From 5163c1eede8e9073e5b6bf1a988ed07d35820343 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 Feb 2015 13:26:01 +0200 Subject: [PATCH 001/411] ASoC: omap: Kconfig: Support for omap5-uevm analog audio The analog audio is supported via omap-abe-twl6040 machine driver on omap5-uevm. Update the Kconfig file to reflect this and select the Palmas clock driver which is providing the 32K clock for audio on omap5-uevm. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a2cd3486ac554a..e7c78b0406b59c 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,17 +100,19 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC + select COMMON_CLK_PALMAS if SOC_OMAP5 help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: - SDP4430/Blaze boards - PandaBoard (4430) - PandaBoardES (4460) + - omap5-uevm (5432) config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" From b6d6c6e95ff0e78f9b8393e6b9f25d5a4341ae1a Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 9 Feb 2015 16:08:25 +0100 Subject: [PATCH 002/411] ASoC: atmel_ssc_dai: Allow more rates When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to the SSC master clock, divided by either of 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? The divider of 2, 3 or 6 is selected based on the Serial Clock Ratio Considerations section from the SSC documentation: The Transmitter and the Receiver can be programmed to operate with the clock signals provided on either the TK or RK pins. This allows the SSC to support many slave-mode data transfers. In this case, the maximum clock speed allowed on the RK pin is: - Peripheral clock divided by 2 if Receiver Frame Synchro is input - Peripheral clock divided by 3 if Receiver Frame Synchro is output In addition, the maximum clock speed allowed on the TK pin is: - Peripheral clock divided by 6 if Transmit Frame Synchro is input - Peripheral clock divided by 2 if Transmit Frame Synchro is output Signed-off-by: Peter Rosin Acked-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 111 ++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 108 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 379ac2a6ab16a0..6b8e648992053f 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,94 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +288,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +296,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +309,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +884,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +903,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d51149526..80b153857a88ac 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); From b6a42670e074da39b5a9f990774359e0733ca9cd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2015 22:37:16 +0100 Subject: [PATCH 003/411] ALSA: seq: Move EXPORT_SYMBOL() after each function ... to follow the standard coding style. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_device.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 0631bdadd12bd6..a752a79a8d3a08 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -133,11 +133,13 @@ void snd_seq_autoload_lock(void) { atomic_inc(&snd_seq_in_init); } +EXPORT_SYMBOL(snd_seq_autoload_lock); void snd_seq_autoload_unlock(void) { atomic_dec(&snd_seq_in_init); } +EXPORT_SYMBOL(snd_seq_autoload_unlock); static void autoload_drivers(void) { @@ -195,10 +197,12 @@ void snd_seq_autoload_init(void) queue_autoload_drivers(); #endif } +EXPORT_SYMBOL(snd_seq_autoload_init); #else #define try_autoload(ops) /* NOP */ #endif + void snd_seq_device_load_drivers(void) { #ifdef CONFIG_MODULES @@ -206,6 +210,7 @@ void snd_seq_device_load_drivers(void) flush_work(&autoload_work); #endif } +EXPORT_SYMBOL(snd_seq_device_load_drivers); /* * register a sequencer device @@ -268,6 +273,7 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, return 0; } +EXPORT_SYMBOL(snd_seq_device_new); /* * free the existing device @@ -326,6 +332,7 @@ static int snd_seq_device_dev_register(struct snd_device *device) unlock_driver(ops); return 0; } +EXPORT_SYMBOL(snd_seq_device_register_driver); /* * disconnect the device @@ -344,6 +351,7 @@ static int snd_seq_device_dev_disconnect(struct snd_device *device) unlock_driver(ops); return 0; } +EXPORT_SYMBOL(snd_seq_device_unregister_driver); /* * register device driver @@ -604,13 +612,3 @@ static void __exit alsa_seq_device_exit(void) module_init(alsa_seq_device_init) module_exit(alsa_seq_device_exit) - -EXPORT_SYMBOL(snd_seq_device_load_drivers); -EXPORT_SYMBOL(snd_seq_device_new); -EXPORT_SYMBOL(snd_seq_device_register_driver); -EXPORT_SYMBOL(snd_seq_device_unregister_driver); -#ifdef CONFIG_MODULES -EXPORT_SYMBOL(snd_seq_autoload_init); -EXPORT_SYMBOL(snd_seq_autoload_lock); -EXPORT_SYMBOL(snd_seq_autoload_unlock); -#endif From 72496edcf85e048b4c5373d518e4f27938d9594e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2015 22:39:51 +0100 Subject: [PATCH 004/411] ALSA: seq: Don't compile snd_seq_device_load_drivers() for built-in Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 4 ++++ sound/core/seq/seq_device.c | 9 +++------ 2 files changed, 7 insertions(+), 6 deletions(-) diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index 2b5f24cc754888..d52433563da225 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -67,7 +67,11 @@ struct snd_seq_dev_ops { /* * prototypes */ +#ifdef CONFIG_MODULES void snd_seq_device_load_drivers(void); +#else +#define snd_seq_device_load_drivers() +#endif int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result); int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize); int snd_seq_device_unregister_driver(char *id); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index a752a79a8d3a08..075a66c0cc6a4c 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -198,19 +198,16 @@ void snd_seq_autoload_init(void) #endif } EXPORT_SYMBOL(snd_seq_autoload_init); -#else -#define try_autoload(ops) /* NOP */ -#endif - void snd_seq_device_load_drivers(void) { -#ifdef CONFIG_MODULES queue_autoload_drivers(); flush_work(&autoload_work); -#endif } EXPORT_SYMBOL(snd_seq_device_load_drivers); +#else +#define try_autoload(ops) /* NOP */ +#endif /* * register a sequencer device From 7c37ae5c625aaa4836466cfaea829a3199dfc571 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 10:51:59 +0100 Subject: [PATCH 005/411] ALSA: seq: Rewrite sequencer device binding with standard bus We've used the old house-made code for binding the sequencer device and driver. This can be far better implemented with the standard bus nowadays. This patch refactors the whole sequencer binding code with the bus /sys/bus/snd_seq. The devices appear as id-card-device on this bus and are bound with the drivers corresponding to the given id like the former implementation. The module autoload is also kept like before. There is no change in API functions by this patch, and almost all transitions are kept inside seq_device.c. The proc file output will change slightly but kept compatible as much as possible. Further integration works will follow in later patches. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 3 + sound/core/seq/seq_device.c | 541 +++++++++++------------------------- 2 files changed, 170 insertions(+), 374 deletions(-) diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index d52433563da225..ea256b82514664 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -43,8 +43,11 @@ struct snd_seq_device { void *private_data; /* private data for the caller */ void (*private_free)(struct snd_seq_device *device); struct list_head list; /* link to next device */ + struct device dev; }; +#define to_seq_dev(_dev) \ + container_of(_dev, struct snd_seq_device, dev) /* driver operators * init_device: diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 075a66c0cc6a4c..d3320ffe43c22f 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -36,6 +36,7 @@ * */ +#include #include #include #include @@ -51,77 +52,57 @@ MODULE_AUTHOR("Takashi Iwai "); MODULE_DESCRIPTION("ALSA sequencer device management"); MODULE_LICENSE("GPL"); -/* driver state */ -#define DRIVER_EMPTY 0 -#define DRIVER_LOADED (1<<0) -#define DRIVER_REQUESTED (1<<1) -#define DRIVER_LOCKED (1<<2) -#define DRIVER_REQUESTING (1<<3) +struct snd_seq_driver { + struct device_driver driver; + char id[ID_LEN]; + int argsize; + struct snd_seq_dev_ops ops; +}; -struct ops_list { - char id[ID_LEN]; /* driver id */ - int driver; /* driver state */ - int used; /* reference counter */ - int argsize; /* argument size */ +#define to_seq_drv(_drv) \ + container_of(_drv, struct snd_seq_driver, driver) - /* operators */ - struct snd_seq_dev_ops ops; +/* + * bus definition + */ +static int snd_seq_bus_match(struct device *dev, struct device_driver *drv) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + struct snd_seq_driver *sdrv = to_seq_drv(drv); - /* registered devices */ - struct list_head dev_list; /* list of devices */ - int num_devices; /* number of associated devices */ - int num_init_devices; /* number of initialized devices */ - struct mutex reg_mutex; + return strcmp(sdrv->id, sdev->id) == 0 && + sdrv->argsize == sdev->argsize; +} - struct list_head list; /* next driver */ +static struct bus_type snd_seq_bus_type = { + .name = "snd_seq", + .match = snd_seq_bus_match, }; - -static LIST_HEAD(opslist); -static int num_ops; -static DEFINE_MUTEX(ops_mutex); +/* + * proc interface -- just for compatibility + */ #ifdef CONFIG_PROC_FS static struct snd_info_entry *info_entry; -#endif -/* - * prototypes - */ -static int snd_seq_device_free(struct snd_seq_device *dev); -static int snd_seq_device_dev_free(struct snd_device *device); -static int snd_seq_device_dev_register(struct snd_device *device); -static int snd_seq_device_dev_disconnect(struct snd_device *device); - -static int init_device(struct snd_seq_device *dev, struct ops_list *ops); -static int free_device(struct snd_seq_device *dev, struct ops_list *ops); -static struct ops_list *find_driver(char *id, int create_if_empty); -static struct ops_list *create_driver(char *id); -static void unlock_driver(struct ops_list *ops); -static void remove_drivers(void); +static int print_dev_info(struct device *dev, void *data) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + struct snd_info_buffer *buffer = data; -/* - * show all drivers and their status - */ + snd_iprintf(buffer, "snd-%s,%s,%d\n", sdev->id, + dev->driver ? "loaded" : "empty", + dev->driver ? 1 : 0); + return 0; +} -#ifdef CONFIG_PROC_FS static void snd_seq_device_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - snd_iprintf(buffer, "snd-%s%s%s%s,%d\n", - ops->id, - ops->driver & DRIVER_LOADED ? ",loaded" : (ops->driver == DRIVER_EMPTY ? ",empty" : ""), - ops->driver & DRIVER_REQUESTED ? ",requested" : "", - ops->driver & DRIVER_LOCKED ? ",locked" : "", - ops->num_devices); - } - mutex_unlock(&ops_mutex); + bus_for_each_dev(&snd_seq_bus_type, NULL, buffer, print_dev_info); } #endif - + /* * load all registered drivers (called from seq_clientmgr.c) */ @@ -141,52 +122,29 @@ void snd_seq_autoload_unlock(void) } EXPORT_SYMBOL(snd_seq_autoload_unlock); -static void autoload_drivers(void) +static int request_seq_drv(struct device *dev, void *data) { - /* avoid reentrance */ - if (atomic_inc_return(&snd_seq_in_init) == 1) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - if ((ops->driver & DRIVER_REQUESTING) && - !(ops->driver & DRIVER_REQUESTED)) { - ops->used++; - mutex_unlock(&ops_mutex); - ops->driver |= DRIVER_REQUESTED; - request_module("snd-%s", ops->id); - mutex_lock(&ops_mutex); - ops->used--; - } - } - mutex_unlock(&ops_mutex); - } - atomic_dec(&snd_seq_in_init); -} + struct snd_seq_device *sdev = to_seq_dev(dev); -static void call_autoload(struct work_struct *work) -{ - autoload_drivers(); + if (!dev->driver) + request_module("snd-%s", sdev->id); + return 0; } -static DECLARE_WORK(autoload_work, call_autoload); - -static void try_autoload(struct ops_list *ops) +static void autoload_drivers(struct work_struct *work) { - if (!ops->driver) { - ops->driver |= DRIVER_REQUESTING; - schedule_work(&autoload_work); - } + /* avoid reentrance */ + if (atomic_inc_return(&snd_seq_in_init) == 1) + bus_for_each_dev(&snd_seq_bus_type, NULL, NULL, + request_seq_drv); + atomic_dec(&snd_seq_in_init); } +static DECLARE_WORK(autoload_work, autoload_drivers); + static void queue_autoload_drivers(void) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) - try_autoload(ops); - mutex_unlock(&ops_mutex); + schedule_work(&autoload_work); } void snd_seq_autoload_init(void) @@ -206,9 +164,50 @@ void snd_seq_device_load_drivers(void) } EXPORT_SYMBOL(snd_seq_device_load_drivers); #else -#define try_autoload(ops) /* NOP */ +#define queue_autoload_drivers() /* NOP */ #endif +/* + * device management + */ +static int snd_seq_device_dev_free(struct snd_device *device) +{ + struct snd_seq_device *dev = device->device_data; + + put_device(&dev->dev); + return 0; +} + +static int snd_seq_device_dev_register(struct snd_device *device) +{ + struct snd_seq_device *dev = device->device_data; + int err; + + err = device_add(&dev->dev); + if (err < 0) + return err; + if (!dev->dev.driver) + queue_autoload_drivers(); + return 0; +} + +static int snd_seq_device_dev_disconnect(struct snd_device *device) +{ + struct snd_seq_device *dev = device->device_data; + + device_del(&dev->dev); + return 0; +} + +static void snd_seq_dev_release(struct device *dev) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + + if (sdev->private_free) + sdev->private_free(sdev); + kfree(sdev); +} + /* * register a sequencer device * card = card info @@ -220,7 +219,6 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result) { struct snd_seq_device *dev; - struct ops_list *ops; int err; static struct snd_device_ops dops = { .dev_free = snd_seq_device_dev_free, @@ -234,15 +232,9 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, if (snd_BUG_ON(!id)) return -EINVAL; - ops = find_driver(id, 1); - if (ops == NULL) - return -ENOMEM; - - dev = kzalloc(sizeof(*dev)*2 + argsize, GFP_KERNEL); - if (dev == NULL) { - unlock_driver(ops); + dev = kzalloc(sizeof(*dev) + argsize, GFP_KERNEL); + if (!dev) return -ENOMEM; - } /* set up device info */ dev->card = card; @@ -251,20 +243,19 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, dev->argsize = argsize; dev->status = SNDRV_SEQ_DEVICE_FREE; - /* add this device to the list */ - mutex_lock(&ops->reg_mutex); - list_add_tail(&dev->list, &ops->dev_list); - ops->num_devices++; - mutex_unlock(&ops->reg_mutex); + device_initialize(&dev->dev); + dev->dev.parent = &card->card_dev; + dev->dev.bus = &snd_seq_bus_type; + dev->dev.release = snd_seq_dev_release; + dev_set_name(&dev->dev, "%s-%d-%d", dev->id, card->number, device); - if ((err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops)) < 0) { - snd_seq_device_free(dev); + /* add this device to the list */ + err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops); + if (err < 0) { + put_device(&dev->dev); return err; } - try_autoload(ops); - unlock_driver(ops); - if (result) *result = dev; @@ -273,82 +264,33 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, EXPORT_SYMBOL(snd_seq_device_new); /* - * free the existing device - */ -static int snd_seq_device_free(struct snd_seq_device *dev) -{ - struct ops_list *ops; - - if (snd_BUG_ON(!dev)) - return -EINVAL; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENXIO; - - /* remove the device from the list */ - mutex_lock(&ops->reg_mutex); - list_del(&dev->list); - ops->num_devices--; - mutex_unlock(&ops->reg_mutex); - - free_device(dev, ops); - if (dev->private_free) - dev->private_free(dev); - kfree(dev); - - unlock_driver(ops); - - return 0; -} - -static int snd_seq_device_dev_free(struct snd_device *device) -{ - struct snd_seq_device *dev = device->device_data; - return snd_seq_device_free(dev); -} - -/* - * register the device + * driver binding - just pass to each driver callback */ -static int snd_seq_device_dev_register(struct snd_device *device) +static int snd_seq_drv_probe(struct device *dev) { - struct snd_seq_device *dev = device->device_data; - struct ops_list *ops; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENOENT; - - /* initialize this device if the corresponding driver was - * already loaded - */ - if (ops->driver & DRIVER_LOADED) - init_device(dev, ops); + struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); + struct snd_seq_device *sdev = to_seq_dev(dev); + int err; - unlock_driver(ops); + err = sdrv->ops.init_device(sdev); + if (err < 0) + return err; + sdev->status = SNDRV_SEQ_DEVICE_REGISTERED; return 0; } -EXPORT_SYMBOL(snd_seq_device_register_driver); -/* - * disconnect the device - */ -static int snd_seq_device_dev_disconnect(struct snd_device *device) +static int snd_seq_drv_remove(struct device *dev) { - struct snd_seq_device *dev = device->device_data; - struct ops_list *ops; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENOENT; - - free_device(dev, ops); + struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); + struct snd_seq_device *sdev = to_seq_dev(dev); + int err; - unlock_driver(ops); + err = sdrv->ops.free_device(sdev); + if (err < 0) + return err; + sdev->status = SNDRV_SEQ_DEVICE_FREE; return 0; } -EXPORT_SYMBOL(snd_seq_device_unregister_driver); /* * register device driver @@ -358,226 +300,66 @@ EXPORT_SYMBOL(snd_seq_device_unregister_driver); int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize) { - struct ops_list *ops; - struct snd_seq_device *dev; + struct snd_seq_driver *sdrv; + int err; if (id == NULL || entry == NULL || entry->init_device == NULL || entry->free_device == NULL) return -EINVAL; - ops = find_driver(id, 1); - if (ops == NULL) + sdrv = kzalloc(sizeof(*sdrv), GFP_KERNEL); + if (!sdrv) return -ENOMEM; - if (ops->driver & DRIVER_LOADED) { - pr_warn("ALSA: seq: driver_register: driver '%s' already exists\n", id); - unlock_driver(ops); - return -EBUSY; - } - - mutex_lock(&ops->reg_mutex); - /* copy driver operators */ - ops->ops = *entry; - ops->driver |= DRIVER_LOADED; - ops->argsize = argsize; - /* initialize existing devices if necessary */ - list_for_each_entry(dev, &ops->dev_list, list) { - init_device(dev, ops); - } - mutex_unlock(&ops->reg_mutex); - - unlock_driver(ops); - - return 0; + sdrv->driver.name = id; + sdrv->driver.bus = &snd_seq_bus_type; + sdrv->driver.probe = snd_seq_drv_probe; + sdrv->driver.remove = snd_seq_drv_remove; + strlcpy(sdrv->id, id, sizeof(sdrv->id)); + sdrv->argsize = argsize; + sdrv->ops = *entry; + + err = driver_register(&sdrv->driver); + if (err < 0) + kfree(sdrv); + return err; } +EXPORT_SYMBOL(snd_seq_device_register_driver); - -/* - * create driver record +/* callback to find a specific driver; data is a pointer to the id string ptr. + * when the id matches, store the driver pointer in return and break the loop. */ -static struct ops_list * create_driver(char *id) +static int find_drv(struct device_driver *drv, void *data) { - struct ops_list *ops; - - ops = kzalloc(sizeof(*ops), GFP_KERNEL); - if (ops == NULL) - return ops; - - /* set up driver entry */ - strlcpy(ops->id, id, sizeof(ops->id)); - mutex_init(&ops->reg_mutex); - /* - * The ->reg_mutex locking rules are per-driver, so we create - * separate per-driver lock classes: - */ - lockdep_set_class(&ops->reg_mutex, (struct lock_class_key *)id); - - ops->driver = DRIVER_EMPTY; - INIT_LIST_HEAD(&ops->dev_list); - /* lock this instance */ - ops->used = 1; - - /* register driver entry */ - mutex_lock(&ops_mutex); - list_add_tail(&ops->list, &opslist); - num_ops++; - mutex_unlock(&ops_mutex); - - return ops; -} + struct snd_seq_driver *sdrv = to_seq_drv(drv); + void **ptr = (void **)data; + if (strcmp(sdrv->id, *ptr)) + return 0; /* id don't match, continue the loop */ + *ptr = sdrv; + return 1; /* break the loop */ +} /* * unregister the specified driver */ int snd_seq_device_unregister_driver(char *id) { - struct ops_list *ops; - struct snd_seq_device *dev; + struct snd_seq_driver *sdrv = (struct snd_seq_driver *)id; - ops = find_driver(id, 0); - if (ops == NULL) + if (!bus_for_each_drv(&snd_seq_bus_type, NULL, &sdrv, find_drv)) return -ENXIO; - if (! (ops->driver & DRIVER_LOADED) || - (ops->driver & DRIVER_LOCKED)) { - pr_err("ALSA: seq: driver_unregister: cannot unload driver '%s': status=%x\n", - id, ops->driver); - unlock_driver(ops); - return -EBUSY; - } - - /* close and release all devices associated with this driver */ - mutex_lock(&ops->reg_mutex); - ops->driver |= DRIVER_LOCKED; /* do not remove this driver recursively */ - list_for_each_entry(dev, &ops->dev_list, list) { - free_device(dev, ops); - } - - ops->driver = 0; - if (ops->num_init_devices > 0) - pr_err("ALSA: seq: free_driver: init_devices > 0!! (%d)\n", - ops->num_init_devices); - mutex_unlock(&ops->reg_mutex); - - unlock_driver(ops); - - /* remove empty driver entries */ - remove_drivers(); - + driver_unregister(&sdrv->driver); + kfree(sdrv); return 0; } - - -/* - * remove empty driver entries - */ -static void remove_drivers(void) -{ - struct list_head *head; - - mutex_lock(&ops_mutex); - head = opslist.next; - while (head != &opslist) { - struct ops_list *ops = list_entry(head, struct ops_list, list); - if (! (ops->driver & DRIVER_LOADED) && - ops->used == 0 && ops->num_devices == 0) { - head = head->next; - list_del(&ops->list); - kfree(ops); - num_ops--; - } else - head = head->next; - } - mutex_unlock(&ops_mutex); -} - -/* - * initialize the device - call init_device operator - */ -static int init_device(struct snd_seq_device *dev, struct ops_list *ops) -{ - if (! (ops->driver & DRIVER_LOADED)) - return 0; /* driver is not loaded yet */ - if (dev->status != SNDRV_SEQ_DEVICE_FREE) - return 0; /* already initialized */ - if (ops->argsize != dev->argsize) { - pr_err("ALSA: seq: incompatible device '%s' for plug-in '%s' (%d %d)\n", - dev->name, ops->id, ops->argsize, dev->argsize); - return -EINVAL; - } - if (ops->ops.init_device(dev) >= 0) { - dev->status = SNDRV_SEQ_DEVICE_REGISTERED; - ops->num_init_devices++; - } else { - pr_err("ALSA: seq: init_device failed: %s: %s\n", - dev->name, dev->id); - } - - return 0; -} - -/* - * release the device - call free_device operator - */ -static int free_device(struct snd_seq_device *dev, struct ops_list *ops) -{ - int result; - - if (! (ops->driver & DRIVER_LOADED)) - return 0; /* driver is not loaded yet */ - if (dev->status != SNDRV_SEQ_DEVICE_REGISTERED) - return 0; /* not registered */ - if (ops->argsize != dev->argsize) { - pr_err("ALSA: seq: incompatible device '%s' for plug-in '%s' (%d %d)\n", - dev->name, ops->id, ops->argsize, dev->argsize); - return -EINVAL; - } - if ((result = ops->ops.free_device(dev)) >= 0 || result == -ENXIO) { - dev->status = SNDRV_SEQ_DEVICE_FREE; - dev->driver_data = NULL; - ops->num_init_devices--; - } else { - pr_err("ALSA: seq: free_device failed: %s: %s\n", - dev->name, dev->id); - } - - return 0; -} - -/* - * find the matching driver with given id - */ -static struct ops_list * find_driver(char *id, int create_if_empty) -{ - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - if (strcmp(ops->id, id) == 0) { - ops->used++; - mutex_unlock(&ops_mutex); - return ops; - } - } - mutex_unlock(&ops_mutex); - if (create_if_empty) - return create_driver(id); - return NULL; -} - -static void unlock_driver(struct ops_list *ops) -{ - mutex_lock(&ops_mutex); - ops->used--; - mutex_unlock(&ops_mutex); -} - +EXPORT_SYMBOL(snd_seq_device_unregister_driver); /* * module part */ -static int __init alsa_seq_device_init(void) +static int __init seq_dev_proc_init(void) { #ifdef CONFIG_PROC_FS info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers", @@ -594,17 +376,28 @@ static int __init alsa_seq_device_init(void) return 0; } +static int __init alsa_seq_device_init(void) +{ + int err; + + err = bus_register(&snd_seq_bus_type); + if (err < 0) + return err; + err = seq_dev_proc_init(); + if (err < 0) + bus_unregister(&snd_seq_bus_type); + return err; +} + static void __exit alsa_seq_device_exit(void) { #ifdef CONFIG_MODULES cancel_work_sync(&autoload_work); #endif - remove_drivers(); #ifdef CONFIG_PROC_FS snd_info_free_entry(info_entry); #endif - if (num_ops) - pr_err("ALSA: seq: drivers not released (%d)\n", num_ops); + bus_unregister(&snd_seq_bus_type); } module_init(alsa_seq_device_init) From af03c243a1f014145dae34368fe975b2f08ed964 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 13:40:50 +0100 Subject: [PATCH 006/411] ALSA: seq: Clean up device and driver structs Use const string pointer instead of copying the id string to each object. Also drop the status and list fields of snd_seq_device struct that are no longer used. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 18 ++++++------------ sound/core/seq/seq_device.c | 31 ++++++++++--------------------- 2 files changed, 16 insertions(+), 33 deletions(-) diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index ea256b82514664..b13cd2930d32b0 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -25,24 +25,16 @@ * registered device information */ -#define ID_LEN 32 - -/* status flag */ -#define SNDRV_SEQ_DEVICE_FREE 0 -#define SNDRV_SEQ_DEVICE_REGISTERED 1 - struct snd_seq_device { /* device info */ struct snd_card *card; /* sound card */ int device; /* device number */ - char id[ID_LEN]; /* driver id */ + const char *id; /* driver id */ char name[80]; /* device name */ int argsize; /* size of the argument */ void *driver_data; /* private data for driver */ - int status; /* flag - read only */ void *private_data; /* private data for the caller */ void (*private_free)(struct snd_seq_device *device); - struct list_head list; /* link to next device */ struct device dev; }; @@ -75,9 +67,11 @@ void snd_seq_device_load_drivers(void); #else #define snd_seq_device_load_drivers() #endif -int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result); -int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize); -int snd_seq_device_unregister_driver(char *id); +int snd_seq_device_new(struct snd_card *card, int device, const char *id, + int argsize, struct snd_seq_device **result); +int snd_seq_device_register_driver(const char *id, + struct snd_seq_dev_ops *entry, int argsize); +int snd_seq_device_unregister_driver(const char *id); #define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device)) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index d3320ffe43c22f..49daf6e3a387a7 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -54,7 +54,7 @@ MODULE_LICENSE("GPL"); struct snd_seq_driver { struct device_driver driver; - char id[ID_LEN]; + const char *id; int argsize; struct snd_seq_dev_ops ops; }; @@ -215,8 +215,8 @@ static void snd_seq_dev_release(struct device *dev) * id = id of driver * result = return pointer (NULL allowed if unnecessary) */ -int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, - struct snd_seq_device **result) +int snd_seq_device_new(struct snd_card *card, int device, const char *id, + int argsize, struct snd_seq_device **result) { struct snd_seq_device *dev; int err; @@ -239,9 +239,8 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, /* set up device info */ dev->card = card; dev->device = device; - strlcpy(dev->id, id, sizeof(dev->id)); + dev->id = id; dev->argsize = argsize; - dev->status = SNDRV_SEQ_DEVICE_FREE; device_initialize(&dev->dev); dev->dev.parent = &card->card_dev; @@ -270,26 +269,16 @@ static int snd_seq_drv_probe(struct device *dev) { struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); struct snd_seq_device *sdev = to_seq_dev(dev); - int err; - err = sdrv->ops.init_device(sdev); - if (err < 0) - return err; - sdev->status = SNDRV_SEQ_DEVICE_REGISTERED; - return 0; + return sdrv->ops.init_device(sdev); } static int snd_seq_drv_remove(struct device *dev) { struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); struct snd_seq_device *sdev = to_seq_dev(dev); - int err; - err = sdrv->ops.free_device(sdev); - if (err < 0) - return err; - sdev->status = SNDRV_SEQ_DEVICE_FREE; - return 0; + return sdrv->ops.free_device(sdev); } /* @@ -297,8 +286,8 @@ static int snd_seq_drv_remove(struct device *dev) * id = driver id * entry = driver operators - duplicated to each instance */ -int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, - int argsize) +int snd_seq_device_register_driver(const char *id, + struct snd_seq_dev_ops *entry, int argsize) { struct snd_seq_driver *sdrv; int err; @@ -315,7 +304,7 @@ int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, sdrv->driver.bus = &snd_seq_bus_type; sdrv->driver.probe = snd_seq_drv_probe; sdrv->driver.remove = snd_seq_drv_remove; - strlcpy(sdrv->id, id, sizeof(sdrv->id)); + sdrv->id = id; sdrv->argsize = argsize; sdrv->ops = *entry; @@ -343,7 +332,7 @@ static int find_drv(struct device_driver *drv, void *data) /* * unregister the specified driver */ -int snd_seq_device_unregister_driver(char *id) +int snd_seq_device_unregister_driver(const char *id) { struct snd_seq_driver *sdrv = (struct snd_seq_driver *)id; From 056622053b8ae02978678ac1321b5bd956e7c812 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 13:43:22 +0100 Subject: [PATCH 007/411] ALSA: seq: Define driver object in each driver This patch moves the driver object initialization and allocation to each driver's module init/exit code like other normal drivers. The snd_seq_driver struct is now published in seq_device.h, and each driver is responsible to define it with proper driver attributes (name, probe and remove) with snd_seq_driver specific attributes as id and argsize fields. The helper functions snd_seq_driver_register(), snd_seq_driver_unregister() and module_snd_seq_driver() are used for simplifying codes. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 27 ++++++--- sound/core/seq/oss/seq_oss.c | 20 ++++--- sound/core/seq/oss/seq_oss_synth.c | 6 +- sound/core/seq/oss/seq_oss_synth.h | 4 +- sound/core/seq/seq_device.c | 93 ++++-------------------------- sound/core/seq/seq_midi.c | 28 +++++---- sound/drivers/opl3/opl3_seq.c | 34 +++++------ sound/drivers/opl4/opl4_seq.c | 33 +++++------ sound/isa/sb/emu8000_synth.c | 35 +++++------ sound/pci/emu10k1/emu10k1_synth.c | 35 +++++------ 10 files changed, 124 insertions(+), 191 deletions(-) diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index b13cd2930d32b0..ddc0d504cf39b3 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -41,8 +41,10 @@ struct snd_seq_device { #define to_seq_dev(_dev) \ container_of(_dev, struct snd_seq_device, dev) +/* sequencer driver */ + /* driver operators - * init_device: + * probe: * Initialize the device with given parameters. * Typically, * 1. call snd_hwdep_new @@ -50,15 +52,19 @@ struct snd_seq_device { * 3. call snd_hwdep_register * 4. store the instance to dev->driver_data pointer. * - * free_device: + * remove: * Release the private data. * Typically, call snd_device_free(dev->card, dev->driver_data) */ -struct snd_seq_dev_ops { - int (*init_device)(struct snd_seq_device *dev); - int (*free_device)(struct snd_seq_device *dev); +struct snd_seq_driver { + struct device_driver driver; + char *id; + int argsize; }; +#define to_seq_drv(_drv) \ + container_of(_drv, struct snd_seq_driver, driver) + /* * prototypes */ @@ -69,12 +75,17 @@ void snd_seq_device_load_drivers(void); #endif int snd_seq_device_new(struct snd_card *card, int device, const char *id, int argsize, struct snd_seq_device **result); -int snd_seq_device_register_driver(const char *id, - struct snd_seq_dev_ops *entry, int argsize); -int snd_seq_device_unregister_driver(const char *id); #define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device)) +int __must_check __snd_seq_driver_register(struct snd_seq_driver *drv, + struct module *mod); +#define snd_seq_driver_register(drv) \ + __snd_seq_driver_register(drv, THIS_MODULE) +void snd_seq_driver_unregister(struct snd_seq_driver *drv); + +#define module_snd_seq_driver(drv) \ + module_driver(drv, snd_seq_driver_register, snd_seq_driver_unregister) /* * id strings for generic devices diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 16d42679e43fc2..ae1814aa767e9e 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -65,13 +65,19 @@ static unsigned int odev_poll(struct file *file, poll_table * wait); * module interface */ +static struct snd_seq_driver seq_oss_synth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_seq_oss_synth_probe, + .remove = snd_seq_oss_synth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OSS, + .argsize = sizeof(struct snd_seq_oss_reg), +}; + static int __init alsa_seq_oss_init(void) { int rc; - static struct snd_seq_dev_ops ops = { - snd_seq_oss_synth_register, - snd_seq_oss_synth_unregister, - }; snd_seq_autoload_lock(); if ((rc = register_device()) < 0) @@ -86,8 +92,8 @@ static int __init alsa_seq_oss_init(void) goto error; } - if ((rc = snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OSS, &ops, - sizeof(struct snd_seq_oss_reg))) < 0) { + rc = snd_seq_driver_register(&seq_oss_synth_driver); + if (rc < 0) { snd_seq_oss_delete_client(); unregister_proc(); unregister_device(); @@ -104,7 +110,7 @@ static int __init alsa_seq_oss_init(void) static void __exit alsa_seq_oss_exit(void) { - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OSS); + snd_seq_driver_unregister(&seq_oss_synth_driver); snd_seq_oss_delete_client(); unregister_proc(); unregister_device(); diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 701feb71b700c7..835edc80f9186f 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -98,8 +98,9 @@ snd_seq_oss_synth_init(void) * registration of the synth device */ int -snd_seq_oss_synth_register(struct snd_seq_device *dev) +snd_seq_oss_synth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); int i; struct seq_oss_synth *rec; struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev); @@ -149,8 +150,9 @@ snd_seq_oss_synth_register(struct snd_seq_device *dev) int -snd_seq_oss_synth_unregister(struct snd_seq_device *dev) +snd_seq_oss_synth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); int index; struct seq_oss_synth *rec = dev->driver_data; unsigned long flags; diff --git a/sound/core/seq/oss/seq_oss_synth.h b/sound/core/seq/oss/seq_oss_synth.h index dbdfcbb80eaa2b..74ac55f166b651 100644 --- a/sound/core/seq/oss/seq_oss_synth.h +++ b/sound/core/seq/oss/seq_oss_synth.h @@ -28,8 +28,8 @@ #include void snd_seq_oss_synth_init(void); -int snd_seq_oss_synth_register(struct snd_seq_device *dev); -int snd_seq_oss_synth_unregister(struct snd_seq_device *dev); +int snd_seq_oss_synth_probe(struct device *dev); +int snd_seq_oss_synth_remove(struct device *dev); void snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp); void snd_seq_oss_synth_setup_midi(struct seq_oss_devinfo *dp); void snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 49daf6e3a387a7..48b20f009598ac 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -52,16 +52,6 @@ MODULE_AUTHOR("Takashi Iwai "); MODULE_DESCRIPTION("ALSA sequencer device management"); MODULE_LICENSE("GPL"); -struct snd_seq_driver { - struct device_driver driver; - const char *id; - int argsize; - struct snd_seq_dev_ops ops; -}; - -#define to_seq_drv(_drv) \ - container_of(_drv, struct snd_seq_driver, driver) - /* * bus definition */ @@ -263,86 +253,23 @@ int snd_seq_device_new(struct snd_card *card, int device, const char *id, EXPORT_SYMBOL(snd_seq_device_new); /* - * driver binding - just pass to each driver callback + * driver registration */ -static int snd_seq_drv_probe(struct device *dev) +int __snd_seq_driver_register(struct snd_seq_driver *drv, struct module *mod) { - struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); - struct snd_seq_device *sdev = to_seq_dev(dev); - - return sdrv->ops.init_device(sdev); -} - -static int snd_seq_drv_remove(struct device *dev) -{ - struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); - struct snd_seq_device *sdev = to_seq_dev(dev); - - return sdrv->ops.free_device(sdev); -} - -/* - * register device driver - * id = driver id - * entry = driver operators - duplicated to each instance - */ -int snd_seq_device_register_driver(const char *id, - struct snd_seq_dev_ops *entry, int argsize) -{ - struct snd_seq_driver *sdrv; - int err; - - if (id == NULL || entry == NULL || - entry->init_device == NULL || entry->free_device == NULL) + if (WARN_ON(!drv->driver.name || !drv->id)) return -EINVAL; - - sdrv = kzalloc(sizeof(*sdrv), GFP_KERNEL); - if (!sdrv) - return -ENOMEM; - - sdrv->driver.name = id; - sdrv->driver.bus = &snd_seq_bus_type; - sdrv->driver.probe = snd_seq_drv_probe; - sdrv->driver.remove = snd_seq_drv_remove; - sdrv->id = id; - sdrv->argsize = argsize; - sdrv->ops = *entry; - - err = driver_register(&sdrv->driver); - if (err < 0) - kfree(sdrv); - return err; -} -EXPORT_SYMBOL(snd_seq_device_register_driver); - -/* callback to find a specific driver; data is a pointer to the id string ptr. - * when the id matches, store the driver pointer in return and break the loop. - */ -static int find_drv(struct device_driver *drv, void *data) -{ - struct snd_seq_driver *sdrv = to_seq_drv(drv); - void **ptr = (void **)data; - - if (strcmp(sdrv->id, *ptr)) - return 0; /* id don't match, continue the loop */ - *ptr = sdrv; - return 1; /* break the loop */ + drv->driver.bus = &snd_seq_bus_type; + drv->driver.owner = mod; + return driver_register(&drv->driver); } +EXPORT_SYMBOL_GPL(__snd_seq_driver_register); -/* - * unregister the specified driver - */ -int snd_seq_device_unregister_driver(const char *id) +void snd_seq_driver_unregister(struct snd_seq_driver *drv) { - struct snd_seq_driver *sdrv = (struct snd_seq_driver *)id; - - if (!bus_for_each_drv(&snd_seq_bus_type, NULL, &sdrv, find_drv)) - return -ENXIO; - driver_unregister(&sdrv->driver); - kfree(sdrv); - return 0; + driver_unregister(&drv->driver); } -EXPORT_SYMBOL(snd_seq_device_unregister_driver); +EXPORT_SYMBOL_GPL(snd_seq_driver_unregister); /* * module part diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 68fec776da2619..79c73119cedc5c 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -273,8 +273,9 @@ static void snd_seq_midisynth_delete(struct seq_midisynth *msynth) /* register new midi synth port */ static int -snd_seq_midisynth_register_port(struct snd_seq_device *dev) +snd_seq_midisynth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct seq_midisynth_client *client; struct seq_midisynth *msynth, *ms; struct snd_seq_port_info *port; @@ -427,8 +428,9 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) /* release midi synth port */ static int -snd_seq_midisynth_unregister_port(struct snd_seq_device *dev) +snd_seq_midisynth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct seq_midisynth_client *client; struct seq_midisynth *msynth; struct snd_card *card = dev->card; @@ -457,23 +459,29 @@ snd_seq_midisynth_unregister_port(struct snd_seq_device *dev) return 0; } +static struct snd_seq_driver seq_midisynth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_seq_midisynth_probe, + .remove = snd_seq_midisynth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_MIDISYNTH, + .argsize = 0, +}; static int __init alsa_seq_midi_init(void) { - static struct snd_seq_dev_ops ops = { - snd_seq_midisynth_register_port, - snd_seq_midisynth_unregister_port, - }; - memset(&synths, 0, sizeof(synths)); + int err; + snd_seq_autoload_lock(); - snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH, &ops, 0); + err = snd_seq_driver_register(&seq_midisynth_driver); snd_seq_autoload_unlock(); - return 0; + return err; } static void __exit alsa_seq_midi_exit(void) { - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH); + snd_seq_driver_unregister(&seq_midisynth_driver); } module_init(alsa_seq_midi_init) diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index a9f618e06a22bc..fdae5d7f421ff7 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -216,8 +216,9 @@ static int snd_opl3_synth_create_port(struct snd_opl3 * opl3) /* ------------------------------ */ -static int snd_opl3_seq_new_device(struct snd_seq_device *dev) +static int snd_opl3_seq_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl3 *opl3; int client, err; char name[32]; @@ -257,8 +258,9 @@ static int snd_opl3_seq_new_device(struct snd_seq_device *dev) return 0; } -static int snd_opl3_seq_delete_device(struct snd_seq_device *dev) +static int snd_opl3_seq_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl3 *opl3; opl3 = *(struct snd_opl3 **)SNDRV_SEQ_DEVICE_ARGPTR(dev); @@ -275,22 +277,14 @@ static int snd_opl3_seq_delete_device(struct snd_seq_device *dev) return 0; } -static int __init alsa_opl3_seq_init(void) -{ - static struct snd_seq_dev_ops ops = - { - snd_opl3_seq_new_device, - snd_opl3_seq_delete_device - }; - - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL3, &ops, - sizeof(struct snd_opl3 *)); -} - -static void __exit alsa_opl3_seq_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL3); -} +static struct snd_seq_driver opl3_seq_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_opl3_seq_probe, + .remove = snd_opl3_seq_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OPL3, + .argsize = sizeof(struct snd_opl3 *), +}; -module_init(alsa_opl3_seq_init) -module_exit(alsa_opl3_seq_exit) +module_snd_seq_driver(opl3_seq_driver); diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c index 99197699c55a63..03d6202f482931 100644 --- a/sound/drivers/opl4/opl4_seq.c +++ b/sound/drivers/opl4/opl4_seq.c @@ -124,8 +124,9 @@ static void snd_opl4_seq_free_port(void *private_data) snd_midi_channel_free_set(opl4->chset); } -static int snd_opl4_seq_new_device(struct snd_seq_device *dev) +static int snd_opl4_seq_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl4 *opl4; int client; struct snd_seq_port_callback pcallbacks; @@ -180,8 +181,9 @@ static int snd_opl4_seq_new_device(struct snd_seq_device *dev) return 0; } -static int snd_opl4_seq_delete_device(struct snd_seq_device *dev) +static int snd_opl4_seq_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl4 *opl4; opl4 = *(struct snd_opl4 **)SNDRV_SEQ_DEVICE_ARGPTR(dev); @@ -195,21 +197,14 @@ static int snd_opl4_seq_delete_device(struct snd_seq_device *dev) return 0; } -static int __init alsa_opl4_synth_init(void) -{ - static struct snd_seq_dev_ops ops = { - snd_opl4_seq_new_device, - snd_opl4_seq_delete_device - }; - - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL4, &ops, - sizeof(struct snd_opl4 *)); -} - -static void __exit alsa_opl4_synth_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL4); -} +static struct snd_seq_driver opl4_seq_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_opl4_seq_probe, + .remove = snd_opl4_seq_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OPL4, + .argsize = sizeof(struct snd_opl4 *), +}; -module_init(alsa_opl4_synth_init) -module_exit(alsa_opl4_synth_exit) +module_snd_seq_driver(opl4_seq_driver); diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c index 72332dfada9a3f..4aa719cad33191 100644 --- a/sound/isa/sb/emu8000_synth.c +++ b/sound/isa/sb/emu8000_synth.c @@ -34,8 +34,9 @@ MODULE_LICENSE("GPL"); /* * create a new hardware dependent device for Emu8000 */ -static int snd_emu8000_new_device(struct snd_seq_device *dev) +static int snd_emu8000_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emu8000 *hw; struct snd_emux *emu; @@ -93,8 +94,9 @@ static int snd_emu8000_new_device(struct snd_seq_device *dev) /* * free all resources */ -static int snd_emu8000_delete_device(struct snd_seq_device *dev) +static int snd_emu8000_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emu8000 *hw; if (dev->driver_data == NULL) @@ -114,21 +116,14 @@ static int snd_emu8000_delete_device(struct snd_seq_device *dev) * INIT part */ -static int __init alsa_emu8000_init(void) -{ - - static struct snd_seq_dev_ops ops = { - snd_emu8000_new_device, - snd_emu8000_delete_device, - }; - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU8000, &ops, - sizeof(struct snd_emu8000*)); -} - -static void __exit alsa_emu8000_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU8000); -} - -module_init(alsa_emu8000_init) -module_exit(alsa_emu8000_exit) +static struct snd_seq_driver emu8000_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_emu8000_probe, + .remove = snd_emu8000_remove, + }, + .id = SNDRV_SEQ_DEV_ID_EMU8000, + .argsize = sizeof(struct snd_emu8000 *), +}; + +module_snd_seq_driver(emu8000_driver); diff --git a/sound/pci/emu10k1/emu10k1_synth.c b/sound/pci/emu10k1/emu10k1_synth.c index 4c41c903a840b5..5457d5613f6b46 100644 --- a/sound/pci/emu10k1/emu10k1_synth.c +++ b/sound/pci/emu10k1/emu10k1_synth.c @@ -29,8 +29,9 @@ MODULE_LICENSE("GPL"); /* * create a new hardware dependent device for Emu10k1 */ -static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) +static int snd_emu10k1_synth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emux *emux; struct snd_emu10k1 *hw; struct snd_emu10k1_synth_arg *arg; @@ -79,8 +80,9 @@ static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) return 0; } -static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev) +static int snd_emu10k1_synth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emux *emux; struct snd_emu10k1 *hw; unsigned long flags; @@ -104,21 +106,14 @@ static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev) * INIT part */ -static int __init alsa_emu10k1_synth_init(void) -{ - - static struct snd_seq_dev_ops ops = { - snd_emu10k1_synth_new_device, - snd_emu10k1_synth_delete_device, - }; - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH, &ops, - sizeof(struct snd_emu10k1_synth_arg)); -} - -static void __exit alsa_emu10k1_synth_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH); -} - -module_init(alsa_emu10k1_synth_init) -module_exit(alsa_emu10k1_synth_exit) +static struct snd_seq_driver emu10k1_synth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_emu10k1_synth_probe, + .remove = snd_emu10k1_synth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH, + .argsize = sizeof(struct snd_emu10k1_synth_arg), +}; + +module_snd_seq_driver(emu10k1_synth_driver); From 54a721abd7953a58e5479065c0cfdd8679d785c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 14:20:24 +0100 Subject: [PATCH 008/411] ALSA: seq: Drop snd_seq_autoload_lock() and _unlock() The autoload lock became already superfluous due to the recent rework of autoload code. Let's drop them now. This allows us to simplify a few codes nicely. Signed-off-by: Takashi Iwai --- include/sound/seq_kernel.h | 6 +----- sound/core/seq/oss/seq_oss.c | 2 -- sound/core/seq/seq_device.c | 19 +++++++------------ sound/core/seq/seq_dummy.c | 6 +----- sound/core/seq/seq_midi.c | 18 +----------------- 5 files changed, 10 insertions(+), 41 deletions(-) diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index 18a2ac58b88f99..feb58d45556064 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -99,13 +99,9 @@ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp, int snd_seq_event_port_detach(int client, int port); #ifdef CONFIG_MODULES -void snd_seq_autoload_lock(void); -void snd_seq_autoload_unlock(void); void snd_seq_autoload_init(void); -#define snd_seq_autoload_exit() snd_seq_autoload_lock() +void snd_seq_autoload_exit(void); #else -#define snd_seq_autoload_lock() -#define snd_seq_autoload_unlock() #define snd_seq_autoload_init() #define snd_seq_autoload_exit() #endif diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index ae1814aa767e9e..72873a46afeb39 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -79,7 +79,6 @@ static int __init alsa_seq_oss_init(void) { int rc; - snd_seq_autoload_lock(); if ((rc = register_device()) < 0) goto error; if ((rc = register_proc()) < 0) { @@ -104,7 +103,6 @@ static int __init alsa_seq_oss_init(void) snd_seq_oss_synth_init(); error: - snd_seq_autoload_unlock(); return rc; } diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 48b20f009598ac..355b34269bd19f 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -98,19 +98,8 @@ static void snd_seq_device_info(struct snd_info_entry *entry, */ #ifdef CONFIG_MODULES -/* avoid auto-loading during module_init() */ +/* flag to block auto-loading */ static atomic_t snd_seq_in_init = ATOMIC_INIT(1); /* blocked as default */ -void snd_seq_autoload_lock(void) -{ - atomic_inc(&snd_seq_in_init); -} -EXPORT_SYMBOL(snd_seq_autoload_lock); - -void snd_seq_autoload_unlock(void) -{ - atomic_dec(&snd_seq_in_init); -} -EXPORT_SYMBOL(snd_seq_autoload_unlock); static int request_seq_drv(struct device *dev, void *data) { @@ -147,6 +136,12 @@ void snd_seq_autoload_init(void) } EXPORT_SYMBOL(snd_seq_autoload_init); +void snd_seq_autoload_exit(void) +{ + atomic_inc(&snd_seq_in_init); +} +EXPORT_SYMBOL(snd_seq_autoload_exit); + void snd_seq_device_load_drivers(void) { queue_autoload_drivers(); diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index 5d905d90d504c0..d3a2ec4f0561e7 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -214,11 +214,7 @@ delete_client(void) static int __init alsa_seq_dummy_init(void) { - int err; - snd_seq_autoload_lock(); - err = register_client(); - snd_seq_autoload_unlock(); - return err; + return register_client(); } static void __exit alsa_seq_dummy_exit(void) diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 79c73119cedc5c..5dd0ee2583592e 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -469,20 +469,4 @@ static struct snd_seq_driver seq_midisynth_driver = { .argsize = 0, }; -static int __init alsa_seq_midi_init(void) -{ - int err; - - snd_seq_autoload_lock(); - err = snd_seq_driver_register(&seq_midisynth_driver); - snd_seq_autoload_unlock(); - return err; -} - -static void __exit alsa_seq_midi_exit(void) -{ - snd_seq_driver_unregister(&seq_midisynth_driver); -} - -module_init(alsa_seq_midi_init) -module_exit(alsa_seq_midi_exit) +module_snd_seq_driver(seq_midisynth_driver); From 226e2f1b0bb4a5f724dd119c1eeb8b8e89e87fab Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 12 Feb 2015 16:41:26 +0200 Subject: [PATCH 009/411] ASoC: davinci-mcasp: Add support for CBS_CFM mode Support for setups where codec is bitclock slave and frame master. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155a50116f..031c1fb44ae72b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -441,6 +441,18 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); mcasp->bclk_master = 1; break; + case SND_SOC_DAIFMT_CBS_CFM: + /* codec is clock slave and frame master */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; + break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); From 38ebb7034970efe5c7419267e499295e5893b565 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2015 14:45:33 +0100 Subject: [PATCH 010/411] ALSA: Consolidate snd_find_free_minor() A really small cleanup to consolidate snd_find_free_minor() and snd_kernel_minor() so that we can get rid of one more ifdef. Signed-off-by: Takashi Iwai --- sound/core/sound.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/core/sound.c b/sound/core/sound.c index 185cec01ee258c..5fc93d00572a5f 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -186,7 +186,7 @@ static const struct file_operations snd_fops = }; #ifdef CONFIG_SND_DYNAMIC_MINORS -static int snd_find_free_minor(int type) +static int snd_find_free_minor(int type, struct snd_card *card, int dev) { int minor; @@ -209,7 +209,7 @@ static int snd_find_free_minor(int type) return -EBUSY; } #else -static int snd_kernel_minor(int type, struct snd_card *card, int dev) +static int snd_find_free_minor(int type, struct snd_card *card, int dev) { int minor; @@ -237,6 +237,8 @@ static int snd_kernel_minor(int type, struct snd_card *card, int dev) } if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OS_MINORS)) return -EINVAL; + if (snd_minors[minor]) + return -EBUSY; return minor; } #endif @@ -276,13 +278,7 @@ int snd_register_device(int type, struct snd_card *card, int dev, preg->private_data = private_data; preg->card_ptr = card; mutex_lock(&sound_mutex); -#ifdef CONFIG_SND_DYNAMIC_MINORS - minor = snd_find_free_minor(type); -#else - minor = snd_kernel_minor(type, card, dev); - if (minor >= 0 && snd_minors[minor]) - minor = -EBUSY; -#endif + minor = snd_find_free_minor(type, card, dev); if (minor < 0) { err = minor; goto error; From 5ecc5dc720307d3fb0167e2b14f50e97dd9a2233 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Tue, 17 Feb 2015 00:05:04 +0100 Subject: [PATCH 011/411] ALSA: hdspm - DRY cleanup in .open callbacks This commit removes code duplication between snd_hdspm_{capture,playback}_open. No semantic changes intended, this is purely cosmetic. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 100 +++++++++----------------------------- 1 file changed, 24 insertions(+), 76 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ca67f896d11757..51e984170b3d11 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6043,23 +6043,30 @@ hdspm_hw_constraints_aes32_sample_rates = { .mask = 0 }; -static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) +static int snd_hdspm_open(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); spin_lock_irq(&hdspm->lock); - snd_pcm_set_sync(substream); + runtime->hw = (playback) ? snd_hdspm_playback_subinfo : + snd_hdspm_capture_subinfo; + if (playback) { + if (hdspm->capture_substream == NULL) + hdspm_stop_audio(hdspm); - runtime->hw = snd_hdspm_playback_subinfo; - - if (hdspm->capture_substream == NULL) - hdspm_stop_audio(hdspm); + hdspm->playback_pid = current->pid; + hdspm->playback_substream = substream; + } else { + if (hdspm->playback_substream == NULL) + hdspm_stop_audio(hdspm); - hdspm->playback_pid = current->pid; - hdspm->playback_substream = substream; + hdspm->capture_pid = current->pid; + hdspm->capture_substream = substream; + } spin_unlock_irq(&hdspm->lock); @@ -6094,16 +6101,20 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) &hdspm_hw_constraints_aes32_sample_rates); } else { snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_hdspm_hw_rule_rate_out_channels, hdspm, + (playback ? + snd_hdspm_hw_rule_rate_out_channels : + snd_hdspm_hw_rule_rate_in_channels), hdspm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); } snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_out_channels, hdspm, + (playback ? snd_hdspm_hw_rule_out_channels : + snd_hdspm_hw_rule_in_channels), hdspm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_out_channels_rate, hdspm, + (playback ? snd_hdspm_hw_rule_out_channels_rate : + snd_hdspm_hw_rule_in_channels_rate), hdspm, SNDRV_PCM_HW_PARAM_RATE, -1); return 0; @@ -6123,69 +6134,6 @@ static int snd_hdspm_playback_release(struct snd_pcm_substream *substream) return 0; } - -static int snd_hdspm_capture_open(struct snd_pcm_substream *substream) -{ - struct hdspm *hdspm = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - spin_lock_irq(&hdspm->lock); - snd_pcm_set_sync(substream); - runtime->hw = snd_hdspm_capture_subinfo; - - if (hdspm->playback_substream == NULL) - hdspm_stop_audio(hdspm); - - hdspm->capture_pid = current->pid; - hdspm->capture_substream = substream; - - spin_unlock_irq(&hdspm->lock); - - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); - - switch (hdspm->io_type) { - case AIO: - case RayDAT: - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - 32, 4096); - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - 16384, 16384); - break; - - default: - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - 64, 8192); - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIODS, - 2, 2); - break; - } - - if (AES32 == hdspm->io_type) { - runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &hdspm_hw_constraints_aes32_sample_rates); - } else { - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_hdspm_hw_rule_rate_in_channels, hdspm, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - } - - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_in_channels, hdspm, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_in_channels_rate, hdspm, - SNDRV_PCM_HW_PARAM_RATE, -1); - - return 0; -} - static int snd_hdspm_capture_release(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); @@ -6414,7 +6362,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, } static struct snd_pcm_ops snd_hdspm_playback_ops = { - .open = snd_hdspm_playback_open, + .open = snd_hdspm_open, .close = snd_hdspm_playback_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, @@ -6426,7 +6374,7 @@ static struct snd_pcm_ops snd_hdspm_playback_ops = { }; static struct snd_pcm_ops snd_hdspm_capture_ops = { - .open = snd_hdspm_capture_open, + .open = snd_hdspm_open, .close = snd_hdspm_capture_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, From 8b73b867294364ae3007affdf8cfd28f022b158c Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Tue, 17 Feb 2015 00:05:05 +0100 Subject: [PATCH 012/411] ALSA: hdspm - DRY cleanup in .release callback This commit removes code duplication between snd_hdspm_{capture,playback}_release. No semantic changes intended, this is purely cosmetic. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 51e984170b3d11..4e1cfb91a8d834 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6120,33 +6120,26 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream) return 0; } -static int snd_hdspm_playback_release(struct snd_pcm_substream *substream) +static int snd_hdspm_release(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); spin_lock_irq(&hdspm->lock); - hdspm->playback_pid = -1; - hdspm->playback_substream = NULL; + if (playback) { + hdspm->playback_pid = -1; + hdspm->playback_substream = NULL; + } else { + hdspm->capture_pid = -1; + hdspm->capture_substream = NULL; + } spin_unlock_irq(&hdspm->lock); return 0; } -static int snd_hdspm_capture_release(struct snd_pcm_substream *substream) -{ - struct hdspm *hdspm = snd_pcm_substream_chip(substream); - - spin_lock_irq(&hdspm->lock); - - hdspm->capture_pid = -1; - hdspm->capture_substream = NULL; - - spin_unlock_irq(&hdspm->lock); - return 0; -} - static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file) { /* we have nothing to initialize but the call is required */ @@ -6363,7 +6356,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, static struct snd_pcm_ops snd_hdspm_playback_ops = { .open = snd_hdspm_open, - .close = snd_hdspm_playback_release, + .close = snd_hdspm_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, .hw_free = snd_hdspm_hw_free, @@ -6375,7 +6368,7 @@ static struct snd_pcm_ops snd_hdspm_playback_ops = { static struct snd_pcm_ops snd_hdspm_capture_ops = { .open = snd_hdspm_open, - .close = snd_hdspm_capture_release, + .close = snd_hdspm_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, .hw_free = snd_hdspm_hw_free, From 0c8d948565490d2a2db9d9a5aec388342c7d38ce Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Tue, 17 Feb 2015 00:05:06 +0100 Subject: [PATCH 013/411] ALSA: hdspm - DRY cleanup of snd_pcm_ops This commit removes code duplication between snd_hdspm_{capture,playback}_ops. No semantic changes intended, this is purely cosmetic. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 18 +++--------------- 1 file changed, 3 insertions(+), 15 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 4e1cfb91a8d834..cb666c73712d15 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6354,19 +6354,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, return 0; } -static struct snd_pcm_ops snd_hdspm_playback_ops = { - .open = snd_hdspm_open, - .close = snd_hdspm_release, - .ioctl = snd_hdspm_ioctl, - .hw_params = snd_hdspm_hw_params, - .hw_free = snd_hdspm_hw_free, - .prepare = snd_hdspm_prepare, - .trigger = snd_hdspm_trigger, - .pointer = snd_hdspm_hw_pointer, - .page = snd_pcm_sgbuf_ops_page, -}; - -static struct snd_pcm_ops snd_hdspm_capture_ops = { +static struct snd_pcm_ops snd_hdspm_ops = { .open = snd_hdspm_open, .close = snd_hdspm_release, .ioctl = snd_hdspm_ioctl, @@ -6462,9 +6450,9 @@ static int snd_hdspm_create_pcm(struct snd_card *card, strcpy(pcm->name, hdspm->card_name); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_hdspm_playback_ops); + &snd_hdspm_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_hdspm_capture_ops); + &snd_hdspm_ops); pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; From 76a3aeac2f6c02ecf065fa9baa279dd54bf2d819 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:27 +0100 Subject: [PATCH 014/411] hdspm.h: include stdint.h in userspace MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes compilation error: sound/hdspm.h:43:2: error: unknown type name ‘uint32_t’ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/hdspm.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index b357f1a5e29cfa..5737332d38f2ce 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,6 +20,12 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __KERNEL__ +#include +#else +#include +#endif + /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 From 4bebf7091aa15ec60edf0dcbc654410a87ca21fe Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:38 +0100 Subject: [PATCH 015/411] include/uapi/sound/asound.h: include stdlib.h in userspace MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes compiler errors like: error: field ‘trigger_tstamp’ has incomplete type error: invalid application of ‘sizeof’ to incomplete t ype ‘struct timespec’ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd63595784..1fe3f4f3d69699 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -25,6 +25,9 @@ #include +#ifndef __KERNEL__ +#include +#endif /* * protocol version From bbf91c1c5bfc00c2961f15657359ee7e87de4269 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:42 +0100 Subject: [PATCH 016/411] include/uapi/sound/asequencer.h: include sound/asound.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes userspace compilation error: error: unknown type name ‘snd_seq_client_type_t’ snd_seq_client_type_t type; /* client type */ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/asequencer.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 09c8a00ea503f7..5a5fa4956ebd66 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -22,6 +22,7 @@ #ifndef _UAPI__SOUND_ASEQUENCER_H #define _UAPI__SOUND_ASEQUENCER_H +#include /** version of the sequencer */ #define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) From b9956409c281931c74ba8d0a2b61a98076a58602 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:43 +0100 Subject: [PATCH 017/411] include/uapi/sound/emu10k1.h: include sound/asound.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes userspace compilation errors like: error: field ‘id’ has incomplete type struct snd_ctl_elem_id id; /* full control ID definition */ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index d1bbaf78457aed..ec1535bb6aedd0 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,8 +23,7 @@ #define _UAPI__SOUND_EMU10K1_H #include - - +#include /* * ---- FX8010 ---- From ef7449780ebb596b47d5019eb7ba7878c30a3404 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 13:56:29 +0100 Subject: [PATCH 018/411] ALSA: hda - Drop hda_bus_template for snd_hda_bus_new() Instead of copying from the given template, let the caller fills the fields after creation. This simplifies the code after all. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 15 +--------- sound/pci/hda/hda_codec.h | 13 +------- sound/pci/hda/hda_controller.c | 54 +++++++++++++++++++--------------- 3 files changed, 32 insertions(+), 50 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2fe86d2e1b09df..215bf048c66876 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -861,14 +861,12 @@ static int snd_hda_bus_dev_disconnect(struct snd_device *device) /** * snd_hda_bus_new - create a HDA bus * @card: the card entry - * @temp: the template for hda_bus information * @busp: the pointer to store the created bus instance * * Returns 0 if successful, or a negative error code. */ int snd_hda_bus_new(struct snd_card *card, - const struct hda_bus_template *temp, - struct hda_bus **busp) + struct hda_bus **busp) { struct hda_bus *bus; int err; @@ -877,11 +875,6 @@ int snd_hda_bus_new(struct snd_card *card, .dev_free = snd_hda_bus_dev_free, }; - if (snd_BUG_ON(!temp)) - return -EINVAL; - if (snd_BUG_ON(!temp->ops.command || !temp->ops.get_response)) - return -EINVAL; - if (busp) *busp = NULL; @@ -892,12 +885,6 @@ int snd_hda_bus_new(struct snd_card *card, } bus->card = card; - bus->private_data = temp->private_data; - bus->pci = temp->pci; - bus->modelname = temp->modelname; - bus->power_save = temp->power_save; - bus->ops = temp->ops; - mutex_init(&bus->cmd_mutex); mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9c8820f21f948f..5a659488406924 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -101,15 +101,6 @@ struct hda_bus_ops { #endif }; -/* template to pass to the bus constructor */ -struct hda_bus_template { - void *private_data; - struct pci_dev *pci; - const char *modelname; - int *power_save; - struct hda_bus_ops ops; -}; - /* * codec bus * @@ -119,7 +110,6 @@ struct hda_bus_template { struct hda_bus { struct snd_card *card; - /* copied from template */ void *private_data; struct pci_dev *pci; const char *modelname; @@ -420,8 +410,7 @@ enum { /* * constructors */ -int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, - struct hda_bus **busp); +int snd_hda_bus_new(struct snd_card *card, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index dfcb5e929f9fc9..31ff8b55f3867b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1815,39 +1815,45 @@ static int get_jackpoll_interval(struct azx *chip) return j; } +static struct hda_bus_ops bus_ops = { + .command = azx_send_cmd, + .get_response = azx_get_response, + .attach_pcm = azx_attach_pcm_stream, + .bus_reset = azx_bus_reset, +#ifdef CONFIG_PM + .pm_notify = azx_power_notify, +#endif +#ifdef CONFIG_SND_HDA_DSP_LOADER + .load_dsp_prepare = azx_load_dsp_prepare, + .load_dsp_trigger = azx_load_dsp_trigger, + .load_dsp_cleanup = azx_load_dsp_cleanup, +#endif +}; + /* Codec initialization */ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, int *power_save_to) { - struct hda_bus_template bus_temp; + struct hda_bus *bus; int c, codecs, err; - memset(&bus_temp, 0, sizeof(bus_temp)); - bus_temp.private_data = chip; - bus_temp.modelname = model; - bus_temp.pci = chip->pci; - bus_temp.ops.command = azx_send_cmd; - bus_temp.ops.get_response = azx_get_response; - bus_temp.ops.attach_pcm = azx_attach_pcm_stream; - bus_temp.ops.bus_reset = azx_bus_reset; -#ifdef CONFIG_PM - bus_temp.power_save = power_save_to; - bus_temp.ops.pm_notify = azx_power_notify; -#endif -#ifdef CONFIG_SND_HDA_DSP_LOADER - bus_temp.ops.load_dsp_prepare = azx_load_dsp_prepare; - bus_temp.ops.load_dsp_trigger = azx_load_dsp_trigger; - bus_temp.ops.load_dsp_cleanup = azx_load_dsp_cleanup; -#endif - - err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus); + err = snd_hda_bus_new(chip->card, &bus); if (err < 0) return err; + chip->bus = bus; + bus->private_data = chip; + bus->pci = chip->pci; + bus->modelname = model; + bus->ops = bus_ops; +#ifdef CONFIG_PM + bus->power_save = power_save_to; +#endif + if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); - chip->bus->needs_damn_long_delay = 1; + bus->needs_damn_long_delay = 1; } codecs = 0; @@ -1883,15 +1889,15 @@ int azx_codec_create(struct azx *chip, const char *model, */ if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - chip->bus->sync_write = 1; - chip->bus->allow_bus_reset = 1; + bus->sync_write = 1; + bus->allow_bus_reset = 1; } /* Then create codec instances */ for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, &codec); + err = snd_hda_codec_new(bus, c, &codec); if (err < 0) continue; codec->jackpoll_interval = get_jackpoll_interval(chip); From 922c88a8368a61ee93653d4a2888a7f4ce263102 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 14:46:40 +0100 Subject: [PATCH 019/411] ALSA: hda - Embed struct hda_bus_unsolicited into struct hda_bus There is no big merit to handle hda_bus_unsolicited object individually, as it's tightly coupled with the hda_bus object itself. Embedding it makes the code simpler in the end. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 39 ++++----------------------------------- sound/pci/hda/hda_codec.h | 13 +++++++++++-- sound/pci/hda/hda_local.h | 17 ----------------- 3 files changed, 15 insertions(+), 54 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 215bf048c66876..0734cb35887d7a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -762,10 +762,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) return 0; trace_hda_unsol_event(bus, res, res_ex); - unsol = bus->unsol; - if (!unsol) - return 0; - + unsol = &bus->unsol; wp = (unsol->wp + 1) % HDA_UNSOL_QUEUE_SIZE; unsol->wp = wp; @@ -784,9 +781,8 @@ EXPORT_SYMBOL_GPL(snd_hda_queue_unsol_event); */ static void process_unsol_events(struct work_struct *work) { - struct hda_bus_unsolicited *unsol = - container_of(work, struct hda_bus_unsolicited, work); - struct hda_bus *bus = unsol->bus; + struct hda_bus *bus = container_of(work, struct hda_bus, unsol.work); + struct hda_bus_unsolicited *unsol = &bus->unsol; struct hda_codec *codec; unsigned int rp, caddr, res; @@ -804,27 +800,6 @@ static void process_unsol_events(struct work_struct *work) } } -/* - * initialize unsolicited queue - */ -static int init_unsol_queue(struct hda_bus *bus) -{ - struct hda_bus_unsolicited *unsol; - - if (bus->unsol) /* already initialized */ - return 0; - - unsol = kzalloc(sizeof(*unsol), GFP_KERNEL); - if (!unsol) { - dev_err(bus->card->dev, "can't allocate unsolicited queue\n"); - return -ENOMEM; - } - INIT_WORK(&unsol->work, process_unsol_events); - unsol->bus = bus; - bus->unsol = unsol; - return 0; -} - /* * destructor */ @@ -836,7 +811,6 @@ static void snd_hda_bus_free(struct hda_bus *bus) WARN_ON(!list_empty(&bus->codec_list)); if (bus->workq) flush_workqueue(bus->workq); - kfree(bus->unsol); if (bus->ops.private_free) bus->ops.private_free(bus); if (bus->workq) @@ -888,6 +862,7 @@ int snd_hda_bus_new(struct snd_card *card, mutex_init(&bus->cmd_mutex); mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); + INIT_WORK(&bus->unsol.work, process_unsol_events); snprintf(bus->workq_name, sizeof(bus->workq_name), "hd-audio%d", card->number); @@ -1689,12 +1664,6 @@ int snd_hda_codec_configure(struct hda_codec *codec) return err; } - if (codec->patch_ops.unsol_event) { - err = init_unsol_queue(codec->bus); - if (err < 0) - return err; - } - /* audio codec should override the mixer name */ if (codec->afg || !*codec->bus->card->mixername) snprintf(codec->bus->card->mixername, diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5a659488406924..96421a3b32cd94 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -66,7 +66,6 @@ struct hda_beep; struct hda_codec; struct hda_pcm; struct hda_pcm_stream; -struct hda_bus_unsolicited; /* NID type */ typedef u16 hda_nid_t; @@ -101,6 +100,16 @@ struct hda_bus_ops { #endif }; +/* unsolicited event handler */ +#define HDA_UNSOL_QUEUE_SIZE 64 +struct hda_bus_unsolicited { + /* ring buffer */ + u32 queue[HDA_UNSOL_QUEUE_SIZE * 2]; + unsigned int rp, wp; + /* workqueue */ + struct work_struct work; +}; + /* * codec bus * @@ -126,7 +135,7 @@ struct hda_bus { struct mutex prepare_mutex; /* unsolicited event queue */ - struct hda_bus_unsolicited *unsol; + struct hda_bus_unsolicited unsol; char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 62658f2f8c9f2d..49c08a7278c449 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -466,23 +466,6 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, const struct hda_fixup *fixlist); - -/* - * unsolicited event handler - */ - -#define HDA_UNSOL_QUEUE_SIZE 64 - -struct hda_bus_unsolicited { - /* ring buffer */ - u32 queue[HDA_UNSOL_QUEUE_SIZE * 2]; - unsigned int rp, wp; - - /* workqueue */ - struct work_struct work; - struct hda_bus *bus; -}; - /* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT) From 364aa716f43c991052cbb4fa05e3754bacccb95c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 16:51:17 +0100 Subject: [PATCH 020/411] ALSA: hda - Introduce azx_has_pm_runtime() macro For making the debugging of runtime PM easier, introduce azx_has_pm_runtime() and use it in all places checking the runtime pm driver capability. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 4 ++-- sound/pci/hda/hda_intel.c | 12 +++++------- sound/pci/hda/hda_priv.h | 3 +++ 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 31ff8b55f3867b..3589fc2165b09a 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1681,7 +1681,7 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) int i; #ifdef CONFIG_PM - if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) + if (azx_has_pm_runtime(chip)) if (!pm_runtime_active(chip->card->dev)) return IRQ_NONE; #endif @@ -1784,7 +1784,7 @@ static void azx_power_notify(struct hda_bus *bus, bool power_up) { struct azx *chip = bus->private_data; - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!azx_has_pm_runtime(chip)) return; if (power_up) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 36d2f20db7a420..589883291f4e52 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -852,7 +852,7 @@ static int azx_runtime_suspend(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!azx_has_pm_runtime(chip)) return 0; /* enable controller wake up event */ @@ -885,7 +885,7 @@ static int azx_runtime_resume(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!azx_has_pm_runtime(chip)) return 0; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { @@ -928,8 +928,7 @@ static int azx_runtime_idle(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!power_save_controller || - !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!power_save_controller || !azx_has_pm_runtime(chip)) return -EBUSY; return 0; @@ -1071,8 +1070,7 @@ static int azx_free(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); int i; - if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) - && chip->running) + if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); azx_del_card_list(chip); @@ -1938,7 +1936,7 @@ static int azx_probe_continue(struct azx *chip) power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); - if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || hda->use_vga_switcheroo) + if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); out_free: diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index daf458299753fa..a7b4a25c571caf 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -403,4 +403,7 @@ struct azx { #define azx_sd_readb(chip, dev, reg) \ ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) +#define azx_has_pm_runtime(chip) \ + (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) + #endif /* __SOUND_HDA_PRIV_H */ From 89a93fea6182a71cedce9de1d901e4f379322cf3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 18:04:17 +0100 Subject: [PATCH 021/411] ALSA: hda - Fold hda_priv.h into hda_controller.h There is no big reason to keep them separately. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 1 - sound/pci/hda/hda_controller.h | 391 ++++++++++++++++++++++++++++++- sound/pci/hda/hda_i915.c | 2 +- sound/pci/hda/hda_intel.c | 1 - sound/pci/hda/hda_intel.h | 2 +- sound/pci/hda/hda_priv.h | 409 --------------------------------- sound/pci/hda/hda_tegra.c | 1 - 7 files changed, 392 insertions(+), 415 deletions(-) delete mode 100644 sound/pci/hda/hda_priv.h diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 3589fc2165b09a..6b3254d479e1cb 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -30,7 +30,6 @@ #include #include #include -#include "hda_priv.h" #include "hda_controller.h" #define CREATE_TRACE_POINTS diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index c90d10fd4d8fc2..bd50b49cdd0d72 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -15,10 +15,399 @@ #ifndef __SOUND_HDA_CONTROLLER_H #define __SOUND_HDA_CONTROLLER_H +#include +#include #include +#include #include #include "hda_codec.h" -#include "hda_priv.h" + +/* + * registers + */ +#define AZX_REG_GCAP 0x00 +#define AZX_GCAP_64OK (1 << 0) /* 64bit address support */ +#define AZX_GCAP_NSDO (3 << 1) /* # of serial data out signals */ +#define AZX_GCAP_BSS (31 << 3) /* # of bidirectional streams */ +#define AZX_GCAP_ISS (15 << 8) /* # of input streams */ +#define AZX_GCAP_OSS (15 << 12) /* # of output streams */ +#define AZX_REG_VMIN 0x02 +#define AZX_REG_VMAJ 0x03 +#define AZX_REG_OUTPAY 0x04 +#define AZX_REG_INPAY 0x06 +#define AZX_REG_GCTL 0x08 +#define AZX_GCTL_RESET (1 << 0) /* controller reset */ +#define AZX_GCTL_FCNTRL (1 << 1) /* flush control */ +#define AZX_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ +#define AZX_REG_WAKEEN 0x0c +#define AZX_REG_STATESTS 0x0e +#define AZX_REG_GSTS 0x10 +#define AZX_GSTS_FSTS (1 << 1) /* flush status */ +#define AZX_REG_INTCTL 0x20 +#define AZX_REG_INTSTS 0x24 +#define AZX_REG_WALLCLK 0x30 /* 24Mhz source */ +#define AZX_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */ +#define AZX_REG_SSYNC 0x38 +#define AZX_REG_CORBLBASE 0x40 +#define AZX_REG_CORBUBASE 0x44 +#define AZX_REG_CORBWP 0x48 +#define AZX_REG_CORBRP 0x4a +#define AZX_CORBRP_RST (1 << 15) /* read pointer reset */ +#define AZX_REG_CORBCTL 0x4c +#define AZX_CORBCTL_RUN (1 << 1) /* enable DMA */ +#define AZX_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ +#define AZX_REG_CORBSTS 0x4d +#define AZX_CORBSTS_CMEI (1 << 0) /* memory error indication */ +#define AZX_REG_CORBSIZE 0x4e + +#define AZX_REG_RIRBLBASE 0x50 +#define AZX_REG_RIRBUBASE 0x54 +#define AZX_REG_RIRBWP 0x58 +#define AZX_RIRBWP_RST (1 << 15) /* write pointer reset */ +#define AZX_REG_RINTCNT 0x5a +#define AZX_REG_RIRBCTL 0x5c +#define AZX_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ +#define AZX_RBCTL_DMA_EN (1 << 1) /* enable DMA */ +#define AZX_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ +#define AZX_REG_RIRBSTS 0x5d +#define AZX_RBSTS_IRQ (1 << 0) /* response irq */ +#define AZX_RBSTS_OVERRUN (1 << 2) /* overrun irq */ +#define AZX_REG_RIRBSIZE 0x5e + +#define AZX_REG_IC 0x60 +#define AZX_REG_IR 0x64 +#define AZX_REG_IRS 0x68 +#define AZX_IRS_VALID (1<<1) +#define AZX_IRS_BUSY (1<<0) + +#define AZX_REG_DPLBASE 0x70 +#define AZX_REG_DPUBASE 0x74 +#define AZX_DPLBASE_ENABLE 0x1 /* Enable position buffer */ + +/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ +enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; + +/* stream register offsets from stream base */ +#define AZX_REG_SD_CTL 0x00 +#define AZX_REG_SD_STS 0x03 +#define AZX_REG_SD_LPIB 0x04 +#define AZX_REG_SD_CBL 0x08 +#define AZX_REG_SD_LVI 0x0c +#define AZX_REG_SD_FIFOW 0x0e +#define AZX_REG_SD_FIFOSIZE 0x10 +#define AZX_REG_SD_FORMAT 0x12 +#define AZX_REG_SD_BDLPL 0x18 +#define AZX_REG_SD_BDLPU 0x1c + +/* PCI space */ +#define AZX_PCIREG_TCSEL 0x44 + +/* + * other constants + */ + +/* max number of fragments - we may use more if allocating more pages for BDL */ +#define BDL_SIZE 4096 +#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) +#define AZX_MAX_FRAG 32 +/* max buffer size - no h/w limit, you can increase as you like */ +#define AZX_MAX_BUF_SIZE (1024*1024*1024) + +/* RIRB int mask: overrun[2], response[0] */ +#define RIRB_INT_RESPONSE 0x01 +#define RIRB_INT_OVERRUN 0x04 +#define RIRB_INT_MASK 0x05 + +/* STATESTS int mask: S3,SD2,SD1,SD0 */ +#define AZX_MAX_CODECS 8 +#define AZX_DEFAULT_CODECS 4 +#define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1) + +/* SD_CTL bits */ +#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ +#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ +#define SD_CTL_STRIPE (3 << 16) /* stripe control */ +#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */ +#define SD_CTL_DIR (1 << 19) /* bi-directional stream */ +#define SD_CTL_STREAM_TAG_MASK (0xf << 20) +#define SD_CTL_STREAM_TAG_SHIFT 20 + +/* SD_CTL and SD_STS */ +#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */ +#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ +#define SD_INT_COMPLETE 0x04 /* completion interrupt */ +#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\ + SD_INT_COMPLETE) + +/* SD_STS */ +#define SD_STS_FIFO_READY 0x20 /* FIFO ready */ + +/* INTCTL and INTSTS */ +#define AZX_INT_ALL_STREAM 0xff /* all stream interrupts */ +#define AZX_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ +#define AZX_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ + +/* below are so far hardcoded - should read registers in future */ +#define AZX_MAX_CORB_ENTRIES 256 +#define AZX_MAX_RIRB_ENTRIES 256 + +/* driver quirks (capabilities) */ +/* bits 0-7 are used for indicating driver type */ +#define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ +#define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ +#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ +#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ +#define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ +#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ +#define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ +#define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ +#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */ +#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ +#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ +#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ +#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ +/* 22 unused */ +#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ +#define AZX_DCAPS_REVERSE_ASSIGN (1 << 24) /* Assign devices in reverse order */ +#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ +#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ +#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ +#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ +#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ + +enum { + AZX_SNOOP_TYPE_NONE, + AZX_SNOOP_TYPE_SCH, + AZX_SNOOP_TYPE_ATI, + AZX_SNOOP_TYPE_NVIDIA, +}; + +/* HD Audio class code */ +#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 + +struct azx_dev { + struct snd_dma_buffer bdl; /* BDL buffer */ + u32 *posbuf; /* position buffer pointer */ + + unsigned int bufsize; /* size of the play buffer in bytes */ + unsigned int period_bytes; /* size of the period in bytes */ + unsigned int frags; /* number for period in the play buffer */ + unsigned int fifo_size; /* FIFO size */ + unsigned long start_wallclk; /* start + minimum wallclk */ + unsigned long period_wallclk; /* wallclk for period */ + + void __iomem *sd_addr; /* stream descriptor pointer */ + + u32 sd_int_sta_mask; /* stream int status mask */ + + /* pcm support */ + struct snd_pcm_substream *substream; /* assigned substream, + * set in PCM open + */ + unsigned int format_val; /* format value to be set in the + * controller and the codec + */ + unsigned char stream_tag; /* assigned stream */ + unsigned char index; /* stream index */ + int assigned_key; /* last device# key assigned to */ + + unsigned int opened:1; + unsigned int running:1; + unsigned int irq_pending:1; + unsigned int prepared:1; + unsigned int locked:1; + /* + * For VIA: + * A flag to ensure DMA position is 0 + * when link position is not greater than FIFO size + */ + unsigned int insufficient:1; + unsigned int wc_marked:1; + unsigned int no_period_wakeup:1; + + struct timecounter azx_tc; + struct cyclecounter azx_cc; + + int delay_negative_threshold; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + /* Allows dsp load to have sole access to the playback stream. */ + struct mutex dsp_mutex; +#endif +}; + +/* CORB/RIRB */ +struct azx_rb { + u32 *buf; /* CORB/RIRB buffer + * Each CORB entry is 4byte, RIRB is 8byte + */ + dma_addr_t addr; /* physical address of CORB/RIRB buffer */ + /* for RIRB */ + unsigned short rp, wp; /* read/write pointers */ + int cmds[AZX_MAX_CODECS]; /* number of pending requests */ + u32 res[AZX_MAX_CODECS]; /* last read value */ +}; + +struct azx; + +/* Functions to read/write to hda registers. */ +struct hda_controller_ops { + /* Register Access */ + void (*reg_writel)(u32 value, u32 __iomem *addr); + u32 (*reg_readl)(u32 __iomem *addr); + void (*reg_writew)(u16 value, u16 __iomem *addr); + u16 (*reg_readw)(u16 __iomem *addr); + void (*reg_writeb)(u8 value, u8 __iomem *addr); + u8 (*reg_readb)(u8 __iomem *addr); + /* Disable msi if supported, PCI only */ + int (*disable_msi_reset_irq)(struct azx *); + /* Allocation ops */ + int (*dma_alloc_pages)(struct azx *chip, + int type, + size_t size, + struct snd_dma_buffer *buf); + void (*dma_free_pages)(struct azx *chip, struct snd_dma_buffer *buf); + int (*substream_alloc_pages)(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size); + int (*substream_free_pages)(struct azx *chip, + struct snd_pcm_substream *substream); + void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, + struct vm_area_struct *area); + /* Check if current position is acceptable */ + int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); +}; + +struct azx_pcm { + struct azx *chip; + struct snd_pcm *pcm; + struct hda_codec *codec; + struct hda_pcm_stream *hinfo[2]; + struct list_head list; +}; + +typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *); +typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos); + +struct azx { + struct snd_card *card; + struct pci_dev *pci; + int dev_index; + + /* chip type specific */ + int driver_type; + unsigned int driver_caps; + int playback_streams; + int playback_index_offset; + int capture_streams; + int capture_index_offset; + int num_streams; + const int *jackpoll_ms; /* per-card jack poll interval */ + + /* Register interaction. */ + const struct hda_controller_ops *ops; + + /* position adjustment callbacks */ + azx_get_pos_callback_t get_position[2]; + azx_get_delay_callback_t get_delay[2]; + + /* pci resources */ + unsigned long addr; + void __iomem *remap_addr; + int irq; + + /* locks */ + spinlock_t reg_lock; + struct mutex open_mutex; /* Prevents concurrent open/close operations */ + + /* streams (x num_streams) */ + struct azx_dev *azx_dev; + + /* PCM */ + struct list_head pcm_list; /* azx_pcm list */ + + /* HD codec */ + unsigned short codec_mask; + int codec_probe_mask; /* copied from probe_mask option */ + struct hda_bus *bus; + unsigned int beep_mode; + + /* CORB/RIRB */ + struct azx_rb corb; + struct azx_rb rirb; + + /* CORB/RIRB and position buffers */ + struct snd_dma_buffer rb; + struct snd_dma_buffer posbuf; + +#ifdef CONFIG_SND_HDA_PATCH_LOADER + const struct firmware *fw; +#endif + + /* flags */ + const int *bdl_pos_adj; + int poll_count; + unsigned int running:1; + unsigned int initialized:1; + unsigned int single_cmd:1; + unsigned int polling_mode:1; + unsigned int msi:1; + unsigned int probing:1; /* codec probing phase */ + unsigned int snoop:1; + unsigned int align_buffer_size:1; + unsigned int region_requested:1; + unsigned int disabled:1; /* disabled by VGA-switcher */ + + /* for debugging */ + unsigned int last_cmd[AZX_MAX_CODECS]; + + /* reboot notifier (for mysterious hangup problem at power-down) */ + struct notifier_block reboot_notifier; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct azx_dev saved_azx_dev; +#endif +}; + +#ifdef CONFIG_X86 +#define azx_snoop(chip) ((chip)->snoop) +#else +#define azx_snoop(chip) true +#endif + +/* + * macros for easy use + */ + +#define azx_writel(chip, reg, value) \ + ((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readl(chip, reg) \ + ((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg)) +#define azx_writew(chip, reg, value) \ + ((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readw(chip, reg) \ + ((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg)) +#define azx_writeb(chip, reg, value) \ + ((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readb(chip, reg) \ + ((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg)) + +#define azx_sd_writel(chip, dev, reg, value) \ + ((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readl(chip, dev, reg) \ + ((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_writew(chip, dev, reg, value) \ + ((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readw(chip, dev, reg) \ + ((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_writeb(chip, dev, reg, value) \ + ((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readb(chip, dev, reg) \ + ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) + +#define azx_has_pm_runtime(chip) \ + (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) /* PCM setup */ static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream) diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 714894527e06a5..52a85d87c23c58 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -22,7 +22,7 @@ #include #include #include -#include "hda_priv.h" +#include "hda_controller.h" #include "hda_intel.h" /* Intel HSW/BDW display HDA controller Extended Mode registers. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 589883291f4e52..ced44a75f8eaa0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -62,7 +62,6 @@ #include #include "hda_codec.h" #include "hda_controller.h" -#include "hda_priv.h" #include "hda_intel.h" /* position fix mode */ diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 3486118354766c..d5231f7216a700 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -17,7 +17,7 @@ #define __SOUND_HDA_INTEL_H #include -#include "hda_priv.h" +#include "hda_controller.h" struct hda_intel { struct azx chip; diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h deleted file mode 100644 index a7b4a25c571caf..00000000000000 --- a/sound/pci/hda/hda_priv.h +++ /dev/null @@ -1,409 +0,0 @@ -/* - * Common defines for the alsa driver code base for HD Audio. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the Free - * Software Foundation; either version 2 of the License, or (at your option) - * any later version. - * - * This program is distributed in the hope that it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ - -#ifndef __SOUND_HDA_PRIV_H -#define __SOUND_HDA_PRIV_H - -#include -#include -#include - -/* - * registers - */ -#define AZX_REG_GCAP 0x00 -#define AZX_GCAP_64OK (1 << 0) /* 64bit address support */ -#define AZX_GCAP_NSDO (3 << 1) /* # of serial data out signals */ -#define AZX_GCAP_BSS (31 << 3) /* # of bidirectional streams */ -#define AZX_GCAP_ISS (15 << 8) /* # of input streams */ -#define AZX_GCAP_OSS (15 << 12) /* # of output streams */ -#define AZX_REG_VMIN 0x02 -#define AZX_REG_VMAJ 0x03 -#define AZX_REG_OUTPAY 0x04 -#define AZX_REG_INPAY 0x06 -#define AZX_REG_GCTL 0x08 -#define AZX_GCTL_RESET (1 << 0) /* controller reset */ -#define AZX_GCTL_FCNTRL (1 << 1) /* flush control */ -#define AZX_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ -#define AZX_REG_WAKEEN 0x0c -#define AZX_REG_STATESTS 0x0e -#define AZX_REG_GSTS 0x10 -#define AZX_GSTS_FSTS (1 << 1) /* flush status */ -#define AZX_REG_INTCTL 0x20 -#define AZX_REG_INTSTS 0x24 -#define AZX_REG_WALLCLK 0x30 /* 24Mhz source */ -#define AZX_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */ -#define AZX_REG_SSYNC 0x38 -#define AZX_REG_CORBLBASE 0x40 -#define AZX_REG_CORBUBASE 0x44 -#define AZX_REG_CORBWP 0x48 -#define AZX_REG_CORBRP 0x4a -#define AZX_CORBRP_RST (1 << 15) /* read pointer reset */ -#define AZX_REG_CORBCTL 0x4c -#define AZX_CORBCTL_RUN (1 << 1) /* enable DMA */ -#define AZX_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ -#define AZX_REG_CORBSTS 0x4d -#define AZX_CORBSTS_CMEI (1 << 0) /* memory error indication */ -#define AZX_REG_CORBSIZE 0x4e - -#define AZX_REG_RIRBLBASE 0x50 -#define AZX_REG_RIRBUBASE 0x54 -#define AZX_REG_RIRBWP 0x58 -#define AZX_RIRBWP_RST (1 << 15) /* write pointer reset */ -#define AZX_REG_RINTCNT 0x5a -#define AZX_REG_RIRBCTL 0x5c -#define AZX_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ -#define AZX_RBCTL_DMA_EN (1 << 1) /* enable DMA */ -#define AZX_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ -#define AZX_REG_RIRBSTS 0x5d -#define AZX_RBSTS_IRQ (1 << 0) /* response irq */ -#define AZX_RBSTS_OVERRUN (1 << 2) /* overrun irq */ -#define AZX_REG_RIRBSIZE 0x5e - -#define AZX_REG_IC 0x60 -#define AZX_REG_IR 0x64 -#define AZX_REG_IRS 0x68 -#define AZX_IRS_VALID (1<<1) -#define AZX_IRS_BUSY (1<<0) - -#define AZX_REG_DPLBASE 0x70 -#define AZX_REG_DPUBASE 0x74 -#define AZX_DPLBASE_ENABLE 0x1 /* Enable position buffer */ - -/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ -enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; - -/* stream register offsets from stream base */ -#define AZX_REG_SD_CTL 0x00 -#define AZX_REG_SD_STS 0x03 -#define AZX_REG_SD_LPIB 0x04 -#define AZX_REG_SD_CBL 0x08 -#define AZX_REG_SD_LVI 0x0c -#define AZX_REG_SD_FIFOW 0x0e -#define AZX_REG_SD_FIFOSIZE 0x10 -#define AZX_REG_SD_FORMAT 0x12 -#define AZX_REG_SD_BDLPL 0x18 -#define AZX_REG_SD_BDLPU 0x1c - -/* PCI space */ -#define AZX_PCIREG_TCSEL 0x44 - -/* - * other constants - */ - -/* max number of fragments - we may use more if allocating more pages for BDL */ -#define BDL_SIZE 4096 -#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) -#define AZX_MAX_FRAG 32 -/* max buffer size - no h/w limit, you can increase as you like */ -#define AZX_MAX_BUF_SIZE (1024*1024*1024) - -/* RIRB int mask: overrun[2], response[0] */ -#define RIRB_INT_RESPONSE 0x01 -#define RIRB_INT_OVERRUN 0x04 -#define RIRB_INT_MASK 0x05 - -/* STATESTS int mask: S3,SD2,SD1,SD0 */ -#define AZX_MAX_CODECS 8 -#define AZX_DEFAULT_CODECS 4 -#define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1) - -/* SD_CTL bits */ -#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ -#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ -#define SD_CTL_STRIPE (3 << 16) /* stripe control */ -#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */ -#define SD_CTL_DIR (1 << 19) /* bi-directional stream */ -#define SD_CTL_STREAM_TAG_MASK (0xf << 20) -#define SD_CTL_STREAM_TAG_SHIFT 20 - -/* SD_CTL and SD_STS */ -#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */ -#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ -#define SD_INT_COMPLETE 0x04 /* completion interrupt */ -#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\ - SD_INT_COMPLETE) - -/* SD_STS */ -#define SD_STS_FIFO_READY 0x20 /* FIFO ready */ - -/* INTCTL and INTSTS */ -#define AZX_INT_ALL_STREAM 0xff /* all stream interrupts */ -#define AZX_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ -#define AZX_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ - -/* below are so far hardcoded - should read registers in future */ -#define AZX_MAX_CORB_ENTRIES 256 -#define AZX_MAX_RIRB_ENTRIES 256 - -/* driver quirks (capabilities) */ -/* bits 0-7 are used for indicating driver type */ -#define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ -#define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ -#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ -#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ -#define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ -#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ -#define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ -#define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ -#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */ -#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ -#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ -#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ -#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ -/* 22 unused */ -#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ -#define AZX_DCAPS_REVERSE_ASSIGN (1 << 24) /* Assign devices in reverse order */ -#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ -#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ -#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ -#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ -#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ - -enum { - AZX_SNOOP_TYPE_NONE , - AZX_SNOOP_TYPE_SCH, - AZX_SNOOP_TYPE_ATI, - AZX_SNOOP_TYPE_NVIDIA, -}; - -/* HD Audio class code */ -#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 - -struct azx_dev { - struct snd_dma_buffer bdl; /* BDL buffer */ - u32 *posbuf; /* position buffer pointer */ - - unsigned int bufsize; /* size of the play buffer in bytes */ - unsigned int period_bytes; /* size of the period in bytes */ - unsigned int frags; /* number for period in the play buffer */ - unsigned int fifo_size; /* FIFO size */ - unsigned long start_wallclk; /* start + minimum wallclk */ - unsigned long period_wallclk; /* wallclk for period */ - - void __iomem *sd_addr; /* stream descriptor pointer */ - - u32 sd_int_sta_mask; /* stream int status mask */ - - /* pcm support */ - struct snd_pcm_substream *substream; /* assigned substream, - * set in PCM open - */ - unsigned int format_val; /* format value to be set in the - * controller and the codec - */ - unsigned char stream_tag; /* assigned stream */ - unsigned char index; /* stream index */ - int assigned_key; /* last device# key assigned to */ - - unsigned int opened:1; - unsigned int running:1; - unsigned int irq_pending:1; - unsigned int prepared:1; - unsigned int locked:1; - /* - * For VIA: - * A flag to ensure DMA position is 0 - * when link position is not greater than FIFO size - */ - unsigned int insufficient:1; - unsigned int wc_marked:1; - unsigned int no_period_wakeup:1; - - struct timecounter azx_tc; - struct cyclecounter azx_cc; - - int delay_negative_threshold; - -#ifdef CONFIG_SND_HDA_DSP_LOADER - /* Allows dsp load to have sole access to the playback stream. */ - struct mutex dsp_mutex; -#endif -}; - -/* CORB/RIRB */ -struct azx_rb { - u32 *buf; /* CORB/RIRB buffer - * Each CORB entry is 4byte, RIRB is 8byte - */ - dma_addr_t addr; /* physical address of CORB/RIRB buffer */ - /* for RIRB */ - unsigned short rp, wp; /* read/write pointers */ - int cmds[AZX_MAX_CODECS]; /* number of pending requests */ - u32 res[AZX_MAX_CODECS]; /* last read value */ -}; - -struct azx; - -/* Functions to read/write to hda registers. */ -struct hda_controller_ops { - /* Register Access */ - void (*reg_writel)(u32 value, u32 __iomem *addr); - u32 (*reg_readl)(u32 __iomem *addr); - void (*reg_writew)(u16 value, u16 __iomem *addr); - u16 (*reg_readw)(u16 __iomem *addr); - void (*reg_writeb)(u8 value, u8 __iomem *addr); - u8 (*reg_readb)(u8 __iomem *addr); - /* Disable msi if supported, PCI only */ - int (*disable_msi_reset_irq)(struct azx *); - /* Allocation ops */ - int (*dma_alloc_pages)(struct azx *chip, - int type, - size_t size, - struct snd_dma_buffer *buf); - void (*dma_free_pages)(struct azx *chip, struct snd_dma_buffer *buf); - int (*substream_alloc_pages)(struct azx *chip, - struct snd_pcm_substream *substream, - size_t size); - int (*substream_free_pages)(struct azx *chip, - struct snd_pcm_substream *substream); - void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, - struct vm_area_struct *area); - /* Check if current position is acceptable */ - int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); -}; - -struct azx_pcm { - struct azx *chip; - struct snd_pcm *pcm; - struct hda_codec *codec; - struct hda_pcm_stream *hinfo[2]; - struct list_head list; -}; - -typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *); -typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos); - -struct azx { - struct snd_card *card; - struct pci_dev *pci; - int dev_index; - - /* chip type specific */ - int driver_type; - unsigned int driver_caps; - int playback_streams; - int playback_index_offset; - int capture_streams; - int capture_index_offset; - int num_streams; - const int *jackpoll_ms; /* per-card jack poll interval */ - - /* Register interaction. */ - const struct hda_controller_ops *ops; - - /* position adjustment callbacks */ - azx_get_pos_callback_t get_position[2]; - azx_get_delay_callback_t get_delay[2]; - - /* pci resources */ - unsigned long addr; - void __iomem *remap_addr; - int irq; - - /* locks */ - spinlock_t reg_lock; - struct mutex open_mutex; /* Prevents concurrent open/close operations */ - - /* streams (x num_streams) */ - struct azx_dev *azx_dev; - - /* PCM */ - struct list_head pcm_list; /* azx_pcm list */ - - /* HD codec */ - unsigned short codec_mask; - int codec_probe_mask; /* copied from probe_mask option */ - struct hda_bus *bus; - unsigned int beep_mode; - - /* CORB/RIRB */ - struct azx_rb corb; - struct azx_rb rirb; - - /* CORB/RIRB and position buffers */ - struct snd_dma_buffer rb; - struct snd_dma_buffer posbuf; - -#ifdef CONFIG_SND_HDA_PATCH_LOADER - const struct firmware *fw; -#endif - - /* flags */ - const int *bdl_pos_adj; - int poll_count; - unsigned int running:1; - unsigned int initialized:1; - unsigned int single_cmd:1; - unsigned int polling_mode:1; - unsigned int msi:1; - unsigned int probing:1; /* codec probing phase */ - unsigned int snoop:1; - unsigned int align_buffer_size:1; - unsigned int region_requested:1; - unsigned int disabled:1; /* disabled by VGA-switcher */ - - /* for debugging */ - unsigned int last_cmd[AZX_MAX_CODECS]; - - /* reboot notifier (for mysterious hangup problem at power-down) */ - struct notifier_block reboot_notifier; - -#ifdef CONFIG_SND_HDA_DSP_LOADER - struct azx_dev saved_azx_dev; -#endif -}; - -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif - -/* - * macros for easy use - */ - -#define azx_writel(chip, reg, value) \ - ((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readl(chip, reg) \ - ((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg)) -#define azx_writew(chip, reg, value) \ - ((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readw(chip, reg) \ - ((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg)) -#define azx_writeb(chip, reg, value) \ - ((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readb(chip, reg) \ - ((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg)) - -#define azx_sd_writel(chip, dev, reg, value) \ - ((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readl(chip, dev, reg) \ - ((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_writew(chip, dev, reg, value) \ - ((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readw(chip, dev, reg) \ - ((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_writeb(chip, dev, reg, value) \ - ((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readb(chip, dev, reg) \ - ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) - -#define azx_has_pm_runtime(chip) \ - (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) - -#endif /* __SOUND_HDA_PRIV_H */ diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 375e94f4cf5265..7d0d04480f4896 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -37,7 +37,6 @@ #include "hda_codec.h" #include "hda_controller.h" -#include "hda_priv.h" /* Defines for Nvidia Tegra HDA support */ #define HDA_BAR0 0x8000 From b8f28d53641f13902790904ab15028ff8ecd0882 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 18:06:45 +0100 Subject: [PATCH 022/411] ALSA: hda - Drop azx_mixer_create() It's just an indirection, so let the caller directly calling snd_hda_build_controls(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 7 ------- sound/pci/hda/hda_controller.h | 1 - sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- 4 files changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6b3254d479e1cb..6fab39133051bc 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1923,13 +1923,6 @@ int azx_codec_configure(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_codec_configure); -/* mixer creation - all stuff is implemented in hda module */ -int azx_mixer_create(struct azx *chip) -{ - return snd_hda_build_controls(chip->bus); -} -EXPORT_SYMBOL_GPL(azx_mixer_create); - static bool is_input_stream(struct azx *chip, unsigned char index) { diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index bd50b49cdd0d72..79808e1a79ac58 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -436,7 +436,6 @@ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, int *power_save_to); int azx_codec_configure(struct azx *chip); -int azx_mixer_create(struct azx *chip); int azx_init_stream(struct azx *chip); void azx_notifier_register(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ced44a75f8eaa0..b0a05691abe70c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1923,7 +1923,7 @@ static int azx_probe_continue(struct azx *chip) goto out_free; /* create mixer controls */ - err = azx_mixer_create(chip); + err = snd_hda_build_controls(chip->bus); if (err < 0) goto out_free; diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 7d0d04480f4896..f305c2a99206dd 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -516,7 +516,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create mixer controls */ - err = azx_mixer_create(chip); + err = snd_hda_build_controls(chip->bus); if (err < 0) goto out_free; From 96d2bd6e3cdf57926f80605d6e28051bb6b24eb3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 18:12:22 +0100 Subject: [PATCH 023/411] ALSA: hda - Split azx_codec_create() to two phases azx_create_codec() function does actually two things: create a bus and probe codecs. For the future work, split this to two logical functions, azx_bus_create() and azx_probe_codecs(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 40 ++++++++++++++++++++-------------- sound/pci/hda/hda_controller.h | 5 ++--- sound/pci/hda/hda_intel.c | 8 ++++--- sound/pci/hda/hda_tegra.c | 6 ++++- 4 files changed, 36 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6fab39133051bc..2a674525e56f87 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1829,13 +1829,11 @@ static struct hda_bus_ops bus_ops = { #endif }; -/* Codec initialization */ -int azx_codec_create(struct azx *chip, const char *model, - unsigned int max_slots, - int *power_save_to) +/* HD-audio bus initialization */ +int azx_bus_create(struct azx *chip, const char *model, int *power_save_to) { struct hda_bus *bus; - int c, codecs, err; + int err; err = snd_hda_bus_new(chip->card, &bus); if (err < 0) @@ -1855,6 +1853,26 @@ int azx_codec_create(struct azx *chip, const char *model, bus->needs_damn_long_delay = 1; } + /* AMD chipsets often cause the communication stalls upon certain + * sequence like the pin-detection. It seems that forcing the synced + * access works around the stall. Grrr... + */ + if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { + dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); + bus->sync_write = 1; + bus->allow_bus_reset = 1; + } + + return 0; +} +EXPORT_SYMBOL_GPL(azx_bus_create); + +/* Probe codecs */ +int azx_probe_codecs(struct azx *chip, unsigned int max_slots) +{ + struct hda_bus *bus = chip->bus; + int c, codecs, err; + codecs = 0; if (!max_slots) max_slots = AZX_DEFAULT_CODECS; @@ -1882,16 +1900,6 @@ int azx_codec_create(struct azx *chip, const char *model, } } - /* AMD chipsets often cause the communication stalls upon certain - * sequence like the pin-detection. It seems that forcing the synced - * access works around the stall. Grrr... - */ - if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - bus->sync_write = 1; - bus->allow_bus_reset = 1; - } - /* Then create codec instances */ for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { @@ -1910,7 +1918,7 @@ int azx_codec_create(struct azx *chip, const char *model, } return 0; } -EXPORT_SYMBOL_GPL(azx_codec_create); +EXPORT_SYMBOL_GPL(azx_probe_codecs); /* configure each codec instance */ int azx_codec_configure(struct azx *chip) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 79808e1a79ac58..0d09aa6b49ac82 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -432,9 +432,8 @@ void azx_enter_link_reset(struct azx *chip); irqreturn_t azx_interrupt(int irq, void *dev_id); /* Codec interface */ -int azx_codec_create(struct azx *chip, const char *model, - unsigned int max_slots, - int *power_save_to); +int azx_bus_create(struct azx *chip, const char *model, int *power_save_to); +int azx_probe_codecs(struct azx *chip, unsigned int max_slots); int azx_codec_configure(struct azx *chip); int azx_init_stream(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b0a05691abe70c..5e00cc4a722f4c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1893,12 +1893,14 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_codec_create(chip, model[dev], - azx_max_codecs[chip->driver_type], - power_save_addr); + err = azx_bus_create(chip, model[dev], power_save_addr); + if (err < 0) + goto out_free; + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); if (err < 0) goto out_free; + #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { err = snd_hda_load_patch(chip->bus, chip->fw->size, diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index f305c2a99206dd..1bd7a9e040465b 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -502,7 +502,11 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create codec instances */ - err = azx_codec_create(chip, NULL, 0, &power_save); + err = azx_bus_create(chip, NULL, &power_save); + if (err < 0) + goto out_free; + + err = azx_probe_codecs(chip, 0); if (err < 0) goto out_free; From 781c7b9615dc1441d6743124086eb3da14bf46e9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 10:26:25 +0100 Subject: [PATCH 024/411] ALSA: hda - Avoid unnecessary power-up at mixer amp changes When the mixer amp is touched by control elements, we don't have to power up always; if the codec was suspended at the time, we can just update the amp cache and it's reflected to the hardware upon resume. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0734cb35887d7a..5e755eb6f19a0b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2148,11 +2148,10 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read); static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val, - bool init_only) + bool init_only, bool cache_only) { struct hda_amp_info *info; unsigned int caps; - unsigned int cache_only; if (snd_BUG_ON(mask & ~0xff)) mask &= 0xff; @@ -2170,7 +2169,7 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; } info->vol[ch] = val; - cache_only = info->head.dirty = codec->cached_write; + info->head.dirty |= cache_only; caps = info->amp_caps; mutex_unlock(&codec->hash_mutex); if (!cache_only) @@ -2194,7 +2193,8 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, direction, idx, mask, val, false); + return codec_amp_update(codec, nid, ch, direction, idx, mask, val, + false, codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); @@ -2241,7 +2241,8 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true); + return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true, + codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); @@ -2383,8 +2384,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, maxval = get_amp_max_value(codec, nid, dir, 0); if (val > maxval) val = maxval; - return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, - HDA_AMP_VOLMASK, val); + return codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val, + false, !hda_codec_is_power_on(codec)); } /** @@ -2434,14 +2435,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); @@ -3109,19 +3108,19 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change = codec_amp_update(codec, nid, 0, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change |= codec_amp_update(codec, nid, 1, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); hda_call_check_power_status(codec, nid); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); From 229d043096ea8e58829d37d35767afeac15997f5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:03 -0600 Subject: [PATCH 025/411] ALSA: core: selection of audio_tstamp type and accuracy reports Audio timestamps can be extracted from sample counters, wall clocks, PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This patch provides the ability to report timestamping capabilities, select timestamp types and retrieve timestamp accuracy, if supported. Details can be found in Documentations/sound/alsa/timestamping.txt This functionality is introduced by reclaiming the reserved_aligned field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a in snd_pcm_status to provide userspace with selection/query capabilities. Additional driver_tstamp and audio_tstamp_accuracy fields are also added. snd_pcm_mmap_status remains a read-only structure with only the audio timestamp value accessible from user space. The selection of audio timestamp type is done through snd_pcm_status only This commit does not impact ABI and does not impact the default behavior. By default audio timestamp is aligned with hw_pointer and reports the DMA position. Backwards compatibility is handled by using the HDAudio wall clock for playback and the hw_ptr for all other cases. For timestamp selection a new STATUS_EXT ioctl is introduced with read/write parameters. Alsa-lib will be modified to make use of STATUS_EXT. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++ include/sound/pcm.h | 60 +++++++ include/uapi/sound/asound.h | 34 +++- 3 files changed, 290 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt new file mode 100644 index 00000000000000..0b191a23f534d1 --- /dev/null +++ b/Documentation/sound/alsa/timestamping.txt @@ -0,0 +1,200 @@ +The ALSA API can provide two different system timestamps: + +- Trigger_tstamp is the system time snapshot taken when the .trigger +callback is invoked. This snapshot is taken by the ALSA core in the +general case, but specific hardware may have synchronization +capabilities or conversely may only be able to provide a correct +estimate with a delay. In the latter two cases, the low-level driver +is responsible for updating the trigger_tstamp at the most appropriate +and precise moment. Applications should not rely solely on the first +trigger_tstamp but update their internal calculations if the driver +provides a refined estimate with a delay. + +- tstamp is the current system timestamp updated during the last +event or application query. +The difference (tstamp - trigger_tstamp) defines the elapsed time. + +The ALSA API provides reports two basic pieces of information, avail +and delay, which combined with the trigger and current system +timestamps allow for applications to keep track of the 'fullness' of +the ring buffer and the amount of queued samples. + +The use of these different pointers and time information depends on +the application needs: + +- 'avail' reports how much can be written in the ring buffer +- 'delay' reports the time it will take to hear a new sample after all +queued samples have been played out. + +When timestamps are enabled, the avail/delay information is reported +along with a snapshot of system time. Applications can select from +CLOCK_REALTIME (NTP corrections including going backwards), +CLOCK_MONOTONIC (NTP corrections but never going backwards), +CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode +dynamically with sw_params + + +The ALSA API also provide an audio_tstamp which reflects the passage +of time as measured by different components of audio hardware. In +ascii-art, this could be represented as follows (for the playback +case): + + +--------------------------------------------------------------> time + ^ ^ ^ ^ ^ + | | | | | + analog link dma app FullBuffer + time time time time time + | | | | | + |< codec delay >|<--hw delay-->||<---avail->| + |<----------------- delay---------------------->| | + |<----ring buffer length---->| + +The analog time is taken at the last stage of the playback, as close +as possible to the actual transducer + +The link time is taken at the output of the SOC/chipset as the samples +are pushed on a link. The link time can be directly measured if +supported in hardware by sample counters or wallclocks (e.g. with +HDAudio 24MHz or PTP clock for networked solutions) or indirectly +estimated (e.g. with the frame counter in USB). + +The DMA time is measured using counters - typically the least reliable +of all measurements due to the bursty natured of DMA transfers. + +The app time corresponds to the time tracked by an application after +writing in the ring buffer. + +The application can query what the hardware supports, define which +audio time it wants reported by selecting the relevant settings in +audio_tstamp_config fields, get an estimate of the timestamp +accuracy. It can also request the delay-to-analog be included in the +measurement. Direct access to the link time is very interesting on +platforms that provide an embedded DSP; measuring directly the link +time with dedicated hardware, possibly synchronized with system time, +removes the need to keep track of internal DSP processing times and +latency. + +In case the application requests an audio tstamp that is not supported +in hardware/low-level driver, the type is overridden as DEFAULT and the +timestamp will report the DMA time based on the hw_pointer value. + +For backwards compatibility with previous implementations that did not +provide timestamp selection, with a zero-valued COMPAT timestamp type +the results will default to the HDAudio wall clock for playback +streams and to the DMA time (hw_ptr) in all other cases. + +The audio timestamp accuracy can be returned to user-space, so that +appropriate decisions are made: + +- for dma time (default), the granularity of the transfers can be + inferred from the steps between updates and in turn provide + information on how much the application pointer can be rewound + safely. + +- the link time can be used to track long-term drifts between audio + and system time using the (tstamp-trigger_tstamp)/audio_tstamp + ratio, the precision helps define how much smoothing/low-pass + filtering is required. The link time can be either reset on startup + or reported as is (the latter being useful to compare progress of + different streams - but may require the wallclock to be always + running and not wrap-around during idle periods). If supported in + hardware, the absolute link time could also be used to define a + precise start time (patches WIP) + +- including the delay in the audio timestamp may + counter-intuitively not increase the precision of timestamps, e.g. if a + codec includes variable-latency DSP processing or a chain of + hardware components the delay is typically not known with precision. + +The accuracy is reported in nanosecond units (using an unsigned 32-bit +word), which gives a max precision of 4.29s, more than enough for +audio applications... + +Due to the varied nature of timestamping needs, even for a single +application, the audio_tstamp_config can be changed dynamically. In +the STATUS ioctl, the parameters are read-only and do not allow for +any application selection. To work around this limitation without +impacting legacy applications, a new STATUS_EXT ioctl is introduced +with read/write parameters. ALSA-lib will be modified to make use of +STATUS_EXT and effectively deprecate STATUS. + +The ALSA API only allows for a single audio timestamp to be reported +at a time. This is a conscious design decision, reading the audio +timestamps from hardware registers or from IPC takes time, the more +timestamps are read the more imprecise the combined measurements +are. To avoid any interpretation issues, a single (system, audio) +timestamp is reported. Applications that need different timestamps +will be required to issue multiple queries and perform an +interpolation of the results + +In some hardware-specific configuration, the system timestamp is +latched by a low-level audio subsytem, and the information provided +back to the driver. Due to potential delays in the communication with +the hardware, there is a risk of misalignment with the avail and delay +information. To make sure applications are not confused, a +driver_timestamp field is added in the snd_pcm_status structure; this +timestamp shows when the information is put together by the driver +before returning from the STATUS and STATUS_EXT ioctl. in most cases +this driver_timestamp will be identical to the regular system tstamp. + +Examples of typestamping with HDaudio: + +1. DMA timestamp, no compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 +playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662 +playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837 +playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420 +playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051 +playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751 +playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822 + +2. DMA timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 -d +playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153 +playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947 +playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685 +playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349 +playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694 + +3. link timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=2 -d +playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787 +playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801 +playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591 +playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779 +playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687 +playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146 + +Example 1 shows that the timestamp at the DMA level is close to 1ms +ahead of the actual playback time (as a side time this sort of +measurement can help define rewind safeguards). Compensating for the +DMA-link delay in example 2 helps remove the hardware buffering abut +the information is still very jittery, with up to one sample of +error. In example 3 where the timestamps are measured with the link +wallclock, the timestamps show a monotonic behavior and a lower +dispersion. + +Example 3 and 4 are with USB audio class. Example 3 shows a high +offset between audio time and system time due to buffering. Example 4 +shows how compensating for the delay exposes a 1ms accuracy (due to +the use of the frame counter by the driver) + +Example 3: DMA timestamp, no compensation for delay, delta of ~5ms +$ ./audio_time -p -Dhw:1 -t1 +playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981 +playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864 +playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912 +playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935 +playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821 +playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259 +playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664 + +Example 4: DMA timestamp, compensation for delay, delay of ~1ms +$ ./audio_time -p -Dhw:1 -t1 -d +playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520 +playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740 +playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081 +playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907 +playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824 +playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847 diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c0ddb7e69c2863..60f0e48f79056d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -60,6 +60,9 @@ struct snd_pcm_hardware { struct snd_pcm_substream; +struct snd_pcm_audio_tstamp_config; /* definitions further down */ +struct snd_pcm_audio_tstamp_report; + struct snd_pcm_ops { int (*open)(struct snd_pcm_substream *substream); int (*close)(struct snd_pcm_substream *substream); @@ -281,6 +284,58 @@ struct snd_pcm_hw_constraint_ranges { struct snd_pcm_hwptr_log; +/* + * userspace-provided audio timestamp config to kernel, + * structure is for internal use only and filled with dedicated unpack routine + */ +struct snd_pcm_audio_tstamp_config { + /* 5 of max 16 bits used */ + u32 type_requested:4; + u32 report_delay:1; /* add total delay to A/D or D/A */ +}; + +static inline void snd_pcm_unpack_audio_tstamp_config(__u32 data, + struct snd_pcm_audio_tstamp_config *config) +{ + config->type_requested = data & 0xF; + config->report_delay = (data >> 4) & 1; +} + +/* + * kernel-provided audio timestamp report to user-space + * structure is for internal use only and read by dedicated pack routine + */ +struct snd_pcm_audio_tstamp_report { + /* 6 of max 16 bits used for bit-fields */ + + /* for backwards compatibility */ + u32 valid:1; + + /* actual type if hardware could not support requested timestamp */ + u32 actual_type:4; + + /* accuracy represented in ns units */ + u32 accuracy_report:1; /* 0 if accuracy unknown, 1 if accuracy field is valid */ + u32 accuracy; /* up to 4.29s, will be packed in separate field */ +}; + +static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy, + const struct snd_pcm_audio_tstamp_report *report) +{ + u32 tmp; + + tmp = report->accuracy_report; + tmp <<= 4; + tmp |= report->actual_type; + tmp <<= 1; + tmp |= report->valid; + + *data &= 0xffff; /* zero-clear MSBs */ + *data |= (tmp << 16); + *accuracy = report->accuracy; +} + + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -361,6 +416,11 @@ struct snd_pcm_runtime { struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + /* -- audio timestamp config -- */ + struct snd_pcm_audio_tstamp_config audio_tstamp_config; + struct snd_pcm_audio_tstamp_report audio_tstamp_report; + struct timespec driver_tstamp; + #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 0e88e7a0f0ebf2..acef4e4d2735ec 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -267,10 +267,17 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ -#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ +#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ + #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ + + typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ #define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */ @@ -408,6 +415,22 @@ struct snd_pcm_channel_info { unsigned int step; /* samples distance in bits */ }; +enum { + /* + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + + /* timestamp definitions */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +}; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ @@ -419,9 +442,11 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - __u32 reserved_alignment; /* must be filled with zero */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */ + struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */ + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -534,6 +559,7 @@ enum { #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) #define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) #define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) From 38ca2a4d58bbc45973ee5cd14e4b803ee5ad69f0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:04 -0600 Subject: [PATCH 026/411] ALSA: core: pass audio tstamp config from userspace Let userspace select audio timestamp config when the STATUS_EXT ioctl is used, ignore and zero all other fields No change for the existing STATUS ioctl, parameters are treated as read-only. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b03a638b420c18..72323a8c35c85e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -753,12 +753,21 @@ int snd_pcm_status(struct snd_pcm_substream *substream, } static int snd_pcm_status_user(struct snd_pcm_substream *substream, - struct snd_pcm_status __user * _status) + struct snd_pcm_status __user * _status, + bool ext) { struct snd_pcm_status status; int res; - + memset(&status, 0, sizeof(status)); + /* + * with extension, parameters are read/write, + * get audio_tstamp_data from user, + * ignore rest of status structure + */ + if (ext && get_user(status.audio_tstamp_data, + (u32 __user *)(&_status->audio_tstamp_data))) + return -EFAULT; res = snd_pcm_status(substream, &status); if (res < 0) return res; @@ -2723,7 +2732,9 @@ static int snd_pcm_common_ioctl1(struct file *file, case SNDRV_PCM_IOCTL_SW_PARAMS: return snd_pcm_sw_params_user(substream, arg); case SNDRV_PCM_IOCTL_STATUS: - return snd_pcm_status_user(substream, arg); + return snd_pcm_status_user(substream, arg, false); + case SNDRV_PCM_IOCTL_STATUS_EXT: + return snd_pcm_status_user(substream, arg, true); case SNDRV_PCM_IOCTL_CHANNEL_INFO: return snd_pcm_channel_info_user(substream, arg); case SNDRV_PCM_IOCTL_PREPARE: From 5442a73a009231598fb5ea065c4e3f9daa30d8cc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:05 -0600 Subject: [PATCH 027/411] ALSA: core: pass audio tstamp config from userspace in compat mode Let userspace select audio timestamp config, ignore and zero all other fields Use audio_tstamp_data to retrieve config and pass report back to user space Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/core/pcm_compat.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 2d957ba6355787..b48b434444ed0e 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -194,18 +194,30 @@ struct snd_pcm_status32 { u32 avail_max; u32 overrange; s32 suspended_state; - u32 reserved_alignment; + u32 audio_tstamp_data; struct compat_timespec audio_tstamp; - unsigned char reserved[56-sizeof(struct compat_timespec)]; + struct compat_timespec driver_tstamp; + u32 audio_tstamp_accuracy; + unsigned char reserved[52-2*sizeof(struct compat_timespec)]; } __attribute__((packed)); static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, - struct snd_pcm_status32 __user *src) + struct snd_pcm_status32 __user *src, + bool ext) { struct snd_pcm_status status; int err; + memset(&status, 0, sizeof(status)); + /* + * with extension, parameters are read/write, + * get audio_tstamp_data from user, + * ignore rest of status structure + */ + if (ext && get_user(status.audio_tstamp_data, + (u32 __user *)(&src->audio_tstamp_data))) + return -EFAULT; err = snd_pcm_status(substream, &status); if (err < 0) return err; @@ -222,7 +234,10 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, put_user(status.avail_max, &src->avail_max) || put_user(status.overrange, &src->overrange) || put_user(status.suspended_state, &src->suspended_state) || - compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp)) + put_user(status.audio_tstamp_data, &src->audio_tstamp_data) || + compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp) || + compat_put_timespec(&status.driver_tstamp, &src->driver_tstamp) || + put_user(status.audio_tstamp_accuracy, &src->audio_tstamp_accuracy)) return -EFAULT; return err; @@ -457,6 +472,7 @@ enum { SNDRV_PCM_IOCTL_HW_PARAMS32 = _IOWR('A', 0x11, struct snd_pcm_hw_params32), SNDRV_PCM_IOCTL_SW_PARAMS32 = _IOWR('A', 0x13, struct snd_pcm_sw_params32), SNDRV_PCM_IOCTL_STATUS32 = _IOR('A', 0x20, struct snd_pcm_status32), + SNDRV_PCM_IOCTL_STATUS_EXT32 = _IOWR('A', 0x24, struct snd_pcm_status32), SNDRV_PCM_IOCTL_DELAY32 = _IOR('A', 0x21, s32), SNDRV_PCM_IOCTL_CHANNEL_INFO32 = _IOR('A', 0x32, struct snd_pcm_channel_info32), SNDRV_PCM_IOCTL_REWIND32 = _IOW('A', 0x46, u32), @@ -517,7 +533,9 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case SNDRV_PCM_IOCTL_SW_PARAMS32: return snd_pcm_ioctl_sw_params_compat(substream, argp); case SNDRV_PCM_IOCTL_STATUS32: - return snd_pcm_status_user_compat(substream, argp); + return snd_pcm_status_user_compat(substream, argp, false); + case SNDRV_PCM_IOCTL_STATUS_EXT32: + return snd_pcm_status_user_compat(substream, argp, true); case SNDRV_PCM_IOCTL_SYNC_PTR32: return snd_pcm_ioctl_sync_ptr_compat(substream, argp); case SNDRV_PCM_IOCTL_CHANNEL_INFO32: From 3179f62001880e588e229db3006a59ad87b7792a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:06 -0600 Subject: [PATCH 028/411] ALSA: core: add .get_time_info Introduce more generic .get_time_info to retrieve system timestamp and audio timestamp in single routine. Backwards compatibility is preserved with same functionality as with .wall_clock method (to be removed in following commits to avoid breaking git bisect) Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ++ sound/core/pcm_lib.c | 88 +++++++++++++++++++++++++++-------------- sound/core/pcm_native.c | 24 +++++++++++ 3 files changed, 87 insertions(+), 29 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 60f0e48f79056d..04f2d492ae57f3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -76,6 +76,10 @@ struct snd_pcm_ops { snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); int (*wall_clock)(struct snd_pcm_substream *substream, struct timespec *audio_ts); + int (*get_time_info)(struct snd_pcm_substream *substream, + struct timespec *system_ts, struct timespec *audio_ts, + struct snd_pcm_audio_tstamp_config *audio_tstamp_config, + struct snd_pcm_audio_tstamp_report *audio_tstamp_report); int (*copy)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ffd656012ab8cf..ac6b33f3779c2e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -232,6 +232,49 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return 0; } +static void update_audio_tstamp(struct snd_pcm_substream *substream, + struct timespec *curr_tstamp, + struct timespec *audio_tstamp) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + u64 audio_frames, audio_nsecs; + struct timespec driver_tstamp; + + if (runtime->tstamp_mode != SNDRV_PCM_TSTAMP_ENABLE) + return; + + if (!(substream->ops->get_time_info) || + (runtime->audio_tstamp_report.actual_type == + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) { + + /* + * provide audio timestamp derived from pointer position + * add delay only if requested + */ + + audio_frames = runtime->hw_ptr_wrap + runtime->status->hw_ptr; + + if (runtime->audio_tstamp_config.report_delay) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + audio_frames -= runtime->delay; + else + audio_frames += runtime->delay; + } + audio_nsecs = div_u64(audio_frames * 1000000000LL, + runtime->rate); + *audio_tstamp = ns_to_timespec(audio_nsecs); + } + runtime->status->audio_tstamp = *audio_tstamp; + runtime->status->tstamp = *curr_tstamp; + + /* + * re-take a driver timestamp to let apps detect if the reference tstamp + * read by low-level hardware was provided with a delay + */ + snd_pcm_gettime(substream->runtime, (struct timespec *)&driver_tstamp); + runtime->driver_tstamp = driver_tstamp; +} + static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned int in_interrupt) { @@ -256,11 +299,18 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, pos = substream->ops->pointer(substream); curr_jiffies = jiffies; if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { - snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); - - if ((runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) && - (substream->ops->wall_clock)) - substream->ops->wall_clock(substream, &audio_tstamp); + if ((substream->ops->get_time_info) && + (runtime->audio_tstamp_config.type_requested != SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) { + substream->ops->get_time_info(substream, &curr_tstamp, + &audio_tstamp, + &runtime->audio_tstamp_config, + &runtime->audio_tstamp_report); + + /* re-test in case tstamp type is not supported in hardware and was demoted to DEFAULT */ + if (runtime->audio_tstamp_report.actual_type == SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT) + snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); + } else + snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); } if (pos == SNDRV_PCM_POS_XRUN) { @@ -403,8 +453,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } no_delta_check: - if (runtime->status->hw_ptr == new_hw_ptr) + if (runtime->status->hw_ptr == new_hw_ptr) { + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return 0; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) @@ -426,30 +478,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_BUG_ON(crossed_boundary != 1); runtime->hw_ptr_wrap += runtime->boundary; } - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { - runtime->status->tstamp = curr_tstamp; - if (!(runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)) { - /* - * no wall clock available, provide audio timestamp - * derived from pointer position+delay - */ - u64 audio_frames, audio_nsecs; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - audio_frames = runtime->hw_ptr_wrap - + runtime->status->hw_ptr - - runtime->delay; - else - audio_frames = runtime->hw_ptr_wrap - + runtime->status->hw_ptr - + runtime->delay; - audio_nsecs = div_u64(audio_frames * 1000000000LL, - runtime->rate); - audio_tstamp = ns_to_timespec(audio_nsecs); - } - runtime->status->audio_tstamp = audio_tstamp; - } + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return snd_pcm_update_state(substream, runtime); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 72323a8c35c85e..9c2c6f801fed2f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -707,6 +707,23 @@ int snd_pcm_status(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_stream_lock_irq(substream); + + snd_pcm_unpack_audio_tstamp_config(status->audio_tstamp_data, + &runtime->audio_tstamp_config); + + /* backwards compatible behavior */ + if (runtime->audio_tstamp_config.type_requested == + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT) { + if (runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) + runtime->audio_tstamp_config.type_requested = + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK; + else + runtime->audio_tstamp_config.type_requested = + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; + runtime->audio_tstamp_report.valid = 0; + } else + runtime->audio_tstamp_report.valid = 1; + status->state = runtime->status->state; status->suspended_state = runtime->status->suspended_state; if (status->state == SNDRV_PCM_STATE_OPEN) @@ -716,8 +733,15 @@ int snd_pcm_status(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { status->tstamp = runtime->status->tstamp; + status->driver_tstamp = runtime->driver_tstamp; status->audio_tstamp = runtime->status->audio_tstamp; + if (runtime->audio_tstamp_report.valid == 1) + /* backwards compatibility, no report provided in COMPAT mode */ + snd_pcm_pack_audio_tstamp_report(&status->audio_tstamp_data, + &status->audio_tstamp_accuracy, + &runtime->audio_tstamp_report); + goto _tstamp_end; } } else { From 9e94df3a624b1b485f2c2ac5ab94032da01e45b3 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:07 -0600 Subject: [PATCH 029/411] ALSA: hda: replace .wallclock by .get_time_info No real functional change, only take wall clock and system time in same routine and add accuracy report. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 40 ++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index dfcb5e929f9fc9..7806f1d297cb31 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -732,17 +732,32 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) azx_get_position(chip, azx_dev)); } -static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, - struct timespec *ts) +static int azx_get_time_info(struct snd_pcm_substream *substream, + struct timespec *system_ts, struct timespec *audio_ts, + struct snd_pcm_audio_tstamp_config *audio_tstamp_config, + struct snd_pcm_audio_tstamp_report *audio_tstamp_report) { struct azx_dev *azx_dev = get_azx_dev(substream); u64 nsec; - nsec = timecounter_read(&azx_dev->azx_tc); - nsec = div_u64(nsec, 3); /* can be optimized */ - nsec = azx_adjust_codec_delay(substream, nsec); + if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) && + (audio_tstamp_config->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK)) { - *ts = ns_to_timespec(nsec); + snd_pcm_gettime(substream->runtime, system_ts); + + nsec = timecounter_read(&azx_dev->azx_tc); + nsec = div_u64(nsec, 3); /* can be optimized */ + if (audio_tstamp_config->report_delay) + nsec = azx_adjust_codec_delay(substream, nsec); + + *audio_ts = ns_to_timespec(nsec); + + audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK; + audio_tstamp_report->accuracy_report = 1; /* rest of structure is valid */ + audio_tstamp_report->accuracy = 42; /* 24 MHz WallClock == 42ns resolution */ + + } else + audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; return 0; } @@ -756,7 +771,8 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_HAS_WALL_CLOCK | + SNDRV_PCM_INFO_HAS_WALL_CLOCK | /* legacy */ + SNDRV_PCM_INFO_HAS_LINK_ATIME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, @@ -842,10 +858,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) return -EINVAL; } - /* disable WALLCLOCK timestamps for capture streams + /* disable LINK_ATIME timestamps for capture streams until we figure out how to handle digital inputs */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; /* legacy */ + runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_LINK_ATIME; + } spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; @@ -877,7 +895,7 @@ static struct snd_pcm_ops azx_pcm_ops = { .prepare = azx_pcm_prepare, .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, - .wall_clock = azx_get_wallclock_tstamp, + .get_time_info = azx_get_time_info, .mmap = azx_pcm_mmap, .page = snd_pcm_sgbuf_ops_page, }; From 2d52a5abd5be35340296a251092c8a594679df54 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:08 -0600 Subject: [PATCH 030/411] ALSA: core: remove .wall_clock can be removed without breaking git-bisect now Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 04f2d492ae57f3..0cb7f3f5df7b9f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -74,8 +74,6 @@ struct snd_pcm_ops { int (*prepare)(struct snd_pcm_substream *substream); int (*trigger)(struct snd_pcm_substream *substream, int cmd); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); - int (*wall_clock)(struct snd_pcm_substream *substream, - struct timespec *audio_ts); int (*get_time_info)(struct snd_pcm_substream *substream, struct timespec *system_ts, struct timespec *audio_ts, struct snd_pcm_audio_tstamp_config *audio_tstamp_config, From c72638bdaabe9ea4b09003b9db7e1754f472fbed Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:09 -0600 Subject: [PATCH 031/411] ALSA: bump PCM protocol to 2.0.13 Bump PCM protocol to enable use of STATUS_EXT ioctls, older apps will still use STATUS and audio timestamp configuration is not supported (backwards compatible behavior). Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index acef4e4d2735ec..3d46e9a0cd2ee6 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -140,7 +140,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; From ad876c862278be59147d4004f1a7c4d492e4ec96 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 16:26:00 +0100 Subject: [PATCH 032/411] ALSA: pcm: Minor refactoring in snd_pcm_attach_substream() No functional changes at all. Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 41 +++++++++++++++-------------------------- 1 file changed, 15 insertions(+), 26 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 0345e53a340ca9..d63d262c571ec5 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -919,6 +919,9 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, if (snd_BUG_ON(!pcm || !rsubstream)) return -ENXIO; + if (snd_BUG_ON(stream != SNDRV_PCM_STREAM_PLAYBACK && + stream != SNDRV_PCM_STREAM_CAPTURE)) + return -EINVAL; *rsubstream = NULL; pstr = &pcm->streams[stream]; if (pstr->substream == NULL || pstr->substream_count == 0) @@ -927,25 +930,14 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, card = pcm->card; prefer_subdevice = snd_ctl_get_preferred_subdevice(card, SND_CTL_SUBDEV_PCM); - switch (stream) { - case SNDRV_PCM_STREAM_PLAYBACK: - if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { - for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; substream; substream = substream->next) { - if (SUBSTREAM_BUSY(substream)) - return -EAGAIN; - } - } - break; - case SNDRV_PCM_STREAM_CAPTURE: - if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { - for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next) { - if (SUBSTREAM_BUSY(substream)) - return -EAGAIN; - } + if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { + int opposite = !stream; + + for (substream = pcm->streams[opposite].substream; substream; + substream = substream->next) { + if (SUBSTREAM_BUSY(substream)) + return -EAGAIN; } - break; - default: - return -EINVAL; } if (file->f_flags & O_APPEND) { @@ -968,15 +960,12 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, return 0; } - if (prefer_subdevice >= 0) { - for (substream = pstr->substream; substream; substream = substream->next) - if (!SUBSTREAM_BUSY(substream) && substream->number == prefer_subdevice) - goto __ok; - } - for (substream = pstr->substream; substream; substream = substream->next) - if (!SUBSTREAM_BUSY(substream)) + for (substream = pstr->substream; substream; substream = substream->next) { + if (!SUBSTREAM_BUSY(substream) && + (prefer_subdevice == -1 || + substream->number == prefer_subdevice)) break; - __ok: + } if (substream == NULL) return -EAGAIN; From b95bd3a454cf9e9e111b6b87c02550368fe6e802 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 16:49:04 +0100 Subject: [PATCH 033/411] ALSA: pcm: Don't add internal PCMs to PCM device list An internal PCM object shouldn't be added to the PCM device list, as it's never accessed directly from the user-space, and it has no proc or any similar accesses. Currently, it's excluded in snd_pcm_get() and snd_pcm_next(), but it's easier not to add such an object to the list. Actually, the whole snd_pcm_dev_register() can be skipped for an internal PCM. So this patch changes the code there, but also addresses the uninitialized list_head access. Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index d63d262c571ec5..d440629e08e2da 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -49,8 +49,6 @@ static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { - if (pcm->internal) - continue; if (pcm->card == card && pcm->device == device) return pcm; } @@ -62,8 +60,6 @@ static int snd_pcm_next(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { - if (pcm->internal) - continue; if (pcm->card == card && pcm->device > device) return pcm->device; else if (pcm->card->number > card->number) @@ -76,6 +72,9 @@ static int snd_pcm_add(struct snd_pcm *newpcm) { struct snd_pcm *pcm; + if (newpcm->internal) + return 0; + list_for_each_entry(pcm, &snd_pcm_devices, list) { if (pcm->card == newpcm->card && pcm->device == newpcm->device) return -EBUSY; @@ -782,6 +781,9 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, pcm->card = card; pcm->device = device; pcm->internal = internal; + mutex_init(&pcm->open_mutex); + init_waitqueue_head(&pcm->open_wait); + INIT_LIST_HEAD(&pcm->list); if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { @@ -792,8 +794,6 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, snd_pcm_free(pcm); return err; } - mutex_init(&pcm->open_mutex); - init_waitqueue_head(&pcm->open_wait); if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { snd_pcm_free(pcm); return err; @@ -1075,15 +1075,16 @@ static int snd_pcm_dev_register(struct snd_device *device) if (snd_BUG_ON(!device || !device->device_data)) return -ENXIO; pcm = device->device_data; + if (pcm->internal) + return 0; + mutex_lock(®ister_mutex); err = snd_pcm_add(pcm); - if (err) { - mutex_unlock(®ister_mutex); - return err; - } + if (err) + goto unlock; for (cidx = 0; cidx < 2; cidx++) { int devtype = -1; - if (pcm->streams[cidx].substream == NULL || pcm->internal) + if (pcm->streams[cidx].substream == NULL) continue; switch (cidx) { case SNDRV_PCM_STREAM_PLAYBACK: @@ -1098,9 +1099,8 @@ static int snd_pcm_dev_register(struct snd_device *device) &snd_pcm_f_ops[cidx], pcm, &pcm->streams[cidx].dev); if (err < 0) { - list_del(&pcm->list); - mutex_unlock(®ister_mutex); - return err; + list_del_init(&pcm->list); + goto unlock; } for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) @@ -1110,8 +1110,9 @@ static int snd_pcm_dev_register(struct snd_device *device) list_for_each_entry(notify, &snd_pcm_notify_list, list) notify->n_register(pcm); + unlock: mutex_unlock(®ister_mutex); - return 0; + return err; } static int snd_pcm_dev_disconnect(struct snd_device *device) From 646e1dd8f9f47c57560ce81c02fdd57ff0929bc6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 17:04:08 +0100 Subject: [PATCH 034/411] ALSA: pcm: Don't notify internal PCMs Notifier shouldn't listen to the changes of internal PCMs. Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index d440629e08e2da..542dbc6f54e822 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -888,8 +888,9 @@ static int snd_pcm_free(struct snd_pcm *pcm) if (!pcm) return 0; - list_for_each_entry(notify, &snd_pcm_notify_list, list) { - notify->n_unregister(pcm); + if (!pcm->internal) { + list_for_each_entry(notify, &snd_pcm_notify_list, list) + notify->n_unregister(pcm); } if (pcm->private_free) pcm->private_free(pcm); @@ -1129,7 +1130,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) mutex_lock(&pcm->open_mutex); wake_up(&pcm->open_wait); list_del_init(&pcm->list); - for (cidx = 0; cidx < 2; cidx++) + for (cidx = 0; cidx < 2; cidx++) { for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) { snd_pcm_stream_lock_irq(substream); if (substream->runtime) { @@ -1139,8 +1140,10 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) } snd_pcm_stream_unlock_irq(substream); } - list_for_each_entry(notify, &snd_pcm_notify_list, list) { - notify->n_disconnect(pcm); + } + if (!pcm->internal) { + list_for_each_entry(notify, &snd_pcm_notify_list, list) + notify->n_disconnect(pcm); } for (cidx = 0; cidx < 2; cidx++) { snd_unregister_device(&pcm->streams[cidx].dev); From b20221385c40155f13068be75b865170a1ad5d1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 17:05:27 +0100 Subject: [PATCH 035/411] ALSA: pcm: Don't ignore internal PCMs in snd_pcm_dev_disconnect() Some codes in snd_pcm_dev_disconnect() are still valid even for internal PCMs, but they are skipped because of the check of list_empty(&pcm->list) at the beginning. Remove this check and put pcm->internal checks appropriately for internal PCM object to process through this function. Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 542dbc6f54e822..e9b87465c73da6 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1124,9 +1124,6 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) int cidx; mutex_lock(®ister_mutex); - if (list_empty(&pcm->list)) - goto unlock; - mutex_lock(&pcm->open_mutex); wake_up(&pcm->open_wait); list_del_init(&pcm->list); @@ -1146,14 +1143,14 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) notify->n_disconnect(pcm); } for (cidx = 0; cidx < 2; cidx++) { - snd_unregister_device(&pcm->streams[cidx].dev); + if (!pcm->internal) + snd_unregister_device(&pcm->streams[cidx].dev); if (pcm->streams[cidx].chmap_kctl) { snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl); pcm->streams[cidx].chmap_kctl = NULL; } } mutex_unlock(&pcm->open_mutex); - unlock: mutex_unlock(®ister_mutex); return 0; } From 7371bd1f4aeb4e1c44b8c1ba8e36ebba4250b59c Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 17 Feb 2015 13:59:26 +0800 Subject: [PATCH 036/411] ASoC: rt5670: Add disabled item in dmic pin enum Currently, we will configure dmic related pin definition if pdata.dmic_en is true. However, there is no disable option in the enum. So, any dmic is used, all 3 dmic related pins will be configured. It may cause unexpected pin definition. This patch adds a disable item for each dmic enum and take it as default. So the driver will not set the pin configuration if we don't set dmicx_data_pin in platform data. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index d11b9c207e2679..84857bdaa78bff 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1967,17 +1967,20 @@ enum { }; enum { + RT5670_DMIC1_DISABLED, RT5670_DMIC_DATA_GPIO6, RT5670_DMIC_DATA_IN2P, RT5670_DMIC_DATA_GPIO7, }; enum { + RT5670_DMIC2_DISABLED, RT5670_DMIC_DATA_GPIO8, RT5670_DMIC_DATA_IN3N, }; enum { + RT5670_DMIC3_DISABLED, RT5670_DMIC_DATA_GPIO9, RT5670_DMIC_DATA_GPIO10, RT5670_DMIC_DATA_GPIO5, From 65d17a9ce9f24a3aaf7d614251fdcc1b2121765f Mon Sep 17 00:00:00 2001 From: Nikesh Oswal Date: Mon, 16 Feb 2015 15:25:48 +0000 Subject: [PATCH 037/411] ASoC: wm_adsp: Ensure DSP controls are always persistent Currently DSP controls are persistent (across DSP On/Off) only if they were set whilst the DSP is off. This change makes the controls persistent irrespective of when they are set. Signed-off-by: Nikesh Oswal Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 720d6e852986c3..14414ea23b555b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -420,10 +420,9 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, memcpy(ctl->cache, p, ctl->len); - if (!ctl->enabled) { - ctl->set = 1; + ctl->set = 1; + if (!ctl->enabled) return 0; - } return wm_coeff_write_control(kcontrol, p, ctl->len); } From be951017453cba2f3eb789413f697b8f14393eec Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 16 Feb 2015 15:25:49 +0000 Subject: [PATCH 038/411] ASoC: wm_adsp: Improve round to next 4-byte boundary Whilst the existing code does correctly round to the next 4-byte boundary it does so rather inefficiently. This patch changes the rounding to be simpler and more efficient. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 14414ea23b555b..e625cedb0fa962 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1184,7 +1184,6 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) int ret, pos, blocks, type, offset, reg; char *file; struct wm_adsp_buf *buf; - int tmp; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -1334,12 +1333,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } } - tmp = le32_to_cpu(blk->len) % 4; - if (tmp) - pos += le32_to_cpu(blk->len) + (4 - tmp) + sizeof(*blk); - else - pos += le32_to_cpu(blk->len) + sizeof(*blk); - + pos += (le32_to_cpu(blk->len) + sizeof(*blk) + 3) & ~0x03; blocks++; } From 7ff5eabce4231d199dadc14c23f14a6619f926c0 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 17 Feb 2015 00:53:12 -0800 Subject: [PATCH 039/411] ASoC: max98357a: Remove use of DRV_NAME Remove use of DRV_NAME define. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index e9e6efbc21dd52..4ee23fbc4e1237 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -26,8 +26,6 @@ #include #include -#define DRV_NAME "max98357a" - static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -87,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "max98357a", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "max98357a-playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -127,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -135,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -145,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); From 327ef4f02582d01f7eedb291794106823b44a0cf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 17:15:04 +0100 Subject: [PATCH 040/411] ALSA: hda - Decouple PCM and hwdep devices from codec object This is a preliminary patch for the hda_bus implementation, removing the parent device setup to codec device. Since the bus and the class devices can't be crossed over, leave the sound devices to the default parent device as is. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 - sound/pci/hda/hda_controller.c | 3 --- sound/pci/hda/hda_hwdep.c | 3 --- 3 files changed, 7 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 1e7de08e77cbf4..d6be4e852c4d5c 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -168,7 +168,6 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->evbit[0] = BIT_MASK(EV_SND); input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); input_dev->event = snd_hda_beep_event; - input_dev->dev.parent = &codec->dev; input_set_drvdata(input_dev, beep); beep->dev = input_dev; diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ebb7a644bd8667..4c7a6f9bfcde04 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -958,9 +958,6 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, chip->card->dev, size, MAX_PREALLOC_SIZE); - /* link to codec */ - for (s = 0; s < 2; s++) - pcm->streams[s].dev.parent = &codec->dev; return 0; } diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 11b5a42b4ec8cf..125f3420fa6ae1 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -116,9 +116,6 @@ int snd_hda_create_hwdep(struct hda_codec *codec) hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat; #endif - /* link to codec */ - hwdep->dev.parent = &codec->dev; - /* for sysfs */ hwdep->dev.groups = snd_hda_dev_attr_groups; dev_set_drvdata(&hwdep->dev, codec); From d8a766a16ed90c4b3bd7afa6e1417f8d715db507 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 15:25:37 +0100 Subject: [PATCH 041/411] ALSA: hda - Bind codecs via standard bus Now we create the standard HD-audio bus (/sys/bus/hdaudio), and bind the codec driver with the codec device over there. This is the first step of the whole transition so that the changes to each codec driver are kept as minimal as possible. Each codec driver needs to register hda_codec_driver struct containing the currently existing preset via the new helper macro module_hda_codec_driver(). The old hda_codec_preset_list is replaced with this infrastructure. The generic parsers (for HDMI and other) are also included in the preset with the special IDs to bind uniquely. In HD-audio core side, the device binding code is split to hda_bind.c. It provides the snd_hda_bus_type implementation to match the codec driver with the given codec vendor ID. It also manages the module auto-loading by itself like before: when the matching isn't found, it tries to probe the corresponding codec modules, and finally falls back to the generic drivers. (The special ID mentioned above is set at this stage.) The only visible change to outside is that the hdaudio sysfs entry now appears in /sys/bus/devices, not as a sound class device. More works to move the suspend/resume and remove ops will be (hopefully) done in later patches. Signed-off-by: Takashi Iwai --- sound/pci/hda/Makefile | 2 +- sound/pci/hda/hda_bind.c | 320 +++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.c | 307 ++----------------------------- sound/pci/hda/hda_codec.h | 27 ++- sound/pci/hda/hda_generic.c | 18 +- sound/pci/hda/hda_local.h | 18 +- sound/pci/hda/patch_analog.c | 16 +- sound/pci/hda/patch_ca0110.c | 16 +- sound/pci/hda/patch_ca0132.c | 16 +- sound/pci/hda/patch_cirrus.c | 16 +- sound/pci/hda/patch_cmedia.c | 16 +- sound/pci/hda/patch_conexant.c | 16 +- sound/pci/hda/patch_hdmi.c | 27 +-- sound/pci/hda/patch_realtek.c | 16 +- sound/pci/hda/patch_si3054.c | 16 +- sound/pci/hda/patch_sigmatel.c | 16 +- sound/pci/hda/patch_via.c | 16 +- 17 files changed, 401 insertions(+), 478 deletions(-) create mode 100644 sound/pci/hda/hda_bind.c diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 194f30935e776f..96caaebfc19d25 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -4,7 +4,7 @@ snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o -snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o +snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c new file mode 100644 index 00000000000000..adf6b475dee135 --- /dev/null +++ b/sound/pci/hda/hda_bind.c @@ -0,0 +1,320 @@ +/* + * HD-audio codec driver binding + * Copyright (c) Takashi Iwai + */ + +#include +#include +#include +#include +#include +#include +#include "hda_codec.h" +#include "hda_local.h" + +/* codec vendor labels */ +struct hda_vendor_id { + unsigned int id; + const char *name; +}; + +static struct hda_vendor_id hda_vendor_ids[] = { + { 0x1002, "ATI" }, + { 0x1013, "Cirrus Logic" }, + { 0x1057, "Motorola" }, + { 0x1095, "Silicon Image" }, + { 0x10de, "Nvidia" }, + { 0x10ec, "Realtek" }, + { 0x1102, "Creative" }, + { 0x1106, "VIA" }, + { 0x111d, "IDT" }, + { 0x11c1, "LSI" }, + { 0x11d4, "Analog Devices" }, + { 0x13f6, "C-Media" }, + { 0x14f1, "Conexant" }, + { 0x17e8, "Chrontel" }, + { 0x1854, "LG" }, + { 0x1aec, "Wolfson Microelectronics" }, + { 0x1af4, "QEMU" }, + { 0x434d, "C-Media" }, + { 0x8086, "Intel" }, + { 0x8384, "SigmaTel" }, + {} /* terminator */ +}; + +/* + * find a matching codec preset + */ +static int hda_bus_match(struct device *dev, struct device_driver *drv) +{ + struct hda_codec *codec = container_of(dev, struct hda_codec, dev); + struct hda_codec_driver *driver = + container_of(drv, struct hda_codec_driver, driver); + const struct hda_codec_preset *preset; + /* check probe_id instead of vendor_id if set */ + u32 id = codec->probe_id ? codec->probe_id : codec->vendor_id; + + for (preset = driver->preset; preset->id; preset++) { + u32 mask = preset->mask; + + if (preset->afg && preset->afg != codec->afg) + continue; + if (preset->mfg && preset->mfg != codec->mfg) + continue; + if (!mask) + mask = ~0; + if (preset->id == (id & mask) && + (!preset->rev || preset->rev == codec->revision_id)) { + codec->preset = preset; + return 1; + } + } + return 0; +} + +/* reset the codec name from the preset */ +static int codec_refresh_name(struct hda_codec *codec, const char *name) +{ + char tmp[16]; + + kfree(codec->chip_name); + if (!name) { + sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); + name = tmp; + } + codec->chip_name = kstrdup(name, GFP_KERNEL); + return codec->chip_name ? 0 : -ENOMEM; +} + +static int hda_codec_driver_probe(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + struct module *owner = dev->driver->owner; + int err; + + if (WARN_ON(!codec->preset)) + return -EINVAL; + + err = codec_refresh_name(codec, codec->preset->name); + if (err < 0) + goto error; + + if (!try_module_get(owner)) { + err = -EINVAL; + goto error; + } + + err = codec->preset->patch(codec); + if (err < 0) { + module_put(owner); + goto error; + } + + return 0; + + error: + codec->preset = NULL; + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + return err; +} + +static int hda_codec_driver_remove(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + + if (codec->patch_ops.free) + codec->patch_ops.free(codec); + codec->preset = NULL; + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + module_put(dev->driver->owner); + return 0; +} + +int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, + struct module *owner) +{ + drv->driver.name = name; + drv->driver.owner = owner; + drv->driver.bus = &snd_hda_bus_type; + drv->driver.probe = hda_codec_driver_probe; + drv->driver.remove = hda_codec_driver_remove; + /* TODO: PM and others */ + return driver_register(&drv->driver); +} +EXPORT_SYMBOL_GPL(__hda_codec_driver_register); + +void hda_codec_driver_unregister(struct hda_codec_driver *drv) +{ + driver_unregister(&drv->driver); +} +EXPORT_SYMBOL_GPL(hda_codec_driver_unregister); + +static inline bool codec_probed(struct hda_codec *codec) +{ + return device_attach(hda_codec_dev(codec)) > 0 && codec->preset; +} + +/* try to auto-load and bind the codec module */ +static void codec_bind_module(struct hda_codec *codec) +{ +#ifdef MODULE + request_module("snd-hda-codec-id:%08x", codec->vendor_id); + if (codec_probed(codec)) + return; + request_module("snd-hda-codec-id:%04x*", + (codec->vendor_id >> 16) & 0xffff); + if (codec_probed(codec)) + return; +#endif +} + +/* store the codec vendor name */ +static int get_codec_vendor_name(struct hda_codec *codec) +{ + const struct hda_vendor_id *c; + const char *vendor = NULL; + u16 vendor_id = codec->vendor_id >> 16; + char tmp[16]; + + for (c = hda_vendor_ids; c->id; c++) { + if (c->id == vendor_id) { + vendor = c->name; + break; + } + } + if (!vendor) { + sprintf(tmp, "Generic %04x", vendor_id); + vendor = tmp; + } + codec->vendor_name = kstrdup(vendor, GFP_KERNEL); + if (!codec->vendor_name) + return -ENOMEM; + return 0; +} + +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) +/* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ +static bool is_likely_hdmi_codec(struct hda_codec *codec) +{ + hda_nid_t nid = codec->start_nid; + int i; + + for (i = 0; i < codec->num_nodes; i++, nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + switch (get_wcaps_type(wcaps)) { + case AC_WID_AUD_IN: + return false; /* HDMI parser supports only HDMI out */ + case AC_WID_AUD_OUT: + if (!(wcaps & AC_WCAP_DIGITAL)) + return false; + break; + } + } + return true; +} +#else +/* no HDMI codec parser support */ +#define is_likely_hdmi_codec(codec) false +#endif /* CONFIG_SND_HDA_CODEC_HDMI */ + +static int codec_bind_generic(struct hda_codec *codec) +{ + if (codec->probe_id) + return -ENODEV; + + if (is_likely_hdmi_codec(codec)) { + codec->probe_id = HDA_CODEC_ID_GENERIC_HDMI; +#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) + request_module("snd-hda-codec-hdmi"); +#endif + if (codec_probed(codec)) + return 0; + } + + codec->probe_id = HDA_CODEC_ID_GENERIC; +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) + request_module("snd-hda-codec-generic"); +#endif + if (codec_probed(codec)) + return 0; + return -ENODEV; +} + +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) +#define is_generic_config(codec) \ + (codec->modelname && !strcmp(codec->modelname, "generic")) +#else +#define is_generic_config(codec) 0 +#endif + +/** + * snd_hda_codec_configure - (Re-)configure the HD-audio codec + * @codec: the HDA codec + * + * Start parsing of the given codec tree and (re-)initialize the whole + * patch instance. + * + * Returns 0 if successful or a negative error code. + */ +int snd_hda_codec_configure(struct hda_codec *codec) +{ + int err; + + if (!codec->vendor_name) { + err = get_codec_vendor_name(codec); + if (err < 0) + return err; + } + + if (is_generic_config(codec)) + codec->probe_id = HDA_CODEC_ID_GENERIC; + else + codec->probe_id = 0; + + err = device_add(hda_codec_dev(codec)); + if (err < 0) + return err; + + if (!codec->preset) + codec_bind_module(codec); + if (!codec->preset) { + err = codec_bind_generic(codec); + if (err < 0) { + codec_err(codec, "Unable to bind the codec\n"); + goto error; + } + } + + /* audio codec should override the mixer name */ + if (codec->afg || !*codec->bus->card->mixername) + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + return 0; + + error: + device_del(hda_codec_dev(codec)); + return err; +} +EXPORT_SYMBOL_GPL(snd_hda_codec_configure); + +/* + * bus registration + */ +struct bus_type snd_hda_bus_type = { + .name = "hdaudio", + .match = hda_bus_match, +}; + +static int __init hda_codec_init(void) +{ + return bus_register(&snd_hda_bus_type); +} + +static void __exit hda_codec_exit(void) +{ + bus_unregister(&snd_hda_bus_type); +} + +module_init(hda_codec_init); +module_exit(hda_codec_exit); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5e755eb6f19a0b..61c8174e8013c4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -40,69 +40,6 @@ #define CREATE_TRACE_POINTS #include "hda_trace.h" -/* - * vendor / preset table - */ - -struct hda_vendor_id { - unsigned int id; - const char *name; -}; - -/* codec vendor labels */ -static struct hda_vendor_id hda_vendor_ids[] = { - { 0x1002, "ATI" }, - { 0x1013, "Cirrus Logic" }, - { 0x1057, "Motorola" }, - { 0x1095, "Silicon Image" }, - { 0x10de, "Nvidia" }, - { 0x10ec, "Realtek" }, - { 0x1102, "Creative" }, - { 0x1106, "VIA" }, - { 0x111d, "IDT" }, - { 0x11c1, "LSI" }, - { 0x11d4, "Analog Devices" }, - { 0x13f6, "C-Media" }, - { 0x14f1, "Conexant" }, - { 0x17e8, "Chrontel" }, - { 0x1854, "LG" }, - { 0x1aec, "Wolfson Microelectronics" }, - { 0x1af4, "QEMU" }, - { 0x434d, "C-Media" }, - { 0x8086, "Intel" }, - { 0x8384, "SigmaTel" }, - {} /* terminator */ -}; - -static DEFINE_MUTEX(preset_mutex); -static LIST_HEAD(hda_preset_tables); - -/** - * snd_hda_add_codec_preset - Add a codec preset to the chain - * @preset: codec preset table to add - */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) -{ - mutex_lock(&preset_mutex); - list_add_tail(&preset->list, &hda_preset_tables); - mutex_unlock(&preset_mutex); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_add_codec_preset); - -/** - * snd_hda_delete_codec_preset - Delete a codec preset from the chain - * @preset: codec preset table to delete - */ -int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset) -{ - mutex_lock(&preset_mutex); - list_del(&preset->list); - mutex_unlock(&preset_mutex); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_delete_codec_preset); - #ifdef CONFIG_PM #define codec_in_pm(codec) ((codec)->in_pm) static void hda_power_work(struct work_struct *work); @@ -885,111 +822,6 @@ int snd_hda_bus_new(struct snd_card *card, } EXPORT_SYMBOL_GPL(snd_hda_bus_new); -#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) -#define is_generic_config(codec) \ - (codec->modelname && !strcmp(codec->modelname, "generic")) -#else -#define is_generic_config(codec) 0 -#endif - -#ifdef MODULE -#define HDA_MODREQ_MAX_COUNT 2 /* two request_modules()'s */ -#else -#define HDA_MODREQ_MAX_COUNT 0 /* all presets are statically linked */ -#endif - -/* - * find a matching codec preset - */ -static const struct hda_codec_preset * -find_codec_preset(struct hda_codec *codec) -{ - struct hda_codec_preset_list *tbl; - const struct hda_codec_preset *preset; - unsigned int mod_requested = 0; - - again: - mutex_lock(&preset_mutex); - list_for_each_entry(tbl, &hda_preset_tables, list) { - if (!try_module_get(tbl->owner)) { - codec_err(codec, "cannot module_get\n"); - continue; - } - for (preset = tbl->preset; preset->id; preset++) { - u32 mask = preset->mask; - if (preset->afg && preset->afg != codec->afg) - continue; - if (preset->mfg && preset->mfg != codec->mfg) - continue; - if (!mask) - mask = ~0; - if (preset->id == (codec->vendor_id & mask) && - (!preset->rev || - preset->rev == codec->revision_id)) { - mutex_unlock(&preset_mutex); - codec->owner = tbl->owner; - return preset; - } - } - module_put(tbl->owner); - } - mutex_unlock(&preset_mutex); - - if (mod_requested < HDA_MODREQ_MAX_COUNT) { - if (!mod_requested) - request_module("snd-hda-codec-id:%08x", - codec->vendor_id); - else - request_module("snd-hda-codec-id:%04x*", - (codec->vendor_id >> 16) & 0xffff); - mod_requested++; - goto again; - } - return NULL; -} - -/* - * get_codec_name - store the codec name - */ -static int get_codec_name(struct hda_codec *codec) -{ - const struct hda_vendor_id *c; - const char *vendor = NULL; - u16 vendor_id = codec->vendor_id >> 16; - char tmp[16]; - - if (codec->vendor_name) - goto get_chip_name; - - for (c = hda_vendor_ids; c->id; c++) { - if (c->id == vendor_id) { - vendor = c->name; - break; - } - } - if (!vendor) { - sprintf(tmp, "Generic %04x", vendor_id); - vendor = tmp; - } - codec->vendor_name = kstrdup(vendor, GFP_KERNEL); - if (!codec->vendor_name) - return -ENOMEM; - - get_chip_name: - if (codec->chip_name) - return 0; - - if (codec->preset && codec->preset->name) - codec->chip_name = kstrdup(codec->preset->name, GFP_KERNEL); - else { - sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); - codec->chip_name = kstrdup(tmp, GFP_KERNEL); - } - if (!codec->chip_name) - return -ENOMEM; - return 0; -} - /* * look for an AFG and MFG nodes */ @@ -1300,20 +1132,6 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) return p; } -/* - * Dynamic symbol binding for the codec parsers - */ - -#define load_parser(codec, sym) \ - ((codec)->parser = (int (*)(struct hda_codec *))symbol_request(sym)) - -static void unload_parser(struct hda_codec *codec) -{ - if (codec->parser) - symbol_put_addr(codec->parser); - codec->parser = NULL; -} - /* * codec destructor */ @@ -1322,6 +1140,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) if (!codec) return; cancel_delayed_work_sync(&codec->jackpoll_work); + if (device_is_registered(hda_codec_dev(codec))) + device_del(hda_codec_dev(codec)); snd_hda_jack_tbl_clear(codec); free_init_pincfgs(codec); #ifdef CONFIG_PM @@ -1335,12 +1155,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) snd_array_free(&codec->spdif_out); remove_conn_list(codec); codec->bus->caddr_tbl[codec->addr] = NULL; - if (codec->patch_ops.free) - codec->patch_ops.free(codec); hda_call_pm_notify(codec, false); /* cancel leftover refcounts */ snd_hda_sysfs_clear(codec); - unload_parser(codec); - module_put(codec->owner); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); kfree(codec->vendor_name); @@ -1348,7 +1164,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec->modelname); kfree(codec->wcaps); codec->bus->num_codecs--; - put_device(&codec->dev); + put_device(hda_codec_dev(codec)); } static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, @@ -1360,10 +1176,7 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, static int snd_hda_codec_dev_register(struct snd_device *device) { struct hda_codec *codec = device->device_data; - int err = device_add(&codec->dev); - if (err < 0) - return err; snd_hda_register_beep_device(codec); return 0; } @@ -1373,7 +1186,6 @@ static int snd_hda_codec_dev_disconnect(struct snd_device *device) struct hda_codec *codec = device->device_data; snd_hda_detach_beep_device(codec); - device_del(&codec->dev); return 0; } @@ -1386,7 +1198,7 @@ static int snd_hda_codec_dev_free(struct snd_device *device) /* just free the container */ static void snd_hda_codec_dev_release(struct device *dev) { - kfree(container_of(dev, struct hda_codec, dev)); + kfree(dev_to_hda_codec(dev)); } /** @@ -1402,6 +1214,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct hda_codec **codecp) { struct hda_codec *codec; + struct device *dev; char component[31]; hda_nid_t fg; int err; @@ -1429,14 +1242,14 @@ int snd_hda_codec_new(struct hda_bus *bus, return -ENOMEM; } - device_initialize(&codec->dev); - codec->dev.parent = &bus->card->card_dev; - codec->dev.class = sound_class; - codec->dev.release = snd_hda_codec_dev_release; - codec->dev.groups = snd_hda_dev_attr_groups; - dev_set_name(&codec->dev, "hdaudioC%dD%d", bus->card->number, - codec_addr); - dev_set_drvdata(&codec->dev, codec); /* for sysfs */ + dev = hda_codec_dev(codec); + device_initialize(dev); + dev->parent = bus->card->dev; + dev->bus = &snd_hda_bus_type; + dev->release = snd_hda_codec_dev_release; + dev->groups = snd_hda_dev_attr_groups; + dev_set_name(dev, "hdaudioC%dD%d", bus->card->number, codec_addr); + dev_set_drvdata(dev, codec); /* for sysfs */ codec->bus = bus; codec->addr = codec_addr; @@ -1587,92 +1400,6 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets); - -#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) -/* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ -static bool is_likely_hdmi_codec(struct hda_codec *codec) -{ - hda_nid_t nid = codec->start_nid; - int i; - - for (i = 0; i < codec->num_nodes; i++, nid++) { - unsigned int wcaps = get_wcaps(codec, nid); - switch (get_wcaps_type(wcaps)) { - case AC_WID_AUD_IN: - return false; /* HDMI parser supports only HDMI out */ - case AC_WID_AUD_OUT: - if (!(wcaps & AC_WCAP_DIGITAL)) - return false; - break; - } - } - return true; -} -#else -/* no HDMI codec parser support */ -#define is_likely_hdmi_codec(codec) false -#endif /* CONFIG_SND_HDA_CODEC_HDMI */ - -/** - * snd_hda_codec_configure - (Re-)configure the HD-audio codec - * @codec: the HDA codec - * - * Start parsing of the given codec tree and (re-)initialize the whole - * patch instance. - * - * Returns 0 if successful or a negative error code. - */ -int snd_hda_codec_configure(struct hda_codec *codec) -{ - int (*patch)(struct hda_codec *) = NULL; - int err; - - codec->preset = find_codec_preset(codec); - if (!codec->vendor_name || !codec->chip_name) { - err = get_codec_name(codec); - if (err < 0) - return err; - } - - if (!is_generic_config(codec) && codec->preset) - patch = codec->preset->patch; - if (!patch) { - unload_parser(codec); /* to be sure */ - if (is_likely_hdmi_codec(codec)) { -#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) - patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#elif IS_BUILTIN(CONFIG_SND_HDA_CODEC_HDMI) - patch = snd_hda_parse_hdmi_codec; -#endif - } - if (!patch) { -#if IS_MODULE(CONFIG_SND_HDA_GENERIC) - patch = load_parser(codec, snd_hda_parse_generic_codec); -#elif IS_BUILTIN(CONFIG_SND_HDA_GENERIC) - patch = snd_hda_parse_generic_codec; -#endif - } - if (!patch) { - codec_err(codec, "No codec parser is available\n"); - return -ENODEV; - } - } - - err = patch(codec); - if (err < 0) { - unload_parser(codec); - return err; - } - - /* audio codec should override the mixer name */ - if (codec->afg || !*codec->bus->card->mixername) - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), - "%s %s", codec->vendor_name, codec->chip_name); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_codec_configure); - /* update the stream-id if changed */ static void update_pcm_stream_id(struct hda_codec *codec, struct hda_cvt_setup *p, hda_nid_t nid, @@ -2739,8 +2466,9 @@ int snd_hda_codec_reset(struct hda_codec *codec) } } snd_hda_detach_beep_device(codec); - if (codec->patch_ops.free) - codec->patch_ops.free(codec); + if (device_is_registered(hda_codec_dev(codec))) + device_del(hda_codec_dev(codec)); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; @@ -2759,9 +2487,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->preset = NULL; codec->slave_dig_outs = NULL; codec->spdif_status_reset = 0; - unload_parser(codec); - module_put(codec->owner); - codec->owner = NULL; /* allow device access again */ snd_hda_unlock_devices(bus); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 96421a3b32cd94..3d42717e0e653a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -174,15 +174,22 @@ struct hda_codec_preset { int (*patch)(struct hda_codec *codec); }; -struct hda_codec_preset_list { +#define HDA_CODEC_ID_GENERIC_HDMI 0x00000101 +#define HDA_CODEC_ID_GENERIC 0x00000201 + +struct hda_codec_driver { + struct device_driver driver; const struct hda_codec_preset *preset; - struct module *owner; - struct list_head list; }; -/* initial hook */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); -int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset); +int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, + struct module *owner); +#define hda_codec_driver_register(drv) \ + __hda_codec_driver_register(drv, KBUILD_MODNAME, THIS_MODULE) +void hda_codec_driver_unregister(struct hda_codec_driver *drv); +#define module_hda_codec_driver(drv) \ + module_driver(drv, hda_codec_driver_register, \ + hda_codec_driver_unregister) /* ops set by the preset patch */ struct hda_codec_ops { @@ -286,11 +293,10 @@ struct hda_codec { u32 vendor_id; u32 subsystem_id; u32 revision_id; + u32 probe_id; /* overridden id for probing */ /* detected preset */ const struct hda_codec_preset *preset; - struct module *owner; - int (*parser)(struct hda_codec *codec); const char *vendor_name; /* codec vendor name */ const char *chip_name; /* codec chip name */ const char *modelname; /* model name for preset */ @@ -408,6 +414,11 @@ struct hda_codec { struct snd_array verbs; }; +#define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, dev) +#define hda_codec_dev(_dev) (&(_dev)->dev) + +extern struct bus_type snd_hda_bus_type; + /* direction */ enum { HDA_INPUT, HDA_OUTPUT diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec63313c..947d1a50f3840b 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5493,13 +5493,11 @@ static const struct hda_codec_ops generic_patch_ops = { #endif }; -/** +/* * snd_hda_parse_generic_codec - Generic codec parser * @codec: the HDA codec - * - * This should be called from the HDA codec core. */ -int snd_hda_parse_generic_codec(struct hda_codec *codec) +static int snd_hda_parse_generic_codec(struct hda_codec *codec) { struct hda_gen_spec *spec; int err; @@ -5525,7 +5523,17 @@ int snd_hda_parse_generic_codec(struct hda_codec *codec) snd_hda_gen_free(codec); return err; } -EXPORT_SYMBOL_GPL(snd_hda_parse_generic_codec); + +static const struct hda_codec_preset snd_hda_preset_generic[] = { + { .id = HDA_CODEC_ID_GENERIC, .patch = snd_hda_parse_generic_codec }, + {} /* terminator */ +}; + +static struct hda_codec_driver generic_driver = { + .preset = snd_hda_preset_generic, +}; + +module_hda_codec_driver(generic_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Generic HD-audio codec parser"); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 49c08a7278c449..2f7d9646a41d5c 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -350,12 +350,6 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout); -/* - * generic codec parser - */ -int snd_hda_parse_generic_codec(struct hda_codec *codec); -int snd_hda_parse_hdmi_codec(struct hda_codec *codec); - /* * generic proc interface */ @@ -783,9 +777,13 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); /* */ -#define codec_err(codec, fmt, args...) dev_err(&(codec)->dev, fmt, ##args) -#define codec_warn(codec, fmt, args...) dev_warn(&(codec)->dev, fmt, ##args) -#define codec_info(codec, fmt, args...) dev_info(&(codec)->dev, fmt, ##args) -#define codec_dbg(codec, fmt, args...) dev_dbg(&(codec)->dev, fmt, ##args) +#define codec_err(codec, fmt, args...) \ + dev_err(hda_codec_dev(codec), fmt, ##args) +#define codec_warn(codec, fmt, args...) \ + dev_warn(hda_codec_dev(codec), fmt, ##args) +#define codec_info(codec, fmt, args...) \ + dev_info(hda_codec_dev(codec), fmt, ##args) +#define codec_dbg(codec, fmt, args...) \ + dev_dbg(hda_codec_dev(codec), fmt, ##args) #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d285904cdb64a0..af4c7be86c27b8 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1194,20 +1194,8 @@ MODULE_ALIAS("snd-hda-codec-id:11d4*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Analog Devices HD-audio codec"); -static struct hda_codec_preset_list analog_list = { +static struct hda_codec_driver analog_driver = { .preset = snd_hda_preset_analog, - .owner = THIS_MODULE, }; -static int __init patch_analog_init(void) -{ - return snd_hda_add_codec_preset(&analog_list); -} - -static void __exit patch_analog_exit(void) -{ - snd_hda_delete_codec_preset(&analog_list); -} - -module_init(patch_analog_init) -module_exit(patch_analog_exit) +module_hda_codec_driver(analog_driver); diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 5e65999e0d8e6a..44730269519572 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -98,20 +98,8 @@ MODULE_ALIAS("snd-hda-codec-id:1102000d"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec"); -static struct hda_codec_preset_list ca0110_list = { +static struct hda_codec_driver ca0110_driver = { .preset = snd_hda_preset_ca0110, - .owner = THIS_MODULE, }; -static int __init patch_ca0110_init(void) -{ - return snd_hda_add_codec_preset(&ca0110_list); -} - -static void __exit patch_ca0110_exit(void) -{ - snd_hda_delete_codec_preset(&ca0110_list); -} - -module_init(patch_ca0110_init) -module_exit(patch_ca0110_exit) +module_hda_codec_driver(ca0110_driver); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e0383eea988024..81991b4134cd10 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4702,20 +4702,8 @@ MODULE_ALIAS("snd-hda-codec-id:11020011"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Creative Sound Core3D codec"); -static struct hda_codec_preset_list ca0132_list = { +static struct hda_codec_driver ca0132_driver = { .preset = snd_hda_preset_ca0132, - .owner = THIS_MODULE, }; -static int __init patch_ca0132_init(void) -{ - return snd_hda_add_codec_preset(&ca0132_list); -} - -static void __exit patch_ca0132_exit(void) -{ - snd_hda_delete_codec_preset(&ca0132_list); -} - -module_init(patch_ca0132_init) -module_exit(patch_ca0132_exit) +module_hda_codec_driver(ca0132_driver); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 1589c9bcce3e15..1af133933bc09e 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1219,20 +1219,8 @@ MODULE_ALIAS("snd-hda-codec-id:10134213"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -static struct hda_codec_preset_list cirrus_list = { +static struct hda_codec_driver cirrus_driver = { .preset = snd_hda_preset_cirrus, - .owner = THIS_MODULE, }; -static int __init patch_cirrus_init(void) -{ - return snd_hda_add_codec_preset(&cirrus_list); -} - -static void __exit patch_cirrus_exit(void) -{ - snd_hda_delete_codec_preset(&cirrus_list); -} - -module_init(patch_cirrus_init) -module_exit(patch_cirrus_exit) +module_hda_codec_driver(cirrus_driver); diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index c895a8f2119224..617d9012e78abe 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -137,20 +137,8 @@ MODULE_ALIAS("snd-hda-codec-id:434d4980"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("C-Media HD-audio codec"); -static struct hda_codec_preset_list cmedia_list = { +static struct hda_codec_driver cmedia_driver = { .preset = snd_hda_preset_cmedia, - .owner = THIS_MODULE, }; -static int __init patch_cmedia_init(void) -{ - return snd_hda_add_codec_preset(&cmedia_list); -} - -static void __exit patch_cmedia_exit(void) -{ - snd_hda_delete_codec_preset(&cmedia_list); -} - -module_init(patch_cmedia_init) -module_exit(patch_cmedia_exit) +module_hda_codec_driver(cmedia_driver); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fd3ed18670e9c4..15a0a7d38c3576 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1007,20 +1007,8 @@ MODULE_ALIAS("snd-hda-codec-id:14f151d7"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -static struct hda_codec_preset_list conexant_list = { +static struct hda_codec_driver conexant_driver = { .preset = snd_hda_preset_conexant, - .owner = THIS_MODULE, }; -static int __init patch_conexant_init(void) -{ - return snd_hda_add_codec_preset(&conexant_list); -} - -static void __exit patch_conexant_exit(void) -{ - snd_hda_delete_codec_preset(&conexant_list); -} - -module_init(patch_conexant_init) -module_exit(patch_conexant_exit) +module_hda_codec_driver(conexant_driver); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b422e406a9cb3b..f1812aabd63e25 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3300,15 +3300,6 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } -/* - * called from hda_codec.c for generic HDMI support - */ -int snd_hda_parse_hdmi_codec(struct hda_codec *codec) -{ - return patch_generic_hdmi(codec); -} -EXPORT_SYMBOL_GPL(snd_hda_parse_hdmi_codec); - /* * patch entries */ @@ -3373,6 +3364,8 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, +/* special ID for generic HDMI */ +{ .id = HDA_CODEC_ID_GENERIC_HDMI, .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -3442,20 +3435,8 @@ MODULE_ALIAS("snd-hda-codec-intelhdmi"); MODULE_ALIAS("snd-hda-codec-nvhdmi"); MODULE_ALIAS("snd-hda-codec-atihdmi"); -static struct hda_codec_preset_list intel_list = { +static struct hda_codec_driver hdmi_driver = { .preset = snd_hda_preset_hdmi, - .owner = THIS_MODULE, }; -static int __init patch_hdmi_init(void) -{ - return snd_hda_add_codec_preset(&intel_list); -} - -static void __exit patch_hdmi_exit(void) -{ - snd_hda_delete_codec_preset(&intel_list); -} - -module_init(patch_hdmi_init) -module_exit(patch_hdmi_exit) +module_hda_codec_driver(hdmi_driver); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b2b24a8b3dac8c..70808f92276ada 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6514,20 +6514,8 @@ MODULE_ALIAS("snd-hda-codec-id:10ec*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Realtek HD-audio codec"); -static struct hda_codec_preset_list realtek_list = { +static struct hda_codec_driver realtek_driver = { .preset = snd_hda_preset_realtek, - .owner = THIS_MODULE, }; -static int __init patch_realtek_init(void) -{ - return snd_hda_add_codec_preset(&realtek_list); -} - -static void __exit patch_realtek_exit(void) -{ - snd_hda_delete_codec_preset(&realtek_list); -} - -module_init(patch_realtek_init) -module_exit(patch_realtek_exit) +module_hda_codec_driver(realtek_driver); diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 3208ad69583ee5..38a4773333218d 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -319,20 +319,8 @@ MODULE_ALIAS("snd-hda-codec-id:18540018"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Si3054 HD-audio modem codec"); -static struct hda_codec_preset_list si3054_list = { +static struct hda_codec_driver si3054_driver = { .preset = snd_hda_preset_si3054, - .owner = THIS_MODULE, }; -static int __init patch_si3054_init(void) -{ - return snd_hda_add_codec_preset(&si3054_list); -} - -static void __exit patch_si3054_exit(void) -{ - snd_hda_delete_codec_preset(&si3054_list); -} - -module_init(patch_si3054_init) -module_exit(patch_si3054_exit) +module_hda_codec_driver(si3054_driver); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 87eff3173ce924..6a216305621666 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5091,20 +5091,8 @@ MODULE_ALIAS("snd-hda-codec-id:111d*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec"); -static struct hda_codec_preset_list sigmatel_list = { +static struct hda_codec_driver sigmatel_driver = { .preset = snd_hda_preset_sigmatel, - .owner = THIS_MODULE, }; -static int __init patch_sigmatel_init(void) -{ - return snd_hda_add_codec_preset(&sigmatel_list); -} - -static void __exit patch_sigmatel_exit(void) -{ - snd_hda_delete_codec_preset(&sigmatel_list); -} - -module_init(patch_sigmatel_init) -module_exit(patch_sigmatel_exit) +module_hda_codec_driver(sigmatel_driver); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 3de6d3d779c994..2045f33b1ace3c 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1884,23 +1884,11 @@ static const struct hda_codec_preset snd_hda_preset_via[] = { MODULE_ALIAS("snd-hda-codec-id:1106*"); -static struct hda_codec_preset_list via_list = { +static struct hda_codec_driver via_driver = { .preset = snd_hda_preset_via, - .owner = THIS_MODULE, }; MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("VIA HD-audio codec"); -static int __init patch_via_init(void) -{ - return snd_hda_add_codec_preset(&via_list); -} - -static void __exit patch_via_exit(void) -{ - snd_hda_delete_codec_preset(&via_list); -} - -module_init(patch_via_init) -module_exit(patch_via_exit) +module_hda_codec_driver(via_driver); From 59ed1eade1d6ec24751baca99305f9713a5d779e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Feb 2015 15:39:59 +0100 Subject: [PATCH 042/411] ALSA: hda - Move codec suspend/resume to codec driver This patch moves the suspend/resume mechanisms down to each codec driver level, as we have a proper codec driver bound on the bus now. Then we get the asynchronous PM gratis without fiddling much in the driver level. As a soft-landing transition, implement the common suspend/resume pm ops for hda_codec_driver and keep the each codec driver intact. Only the callers of suspend/resume in the controller side (azx_suspend() and azx_resume()) are removed. Another involved place is azx_bus_reset() calling the temporary suspend and resume as a hackish method of bus reset. The HD-audio core provide a helper function snd_hda_bus_reset() instead. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 3 +- sound/pci/hda/hda_codec.c | 93 ++++++++++++---------------------- sound/pci/hda/hda_codec.h | 6 +-- sound/pci/hda/hda_controller.c | 11 +--- sound/pci/hda/hda_intel.c | 6 --- sound/pci/hda/hda_tegra.c | 6 --- 6 files changed, 39 insertions(+), 86 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index adf6b475dee135..ce2dd7b0dc078f 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -8,6 +8,7 @@ #include #include #include +#include #include #include "hda_codec.h" #include "hda_local.h" @@ -138,7 +139,7 @@ int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, drv->driver.bus = &snd_hda_bus_type; drv->driver.probe = hda_codec_driver_probe; drv->driver.remove = hda_codec_driver_remove; - /* TODO: PM and others */ + drv->driver.pm = &hda_codec_driver_pm; return driver_register(&drv->driver); } EXPORT_SYMBOL_GPL(__hda_codec_driver_register); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 61c8174e8013c4..3d6ff60e6c144b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1250,6 +1250,7 @@ int snd_hda_codec_new(struct hda_bus *bus, dev->groups = snd_hda_dev_attr_groups; dev_set_name(dev, "hdaudioC%dD%d", bus->card->number, codec_addr); dev_set_drvdata(dev, codec); /* for sysfs */ + device_enable_async_suspend(dev); codec->bus = bus; codec->addr = codec_addr; @@ -3970,8 +3971,31 @@ static void hda_call_codec_resume(struct hda_codec *codec) codec->in_pm = 0; snd_hda_power_down(codec); /* flag down before returning */ } + +static int hda_codec_driver_suspend(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + int i; + + cancel_delayed_work_sync(&codec->jackpoll_work); + for (i = 0; i < codec->num_pcms; i++) + snd_pcm_suspend_all(codec->pcm_info[i].pcm); + hda_call_codec_suspend(codec, false); + return 0; +} + +static int hda_codec_driver_resume(struct device *dev) +{ + hda_call_codec_resume(dev_to_hda_codec(dev)); + return 0; +} #endif /* CONFIG_PM */ +/* referred in hda_bind.c */ +const struct dev_pm_ops hda_codec_driver_pm = { + SET_SYSTEM_SLEEP_PM_OPS(hda_codec_driver_suspend, + hda_codec_driver_resume) +}; /** * snd_hda_build_controls - build mixer controls @@ -5505,77 +5529,26 @@ int snd_hda_add_imux_item(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_add_imux_item); - -#ifdef CONFIG_PM -/* - * power management - */ - - -static void hda_async_suspend(void *data, async_cookie_t cookie) -{ - hda_call_codec_suspend(data, false); -} - -static void hda_async_resume(void *data, async_cookie_t cookie) -{ - hda_call_codec_resume(data); -} - /** - * snd_hda_suspend - suspend the codecs - * @bus: the HDA bus - * - * Returns 0 if successful. + * snd_hda_bus_reset - Reset the bus + * @bus: HD-audio bus */ -int snd_hda_suspend(struct hda_bus *bus) +void snd_hda_bus_reset(struct hda_bus *bus) { struct hda_codec *codec; - ASYNC_DOMAIN_EXCLUSIVE(domain); list_for_each_entry(codec, &bus->codec_list, list) { + /* FIXME: maybe a better way needed for forced reset */ cancel_delayed_work_sync(&codec->jackpoll_work); +#ifdef CONFIG_PM if (hda_codec_is_power_on(codec)) { - if (bus->num_codecs > 1) - async_schedule_domain(hda_async_suspend, codec, - &domain); - else - hda_call_codec_suspend(codec, false); - } - } - - if (bus->num_codecs > 1) - async_synchronize_full_domain(&domain); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_suspend); - -/** - * snd_hda_resume - resume the codecs - * @bus: the HDA bus - * - * Returns 0 if successful. - */ -int snd_hda_resume(struct hda_bus *bus) -{ - struct hda_codec *codec; - ASYNC_DOMAIN_EXCLUSIVE(domain); - - list_for_each_entry(codec, &bus->codec_list, list) { - if (bus->num_codecs > 1) - async_schedule_domain(hda_async_resume, codec, &domain); - else + hda_call_codec_suspend(codec, false); hda_call_codec_resume(codec); + } +#endif } - - if (bus->num_codecs > 1) - async_synchronize_full_domain(&domain); - - return 0; } -EXPORT_SYMBOL_GPL(snd_hda_resume); -#endif /* CONFIG_PM */ +EXPORT_SYMBOL_GPL(snd_hda_bus_reset); /* * generic arrays diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 3d42717e0e653a..1fa3dd9baf5225 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -567,14 +567,12 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, int snd_hda_lock_devices(struct hda_bus *bus); void snd_hda_unlock_devices(struct hda_bus *bus); +void snd_hda_bus_reset(struct hda_bus *bus); /* * power management */ -#ifdef CONFIG_PM -int snd_hda_suspend(struct hda_bus *bus); -int snd_hda_resume(struct hda_bus *bus); -#endif +extern const struct dev_pm_ops hda_codec_driver_pm; static inline int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 4c7a6f9bfcde04..30ddb751806acc 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1780,15 +1780,8 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 1; azx_stop_chip(chip); azx_init_chip(chip, true); -#ifdef CONFIG_PM - if (chip->initialized) { - struct azx_pcm *p; - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); - } -#endif + if (chip->initialized) + snd_hda_bus_reset(chip->bus); bus->in_reset = 0; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5e00cc4a722f4c..9db1b078801f29 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -772,7 +772,6 @@ static int azx_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; - struct azx_pcm *p; if (!card) return 0; @@ -784,10 +783,6 @@ static int azx_suspend(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_clear_irq_pending(chip); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - if (chip->initialized) - snd_hda_suspend(chip->bus); azx_stop_chip(chip); azx_enter_link_reset(chip); if (chip->irq >= 0) { @@ -830,7 +825,6 @@ static int azx_resume(struct device *dev) azx_init_chip(chip, true); - snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 1bd7a9e040465b..f6949e413a501a 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -249,14 +249,9 @@ static int hda_tegra_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - struct azx_pcm *p; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - if (chip->initialized) - snd_hda_suspend(chip->bus); azx_stop_chip(chip); azx_enter_link_reset(chip); @@ -277,7 +272,6 @@ static int hda_tegra_resume(struct device *dev) azx_init_chip(chip, 1); - snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; From cc72da7d4d063ab9e690e56e0ef1ca1c24ee1635 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 16:00:22 +0100 Subject: [PATCH 043/411] ALSA: hda - Use standard runtime PM for codec power-save control Like the previous transition of suspend/resume, now move the power-save code to the standard runtime PM. As usual for runtime PM, it's a bit tricky, but this simplified codes a lot in the end. For keeping the usage compatibility, power_save module option still controls the whole power-saving behavior on all codecs. The value is translated to pm_runtime_*_autosuspend() and pm_runtime_allow() / pm_runtime_forbid() calls. snd_hda_power_up() and snd_hda_power_down() are translated to pm_runtime_get_sync() and pm_runtime_put_autosuspend(), respectively. Since we can do call pm_runtime_get_sync() more reliably, the sync version is used always and snd_hda_power_up_d3wait() is dropped. Another slight difference is that snd_hda_power_up()/down() don't call runtime_pm code during the suspend/resume transition phase. Calling them there isn't safe unlike our own code, resulted in unexpected behavior (endless wakeups). The hda_power_count tracepoint was removed, as it doesn't match well with the new code. Last but not least, we need to set ignore_children flag in the parent dev.power field so that the runtime PM of the controller chip won't get confused. The notification is still done in the bus pm_notify callback. We'll get rid of this hack in the later patch. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 241 +++++++++++++-------------------- sound/pci/hda/hda_codec.h | 67 +-------- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_intel.c | 2 + sound/pci/hda/hda_trace.h | 24 ---- 5 files changed, 106 insertions(+), 230 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3d6ff60e6c144b..d0dbc62c91474a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -26,6 +26,8 @@ #include #include #include +#include +#include #include #include "hda_codec.h" #include @@ -41,10 +43,9 @@ #include "hda_trace.h" #ifdef CONFIG_PM -#define codec_in_pm(codec) ((codec)->in_pm) -static void hda_power_work(struct work_struct *work); -static void hda_keep_power_on(struct hda_codec *codec); -#define hda_codec_is_power_on(codec) ((codec)->power_on) +#define codec_in_pm(codec) atomic_read(&(codec)->in_pm) +#define hda_codec_is_power_on(codec) \ + (!pm_runtime_suspended(hda_codec_dev(codec))) static void hda_call_pm_notify(struct hda_codec *codec, bool power_up) { @@ -60,7 +61,6 @@ static void hda_call_pm_notify(struct hda_codec *codec, bool power_up) #else #define codec_in_pm(codec) 0 -static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 #define hda_call_pm_notify(codec, state) {} #endif @@ -1144,10 +1144,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) device_del(hda_codec_dev(codec)); snd_hda_jack_tbl_clear(codec); free_init_pincfgs(codec); -#ifdef CONFIG_PM - cancel_delayed_work(&codec->power_work); flush_workqueue(codec->bus->workq); -#endif list_del(&codec->list); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); @@ -1178,6 +1175,10 @@ static int snd_hda_codec_dev_register(struct snd_device *device) struct hda_codec *codec = device->device_data; snd_hda_register_beep_device(codec); + if (device_is_registered(hda_codec_dev(codec))) { + snd_hda_power_sync(codec); + pm_runtime_enable(hda_codec_dev(codec)); + } return 0; } @@ -1274,13 +1275,14 @@ int snd_hda_codec_new(struct hda_bus *bus, codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM - spin_lock_init(&codec->power_lock); - INIT_DELAYED_WORK(&codec->power_work, hda_power_work); /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. * the caller has to power down appropriatley after initialization * phase. */ - hda_keep_power_on(codec); + pm_runtime_set_active(hda_codec_dev(codec)); + pm_runtime_get_noresume(hda_codec_dev(codec)); + codec->power_jiffies = jiffies; + hda_call_pm_notify(codec, true); #endif snd_hda_sysfs_init(codec); @@ -2453,10 +2455,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) /* OK, let it free */ cancel_delayed_work_sync(&codec->jackpoll_work); -#ifdef CONFIG_PM - cancel_delayed_work_sync(&codec->power_work); flush_workqueue(bus->workq); -#endif snd_hda_ctls_clear(codec); /* release PCMs */ for (i = 0; i < codec->num_pcms; i++) { @@ -3893,31 +3892,40 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} #endif #ifdef CONFIG_PM +/* update the power on/off account with the current jiffies */ +static void update_power_acct(struct hda_codec *codec, bool on) +{ + unsigned long delta = jiffies - codec->power_jiffies; + + if (on) + codec->power_on_acct += delta; + else + codec->power_off_acct += delta; + codec->power_jiffies += delta; +} + +void snd_hda_update_power_acct(struct hda_codec *codec) +{ + update_power_acct(codec, hda_codec_is_power_on(codec)); +} + /* * call suspend and power-down; used both from PM and power-save * this function returns the power state in the end */ -static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) +static unsigned int hda_call_codec_suspend(struct hda_codec *codec) { unsigned int state; - codec->in_pm = 1; + atomic_inc(&codec->in_pm); if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); - /* Cancel delayed work if we aren't currently running from it. */ - if (!in_wq) - cancel_delayed_work_sync(&codec->power_work); - spin_lock(&codec->power_lock); - snd_hda_update_power_acct(codec); trace_hda_power_down(codec); - codec->power_on = 0; - codec->power_transition = 0; - codec->power_jiffies = jiffies; - spin_unlock(&codec->power_lock); - codec->in_pm = 0; + update_power_acct(codec, true); + atomic_dec(&codec->in_pm); return state; } @@ -3942,14 +3950,14 @@ static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { - codec->in_pm = 1; + atomic_inc(&codec->in_pm); + trace_hda_power_up(codec); hda_mark_cmd_cache_dirty(codec); - /* set as if powered on for avoiding re-entering the resume - * in the resume / power-save sequence - */ - hda_keep_power_on(codec); + codec->power_jiffies = jiffies; + hda_call_pm_notify(codec, true); + hda_set_power_state(codec, AC_PWRST_D0); restore_shutup_pins(codec); hda_exec_init_verbs(codec); @@ -3967,34 +3975,38 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_jackpoll_work(&codec->jackpoll_work.work); else snd_hda_jack_report_sync(codec); - - codec->in_pm = 0; - snd_hda_power_down(codec); /* flag down before returning */ + atomic_dec(&codec->in_pm); } -static int hda_codec_driver_suspend(struct device *dev) +static int hda_codec_runtime_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + unsigned int state; int i; cancel_delayed_work_sync(&codec->jackpoll_work); for (i = 0; i < codec->num_pcms; i++) snd_pcm_suspend_all(codec->pcm_info[i].pcm); - hda_call_codec_suspend(codec, false); + state = hda_call_codec_suspend(codec); + if (!codec->bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) + hda_call_pm_notify(codec, false); return 0; } -static int hda_codec_driver_resume(struct device *dev) +static int hda_codec_runtime_resume(struct device *dev) { hda_call_codec_resume(dev_to_hda_codec(dev)); + pm_runtime_mark_last_busy(dev); return 0; } #endif /* CONFIG_PM */ /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { - SET_SYSTEM_SLEEP_PM_OPS(hda_codec_driver_suspend, - hda_codec_driver_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) + SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume, + NULL) }; /** @@ -4733,127 +4745,66 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls); #ifdef CONFIG_PM -static void hda_power_work(struct work_struct *work) +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the usage counter and resume the device if not done yet. + */ +void snd_hda_power_up(struct hda_codec *codec) { - struct hda_codec *codec = - container_of(work, struct hda_codec, power_work.work); - struct hda_bus *bus = codec->bus; - unsigned int state; + struct device *dev = hda_codec_dev(codec); - spin_lock(&codec->power_lock); - if (codec->power_transition > 0) { /* during power-up sequence? */ - spin_unlock(&codec->power_lock); + if (codec_in_pm(codec)) return; - } - if (!codec->power_on || codec->power_count) { - codec->power_transition = 0; - spin_unlock(&codec->power_lock); - return; - } - spin_unlock(&codec->power_lock); - - state = hda_call_codec_suspend(codec, true); - if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) - hda_call_pm_notify(codec, false); + pm_runtime_get_sync(dev); } +EXPORT_SYMBOL_GPL(snd_hda_power_up); -static void hda_keep_power_on(struct hda_codec *codec) +/** + * snd_hda_power_down - Power-down the codec + * @codec: HD-audio codec + * + * Decrement the usage counter and schedules the autosuspend if none used. + */ +void snd_hda_power_down(struct hda_codec *codec) { - spin_lock(&codec->power_lock); - codec->power_count++; - codec->power_on = 1; - codec->power_jiffies = jiffies; - spin_unlock(&codec->power_lock); - hda_call_pm_notify(codec, true); -} + struct device *dev = hda_codec_dev(codec); -/* update the power on/off account with the current jiffies */ -void snd_hda_update_power_acct(struct hda_codec *codec) -{ - unsigned long delta = jiffies - codec->power_jiffies; - if (codec->power_on) - codec->power_on_acct += delta; - else - codec->power_off_acct += delta; - codec->power_jiffies += delta; -} - -/* Transition to powered up, if wait_power_down then wait for a pending - * transition to D3 to complete. A pending D3 transition is indicated - * with power_transition == -1. */ -/* call this with codec->power_lock held! */ -static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) -{ - /* Return if power_on or transitioning to power_on, unless currently - * powering down. */ - if ((codec->power_on || codec->power_transition > 0) && - !(wait_power_down && codec->power_transition < 0)) + if (codec_in_pm(codec)) return; - spin_unlock(&codec->power_lock); - - cancel_delayed_work_sync(&codec->power_work); - - spin_lock(&codec->power_lock); - /* If the power down delayed work was cancelled above before starting, - * then there is no need to go through power up here. - */ - if (codec->power_on) { - if (codec->power_transition < 0) - codec->power_transition = 0; - return; - } - - trace_hda_power_up(codec); - snd_hda_update_power_acct(codec); - codec->power_on = 1; - codec->power_jiffies = jiffies; - codec->power_transition = 1; /* avoid reentrance */ - spin_unlock(&codec->power_lock); - - hda_call_codec_resume(codec); - - spin_lock(&codec->power_lock); - codec->power_transition = 0; -} - -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) - -/* Transition to powered down */ -static void __snd_hda_power_down(struct hda_codec *codec) -{ - if (!codec->power_on || codec->power_count || codec->power_transition) - return; - - if (power_save(codec)) { - codec->power_transition = -1; /* avoid reentrance */ - queue_delayed_work(codec->bus->workq, &codec->power_work, - msecs_to_jiffies(power_save(codec) * 1000)); - } + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); } +EXPORT_SYMBOL_GPL(snd_hda_power_down); /** - * snd_hda_power_save - Power-up/down/sync the codec + * snd_hda_power_sync - Synchronize the power_save option * @codec: HD-audio codec - * @delta: the counter delta to change - * @d3wait: sync for D3 transition complete * - * Change the power-up counter via @delta, and power up or down the hardware - * appropriately. For the power-down, queue to the delayed action. - * Passing zero to @delta means to synchronize the power state. + * Synchronize the runtime PM autosuspend state from the power_save option. */ -void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) +void snd_hda_power_sync(struct hda_codec *codec) { - spin_lock(&codec->power_lock); - codec->power_count += delta; - trace_hda_power_count(codec); - if (delta > 0) - __snd_hda_power_up(codec, d3wait); - else - __snd_hda_power_down(codec); - spin_unlock(&codec->power_lock); + struct device *dev = hda_codec_dev(codec); + int delay; + + if (!codec->bus->power_save) + return; + + delay = *codec->bus->power_save * 1000; + if (delay > 0) { + pm_runtime_set_autosuspend_delay(dev, delay); + pm_runtime_use_autosuspend(dev); + pm_runtime_allow(dev); + if (!pm_runtime_suspended(dev)) + pm_runtime_mark_last_busy(dev); + } else { + pm_runtime_dont_use_autosuspend(dev); + pm_runtime_forbid(dev); + } } -EXPORT_SYMBOL_GPL(snd_hda_power_save); +EXPORT_SYMBOL_GPL(snd_hda_power_sync); /** * snd_hda_check_amp_list_power - Check the amp list and update the power @@ -5542,7 +5493,7 @@ void snd_hda_bus_reset(struct hda_bus *bus) cancel_delayed_work_sync(&codec->jackpoll_work); #ifdef CONFIG_PM if (hda_codec_is_power_on(codec)) { - hda_call_codec_suspend(codec, false); + hda_call_codec_suspend(codec); hda_call_codec_resume(codec); } #endif diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1fa3dd9baf5225..593956fc384b09 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -372,17 +372,12 @@ struct hda_codec { unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ #ifdef CONFIG_PM - unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_up_notified:1; /* PM notified to controller */ - unsigned int in_pm:1; /* suspend/resume being performed */ - int power_transition; /* power-state in transition */ - int power_count; /* current (global) power refcount */ - struct delayed_work power_work; /* delayed task for powerdown */ + atomic_t in_pm; /* suspend/resume being performed */ unsigned long power_on_acct; unsigned long power_off_acct; unsigned long power_jiffies; - spinlock_t power_lock; #endif /* filter the requested power state per nid */ @@ -595,64 +590,16 @@ const char *snd_hda_get_jack_location(u32 cfg); * power saving */ #ifdef CONFIG_PM -void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); +void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_down(struct hda_codec *codec); +void snd_hda_power_sync(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); #else -static inline void snd_hda_power_save(struct hda_codec *codec, int delta, - bool d3wait) {} +static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_down(struct hda_codec *codec) {} +static inline void snd_hda_power_sync(struct hda_codec *codec) {} #endif -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -static inline void snd_hda_power_up(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 1, false); -} - -/** - * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending - * D3 transition to complete. This differs from snd_hda_power_up() when - * power_transition == -1. snd_hda_power_up sees this case as a nop, - * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers - * back up. - * @codec: HD-audio codec - * - * Cancel any power down operation hapenning on the work queue, then power up. - */ -static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 1, true); -} - -/** - * snd_hda_power_down - Power-down the codec - * @codec: HD-audio codec - * - * Decrement the power-up counter and schedules the power-off work if - * the counter rearches to zero. - */ -static inline void snd_hda_power_down(struct hda_codec *codec) -{ - snd_hda_power_save(codec, -1, false); -} - -/** - * snd_hda_power_sync - Synchronize the power-save status - * @codec: HD-audio codec - * - * Synchronize the actual power state with the power account; - * called when power_save parameter is changed - */ -static inline void snd_hda_power_sync(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 0, false); -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 30ddb751806acc..522c54f9474077 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -836,7 +836,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up_d3wait(apcm->codec); + snd_hda_power_up(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { azx_release_device(azx_dev); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9db1b078801f29..26510e60cbe2b3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1087,6 +1087,7 @@ static int azx_free(struct azx *chip) azx_stop_chip(chip); } + pci->dev.power.ignore_children = 0; /* FIXME */ if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); if (chip->msi) @@ -1796,6 +1797,7 @@ static int azx_probe(struct pci_dev *pci, return err; } + pci->dev.power.ignore_children = 1; /* FIXME */ err = azx_create(card, pci, dev, pci_id->driver_data, &pci_hda_ops, &chip); if (err < 0) diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h index 3a1c63161eb129..c0e1c7d24dbebe 100644 --- a/sound/pci/hda/hda_trace.h +++ b/sound/pci/hda/hda_trace.h @@ -87,30 +87,6 @@ DEFINE_EVENT(hda_power, hda_power_up, TP_PROTO(struct hda_codec *codec), TP_ARGS(codec) ); - -TRACE_EVENT(hda_power_count, - TP_PROTO(struct hda_codec *codec), - TP_ARGS(codec), - TP_STRUCT__entry( - __field( unsigned int, card ) - __field( unsigned int, addr ) - __field( int, power_count ) - __field( int, power_on ) - __field( int, power_transition ) - ), - - TP_fast_assign( - __entry->card = (codec)->bus->card->number; - __entry->addr = (codec)->addr; - __entry->power_count = (codec)->power_count; - __entry->power_on = (codec)->power_on; - __entry->power_transition = (codec)->power_transition; - ), - - TP_printk("[%d:%d] power_count=%d, power_on=%d, power_transition=%d", - __entry->card, __entry->addr, __entry->power_count, - __entry->power_on, __entry->power_transition) -); #endif /* CONFIG_PM */ TRACE_EVENT(hda_unsol_event, From 052a9f698268e606ca01eb1ce2a672e548f2ce11 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 9 Feb 2015 00:18:11 -0800 Subject: [PATCH 044/411] ALSA: Add params_set_format helper Add a helper to set pcm format directly from params Signed-off-by: Fang, Yang A Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/pcm_params.h | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 3c45f3924ba7e7..c704357775fc4e 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -366,4 +366,11 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p) return snd_pcm_format_physical_width(params_format(p)); } +static inline void +params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt) +{ + snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), + (__force int)fmt); +} + #endif /* __SOUND_PCM_PARAMS_H */ From 369a9f5f397fe3258ab937ec7a9c2229d0b1a015 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 9 Feb 2015 00:18:12 -0800 Subject: [PATCH 045/411] ASoC: Intel: fix machine driver warnings this patch will fix below sparse warnings warning: incorrect type in argument 2 (different base types) expected unsigned int [unsigned] val got restricted snd_pcm_format_t [usertype] sound/soc/intel/haswell.c:61:37 sound/soc/intel/broadwell.c:115:37: sound/soc/intel/bytcr_dpcm_rt5640.c:118:37: sound/soc/intel/cht_bsw_rt5672.c:183:37: sound/soc/intel/cht_bsw_rt5645.c:208:37: Signed-off-by: Fang, Yang A Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 4 +--- sound/soc/intel/bytcr_dpcm_rt5640.c | 4 +--- sound/soc/intel/cht_bsw_rt5645.c | 4 +--- sound/soc/intel/cht_bsw_rt5672.c | 4 +--- sound/soc/intel/haswell.c | 4 +--- 5 files changed, 5 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01479adcb..fba2ef5dac4280 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -110,9 +110,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index 59308629043e62..3b262d01c1b370 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -113,9 +113,7 @@ static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617a9ab9d2..dd935255a0206f 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -203,9 +203,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index ff016621583a63..c56f9dfe2129dc 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -178,9 +178,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 35edf51a52aa92..00fddd3f5dfb8f 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -56,9 +56,7 @@ static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } From 48c7699fb2c799d084ce490bceea14fe04ad12a1 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:53 +0530 Subject: [PATCH 046/411] ASoC: core: allow pcms to be registered as nonatomic ALSA core with commit 257f8cce5d40 - "ALSA: pcm: Allow nonatomic trigger operations" allows trigger ops to implemented as nonatomic. For ASoC, we can specify this in dailinks and is updated while snd_pcm is created Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Cc: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 1 + 2 files changed, 4 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade19562857..76bc944dcb5cbf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -954,6 +954,9 @@ struct snd_soc_dai_link { unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; + /* Mark this pcm with non atomic ops */ + bool nonatomic; + /* Do not create a PCM for this DAI link (Backend link) */ unsigned int no_pcm:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e7cb88c2..6e3781e88f9adf 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2511,6 +2511,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; pcm->private_data = rtd; From 76ca1c2cd8fc0b8764c6360263e2fbca43495ab2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:54 +0530 Subject: [PATCH 047/411] ASoC: Intel: mark cht machine driver with nonatomic trigger The DSP messages are sent with nonatomic context, which include trigger messages, so mark the driver as nonatomic Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index c56f9dfe2129dc..a5098d6f988be5 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -216,6 +216,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -238,6 +239,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, + .nonatomic = true, .codec_dai_name = "rt5670-aif1", .codec_name = "i2c-10EC5670:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF From 7b9ca9d7e561ebdc93b43277eb69d20a0dc8f5cd Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:55 +0530 Subject: [PATCH 048/411] ASoC: Intel: update MMX ID to 3 The updated firmware expects the MMX ID to be used as 3, so update the driver as well Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index dfebfdd5eb2aaa..daecc58f28afaf 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -150,7 +150,7 @@ enum sst_cmd_type { enum sst_task { SST_TASK_SBA = 1, - SST_TASK_MMX, + SST_TASK_MMX = 3, }; enum sst_type { From e0b87d476bc13fc55e7000a84cd1d87c8fdc1f2f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:56 +0530 Subject: [PATCH 049/411] ASoC: Intel: add support for pause and resume in sst This adds missing pcm pause and resume ops in the driver Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 31 +++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 5f75ef3cdd2296..5d56fcdd58d882 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -572,6 +572,35 @@ static int sst_stream_drop(struct device *dev, int str_id) return sst_drop_stream(ctx, str_id); } +static int sst_stream_pause(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + + return sst_pause_stream(ctx, str_id); +} + +static int sst_stream_resume(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + return sst_resume_stream(ctx, str_id); +} + static int sst_stream_init(struct device *dev, struct pcm_stream_info *str_info) { int str_id = 0; @@ -633,6 +662,8 @@ static struct sst_ops pcm_ops = { .stream_init = sst_stream_init, .stream_start = sst_stream_start, .stream_drop = sst_stream_drop, + .stream_pause = sst_stream_pause, + .stream_pause_release = sst_stream_resume, .stream_read_tstamp = sst_read_timestamp, .send_byte_stream = sst_send_byte_stream, .close = sst_close_pcm_stream, From fc9406ab9b4a9aac0ab9ad213993824cbe9c65ac Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:57 +0530 Subject: [PATCH 050/411] ASoC: Intel: add support for pcm stream suspend/resume The driver didn't implement support for pcm stream suspend and resume, so add it Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7523cbef87807f..ea0fa4b90bb0e1 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -594,11 +594,13 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: dev_dbg(rtd->dev, "sst: in pause\n"); status = SST_PLATFORM_PAUSED; ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: dev_dbg(rtd->dev, "sst: in pause release\n"); status = SST_PLATFORM_RUNNING; ret_val = stream->ops->stream_pause_release(sst->dev, str_id); From 54e6beccc67129c474aad7578951112c6cf28e01 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:58 +0530 Subject: [PATCH 051/411] ASoC: Intel: add support for platform suspend This adds support for platform suspend and resume. We ensure all pcms are suspended by invoking snd_soc_suspend() and then stop the DSP Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 58 +++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 1 + 2 files changed, 59 insertions(+) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index ea0fa4b90bb0e1..2fbaf2c75d1709 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -667,6 +667,9 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) static int sst_soc_probe(struct snd_soc_platform *platform) { + struct sst_data *drv = dev_get_drvdata(platform->dev); + + drv->soc_card = platform->component.card; return sst_dsp_init_v2_dpcm(platform); } @@ -729,9 +732,64 @@ static int sst_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP + +static int sst_soc_prepare(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* suspend all pcms first */ + snd_soc_suspend(drv->soc_card->dev); + snd_soc_poweroff(drv->soc_card->dev); + + /* set the SSPs to idle */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + send_ssp_cmd(dai, dai->name, 0); + sst_handle_vb_timer(dai, false); + } + } + + return 0; +} + +static void sst_soc_complete(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* restart SSPs */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + sst_handle_vb_timer(dai, true); + send_ssp_cmd(dai, dai->name, 1); + } + } + snd_soc_resume(drv->soc_card->dev); +} + +#else + +#define sst_soc_prepare NULL +#define sst_soc_complete NULL + +#endif + + +static const struct dev_pm_ops sst_platform_pm = { + .prepare = sst_soc_prepare, + .complete = sst_soc_complete, +}; + static struct platform_driver sst_platform_driver = { .driver = { .name = "sst-mfld-platform", + .pm = &sst_platform_pm, }, .probe = sst_platform_probe, .remove = sst_platform_remove, diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 79c8d1246a8fb3..9094314be2b043 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -174,6 +174,7 @@ struct sst_data { struct sst_platform_data *pdata; struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; + struct snd_soc_card *soc_card; }; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); From 5c88b4e91d3b6a3d701d7b134fa945e6309e7068 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 12 Feb 2015 10:00:01 +0530 Subject: [PATCH 052/411] ASoC: Intel: Add memcpy32_fromio as well Export 32-bit version of memcpy for use in suspend/resume. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.h | 3 +++ sound/soc/intel/sst/sst_loader.c | 10 +++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 562bc483d6b7f7..f793780a50a206 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -544,4 +544,7 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, int sst_context_init(struct intel_sst_drv *ctx); void sst_context_cleanup(struct intel_sst_drv *ctx); void sst_configure_runtime_pm(struct intel_sst_drv *ctx); +void memcpy32_toio(void __iomem *dst, const void *src, int count); +void memcpy32_fromio(void *dst, const void __iomem *src, int count); + #endif diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index 7888cd707853db..e88907ae8b154f 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -39,7 +39,15 @@ #include "sst.h" #include "../sst-dsp.h" -static inline void memcpy32_toio(void __iomem *dst, const void *src, int count) +void memcpy32_toio(void __iomem *dst, const void *src, int count) +{ + /* __iowrite32_copy uses 32-bit count values so divide by 4 for + * right count in words + */ + __iowrite32_copy(dst, src, count/4); +} + +void memcpy32_fromio(void *dst, const void __iomem *src, int count) { /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words From 4a8448d4289d7210053a43f9f21e42929beb159b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 24 Feb 2015 11:39:44 +0530 Subject: [PATCH 053/411] ASoC: Intel: add pm support in sst ipc driver This adds support for system pm support. We need to save the dsp memory which gets lost on suspend and restore that on resume Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 128 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst.h | 9 +++ 2 files changed, 137 insertions(+) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a146e75a..8f938112a01fc4 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -415,6 +415,83 @@ static int intel_sst_runtime_suspend(struct device *dev) return ret; } +static int intel_sst_suspend(struct device *dev) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save; + int i, ret = 0; + + /* check first if we are already in SW reset */ + if (ctx->sst_state == SST_RESET) + return 0; + + /* + * check if any stream is active and running + * they should already by suspend by soc_suspend + */ + for (i = 1; i <= ctx->info.max_streams; i++) { + struct stream_info *stream = &ctx->streams[i]; + + if (stream->status == STREAM_RUNNING) { + dev_err(dev, "stream %d is running, cant susupend, abort\n", i); + return -EBUSY; + } + } + synchronize_irq(ctx->irq_num); + flush_workqueue(ctx->post_msg_wq); + + /* Move the SST state to Reset */ + sst_set_fw_state_locked(ctx, SST_RESET); + + /* tell DSP we are suspending */ + if (ctx->ops->save_dsp_context(ctx)) + return -EBUSY; + + /* save the memories */ + fw_save = kzalloc(sizeof(*fw_save), GFP_KERNEL); + if (!fw_save) + return -ENOMEM; + fw_save->iram = kzalloc(ctx->iram_end - ctx->iram_base, GFP_KERNEL); + if (!fw_save->iram) { + ret = -ENOMEM; + goto iram; + } + fw_save->dram = kzalloc(ctx->dram_end - ctx->dram_base, GFP_KERNEL); + if (!fw_save->dram) { + ret = -ENOMEM; + goto dram; + } + fw_save->sram = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); + if (!fw_save->sram) { + ret = -ENOMEM; + goto sram; + } + + fw_save->ddr = kzalloc(ctx->ddr_end - ctx->ddr_base, GFP_KERNEL); + if (!fw_save->ddr) { + ret = -ENOMEM; + goto ddr; + } + + memcpy32_fromio(fw_save->iram, ctx->iram, ctx->iram_end - ctx->iram_base); + memcpy32_fromio(fw_save->dram, ctx->dram, ctx->dram_end - ctx->dram_base); + memcpy32_fromio(fw_save->sram, ctx->mailbox, SST_MAILBOX_SIZE); + memcpy32_fromio(fw_save->ddr, ctx->ddr, ctx->ddr_end - ctx->ddr_base); + + ctx->fw_save = fw_save; + ctx->ops->reset(ctx); + return 0; +ddr: + kfree(fw_save->sram); +sram: + kfree(fw_save->dram); +dram: + kfree(fw_save->iram); +iram: + kfree(fw_save); + return ret; +} + static int intel_sst_runtime_resume(struct device *dev) { int ret = 0; @@ -430,7 +507,58 @@ static int intel_sst_runtime_resume(struct device *dev) return ret; } +static int intel_sst_resume(struct device *dev) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save = ctx->fw_save; + int ret = 0; + struct sst_block *block; + + if (!fw_save) + return intel_sst_runtime_resume(dev); + + sst_set_fw_state_locked(ctx, SST_FW_LOADING); + + /* we have to restore the memory saved */ + ctx->ops->reset(ctx); + + ctx->fw_save = NULL; + + memcpy32_toio(ctx->iram, fw_save->iram, ctx->iram_end - ctx->iram_base); + memcpy32_toio(ctx->dram, fw_save->dram, ctx->dram_end - ctx->dram_base); + memcpy32_toio(ctx->mailbox, fw_save->sram, SST_MAILBOX_SIZE); + memcpy32_toio(ctx->ddr, fw_save->ddr, ctx->ddr_end - ctx->ddr_base); + + kfree(fw_save->sram); + kfree(fw_save->dram); + kfree(fw_save->iram); + kfree(fw_save->ddr); + kfree(fw_save); + + block = sst_create_block(ctx, 0, FW_DWNL_ID); + if (block == NULL) + return -ENOMEM; + + + /* start and wait for ack */ + ctx->ops->start(ctx); + ret = sst_wait_timeout(ctx, block); + if (ret) { + dev_err(ctx->dev, "fw download failed %d\n", ret); + /* FW download failed due to timeout */ + ret = -EBUSY; + + } else { + sst_set_fw_state_locked(ctx, SST_FW_RUNNING); + } + + sst_free_block(ctx, block); + return ret; +} + const struct dev_pm_ops intel_sst_pm = { + .suspend = intel_sst_suspend, + .resume = intel_sst_resume, .runtime_suspend = intel_sst_runtime_suspend, .runtime_resume = intel_sst_runtime_resume, }; diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index f793780a50a206..3f493862e98dd0 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -337,6 +337,13 @@ struct sst_shim_regs64 { u64 csr2; }; +struct sst_fw_save { + void *iram; + void *dram; + void *sram; + void *ddr; +}; + /** * struct intel_sst_drv - driver ops * @@ -428,6 +435,8 @@ struct intel_sst_drv { * persistent till worker thread gets called */ char firmware_name[FW_NAME_SIZE]; + + struct sst_fw_save *fw_save; }; /* misc definitions */ From 9308d1456e30e374d93957e3376a09370be9dc52 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 24 Feb 2015 11:39:45 +0530 Subject: [PATCH 054/411] ASoC: Intel: Move the fw download to power_control Thus removing the runtime_resume handler. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 18 +---------------- sound/soc/intel/sst/sst_drv_interface.c | 27 +++++++++++++++++++++---- 2 files changed, 24 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8f938112a01fc4..4d8f73ac5dd95f 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -492,21 +492,6 @@ static int intel_sst_suspend(struct device *dev) return ret; } -static int intel_sst_runtime_resume(struct device *dev) -{ - int ret = 0; - struct intel_sst_drv *ctx = dev_get_drvdata(dev); - - if (ctx->sst_state == SST_RESET) { - ret = sst_load_fw(ctx); - if (ret) { - dev_err(dev, "FW download fail %d\n", ret); - sst_set_fw_state_locked(ctx, SST_RESET); - } - } - return ret; -} - static int intel_sst_resume(struct device *dev) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); @@ -515,7 +500,7 @@ static int intel_sst_resume(struct device *dev) struct sst_block *block; if (!fw_save) - return intel_sst_runtime_resume(dev); + return 0; sst_set_fw_state_locked(ctx, SST_FW_LOADING); @@ -560,6 +545,5 @@ const struct dev_pm_ops intel_sst_pm = { .suspend = intel_sst_suspend, .resume = intel_sst_resume, .runtime_suspend = intel_sst_runtime_suspend, - .runtime_resume = intel_sst_runtime_resume, }; EXPORT_SYMBOL_GPL(intel_sst_pm); diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 5d56fcdd58d882..549af7d7f6d070 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -138,12 +138,31 @@ int sst_get_stream(struct intel_sst_drv *ctx, static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); + int ret = 0; - dev_dbg(ctx->dev, "state:%d", state); - if (state == true) - return pm_runtime_get_sync(dev); - else + if (state == true) { + ret = pm_runtime_get_sync(dev); + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", + atomic_read(&dev->power.usage_count)); + if (ret < 0) { + dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); + return ret; + } + if ((ctx->sst_state == SST_RESET) && + (atomic_read(&dev->power.usage_count) == 1)) { + ret = sst_load_fw(ctx); + if (ret) { + dev_err(dev, "FW download fail %d\n", ret); + sst_set_fw_state_locked(ctx, SST_RESET); + ret = sst_pm_runtime_put(ctx); + } + } + } else { + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", + atomic_read(&dev->power.usage_count)); return sst_pm_runtime_put(ctx); + } + return ret; } /* From 583e58a0f0e996008780fe4df0f7640890a9b69a Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 24 Feb 2015 11:39:46 +0530 Subject: [PATCH 055/411] ASoC: Intel: Remove ignore suspend support In our platform we want platform and codec driver routines to get invoked and don't need the machine routines so remove here Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index a5098d6f988be5..67db5101bc89cc 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -215,7 +215,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, @@ -246,7 +245,6 @@ static struct snd_soc_dai_link cht_dailink[] = { | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &cht_be_ssp2_ops, From 3f2dcbeaeb2badb951a68e7d525ff4e55d0b0678 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 24 Feb 2015 11:39:47 +0530 Subject: [PATCH 056/411] ASoC: Intel: Remove soc pm handling to allow platform driver handle it Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index 67db5101bc89cc..bc8dcacd5e6a08 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -283,7 +283,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-rt5672", - .pm = &snd_soc_pm_ops, }, .probe = snd_cht_mc_probe, }; From 34d7c3905adb9a9d8f8155857c76314125510817 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Sat, 21 Feb 2015 16:33:24 +0100 Subject: [PATCH 057/411] ASoC: improve usage of gpiod API MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Simplify drivers accordingly. Also there is an *_optional variant that serves well here. The sematics is slightly changed here by using it as error checking is more strict now: If GPIOLIB is not enabled an error is returned instead of just ignoring the gpio. On one hand this is bad for devices that don't "have" the respective gpio as the driver is failing now. On the other hand there is no means to assert that this gpio is really not needed or if only the driver to control it is not available. The latter is a real reason to fail and so it's defensive to fail here, too. Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/codecs/adau1977.c | 17 +++++------------ sound/soc/codecs/cs35l32.c | 19 ++++++------------- sound/soc/codecs/cs4265.c | 19 ++++++------------- sound/soc/codecs/sta350.c | 30 +++++++++--------------------- sound/soc/codecs/tas2552.c | 13 +++---------- 5 files changed, 29 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 70ab35744abadd..7ad8e156e2df90 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, adau1977->dvdd_reg = NULL; } - adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); - if (IS_ERR(adau1977->reset_gpio)) { - ret = PTR_ERR(adau1977->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return PTR_ERR(adau1977->reset_gpio); - adau1977->reset_gpio = NULL; - } + adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(adau1977->reset_gpio)) + return PTR_ERR(adau1977->reset_gpio); dev_set_drvdata(dev, adau1977); - if (adau1977->reset_gpio) { - ret = gpiod_direction_output(adau1977->reset_gpio, 0); - if (ret) - return ret; + if (adau1977->reset_gpio) ndelay(100); - } ret = adau1977_power_enable(adau1977); if (ret) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index f2b8aad21274ae..60598b23034111 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, } /* Reset the Device */ - cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs35l32->reset_gpio)) { - ret = PTR_ERR(cs35l32->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs35l32->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs35l32->reset_gpio, 0); - if (ret) - return ret; + cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l32->reset_gpio)) + return PTR_ERR(cs35l32->reset_gpio); + + if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); - } /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ce6086835ebdbe..cac48ddf3ba6bb 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, return ret; } - cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs4265->reset_gpio)) { - ret = PTR_ERR(cs4265->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs4265->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs4265->reset_gpio, 0); - if (ret) - return ret; + cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4265->reset_gpio)) + return PTR_ERR(cs4265->reset_gpio); + + if (cs4265->reset_gpio) { mdelay(1); gpiod_set_value_cansleep(cs4265->reset_gpio, 1); - } i2c_set_clientdata(i2c_client, cs4265); diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bda2ee18769ed7..669e3228241e39 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1213,27 +1213,15 @@ static int sta350_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta350->gpiod_nreset)) { - ret = PTR_ERR(sta350->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta350->gpiod_nreset, 0); - } - - sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); - if (IS_ERR(sta350->gpiod_power_down)) { - ret = PTR_ERR(sta350->gpiod_power_down); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_power_down = NULL; - } else { - gpiod_direction_output(sta350->gpiod_power_down, 0); - } + sta350->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_nreset)) + return PTR_ERR(sta350->gpiod_nreset); + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_power_down)) + return PTR_ERR(sta350->gpiod_power_down); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index ae23acdd270884..dfb4ff5cc9ea1a 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -485,16 +485,9 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable"); - if (IS_ERR(data->enable_gpio)) { - ret = PTR_ERR(data->enable_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - data->enable_gpio = NULL; - } else { - gpiod_direction_output(data->enable_gpio, 0); - } + data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); From 5890bd5256bc026c425361fa087dc05c7a24d853 Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 16 Feb 2015 22:02:47 +0100 Subject: [PATCH 058/411] ASoC: pcm512x: Rearrange to not repeat dacsrc_rate / dac_div Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 9974f201a08f44..f11c76f1acfedf 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -863,28 +863,29 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = sck_rate; } + osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); + if (osr_div > 128) { + dev_err(dev, "Failed to find OSR divider\n"); + return -EINVAL; + } + dac_div = DIV_ROUND_CLOSEST(dacsrc_rate, dac_rate); if (dac_div > 128) { dev_err(dev, "Failed to find DAC divider\n"); return -EINVAL; } + dac_rate = dacsrc_rate / dac_div; - ncp_div = DIV_ROUND_CLOSEST(dacsrc_rate / dac_div, 1536000); - if (ncp_div > 128 || dacsrc_rate / dac_div / ncp_div > 2048000) { + ncp_div = DIV_ROUND_CLOSEST(dac_rate, 1536000); + if (ncp_div > 128 || dac_rate / ncp_div > 2048000) { /* run NCP no faster than 2048000 Hz, but why? */ - ncp_div = DIV_ROUND_UP(dacsrc_rate / dac_div, 2048000); + ncp_div = DIV_ROUND_UP(dac_rate, 2048000); if (ncp_div > 128) { dev_err(dev, "Failed to find NCP divider\n"); return -EINVAL; } } - osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); - if (osr_div > 128) { - dev_err(dev, "Failed to find OSR divider\n"); - return -EINVAL; - } - idac = mck_rate / (dsp_div * sample_rate); ret = regmap_write(pcm512x->regmap, PCM512x_DSP_CLKDIV, dsp_div - 1); From f29933c9ae4b8f30c713186d3babb630c7cfb4f2 Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 23 Feb 2015 21:03:33 +0100 Subject: [PATCH 059/411] ASoC: pcm512x: Allow independently overclocking PLL, DAC and DSP When using non-standard rates, a relatively small amount of overclocking can make a big difference to a number of cases. - Not all rates are possible to achieve with the PLL, due to divider restrictions. - The higher oversampling rates that can be used by the DAC, the simpler the analog output filters get (mirror frequencies move up, away from the desired spectrum). - The more work the DSP can perform per sample, the better. For standard rates, there is little to gain as everything is designed just right, and the needed overclocking to make a real difference would be significant. Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 161 ++++++++++++++++++++++++++++++++++--- 1 file changed, 150 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index f11c76f1acfedf..4b5f1fe9be973a 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -54,6 +54,9 @@ struct pcm512x_priv { int pll_d; int pll_p; unsigned long real_pll; + unsigned long overclock_pll; + unsigned long overclock_dac; + unsigned long overclock_dsp; }; /* @@ -224,6 +227,90 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) } } +static int pcm512x_overclock_pll_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_pll; + return 0; +} + +static int pcm512x_overclock_pll_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_pll = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dsp; + return 0; +} + +static int pcm512x_overclock_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dsp = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dac_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dac; + return 0; +} + +static int pcm512x_overclock_dac_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dac = ucontrol->value.integer.value[0]; + return 0; +} + static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); @@ -328,6 +415,13 @@ SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), + +SOC_SINGLE_EXT("Max Overclock PLL", SND_SOC_NOPM, 0, 20, 0, + pcm512x_overclock_pll_get, pcm512x_overclock_pll_put), +SOC_SINGLE_EXT("Max Overclock DSP", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dsp_get, pcm512x_overclock_dsp_put), +SOC_SINGLE_EXT("Max Overclock DAC", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dac_get, pcm512x_overclock_dac_put), }; static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { @@ -346,6 +440,45 @@ static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { { "OUTR", NULL, "DACR" }, }; +static unsigned long pcm512x_pll_max(struct pcm512x_priv *pcm512x) +{ + return 25000000 + 25000000 * pcm512x->overclock_pll / 100; +} + +static unsigned long pcm512x_dsp_max(struct pcm512x_priv *pcm512x) +{ + return 50000000 + 50000000 * pcm512x->overclock_dsp / 100; +} + +static unsigned long pcm512x_dac_max(struct pcm512x_priv *pcm512x, + unsigned long rate) +{ + return rate + rate * pcm512x->overclock_dac / 100; +} + +static unsigned long pcm512x_sck_max(struct pcm512x_priv *pcm512x) +{ + if (!pcm512x->pll_out) + return 25000000; + return pcm512x_pll_max(pcm512x); +} + +static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x, + unsigned long dac_rate) +{ + /* + * If the DAC is not actually overclocked, use the good old + * NCP target rate... + */ + if (dac_rate <= 6144000) + return 1536000; + /* + * ...but if the DAC is in fact overclocked, bump the NCP target + * rate to get the recommended dividers even when overclocking. + */ + return pcm512x_dac_max(pcm512x, 1536000); +} + static const u32 pcm512x_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 384000, @@ -359,6 +492,7 @@ static const struct snd_pcm_hw_constraint_list constraints_slave = { static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { + struct pcm512x_priv *pcm512x = rule->private; struct snd_interval ranges[2]; int frame_size; @@ -377,7 +511,7 @@ static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, */ memset(ranges, 0, sizeof(ranges)); ranges[0].min = 8000; - ranges[0].max = 25000000 / frame_size / 2; + ranges[0].max = pcm512x_sck_max(pcm512x) / frame_size / 2; ranges[1].min = DIV_ROUND_UP(16000000, frame_size); ranges[1].max = 384000; break; @@ -408,7 +542,7 @@ static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream, return snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, pcm512x_hw_rule_rate, - NULL, + pcm512x, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -517,6 +651,8 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, unsigned long bclk_rate) { struct device *dev = dai->dev; + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); unsigned long sck_rate; int pow2; @@ -527,9 +663,10 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, * as many factors of 2 as possible, as that makes it easier * to find a fast DAC rate */ - pow2 = 1 << fls((25000000 - 16000000) / bclk_rate); + pow2 = 1 << fls((pcm512x_pll_max(pcm512x) - 16000000) / bclk_rate); for (; pow2; pow2 >>= 1) { - sck_rate = rounddown(25000000, bclk_rate * pow2); + sck_rate = rounddown(pcm512x_pll_max(pcm512x), + bclk_rate * pow2); if (sck_rate >= 16000000) break; } @@ -678,7 +815,7 @@ static unsigned long pcm512x_pllin_dac_rate(struct snd_soc_dai *dai, return 0; /* futile, quit early */ /* run DAC no faster than 6144000 Hz */ - for (dac_rate = rounddown(6144000, osr_rate); + for (dac_rate = rounddown(pcm512x_dac_max(pcm512x, 6144000), osr_rate); dac_rate; dac_rate -= osr_rate) { @@ -805,7 +942,7 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, osr_rate = 16 * sample_rate; /* run DSP no faster than 50 MHz */ - dsp_div = mck_rate > 50000000 ? 2 : 1; + dsp_div = mck_rate > pcm512x_dsp_max(pcm512x) ? 2 : 1; dac_rate = pcm512x_pllin_dac_rate(dai, osr_rate, pllin_rate); if (dac_rate) { @@ -836,7 +973,8 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = pllin_rate; } else { /* run DAC no faster than 6144000 Hz */ - unsigned long dac_mul = 6144000 / osr_rate; + unsigned long dac_mul = pcm512x_dac_max(pcm512x, 6144000) + / osr_rate; unsigned long sck_mul = sck_rate / osr_rate; for (; dac_mul; dac_mul--) { @@ -876,7 +1014,8 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, } dac_rate = dacsrc_rate / dac_div; - ncp_div = DIV_ROUND_CLOSEST(dac_rate, 1536000); + ncp_div = DIV_ROUND_CLOSEST(dac_rate, + pcm512x_ncp_target(pcm512x, dac_rate)); if (ncp_div > 128 || dac_rate / ncp_div > 2048000) { /* run NCP no faster than 2048000 Hz, but why? */ ncp_div = DIV_ROUND_UP(dac_rate, 2048000); @@ -938,11 +1077,11 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, return ret; } - if (sample_rate <= 48000) + if (sample_rate <= pcm512x_dac_max(pcm512x, 48000)) fssp = PCM512x_FSSP_48KHZ; - else if (sample_rate <= 96000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 96000)) fssp = PCM512x_FSSP_96KHZ; - else if (sample_rate <= 192000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 192000)) fssp = PCM512x_FSSP_192KHZ; else fssp = PCM512x_FSSP_384KHZ; From f23e860edbb3f2208c0ab3448e756689bb4a3760 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Sat, 14 Feb 2015 17:22:49 -0800 Subject: [PATCH 060/411] ASoC: core: Add extra dapm properties for Device Tree The current helper functions, snd_soc_of_parse_audio_simple_widgets() and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes without caring if they are already set by using build-in widgets and routes in the card driver. So there could be one of them, build-in one or Device Tree one, overrided by the other depending on which one was assigned later. This patch adds an extra pair of dapm_widgets and dapm_routes for DT use only so as to prevent unexpected overriding. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 16 ++++++++++++---- 2 files changed, 17 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade19562857..f66a1ef98a40bd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1071,11 +1071,16 @@ struct snd_soc_card { /* * Card-specific routes and widgets. + * Note: of_dapm_xxx for Device Tree; Otherwise for driver build-in. */ const struct snd_soc_dapm_widget *dapm_widgets; int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + const struct snd_soc_dapm_widget *of_dapm_widgets; + int num_of_dapm_widgets; + const struct snd_soc_dapm_route *of_dapm_routes; + int num_of_dapm_routes; bool fully_routed; struct work_struct deferred_resume_work; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca5bacb98..5c0658d49609d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1561,6 +1561,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); + if (card->of_dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets, + card->num_of_dapm_widgets); + /* initialise the sound card only once */ if (card->probe) { ret = card->probe(card); @@ -1616,6 +1620,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + if (card->of_dapm_routes) + snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, + card->num_of_dapm_routes); + for (i = 0; i < card->num_links; i++) { if (card->dai_link[i].dai_fmt) snd_soc_runtime_set_dai_fmt(&card->rtd[i], @@ -3223,8 +3231,8 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, widgets[i].name = wname; } - card->dapm_widgets = widgets; - card->num_dapm_widgets = num_widgets; + card->of_dapm_widgets = widgets; + card->num_of_dapm_widgets = num_widgets; return 0; } @@ -3308,8 +3316,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; - card->dapm_routes = routes; + card->num_of_dapm_routes = num_routes; + card->of_dapm_routes = routes; return 0; } From 3185878a70e721644b0e32ebbc0a039616551949 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Sat, 14 Feb 2015 17:22:50 -0800 Subject: [PATCH 061/411] ASoC: fsl-asoc-card: Add snd_soc_of_parse_audio_routing() This patch adds snd_soc_of_parse_audio_routing() to get dapm routes configurations via Device Tree. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 3f6959c8e2f71b..de438871040bd4 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -512,6 +512,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + goto asrc_fail; + } + /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; From e2cef68d5903cc2052e9f6e46b323b7ead695e73 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 10 Feb 2015 17:01:56 +0800 Subject: [PATCH 062/411] ASoC: rt286: add jack detection disable with NULL jack passed Some platforms, e.g. WSB, don't need jack detection when system is in Suspend, for power save reason. Here add headphone/mic jack detection disable feature with NULL jack passed in, when disabled, it will disable interrupt, and disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang Reviewed-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index f374840a5a7ce3..16723b167fbf0d 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -395,9 +395,20 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) rt286->jack = jack; - /* Send an initial empty report */ - snd_soc_jack_report(rt286->jack, 0, - SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + if (jack) { + /* enable IRQ */ + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1"); + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, rt286->jack->status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + } else { + /* disable IRQ */ + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0); + snd_soc_dapm_disable_pin(&codec->dapm, "LDO1"); + } + snd_soc_dapm_sync(&codec->dapm); return 0; } From 19449593d60b75654fe33a98c4fb8ff8a38ac1e0 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:27:10 +0300 Subject: [PATCH 063/411] sound: sys_timer: indent poll_def_tmr() correctly The indenting here was really whacky and not consistent from one line to the next. I also reverse the "if (opened)" and "if (tmr_running)" tests so that I could remove two indent levels. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sys_timer.c | 35 +++++++++++++++-------------------- 1 file changed, 15 insertions(+), 20 deletions(-) diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 9f039831114c2f..2226dda0eff0d9 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -50,29 +50,24 @@ tmr2ticks(int tmr_value) static void poll_def_tmr(unsigned long dummy) { + if (!opened) + return; + def_tmr.expires = (1) + jiffies; + add_timer(&def_tmr); - if (opened) - { + if (!tmr_running) + return; - { - def_tmr.expires = (1) + jiffies; - add_timer(&def_tmr); - } + spin_lock(&lock); + tmr_ctr++; + curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - if (tmr_running) - { - spin_lock(&lock); - tmr_ctr++; - curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - - if (curr_ticks >= next_event_time) - { - next_event_time = (unsigned long) -1; - sequencer_timer(0); - } - spin_unlock(&lock); - } - } + if (curr_ticks >= next_event_time) { + next_event_time = (unsigned long) -1; + sequencer_timer(0); + } + + spin_unlock(&lock); } static void From df403869e35f88e1e45483d31ee70b4aa3ed8896 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:30:36 +0300 Subject: [PATCH 064/411] sound/oss/opl3: remove some stray whitespace Removed an extra tab and a extra space character. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/opl3.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 607cee4d545ec6..b6d19adf8f4111 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -666,7 +666,7 @@ static int opl3_start_note (int dev, int voice, int note, int volume) opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data); data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3); - devc->voc[voice].keyon_byte = data; + devc->voc[voice].keyon_byte = data; opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data); if (voice_mode == 4) opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num + 3, data); @@ -717,7 +717,7 @@ static void freq_to_fnum (int freq, int *block, int *fnum) static void opl3_command (int io_addr, unsigned int addr, unsigned int val) { - int i; + int i; /* * The original 2-OP synth requires a quite long delay after writing to a From e214e5183d9da3b61f775d3ae7202ea8aa10ebed Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:31:43 +0300 Subject: [PATCH 065/411] sound/sb_ess: white space cleanups These weren't aligned on the same lines as the surrounding code and the printk was quite messy. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sb_ess.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index b47a69026f1b08..57f7d25a2cd300 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -604,7 +604,7 @@ static void ess_audio_output_block_audio2 ess_chgmixer (devc, 0x78, 0x03, 0x03); /* Go */ devc->irq_mode_16 = IMODE_OUTPUT; - devc->intr_active_16 = 1; + devc->intr_active_16 = 1; } static void ess_audio_output_block @@ -1183,17 +1183,12 @@ FKS_test (devc); chip = "ES1688"; } - printk ( KERN_INFO "ESS chip %s %s%s\n" - , chip - , ( devc->sbmo.esstype == ESSTYPE_DETECT || devc->sbmo.esstype == ESSTYPE_LIKE20 - ? "detected" - : "specified" - ) - , ( devc->sbmo.esstype == ESSTYPE_LIKE20 - ? " (kernel 2.0 compatible)" - : "" - ) - ); + printk(KERN_INFO "ESS chip %s %s%s\n", chip, + (devc->sbmo.esstype == ESSTYPE_DETECT || + devc->sbmo.esstype == ESSTYPE_LIKE20) ? + "detected" : "specified", + devc->sbmo.esstype == ESSTYPE_LIKE20 ? + " (kernel 2.0 compatible)" : ""); sprintf(name,"ESS %s AudioDrive (rev %d)", chip, ess_minor & 0x0f); } else { From f0418d46d6ad25be991d557c6c50d1e61b4ba690 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:32:16 +0300 Subject: [PATCH 066/411] sound/sb_midi: a couple indenting fixes Let's make things line up a little bit better. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sb_midi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c index f139028e85c0dd..551ee7557b4efb 100644 --- a/sound/oss/sb_midi.c +++ b/sound/oss/sb_midi.c @@ -179,14 +179,14 @@ void sb_dsp_midi_init(sb_devc * devc, struct module *owner) { printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n"); sound_unload_mididev(dev); - return; + return; } memcpy((char *) midi_devs[dev], (char *) &sb_midi_operations, sizeof(struct midi_operations)); if (owner) - midi_devs[dev]->owner = owner; - + midi_devs[dev]->owner = owner; + midi_devs[dev]->devc = devc; From 7f788e0cc07dba7f4eee6ffea30edee3af86e2a5 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:32:45 +0300 Subject: [PATCH 067/411] ALSA: azt3328: some indenting cleanups A few minor tweaks to make things line up correctly. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index a40a2b4c8fd7ac..33b2a0af1b5940 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1385,8 +1385,8 @@ snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, .running) && (!chip->codecs[peer_codecs[codec_type].other2] .running)); - } - if (call_function) + } + if (call_function) snd_azf3328_ctrl_enable_codecs(chip, enable); /* ...and adjust clock, too @@ -2126,7 +2126,8 @@ static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { static int snd_azf3328_pcm(struct snd_azf3328 *chip) { -enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */ + /* pcm devices */ + enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; struct snd_pcm *pcm; int err; From 9603cded0e2cef003a822985d84b5daff1c7232f Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:33:57 +0300 Subject: [PATCH 068/411] ALSA: cmipci: remove a stray space character Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1d0f2cad2f5a64..6cf464d9043d29 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2062,7 +2062,7 @@ static int snd_cmipci_get_volume(struct snd_kcontrol *kcontrol, val = (snd_cmipci_mixer_read(cm, reg.right_reg) >> reg.right_shift) & reg.mask; if (reg.invert) val = reg.mask - val; - ucontrol->value.integer.value[1] = val; + ucontrol->value.integer.value[1] = val; } spin_unlock_irq(&cm->reg_lock); return 0; From 5bb400ce4a9b100a2dd3f5db17c4c76877cecade Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 25 Feb 2015 21:37:52 +0530 Subject: [PATCH 069/411] ASoC: Intel: wrap runtime_pm usage count under CONFIG_PM The struct dev_pm_ops defines usage_count only when CONFIG_PM is defined. So we should use this variable only in cases where this falg is true. So we define a local variable and read the value under this flag. In non PM cases, we set this to 1. Reported-by: kbuild test robot Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 549af7d7f6d070..f0e4b99b3aeb13 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -139,17 +139,23 @@ static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); int ret = 0; + int usage_count = 0; + +#ifdef CONFIG_PM + usage_count = atomic_read(&dev->power.usage_count); +#else + usage_count = 1; +#endif if (state == true) { ret = pm_runtime_get_sync(dev); - dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } - if ((ctx->sst_state == SST_RESET) && - (atomic_read(&dev->power.usage_count) == 1)) { + if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { ret = sst_load_fw(ctx); if (ret) { dev_err(dev, "FW download fail %d\n", ret); @@ -158,8 +164,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { - dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } return ret; From 92b2ad2c9e18ca2bfa8727af7edcd372d9acaac4 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 24 Feb 2015 22:39:04 -0800 Subject: [PATCH 070/411] ASoC: max98357a: Use standard DAI names Use the standard naming convention for the codec DAI. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 4ee23fbc4e1237..bf3e933ee89575 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -85,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = "max98357a", + .name = "HiFi", .playback = { - .stream_name = "max98357a-playback", + .stream_name = "HiFi Playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, From bb573928e187fc5b1f91c3a7684791d5dfcca640 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 09:26:04 +0100 Subject: [PATCH 071/411] ALSA: hda - Drop power_save value indirection in hda_bus We used to pass the power_save option value to hda_bus via a given pointer. This was needed to refer to the value from the HD-audio core side. However, after the transition to the runtime PM, this is no longer needed. This patch drops the power_save value indirection in hda_bus above, and let the controller driver reprograms the autosuspend value explicitly by a new helper, snd_hda_set_power_save(). Without this call, the HD-audio core doesn't set up the autosuspend and flip the runtime PM. (User may still be able to set up via sysfs, though.) Along with this change, the pointer argument of azx_bus_create() is dropped as well. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 34 ++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 5 ++--- sound/pci/hda/hda_controller.c | 5 +---- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 10 ++++------ sound/pci/hda/hda_tegra.c | 5 +++-- 6 files changed, 29 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d0dbc62c91474a..36cebe0e76b2f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1175,10 +1175,8 @@ static int snd_hda_codec_dev_register(struct snd_device *device) struct hda_codec *codec = device->device_data; snd_hda_register_beep_device(codec); - if (device_is_registered(hda_codec_dev(codec))) { - snd_hda_power_sync(codec); + if (device_is_registered(hda_codec_dev(codec))) pm_runtime_enable(hda_codec_dev(codec)); - } return 0; } @@ -4778,21 +4776,10 @@ void snd_hda_power_down(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_power_down); -/** - * snd_hda_power_sync - Synchronize the power_save option - * @codec: HD-audio codec - * - * Synchronize the runtime PM autosuspend state from the power_save option. - */ -void snd_hda_power_sync(struct hda_codec *codec) +static void codec_set_power_save(struct hda_codec *codec, int delay) { struct device *dev = hda_codec_dev(codec); - int delay; - if (!codec->bus->power_save) - return; - - delay = *codec->bus->power_save * 1000; if (delay > 0) { pm_runtime_set_autosuspend_delay(dev, delay); pm_runtime_use_autosuspend(dev); @@ -4804,7 +4791,22 @@ void snd_hda_power_sync(struct hda_codec *codec) pm_runtime_forbid(dev); } } -EXPORT_SYMBOL_GPL(snd_hda_power_sync); + +/** + * snd_hda_set_power_save - reprogram autosuspend for the given delay + * @bus: HD-audio bus + * @delay: autosuspend delay in msec, 0 = off + * + * Synchronize the runtime PM autosuspend state from the power_save option. + */ +void snd_hda_set_power_save(struct hda_bus *bus, int delay) +{ + struct hda_codec *c; + + list_for_each_entry(c, &bus->codec_list, list) + codec_set_power_save(c, delay); +} +EXPORT_SYMBOL_GPL(snd_hda_set_power_save); /** * snd_hda_check_amp_list_power - Check the amp list and update the power diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 593956fc384b09..89908f5b9601af 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -122,7 +122,6 @@ struct hda_bus { void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; /* codec linked list */ @@ -592,12 +591,12 @@ const char *snd_hda_get_jack_location(u32 cfg); #ifdef CONFIG_PM void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); -void snd_hda_power_sync(struct hda_codec *codec); +void snd_hda_set_power_save(struct hda_bus *bus, int delay); void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} -static inline void snd_hda_power_sync(struct hda_codec *codec) {} +static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 522c54f9474077..cfe2c55296b6c4 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1838,7 +1838,7 @@ static struct hda_bus_ops bus_ops = { }; /* HD-audio bus initialization */ -int azx_bus_create(struct azx *chip, const char *model, int *power_save_to) +int azx_bus_create(struct azx *chip, const char *model) { struct hda_bus *bus; int err; @@ -1852,9 +1852,6 @@ int azx_bus_create(struct azx *chip, const char *model, int *power_save_to) bus->pci = chip->pci; bus->modelname = model; bus->ops = bus_ops; -#ifdef CONFIG_PM - bus->power_save = power_save_to; -#endif if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 0d09aa6b49ac82..e4f46a21698a58 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -432,7 +432,7 @@ void azx_enter_link_reset(struct azx *chip); irqreturn_t azx_interrupt(int irq, void *dev_id); /* Codec interface */ -int azx_bus_create(struct azx *chip, const char *model, int *power_save_to); +int azx_bus_create(struct azx *chip, const char *model); int azx_probe_codecs(struct azx *chip, unsigned int max_slots); int azx_codec_configure(struct azx *chip); int azx_init_stream(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 26510e60cbe2b3..40540048b00298 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -173,7 +173,6 @@ static struct kernel_param_ops param_ops_xint = { #define param_check_xint param_check_int static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; -static int *power_save_addr = &power_save; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -186,7 +185,7 @@ static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #else -static int *power_save_addr; +#define power_save 0 #endif /* CONFIG_PM */ static int align_buffer_size = -1; @@ -740,7 +739,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) { struct hda_intel *hda; struct azx *chip; - struct hda_codec *c; int prev = power_save; int ret = param_set_int(val, kp); @@ -752,8 +750,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) chip = &hda->chip; if (!chip->bus || chip->disabled) continue; - list_for_each_entry(c, &chip->bus->codec_list, list) - snd_hda_power_sync(c); + snd_hda_set_power_save(chip->bus, power_save * 1000); } mutex_unlock(&card_list_lock); return 0; @@ -1889,7 +1886,7 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_bus_create(chip, model[dev], power_save_addr); + err = azx_bus_create(chip, model[dev]); if (err < 0) goto out_free; @@ -1933,6 +1930,7 @@ static int azx_probe_continue(struct azx *chip) power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); + snd_hda_set_power_save(chip->bus, power_save * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index f6949e413a501a..42bc17655df055 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -81,7 +81,7 @@ module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); #else -static int power_save = 0; +#define power_save 0 #endif /* @@ -496,7 +496,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create codec instances */ - err = azx_bus_create(chip, NULL, &power_save); + err = azx_bus_create(chip, NULL); if (err < 0) goto out_free; @@ -525,6 +525,7 @@ static int hda_tegra_probe(struct platform_device *pdev) chip->running = 1; power_down_all_codecs(chip); azx_notifier_register(chip); + snd_hda_set_power_save(chip->bus, power_save * 1000); return 0; From 55ed9cd1feee80764937913afe760161b86cfb11 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 17:35:32 +0100 Subject: [PATCH 072/411] ALSA: hda - Replace bus pm_notify with the standard runtime PM framework Now the final bit of runtime PM cleanup: instead of manual notification of the power up/down of the codec via hda_bus pm_notify ops, use the standard runtime PM feature. The child codec device will kick off the runtime PM of the parent (PCI) device upon suspend/resume automatically. For managing whether the link can be really turned off, we use the bit flags bus->codec_powered instead of the earlier bus->power_keep_link_on. flag. Each codec driver is responsible to set/clear the bit flag, and the controller device can be turned off only when all these bits are cleared. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 33 ++++++++------------------------- sound/pci/hda/hda_codec.h | 7 +------ sound/pci/hda/hda_controller.c | 19 ------------------- sound/pci/hda/hda_intel.c | 5 ++--- sound/pci/hda/patch_sigmatel.c | 4 +++- 5 files changed, 14 insertions(+), 54 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 36cebe0e76b2f9..33b8b71f8eaf53 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -46,23 +46,9 @@ #define codec_in_pm(codec) atomic_read(&(codec)->in_pm) #define hda_codec_is_power_on(codec) \ (!pm_runtime_suspended(hda_codec_dev(codec))) - -static void hda_call_pm_notify(struct hda_codec *codec, bool power_up) -{ - struct hda_bus *bus = codec->bus; - - if ((power_up && codec->pm_up_notified) || - (!power_up && !codec->pm_up_notified)) - return; - if (bus->ops.pm_notify) - bus->ops.pm_notify(bus, power_up); - codec->pm_up_notified = power_up; -} - #else #define codec_in_pm(codec) 0 #define hda_codec_is_power_on(codec) 1 -#define hda_call_pm_notify(codec, state) {} #endif /** @@ -1152,7 +1138,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) snd_array_free(&codec->spdif_out); remove_conn_list(codec); codec->bus->caddr_tbl[codec->addr] = NULL; - hda_call_pm_notify(codec, false); /* cancel leftover refcounts */ + clear_bit(codec->addr, &codec->bus->codec_powered); snd_hda_sysfs_clear(codec); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); @@ -1277,10 +1263,10 @@ int snd_hda_codec_new(struct hda_bus *bus, * the caller has to power down appropriatley after initialization * phase. */ + set_bit(codec->addr, &bus->codec_powered); pm_runtime_set_active(hda_codec_dev(codec)); pm_runtime_get_noresume(hda_codec_dev(codec)); codec->power_jiffies = jiffies; - hda_call_pm_notify(codec, true); #endif snd_hda_sysfs_init(codec); @@ -1340,11 +1326,6 @@ int snd_hda_codec_new(struct hda_bus *bus, #endif codec->epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); -#ifdef CONFIG_PM - if (!codec->d3_stop_clk || !codec->epss) - bus->power_keep_link_on = 1; -#endif - /* power-up all before initialization */ hda_set_power_state(codec, AC_PWRST_D0); @@ -3954,7 +3935,6 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_mark_cmd_cache_dirty(codec); codec->power_jiffies = jiffies; - hda_call_pm_notify(codec, true); hda_set_power_state(codec, AC_PWRST_D0); restore_shutup_pins(codec); @@ -3986,14 +3966,17 @@ static int hda_codec_runtime_suspend(struct device *dev) for (i = 0; i < codec->num_pcms; i++) snd_pcm_suspend_all(codec->pcm_info[i].pcm); state = hda_call_codec_suspend(codec); - if (!codec->bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) - hda_call_pm_notify(codec, false); + if (codec->d3_stop_clk && codec->epss && (state & AC_PWRST_CLK_STOP_OK)) + clear_bit(codec->addr, &codec->bus->codec_powered); return 0; } static int hda_codec_runtime_resume(struct device *dev) { - hda_call_codec_resume(dev_to_hda_codec(dev)); + struct hda_codec *codec = dev_to_hda_codec(dev); + + set_bit(codec->addr, &codec->bus->codec_powered); + hda_call_codec_resume(codec); pm_runtime_mark_last_busy(dev); return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 89908f5b9601af..457fc589eb469d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -83,10 +83,6 @@ struct hda_bus_ops { struct hda_pcm *pcm); /* reset bus for retry verb */ void (*bus_reset)(struct hda_bus *bus); -#ifdef CONFIG_PM - /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus, bool power_up); -#endif #ifdef CONFIG_SND_HDA_DSP_LOADER /* prepare DSP transfer */ int (*load_dsp_prepare)(struct hda_bus *bus, unsigned int format, @@ -150,10 +146,10 @@ struct hda_bus { unsigned int rirb_error:1; /* error in codec communication */ unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ - unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ + unsigned long codec_powered; /* bit flags of powered codecs */ }; /* @@ -372,7 +368,6 @@ struct hda_codec { unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - unsigned int pm_up_notified:1; /* PM notified to controller */ atomic_t in_pm; /* suspend/resume being performed */ unsigned long power_on_acct; unsigned long power_off_acct; diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index cfe2c55296b6c4..789ca66c309436 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1785,22 +1785,6 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 0; } -#ifdef CONFIG_PM -/* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus, bool power_up) -{ - struct azx *chip = bus->private_data; - - if (!azx_has_pm_runtime(chip)) - return; - - if (power_up) - pm_runtime_get_sync(chip->card->dev); - else - pm_runtime_put_sync(chip->card->dev); -} -#endif - static int get_jackpoll_interval(struct azx *chip) { int i; @@ -1827,9 +1811,6 @@ static struct hda_bus_ops bus_ops = { .get_response = azx_get_response, .attach_pcm = azx_attach_pcm_stream, .bus_reset = azx_bus_reset, -#ifdef CONFIG_PM - .pm_notify = azx_power_notify, -#endif #ifdef CONFIG_SND_HDA_DSP_LOADER .load_dsp_prepare = azx_load_dsp_prepare, .load_dsp_trigger = azx_load_dsp_trigger, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 40540048b00298..738d332351d5ba 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -918,7 +918,8 @@ static int azx_runtime_idle(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!power_save_controller || !azx_has_pm_runtime(chip)) + if (!power_save_controller || !azx_has_pm_runtime(chip) || + chip->bus->codec_powered) return -EBUSY; return 0; @@ -1084,7 +1085,6 @@ static int azx_free(struct azx *chip) azx_stop_chip(chip); } - pci->dev.power.ignore_children = 0; /* FIXME */ if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); if (chip->msi) @@ -1794,7 +1794,6 @@ static int azx_probe(struct pci_dev *pci, return err; } - pci->dev.power.ignore_children = 1; /* FIXME */ err = azx_create(card, pci, dev, pci_id->driver_data, &pci_hda_ops, &chip); if (err < 0) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6a216305621666..2956a6ba6bf018 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2132,8 +2132,10 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ +#ifdef CONFIG_PM /* resetting controller clears GPIO, so we need to keep on */ - codec->bus->power_keep_link_on = 1; + codec->d3_stop_clk = 0; +#endif } } From 709949fbe9632941585dcacabc8a66010030ed10 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 09:58:14 +0100 Subject: [PATCH 073/411] ALSA: hda - Power down codec automatically at registration So far, we let the controller driver power down the all codecs at the end of probe. But this can be done better in the codec's dev_register callback. This results in the reduction of duplicated codes in each control driver. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 +++-- sound/pci/hda/hda_intel.c | 14 -------------- sound/pci/hda/hda_tegra.c | 12 ------------ 3 files changed, 3 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 33b8b71f8eaf53..6580a367023e25 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1163,6 +1163,8 @@ static int snd_hda_codec_dev_register(struct snd_device *device) snd_hda_register_beep_device(codec); if (device_is_registered(hda_codec_dev(codec))) pm_runtime_enable(hda_codec_dev(codec)); + /* it was powered up in snd_hda_codec_new(), now all done */ + snd_hda_power_down(codec); return 0; } @@ -1260,8 +1262,7 @@ int snd_hda_codec_new(struct hda_bus *bus, #ifdef CONFIG_PM /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. - * the caller has to power down appropriatley after initialization - * phase. + * it's powered down later in snd_hda_codec_dev_register(). */ set_bit(codec->addr, &bus->codec_powered); pm_runtime_set_active(hda_codec_dev(codec)); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 738d332351d5ba..e75e8137e296cd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1604,19 +1604,6 @@ static int azx_first_init(struct azx *chip) return 0; } -static void power_down_all_codecs(struct azx *chip) -{ -#ifdef CONFIG_PM - /* The codecs were powered up in snd_hda_codec_new(). - * Now all initialization done, so turn them down if possible - */ - struct hda_codec *codec; - list_for_each_entry(codec, &chip->bus->codec_list, list) { - snd_hda_power_down(codec); - } -#endif -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* callback from request_firmware_nowait() */ static void azx_firmware_cb(const struct firmware *fw, void *context) @@ -1926,7 +1913,6 @@ static int azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; - power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); snd_hda_set_power_save(chip->bus, power_save * 1000); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 42bc17655df055..1359fdd20f02bd 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -337,17 +337,6 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) return 0; } -/* - * The codecs were powered up in snd_hda_codec_new(). - * Now all initialization done, so turn them down if possible - */ -static void power_down_all_codecs(struct azx *chip) -{ - struct hda_codec *codec; - list_for_each_entry(codec, &chip->bus->codec_list, list) - snd_hda_power_down(codec); -} - static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) { struct snd_card *card = chip->card; @@ -523,7 +512,6 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; chip->running = 1; - power_down_all_codecs(chip); azx_notifier_register(chip); snd_hda_set_power_save(chip->bus, power_save * 1000); From 777ae1946813f1dea24d908c8f318754f090f41f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2015 14:35:59 +0100 Subject: [PATCH 074/411] ALSA: hda - Set parent of input beep devices Set the card device as the parent like other sound devices instead of leaving it empty. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index d6be4e852c4d5c..e98438e95e79dd 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -160,6 +160,7 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->name = "HDA Digital PCBeep"; input_dev->phys = beep->phys; input_dev->id.bustype = BUS_PCI; + input_dev->dev.parent = &codec->bus->card->card_dev; input_dev->id.vendor = codec->vendor_id >> 16; input_dev->id.product = codec->vendor_id & 0xffff; From 7e40b80da452770878943edfe7da80f10f8d25da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2015 09:42:04 +0100 Subject: [PATCH 075/411] ALSA: hda - Remove channel mode helper functions They are no longer used, let's kill them. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 82 --------------------------------------- sound/pci/hda/hda_local.h | 23 ----------- 2 files changed, 105 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6580a367023e25..db86b446743c91 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4843,88 +4843,6 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); #endif -/* - * Channel mode helper - */ - -/** - * snd_hda_ch_mode_info - Info callback helper for the channel mode enum - * @codec: the HDA codec - * @uinfo: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - */ -int snd_hda_ch_mode_info(struct hda_codec *codec, - struct snd_ctl_elem_info *uinfo, - const struct hda_channel_mode *chmode, - int num_chmodes) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = num_chmodes; - if (uinfo->value.enumerated.item >= num_chmodes) - uinfo->value.enumerated.item = num_chmodes - 1; - sprintf(uinfo->value.enumerated.name, "%dch", - chmode[uinfo->value.enumerated.item].channels); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info); - -/** - * snd_hda_ch_mode_get - Get callback helper for the channel mode enum - * @codec: the HDA codec - * @ucontrol: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - * @max_channels: max number of channels - */ -int snd_hda_ch_mode_get(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int max_channels) -{ - int i; - - for (i = 0; i < num_chmodes; i++) { - if (max_channels == chmode[i].channels) { - ucontrol->value.enumerated.item[0] = i; - break; - } - } - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get); - -/** - * snd_hda_ch_mode_put - Put callback helper for the channel mode enum - * @codec: the HDA codec - * @ucontrol: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - * @max_channelsp: pointer to store the max channels - */ -int snd_hda_ch_mode_put(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int *max_channelsp) -{ - unsigned int mode; - - mode = ucontrol->value.enumerated.item[0]; - if (mode >= num_chmodes) - return -EINVAL; - if (*max_channelsp == chmode[mode].channels) - return 0; - /* change the current channel setting */ - *max_channelsp = chmode[mode].channels; - if (chmode[mode].sequence) - snd_hda_sequence_write_cache(codec, chmode[mode].sequence); - return 1; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put); - /* * input MUX helper */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 2f7d9646a41d5c..8588813163e31d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -272,29 +272,6 @@ int snd_hda_add_imux_item(struct hda_codec *codec, struct hda_input_mux *imux, const char *label, int index, int *type_index_ret); -/* - * Channel mode helper - */ -struct hda_channel_mode { - int channels; - const struct hda_verb *sequence; -}; - -int snd_hda_ch_mode_info(struct hda_codec *codec, - struct snd_ctl_elem_info *uinfo, - const struct hda_channel_mode *chmode, - int num_chmodes); -int snd_hda_ch_mode_get(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int max_channels); -int snd_hda_ch_mode_put(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int *max_channelsp); - /* * Multi-channel / digital-out PCM helper */ From 820cc6cf2c552155ea919e596a85e1f4e5dfa2b5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 12:50:46 +0100 Subject: [PATCH 076/411] ALSA: hda - Clear pcm pointer assigned to hda_pcm at device removal We leave the pcm field of struct hda_pcm at removal of each device, so far. This hasn't been a problem since unbinding the codec driver isn't supposed to happen and another route via snd_hda_codec_reset() clears all the once. However, for a proper unbind implementation, we need to care about it. This patch does the thing above properly: - Include struct hda_pcm pointer instead of struct hda_pcm_stream pointers in struct azx_dev. This allows us to point the hda_pcm object at dev_free callback. - Introduce to_hda_pcm_stream() macro for better readability. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 22 +++++++++++++++------- sound/pci/hda/hda_controller.h | 2 +- 2 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 789ca66c309436..1695f0e2bd9d47 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -258,11 +258,18 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static inline struct hda_pcm_stream * +to_hda_pcm_stream(struct snd_pcm_substream *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + return &apcm->info->stream[substream->stream]; +} + static u64 azx_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); u64 codec_frames, codec_nsecs; if (!hinfo->ops.get_delay) @@ -398,7 +405,7 @@ static int azx_setup_periods(struct azx *chip, static int azx_pcm_close(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev = get_azx_dev(substream); unsigned long flags; @@ -440,7 +447,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx_dev *azx_dev = get_azx_dev(substream); struct azx *chip = apcm->chip; - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); int err; /* reset BDL address */ @@ -467,7 +474,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev = get_azx_dev(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int bufsize, period_bytes, format_val, stream_tag; int err; @@ -707,7 +714,7 @@ unsigned int azx_get_position(struct azx *chip, if (substream->runtime) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); if (chip->get_delay[stream]) delay += chip->get_delay[stream](chip, azx_dev, pos); @@ -790,7 +797,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { static int azx_pcm_open(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev; struct snd_pcm_runtime *runtime = substream->runtime; @@ -904,6 +911,7 @@ static void azx_pcm_free(struct snd_pcm *pcm) struct azx_pcm *apcm = pcm->private_data; if (apcm) { list_del(&apcm->list); + apcm->info->pcm = NULL; kfree(apcm); } } @@ -940,6 +948,7 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; + apcm->info = cpcm; pcm->private_data = apcm; pcm->private_free = azx_pcm_free; if (cpcm->pcm_type == HDA_PCM_TYPE_MODEM) @@ -947,7 +956,6 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, list_add_tail(&apcm->list, &chip->pcm_list); cpcm->pcm = pcm; for (s = 0; s < 2; s++) { - apcm->hinfo[s] = &cpcm->stream[s]; if (cpcm->stream[s].substreams) snd_pcm_set_ops(pcm, s, &azx_pcm_ops); } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index e4f46a21698a58..94c1a4719f7fbd 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -283,7 +283,7 @@ struct azx_pcm { struct azx *chip; struct snd_pcm *pcm; struct hda_codec *codec; - struct hda_pcm_stream *hinfo[2]; + struct hda_pcm *info; struct list_head list; }; From 223c055aa0eb7e606eb7132e339ce66bb8d7be0d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 18 Dec 2014 11:32:52 +0800 Subject: [PATCH 077/411] ASoC: rt5670: set platform data by dmi This patch set specific data according to dmi data. Signed-off-by: Jin, Yao Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8a0833de1665bc..cd47ef1f5561d5 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_intel_braswell[] = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_intel_braswell)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); From 64e89e5f55484d289c8b326521e5a12291e2283e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 15 Dec 2014 15:42:33 +0800 Subject: [PATCH 078/411] ASoC: rt5670: Add runtime PM support This patch adds runtime PM support on rt5670 codec. Signed-off-by: Lin Mengdong Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index cd47ef1f5561d5..78d85de8af6fc0 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -2734,18 +2735,26 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) goto err; + pm_runtime_put(&i2c->dev); + return 0; err: + pm_runtime_disable(&i2c->dev); + return ret; } static int rt5670_i2c_remove(struct i2c_client *i2c) { + pm_runtime_disable(&i2c->dev); snd_soc_unregister_codec(&i2c->dev); return 0; From 77e3ea2801c8ca4700e6b17053b625b8a981ac77 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 15 Dec 2014 15:42:34 +0800 Subject: [PATCH 079/411] ASoC: rt5670: Keep sysclk on if JD func is used System clock is necessary for rt5670 JD function. We assume system clock source will be set in machine driver. So there are two things left we should do in codec driver. 1. Set sysclk to codec internal clock in probe since machine driver may not do that before JD function is registered. 2. Power up PLL once sysclk source is switched to PLL. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 78d85de8af6fc0..0a027bc943991c 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2190,6 +2190,13 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) return 0; + if (rt5670->pdata.jd_mode) { + if (clk_id == RT5670_SCLK_S_PLL1) + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + else + snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_sync(&codec->dapm); + } switch (clk_id) { case RT5670_SCLK_S_MCLK: reg_val |= RT5670_SCLK_SRC_MCLK; @@ -2628,6 +2635,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } if (rt5670->pdata.jd_mode) { + regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); + rt5670->sysclk = 0; + rt5670->sysclk_src = RT5670_SCLK_S_RCCLK; regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, RT5670_PWR_MB, RT5670_PWR_MB); regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, From 3aebec3a701e70d6fe2816891e5abea066492779 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 7 Jan 2015 10:19:06 +0800 Subject: [PATCH 080/411] ASoC: rt5670: redefine ASRC control registers 0x84 and 0x85 The previous definition of registers 0x84 and 0x85 doesn't match the datasheet. So this patch removes the wrong definition and writes a new one for the two registers. Signed-off-by: Bard Liao Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.h | 65 +++++++++++++++------------------------ 1 file changed, 24 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 84857bdaa78bff..82553b1726cdcd 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1023,50 +1023,33 @@ #define RT5670_DMIC_2_M_NOR (0x0 << 8) #define RT5670_DMIC_2_M_ASYN (0x1 << 8) +/* ASRC clock source selection (0x84, 0x85) */ +#define RT5670_CLK_SEL_SYS (0x0) +#define RT5670_CLK_SEL_I2S1_ASRC (0x1) +#define RT5670_CLK_SEL_I2S2_ASRC (0x2) +#define RT5670_CLK_SEL_I2S3_ASRC (0x3) +#define RT5670_CLK_SEL_SYS2 (0x5) +#define RT5670_CLK_SEL_SYS3 (0x6) + /* ASRC Control 2 (0x84) */ -#define RT5670_MDA_L_M_MASK (0x1 << 15) -#define RT5670_MDA_L_M_SFT 15 -#define RT5670_MDA_L_M_NOR (0x0 << 15) -#define RT5670_MDA_L_M_ASYN (0x1 << 15) -#define RT5670_MDA_R_M_MASK (0x1 << 14) -#define RT5670_MDA_R_M_SFT 14 -#define RT5670_MDA_R_M_NOR (0x0 << 14) -#define RT5670_MDA_R_M_ASYN (0x1 << 14) -#define RT5670_MAD_L_M_MASK (0x1 << 13) -#define RT5670_MAD_L_M_SFT 13 -#define RT5670_MAD_L_M_NOR (0x0 << 13) -#define RT5670_MAD_L_M_ASYN (0x1 << 13) -#define RT5670_MAD_R_M_MASK (0x1 << 12) -#define RT5670_MAD_R_M_SFT 12 -#define RT5670_MAD_R_M_NOR (0x0 << 12) -#define RT5670_MAD_R_M_ASYN (0x1 << 12) -#define RT5670_ADC_M_MASK (0x1 << 11) -#define RT5670_ADC_M_SFT 11 -#define RT5670_ADC_M_NOR (0x0 << 11) -#define RT5670_ADC_M_ASYN (0x1 << 11) -#define RT5670_STO_DAC_M_MASK (0x1 << 5) -#define RT5670_STO_DAC_M_SFT 5 -#define RT5670_STO_DAC_M_NOR (0x0 << 5) -#define RT5670_STO_DAC_M_ASYN (0x1 << 5) -#define RT5670_I2S1_R_D_MASK (0x1 << 4) -#define RT5670_I2S1_R_D_SFT 4 -#define RT5670_I2S1_R_D_DIS (0x0 << 4) -#define RT5670_I2S1_R_D_EN (0x1 << 4) -#define RT5670_I2S2_R_D_MASK (0x1 << 3) -#define RT5670_I2S2_R_D_SFT 3 -#define RT5670_I2S2_R_D_DIS (0x0 << 3) -#define RT5670_I2S2_R_D_EN (0x1 << 3) -#define RT5670_PRE_SCLK_MASK (0x3) -#define RT5670_PRE_SCLK_SFT 0 -#define RT5670_PRE_SCLK_512 (0x0) -#define RT5670_PRE_SCLK_1024 (0x1) -#define RT5670_PRE_SCLK_2048 (0x2) +#define RT5670_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5670_DA_STO_CLK_SEL_SFT 12 +#define RT5670_DA_MONOL_CLK_SEL_MASK (0xf << 8) +#define RT5670_DA_MONOL_CLK_SEL_SFT 8 +#define RT5670_DA_MONOR_CLK_SEL_MASK (0xf << 4) +#define RT5670_DA_MONOR_CLK_SEL_SFT 4 +#define RT5670_AD_STO1_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_STO1_CLK_SEL_SFT 0 /* ASRC Control 3 (0x85) */ -#define RT5670_I2S1_RATE_MASK (0xf << 12) -#define RT5670_I2S1_RATE_SFT 12 -#define RT5670_I2S2_RATE_MASK (0xf << 8) -#define RT5670_I2S2_RATE_SFT 8 +#define RT5670_UP_CLK_SEL_MASK (0xf << 12) +#define RT5670_UP_CLK_SEL_SFT 12 +#define RT5670_DOWN_CLK_SEL_MASK (0xf << 8) +#define RT5670_DOWN_CLK_SEL_SFT 8 +#define RT5670_AD_MONOL_CLK_SEL_MASK (0xf << 4) +#define RT5670_AD_MONOL_CLK_SEL_SFT 4 +#define RT5670_AD_MONOR_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_MONOR_CLK_SEL_SFT 0 /* ASRC Control 4 (0x89) */ #define RT5670_I2S1_PD_MASK (0x7 << 12) From ea232b3f7233f9242e5a1287377fedb6972dfea4 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 7 Jan 2015 10:19:12 +0800 Subject: [PATCH 081/411] ASoC: rt5670: add API to select ASRC clock source When codec is in slave mode, ASRC can suppress noise for asynchronous MCLK and LRCLK or special I2S format. This patch defines an API to select the clock source for specified filters. And the codec driver will turn on ASRC for these filters if ASRC is selected as their clock source. Signed-off-by: Bard Liao Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 83 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670.h | 15 +++++++ 2 files changed, 98 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0a027bc943991c..0632b7458a5322 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -590,6 +590,89 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } + +/** + * rt5670_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5670 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + unsigned int asrc2_mask = 0, asrc2_value = 0; + unsigned int asrc3_mask = 0, asrc3_value = 0; + + if (clk_src > RT5670_CLK_SEL_SYS3) + return -EINVAL; + + if (filter_mask & RT5670_DA_STEREO_FILTER) { + asrc2_mask |= RT5670_DA_STO_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5670_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_L_FILTER) { + asrc2_mask |= RT5670_DA_MONOL_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_R_FILTER) { + asrc2_mask |= RT5670_DA_MONOR_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_STEREO_FILTER) { + asrc2_mask |= RT5670_AD_STO1_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5670_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_L_FILTER) { + asrc3_mask |= RT5670_AD_MONOL_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_R_FILTER) { + asrc3_mask |= RT5670_AD_MONOR_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_UP_RATE_FILTER) { + asrc3_mask |= RT5670_UP_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_UP_CLK_SEL_MASK) + | (clk_src << RT5670_UP_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DOWN_RATE_FILTER) { + asrc3_mask |= RT5670_DOWN_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_DOWN_CLK_SEL_MASK) + | (clk_src << RT5670_DOWN_CLK_SEL_SFT); + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5670_ASRC_2, + asrc2_mask, asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5670_ASRC_3, + asrc3_mask, asrc3_value); + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 82553b1726cdcd..0a67adbcfbc37f 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1969,6 +1969,21 @@ enum { RT5670_DMIC_DATA_GPIO5, }; +/* filter mask */ +enum { + RT5670_DA_STEREO_FILTER = 0x1, + RT5670_DA_MONO_L_FILTER = (0x1 << 1), + RT5670_DA_MONO_R_FILTER = (0x1 << 2), + RT5670_AD_STEREO_FILTER = (0x1 << 3), + RT5670_AD_MONO_L_FILTER = (0x1 << 4), + RT5670_AD_MONO_R_FILTER = (0x1 << 5), + RT5670_UP_RATE_FILTER = (0x1 << 6), + RT5670_DOWN_RATE_FILTER = (0x1 << 7), +}; + +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; From ab1f70952f61504f60805f13660c8740adcbe14f Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 11 Feb 2015 19:18:51 +0800 Subject: [PATCH 082/411] ASoC: rt5677: Add the chip type to distinguish the setting of the clock source There is only one clock source in the rt5676, so the chip type is added to distinguish the setting of the clock source in the VAD function. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 28 ++++++++++++++++++++++------ sound/soc/codecs/rt5677.h | 6 ++++++ 2 files changed, 28 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb8748dd1df..ab62777dbd331c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -718,11 +718,24 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); + switch (rt5677->type) { + case RT5677: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | + RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | + RT5677_DSP_CLK_SRC_BYPASS); + break; + case RT5676: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_MASK, + RT5677_DSP_CLK_SRC_BYPASS); + break; + default: + break; + } regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); rt5677_set_dsp_mode(codec, true); @@ -4733,7 +4746,8 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", 0 }, + { "rt5677", RT5677 }, + { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -4850,6 +4864,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); + rt5677->type = id->driver_data; + if (pdata) rt5677->pdata = *pdata; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index c0a625f290cc7c..07df96b43f5962 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1665,6 +1665,11 @@ enum { RT5677_IRQ_JD3, }; +enum rt5677_type { + RT5677, + RT5676, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1681,6 +1686,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif From cbca4076d156c93cedadabb0e463ba0db16bb166 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 25 Feb 2015 17:36:14 +0800 Subject: [PATCH 083/411] ASoC: rt5677: Keep the LDO2 powered while used in the suspend mode The patch keeps the ldo2 power while the DSP function of "Voice Wake Up" used in the suspend mode. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ab62777dbd331c..5ff7ffaec5cc90 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4513,10 +4513,10 @@ static int rt5677_suspend(struct snd_soc_codec *codec) if (!rt5677->dsp_vad_en) { regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); - } - if (gpio_is_valid(rt5677->pow_ldo2)) - gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + } return 0; } @@ -4525,12 +4525,12 @@ static int rt5677_resume(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - if (gpio_is_valid(rt5677->pow_ldo2)) { - gpio_set_value_cansleep(rt5677->pow_ldo2, 1); - msleep(10); - } - if (!rt5677->dsp_vad_en) { + if (gpio_is_valid(rt5677->pow_ldo2)) { + gpio_set_value_cansleep(rt5677->pow_ldo2, 1); + msleep(10); + } + regcache_cache_only(rt5677->regmap, false); regcache_sync(rt5677->regmap); } From a0cf43e2f0f391dad7882febbf04423e73e3ff99 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:55 +0100 Subject: [PATCH 084/411] ASoC: tegra: Expose Headphones pin to userspace So userspace can enable or disable it based on the current policy. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb997b75228..8df71a436f1199 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -136,6 +136,7 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), }; From 3a4562f756617b4b210fc487bfe23853a450d3c1 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:56 +0100 Subject: [PATCH 085/411] ASoC: tegra: Add sink for the internal mic to tegra_max98090 Also adds a control for the pin of the internal mic, so userspace can apply policy when the state of the external mic jack changes. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt index c949abc2992f5f..c3495beba35841 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -18,6 +18,7 @@ Required properties: * Headphones * Speakers * Mic Jack + * Int Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 8df71a436f1199..29ea87cd852efc 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -133,11 +133,13 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Int Mic"), }; static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) From dd3001490834e10615d9eb229b3e9bbcc0070541 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:57 +0100 Subject: [PATCH 086/411] ASoC: tegra: Add control for the Mic Jack pin So userspace can enable and disable the external microphone. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 29ea87cd852efc..1f20c2c40a5af0 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -139,6 +139,7 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { static const struct snd_kcontrol_new tegra_max98090_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), SOC_DAPM_PIN_SWITCH("Int Mic"), }; From 1a4ba30cced3002add8459eadcd65b8d3cd1515e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Feb 2015 11:50:11 +0100 Subject: [PATCH 087/411] ALSA: hda - Split snd_hda_build_pcms() snd_hda_build_pcms() does actually three things: let the codec driver build up hda_pcm list, set the PCM default values, and call the attach_pcm bus ops for each hda_pcm instance. The former two are basically independent from the bus implementation, so it'd make the code a bit more readable. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 102 ++++++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 60 insertions(+), 43 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index db86b446743c91..40300fcc530891 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4569,71 +4569,87 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) return -EAGAIN; } -/* - * attach a new PCM stream - */ -static int snd_hda_attach_pcm(struct hda_codec *codec, struct hda_pcm *pcm) +/* call build_pcms ops of the given codec and set up the default parameters */ +int snd_hda_codec_parse_pcms(struct hda_codec *codec) { - struct hda_bus *bus = codec->bus; - struct hda_pcm_stream *info; - int stream, err; + unsigned int pcm; + int err; - if (snd_BUG_ON(!pcm->name)) - return -EINVAL; - for (stream = 0; stream < 2; stream++) { - info = &pcm->stream[stream]; - if (info->substreams) { + if (codec->num_pcms) + return 0; /* already parsed */ + + if (!codec->patch_ops.build_pcms) + return 0; + + err = codec->patch_ops.build_pcms(codec); + if (err < 0) { + codec_err(codec, "cannot build PCMs for #%d (error %d)\n", + codec->addr, err); + return err; + } + + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + int stream; + + for (stream = 0; stream < 2; stream++) { + struct hda_pcm_stream *info = &cpcm->stream[stream]; + + if (!info->substreams) + continue; + if (snd_BUG_ON(!cpcm->name)) + return -EINVAL; err = set_pcm_default_values(codec, info); - if (err < 0) + if (err < 0) { + codec_warn(codec, + "fail to setup default for PCM %s\n", + cpcm->name); return err; + } } } - return bus->ops.attach_pcm(bus, codec, pcm); + + return 0; } /* assign all PCMs of the given codec */ int snd_hda_codec_build_pcms(struct hda_codec *codec) { + struct hda_bus *bus = codec->bus; unsigned int pcm; - int err; + int dev, err; - if (!codec->num_pcms) { - if (!codec->patch_ops.build_pcms) - return 0; - err = codec->patch_ops.build_pcms(codec); - if (err < 0) { - codec_err(codec, - "cannot build PCMs for #%d (error %d)\n", - codec->addr, err); - err = snd_hda_codec_reset(codec); - if (err < 0) { - codec_err(codec, - "cannot revert codec\n"); - return err; - } - } + if (snd_BUG_ON(!bus->ops.attach_pcm)) + return -EINVAL; + + err = snd_hda_codec_parse_pcms(codec); + if (err < 0) { + snd_hda_codec_reset(codec); + return err; } + + /* attach a new PCM streams */ for (pcm = 0; pcm < codec->num_pcms; pcm++) { struct hda_pcm *cpcm = &codec->pcm_info[pcm]; - int dev; + if (cpcm->pcm) + continue; /* already attached */ if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) continue; /* no substreams assigned */ - if (!cpcm->pcm) { - dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); - if (dev < 0) - continue; /* no fatal error */ - cpcm->device = dev; - err = snd_hda_attach_pcm(codec, cpcm); - if (err < 0) { - codec_err(codec, - "cannot attach PCM stream %d for codec #%d\n", - dev, codec->addr); - continue; /* no fatal error */ - } + dev = get_empty_pcm_device(bus, cpcm->pcm_type); + if (dev < 0) + continue; /* no fatal error */ + cpcm->device = dev; + err = bus->ops.attach_pcm(bus, codec, cpcm); + if (err < 0) { + codec_err(codec, + "cannot attach PCM stream %d for codec #%d\n", + dev, codec->addr); + continue; /* no fatal error */ } } + return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 457fc589eb469d..8cf70369cd1016 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -517,6 +517,7 @@ int snd_hda_codec_build_controls(struct hda_codec *codec); * PCM */ int snd_hda_build_pcms(struct hda_bus *bus); +int snd_hda_codec_parse_pcms(struct hda_codec *codec); int snd_hda_codec_build_pcms(struct hda_codec *codec); int snd_hda_codec_prepare(struct hda_codec *codec, From 6efdd8513f182492c21fb7238592d4539d5c751a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 16:09:22 +0100 Subject: [PATCH 088/411] ALSA: hda - Add card field to hda_codec struct Allow the codec object to have an individual card pointer. Not only this simplifies the redirections in many places, also this will allow us to make each codec assigned to a different card object. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 4 ++-- sound/pci/hda/hda_bind.c | 6 +++--- sound/pci/hda/hda_codec.c | 30 +++++++++++++++--------------- sound/pci/hda/hda_codec.h | 5 +++-- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_jack.c | 8 ++++---- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_trace.h | 4 ++-- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_hdmi.c | 6 +++--- sound/pci/hda/patch_via.c | 8 ++++---- 13 files changed, 41 insertions(+), 40 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index e98438e95e79dd..581b7fdef0e379 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -160,7 +160,7 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->name = "HDA Digital PCBeep"; input_dev->phys = beep->phys; input_dev->id.bustype = BUS_PCI; - input_dev->dev.parent = &codec->bus->card->card_dev; + input_dev->dev.parent = &codec->card->card_dev; input_dev->id.vendor = codec->vendor_id >> 16; input_dev->id.product = codec->vendor_id & 0xffff; @@ -224,7 +224,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) if (beep == NULL) return -ENOMEM; snprintf(beep->phys, sizeof(beep->phys), - "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); + "card%d/codec#%d/beep0", codec->card->number, codec->addr); /* enable linear scale */ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0x01); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index ce2dd7b0dc078f..2d00417494e25a 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -287,9 +287,9 @@ int snd_hda_codec_configure(struct hda_codec *codec) } /* audio codec should override the mixer name */ - if (codec->afg || !*codec->bus->card->mixername) - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), + if (codec->afg || !*codec->card->mixername) + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", codec->vendor_name, codec->chip_name); return 0; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 40300fcc530891..0533c862b53946 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1196,9 +1196,8 @@ static void snd_hda_codec_dev_release(struct device *dev) * * Returns 0 if successful, or a negative error code. */ -int snd_hda_codec_new(struct hda_bus *bus, - unsigned int codec_addr, - struct hda_codec **codecp) +int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp) { struct hda_codec *codec; struct device *dev; @@ -1217,7 +1216,7 @@ int snd_hda_codec_new(struct hda_bus *bus, return -EINVAL; if (bus->caddr_tbl[codec_addr]) { - dev_err(bus->card->dev, + dev_err(card->dev, "address 0x%x is already occupied\n", codec_addr); return -EBUSY; @@ -1225,21 +1224,22 @@ int snd_hda_codec_new(struct hda_bus *bus, codec = kzalloc(sizeof(*codec), GFP_KERNEL); if (codec == NULL) { - dev_err(bus->card->dev, "can't allocate struct hda_codec\n"); + dev_err(card->dev, "can't allocate struct hda_codec\n"); return -ENOMEM; } dev = hda_codec_dev(codec); device_initialize(dev); - dev->parent = bus->card->dev; + dev->parent = card->dev; dev->bus = &snd_hda_bus_type; dev->release = snd_hda_codec_dev_release; dev->groups = snd_hda_dev_attr_groups; - dev_set_name(dev, "hdaudioC%dD%d", bus->card->number, codec_addr); + dev_set_name(dev, "hdaudioC%dD%d", card->number, codec_addr); dev_set_drvdata(dev, codec); /* for sysfs */ device_enable_async_suspend(dev); codec->bus = bus; + codec->card = card; codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); @@ -1300,7 +1300,7 @@ int snd_hda_codec_new(struct hda_bus *bus, setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { - dev_err(bus->card->dev, "no AFG or MFG node found\n"); + dev_err(card->dev, "no AFG or MFG node found\n"); err = -ENODEV; goto error; } @@ -1308,7 +1308,7 @@ int snd_hda_codec_new(struct hda_bus *bus, fg = codec->afg ? codec->afg : codec->mfg; err = read_widget_caps(codec, fg); if (err < 0) { - dev_err(bus->card->dev, "cannot malloc\n"); + dev_err(card->dev, "cannot malloc\n"); goto error; } err = read_pin_defaults(codec); @@ -1337,9 +1337,9 @@ int snd_hda_codec_new(struct hda_bus *bus, sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, codec->subsystem_id, codec->revision_id); - snd_component_add(codec->bus->card, component); + snd_component_add(card, component); - err = snd_device_new(bus->card, SNDRV_DEV_CODEC, codec, &dev_ops); + err = snd_device_new(card, SNDRV_DEV_CODEC, codec, &dev_ops); if (err < 0) goto error; @@ -2237,7 +2237,7 @@ find_mixer_ctl(struct hda_codec *codec, const char *name, int dev, int idx) if (snd_BUG_ON(strlen(name) >= sizeof(id.name))) return NULL; strcpy(id.name, name); - return snd_ctl_find_id(codec->bus->card, &id); + return snd_ctl_find_id(codec->card, &id); } /** @@ -2301,7 +2301,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, nid = kctl->id.subdevice & 0xffff; if (kctl->id.subdevice & (HDA_SUBDEV_NID_FLAG|HDA_SUBDEV_AMP_FLAG)) kctl->id.subdevice = 0; - err = snd_ctl_add(codec->bus->card, kctl); + err = snd_ctl_add(codec->card, kctl); if (err < 0) return err; item = snd_array_new(&codec->mixers); @@ -2354,7 +2354,7 @@ void snd_hda_ctls_clear(struct hda_codec *codec) int i; struct hda_nid_item *items = codec->mixers.list; for (i = 0; i < codec->mixers.used; i++) - snd_ctl_remove(codec->bus->card, items[i].kctl); + snd_ctl_remove(codec->card, items[i].kctl); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); } @@ -2427,7 +2427,7 @@ EXPORT_SYMBOL_GPL(snd_hda_unlock_devices); int snd_hda_codec_reset(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; - struct snd_card *card = bus->card; + struct snd_card *card = codec->card; int i; if (snd_hda_lock_devices(bus) < 0) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8cf70369cd1016..8908a07687361a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -274,6 +274,7 @@ struct hda_pcm { struct hda_codec { struct device dev; struct hda_bus *bus; + struct snd_card *card; unsigned int addr; /* codec addr*/ struct list_head list; /* list point */ @@ -420,8 +421,8 @@ enum { * constructors */ int snd_hda_bus_new(struct snd_card *card, struct hda_bus **busp); -int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); +int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 1695f0e2bd9d47..f50863a5159df5 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1898,7 +1898,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots) for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(bus, c, &codec); + err = snd_hda_codec_new(bus, bus->card, c, &codec); if (err < 0) continue; codec->jackpoll_interval = get_jackpoll_interval(chip); diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 125f3420fa6ae1..57df06e76968ac 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -101,7 +101,7 @@ int snd_hda_create_hwdep(struct hda_codec *codec) int err; sprintf(hwname, "HDA Codec %d", codec->addr); - err = snd_hwdep_new(codec->bus->card, hwname, codec->addr, &hwdep); + err = snd_hwdep_new(codec->card, hwname, codec->addr, &hwdep); if (err < 0) return err; codec->hwdep = hwdep; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index e664307617bd5f..d7cfe7b8c32b10 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -135,7 +135,7 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_INPUT_JACK /* free jack instances manually when clearing/reconfiguring */ if (!codec->bus->shutdown && jack->jack) - snd_device_free(codec->bus->card, jack->jack); + snd_device_free(codec->card, jack->jack); #endif for (cb = jack->callback; cb; cb = next) { next = cb->next; @@ -340,7 +340,7 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) if (!jack->kctl || jack->block_report) continue; state = get_jack_plug_state(jack->pin_sense); - snd_kctl_jack_report(codec->bus->card, jack->kctl, state); + snd_kctl_jack_report(codec->card, jack->kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK if (jack->jack) snd_jack_report(jack->jack, @@ -412,11 +412,11 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->phantom_jack = !!phantom_jack; state = snd_hda_jack_detect(codec, nid); - snd_kctl_jack_report(codec->bus->card, kctl, state); + snd_kctl_jack_report(codec->card, kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK if (!phantom_jack) { jack->type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->bus->card, name, jack->type, + err = snd_jack_new(codec->card, name, jack->type, &jack->jack); if (err < 0) return err; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index ce5a6da834199b..cc32b878ae2e8e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -839,7 +839,7 @@ int snd_hda_codec_proc_new(struct hda_codec *codec) int err; snprintf(name, sizeof(name), "codec#%d", codec->addr); - err = snd_card_proc_new(codec->bus->card, name, &entry); + err = snd_card_proc_new(codec->card, name, &entry); if (err < 0) return err; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index ccc962a1699f1b..e13c75d6784794 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -149,7 +149,7 @@ static int reconfig_codec(struct hda_codec *codec) err = snd_hda_codec_build_controls(codec); if (err < 0) goto error; - err = snd_card_register(codec->bus->card); + err = snd_card_register(codec->card); error: snd_hda_power_down(codec); return err; diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h index c0e1c7d24dbebe..7fedfa86241970 100644 --- a/sound/pci/hda/hda_trace.h +++ b/sound/pci/hda/hda_trace.h @@ -23,7 +23,7 @@ DECLARE_EVENT_CLASS(hda_cmd, ), TP_fast_assign( - __entry->card = (codec)->bus->card->number; + __entry->card = (codec)->card->number; __entry->addr = (codec)->addr; __entry->val = (val); ), @@ -71,7 +71,7 @@ DECLARE_EVENT_CLASS(hda_power, ), TP_fast_assign( - __entry->card = (codec)->bus->card->number; + __entry->card = (codec)->card->number; __entry->addr = (codec)->addr; ), diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 81991b4134cd10..ced3e82d9e23ef 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4352,7 +4352,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - if (request_firmware(&fw_entry, EFX_FILE, codec->bus->card->dev) != 0) + if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) return false; dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f1812aabd63e25..0f8354cbc7a7e1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -579,7 +579,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) int err; snprintf(name, sizeof(name), "eld#%d.%d", codec->addr, index); - err = snd_card_proc_new(codec->bus->card, name, &entry); + err = snd_card_proc_new(codec->card, name, &entry); if (err < 0) return err; @@ -594,7 +594,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) static void eld_proc_free(struct hdmi_spec_per_pin *per_pin) { if (!per_pin->codec->bus->shutdown && per_pin->proc_entry) { - snd_device_free(per_pin->codec->bus->card, per_pin->proc_entry); + snd_device_free(per_pin->codec->card, per_pin->proc_entry); per_pin->proc_entry = NULL; } } @@ -1624,7 +1624,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) } if (eld_changed) - snd_ctl_notify(codec->bus->card, + snd_ctl_notify(codec->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &per_pin->eld_ctl->id); unlock: diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 2045f33b1ace3c..57ad503ff9407b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -907,16 +907,16 @@ static int patch_vt1708S(struct hda_codec *codec) if (get_codec_type(codec) == VT1708BCE) { kfree(codec->chip_name); codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", codec->vendor_name, codec->chip_name); } /* correct names for VT1705 */ if (codec->vendor_id == 0x11064397) { kfree(codec->chip_name); codec->chip_name = kstrdup("VT1705", GFP_KERNEL); - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", codec->vendor_name, codec->chip_name); } From f4de8fe6cffb449a779dff61f071bd1af9e18e0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 16:17:18 +0100 Subject: [PATCH 089/411] ALSA: hda - Remove superfluous memory allocation error messages The memory allocators should have already given the kernel warning messages, thus we don't have to annoy again. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 16 ++++------------ sound/pci/hda/hda_controller.c | 18 ++++-------------- sound/pci/hda/hda_intel.c | 5 +---- 3 files changed, 9 insertions(+), 30 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0533c862b53946..262c41a8f96e99 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -776,10 +776,8 @@ int snd_hda_bus_new(struct snd_card *card, *busp = NULL; bus = kzalloc(sizeof(*bus), GFP_KERNEL); - if (bus == NULL) { - dev_err(card->dev, "can't allocate struct hda_bus\n"); + if (!bus) return -ENOMEM; - } bus->card = card; mutex_init(&bus->cmd_mutex); @@ -1223,10 +1221,8 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, } codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (codec == NULL) { - dev_err(card->dev, "can't allocate struct hda_codec\n"); + if (!codec) return -ENOMEM; - } dev = hda_codec_dev(codec); device_initialize(dev); @@ -1307,10 +1303,8 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, fg = codec->afg ? codec->afg : codec->mfg; err = read_widget_caps(codec, fg); - if (err < 0) { - dev_err(card->dev, "cannot malloc\n"); + if (err < 0) goto error; - } err = read_pin_defaults(codec); if (err < 0) goto error; @@ -1371,10 +1365,8 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) kfree(codec->wcaps); fg = codec->afg ? codec->afg : codec->mfg; err = read_widget_caps(codec, fg); - if (err < 0) { - codec_err(codec, "cannot malloc\n"); + if (err < 0) return err; - } snd_array_free(&codec->init_pins); err = read_pin_defaults(codec); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index f50863a5159df5..be02bca6f7e6bc 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -974,14 +974,9 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, */ static int azx_alloc_cmd_io(struct azx *chip) { - int err; - /* single page (at least 4096 bytes) must suffice for both ringbuffes */ - err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, - PAGE_SIZE, &chip->rb); - if (err < 0) - dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n"); - return err; + return chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, + PAGE_SIZE, &chip->rb); } EXPORT_SYMBOL_GPL(azx_alloc_cmd_io); @@ -1472,7 +1467,6 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, int azx_alloc_stream_pages(struct azx *chip) { int i, err; - struct snd_card *card = chip->card; for (i = 0; i < chip->num_streams; i++) { dsp_lock_init(&chip->azx_dev[i]); @@ -1480,18 +1474,14 @@ int azx_alloc_stream_pages(struct azx *chip) err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, BDL_SIZE, &chip->azx_dev[i].bdl); - if (err < 0) { - dev_err(card->dev, "cannot allocate BDL\n"); + if (err < 0) return -ENOMEM; - } } /* allocate memory for the position buffer */ err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, chip->num_streams * 8, &chip->posbuf); - if (err < 0) { - dev_err(card->dev, "cannot allocate posbuf\n"); + if (err < 0) return -ENOMEM; - } /* allocate CORB/RIRB */ err = azx_alloc_cmd_io(chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e75e8137e296cd..f7fb1b51d4464c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1383,7 +1383,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, hda = kzalloc(sizeof(*hda), GFP_KERNEL); if (!hda) { - dev_err(card->dev, "Cannot allocate hda\n"); pci_disable_device(pci); return -ENOMEM; } @@ -1564,10 +1563,8 @@ static int azx_first_init(struct azx *chip) chip->num_streams = chip->playback_streams + chip->capture_streams; chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); - if (!chip->azx_dev) { - dev_err(card->dev, "cannot malloc azx_dev\n"); + if (!chip->azx_dev) return -ENOMEM; - } err = azx_alloc_stream_pages(chip); if (err < 0) From bbbc7e8502c95237dbd86cc4d0a12ca9a6b18c8f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 17:43:19 +0100 Subject: [PATCH 090/411] ALSA: hda - Allocate hda_pcm objects dynamically So far, the hda_codec object kept the hda_pcm list in an array, and the codec driver was expected to assign the array. However, this makes the object life cycle management harder, because the assigned array is freed at the codec driver detach while it might be still accessed by the opened streams. In this patch, we allocate each hda_pcm object dynamically and manage it as a linked list. Each object has a kref refcount, and both the codec driver binder and the PCM open/close touches it, so that the object won't be freed while in use. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 116 ++++++++++++++++++++++------------ sound/pci/hda/hda_codec.h | 18 +++++- sound/pci/hda/hda_generic.c | 28 ++++---- sound/pci/hda/hda_generic.h | 2 +- sound/pci/hda/hda_proc.c | 6 +- sound/pci/hda/patch_ca0132.c | 29 ++++----- sound/pci/hda/patch_hdmi.c | 30 +++------ sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 11 ++-- sound/pci/hda/patch_via.c | 6 +- 10 files changed, 145 insertions(+), 103 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 262c41a8f96e99..20283bead10af2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1116,6 +1116,60 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) return p; } +/* + * PCM device + */ +static void release_pcm(struct kref *kref) +{ + struct hda_pcm *pcm = container_of(kref, struct hda_pcm, kref); + + if (pcm->pcm) + snd_device_free(pcm->codec->card, pcm->pcm); + clear_bit(pcm->device, pcm->codec->bus->pcm_dev_bits); + kfree(pcm->name); + kfree(pcm); +} + +void snd_hda_codec_pcm_put(struct hda_pcm *pcm) +{ + kref_put(&pcm->kref, release_pcm); +} +EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_put); + +struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, + const char *fmt, ...) +{ + struct hda_pcm *pcm; + va_list args; + + va_start(args, fmt); + pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return NULL; + + pcm->codec = codec; + kref_init(&pcm->kref); + pcm->name = kvasprintf(GFP_KERNEL, fmt, args); + if (!pcm->name) { + kfree(pcm); + return NULL; + } + + list_add_tail(&pcm->list, &codec->pcm_list_head); + return pcm; +} +EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); + +static void codec_release_pcms(struct hda_codec *codec) +{ + struct hda_pcm *pcm, *n; + + list_for_each_entry_safe(pcm, n, &codec->pcm_list_head, list) { + list_del_init(&pcm->list); + snd_hda_codec_pcm_put(pcm); + } +} + /* * codec destructor */ @@ -1124,6 +1178,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) if (!codec) return; cancel_delayed_work_sync(&codec->jackpoll_work); + codec_release_pcms(codec); if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); snd_hda_jack_tbl_clear(codec); @@ -1251,6 +1306,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; @@ -2370,9 +2426,8 @@ int snd_hda_lock_devices(struct hda_bus *bus) goto err_clear; list_for_each_entry(codec, &bus->codec_list, list) { - int pcm; - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + struct hda_pcm *cpcm; + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { if (!cpcm->pcm) continue; if (cpcm->pcm->streams[0].substream_opened || @@ -2419,8 +2474,6 @@ EXPORT_SYMBOL_GPL(snd_hda_unlock_devices); int snd_hda_codec_reset(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; - struct snd_card *card = codec->card; - int i; if (snd_hda_lock_devices(bus) < 0) return -EBUSY; @@ -2429,14 +2482,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) cancel_delayed_work_sync(&codec->jackpoll_work); flush_workqueue(bus->workq); snd_hda_ctls_clear(codec); - /* release PCMs */ - for (i = 0; i < codec->num_pcms; i++) { - if (codec->pcm_info[i].pcm) { - snd_device_free(card, codec->pcm_info[i].pcm); - clear_bit(codec->pcm_info[i].device, - bus->pcm_dev_bits); - } - } + codec_release_pcms(codec); snd_hda_detach_beep_device(codec); if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); @@ -2454,8 +2500,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) snd_array_free(&codec->cvt_setups); snd_array_free(&codec->spdif_out); snd_array_free(&codec->verbs); - codec->num_pcms = 0; - codec->pcm_info = NULL; codec->preset = NULL; codec->slave_dig_outs = NULL; codec->spdif_status_reset = 0; @@ -3952,12 +3996,12 @@ static void hda_call_codec_resume(struct hda_codec *codec) static int hda_codec_runtime_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + struct hda_pcm *pcm; unsigned int state; - int i; cancel_delayed_work_sync(&codec->jackpoll_work); - for (i = 0; i < codec->num_pcms; i++) - snd_pcm_suspend_all(codec->pcm_info[i].pcm); + list_for_each_entry(pcm, &codec->pcm_list_head, list) + snd_pcm_suspend_all(pcm->pcm); state = hda_call_codec_suspend(codec); if (codec->d3_stop_clk && codec->epss && (state & AC_PWRST_CLK_STOP_OK)) clear_bit(codec->addr, &codec->bus->codec_powered); @@ -4018,22 +4062,21 @@ EXPORT_SYMBOL_GPL(snd_hda_build_controls); */ static int add_std_chmaps(struct hda_codec *codec) { - int i, str, err; + struct hda_pcm *pcm; + int str, err; - for (i = 0; i < codec->num_pcms; i++) { + list_for_each_entry(pcm, &codec->pcm_list_head, list) { for (str = 0; str < 2; str++) { - struct snd_pcm *pcm = codec->pcm_info[i].pcm; - struct hda_pcm_stream *hinfo = - &codec->pcm_info[i].stream[str]; + struct hda_pcm_stream *hinfo = &pcm->stream[str]; struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (codec->pcm_info[i].own_chmap) + if (pcm->own_chmap) continue; if (!pcm || !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; - err = snd_pcm_add_chmap_ctls(pcm, str, elem, + err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, hinfo->channels_max, 0, &chmap); if (err < 0) @@ -4564,10 +4607,10 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) /* call build_pcms ops of the given codec and set up the default parameters */ int snd_hda_codec_parse_pcms(struct hda_codec *codec) { - unsigned int pcm; + struct hda_pcm *cpcm; int err; - if (codec->num_pcms) + if (!list_empty(&codec->pcm_list_head)) return 0; /* already parsed */ if (!codec->patch_ops.build_pcms) @@ -4580,8 +4623,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) return err; } - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { int stream; for (stream = 0; stream < 2; stream++) { @@ -4589,8 +4631,6 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) if (!info->substreams) continue; - if (snd_BUG_ON(!cpcm->name)) - return -EINVAL; err = set_pcm_default_values(codec, info); if (err < 0) { codec_warn(codec, @@ -4608,7 +4648,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) int snd_hda_codec_build_pcms(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; - unsigned int pcm; + struct hda_pcm *cpcm; int dev, err; if (snd_BUG_ON(!bus->ops.attach_pcm)) @@ -4621,9 +4661,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) } /* attach a new PCM streams */ - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; - + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { if (cpcm->pcm) continue; /* already attached */ if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) @@ -4651,11 +4689,9 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) * * Create PCM information for each codec included in the bus. * - * The build_pcms codec patch is requested to set up codec->num_pcms and - * codec->pcm_info properly. The array is referred by the top-level driver - * to create its PCM instances. - * The allocated codec->pcm_info should be released in codec->patch_ops.free - * callback. + * The build_pcms codec patch is requested to create and assign new + * hda_pcm objects. The codec is responsible to call snd_hda_codec_pcm_new() + * and fills the fields. Later they are instantiated by this function. * * At least, substreams, channels_min and channels_max must be filled for * each stream. substreams = 0 indicates that the stream doesn't exist. diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8908a07687361a..2ccd6f9a91fec5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -21,6 +21,7 @@ #ifndef __SOUND_HDA_CODEC_H #define __SOUND_HDA_CODEC_H +#include #include #include #include @@ -268,6 +269,10 @@ struct hda_pcm { int device; /* device number to assign */ struct snd_pcm *pcm; /* assigned PCM instance */ bool own_chmap; /* codec driver provides own channel maps */ + /* private: */ + struct hda_codec *codec; + struct kref kref; + struct list_head list; }; /* codec information */ @@ -301,8 +306,7 @@ struct hda_codec { struct hda_codec_ops patch_ops; /* PCM to create, set by patch_ops.build_pcms callback */ - unsigned int num_pcms; - struct hda_pcm *pcm_info; + struct list_head pcm_list_head; /* codec specific info */ void *spec; @@ -521,6 +525,16 @@ int snd_hda_build_pcms(struct hda_bus *bus); int snd_hda_codec_parse_pcms(struct hda_codec *codec); int snd_hda_codec_build_pcms(struct hda_codec *codec); +__printf(2, 3) +struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, + const char *fmt, ...); + +static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm) +{ + kref_get(&pcm->kref); +} +void snd_hda_codec_pcm_put(struct hda_pcm *pcm); + int snd_hda_codec_prepare(struct hda_codec *codec, struct hda_pcm_stream *hinfo, unsigned int stream, diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 947d1a50f3840b..092f06fd21bd0c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4644,7 +4644,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) err = snd_hda_create_dig_out_ctls(codec, spec->multiout.dig_out_nid, spec->multiout.dig_out_nid, - spec->pcm_rec[1].pcm_type); + spec->pcm_rec[1]->pcm_type); if (err < 0) return err; if (!spec->no_analog) { @@ -5115,20 +5115,20 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx, int snd_hda_gen_build_pcms(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; + struct hda_pcm *info; const struct hda_pcm_stream *p; bool have_multi_adcs; - codec->num_pcms = 1; - codec->pcm_info = info; - if (spec->no_analog) goto skip_analog; fill_pcm_stream_name(spec->stream_name_analog, sizeof(spec->stream_name_analog), " Analog", codec->chip_name); - info->name = spec->stream_name_analog; + info = snd_hda_codec_pcm_new(codec, "%s", spec->stream_name_analog); + if (!info) + return -ENOMEM; + spec->pcm_rec[0] = info; if (spec->multiout.num_dacs > 0) { p = spec->stream_analog_playback; @@ -5161,10 +5161,12 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) fill_pcm_stream_name(spec->stream_name_digital, sizeof(spec->stream_name_digital), " Digital", codec->chip_name); - codec->num_pcms = 2; + info = snd_hda_codec_pcm_new(codec, "%s", + spec->stream_name_digital); + if (!info) + return -ENOMEM; codec->slave_dig_outs = spec->multiout.slave_dig_outs; - info = spec->pcm_rec + 1; - info->name = spec->stream_name_digital; + spec->pcm_rec[1] = info; if (spec->dig_out_type) info->pcm_type = spec->dig_out_type; else @@ -5198,9 +5200,11 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) fill_pcm_stream_name(spec->stream_name_alt_analog, sizeof(spec->stream_name_alt_analog), " Alt Analog", codec->chip_name); - codec->num_pcms = 3; - info = spec->pcm_rec + 2; - info->name = spec->stream_name_alt_analog; + info = snd_hda_codec_pcm_new(codec, "%s", + spec->stream_name_alt_analog); + if (!info) + return -ENOMEM; + spec->pcm_rec[2] = info; if (spec->alt_dac_nid) { p = spec->stream_analog_alt_playback; if (!p) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 3d852660443aaf..b211f889b335dd 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -144,7 +144,7 @@ struct hda_gen_spec { int const_channel_count; /* channel count for all */ /* PCM information */ - struct hda_pcm pcm_rec[3]; /* used in build_pcms() */ + struct hda_pcm *pcm_rec[3]; /* used in build_pcms() */ /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index cc32b878ae2e8e..aeb983e7506d1c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -99,10 +99,10 @@ static void print_nid_array(struct snd_info_buffer *buffer, static void print_nid_pcms(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { - int pcm, type; + int type; struct hda_pcm *cpcm; - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - cpcm = &codec->pcm_info[pcm]; + + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { for (type = 0; type < 2; type++) { if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL) continue; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index ced3e82d9e23ef..555781fad26f00 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -719,7 +719,6 @@ struct ca0132_spec { unsigned int num_inputs; hda_nid_t shared_mic_nid; hda_nid_t shared_out_nid; - struct hda_pcm pcm_rec[5]; /* PCM information */ /* chip access */ struct mutex chipio_mutex; /* chip access mutex */ @@ -4036,12 +4035,11 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { static int ca0132_build_pcms(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; + struct hda_pcm *info; - codec->pcm_info = info; - codec->num_pcms = 0; - - info->name = "CA0132 Analog"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = @@ -4049,27 +4047,27 @@ static int ca0132_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; - codec->num_pcms++; - info++; - info->name = "CA0132 Analog Mic-In2"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; - codec->num_pcms++; - info++; - info->name = "CA0132 What U Hear"; + info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[2]; - codec->num_pcms++; if (!spec->dig_out && !spec->dig_in) return 0; - info++; - info->name = "CA0132 Digital"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Digital"); + if (!info) + return -ENOMEM; info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->dig_out) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = @@ -4081,7 +4079,6 @@ static int ca0132_build_pcms(struct hda_codec *codec) ca0132_pcm_digital_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; } - codec->num_pcms++; return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0f8354cbc7a7e1..708bbed15ea399 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -86,7 +86,6 @@ struct hdmi_spec_per_pin { bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ - char pcm_name[8]; /* filled in build_pcm callbacks */ #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif @@ -132,7 +131,7 @@ struct hdmi_spec { int num_pins; struct snd_array pins; /* struct hdmi_spec_per_pin */ - struct snd_array pcm_rec; /* struct hda_pcm */ + struct hda_pcm *pcm_rec[16]; unsigned int channels_max; /* max over all cvts */ struct hdmi_eld temp_eld; @@ -355,8 +354,7 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { ((struct hdmi_spec_per_pin *)snd_array_elem(&spec->pins, idx)) #define get_cvt(spec, idx) \ ((struct hdmi_spec_per_cvt *)snd_array_elem(&spec->cvts, idx)) -#define get_pcm_rec(spec, idx) \ - ((struct hda_pcm *)snd_array_elem(&spec->pcm_rec, idx)) +#define get_pcm_rec(spec, idx) ((spec)->pcm_rec[idx]) static int pin_nid_to_pin_index(struct hda_codec *codec, hda_nid_t pin_nid) { @@ -2056,11 +2054,10 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) struct hdmi_spec_per_pin *per_pin; per_pin = get_pin(spec, pin_idx); - sprintf(per_pin->pcm_name, "HDMI %d", pin_idx); - info = snd_array_new(&spec->pcm_rec); + info = snd_hda_codec_pcm_new(codec, "HDMI %d", pin_idx); if (!info) return -ENOMEM; - info->name = per_pin->pcm_name; + spec->pcm_rec[pin_idx] = info; info->pcm_type = HDA_PCM_TYPE_HDMI; info->own_chmap = true; @@ -2070,9 +2067,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) /* other pstr fields are set in open */ } - codec->num_pcms = spec->num_pins; - codec->pcm_info = spec->pcm_rec.list; - return 0; } @@ -2125,13 +2119,15 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) /* add channel maps */ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hda_pcm *pcm; struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; - if (!codec->pcm_info[pin_idx].pcm) + pcm = spec->pcm_rec[pin_idx]; + if (!pcm || !pcm->pcm) break; - err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, + err = snd_pcm_add_chmap_ctls(pcm->pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); if (err < 0) @@ -2186,14 +2182,12 @@ static void hdmi_array_init(struct hdmi_spec *spec, int nums) { snd_array_init(&spec->pins, sizeof(struct hdmi_spec_per_pin), nums); snd_array_init(&spec->cvts, sizeof(struct hdmi_spec_per_cvt), nums); - snd_array_init(&spec->pcm_rec, sizeof(struct hda_pcm), nums); } static void hdmi_array_free(struct hdmi_spec *spec) { snd_array_free(&spec->pins); snd_array_free(&spec->cvts); - snd_array_free(&spec->pcm_rec); } static void generic_hdmi_free(struct hda_codec *codec) @@ -2381,11 +2375,10 @@ static int simple_playback_build_pcms(struct hda_codec *codec) chans = get_wcaps(codec, per_cvt->cvt_nid); chans = get_wcaps_channels(chans); - info = snd_array_new(&spec->pcm_rec); + info = snd_hda_codec_pcm_new(codec, "HDMI 0"); if (!info) return -ENOMEM; - info->name = get_pin(spec, 0)->pcm_name; - sprintf(info->name, "HDMI 0"); + spec->pcm_rec[0] = info; info->pcm_type = HDA_PCM_TYPE_HDMI; pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; *pstr = spec->pcm_playback; @@ -2393,9 +2386,6 @@ static int simple_playback_build_pcms(struct hda_codec *codec) if (pstr->channels_max <= 2 && chans && chans <= 16) pstr->channels_max = chans; - codec->num_pcms = 1; - codec->pcm_info = info; - return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70808f92276ada..0a5a2246e57aff 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5843,7 +5843,7 @@ static void alc_fixup_bass_chmap(struct hda_codec *codec, { if (action == HDA_FIXUP_ACT_BUILD) { struct alc_spec *spec = codec->spec; - spec->gen.pcm_rec[0].stream[0].chmap = asus_pcm_2_1_chmaps; + spec->gen.pcm_rec[0]->stream[0].chmap = asus_pcm_2_1_chmaps; } } diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 38a4773333218d..df243134baa883 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -83,7 +83,6 @@ struct si3054_spec { unsigned international; - struct hda_pcm pcm; }; @@ -199,11 +198,11 @@ static const struct hda_pcm_stream si3054_pcm = { static int si3054_build_pcms(struct hda_codec *codec) { - struct si3054_spec *spec = codec->spec; - struct hda_pcm *info = &spec->pcm; - codec->num_pcms = 1; - codec->pcm_info = info; - info->name = "Si3054 Modem"; + struct hda_pcm *info; + + info = snd_hda_codec_pcm_new(codec, "Si3054 Modem"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = codec->mfg; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 57ad503ff9407b..11a05638e03b9d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -683,8 +683,10 @@ static int vt1708_build_pcms(struct hda_codec *codec) * 24bit samples are used. Until any workaround is found, * disable the 24bit format, so far. */ - for (i = 0; i < codec->num_pcms; i++) { - struct hda_pcm *info = &spec->gen.pcm_rec[i]; + for (i = 0; i < ARRAY_SIZE(spec->gen.pcm_rec); i++) { + struct hda_pcm *info = spec->gen.pcm_rec[i]; + if (!info) + continue; if (!info->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams || info->pcm_type != HDA_PCM_TYPE_AUDIO) continue; From 61ca4107a16828559e2463e49b87002990da0b98 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 17:57:55 +0100 Subject: [PATCH 091/411] ALSA: hda - Don't assume non-NULL PCM ops The PCM ops might be set NULL, or cleared to NULL when the driver is unbound. Give a proper NULL check at each place to be more robust. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 9 +++++++-- sound/pci/hda/hda_controller.c | 30 +++++++++++++++++++----------- 2 files changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 20283bead10af2..3bd9158addc2b8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4525,7 +4525,11 @@ int snd_hda_codec_prepare(struct hda_codec *codec, { int ret; mutex_lock(&codec->bus->prepare_mutex); - ret = hinfo->ops.prepare(hinfo, codec, stream, format, substream); + if (hinfo->ops.prepare) + ret = hinfo->ops.prepare(hinfo, codec, stream, format, + substream); + else + ret = -ENODEV; if (ret >= 0) purify_inactive_streams(codec); mutex_unlock(&codec->bus->prepare_mutex); @@ -4546,7 +4550,8 @@ void snd_hda_codec_cleanup(struct hda_codec *codec, struct snd_pcm_substream *substream) { mutex_lock(&codec->bus->prepare_mutex); - hinfo->ops.cleanup(hinfo, codec, substream); + if (hinfo->ops.cleanup) + hinfo->ops.cleanup(hinfo, codec, substream); mutex_unlock(&codec->bus->prepare_mutex); } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index be02bca6f7e6bc..ad85f9bfaf57bb 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -416,7 +416,8 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) azx_dev->running = 0; spin_unlock_irqrestore(&chip->reg_lock, flags); azx_release_device(azx_dev); - hinfo->ops.close(hinfo, apcm->codec, substream); + if (hinfo->ops.close) + hinfo->ops.close(hinfo, apcm->codec, substream); snd_hda_power_down(apcm->codec); mutex_unlock(&chip->open_mutex); return 0; @@ -808,8 +809,8 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { - mutex_unlock(&chip->open_mutex); - return -EBUSY; + err = -EBUSY; + goto unlock; } runtime->hw = azx_pcm_hw; runtime->hw.channels_min = hinfo->channels_min; @@ -844,12 +845,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); snd_hda_power_up(apcm->codec); - err = hinfo->ops.open(hinfo, apcm->codec, substream); + if (hinfo->ops.open) + err = hinfo->ops.open(hinfo, apcm->codec, substream); + else + err = -ENODEV; if (err < 0) { azx_release_device(azx_dev); - snd_hda_power_down(apcm->codec); - mutex_unlock(&chip->open_mutex); - return err; + goto powerdown; } snd_pcm_limit_hw_rates(runtime); /* sanity check */ @@ -858,10 +860,10 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_BUG_ON(!runtime->hw.formats) || snd_BUG_ON(!runtime->hw.rates)) { azx_release_device(azx_dev); - hinfo->ops.close(hinfo, apcm->codec, substream); - snd_hda_power_down(apcm->codec); - mutex_unlock(&chip->open_mutex); - return -EINVAL; + if (hinfo->ops.close) + hinfo->ops.close(hinfo, apcm->codec, substream); + err = -EINVAL; + goto powerdown; } /* disable LINK_ATIME timestamps for capture streams @@ -880,6 +882,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); return 0; + + powerdown: + snd_hda_power_down(apcm->codec); + unlock: + mutex_unlock(&chip->open_mutex); + return err; } static int azx_pcm_mmap(struct snd_pcm_substream *substream, From e086e3035e0691b362755d1b5e24df631eee335a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 18:01:22 +0100 Subject: [PATCH 092/411] ALSA: core: Re-add snd_device_disconnect() Revive snd_device_disconnect() again so that it can be called from the individual driver. This time, HD-audio will need it. Signed-off-by: Takashi Iwai --- include/sound/core.h | 3 ++- sound/core/device.c | 43 +++++++++++++++++++++++++++++++++---------- sound/core/init.c | 5 +---- 3 files changed, 36 insertions(+), 15 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index da574828996807..b12931f513f4a8 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -278,7 +278,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, void *device_data, struct snd_device_ops *ops); int snd_device_register(struct snd_card *card, void *device_data); int snd_device_register_all(struct snd_card *card); -int snd_device_disconnect_all(struct snd_card *card); +void snd_device_disconnect(struct snd_card *card, void *device_data); +void snd_device_disconnect_all(struct snd_card *card); void snd_device_free(struct snd_card *card, void *device_data); void snd_device_free_all(struct snd_card *card); diff --git a/sound/core/device.c b/sound/core/device.c index 41bec3075ae5b8..446dc452654e51 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -73,7 +73,7 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, } EXPORT_SYMBOL(snd_device_new); -static int __snd_device_disconnect(struct snd_device *dev) +static void __snd_device_disconnect(struct snd_device *dev) { if (dev->state == SNDRV_DEV_REGISTERED) { if (dev->ops->dev_disconnect && @@ -81,7 +81,6 @@ static int __snd_device_disconnect(struct snd_device *dev) dev_err(dev->card->dev, "device disconnect failure\n"); dev->state = SNDRV_DEV_DISCONNECTED; } - return 0; } static void __snd_device_free(struct snd_device *dev) @@ -108,6 +107,34 @@ static struct snd_device *look_for_dev(struct snd_card *card, void *device_data) return NULL; } +/** + * snd_device_disconnect - disconnect the device + * @card: the card instance + * @device_data: the data pointer to disconnect + * + * Turns the device into the disconnection state, invoking + * dev_disconnect callback, if the device was already registered. + * + * Usually called from snd_card_disconnect(). + * + * Return: Zero if successful, or a negative error code on failure or if the + * device not found. + */ +void snd_device_disconnect(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + if (snd_BUG_ON(!card || !device_data)) + return; + dev = look_for_dev(card, device_data); + if (dev) + __snd_device_disconnect(dev); + else + dev_dbg(card->dev, "device disconnect %p (from %pF), not found\n", + device_data, __builtin_return_address(0)); +} +EXPORT_SYMBOL_GPL(snd_device_disconnect); + /** * snd_device_free - release the device from the card * @card: the card instance @@ -195,18 +222,14 @@ int snd_device_register_all(struct snd_card *card) * disconnect all the devices on the card. * called from init.c */ -int snd_device_disconnect_all(struct snd_card *card) +void snd_device_disconnect_all(struct snd_card *card) { struct snd_device *dev; - int err = 0; if (snd_BUG_ON(!card)) - return -ENXIO; - list_for_each_entry_reverse(dev, &card->devices, list) { - if (__snd_device_disconnect(dev) < 0) - err = -ENXIO; - } - return err; + return; + list_for_each_entry_reverse(dev, &card->devices, list) + __snd_device_disconnect(dev); } /* diff --git a/sound/core/init.c b/sound/core/init.c index 35419054821c3d..04734e047bfe47 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -400,7 +400,6 @@ static const struct file_operations snd_shutdown_f_ops = int snd_card_disconnect(struct snd_card *card) { struct snd_monitor_file *mfile; - int err; if (!card) return -EINVAL; @@ -445,9 +444,7 @@ int snd_card_disconnect(struct snd_card *card) #endif /* notify all devices that we are disconnected */ - err = snd_device_disconnect_all(card); - if (err < 0) - dev_err(card->dev, "not all devices for card %i can be disconnected\n", card->number); + snd_device_disconnect_all(card); snd_info_card_disconnect(card); if (card->registered) { From 9a6246ff78ac33af78f82704cde6fec361597eea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 18:17:28 +0100 Subject: [PATCH 093/411] ALSA: hda - Implement unbind more safely Now we have all pieces ready, and put them into places: - add the hda_pcm refcount to azx_pcm_open() and azx_pcm_close(), - call the most of cleanup code in hda_codec_reset() from the codec driver remove, - call the same code also from the hda_codec object free. Then the codec driver can be unbound more safely now. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 3 +- sound/pci/hda/hda_codec.c | 70 ++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_controller.c | 3 ++ sound/pci/hda/hda_local.h | 1 + 5 files changed, 43 insertions(+), 35 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 2d00417494e25a..311896a23cd165 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -125,8 +125,7 @@ static int hda_codec_driver_remove(struct device *dev) if (codec->patch_ops.free) codec->patch_ops.free(codec); - codec->preset = NULL; - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + snd_hda_codec_cleanup_for_unbind(codec); module_put(dev->driver->owner); return 0; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3bd9158addc2b8..2c7e481a917147 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1160,36 +1160,62 @@ struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); +/* + * codec destructor + */ static void codec_release_pcms(struct hda_codec *codec) { struct hda_pcm *pcm, *n; list_for_each_entry_safe(pcm, n, &codec->pcm_list_head, list) { list_del_init(&pcm->list); + if (pcm->pcm) + snd_device_disconnect(codec->card, pcm->pcm); snd_hda_codec_pcm_put(pcm); } } -/* - * codec destructor - */ +void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) +{ + cancel_delayed_work_sync(&codec->jackpoll_work); + flush_workqueue(codec->bus->workq); + if (!codec->in_freeing) + snd_hda_ctls_clear(codec); + codec_release_pcms(codec); + snd_hda_detach_beep_device(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + snd_hda_jack_tbl_clear(codec); + codec->proc_widget_hook = NULL; + codec->spec = NULL; + + free_hda_cache(&codec->amp_cache); + free_hda_cache(&codec->cmd_cache); + init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); + init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + + /* free only driver_pins so that init_pins + user_pins are restored */ + snd_array_free(&codec->driver_pins); + snd_array_free(&codec->cvt_setups); + snd_array_free(&codec->spdif_out); + snd_array_free(&codec->verbs); + codec->preset = NULL; + codec->slave_dig_outs = NULL; + codec->spdif_status_reset = 0; + snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); + remove_conn_list(codec); +} + static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; - cancel_delayed_work_sync(&codec->jackpoll_work); - codec_release_pcms(codec); + codec->in_freeing = 1; if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); - snd_hda_jack_tbl_clear(codec); free_init_pincfgs(codec); flush_workqueue(codec->bus->workq); list_del(&codec->list); - snd_array_free(&codec->mixers); - snd_array_free(&codec->nids); - snd_array_free(&codec->cvt_setups); - snd_array_free(&codec->spdif_out); - remove_conn_list(codec); codec->bus->caddr_tbl[codec->addr] = NULL; clear_bit(codec->addr, &codec->bus->codec_powered); snd_hda_sysfs_clear(codec); @@ -2479,31 +2505,9 @@ int snd_hda_codec_reset(struct hda_codec *codec) return -EBUSY; /* OK, let it free */ - cancel_delayed_work_sync(&codec->jackpoll_work); - flush_workqueue(bus->workq); - snd_hda_ctls_clear(codec); - codec_release_pcms(codec); - snd_hda_detach_beep_device(codec); if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); - snd_hda_jack_tbl_clear(codec); - codec->proc_widget_hook = NULL; - codec->spec = NULL; - free_hda_cache(&codec->amp_cache); - free_hda_cache(&codec->cmd_cache); - init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); - init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - /* free only driver_pins so that init_pins + user_pins are restored */ - snd_array_free(&codec->driver_pins); - snd_array_free(&codec->cvt_setups); - snd_array_free(&codec->spdif_out); - snd_array_free(&codec->verbs); - codec->preset = NULL; - codec->slave_dig_outs = NULL; - codec->spdif_status_reset = 0; - /* allow device access again */ snd_hda_unlock_devices(bus); return 0; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2ccd6f9a91fec5..fc62ca51fd3533 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -350,6 +350,7 @@ struct hda_codec { #endif /* misc flags */ + unsigned int in_freeing:1; /* being released */ unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each * status change * (e.g. Realtek codecs) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ad85f9bfaf57bb..cae50d5ffb814f 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -420,6 +420,7 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) hinfo->ops.close(hinfo, apcm->codec, substream); snd_hda_power_down(apcm->codec); mutex_unlock(&chip->open_mutex); + snd_hda_codec_pcm_put(apcm->info); return 0; } @@ -806,6 +807,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) int err; int buff_step; + snd_hda_codec_pcm_get(apcm->info); mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { @@ -887,6 +889,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_hda_power_down(apcm->codec); unlock: mutex_unlock(&chip->open_mutex); + snd_hda_codec_pcm_put(apcm->info); return err; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 8588813163e31d..1d001647fc47bb 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -150,6 +150,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, #define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL) int snd_hda_codec_reset(struct hda_codec *codec); +void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); enum { HDA_VMUTE_OFF, From bcd96557bd0ab1129fcdde073d5700aed8fcb942 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 20:44:54 +0100 Subject: [PATCH 094/411] ALSA: hda - Build PCMs and controls at codec driver probe This makes the code flow easier -- instead of the controller driver calling snd_hda_build_pcms() and snd_hda_build_controls() explicitly, the codec driver itself builds PCMs and controls at probe time. Then the controller driver only needs to call snd_card_register(). Also, this allows us the full bind/unbind control, too. Even when a codec driver is bound later, it automatically registers the new PCM and controls by itself. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 22 ++++++++++--- sound/pci/hda/hda_codec.c | 67 --------------------------------------- sound/pci/hda/hda_codec.h | 2 -- sound/pci/hda/hda_intel.c | 10 ------ sound/pci/hda/hda_tegra.c | 10 ------ 5 files changed, 17 insertions(+), 94 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 311896a23cd165..a49bc45c2ea5ac 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -106,16 +106,28 @@ static int hda_codec_driver_probe(struct device *dev) } err = codec->preset->patch(codec); - if (err < 0) { - module_put(owner); - goto error; + if (err < 0) + goto error_module; + + err = snd_hda_codec_build_pcms(codec); + if (err < 0) + goto error_module; + err = snd_hda_codec_build_controls(codec); + if (err < 0) + goto error_module; + if (codec->card->registered) { + err = snd_card_register(codec->card); + if (err < 0) + goto error_module; } return 0; + error_module: + module_put(owner); + error: - codec->preset = NULL; - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + snd_hda_codec_cleanup_for_unbind(codec); return err; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c7e481a917147..7085d3733d0dec 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4031,36 +4031,6 @@ const struct dev_pm_ops hda_codec_driver_pm = { NULL) }; -/** - * snd_hda_build_controls - build mixer controls - * @bus: the BUS - * - * Creates mixer controls for each codec included in the bus. - * - * Returns 0 if successful, otherwise a negative error code. - */ -int snd_hda_build_controls(struct hda_bus *bus) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &bus->codec_list, list) { - int err = snd_hda_codec_build_controls(codec); - if (err < 0) { - codec_err(codec, - "cannot build controls for #%d (error %d)\n", - codec->addr, err); - err = snd_hda_codec_reset(codec); - if (err < 0) { - codec_err(codec, - "cannot revert codec\n"); - return err; - } - } - } - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_build_controls); - /* * add standard channel maps if not specified */ @@ -4692,43 +4662,6 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) return 0; } -/** - * snd_hda_build_pcms - build PCM information - * @bus: the BUS - * - * Create PCM information for each codec included in the bus. - * - * The build_pcms codec patch is requested to create and assign new - * hda_pcm objects. The codec is responsible to call snd_hda_codec_pcm_new() - * and fills the fields. Later they are instantiated by this function. - * - * At least, substreams, channels_min and channels_max must be filled for - * each stream. substreams = 0 indicates that the stream doesn't exist. - * When rates and/or formats are zero, the supported values are queried - * from the given nid. The nid is used also by the default ops.prepare - * and ops.cleanup callbacks. - * - * The driver needs to call ops.open in its open callback. Similarly, - * ops.close is supposed to be called in the close callback. - * ops.prepare should be called in the prepare or hw_params callback - * with the proper parameters for set up. - * ops.cleanup should be called in hw_free for clean up of streams. - * - * This function returns 0 if successful, or a negative error code. - */ -int snd_hda_build_pcms(struct hda_bus *bus) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &bus->codec_list, list) { - int err = snd_hda_codec_build_pcms(codec); - if (err < 0) - return err; - } - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_build_pcms); - /** * snd_hda_add_new_ctls - create controls from the array * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fc62ca51fd3533..46d253e2f26637 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -516,13 +516,11 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid); /* * Mixer */ -int snd_hda_build_controls(struct hda_bus *bus); int snd_hda_codec_build_controls(struct hda_codec *codec); /* * PCM */ -int snd_hda_build_pcms(struct hda_bus *bus); int snd_hda_codec_parse_pcms(struct hda_codec *codec); int snd_hda_codec_build_pcms(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f7fb1b51d4464c..e81461a413b8b8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1895,16 +1895,6 @@ static int azx_probe_continue(struct azx *chip) goto out_free; } - /* create PCM streams */ - err = snd_hda_build_pcms(chip->bus); - if (err < 0) - goto out_free; - - /* create mixer controls */ - err = snd_hda_build_controls(chip->bus); - if (err < 0) - goto out_free; - err = snd_card_register(chip->card); if (err < 0) goto out_free; diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 1359fdd20f02bd..7586abe91dfb3f 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -497,16 +497,6 @@ static int hda_tegra_probe(struct platform_device *pdev) if (err < 0) goto out_free; - /* create PCM streams */ - err = snd_hda_build_pcms(chip->bus); - if (err < 0) - goto out_free; - - /* create mixer controls */ - err = snd_hda_build_controls(chip->bus); - if (err < 0) - goto out_free; - err = snd_card_register(chip->card); if (err < 0) goto out_free; From 2f35c630f7d49efdef29b58d81ed2531ddd916d9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 22:43:26 +0100 Subject: [PATCH 095/411] ALSA: hda - Use standard workqueue for unsol and jack events The events that are handled by HD-audio drivers are no frequent and urgent ones, so we can use the standard workqueue without any problem nowadays. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 26 +++++--------------------- sound/pci/hda/hda_codec.h | 2 -- sound/pci/hda/hda_intel.c | 8 ++++---- sound/pci/hda/patch_ca0132.c | 3 +-- sound/pci/hda/patch_hdmi.c | 8 +++----- sound/pci/hda/patch_via.c | 3 +-- 6 files changed, 14 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7085d3733d0dec..f2ccb39a37881f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -681,7 +681,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) struct hda_bus_unsolicited *unsol; unsigned int wp; - if (!bus || !bus->workq) + if (!bus) return 0; trace_hda_unsol_event(bus, res, res_ex); @@ -693,7 +693,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) unsol->queue[wp] = res; unsol->queue[wp + 1] = res_ex; - queue_work(bus->workq, &unsol->work); + schedule_work(&unsol->work); return 0; } @@ -732,13 +732,9 @@ static void snd_hda_bus_free(struct hda_bus *bus) return; WARN_ON(!list_empty(&bus->codec_list)); - if (bus->workq) - flush_workqueue(bus->workq); + cancel_work_sync(&bus->unsol.work); if (bus->ops.private_free) bus->ops.private_free(bus); - if (bus->workq) - destroy_workqueue(bus->workq); - kfree(bus); } @@ -785,16 +781,6 @@ int snd_hda_bus_new(struct snd_card *card, INIT_LIST_HEAD(&bus->codec_list); INIT_WORK(&bus->unsol.work, process_unsol_events); - snprintf(bus->workq_name, sizeof(bus->workq_name), - "hd-audio%d", card->number); - bus->workq = create_singlethread_workqueue(bus->workq_name); - if (!bus->workq) { - dev_err(card->dev, "cannot create workqueue %s\n", - bus->workq_name); - kfree(bus); - return -ENOMEM; - } - err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops); if (err < 0) { snd_hda_bus_free(bus); @@ -1068,8 +1054,8 @@ static void hda_jackpoll_work(struct work_struct *work) if (!codec->jackpoll_interval) return; - queue_delayed_work(codec->bus->workq, &codec->jackpoll_work, - codec->jackpoll_interval); + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); } static void init_hda_cache(struct hda_cache_rec *cache, @@ -1178,7 +1164,6 @@ static void codec_release_pcms(struct hda_codec *codec) void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) { cancel_delayed_work_sync(&codec->jackpoll_work); - flush_workqueue(codec->bus->workq); if (!codec->in_freeing) snd_hda_ctls_clear(codec); codec_release_pcms(codec); @@ -1214,7 +1199,6 @@ static void snd_hda_codec_free(struct hda_codec *codec) if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); free_init_pincfgs(codec); - flush_workqueue(codec->bus->workq); list_del(&codec->list); codec->bus->caddr_tbl[codec->addr] = NULL; clear_bit(codec->addr, &codec->bus->codec_powered); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 46d253e2f26637..bf9efb7e1b9a3f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -132,8 +132,6 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited unsol; - char workq_name[16]; - struct workqueue_struct *workq; /* common workqueue for codecs */ /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e81461a413b8b8..dbc5a593da46af 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -528,10 +528,10 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) if (ok == 1) { azx_dev->irq_pending = 0; return ok; - } else if (ok == 0 && chip->bus && chip->bus->workq) { + } else if (ok == 0) { /* bogus IRQ, process it later */ azx_dev->irq_pending = 1; - queue_work(chip->bus->workq, &hda->irq_pending_work); + schedule_work(&hda->irq_pending_work); } return 0; } @@ -893,8 +893,8 @@ static int azx_runtime_resume(struct device *dev) if (status && bus) { list_for_each_entry(codec, &bus->codec_list, list) if (status & (1 << codec->addr)) - queue_delayed_work(codec->bus->workq, - &codec->jackpoll_work, codec->jackpoll_interval); + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); } /* disable controller Wake Up event*/ diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 555781fad26f00..72d20652df504b 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4410,8 +4410,7 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) * state machine run. */ cancel_delayed_work_sync(&spec->unsol_hp_work); - queue_delayed_work(codec->bus->workq, &spec->unsol_hp_work, - msecs_to_jiffies(500)); + schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); cb->tbl->block_report = 1; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 708bbed15ea399..7e9ff7b16e5658 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1576,9 +1576,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) update_eld = true; } else if (repoll) { - queue_delayed_work(codec->bus->workq, - &per_pin->work, - msecs_to_jiffies(300)); + schedule_delayed_work(&per_pin->work, + msecs_to_jiffies(300)); goto unlock; } } @@ -2198,11 +2197,10 @@ static void generic_hdmi_free(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); - cancel_delayed_work(&per_pin->work); + cancel_delayed_work_sync(&per_pin->work); eld_proc_free(per_pin); } - flush_workqueue(codec->bus->workq); hdmi_array_free(spec); kfree(spec); } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 11a05638e03b9d..2112fbe9e57702 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -222,8 +222,7 @@ static void vt1708_update_hp_work(struct hda_codec *codec) if (!spec->hp_work_active) { codec->jackpoll_interval = msecs_to_jiffies(100); snd_hda_codec_write(codec, 0x1, 0, 0xf81, 0); - queue_delayed_work(codec->bus->workq, - &codec->jackpoll_work, 0); + schedule_delayed_work(&codec->jackpoll_work, 0); spec->hp_work_active = true; } } else if (!hp_detect_with_aa(codec)) From d56db741b8e688a0b9d4d5bb9caa11dfcb7c0b08 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 23:25:35 +0100 Subject: [PATCH 096/411] ALSA: hda - Release resources in device release callback Move the destructor code to device release callback for the codec object instead. This is a safer place to release the resources than dev_free callback in general. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 51 ++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2ccb39a37881f..6fecf57c8d7c5a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1191,28 +1191,6 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) remove_conn_list(codec); } -static void snd_hda_codec_free(struct hda_codec *codec) -{ - if (!codec) - return; - codec->in_freeing = 1; - if (device_is_registered(hda_codec_dev(codec))) - device_del(hda_codec_dev(codec)); - free_init_pincfgs(codec); - list_del(&codec->list); - codec->bus->caddr_tbl[codec->addr] = NULL; - clear_bit(codec->addr, &codec->bus->codec_powered); - snd_hda_sysfs_clear(codec); - free_hda_cache(&codec->amp_cache); - free_hda_cache(&codec->cmd_cache); - kfree(codec->vendor_name); - kfree(codec->chip_name); - kfree(codec->modelname); - kfree(codec->wcaps); - codec->bus->num_codecs--; - put_device(hda_codec_dev(codec)); -} - static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1241,14 +1219,32 @@ static int snd_hda_codec_dev_disconnect(struct snd_device *device) static int snd_hda_codec_dev_free(struct snd_device *device) { - snd_hda_codec_free(device->device_data); + struct hda_codec *codec = device->device_data; + + codec->in_freeing = 1; + if (device_is_registered(hda_codec_dev(codec))) + device_del(hda_codec_dev(codec)); + put_device(hda_codec_dev(codec)); return 0; } -/* just free the container */ static void snd_hda_codec_dev_release(struct device *dev) { - kfree(dev_to_hda_codec(dev)); + struct hda_codec *codec = dev_to_hda_codec(dev); + + free_init_pincfgs(codec); + list_del(&codec->list); + codec->bus->caddr_tbl[codec->addr] = NULL; + clear_bit(codec->addr, &codec->bus->codec_powered); + snd_hda_sysfs_clear(codec); + free_hda_cache(&codec->amp_cache); + free_hda_cache(&codec->cmd_cache); + kfree(codec->vendor_name); + kfree(codec->chip_name); + kfree(codec->modelname); + kfree(codec->wcaps); + codec->bus->num_codecs--; + kfree(codec); } /** @@ -1362,7 +1358,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { - dev_err(card->dev, "no AFG or MFG node found\n"); + codec_err(codec, "no AFG or MFG node found\n"); err = -ENODEV; goto error; } @@ -1408,7 +1404,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, return 0; error: - snd_hda_codec_free(codec); + put_device(hda_codec_dev(codec)); return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_new); @@ -2464,7 +2460,6 @@ void snd_hda_unlock_devices(struct hda_bus *bus) { struct snd_card *card = bus->card; - card = bus->card; spin_lock(&card->files_lock); card->shutdown = 0; spin_unlock(&card->files_lock); From 0004defd4e44d81966b0c4164c2ee01f20ab357b Mon Sep 17 00:00:00 2001 From: Vishal Thanki Date: Tue, 3 Mar 2015 18:59:00 +0530 Subject: [PATCH 097/411] ASoC: simple-card: Add a NULL pointer check in asoc_simple_card_dai_link_of Make sure devm_kzalloc() succeeds. Signed-off-by: Vishal Thanki Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5daee..fb550b5869d21f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto dai_link_of_err; + } + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; From 1a6ab46fa9c2bc9399694b4856ab7ea19c036485 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Wed, 4 Mar 2015 10:56:13 +0900 Subject: [PATCH 098/411] ALSA: Fix spelling typo in Documentation/DocBook/alsa-driver-api.xml This patch fix spelling typo found in alsa-driver-api.xml. It is because this file is generated from comments in source files, I have to fix source files. Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 4 ++-- include/sound/control.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/compress_offload.h | 2 +- sound/core/pcm_dmaengine.c | 4 ++-- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f48089d364c540..fa1d05512c0984 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -70,7 +70,7 @@ struct snd_compr_runtime { * @device: device pointer * @direction: stream direction, playback/recording * @metadata_set: metadata set flag, true when set - * @next_track: has userspace signall next track transistion, true when set + * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -95,7 +95,7 @@ struct snd_compr_stream { * and the stream properties * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream - * @get_metadata: retreives the requested metadata values from stream + * @get_metadata: retrieves the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054023f711..95aad6d3fd1a9b 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * Add a virtual slave control to the given master. * Unlike snd_ctl_add_slave(), the element added via this function * is supposed to have volatile values, and get callback is called - * at each time quried from the master. + * at each time queried from the master. * * When the control peeks the hardware values directly and the value * can be changed by other means than the put callback of the element, diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade19562857..cf0bb156d6da89 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1469,7 +1469,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( } /** - * snd_soc_kcontrol_platform() - Returns the platform that registerd the control + * snd_soc_kcontrol_platform() - Returns the platform that registered the control * @kcontrol: The control for which to get the platform * * Note: This function will only work correctly if the control has been diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 22ed8cb7800b40..e00d8cbfc6282c 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -75,7 +75,7 @@ struct snd_compr_tstamp { /** * struct snd_compr_avail - avail descriptor * @avail: Number of bytes available in ring buffer for writing/reading - * @tstamp: timestamp infomation + * @tstamp: timestamp information */ struct snd_compr_avail { __u64 avail; diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6542c40835949f..fba365a783909f 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -289,7 +289,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); * * The function should usually be called from the pcm open callback. Note that * this function will use private_data field of the substream's runtime. So it - * is not availabe to your pcm driver implementation. + * is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, struct dma_chan *chan) @@ -328,7 +328,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); * This function will request a DMA channel using the passed filter function and * data. The function should usually be called from the pcm open callback. Note * that this function will use private_data field of the substream's runtime. So - * it is not availabe to your pcm driver implementation. + * it is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) From 8b28c93fe5a55873ce22b7126e84eb59290f8603 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Mar 2015 15:37:01 +0100 Subject: [PATCH 099/411] ALSA: usb-audio: Check Marantz/Denon USB DACs in a single place There are three places doing the same check. Let's make them together. Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 40 +++++++++++++++++----------------------- 1 file changed, 17 insertions(+), 23 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 753a47de8459b7..353532b8aee444 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1120,17 +1120,24 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) /* Marantz/Denon USB DACs need a vendor cmd to switch * between PCM and native DSD mode */ +static bool is_marantz_denon_dac(unsigned int id) +{ + switch (id) { + case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ + case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ + case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + return true; + } + return false; +} + int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) { struct usb_device *dev = subs->dev; int err; - switch (subs->stream->chip->usb_id) { - case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ - case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ - case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ - + if (is_marantz_denon_dac(subs->stream->chip->usb_id)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1203,17 +1210,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, /* Marantz/Denon devices with USB DAC functionality need a delay * after each class compliant request */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { - - switch (le16_to_cpu(dev->descriptor.idProduct)) { - case 0x1003: /* Denon DA300-USB */ - case 0x3005: /* Marantz HD-DAC1 */ - case 0x3006: /* Marantz SA-14S1 */ - mdelay(20); - break; - } - } + if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct))) + && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(20); /* Zoom R16/24 needs a tiny delay here, otherwise requests like * get/set frequency return as failed despite actually succeeding. @@ -1268,15 +1268,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* Denon/Marantz devices with USB DAC functionality */ - switch (chip->usb_id) { - case USB_ID(0x154e, 0x1003): /* Denon DA300-USB */ - case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ - case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + if (is_marantz_denon_dac(chip->usb_id)) { if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; - break; - default: - break; } return 0; From c472b93990e02c31f02322ddf0fdd9d571169310 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:16 +0100 Subject: [PATCH 100/411] ASoC: sn95031: Pass CODEC to sn95031_jack_detection() The sn95031 driver currently gets the CODEC implicitly from the jack that is passed to sn95031_jack_detection(). But the codec field is going to be removed from the snd_soc_jack struct, so refactor things to pass the CODEC explicitly. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 14 ++++++++------ sound/soc/codecs/sn95031.h | 3 ++- sound/soc/intel/mfld_machine.c | 13 +++++++------ 3 files changed, 17 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 47b257e4180965..1e5d2643c28606 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -783,19 +783,21 @@ static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); } -static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) +static int sn95031_get_headset_state(struct snd_soc_codec *codec, + struct snd_soc_jack *mfld_jack) { - int micbias = sn95031_get_mic_bias(mfld_jack->codec); + int micbias = sn95031_get_mic_bias(codec); int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) - sn95031_enable_jack_btn(mfld_jack->codec); + sn95031_enable_jack_btn(codec); return jack_type; } -void sn95031_jack_detection(struct mfld_jack_data *jack_data) +void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data) { unsigned int status; unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET; @@ -809,11 +811,11 @@ void sn95031_jack_detection(struct mfld_jack_data *jack_data) status = SND_JACK_HEADSET | SND_JACK_BTN_1; } else if (jack_data->intr_id & 0x4) { pr_debug("headset or headphones inserted\n"); - status = sn95031_get_headset_state(jack_data->mfld_jack); + status = sn95031_get_headset_state(codec, jack_data->mfld_jack); } else if (jack_data->intr_id & 0x8) { pr_debug("headset or headphones removed\n"); status = 0; - sn95031_disable_jack_btn(jack_data->mfld_jack->codec); + sn95031_disable_jack_btn(codec); } else { pr_err("unidentified interrupt\n"); return; diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index 20376d234fb83e..7651fe4e6a4589 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -127,6 +127,7 @@ struct mfld_jack_data { struct snd_soc_jack *mfld_jack; }; -extern void sn95031_jack_detection(struct mfld_jack_data *jack_data); +extern void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data); #endif diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 90b7a57713a023..d22b44db824ee8 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -228,10 +228,13 @@ static void mfld_jack_check(unsigned int intr_status) { struct mfld_jack_data jack_data; + if (!mfld_codec) + return; + jack_data.mfld_jack = &mfld_jack; jack_data.intr_id = intr_status; - sn95031_jack_detection(&jack_data); + sn95031_jack_detection(mfld_codec, &jack_data); /* TODO: add american headset detection post gpiolib support */ } @@ -240,8 +243,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_dapm_context *dapm = &runtime->card->dapm; int ret_val; - mfld_codec = runtime->codec; - /* default is earpiece pin, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ @@ -254,7 +255,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", + ret_val = snd_soc_jack_new(runtime->codec, "Intel(R) MID Audio Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack); if (ret_val) { @@ -275,6 +276,8 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } + mfld_codec = runtime->codec; + /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc * to find if anything is there or not */ @@ -359,8 +362,6 @@ static irqreturn_t snd_mfld_jack_detection(int irq, void *data) { struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - if (mfld_jack.codec == NULL) - return IRQ_HANDLED; mfld_jack_check(mc_drv_ctx->interrupt_status); return IRQ_HANDLED; From 970939964c26db4643f58d4e84487821962e0b65 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:17 +0100 Subject: [PATCH 101/411] ASoC: Allow to register jacks at the card level Jacks are typically card level elements, but are currently registered with a CODEC. When it was originally introduced snd_soc_jack_new() took a snd_soc_card as its parameter, but at that time DAPM was only implemented at the CODEC level and there was only one CODEC per card. This made it clear which CODEC to use for the jack DAPM operations. But the multi-component patchset added support for having multiple CODECs per card and with it the API was updated to register jacks with a specific CODEC instance instead. Subsequently DAPM support at the card level has been introduced, but the snd_soc_jack_new() API has so remained unchanged. This leaves us with the issue that the DAPM pins that are managed by the jack detection logic usually are part of the card DAPM context but are accessed through a CODEC DAPM context. Currently this works fine, but might break in the future if we take a more hierarchical approach to DAPM contexts. Furthermore with componentization progressing systems that do not register a snd_soc_codec might appear, while these system may still want to able to register a jack. This patch addresses these issues by adding a new function called snd_soc_card_jack_new() that can be used to register jacks with the card rather than a CODEC. This new function is mostly identical to snd_soc_jack_new() except that it additionally allows to directly specify the DAPM pins associated with the jack. This was done since most users of snd_soc_jack_new() typically call snd_soc_jack_add_pins() right after it, which is not necessary with the new API and allows to reduce the amount of boiler plate code. The old snd_soc_jack_new() is re-implemented as a wrapper around snd_soc_card_jack_new(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 28 +++++++++++++++++++++++++--- sound/soc/soc-jack.c | 42 ++++++++++++++++++++++++++---------------- 2 files changed, 51 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade19562857..99d9ce27220985 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -450,8 +450,10 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); /* Jack reporting */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack); +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins); + void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); @@ -659,7 +661,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack { struct mutex mutex; struct snd_jack *jack; - struct snd_soc_codec *codec; + struct snd_soc_card *card; struct list_head pins; int status; struct blocking_notifier_head notifier; @@ -1482,6 +1484,26 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform( return snd_soc_component_to_platform(snd_soc_kcontrol_component(kcontrol)); } +/** + * snd_soc_jack_new - Create a new jack + * @codec: ASoC CODEC + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jack: structure to use for the jack + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jack will be initialised. + */ +static inline int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, + int type, struct snd_soc_jack *jack) +{ + return snd_soc_card_jack_new(codec->component.card, id, type, jack, + NULL, 0); +} + int snd_soc_util_init(void); void snd_soc_util_exit(void); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4380dcc064a530..9f60c25c4568e7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,30 +22,42 @@ #include /** - * snd_soc_jack_new - Create a new jack - * @codec: ASoC codec + * snd_soc_card_jack_new - Create a new jack + * @card: ASoC card * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack * @jack: structure to use for the jack + * @pins: Array of jack pins to be added to the jack or NULL + * @num_pins: Number of elements in the @pins array * * Creates a new jack object. * * Returns zero if successful, or a negative error code on failure. * On success jack will be initialised. */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack) +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins) { + int ret; + mutex_init(&jack->mutex); - jack->codec = codec; + jack->card = card; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack); + ret = snd_jack_new(card->snd_card, id, type, &jack->jack); + if (ret) + return ret; + + if (num_pins) + return snd_soc_jack_add_pins(jack, num_pins, pins); + + return 0; } -EXPORT_SYMBOL_GPL(snd_soc_jack_new); +EXPORT_SYMBOL_GPL(snd_soc_card_jack_new); /** * snd_soc_jack_report - Report the current status for a jack @@ -63,7 +75,6 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; unsigned int sync = 0; @@ -74,8 +85,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (!jack) return; - codec = jack->codec; - dapm = &codec->dapm; + dapm = &jack->card->dapm; mutex_lock(&jack->mutex); @@ -175,12 +185,12 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!pins[i].pin) { - dev_err(jack->codec->dev, "ASoC: No name for pin %d\n", + dev_err(jack->card->dev, "ASoC: No name for pin %d\n", i); return -EINVAL; } if (!pins[i].mask) { - dev_err(jack->codec->dev, "ASoC: No mask for pin %d" + dev_err(jack->card->dev, "ASoC: No mask for pin %d" " (%s)\n", i, pins[i].pin); return -EINVAL; } @@ -260,7 +270,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) static irqreturn_t gpio_handler(int irq, void *data) { struct snd_soc_jack_gpio *gpio = data; - struct device *dev = gpio->jack->codec->component.card->dev; + struct device *dev = gpio->jack->card->dev; trace_snd_soc_jack_irq(gpio->name); @@ -299,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!gpios[i].name) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: No name for gpio at index %d\n", i); ret = -EINVAL; goto undo; @@ -320,7 +330,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, } else { /* legacy GPIO number */ if (!gpio_is_valid(gpios[i].gpio)) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Invalid gpio %d\n", gpios[i].gpio); ret = -EINVAL; @@ -350,7 +360,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (gpios[i].wake) { ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1); if (ret != 0) - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Failed to mark GPIO at index %d as wake source: %d\n", i, ret); } From 386669fcec85a16cb81cd19236abe76abe0f1fc1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:18 +0100 Subject: [PATCH 102/411] ASoC: simple-card: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5daee..b8ee47b7ba9c0b 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -176,11 +176,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; if (gpio_is_valid(priv->gpio_hp_det)) { - snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, - &simple_card_hp_jack); - snd_soc_jack_add_pins(&simple_card_hp_jack, - ARRAY_SIZE(simple_card_hp_jack_pins), - simple_card_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &simple_card_hp_jack, + simple_card_hp_jack_pins, + ARRAY_SIZE(simple_card_hp_jack_pins)); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; @@ -189,11 +189,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(priv->gpio_mic_det)) { - snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, - &simple_card_mic_jack); - snd_soc_jack_add_pins(&simple_card_mic_jack, - ARRAY_SIZE(simple_card_mic_jack_pins), - simple_card_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &simple_card_mic_jack, + simple_card_mic_jack_pins, + ARRAY_SIZE(simple_card_mic_jack_pins)); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, From 27cb64b474516421001932d966ca3184795d4e29 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:19 +0100 Subject: [PATCH 103/411] ASoC: imx-es8328: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-es8328.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index f8cf10e16ce9c7..20e7400e2611e4 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -53,9 +53,9 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) /* Headphone jack detection */ if (gpio_is_valid(data->jack_gpio)) { - ret = snd_soc_jack_new(rtd->codec, "Headphone", - SND_JACK_HEADPHONE | SND_JACK_BTN_0, - &headset_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headphone", + SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &headset_jack, NULL, 0); if (ret) return ret; From 47ec96d4ca7e4a7b9b8b115a10d59e89f794ef95 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:20 +0100 Subject: [PATCH 104/411] ASoC: wm1133-ev: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/wm1133-ev1.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index a958937ab405ce..0653aa83c9273c 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -205,16 +205,14 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; /* Headphone jack detection */ - snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); /* Microphone jack detection */ - snd_soc_jack_new(codec, "Microphone", - SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Microphone", + SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack, + mic_jack_pins, ARRAY_SIZE(mic_jack_pins)); wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); From 85c85e5d6d579a5ff8b5471c4e753946eedbf788 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:21 +0100 Subject: [PATCH 105/411] ASoC: broadwell: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01479adcb..9effa3da982f46 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -80,15 +80,9 @@ static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; int ret = 0; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); - - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&broadwell_headset, - ARRAY_SIZE(broadwell_headset_pins), - broadwell_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, + broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); if (ret) return ret; From e0f7dd9d88f4c151aeca45d290e171d907249888 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:22 +0100 Subject: [PATCH 106/411] ASoC: byt-max98090: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 9832afe7d22cb0..d8b1f038da1c41 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -84,7 +84,6 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_codec *codec = runtime->codec; struct snd_soc_card *card = runtime->card; struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); struct snd_soc_jack *jack = &drv->jack; @@ -100,13 +99,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) } /* Enable jack detection */ - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET, jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack, + hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; From fb1edb4b68a829619bcd50a0c23c557000d0d638 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:23 +0100 Subject: [PATCH 107/411] ASoC: cht_bsw_rt5645: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5645.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617a9ab9d2..0bfca2192ca095 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -169,17 +169,17 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - ret = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, - &ctx->hp_jack); + ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", + SND_JACK_HEADPHONE, &ctx->hp_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "HP jack creation failed %d\n", ret); return ret; } - ret = snd_soc_jack_new(codec, "Mic Jack", - SND_JACK_MICROPHONE, - &ctx->mic_jack); + ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", + SND_JACK_MICROPHONE, &ctx->mic_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); return ret; From af13cbc1a288d3921f1af739da84371e6c53aea3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:24 +0100 Subject: [PATCH 108/411] ASoC: mfld_machine: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/mfld_machine.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index d22b44db824ee8..49c09a0add797f 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -255,20 +255,15 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(runtime->codec, "Intel(R) MID Audio Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1, &mfld_jack); + ret_val = snd_soc_card_jack_new(runtime->card, + "Intel(R) MID Audio Jack", SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack, + mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins)); if (ret_val) { pr_err("jack creation failed\n"); return ret_val; } - ret_val = snd_soc_jack_add_pins(&mfld_jack, - ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); - if (ret_val) { - pr_err("adding jack pins failed\n"); - return ret_val; - } ret_val = snd_soc_jack_add_zones(&mfld_jack, ARRAY_SIZE(mfld_zones), mfld_zones); if (ret_val) { From df8c66189dd42f719c75800a526bdc901f300f41 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:25 +0100 Subject: [PATCH 109/411] ASoC: ams-deltea: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 706613077c155e..16cc95fa45731b 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -479,8 +479,8 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ - ret = snd_soc_jack_new(rtd->codec, "hook_switch", - SND_JACK_HEADSET, &ams_delta_hook_switch); + ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET, + &ams_delta_hook_switch, NULL, 0); if (ret) dev_warn(card->dev, "Failed to allocate resources for hook switch, " From 25649592cfa6c210c9f86670472b864782c8d677 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:26 +0100 Subject: [PATCH 110/411] ASoC: omap-abe-twl6040: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index b9c65f1ad5a85e..0843a68f277c2c 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -182,17 +182,17 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* Headset jack detection only if it is supported */ if (priv->jack_detection) { - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - return ret; + return 0; } static const struct snd_soc_dapm_route dmic_audio_map[] = { From da21cf6d65283680247da74c3d03f7e5cdfb40d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:27 +0100 Subject: [PATCH 111/411] ASoC: omap-twl4030: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-twl4030.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index fb1f6bb87cd430..3673ada43bfb24 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -170,14 +170,10 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) if (priv->jack_detect > 0) { hs_jack_gpios[0].gpio = priv->jack_detect; - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &priv->hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&priv->hs_jack, - ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &priv->hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; From 753d45e6b886c93a2a8a88eddaca345643a87f4e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:28 +0100 Subject: [PATCH 112/411] ASoC: rx51: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/rx51.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 7f299357c2d207..c2ddf0fbfa2895 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -311,9 +311,9 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) } /* AV jack detection */ - err = snd_soc_jack_new(codec, "AV Jack", - SND_JACK_HEADSET | SND_JACK_VIDEOOUT, - &rx51_av_jack); + err = snd_soc_card_jack_new(rtd->card, "AV Jack", + SND_JACK_HEADSET | SND_JACK_VIDEOOUT, + &rx51_av_jack, NULL, 0); if (err) { dev_err(card->dev, "Failed to add AV Jack\n"); return err; From f7a4433b498384f0e300c51b654910f3e03b5ca6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:29 +0100 Subject: [PATCH 113/411] ASoC: hx4700: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/hx4700.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 73eb5ddf9753be..9f8be7cd567e48 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -126,17 +126,12 @@ static const struct snd_soc_dapm_route hx4700_audio_map[] = { */ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin), - hs_jack_pin); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin, + ARRAY_SIZE(hs_jack_pin)); if (err) return err; From bc1e2e06a07ad4c0c021165b34fa8259bdf4d8c6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:30 +0100 Subject: [PATCH 114/411] ASoC: palm27x: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/palm27x.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 910336c5ebebd1..c20bbc042425da 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -75,17 +75,12 @@ static struct snd_soc_card palm27x_asoc; static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (err) return err; From 3b14125bc553a0fe091a5d43a22be41cdc43b156 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:31 +0100 Subject: [PATCH 115/411] ASoC: ttc-dkb: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/ttc-dkb.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 5001dbb9b25776..1753c7d9e760e4 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -78,15 +78,12 @@ static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); + snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack, mic_jack_pins, + ARRAY_SIZE(mic_jack_pins)); /* headphone, microphone detection & headset short detection */ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, From d30d141f9cb7eb9fb3f03af11146dc0d2b393ff2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:32 +0100 Subject: [PATCH 116/411] ASoC: z2: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/z2.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 76ccb172d0a77e..bcbfbe8303f7a0 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -143,13 +143,9 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "MONO1"); /* Jack detection API stuff */ - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &hs_jack); - if (ret) - goto err; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) goto err; From dfe11f282c61808f7140d9dd741f7e54cf97cda6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:33 +0100 Subject: [PATCH 117/411] ASoC: h1980_uda1380: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/h1940_uda1380.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 59b044255b781c..c72e9fb26658ce 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -162,13 +162,8 @@ static struct platform_device *s3c24xx_snd_device; static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); From 39ec5109d6089e1acd04b51b9df5349f5b8a7f5c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:34 +0100 Subject: [PATCH 118/411] ASoC: littlemill: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/littlemill.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 141519c21e21d7..31a820eb0ac37b 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -260,12 +260,12 @@ static int littlemill_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_MECHANICAL | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | - SND_JACK_BTN_4 | SND_JACK_BTN_5, - &littlemill_headset); + ret = snd_soc_card_jack_new(card, "Headset", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5, + &littlemill_headset, NULL, 0); if (ret) return ret; From f97e0eacf2b5d9c1a470e53df60519d555ac5a75 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:35 +0100 Subject: [PATCH 119/411] ASoC: lowland: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/lowland.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 243dea7ba38f81..5f156093101e3a 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -56,16 +56,10 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &lowland_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&lowland_headset, - ARRAY_SIZE(lowland_headset_pins), - lowland_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &lowland_headset, lowland_headset_pins, + ARRAY_SIZE(lowland_headset_pins)); if (ret) return ret; From e9c9a723eea5102fa6adedf454e02fff6201a3c3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:36 +0100 Subject: [PATCH 120/411] ASoC: rx1950_uda1380: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/rx1950_uda1380.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 873f2cb4bebe3e..35e37c457f1fd3 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -211,13 +211,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); From 55b2ed2d9dd8c611837f34ca29df881eb0a1de8d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:37 +0100 Subject: [PATCH 121/411] ASoC: smartq: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/smartq_wm8987.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 8291d2a5f15259..dfbe2db1c4078d 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -151,13 +151,10 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); /* Headphone jack detection */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &smartq_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), - smartq_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack, + smartq_jack_pins, + ARRAY_SIZE(smartq_jack_pins)); if (err) return err; From 663976ad478b50664353fdf19a5a3dcad3cb4e22 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:38 +0100 Subject: [PATCH 122/411] ASoC: speyside: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 5ec7c52282f201..2dcb988bdff212 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -153,16 +153,10 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) pr_err("Failed to request HP_SEL GPIO: %d\n", ret); gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity); - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &speyside_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&speyside_headset, - ARRAY_SIZE(speyside_headset_pins), - speyside_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &speyside_headset, speyside_headset_pins, + ARRAY_SIZE(speyside_headset_pins)); if (ret) return ret; From 3fd94f37da000a2b562a3f4e6c553b7ab1ad9e19 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:39 +0100 Subject: [PATCH 123/411] ASoC: tobermory: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/tobermory.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 9c80506527c4f3..85ccfb7188cb2d 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -179,15 +179,10 @@ static int tobermory_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &tobermory_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&tobermory_headset, - ARRAY_SIZE(tobermory_headset_pins), - tobermory_headset_pins); + ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | + SND_JACK_BTN_0, &tobermory_headset, + tobermory_headset_pins, + ARRAY_SIZE(tobermory_headset_pins)); if (ret) return ret; From 12cc6d1dca4d3a9e929090cb0cf9ef452f414518 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:40 +0100 Subject: [PATCH 124/411] ASoC: tegra_alc5632: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 769aca2fc5f5af..6dcd06a966c7c6 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -106,11 +106,10 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &tegra_alc5632_hs_jack); - snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, - ARRAY_SIZE(tegra_alc5632_hs_jack_pins), - tegra_alc5632_hs_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack, + tegra_alc5632_hs_jack_pins, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; From d020e77c61b8a9d563d205cfcec7e71090d1377d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:41 +0100 Subject: [PATCH 125/411] ASoC: tegra_max98090: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 22 ++++++++++------------ 1 file changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb997b75228..6760f0ebc13388 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -141,16 +141,14 @@ static const struct snd_kcontrol_new tegra_max98090_controls[] = { static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_max98090_hp_jack); - snd_soc_jack_add_pins(&tegra_max98090_hp_jack, - ARRAY_SIZE(tegra_max98090_hp_jack_pins), - tegra_max98090_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &tegra_max98090_hp_jack, + tegra_max98090_hp_jack_pins, + ARRAY_SIZE(tegra_max98090_hp_jack_pins)); tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_add_gpios(&tegra_max98090_hp_jack, @@ -159,11 +157,11 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(machine->gpio_mic_det)) { - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_max98090_mic_jack); - snd_soc_jack_add_pins(&tegra_max98090_mic_jack, - ARRAY_SIZE(tegra_max98090_mic_jack_pins), - tegra_max98090_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &tegra_max98090_mic_jack, + tegra_max98090_mic_jack_pins, + ARRAY_SIZE(tegra_max98090_mic_jack_pins)); tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det; snd_soc_jack_add_gpios(&tegra_max98090_mic_jack, From 00eafe3b1b191c9b2611b74c03e1b573ae257b1e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:42 +0100 Subject: [PATCH 126/411] ASoC: tegra_rt5640: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5640.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index ed759a3076b8b5..773daecaa5e89f 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -108,15 +108,11 @@ static const struct snd_kcontrol_new tegra_rt5640_controls[] = { static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_rt5640_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5640_hp_jack, - ARRAY_SIZE(tegra_rt5640_hp_jack_pins), - tegra_rt5640_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", SND_JACK_HEADPHONE, + &tegra_rt5640_hp_jack, tegra_rt5640_hp_jack_pins, + ARRAY_SIZE(tegra_rt5640_hp_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; From 783b1e7948010ded40eba784b558d86d72ae2ef4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:43 +0100 Subject: [PATCH 127/411] ASoC: tegra_rt5677: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5677.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index e4cf978a6e3a9a..68d8b67e79c152 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -146,10 +146,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_rt5677_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, - &tegra_rt5677_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack, + &tegra_rt5677_hp_jack_pins, 1); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; @@ -158,10 +157,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_rt5677_mic_jack); - snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, - &tegra_rt5677_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack, + &tegra_rt5677_mic_jack_pins, 1); if (gpio_is_valid(machine->gpio_mic_present)) { tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; From 7ba8cbb2f0fd9ff232fa19159e2646bf64135126 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:44 +0100 Subject: [PATCH 128/411] ASoC: tegra_wm8903: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index e52420dae2b4b0..4a95b70f0cf082 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -177,21 +177,19 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) if (gpio_is_valid(machine->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_wm8903_hp_jack); - snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, - ARRAY_SIZE(tegra_wm8903_hp_jack_pins), - tegra_wm8903_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &tegra_wm8903_hp_jack, + tegra_wm8903_hp_jack_pins, + ARRAY_SIZE(tegra_wm8903_hp_jack_pins)); snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_wm8903_mic_jack); - snd_soc_jack_add_pins(&tegra_wm8903_mic_jack, - ARRAY_SIZE(tegra_wm8903_mic_jack_pins), - tegra_wm8903_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_wm8903_mic_jack, + tegra_wm8903_mic_jack_pins, + ARRAY_SIZE(tegra_wm8903_mic_jack_pins)); wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, 0); From 77c71765ef78f87dd901fcb4c751908e835a3b2e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:45 +0100 Subject: [PATCH 129/411] ASoC: Remove snd_soc_jack_new() There are no users of snd_soc_jack_new() left and new users should use snd_soc_card_jack_new() instead. So remove the function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 20 -------------------- 1 file changed, 20 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 99d9ce27220985..40b3ee96f3177a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1484,26 +1484,6 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform( return snd_soc_component_to_platform(snd_soc_kcontrol_component(kcontrol)); } -/** - * snd_soc_jack_new - Create a new jack - * @codec: ASoC CODEC - * @id: an identifying string for this jack - * @type: a bitmask of enum snd_jack_type values that can be detected by - * this jack - * @jack: structure to use for the jack - * - * Creates a new jack object. - * - * Returns zero if successful, or a negative error code on failure. - * On success jack will be initialised. - */ -static inline int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, - int type, struct snd_soc_jack *jack) -{ - return snd_soc_card_jack_new(codec->component.card, id, type, jack, - NULL, 0); -} - int snd_soc_util_init(void); void snd_soc_util_exit(void); From 4c03a5ebc7f75e98b32591d1d2c6758c811dcbef Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:17 +0200 Subject: [PATCH 130/411] ASoC: davinci: Select SND_EDMA_SOC when SND_DAVINCI_SOC is enabled edma-pcm going to replace davinci-pcm as platform driver for daVinci platform. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 2b81ca418d2a67..eae4e229f341d0 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,10 +1,11 @@ config SND_DAVINCI_SOC tristate "SoC Audio for TI DAVINCI" depends on ARCH_DAVINCI + select SND_EDMA_SOC config SND_EDMA_SOC tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" - depends on SOC_AM33XX || SOC_AM43XX + depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. From 257ade78b6019cf1570c1239894a7a6a549768e1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:18 +0200 Subject: [PATCH 131/411] ASoC: davinci-i2s: Convert to use edma-pcm The edma-pcm can replace the old davinci-pcm as platform driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 67 +++++++++++++-------------------- 1 file changed, 26 insertions(+), 41 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 15fb28fc8e1b0f..56cb4d95637dda 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -23,8 +23,9 @@ #include #include #include +#include -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" @@ -122,7 +123,8 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { struct davinci_mcbsp_dev { struct device *dev; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 @@ -419,8 +421,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - struct davinci_pcm_dma_params *dma_params = - &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length, master; unsigned int rcr, xcr, srgr, clk_div, freq, framesize; @@ -532,8 +532,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } - dma_params->acnt = dma_params->data_type = data_type[fmt]; - dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; switch (master) { @@ -600,15 +598,6 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -620,7 +609,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -630,7 +618,18 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { }; +static int davinci_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_i2s_dai = { + .probe = davinci_i2s_dai_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -651,11 +650,9 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; - enum dma_event_q asp_chan_q = EVENTQ_0; - enum dma_event_q ram_chan_q = EVENTQ_1; + int *dma; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -676,22 +673,6 @@ static int davinci_i2s_probe(struct platform_device *pdev) GFP_KERNEL); if (!dev) return -ENOMEM; - if (pdata) { - dev->enable_channel_combine = pdata->enable_channel_combine; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = - pdata->sram_size_playback; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = - pdata->sram_size_capture; - dev->clk_input_pin = pdata->clk_input_pin; - dev->i2s_accurate_sck = pdata->i2s_accurate_sck; - asp_chan_q = pdata->asp_chan_q; - ram_chan_q = pdata->ram_chan_q; - } - - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) @@ -705,10 +686,10 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); /* first TX, then RX */ @@ -718,7 +699,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; + dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -726,9 +709,11 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; - dev->dev = &pdev->dev; + dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma; + dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component, @@ -736,7 +721,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); goto err_unregister_component; From 62731d33c41d95914a0a796f319924e22e7ea411 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:19 +0200 Subject: [PATCH 132/411] ASoC: davinci-vcif: Convert to use edma-pcm The edma-pcm can replace the old davinci-pcm as platform driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-vcif.c | 55 ++++++++++++++------------------ 1 file changed, 24 insertions(+), 31 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 5bee04279ebede..fabd05f24aebe9 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -33,8 +33,9 @@ #include #include #include +#include -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" #define MOD_REG_BIT(val, mask, set) do { \ @@ -47,7 +48,8 @@ struct davinci_vcif_dev { struct davinci_vc *davinci_vc; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; }; static void davinci_vcif_start(struct snd_pcm_substream *substream) @@ -93,8 +95,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, { struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; - struct davinci_pcm_dma_params *dma_params = - &davinci_vcif_dev->dma_params[substream->stream]; u32 w; /* Restart the codec before setup */ @@ -113,16 +113,12 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, /* Determine xfer data type */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: - dma_params->data_type = 0; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | DAVINCI_VC_CTRL_WD_UNSIGNED, 1); break; case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_WD_BITS_8, 1); @@ -130,8 +126,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, DAVINCI_VC_CTRL_WD_UNSIGNED, 0); break; case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | @@ -142,8 +136,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - dma_params->acnt = dma_params->data_type; - writel(w, davinci_vc->base + DAVINCI_VC_CTRL); return 0; @@ -172,24 +164,25 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_vcif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - #define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { - .startup = davinci_vcif_startup, .trigger = davinci_vcif_trigger, .hw_params = davinci_vcif_hw_params, }; +static int davinci_vcif_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_vcif_dai = { + .probe = davinci_vcif_dai_probe, .playback = { .channels_min = 1, .channels_max = 2, @@ -225,16 +218,16 @@ static int davinci_vcif_probe(struct platform_device *pdev) /* DMA tx params */ davinci_vcif_dev->davinci_vc = davinci_vc; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = - davinci_vc->davinci_vcif.dma_tx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = - davinci_vc->davinci_vcif.dma_tx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = + &davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + davinci_vc->davinci_vcif.dma_tx_addr; /* DMA rx params */ - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = - davinci_vc->davinci_vcif.dma_rx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = - davinci_vc->davinci_vcif.dma_rx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = + &davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + davinci_vc->davinci_vcif.dma_rx_addr; dev_set_drvdata(&pdev->dev, davinci_vcif_dev); @@ -245,7 +238,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) return ret; } - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); snd_soc_unregister_component(&pdev->dev); From 9759e7ef53138c5ab46ea516ad08977eb5770393 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:20 +0200 Subject: [PATCH 133/411] ASoC: davinci-mcasp: Deprecate the use of davinci-pcm in favor of edma-pcm The edma-pcm performs as good as the old davinci-pcm and it's use does not require the 'ping-pong' mode of davinci-pcm, which was introduced to overcome under/over flow issues when using davinci-pcm. Keep the SND_DAVINCI_SOC config option to select the SND_EDMA_SOC to avoid regression in audio support. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 15 +++--- sound/soc/davinci/davinci-mcasp.c | 87 +++++++------------------------ 2 files changed, 27 insertions(+), 75 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index eae4e229f341d0..3736d9aabc563c 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,15 +1,16 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for TI DAVINCI" + tristate depends on ARCH_DAVINCI select SND_EDMA_SOC config SND_EDMA_SOC - tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" + tristate "SoC Audio for Texas Instruments chips using eDMA" depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. The following line of SoCs are supported by this platform driver: + - daVinci devices - AM335x - AM437x/AM438x @@ -18,7 +19,7 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate "Multichannel Audio Serial Port (McASP) support" - depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC + depends on SND_OMAP_SOC || SND_EDMA_SOC help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: @@ -46,7 +47,7 @@ config SND_AM33XX_SOC_EVM config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on SND_DAVINCI_SOC && I2C + depends on SND_EDMA_SOC && I2C depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_GENERIC_EVM help @@ -74,7 +75,7 @@ endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DM6467_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM select SND_SOC_SPDIF @@ -83,7 +84,7 @@ config SND_DM6467_SOC_EVM config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA830_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help @@ -92,7 +93,7 @@ config SND_DA830_SOC_EVM config SND_DA850_SOC_EVM tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA850_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 031c1fb44ae72b..0c882995a3575a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include @@ -36,7 +37,6 @@ #include #include -#include "davinci-pcm.h" #include "edma-pcm.h" #include "davinci-mcasp.h" @@ -65,7 +65,6 @@ struct davinci_mcasp_context { }; struct davinci_mcasp { - struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; u32 fifo_base; @@ -82,6 +81,7 @@ struct davinci_mcasp { u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; + int dma_request[2]; int sysclk_freq; bool bclk_master; @@ -643,7 +643,6 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int period_words, int channels) { - struct davinci_pcm_dma_params *dma_params = &mcasp->dma_params[stream]; struct snd_dmaengine_dai_dma_data *dma_data = &mcasp->dma_data[stream]; int i; u8 tx_ser = 0; @@ -711,10 +710,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * For example if three serializers are enabled the DMA * need to transfer three words per DMA request. */ - dma_params->fifo_level = active_serializers; dma_data->maxburst = active_serializers; } else { - dma_params->fifo_level = 0; dma_data->maxburst = 0; } return 0; @@ -746,7 +743,6 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, /* Configure the burst size for platform drivers */ if (numevt == 1) numevt = 0; - dma_params->fifo_level = numevt; dma_data->maxburst = numevt; return 0; @@ -872,8 +868,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); - struct davinci_pcm_dma_params *dma_params = - &mcasp->dma_params[substream->stream]; int word_length; int channels = params_channels(params); int period_size = params_period_size(params); @@ -914,31 +908,26 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; word_length = 8; break; case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; word_length = 16; break; case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: - dma_params->data_type = 3; word_length = 24; break; case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: - dma_params->data_type = 4; word_length = 24; break; case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; word_length = 32; break; @@ -947,11 +936,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (mcasp->version == MCASP_VERSION_2 && !dma_params->fifo_level) - dma_params->acnt = 4; - else - dma_params->acnt = dma_params->data_type; - davinci_config_channel_size(mcasp, word_length); if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) @@ -1055,17 +1039,8 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (mcasp->version >= MCASP_VERSION_3) { - /* Using dmaengine PCM */ - dai->playback_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - } else { - /* Using davinci-pcm */ - dai->playback_dma_data = mcasp->dma_params; - dai->capture_dma_data = mcasp->dma_params; - } + dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; return 0; } @@ -1184,28 +1159,24 @@ static const struct snd_soc_component_driver davinci_mcasp_component = { static struct davinci_mcasp_pdata dm646x_mcasp_pdata = { .tx_dma_offset = 0x400, .rx_dma_offset = 0x400, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_1, }; static struct davinci_mcasp_pdata da830_mcasp_pdata = { .tx_dma_offset = 0x2000, .rx_dma_offset = 0x2000, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, }; static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { .tx_dma_offset = 0, .rx_dma_offset = 0, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_3, }; static struct davinci_mcasp_pdata dra7_mcasp_pdata = { .tx_dma_offset = 0x200, .rx_dma_offset = 0x284, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_4, }; @@ -1382,12 +1353,12 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( static int davinci_mcasp_probe(struct platform_device *pdev) { - struct davinci_pcm_dma_params *dma_params; struct snd_dmaengine_dai_dma_data *dma_data; struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; char *irq_name; + int *dma; int irq; int ret; @@ -1521,59 +1492,45 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) mcasp->dat_port = true; - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_playback; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->tx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->tx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->tx_dma_channel; + *dma = pdata->tx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "tx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; /* RX is not valid in DIT mode */ if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_capture; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->rx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->rx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->rx_dma_channel; + *dma = pdata->rx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "rx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; } if (mcasp->version < MCASP_VERSION_3) { @@ -1596,17 +1553,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; switch (mcasp->version) { -#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_DAVINCI_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - ret = davinci_soc_platform_register(&pdev->dev); - break; -#endif #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) + case MCASP_VERSION_1: + case MCASP_VERSION_2: case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); break; From 4da4608c91308d0d15dd022074724446c15710dc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:21 +0200 Subject: [PATCH 134/411] ASoC: davinci: Remove unused davinci-pcm platform driver All DAI drivers has been converted to use edma-pcm instead of davinci-pcm and the driver can be removed from the tree. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-pcm.c | 861 -------------------------------- sound/soc/davinci/davinci-pcm.h | 41 -- 3 files changed, 904 deletions(-) delete mode 100644 sound/soc/davinci/davinci-pcm.c delete mode 100644 sound/soc/davinci/davinci-pcm.h diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 09bf2ba92d38e3..f883933c1a194d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,11 +1,9 @@ # DAVINCI Platform Support -snd-soc-davinci-objs := davinci-pcm.o snd-soc-edma-objs := edma-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o -obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c deleted file mode 100644 index 7809e9d935fc9c..00000000000000 --- a/sound/soc/davinci/davinci-pcm.c +++ /dev/null @@ -1,861 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, - * Copyright: (C) 2007 MontaVista Software, Inc., - * added SRAM ping/pong (C) 2008 Troy Kisky - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include - -#include "davinci-pcm.h" - -#ifdef DEBUG -static void print_buf_info(int slot, char *name) -{ - struct edmacc_param p; - if (slot < 0) - return; - edma_read_slot(slot, &p); - printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", - name, slot, p.opt, p.src, p.a_b_cnt, p.dst); - printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", - p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); -} -#else -static void print_buf_info(int slot, char *name) -{ -} -#endif - -static struct snd_pcm_hardware pcm_hardware_playback = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| - SNDRV_PCM_INFO_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware pcm_hardware_capture = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -/* - * How ping/pong works.... - * - * Playback: - * ram_params - copys 2*ping_size from start of SDRAM to iram, - * links to ram_link2 - * ram_link2 - copys rest of SDRAM to iram in ping_size units, - * links to ram_link - * ram_link - copys entire SDRAM to iram in ping_size uints, - * links to self - * - * asp_params - same as asp_link[0] - * asp_link[0] - copys from lower half of iram to asp port - * links to asp_link[1], triggers iram copy event on completion - * asp_link[1] - copys from upper half of iram to asp port - * links to asp_link[0], triggers iram copy event on completion - * triggers interrupt only needed to let upper SOC levels update position - * in stream on completion - * - * When playback is started: - * ram_params started - * asp_params started - * - * Capture: - * ram_params - same as ram_link, - * links to ram_link - * ram_link - same as playback - * links to self - * - * asp_params - same as playback - * asp_link[0] - same as playback - * asp_link[1] - same as playback - * - * When capture is started: - * asp_params started - */ -struct davinci_runtime_data { - spinlock_t lock; - int period; /* current DMA period */ - int asp_channel; /* Master DMA channel */ - int asp_link[2]; /* asp parameter link channel, ping/pong */ - struct davinci_pcm_dma_params *params; /* DMA params */ - int ram_channel; - int ram_link; - int ram_link2; - struct edmacc_param asp_params; - struct edmacc_param ram_params; -}; - -static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - prtd->period++; - if (unlikely(prtd->period >= runtime->periods)) - prtd->period = 0; -} - -static void davinci_pcm_period_reset(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - prtd->period = 0; -} -/* - * Not used with ping/pong - */ -static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int period_size; - unsigned int dma_offset; - dma_addr_t dma_pos; - dma_addr_t src, dst; - unsigned short src_bidx, dst_bidx; - unsigned short src_cidx, dst_cidx; - unsigned int data_type; - unsigned short acnt; - unsigned int count; - unsigned int fifo_level; - - period_size = snd_pcm_lib_period_bytes(substream); - dma_offset = prtd->period * period_size; - dma_pos = runtime->dma_addr + dma_offset; - fifo_level = prtd->params->fifo_level; - - pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, - period_size); - - data_type = prtd->params->data_type; - count = period_size / data_type; - if (fifo_level) - count /= fifo_level; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = dma_pos; - dst = prtd->params->dma_addr; - src_bidx = data_type; - dst_bidx = 4; - src_cidx = data_type * fifo_level; - dst_cidx = 0; - } else { - src = prtd->params->dma_addr; - dst = dma_pos; - src_bidx = 0; - dst_bidx = data_type; - src_cidx = 0; - dst_cidx = data_type * fifo_level; - } - - acnt = prtd->params->acnt; - edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); - edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); - edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); - - if (!fifo_level) - edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, - ASYNC); - else - edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level, - count, fifo_level, - ABSYNC); -} - -static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) -{ - struct snd_pcm_substream *substream = data; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - print_buf_info(prtd->ram_channel, "i ram_channel"); - pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); - - if (unlikely(ch_status != EDMA_DMA_COMPLETE)) - return; - - if (snd_pcm_running(substream)) { - spin_lock(&prtd->lock); - if (prtd->ram_channel < 0) { - /* No ping/pong must fix up link dma data*/ - davinci_pcm_enqueue_dma(substream); - } - davinci_pcm_period_elapsed(substream); - spin_unlock(&prtd->lock); - snd_pcm_period_elapsed(substream); - } -} - -#ifdef CONFIG_GENERIC_ALLOCATOR -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - struct snd_dma_buffer *buf = &substream->dma_buffer; - struct snd_dma_buffer *iram_dma = NULL; - dma_addr_t iram_phys = 0; - void *iram_virt = NULL; - - if (buf->private_data || !size) - return 0; - - ppcm->period_bytes_max = size; - iram_virt = gen_pool_dma_alloc(sram_pool, size, &iram_phys); - if (!iram_virt) - goto exit1; - iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); - if (!iram_dma) - goto exit2; - iram_dma->area = iram_virt; - iram_dma->addr = iram_phys; - memset(iram_dma->area, 0, size); - iram_dma->bytes = size; - buf->private_data = iram_dma; - return 0; -exit2: - if (iram_virt) - gen_pool_free(sram_pool, (unsigned)iram_virt, size); -exit1: - return -ENOMEM; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct gen_pool *sram_pool = prtd->params->sram_pool; - - gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); -} -#else -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - return 0; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ -} -#endif - -/* - * Only used with ping/pong. - * This is called after runtime->dma_addr, period_bytes and data_type are valid - */ -static int ping_pong_dma_setup(struct snd_pcm_substream *substream) -{ - unsigned short ram_src_cidx, ram_dst_cidx; - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - struct snd_dma_buffer *iram_dma = - (struct snd_dma_buffer *)substream->dma_buffer.private_data; - struct davinci_pcm_dma_params *params = prtd->params; - unsigned int data_type = params->data_type; - unsigned int acnt = params->acnt; - /* divide by 2 for ping/pong */ - unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - unsigned int fifo_level = prtd->params->fifo_level; - unsigned int count; - if ((data_type == 0) || (data_type > 4)) { - printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); - return -EINVAL; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_addr_t asp_src_pong = iram_dma->addr + ping_size; - ram_src_cidx = ping_size; - ram_dst_cidx = -ping_size; - edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_src_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } else { - dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; - ram_src_cidx = -ping_size; - ram_dst_cidx = ping_size; - edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - - edma_set_dest_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_dest_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } - - if (!fifo_level) { - count = ping_size / data_type; - edma_set_transfer_params(prtd->asp_link[0], acnt, count, - 1, 0, ASYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, count, - 1, 0, ASYNC); - } else { - count = ping_size / (data_type * fifo_level); - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, ABSYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, - count, fifo_level, ABSYNC); - } - - edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); - edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); - edma_set_transfer_params(prtd->ram_link, ping_size, 2, - runtime->periods, 2, ASYNC); - - /* init master params */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_read_slot(prtd->ram_link, &prtd->ram_params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - struct edmacc_param p_ram; - /* Copy entire iram buffer before playback started */ - prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); - /* 0 dst_bidx */ - prtd->ram_params.src_dst_bidx = (ping_size << 1); - /* 0 dst_cidx */ - prtd->ram_params.src_dst_cidx = (ping_size << 1); - prtd->ram_params.ccnt = 1; - - /* Skip 1st period */ - edma_read_slot(prtd->ram_link, &p_ram); - p_ram.src += (ping_size << 1); - p_ram.ccnt -= 1; - edma_write_slot(prtd->ram_link2, &p_ram); - /* - * When 1st started, ram -> iram dma channel will fill the - * entire iram. Then, whenever a ping/pong asp buffer finishes, - * 1/2 iram will be filled. - */ - prtd->ram_params.link_bcntrld = - EDMA_CHAN_SLOT(prtd->ram_link2) << 5; - } - return 0; -} - -/* 1 asp tx or rx channel using 2 parameter channels - * 1 ram to/from iram channel using 1 parameter channel - * - * Playback - * ram copy channel kicks off first, - * 1st ram copy of entire iram buffer completion kicks off asp channel - * asp tcc always kicks off ram copy of 1/2 iram buffer - * - * Record - * asp channel starts, tcc kicks off ram copy - */ -static int request_ping_pong(struct snd_pcm_substream *substream, - struct davinci_runtime_data *prtd, - struct snd_dma_buffer *iram_dma) -{ - dma_addr_t asp_src_ping; - dma_addr_t asp_dst_ping; - int ret; - struct davinci_pcm_dma_params *params = prtd->params; - - /* Request ram master channel */ - ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, - davinci_pcm_dma_irq, substream, - prtd->params->ram_chan_q); - if (ret < 0) - goto exit1; - - /* Request ram link channel */ - ret = prtd->ram_link = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - ret = prtd->asp_link[1] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit3; - - prtd->ram_link2 = -1; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = prtd->ram_link2 = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit4; - } - /* circle ping-pong buffers */ - edma_link(prtd->asp_link[0], prtd->asp_link[1]); - edma_link(prtd->asp_link[1], prtd->asp_link[0]); - /* circle ram buffers */ - edma_link(prtd->ram_link, prtd->ram_link); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - asp_src_ping = iram_dma->addr; - asp_dst_ping = params->dma_addr; /* fifo */ - } else { - asp_src_ping = params->dma_addr; /* fifo */ - asp_dst_ping = iram_dma->addr; - } - /* ping */ - edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[0], 0, 0); - edma_set_dest_index(prtd->asp_link[0], 0, 0); - - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); - prtd->asp_params.opt |= TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - - /* pong */ - edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[1], 0, 0); - edma_set_dest_index(prtd->asp_link[1], 0, 0); - - edma_read_slot(prtd->asp_link[1], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); - /* interrupt after every pong completion */ - prtd->asp_params.opt |= TCINTEN | TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[1], &prtd->asp_params); - - /* ram */ - edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," - "for asp:%u %u %u\n", __func__, - prtd->ram_channel, prtd->ram_link, prtd->ram_link2, - prtd->asp_channel, prtd->asp_link[0], - prtd->asp_link[1]); - return 0; -exit4: - edma_free_channel(prtd->asp_link[1]); - prtd->asp_link[1] = -1; -exit3: - edma_free_channel(prtd->ram_link); - prtd->ram_link = -1; -exit2: - edma_free_channel(prtd->ram_channel); - prtd->ram_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) -{ - struct snd_dma_buffer *iram_dma; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct davinci_pcm_dma_params *params = prtd->params; - int ret; - - if (!params) - return -ENODEV; - - /* Request asp master DMA channel */ - ret = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, - prtd->params->asp_chan_q); - if (ret < 0) - goto exit1; - - /* Request asp link channels */ - ret = prtd->asp_link[0] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; - if (iram_dma) { - if (request_ping_pong(substream, prtd, iram_dma) == 0) - return 0; - printk(KERN_WARNING "%s: dma channel allocation failed," - "not using sram\n", __func__); - } - - /* Issue transfer completion IRQ when the channel completes a - * transfer, then always reload from the same slot (by a kind - * of loopback link). The completion IRQ handler will update - * the reload slot with a new buffer. - * - * REVISIT save p_ram here after setting up everything except - * the buffer and its length (ccnt) ... use it as a template - * so davinci_pcm_enqueue_dma() takes less time in IRQ. - */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt |= TCINTEN | - EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - return 0; -exit2: - edma_free_channel(prtd->asp_channel); - prtd->asp_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - edma_start(prtd->asp_channel); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - prtd->ram_channel >= 0) { - /* copy 1st iram buffer */ - edma_start(prtd->ram_channel); - } - break; - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_resume(prtd->asp_channel); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_pause(prtd->asp_channel); - break; - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static int davinci_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - davinci_pcm_period_reset(substream); - if (prtd->ram_channel >= 0) { - int ret = ping_pong_dma_setup(substream); - if (ret < 0) - return ret; - - edma_write_slot(prtd->ram_channel, &prtd->ram_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - - print_buf_info(prtd->ram_channel, "ram_channel"); - print_buf_info(prtd->ram_link, "ram_link"); - print_buf_info(prtd->ram_link2, "ram_link2"); - print_buf_info(prtd->asp_channel, "asp_channel"); - print_buf_info(prtd->asp_link[0], "asp_link[0]"); - print_buf_info(prtd->asp_link[1], "asp_link[1]"); - - /* - * There is a phase offset of 2 periods between the position - * used by dma setup and the position reported in the pointer - * function. - * - * The phase offset, when not using ping-pong buffers, is due to - * the two consecutive calls to davinci_pcm_enqueue_dma() below. - * - * Whereas here, with ping-pong buffers, the phase is due to - * there being an entire buffer transfer complete before the - * first dma completion event triggers davinci_pcm_dma_irq(). - */ - davinci_pcm_period_elapsed(substream); - davinci_pcm_period_elapsed(substream); - - return 0; - } - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - return 0; -} - -static snd_pcm_uframes_t -davinci_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - unsigned int offset; - int asp_count; - unsigned int period_size = snd_pcm_lib_period_bytes(substream); - - /* - * There is a phase offset of 2 periods between the position used by dma - * setup and the position reported in the pointer function. Either +2 in - * the dma setup or -2 here in the pointer function (with wrapping, - * both) accounts for this offset -- choose the latter since it makes - * the first-time setup clearer. - */ - spin_lock(&prtd->lock); - asp_count = prtd->period - 2; - spin_unlock(&prtd->lock); - - if (asp_count < 0) - asp_count += runtime->periods; - asp_count *= period_size; - - offset = bytes_to_frames(runtime, asp_count); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -static int davinci_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd; - struct snd_pcm_hardware *ppcm; - int ret = 0; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa; - struct davinci_pcm_dma_params *params; - - pa = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!pa) - return -ENODEV; - params = &pa[substream->stream]; - - ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, params->sram_pool, params->sram_size, ppcm); - snd_soc_set_runtime_hwparams(substream, ppcm); - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - prtd->params = params; - prtd->asp_channel = -1; - prtd->asp_link[0] = prtd->asp_link[1] = -1; - prtd->ram_channel = -1; - prtd->ram_link = -1; - prtd->ram_link2 = -1; - - runtime->private_data = prtd; - - ret = davinci_pcm_dma_request(substream); - if (ret) { - printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); - kfree(prtd); - } - - return ret; -} - -static int davinci_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - - if (prtd->ram_channel >= 0) - edma_stop(prtd->ram_channel); - if (prtd->asp_channel >= 0) - edma_stop(prtd->asp_channel); - if (prtd->asp_link[0] >= 0) - edma_unlink(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_unlink(prtd->asp_link[1]); - if (prtd->ram_link >= 0) - edma_unlink(prtd->ram_link); - - if (prtd->asp_link[0] >= 0) - edma_free_slot(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_free_slot(prtd->asp_link[1]); - if (prtd->asp_channel >= 0) - edma_free_channel(prtd->asp_channel); - if (prtd->ram_link >= 0) - edma_free_slot(prtd->ram_link); - if (prtd->ram_link2 >= 0) - edma_free_slot(prtd->ram_link2); - if (prtd->ram_channel >= 0) - edma_free_channel(prtd->ram_channel); - - kfree(prtd); - - return 0; -} - -static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); -} - -static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int davinci_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops davinci_pcm_ops = { - .open = davinci_pcm_open, - .close = davinci_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = davinci_pcm_hw_params, - .hw_free = davinci_pcm_hw_free, - .prepare = davinci_pcm_prepare, - .trigger = davinci_pcm_trigger, - .pointer = davinci_pcm_pointer, - .mmap = davinci_pcm_mmap, -}; - -static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, - size_t size) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " - "size=%d\n", (void *) buf->area, (void *) buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static void davinci_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - struct snd_dma_buffer *iram_dma; - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - iram_dma = buf->private_data; - if (iram_dma) { - davinci_free_sram(substream, iram_dma); - kfree(iram_dma); - } - } -} - -static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, - pcm_hardware_playback.buffer_bytes_max); - if (ret) - return ret; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, - pcm_hardware_capture.buffer_bytes_max); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver davinci_soc_platform = { - .ops = &davinci_pcm_ops, - .pcm_new = davinci_pcm_new, - .pcm_free = davinci_pcm_free, -}; - -int davinci_soc_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &davinci_soc_platform); -} -EXPORT_SYMBOL_GPL(davinci_soc_platform_register); - -MODULE_AUTHOR("Vladimir Barinov"); -MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h deleted file mode 100644 index 0fe2346a9aa29e..00000000000000 --- a/sound/soc/davinci/davinci-pcm.h +++ /dev/null @@ -1,41 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, - * Copyright: (C) 2007 MontaVista Software, Inc., - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _DAVINCI_PCM_H -#define _DAVINCI_PCM_H - -#include -#include -#include - -struct davinci_pcm_dma_params { - int channel; /* sync dma channel ID */ - unsigned short acnt; - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned sram_size; - struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */ - enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ - enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ - unsigned char data_type; /* xfer data type */ - unsigned char convert_mono_stereo; - unsigned int fifo_level; -}; - -#if IS_ENABLED(CONFIG_SND_DAVINCI_SOC) -int davinci_soc_platform_register(struct device *dev); -#else -static inline int davinci_soc_platform_register(struct device *dev) -{ - return 0; -} -#endif /* CONFIG_SND_DAVINCI_SOC */ - -#endif From 6742e15cf92a8dc3065843a627952ed518e08267 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 3 Mar 2015 13:28:53 +0200 Subject: [PATCH 135/411] ASoC: omap-pcm: Allow only formats with 1, 2, and 4 byte physical size sDMA support only transfer elements with 1, 2, and 4 byte physical size. Initialize the pcm driver accordingly. Signed-off-by: Jyri Sarha Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f4b05bc23e4bfb..e49ee2383a8841 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -39,7 +39,7 @@ #define pcm_omap1510() 0 #endif -static const struct snd_pcm_hardware omap_pcm_hardware = { +static struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -53,6 +53,24 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { .buffer_bytes_max = 128 * 1024, }; +/* sDMA supports only 1, 2, and 4 byte transfer elements. */ +static void omap_pcm_limit_supported_formats(void) +{ + int i; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + switch (snd_pcm_format_physical_width(i)) { + case 8: + case 16: + case 32: + omap_pcm_hardware.formats |= (1LL << i); + break; + default: + break; + } + } +} + /* this may get called several times by oss emulation */ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -235,6 +253,7 @@ static struct snd_soc_platform_driver omap_soc_platform = { int omap_pcm_platform_register(struct device *dev) { + omap_pcm_limit_supported_formats(); return devm_snd_soc_register_platform(dev, &omap_soc_platform); } EXPORT_SYMBOL_GPL(omap_pcm_platform_register); From 2bf9eba14340a53776a742f2c8a0bfbd9c86d259 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 3 Mar 2015 18:31:29 +0800 Subject: [PATCH 136/411] ASoC: rt5670: Fix the speaker mono output issue We need to set left/right control for the speaker amp to get stereo output on speaker. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0632b7458a5322..592f961b5de581 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2700,6 +2700,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val); + if (val >= 4) + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980); + else + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00); + ret = regmap_register_patch(rt5670->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) From bbed297d373471c8e4c3183bf67472a768576664 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sun, 22 Feb 2015 16:43:21 +0000 Subject: [PATCH 137/411] ASoC: wm8804: Split out bus drivers Simplify dependencies by using new style split out bus interfaces. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 18 ++++- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wm8804-i2c.c | 64 ++++++++++++++++ sound/soc/codecs/wm8804-spi.c | 56 ++++++++++++++ sound/soc/codecs/wm8804.c | 139 ++++------------------------------ sound/soc/codecs/wm8804.h | 7 ++ 6 files changed, 162 insertions(+), 126 deletions(-) create mode 100644 sound/soc/codecs/wm8804-i2c.c create mode 100644 sound/soc/codecs/wm8804-spi.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c18e10923..1d17988df79640 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -141,7 +141,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8782 - select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8804_I2C if I2C + select SND_SOC_WM8804_SPI if SPI_MASTER select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C @@ -744,8 +745,19 @@ config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" - depends on SND_SOC_I2C_AND_SPI + tristate + +config SND_SOC_WM8804_I2C + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" + depends on I2C + select SND_SOC_WM8804 + select REGMAP_I2C + +config SND_SOC_WM8804_SPI + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" + depends on SPI_MASTER + select SND_SOC_WM8804 + select REGMAP_SPI config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666d187a0e..7acb6c174cb48f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -145,6 +145,8 @@ snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o +snd-soc-wm8804-i2c-objs := wm8804-i2c.o +snd-soc-wm8804-spi-objs := wm8804-spi.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o @@ -323,6 +325,8 @@ obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o +obj-$(CONFIG_SND_SOC_WM8804_I2C) += snd-soc-wm8804-i2c.o +obj-$(CONFIG_SND_SOC_WM8804_SPI) += snd-soc-wm8804-spi.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c new file mode 100644 index 00000000000000..5bd4af2b4059c4 --- /dev/null +++ b/sound/soc/codecs/wm8804-i2c.c @@ -0,0 +1,64 @@ +/* + * wm8804-i2c.c -- WM8804 S/PDIF transceiver driver - I2C + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include "wm8804.h" + +static int wm8804_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&i2c->dev, regmap); +} + +static int wm8804_i2c_remove(struct i2c_client *i2c) +{ + wm8804_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id wm8804_i2c_id[] = { + { "wm8804", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct i2c_driver wm8804_i2c_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_i2c_probe, + .remove = wm8804_i2c_remove, + .id_table = wm8804_i2c_id +}; + +module_i2c_driver(wm8804_i2c_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - I2C"); +MODULE_AUTHOR("Charles Keepax "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804-spi.c b/sound/soc/codecs/wm8804-spi.c new file mode 100644 index 00000000000000..287e11e9079428 --- /dev/null +++ b/sound/soc/codecs/wm8804-spi.c @@ -0,0 +1,56 @@ +/* + * wm8804-spi.c -- WM8804 S/PDIF transceiver driver - SPI + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include "wm8804.h" + +static int wm8804_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&spi->dev, regmap); +} + +static int wm8804_spi_remove(struct spi_device *spi) +{ + wm8804_remove(&spi->dev); + return 0; +} + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct spi_driver wm8804_spi_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_spi_probe, + .remove = wm8804_spi_remove +}; + +module_spi_driver(wm8804_spi_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - SPI"); +MODULE_AUTHOR("Charles Keepax "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index b2b0e68f707e32..b5a04fc5060fdd 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -15,10 +15,7 @@ #include #include #include -#include #include -#include -#include #include #include #include @@ -518,7 +515,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8804_remove(struct snd_soc_codec *codec) +static int wm8804_codec_remove(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; int i; @@ -531,7 +528,7 @@ static int wm8804_remove(struct snd_soc_codec *codec) return 0; } -static int wm8804_probe(struct snd_soc_codec *codec) +static int wm8804_codec_probe(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; int i, id1, id2, ret; @@ -649,8 +646,8 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .probe = wm8804_probe, - .remove = wm8804_remove, + .probe = wm8804_codec_probe, + .remove = wm8804_codec_remove, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, @@ -658,13 +655,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .num_controls = ARRAY_SIZE(wm8804_snd_controls), }; -static const struct of_device_id wm8804_of_match[] = { - { .compatible = "wlf,wm8804", }, - { } -}; -MODULE_DEVICE_TABLE(of, wm8804_of_match); - -static const struct regmap_config wm8804_regmap_config = { +const struct regmap_config wm8804_regmap_config = { .reg_bits = 8, .val_bits = 8, @@ -675,128 +666,30 @@ static const struct regmap_config wm8804_regmap_config = { .reg_defaults = wm8804_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8804_reg_defaults), }; +EXPORT_SYMBOL_GPL(wm8804_regmap_config); -#if defined(CONFIG_SPI_MASTER) -static int wm8804_spi_probe(struct spi_device *spi) +int wm8804_probe(struct device *dev, struct regmap *regmap) { struct wm8804_priv *wm8804; - int ret; - wm8804 = devm_kzalloc(&spi->dev, sizeof *wm8804, GFP_KERNEL); + wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL); if (!wm8804) return -ENOMEM; - wm8804->regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; - } - - spi_set_drvdata(spi, wm8804); + dev_set_drvdata(dev, wm8804); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); - - return ret; -} - -static int wm8804_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver wm8804_spi_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_spi_probe, - .remove = wm8804_spi_remove -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -static int wm8804_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct wm8804_priv *wm8804; - int ret; - - wm8804 = devm_kzalloc(&i2c->dev, sizeof *wm8804, GFP_KERNEL); - if (!wm8804) - return -ENOMEM; + wm8804->regmap = regmap; - wm8804->regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; - } - - i2c_set_clientdata(i2c, wm8804); - - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); - return ret; -} - -static int wm8804_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; -} - -static const struct i2c_device_id wm8804_i2c_id[] = { - { "wm8804", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); - -static struct i2c_driver wm8804_i2c_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_i2c_probe, - .remove = wm8804_i2c_remove, - .id_table = wm8804_i2c_id -}; -#endif - -static int __init wm8804_modinit(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&wm8804_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&wm8804_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8804 SPI driver: %d\n", - ret); - } -#endif - return ret; + return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, + &wm8804_dai, 1); } -module_init(wm8804_modinit); +EXPORT_SYMBOL_GPL(wm8804_probe); -static void __exit wm8804_exit(void) +void wm8804_remove(struct device *dev) { -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&wm8804_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8804_spi_driver); -#endif + snd_soc_unregister_codec(dev); } -module_exit(wm8804_exit); +EXPORT_SYMBOL_GPL(wm8804_remove); MODULE_DESCRIPTION("ASoC WM8804 driver"); MODULE_AUTHOR("Dimitris Papastamos "); diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index e72d4f4ba6b154..a39a2563dc6704 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -13,6 +13,8 @@ #ifndef _WM8804_H #define _WM8804_H +#include + /* * Register values. */ @@ -62,4 +64,9 @@ #define WM8804_MCLKDIV_256FS 0 #define WM8804_MCLKDIV_128FS 1 +extern const struct regmap_config wm8804_regmap_config; + +int wm8804_probe(struct device *dev, struct regmap *regmap); +void wm8804_remove(struct device *dev); + #endif /* _WM8804_H */ From 6f2c9348095ae1a489abafe2ab3db7deca406e49 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sun, 22 Feb 2015 16:43:22 +0000 Subject: [PATCH 138/411] ASoC: wm8804: Merge CODEC probe and bus probe All of the things in the CODEC probe, such as getting the regulators and verifying the chip ID, are better done in bus probe. It is better to fail during bus probe if this is the wrong chip and all resource allocation should be done in the bus probe anyway. This patch merges the CODEC probe into bus probe. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 180 +++++++++++++++++--------------------- 1 file changed, 82 insertions(+), 98 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index b5a04fc5060fdd..1bd4ace295948a 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -182,9 +182,9 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct snd_soc_codec *codec) +static int wm8804_reset(struct wm8804_priv *wm8804) { - return snd_soc_write(codec, WM8804_RST_DEVID1, 0x0); + return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } static int wm8804_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -515,100 +515,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8804_codec_remove(struct snd_soc_codec *codec) -{ - struct wm8804_priv *wm8804; - int i; - - wm8804 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) - regulator_unregister_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - return 0; -} - -static int wm8804_codec_probe(struct snd_soc_codec *codec) -{ - struct wm8804_priv *wm8804; - int i, id1, id2, ret; - - wm8804 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) - wm8804->supplies[i].supply = wm8804_supply_names[i]; - - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0; - wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1; - - /* This should really be moved into the regulator core */ - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { - ret = regulator_register_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - if (ret != 0) { - dev_err(codec->dev, - "Failed to register regulator notifier: %d\n", - ret); - } - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - return ret; - } - - id1 = snd_soc_read(codec, WM8804_RST_DEVID1); - if (id1 < 0) { - dev_err(codec->dev, "Failed to read device ID: %d\n", id1); - ret = id1; - goto err_reg_enable; - } - - id2 = snd_soc_read(codec, WM8804_DEVID2); - if (id2 < 0) { - dev_err(codec->dev, "Failed to read device ID: %d\n", id2); - ret = id2; - goto err_reg_enable; - } - - id2 = (id2 << 8) | id1; - - if (id2 != 0x8805) { - dev_err(codec->dev, "Invalid device ID: %#x\n", id2); - ret = -EINVAL; - goto err_reg_enable; - } - - ret = snd_soc_read(codec, WM8804_DEVREV); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_reg_enable; - } - dev_info(codec->dev, "revision %c\n", ret + 'A'); - - ret = wm8804_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; - } - - return 0; - -err_reg_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); - return ret; -} - static const struct snd_soc_dai_ops wm8804_dai_ops = { .hw_params = wm8804_hw_params, .set_fmt = wm8804_set_fmt, @@ -646,8 +552,6 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .probe = wm8804_codec_probe, - .remove = wm8804_codec_remove, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, @@ -671,6 +575,8 @@ EXPORT_SYMBOL_GPL(wm8804_regmap_config); int wm8804_probe(struct device *dev, struct regmap *regmap) { struct wm8804_priv *wm8804; + unsigned int id1, id2; + int i, ret; wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL); if (!wm8804) @@ -680,13 +586,91 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) + wm8804->supplies[i].supply = wm8804_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0; + wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { + ret = regulator_register_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); + if (ret != 0) { + dev_err(dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); + if (ret < 0) { + dev_err(dev, "Failed to read device ID: %d\n", ret); + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_DEVID2, &id2); + if (ret < 0) { + dev_err(dev, "Failed to read device ID: %d\n", ret); + goto err_reg_enable; + } + + id2 = (id2 << 8) | id1; + + if (id2 != 0x8805) { + dev_err(dev, "Invalid device ID: %#x\n", id2); + ret = -EINVAL; + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_DEVREV, &id1); + if (ret < 0) { + dev_err(dev, "Failed to read device revision: %d\n", + ret); + goto err_reg_enable; + } + dev_info(dev, "revision %c\n", id1 + 'A'); + + ret = wm8804_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } + return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, &wm8804_dai, 1); + +err_reg_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); + return ret; } EXPORT_SYMBOL_GPL(wm8804_probe); void wm8804_remove(struct device *dev) { + struct wm8804_priv *wm8804; + int i; + + wm8804 = dev_get_drvdata(dev); + + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) + regulator_unregister_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); + snd_soc_unregister_codec(dev); } EXPORT_SYMBOL_GPL(wm8804_remove); From 6afda7f5075440f6737d953ab00fc82efb6cabae Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 5 Mar 2015 16:55:21 +0200 Subject: [PATCH 139/411] ASoC: davinci-mcasp: Allow complete shutdown of McASP when not in use Rearrange the pm_runtime_get/put_sync calls so the IP will be turned off when it is not in use. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 27 +++++++++++++++++---------- 1 file changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0c882995a3575a..33cea0728cd26e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -62,6 +62,7 @@ struct davinci_mcasp_context { u32 config_regs[ARRAY_SIZE(context_regs)]; u32 afifo_regs[2]; /* for read/write fifo control registers */ u32 *xrsr_regs; /* for serializer configuration */ + bool pm_state; }; struct davinci_mcasp { @@ -519,7 +520,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); } out: - pm_runtime_put_sync(mcasp->dev); + pm_runtime_put(mcasp->dev); return ret; } @@ -528,6 +529,7 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + pm_runtime_get_sync(mcasp->dev); switch (div_id) { case 0: /* MCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, @@ -553,6 +555,7 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, return -EINVAL; } + pm_runtime_put(mcasp->dev); return 0; } @@ -567,6 +570,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + pm_runtime_get_sync(mcasp->dev); if (dir == SND_SOC_CLOCK_OUT) { mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); @@ -579,6 +583,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, mcasp->sysclk_freq = freq; + pm_runtime_put(mcasp->dev); return 0; } @@ -1053,6 +1058,10 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) u32 reg; int i; + context->pm_state = pm_runtime_enabled(mcasp->dev) + if (!context->pm_state) + pm_runtime_get_sync(mcasp->dev); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); @@ -1069,6 +1078,8 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) context->xrsr_regs[i] = mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i)); + pm_runtime_put_sync(mcasp->dev); + return 0; } @@ -1079,6 +1090,8 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) u32 reg; int i; + pm_runtime_get_sync(mcasp->dev); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); @@ -1095,6 +1108,9 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), context->xrsr_regs[i]); + if (!context->pm_state) + pm_runtime_put_sync(mcasp->dev); + return 0; } #else @@ -1398,13 +1414,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - ret = pm_runtime_get_sync(&pdev->dev); - if (IS_ERR_VALUE(ret)) { - dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); - pm_runtime_disable(&pdev->dev); - return ret; - } - mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!mcasp->base) { dev_err(&pdev->dev, "ioremap failed\n"); @@ -1584,14 +1593,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err: - pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); return ret; } static int davinci_mcasp_remove(struct platform_device *pdev) { - pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); return 0; From 0be9653a02830637ed385d99bf898d456e8eae8f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 5 Mar 2015 15:39:21 +0000 Subject: [PATCH 140/411] ASoC: wm8804: Use new devres regulator_register_notifier This is more idiomatic and also fixes an issue where the notifiers were being leaked if probe failed. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 1bd4ace295948a..7804ddf53a04d4 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -601,8 +601,10 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { - ret = regulator_register_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); + struct regulator *regulator = wm8804->supplies[i].consumer; + + ret = devm_regulator_register_notifier(regulator, + &wm8804->disable_nb[i]); if (ret != 0) { dev_err(dev, "Failed to register regulator notifier: %d\n", @@ -662,15 +664,6 @@ EXPORT_SYMBOL_GPL(wm8804_probe); void wm8804_remove(struct device *dev) { - struct wm8804_priv *wm8804; - int i; - - wm8804 = dev_get_drvdata(dev); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) - regulator_unregister_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - snd_soc_unregister_codec(dev); } EXPORT_SYMBOL_GPL(wm8804_remove); From fcf638f9953eb7f3b97fad7e970ae59dcdbd70c1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 5 Mar 2015 15:39:22 +0000 Subject: [PATCH 141/411] ASoC: wm8804: Fix small issues in probe error paths This patch fixes some small issues on the probe error paths. Firstly, fail probe if we can't register the regulator notifiers as this will cause the cache to never be synchronised which will result in odd behaviour if the regulators are controllable. Secondly, we don't need to call regulator_bulk_disable if the enable fails, because the regulator core will handle this clean up for us. Finally, we need to disable the regulators if snd_soc_register_codecs fails. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 7804ddf53a04d4..f44da83f50dc85 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -609,6 +609,7 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) dev_err(dev, "Failed to register regulator notifier: %d\n", ret); + return ret; } } @@ -616,7 +617,7 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->supplies); if (ret) { dev_err(dev, "Failed to enable supplies: %d\n", ret); - goto err_reg_enable; + return ret; } ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); @@ -653,8 +654,14 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) goto err_reg_enable; } - return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, - &wm8804_dai, 1); + ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, + &wm8804_dai, 1); + if (ret < 0) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_reg_enable; + } + + return 0; err_reg_enable: regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); From 45ad3e67f663af3f7988552165e8b8ebddde0ac7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 5 Mar 2015 15:42:51 +0000 Subject: [PATCH 142/411] ASoC: wm8804: Update DT binding document to cover regulator supplies Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/wm8804.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8804.txt b/Documentation/devicetree/bindings/sound/wm8804.txt index 4d3a56f38adc31..10ef07606b84c0 100644 --- a/Documentation/devicetree/bindings/sound/wm8804.txt +++ b/Documentation/devicetree/bindings/sound/wm8804.txt @@ -10,6 +10,9 @@ Required properties: - reg : the I2C address of the device for I2C, the chip select number for SPI. + - PVDD-supply, DVDD-supply : Power supplies for the device, as covered + in Documentation/devicetree/bindings/regulator/regulator.txt + Example: codec: wm8804@1a { From 012baec5c1eaab7ad98b461eb7afae2faf79dbea Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Thu, 5 Mar 2015 03:37:39 +0800 Subject: [PATCH 143/411] ASoC: tegra: fix platform_no_drv_owner.cocci warnings sound/soc/tegra/tegra_rt5677.c:334:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Anatol Pomozov Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5677.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index e4cf978a6e3a9a..4453566405efb0 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -331,7 +331,6 @@ static const struct of_device_id tegra_rt5677_of_match[] = { static struct platform_driver tegra_rt5677_driver = { .driver = { .name = DRV_NAME, - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = tegra_rt5677_of_match, }, From cd59f13823ae65a16b7c99fdd338ebac0c6487f2 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 3 Mar 2015 16:21:53 -0800 Subject: [PATCH 144/411] ASoC: qcom: add LPASS header files Add the LPASS header files for ipq806x SOC. This includes the register definitions for the ipq806x LPAIF, and the structure definition for the driver data. Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-lpaif-ipq806x.h | 172 +++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 51 ++++++++ 2 files changed, 223 insertions(+) create mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass.h diff --git a/sound/soc/qcom/lpass-lpaif-ipq806x.h b/sound/soc/qcom/lpass-lpaif-ipq806x.h new file mode 100644 index 00000000000000..dc423b8888425e --- /dev/null +++ b/sound/soc/qcom/lpass-lpaif-ipq806x.h @@ -0,0 +1,172 @@ +/* + * Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * lpass-lpaif-ipq806x.h -- Definitions for the QTi LPAIF in the ipq806x LPASS + */ + +#ifndef __LPASS_LPAIF_H__ +#define __LPASS_LPAIF_H__ + +#define LPAIF_BANK_OFFSET 0x1000 + +/* LPAIF I2S */ + +#define LPAIF_I2SCTL_REG_BASE 0x0010 +#define LPAIF_I2SCTL_REG_STRIDE 0x4 +#define LPAIF_I2SCTL_REG_ADDR(addr, port) \ + (LPAIF_I2SCTL_REG_BASE + (addr) + (LPAIF_I2SCTL_REG_STRIDE * (port))) + +enum lpaif_i2s_ports { + LPAIF_I2S_PORT_MIN = 0, + + LPAIF_I2S_PORT_CODEC_SPK = 0, + LPAIF_I2S_PORT_CODEC_MIC = 1, + LPAIF_I2S_PORT_SEC_SPK = 2, + LPAIF_I2S_PORT_SEC_MIC = 3, + LPAIF_I2S_PORT_MI2S = 4, + + LPAIF_I2S_PORT_MAX = 4, + LPAIF_I2S_PORT_NUM = 5, +}; + +#define LPAIF_I2SCTL_REG(port) LPAIF_I2SCTL_REG_ADDR(0x0, (port)) + +#define LPAIF_I2SCTL_LOOPBACK_MASK 0x8000 +#define LPAIF_I2SCTL_LOOPBACK_SHIFT 15 +#define LPAIF_I2SCTL_LOOPBACK_DISABLE (0 << LPAIF_I2SCTL_LOOPBACK_SHIFT) +#define LPAIF_I2SCTL_LOOPBACK_ENABLE (1 << LPAIF_I2SCTL_LOOPBACK_SHIFT) + +#define LPAIF_I2SCTL_SPKEN_MASK 0x4000 +#define LPAIF_I2SCTL_SPKEN_SHIFT 14 +#define LPAIF_I2SCTL_SPKEN_DISABLE (0 << LPAIF_I2SCTL_SPKEN_SHIFT) +#define LPAIF_I2SCTL_SPKEN_ENABLE (1 << LPAIF_I2SCTL_SPKEN_SHIFT) + +#define LPAIF_I2SCTL_SPKMODE_MASK 0x3C00 +#define LPAIF_I2SCTL_SPKMODE_SHIFT 10 +#define LPAIF_I2SCTL_SPKMODE_NONE (0 << LPAIF_I2SCTL_SPKMODE_SHIFT) +#define LPAIF_I2SCTL_SPKMODE_SD0 (1 << LPAIF_I2SCTL_SPKMODE_SHIFT) +#define LPAIF_I2SCTL_SPKMODE_SD1 (2 << LPAIF_I2SCTL_SPKMODE_SHIFT) +#define LPAIF_I2SCTL_SPKMODE_SD2 (3 << LPAIF_I2SCTL_SPKMODE_SHIFT) +#define LPAIF_I2SCTL_SPKMODE_SD3 (4 << LPAIF_I2SCTL_SPKMODE_SHIFT) +#define LPAIF_I2SCTL_SPKMODE_QUAD01 (5 << LPAIF_I2SCTL_SPKMODE_SHIFT) +#define LPAIF_I2SCTL_SPKMODE_QUAD23 (6 << LPAIF_I2SCTL_SPKMODE_SHIFT) +#define LPAIF_I2SCTL_SPKMODE_6CH (7 << LPAIF_I2SCTL_SPKMODE_SHIFT) +#define LPAIF_I2SCTL_SPKMODE_8CH (8 << LPAIF_I2SCTL_SPKMODE_SHIFT) + +#define LPAIF_I2SCTL_SPKMONO_MASK 0x0200 +#define LPAIF_I2SCTL_SPKMONO_SHIFT 9 +#define LPAIF_I2SCTL_SPKMONO_STEREO (0 << LPAIF_I2SCTL_SPKMONO_SHIFT) +#define LPAIF_I2SCTL_SPKMONO_MONO (1 << LPAIF_I2SCTL_SPKMONO_SHIFT) + +#define LPAIF_I2SCTL_WSSRC_MASK 0x0004 +#define LPAIF_I2SCTL_WSSRC_SHIFT 2 +#define LPAIF_I2SCTL_WSSRC_INTERNAL (0 << LPAIF_I2SCTL_WSSRC_SHIFT) +#define LPAIF_I2SCTL_WSSRC_EXTERNAL (1 << LPAIF_I2SCTL_WSSRC_SHIFT) + +#define LPAIF_I2SCTL_BITWIDTH_MASK 0x0003 +#define LPAIF_I2SCTL_BITWIDTH_SHIFT 0 +#define LPAIF_I2SCTL_BITWIDTH_16 (0 << LPAIF_I2SCTL_BITWIDTH_SHIFT) +#define LPAIF_I2SCTL_BITWIDTH_24 (1 << LPAIF_I2SCTL_BITWIDTH_SHIFT) +#define LPAIF_I2SCTL_BITWIDTH_32 (2 << LPAIF_I2SCTL_BITWIDTH_SHIFT) + +/* LPAIF IRQ */ + +#define LPAIF_IRQ_REG_BASE 0x3000 +#define LPAIF_IRQ_REG_STRIDE 0x1000 +#define LPAIF_IRQ_REG_ADDR(addr, port) \ + (LPAIF_IRQ_REG_BASE + (addr) + (LPAIF_IRQ_REG_STRIDE * (port))) + +enum lpaif_irq_ports { + LPAIF_IRQ_PORT_MIN = 0, + + LPAIF_IRQ_PORT_HOST = 0, + LPAIF_IRQ_PORT_ADSP = 1, + + LPAIF_IRQ_PORT_MAX = 2, + LPAIF_IRQ_PORT_NUM = 3, +}; + +#define LPAIF_IRQEN_REG(port) LPAIF_IRQ_REG_ADDR(0x0, (port)) +#define LPAIF_IRQSTAT_REG(port) LPAIF_IRQ_REG_ADDR(0x4, (port)) +#define LPAIF_IRQCLEAR_REG(port) LPAIF_IRQ_REG_ADDR(0xC, (port)) + +#define LPAIF_IRQ_BITSTRIDE 3 +#define LPAIF_IRQ_PER(chan) (1 << (LPAIF_IRQ_BITSTRIDE * (chan))) +#define LPAIF_IRQ_XRUN(chan) (2 << (LPAIF_IRQ_BITSTRIDE * (chan))) +#define LPAIF_IRQ_ERR(chan) (4 << (LPAIF_IRQ_BITSTRIDE * (chan))) +#define LPAIF_IRQ_ALL(chan) (7 << (LPAIF_IRQ_BITSTRIDE * (chan))) + +/* LPAIF DMA */ + +#define LPAIF_RDMA_REG_BASE 0x6000 +#define LPAIF_RDMA_REG_STRIDE 0x1000 +#define LPAIF_RDMA_REG_ADDR(addr, chan) \ + (LPAIF_RDMA_REG_BASE + (addr) + (LPAIF_RDMA_REG_STRIDE * (chan))) + +enum lpaif_dma_channels { + LPAIF_RDMA_CHAN_MIN = 0, + + LPAIF_RDMA_CHAN_MI2S = 0, + LPAIF_RDMA_CHAN_PCM0 = 1, + LPAIF_RDMA_CHAN_PCM1 = 2, + + LPAIF_RDMA_CHAN_MAX = 4, + LPAIF_RDMA_CHAN_NUM = 5, +}; + +#define LPAIF_RDMACTL_REG(chan) LPAIF_RDMA_REG_ADDR(0x00, (chan)) +#define LPAIF_RDMABASE_REG(chan) LPAIF_RDMA_REG_ADDR(0x04, (chan)) +#define LPAIF_RDMABUFF_REG(chan) LPAIF_RDMA_REG_ADDR(0x08, (chan)) +#define LPAIF_RDMACURR_REG(chan) LPAIF_RDMA_REG_ADDR(0x0C, (chan)) +#define LPAIF_RDMAPER_REG(chan) LPAIF_RDMA_REG_ADDR(0x10, (chan)) + +#define LPAIF_RDMACTL_BURSTEN_MASK 0x800 +#define LPAIF_RDMACTL_BURSTEN_SHIFT 11 +#define LPAIF_RDMACTL_BURSTEN_SINGLE (0 << LPAIF_RDMACTL_BURSTEN_SHIFT) +#define LPAIF_RDMACTL_BURSTEN_INCR4 (1 << LPAIF_RDMACTL_BURSTEN_SHIFT) + +#define LPAIF_RDMACTL_WPSCNT_MASK 0x700 +#define LPAIF_RDMACTL_WPSCNT_SHIFT 8 +#define LPAIF_RDMACTL_WPSCNT_ONE (0 << LPAIF_RDMACTL_WPSCNT_SHIFT) +#define LPAIF_RDMACTL_WPSCNT_TWO (1 << LPAIF_RDMACTL_WPSCNT_SHIFT) +#define LPAIF_RDMACTL_WPSCNT_THREE (2 << LPAIF_RDMACTL_WPSCNT_SHIFT) +#define LPAIF_RDMACTL_WPSCNT_FOUR (3 << LPAIF_RDMACTL_WPSCNT_SHIFT) +#define LPAIF_RDMACTL_WPSCNT_SIX (5 << LPAIF_RDMACTL_WPSCNT_SHIFT) +#define LPAIF_RDMACTL_WPSCNT_EIGHT (7 << LPAIF_RDMACTL_WPSCNT_SHIFT) + +#define LPAIF_RDMACTL_AUDINTF_MASK 0x0F0 +#define LPAIF_RDMACTL_AUDINTF_SHIFT 4 +#define LPAIF_RDMACTL_AUDINTF_NONE (0 << LPAIF_RDMACTL_AUDINTF_SHIFT) +#define LPAIF_RDMACTL_AUDINTF_CODEC (1 << LPAIF_RDMACTL_AUDINTF_SHIFT) +#define LPAIF_RDMACTL_AUDINTF_PCM (2 << LPAIF_RDMACTL_AUDINTF_SHIFT) +#define LPAIF_RDMACTL_AUDINTF_SEC_I2S (3 << LPAIF_RDMACTL_AUDINTF_SHIFT) +#define LPAIF_RDMACTL_AUDINTF_MI2S (4 << LPAIF_RDMACTL_AUDINTF_SHIFT) +#define LPAIF_RDMACTL_AUDINTF_HDMI (5 << LPAIF_RDMACTL_AUDINTF_SHIFT) +#define LPAIF_RDMACTL_AUDINTF_SEC_PCM (7 << LPAIF_RDMACTL_AUDINTF_SHIFT) + +#define LPAIF_RDMACTL_FIFOWM_MASK 0x00E +#define LPAIF_RDMACTL_FIFOWM_SHIFT 1 +#define LPAIF_RDMACTL_FIFOWM_1 (0 << LPAIF_RDMACTL_FIFOWM_SHIFT) +#define LPAIF_RDMACTL_FIFOWM_2 (1 << LPAIF_RDMACTL_FIFOWM_SHIFT) +#define LPAIF_RDMACTL_FIFOWM_3 (2 << LPAIF_RDMACTL_FIFOWM_SHIFT) +#define LPAIF_RDMACTL_FIFOWM_4 (3 << LPAIF_RDMACTL_FIFOWM_SHIFT) +#define LPAIF_RDMACTL_FIFOWM_5 (4 << LPAIF_RDMACTL_FIFOWM_SHIFT) +#define LPAIF_RDMACTL_FIFOWM_6 (5 << LPAIF_RDMACTL_FIFOWM_SHIFT) +#define LPAIF_RDMACTL_FIFOWM_7 (6 << LPAIF_RDMACTL_FIFOWM_SHIFT) +#define LPAIF_RDMACTL_FIFOWM_8 (7 << LPAIF_RDMACTL_FIFOWM_SHIFT) + +#define LPAIF_RDMACTL_ENABLE_MASK 0x1 +#define LPAIF_RDMACTL_ENABLE_SHIFT 0 +#define LPAIF_RDMACTL_ENABLE_OFF (0 << LPAIF_RDMACTL_ENABLE_SHIFT) +#define LPAIF_RDMACTL_ENABLE_ON (1 << LPAIF_RDMACTL_ENABLE_SHIFT) + +#endif /* __LPASS_LPAIF_H__ */ diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h new file mode 100644 index 00000000000000..5c99b3dace86d3 --- /dev/null +++ b/sound/soc/qcom/lpass.h @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * lpass.h - Definitions for the QTi LPASS + */ + +#ifndef __LPASS_H__ +#define __LPASS_H__ + +#include +#include +#include +#include + +#define LPASS_AHBIX_CLOCK_FREQUENCY 131072000 + +/* Both the CPU DAI and platform drivers will access this data */ +struct lpass_data { + + /* AHB-I/X bus clocks inside the low-power audio subsystem (LPASS) */ + struct clk *ahbix_clk; + + /* MI2S system clock */ + struct clk *mi2s_osr_clk; + + /* MI2S bit clock (derived from system clock by a divider */ + struct clk *mi2s_bit_clk; + + /* low-power audio interface (LPAIF) registers */ + void __iomem *lpaif; + + /* regmap backed by the low-power audio interface (LPAIF) registers */ + struct regmap *lpaif_map; + + /* interrupts from the low-power audio interface (LPAIF) */ + int lpaif_irq; +}; + +/* register the platform driver from the CPU DAI driver */ +int asoc_qcom_lpass_platform_register(struct platform_device *); + +#endif /* __LPASS_H__ */ From 80beab8e1d86d7da843e6c3e439bbca5320c568d Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 3 Mar 2015 16:21:54 -0800 Subject: [PATCH 145/411] ASoC: qcom: Add LPASS CPU DAI driver Add the CPU DAI driver for the Qualcomm Technologies low-power audio subsystem (LPASS). Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 510 +++++++++++++++++++++++++++++++++++++ 1 file changed, 510 insertions(+) create mode 100644 sound/soc/qcom/lpass-cpu.c diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c new file mode 100644 index 00000000000000..d5167131787f3a --- /dev/null +++ b/sound/soc/qcom/lpass-cpu.c @@ -0,0 +1,510 @@ +/* + * Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * lpass-cpu.c -- ALSA SoC CPU DAI driver for QTi LPASS + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "lpass-lpaif-ipq806x.h" +#include "lpass.h" + +static int lpass_cpu_daiops_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_set_rate(drvdata->mi2s_osr_clk, freq); + if (ret) + dev_err(dai->dev, "%s() error setting mi2s osrclk to %u: %d\n", + __func__, freq, ret); + + return ret; +} + +static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(drvdata->mi2s_osr_clk); + if (ret) { + dev_err(dai->dev, "%s() error in enabling mi2s osr clk: %d\n", + __func__, ret); + return ret; + } + + ret = clk_prepare_enable(drvdata->mi2s_bit_clk); + if (ret) { + dev_err(dai->dev, "%s() error in enabling mi2s bit clk: %d\n", + __func__, ret); + clk_disable_unprepare(drvdata->mi2s_osr_clk); + return ret; + } + + return 0; +} + +static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(drvdata->mi2s_bit_clk); + clk_disable_unprepare(drvdata->mi2s_osr_clk); +} + +static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + snd_pcm_format_t format = params_format(params); + unsigned int channels = params_channels(params); + unsigned int rate = params_rate(params); + unsigned int regval; + int bitwidth, ret; + + bitwidth = snd_pcm_format_width(format); + if (bitwidth < 0) { + dev_err(dai->dev, "%s() invalid bit width given: %d\n", + __func__, bitwidth); + return bitwidth; + } + + regval = LPAIF_I2SCTL_LOOPBACK_DISABLE | + LPAIF_I2SCTL_WSSRC_INTERNAL; + + switch (bitwidth) { + case 16: + regval |= LPAIF_I2SCTL_BITWIDTH_16; + break; + case 24: + regval |= LPAIF_I2SCTL_BITWIDTH_24; + break; + case 32: + regval |= LPAIF_I2SCTL_BITWIDTH_32; + break; + default: + dev_err(dai->dev, "%s() invalid bitwidth given: %d\n", + __func__, bitwidth); + return -EINVAL; + } + + switch (channels) { + case 1: + regval |= LPAIF_I2SCTL_SPKMODE_SD0; + regval |= LPAIF_I2SCTL_SPKMONO_MONO; + break; + case 2: + regval |= LPAIF_I2SCTL_SPKMODE_SD0; + regval |= LPAIF_I2SCTL_SPKMONO_STEREO; + break; + case 4: + regval |= LPAIF_I2SCTL_SPKMODE_QUAD01; + regval |= LPAIF_I2SCTL_SPKMONO_STEREO; + break; + case 6: + regval |= LPAIF_I2SCTL_SPKMODE_6CH; + regval |= LPAIF_I2SCTL_SPKMONO_STEREO; + break; + case 8: + regval |= LPAIF_I2SCTL_SPKMODE_8CH; + regval |= LPAIF_I2SCTL_SPKMONO_STEREO; + break; + default: + dev_err(dai->dev, "%s() invalid channels given: %u\n", + __func__, channels); + return -EINVAL; + } + + ret = regmap_write(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(LPAIF_I2S_PORT_MI2S), regval); + if (ret) { + dev_err(dai->dev, "%s() error writing to i2sctl reg: %d\n", + __func__, ret); + return ret; + } + + ret = clk_set_rate(drvdata->mi2s_bit_clk, rate * bitwidth * 2); + if (ret) { + dev_err(dai->dev, "%s() error setting mi2s bitclk to %u: %d\n", + __func__, rate * bitwidth * 2, ret); + return ret; + } + + return 0; +} + +static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = regmap_write(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(LPAIF_I2S_PORT_MI2S), 0); + if (ret) + dev_err(dai->dev, "%s() error writing to i2sctl reg: %d\n", + __func__, ret); + + return ret; +} + +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(LPAIF_I2S_PORT_MI2S), + LPAIF_I2SCTL_SPKEN_MASK, LPAIF_I2SCTL_SPKEN_ENABLE); + if (ret) + dev_err(dai->dev, "%s() error writing to i2sctl reg: %d\n", + __func__, ret); + + return ret; +} + +static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(LPAIF_I2S_PORT_MI2S), + LPAIF_I2SCTL_SPKEN_MASK, + LPAIF_I2SCTL_SPKEN_ENABLE); + if (ret) + dev_err(dai->dev, "%s() error writing to i2sctl reg: %d\n", + __func__, ret); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(LPAIF_I2S_PORT_MI2S), + LPAIF_I2SCTL_SPKEN_MASK, + LPAIF_I2SCTL_SPKEN_DISABLE); + if (ret) + dev_err(dai->dev, "%s() error writing to i2sctl reg: %d\n", + __func__, ret); + break; + } + + return ret; +} + +static struct snd_soc_dai_ops lpass_cpu_dai_ops = { + .set_sysclk = lpass_cpu_daiops_set_sysclk, + .startup = lpass_cpu_daiops_startup, + .shutdown = lpass_cpu_daiops_shutdown, + .hw_params = lpass_cpu_daiops_hw_params, + .hw_free = lpass_cpu_daiops_hw_free, + .prepare = lpass_cpu_daiops_prepare, + .trigger = lpass_cpu_daiops_trigger, +}; + +static int lpass_cpu_dai_probe(struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + int ret; + + /* ensure audio hardware is disabled */ + ret = regmap_write(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(LPAIF_I2S_PORT_MI2S), 0); + if (ret) + dev_err(dai->dev, "%s() error writing to i2sctl reg: %d\n", + __func__, ret); + + return ret; +} + +static struct snd_soc_dai_driver lpass_cpu_dai_driver = { + .playback = { + .stream_name = "lpass-cpu-playback", + .formats = SNDRV_PCM_FMTBIT_S16 | + SNDRV_PCM_FMTBIT_S24 | + SNDRV_PCM_FMTBIT_S32, + .rates = SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 8, + }, + .probe = &lpass_cpu_dai_probe, + .ops = &lpass_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver lpass_cpu_comp_driver = { + .name = "lpass-cpu", +}; + +static bool lpass_cpu_regmap_writeable(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < LPAIF_I2S_PORT_NUM; ++i) + if (reg == LPAIF_I2SCTL_REG(i)) + return true; + + for (i = 0; i < LPAIF_IRQ_PORT_NUM; ++i) { + if (reg == LPAIF_IRQEN_REG(i)) + return true; + if (reg == LPAIF_IRQCLEAR_REG(i)) + return true; + } + + for (i = 0; i < LPAIF_RDMA_CHAN_NUM; ++i) { + if (reg == LPAIF_RDMACTL_REG(i)) + return true; + if (reg == LPAIF_RDMABASE_REG(i)) + return true; + if (reg == LPAIF_RDMABUFF_REG(i)) + return true; + if (reg == LPAIF_RDMAPER_REG(i)) + return true; + } + + return false; +} + +static bool lpass_cpu_regmap_readable(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < LPAIF_I2S_PORT_NUM; ++i) + if (reg == LPAIF_I2SCTL_REG(i)) + return true; + + for (i = 0; i < LPAIF_IRQ_PORT_NUM; ++i) { + if (reg == LPAIF_IRQEN_REG(i)) + return true; + if (reg == LPAIF_IRQSTAT_REG(i)) + return true; + } + + for (i = 0; i < LPAIF_RDMA_CHAN_NUM; ++i) { + if (reg == LPAIF_RDMACTL_REG(i)) + return true; + if (reg == LPAIF_RDMABASE_REG(i)) + return true; + if (reg == LPAIF_RDMABUFF_REG(i)) + return true; + if (reg == LPAIF_RDMACURR_REG(i)) + return true; + if (reg == LPAIF_RDMAPER_REG(i)) + return true; + } + + return false; +} + +static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < LPAIF_IRQ_PORT_NUM; ++i) + if (reg == LPAIF_IRQSTAT_REG(i)) + return true; + + for (i = 0; i < LPAIF_RDMA_CHAN_NUM; ++i) + if (reg == LPAIF_RDMACURR_REG(i)) + return true; + + return false; +} + +static const struct regmap_config lpass_cpu_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = LPAIF_RDMAPER_REG(LPAIF_RDMA_CHAN_MAX), + .writeable_reg = lpass_cpu_regmap_writeable, + .readable_reg = lpass_cpu_regmap_readable, + .volatile_reg = lpass_cpu_regmap_volatile, + .cache_type = REGCACHE_FLAT, +}; + +static int lpass_cpu_parse_of(struct device *dev) +{ + struct device_node *dsp_of_node; + + dsp_of_node = of_get_child_by_name(dev->of_node, "qcom,adsp"); + if (!dsp_of_node) { + dev_err(dev, "%s() error getting qcom,adsp sub-node\n", + __func__); + return -EINVAL; + } + + if (of_device_is_available(dsp_of_node)) { + dev_err(dev, "%s() DSP exists and holds audio resources\n", + __func__); + return -EBUSY; + } + + return 0; +} + +static int lpass_cpu_platform_probe(struct platform_device *pdev) +{ + struct lpass_data *drvdata; + struct resource *res; + int ret; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(struct lpass_data), + GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + platform_set_drvdata(pdev, drvdata); + + ret = lpass_cpu_parse_of(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "%s() error getting DT node info: %d\n", + __func__, ret); + return ret; + } + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "lpass-lpaif"); + if (!res) { + dev_err(&pdev->dev, "%s() error getting resource\n", __func__); + return -ENODEV; + } + + drvdata->lpaif = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR((void const __force *)drvdata->lpaif)) { + dev_err(&pdev->dev, "%s() error mapping reg resource: %ld\n", + __func__, + PTR_ERR((void const __force *)drvdata->lpaif)); + return PTR_ERR((void const __force *)drvdata->lpaif); + } + + drvdata->lpaif_map = devm_regmap_init_mmio(&pdev->dev, drvdata->lpaif, + &lpass_cpu_regmap_config); + if (IS_ERR(drvdata->lpaif_map)) { + dev_err(&pdev->dev, "%s() error initializing regmap: %ld\n", + __func__, PTR_ERR(drvdata->lpaif_map)); + return PTR_ERR(drvdata->lpaif_map); + } + + drvdata->mi2s_osr_clk = devm_clk_get(&pdev->dev, "mi2s-osr-clk"); + if (IS_ERR(drvdata->mi2s_osr_clk)) { + dev_err(&pdev->dev, "%s() error getting mi2s-osr-clk: %ld\n", + __func__, PTR_ERR(drvdata->mi2s_osr_clk)); + return PTR_ERR(drvdata->mi2s_osr_clk); + } + + drvdata->mi2s_bit_clk = devm_clk_get(&pdev->dev, "mi2s-bit-clk"); + if (IS_ERR(drvdata->mi2s_bit_clk)) { + dev_err(&pdev->dev, "%s() error getting mi2s-bit-clk: %ld\n", + __func__, PTR_ERR(drvdata->mi2s_bit_clk)); + return PTR_ERR(drvdata->mi2s_bit_clk); + } + + drvdata->ahbix_clk = devm_clk_get(&pdev->dev, "ahbix-clk"); + if (IS_ERR(drvdata->ahbix_clk)) { + dev_err(&pdev->dev, "%s() error getting ahbix-clk: %ld\n", + __func__, PTR_ERR(drvdata->ahbix_clk)); + return PTR_ERR(drvdata->ahbix_clk); + } + + ret = clk_set_rate(drvdata->ahbix_clk, LPASS_AHBIX_CLOCK_FREQUENCY); + if (ret) { + dev_err(&pdev->dev, "%s() error setting rate on ahbix_clk: %d\n", + __func__, ret); + return ret; + } + dev_dbg(&pdev->dev, "%s() set ahbix_clk rate to %lu\n", __func__, + clk_get_rate(drvdata->ahbix_clk)); + + ret = clk_prepare_enable(drvdata->ahbix_clk); + if (ret) { + dev_err(&pdev->dev, "%s() error enabling ahbix_clk: %d\n", + __func__, ret); + return ret; + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &lpass_cpu_comp_driver, &lpass_cpu_dai_driver, 1); + if (ret) { + dev_err(&pdev->dev, "%s() error registering cpu driver: %d\n", + __func__, ret); + goto err_clk; + } + + ret = asoc_qcom_lpass_platform_register(pdev); + if (ret) { + dev_err(&pdev->dev, "%s() error registering platform driver: %d\n", + __func__, ret); + goto err_clk; + } + + return 0; + +err_clk: + clk_disable_unprepare(drvdata->ahbix_clk); + return ret; +} + +static int lpass_cpu_platform_remove(struct platform_device *pdev) +{ + struct lpass_data *drvdata = platform_get_drvdata(pdev); + + clk_disable_unprepare(drvdata->ahbix_clk); + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id lpass_cpu_device_id[] = { + { .compatible = "qcom,lpass-cpu" }, + {} +}; +MODULE_DEVICE_TABLE(of, lpass_cpu_device_id); +#endif + +static struct platform_driver lpass_cpu_platform_driver = { + .driver = { + .name = "lpass-cpu", + .of_match_table = of_match_ptr(lpass_cpu_device_id), + }, + .probe = lpass_cpu_platform_probe, + .remove = lpass_cpu_platform_remove, +}; +module_platform_driver(lpass_cpu_platform_driver); + +MODULE_DESCRIPTION("QTi LPASS CPU Driver"); +MODULE_LICENSE("GPL v2"); From c5c8635a04711c7a7aca82f90e6b1e6df1c057be Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 3 Mar 2015 16:21:55 -0800 Subject: [PATCH 146/411] ASoC: qcom: Add LPASS platform driver Add platform driver for the Qualcomm Technologies low-power audio subsystem (LPASS) ports. Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 526 ++++++++++++++++++++++++++++++++ 1 file changed, 526 insertions(+) create mode 100644 sound/soc/qcom/lpass-platform.c diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c new file mode 100644 index 00000000000000..2fa6280dfb234b --- /dev/null +++ b/sound/soc/qcom/lpass-platform.c @@ -0,0 +1,526 @@ +/* + * Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * lpass-platform.c -- ALSA SoC platform driver for QTi LPASS + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "lpass-lpaif-ipq806x.h" +#include "lpass.h" + +#define LPASS_PLATFORM_BUFFER_SIZE (16 * 1024) +#define LPASS_PLATFORM_PERIODS 2 + +static struct snd_pcm_hardware lpass_platform_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16 | + SNDRV_PCM_FMTBIT_S24 | + SNDRV_PCM_FMTBIT_S32, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 8, + .buffer_bytes_max = LPASS_PLATFORM_BUFFER_SIZE, + .period_bytes_max = LPASS_PLATFORM_BUFFER_SIZE / + LPASS_PLATFORM_PERIODS, + .period_bytes_min = LPASS_PLATFORM_BUFFER_SIZE / + LPASS_PLATFORM_PERIODS, + .periods_min = LPASS_PLATFORM_PERIODS, + .periods_max = LPASS_PLATFORM_PERIODS, + .fifo_size = 0, +}; + +static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &lpass_platform_pcm_hardware); + + runtime->dma_bytes = lpass_platform_pcm_hardware.buffer_bytes_max; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(soc_runtime->dev, "%s() setting constraints failed: %d\n", + __func__, ret); + return -EINVAL; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int lpass_platform_pcmops_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct lpass_data *drvdata = + snd_soc_platform_get_drvdata(soc_runtime->platform); + snd_pcm_format_t format = params_format(params); + unsigned int channels = params_channels(params); + unsigned int regval; + int bitwidth; + int ret; + + bitwidth = snd_pcm_format_width(format); + if (bitwidth < 0) { + dev_err(soc_runtime->dev, "%s() invalid bit width given: %d\n", + __func__, bitwidth); + return bitwidth; + } + + regval = LPAIF_RDMACTL_BURSTEN_INCR4 | + LPAIF_RDMACTL_AUDINTF_MI2S | + LPAIF_RDMACTL_FIFOWM_8; + + switch (bitwidth) { + case 16: + switch (channels) { + case 1: + case 2: + regval |= LPAIF_RDMACTL_WPSCNT_ONE; + break; + case 4: + regval |= LPAIF_RDMACTL_WPSCNT_TWO; + break; + case 6: + regval |= LPAIF_RDMACTL_WPSCNT_THREE; + break; + case 8: + regval |= LPAIF_RDMACTL_WPSCNT_FOUR; + break; + default: + dev_err(soc_runtime->dev, "%s() invalid PCM config given: bw=%d, ch=%u\n", + __func__, bitwidth, channels); + return -EINVAL; + } + break; + case 24: + case 32: + switch (channels) { + case 1: + regval |= LPAIF_RDMACTL_WPSCNT_ONE; + break; + case 2: + regval |= LPAIF_RDMACTL_WPSCNT_TWO; + break; + case 4: + regval |= LPAIF_RDMACTL_WPSCNT_FOUR; + break; + case 6: + regval |= LPAIF_RDMACTL_WPSCNT_SIX; + break; + case 8: + regval |= LPAIF_RDMACTL_WPSCNT_EIGHT; + break; + default: + dev_err(soc_runtime->dev, "%s() invalid PCM config given: bw=%d, ch=%u\n", + __func__, bitwidth, channels); + return -EINVAL; + } + break; + default: + dev_err(soc_runtime->dev, "%s() invalid PCM config given: bw=%d, ch=%u\n", + __func__, bitwidth, channels); + return -EINVAL; + } + + ret = regmap_write(drvdata->lpaif_map, + LPAIF_RDMACTL_REG(LPAIF_RDMA_CHAN_MI2S), regval); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to rdmactl reg: %d\n", + __func__, ret); + return ret; + } + + return 0; +} + +static int lpass_platform_pcmops_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct lpass_data *drvdata = + snd_soc_platform_get_drvdata(soc_runtime->platform); + int ret; + + ret = regmap_write(drvdata->lpaif_map, + LPAIF_RDMACTL_REG(LPAIF_RDMA_CHAN_MI2S), 0); + if (ret) + dev_err(soc_runtime->dev, "%s() error writing to rdmactl reg: %d\n", + __func__, ret); + + return ret; +} + +static int lpass_platform_pcmops_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct lpass_data *drvdata = + snd_soc_platform_get_drvdata(soc_runtime->platform); + int ret; + + ret = regmap_write(drvdata->lpaif_map, + LPAIF_RDMABASE_REG(LPAIF_RDMA_CHAN_MI2S), + runtime->dma_addr); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to rdmabase reg: %d\n", + __func__, ret); + return ret; + } + + ret = regmap_write(drvdata->lpaif_map, + LPAIF_RDMABUFF_REG(LPAIF_RDMA_CHAN_MI2S), + (snd_pcm_lib_buffer_bytes(substream) >> 2) - 1); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to rdmabuff reg: %d\n", + __func__, ret); + return ret; + } + + ret = regmap_write(drvdata->lpaif_map, + LPAIF_RDMAPER_REG(LPAIF_RDMA_CHAN_MI2S), + (snd_pcm_lib_period_bytes(substream) >> 2) - 1); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to rdmaper reg: %d\n", + __func__, ret); + return ret; + } + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_RDMACTL_REG(LPAIF_RDMA_CHAN_MI2S), + LPAIF_RDMACTL_ENABLE_MASK, LPAIF_RDMACTL_ENABLE_ON); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to rdmactl reg: %d\n", + __func__, ret); + return ret; + } + + return 0; +} + +static int lpass_platform_pcmops_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct lpass_data *drvdata = + snd_soc_platform_get_drvdata(soc_runtime->platform); + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* clear status before enabling interrupts */ + ret = regmap_write(drvdata->lpaif_map, + LPAIF_IRQCLEAR_REG(LPAIF_IRQ_PORT_HOST), + LPAIF_IRQ_ALL(LPAIF_RDMA_CHAN_MI2S)); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to irqclear reg: %d\n", + __func__, ret); + return ret; + } + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_IRQEN_REG(LPAIF_IRQ_PORT_HOST), + LPAIF_IRQ_ALL(LPAIF_RDMA_CHAN_MI2S), + LPAIF_IRQ_ALL(LPAIF_RDMA_CHAN_MI2S)); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to irqen reg: %d\n", + __func__, ret); + return ret; + } + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_RDMACTL_REG(LPAIF_RDMA_CHAN_MI2S), + LPAIF_RDMACTL_ENABLE_MASK, + LPAIF_RDMACTL_ENABLE_ON); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to rdmactl reg: %d\n", + __func__, ret); + return ret; + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_RDMACTL_REG(LPAIF_RDMA_CHAN_MI2S), + LPAIF_RDMACTL_ENABLE_MASK, + LPAIF_RDMACTL_ENABLE_OFF); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to rdmactl reg: %d\n", + __func__, ret); + return ret; + } + + ret = regmap_update_bits(drvdata->lpaif_map, + LPAIF_IRQEN_REG(LPAIF_IRQ_PORT_HOST), + LPAIF_IRQ_ALL(LPAIF_RDMA_CHAN_MI2S), 0); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to irqen reg: %d\n", + __func__, ret); + return ret; + } + break; + } + + return 0; +} + +static snd_pcm_uframes_t lpass_platform_pcmops_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct lpass_data *drvdata = + snd_soc_platform_get_drvdata(soc_runtime->platform); + unsigned int base_addr, curr_addr; + int ret; + + ret = regmap_read(drvdata->lpaif_map, + LPAIF_RDMABASE_REG(LPAIF_RDMA_CHAN_MI2S), &base_addr); + if (ret) { + dev_err(soc_runtime->dev, "%s() error reading from rdmabase reg: %d\n", + __func__, ret); + return ret; + } + + ret = regmap_read(drvdata->lpaif_map, + LPAIF_RDMACURR_REG(LPAIF_RDMA_CHAN_MI2S), &curr_addr); + if (ret) { + dev_err(soc_runtime->dev, "%s() error reading from rdmacurr reg: %d\n", + __func__, ret); + return ret; + } + + return bytes_to_frames(substream->runtime, curr_addr - base_addr); +} + +static int lpass_platform_pcmops_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_coherent(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops lpass_platform_pcm_ops = { + .open = lpass_platform_pcmops_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = lpass_platform_pcmops_hw_params, + .hw_free = lpass_platform_pcmops_hw_free, + .prepare = lpass_platform_pcmops_prepare, + .trigger = lpass_platform_pcmops_trigger, + .pointer = lpass_platform_pcmops_pointer, + .mmap = lpass_platform_pcmops_mmap, +}; + +static irqreturn_t lpass_platform_lpaif_irq(int irq, void *data) +{ + struct snd_pcm_substream *substream = data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct lpass_data *drvdata = + snd_soc_platform_get_drvdata(soc_runtime->platform); + unsigned int interrupts; + irqreturn_t ret = IRQ_NONE; + int rv; + + rv = regmap_read(drvdata->lpaif_map, + LPAIF_IRQSTAT_REG(LPAIF_IRQ_PORT_HOST), &interrupts); + if (rv) { + dev_err(soc_runtime->dev, "%s() error reading from irqstat reg: %d\n", + __func__, rv); + return IRQ_NONE; + } + interrupts &= LPAIF_IRQ_ALL(LPAIF_RDMA_CHAN_MI2S); + + if (interrupts & LPAIF_IRQ_PER(LPAIF_RDMA_CHAN_MI2S)) { + rv = regmap_write(drvdata->lpaif_map, + LPAIF_IRQCLEAR_REG(LPAIF_IRQ_PORT_HOST), + LPAIF_IRQ_PER(LPAIF_RDMA_CHAN_MI2S)); + if (rv) { + dev_err(soc_runtime->dev, "%s() error writing to irqclear reg: %d\n", + __func__, rv); + return IRQ_NONE; + } + snd_pcm_period_elapsed(substream); + ret = IRQ_HANDLED; + } + + if (interrupts & LPAIF_IRQ_XRUN(LPAIF_RDMA_CHAN_MI2S)) { + rv = regmap_write(drvdata->lpaif_map, + LPAIF_IRQCLEAR_REG(LPAIF_IRQ_PORT_HOST), + LPAIF_IRQ_XRUN(LPAIF_RDMA_CHAN_MI2S)); + if (rv) { + dev_err(soc_runtime->dev, "%s() error writing to irqclear reg: %d\n", + __func__, rv); + return IRQ_NONE; + } + dev_warn(soc_runtime->dev, "%s() xrun warning\n", __func__); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + ret = IRQ_HANDLED; + } + + if (interrupts & LPAIF_IRQ_ERR(LPAIF_RDMA_CHAN_MI2S)) { + rv = regmap_write(drvdata->lpaif_map, + LPAIF_IRQCLEAR_REG(LPAIF_IRQ_PORT_HOST), + LPAIF_IRQ_ERR(LPAIF_RDMA_CHAN_MI2S)); + if (rv) { + dev_err(soc_runtime->dev, "%s() error writing to irqclear reg: %d\n", + __func__, rv); + return IRQ_NONE; + } + dev_err(soc_runtime->dev, "%s() bus access error\n", __func__); + snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED); + ret = IRQ_HANDLED; + } + + return ret; +} + +static int lpass_platform_alloc_buffer(struct snd_pcm_substream *substream, + struct snd_soc_pcm_runtime *soc_runtime) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = lpass_platform_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = soc_runtime->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(soc_runtime->dev, size, &buf->addr, + GFP_KERNEL); + if (!buf->area) { + dev_err(soc_runtime->dev, "%s: Could not allocate DMA buffer\n", + __func__); + return -ENOMEM; + } + buf->bytes = size; + + return 0; +} + +static void lpass_platform_free_buffer(struct snd_pcm_substream *substream, + struct snd_soc_pcm_runtime *soc_runtime) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + + if (buf->area) { + dma_free_coherent(soc_runtime->dev, buf->bytes, buf->area, + buf->addr); + } + buf->area = NULL; +} + +static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) +{ + struct snd_pcm *pcm = soc_runtime->pcm; + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct lpass_data *drvdata = + snd_soc_platform_get_drvdata(soc_runtime->platform); + int ret; + + soc_runtime->dev->coherent_dma_mask = DMA_BIT_MASK(32); + soc_runtime->dev->dma_mask = &soc_runtime->dev->coherent_dma_mask; + + ret = lpass_platform_alloc_buffer(substream, soc_runtime); + if (ret) + return ret; + + ret = devm_request_irq(soc_runtime->dev, drvdata->lpaif_irq, + lpass_platform_lpaif_irq, IRQF_TRIGGER_RISING, + "lpass-irq-lpaif", substream); + if (ret) { + dev_err(soc_runtime->dev, "%s() irq request failed: %d\n", + __func__, ret); + goto err_buf; + } + + /* ensure audio hardware is disabled */ + ret = regmap_write(drvdata->lpaif_map, + LPAIF_IRQEN_REG(LPAIF_IRQ_PORT_HOST), 0); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to irqen reg: %d\n", + __func__, ret); + return ret; + } + ret = regmap_write(drvdata->lpaif_map, + LPAIF_RDMACTL_REG(LPAIF_RDMA_CHAN_MI2S), 0); + if (ret) { + dev_err(soc_runtime->dev, "%s() error writing to rdmactl reg: %d\n", + __func__, ret); + return ret; + } + + return 0; + +err_buf: + lpass_platform_free_buffer(substream, soc_runtime); + return ret; +} + +static void lpass_platform_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + + lpass_platform_free_buffer(substream, soc_runtime); +} + +static struct snd_soc_platform_driver lpass_platform_driver = { + .pcm_new = lpass_platform_pcm_new, + .pcm_free = lpass_platform_pcm_free, + .ops = &lpass_platform_pcm_ops, +}; + +int asoc_qcom_lpass_platform_register(struct platform_device *pdev) +{ + struct lpass_data *drvdata = platform_get_drvdata(pdev); + + drvdata->lpaif_irq = platform_get_irq_byname(pdev, "lpass-irq-lpaif"); + if (drvdata->lpaif_irq < 0) { + dev_err(&pdev->dev, "%s() error getting irq handle: %d\n", + __func__, drvdata->lpaif_irq); + return -ENODEV; + } + + return devm_snd_soc_register_platform(&pdev->dev, + &lpass_platform_driver); +} +EXPORT_SYMBOL_GPL(asoc_qcom_lpass_platform_register); + +MODULE_DESCRIPTION("QTi LPASS Platform Driver"); +MODULE_LICENSE("GPL v2"); From 8dd72c42d38e62b82d7e3c47173b502d851c48ad Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 3 Mar 2015 16:21:51 -0800 Subject: [PATCH 147/411] ASoC: qcom: Document LPASS CPU bindings Add documentation to the sound directory of the device-tree bindings for the QTi LPASS CPU DAI device. Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- .../bindings/sound/qcom,lpass-cpu.txt | 49 +++++++++++++++++++ 1 file changed, 49 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt new file mode 100644 index 00000000000000..e7c6e9321863f0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -0,0 +1,49 @@ +* Qualcomm Technologies LPASS CPU DAI + +This node models the Qualcomm Technologies Low-Power Audio SubSystem (LPASS). + +Required properties: + +- compatible : "qcom,lpass-cpu" +- clocks : Must contain an entry for each entry in clock-names. +- clock-names : A list which must include the following entries: + * "ahbix-clk" + * "mi2s-osr-clk" + * "mi2s-bit-clk" +- interrupts : Must contain an entry for each entry in + interrupt-names. +- interrupt-names : A list which must include the following entries: + * "lpass-irq-lpaif" +- pinctrl-N : One property must exist for each entry in + pinctrl-names. See ../pinctrl/pinctrl-bindings.txt + for details of the property values. +- pinctrl-names : Must contain a "default" entry. +- reg : Must contain an address for each entry in reg-names. +- reg-names : A list which must include the following entries: + * "lpass-lpaif" + +Required subnodes: + +- qcom,adsp : Audio DSP sub-node + +Optional Audio DSP subnode properties: + +- status : "disabled" indicates the adsp is not available. + +Example: + +lpass@28100000 { + compatible = "qcom,lpass-cpu"; + clocks = <&lcc AHBIX_CLK>, <&lcc MI2S_OSR_CLK>, <&lcc MI2S_BIT_CLK>; + clock-names = "ahbix-clk", "mi2s-osr-clk", "mi2s-bit-clk"; + interrupts = <0 85 1>; + interrupt-names = "lpass-irq-lpaif"; + pinctrl-names = "default", "idle"; + pinctrl-0 = <&mi2s_default>; + pinctrl-1 = <&mi2s_idle>; + reg = <0x28100000 0x10000>; + reg-names = "lpass-lpaif"; + qcom,adsp { + status = "disabled"; + }; +}; From 2d800897e868ba561733ecffb30d840cbe5c86d4 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 3 Mar 2015 16:21:50 -0800 Subject: [PATCH 148/411] MAINTAINERS: Add QCOM audio ASoC maintainer Add maintainers for the Qualcomm Technologies sound drivers. Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index ddc5a8cf9a8ac0..9514b794b74d7f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5254,6 +5254,13 @@ F: drivers/char/ipmi/ F: include/linux/ipmi* F: include/uapi/linux/ipmi* +QCOM AUDIO (ASoC) DRIVERS +M: Patrick Lai +M: Banajit Goswami +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Supported +F: sound/soc/qcom/ + IPS SCSI RAID DRIVER M: Adaptec OEM Raid Solutions L: linux-scsi@vger.kernel.org From 66e618857ca46433741bf97ceca1b425387400b1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 6 Mar 2015 09:07:32 +0200 Subject: [PATCH 149/411] ASoC: davinci-mcasp: Fix compilation error Introduced by commit: 6afda7f50754 ASoC: davinci-mcasp: Allow complete shutdown of McASP when not in use I'm really sorry for this... Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 33cea0728cd26e..d40b392b3da2d8 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1058,7 +1058,7 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) u32 reg; int i; - context->pm_state = pm_runtime_enabled(mcasp->dev) + context->pm_state = pm_runtime_enabled(mcasp->dev); if (!context->pm_state) pm_runtime_get_sync(mcasp->dev); From d6482288aadcf19e348cbccff7a605790a3b3875 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Fri, 6 Mar 2015 16:59:00 +0100 Subject: [PATCH 150/411] ALSA: ac97: Add VT1613 AC97 codec support Patch to add an VT1613 AC97 codec support. This codec has additional DC offset removal control, headphone output and no video input. Signed-off-by: Maciej Szmigiero Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 1 + sound/pci/ac97/ac97_patch.c | 33 +++++++++++++++++++++++++++++++++ 2 files changed, 34 insertions(+) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 5ee2f17c287c0f..5bca1a33fed69e 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -177,6 +177,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] { 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, +{ 0x56494120, 0xfffffff0, "VIA1613", patch_vt1613, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF { 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index ceaac1c4190699..eca22100f70d4a 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3351,6 +3351,39 @@ static int patch_cm9780(struct snd_ac97 *ac97) return 0; } +/* + * VIA VT1613 codec + */ +static const struct snd_kcontrol_new snd_ac97_controls_vt1613[] = { +AC97_SINGLE("DC Offset removal", 0x5a, 10, 1, 0), +}; + +static int patch_vt1613_specific(struct snd_ac97 *ac97) +{ + int err; + + err = patch_build_controls(ac97, &snd_ac97_controls_vt1613[0], + ARRAY_SIZE(snd_ac97_controls_vt1613)); + if (err) + return err; + + return 0; +}; + +static const struct snd_ac97_build_ops patch_vt1613_ops = { + .build_specific = patch_vt1613_specific +}; + +static int patch_vt1613(struct snd_ac97 *ac97) +{ + ac97->build_ops = &patch_vt1613_ops; + + ac97->flags |= AC97_HAS_NO_VIDEO; + ac97->caps |= AC97_BC_HEADPHONE; + + return 0; +} + /* * VIA VT1616 codec */ From 5371fc0ecdf55b6811ade8a198de8ace2f4e5861 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 6 Mar 2015 13:41:42 -0300 Subject: [PATCH 151/411] ALSA: ac97: ac97_patch: Simplify patch_vt1613_specific() We can simplify the code by returning patch_build_controls() directly. Signed-off-by: Fabio Estevam Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index eca22100f70d4a..f4234edb878c7a 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3360,14 +3360,8 @@ AC97_SINGLE("DC Offset removal", 0x5a, 10, 1, 0), static int patch_vt1613_specific(struct snd_ac97 *ac97) { - int err; - - err = patch_build_controls(ac97, &snd_ac97_controls_vt1613[0], - ARRAY_SIZE(snd_ac97_controls_vt1613)); - if (err) - return err; - - return 0; + return patch_build_controls(ac97, &snd_ac97_controls_vt1613[0], + ARRAY_SIZE(snd_ac97_controls_vt1613)); }; static const struct snd_ac97_build_ops patch_vt1613_ops = { From 3d4cf65e2db58f927ca3cd13b0074b8fe5124659 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 25 Feb 2015 16:42:12 +0100 Subject: [PATCH 152/411] ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency The change to enable OMAP5 support on this platform was a little too eager in adding a 'select' for a particular clock driver that might not be enabled in all configurations, which in turn leads to a build error: warning: (SND_OMAP_SOC_OMAP_ABE_TWL6040) selects COMMON_CLK_PALMAS which has unmet direct dependencies (COMMON_CLK && MFD_PALMAS) drivers/built-in.o: In function `palmas_clks_probe': drivers/clk/clk-palmas.c:228: undefined reference to `palmas_ext_control_req_config' I do not see a strong dependency here, so it's probably better to drop this select and to avoid adding more complexity here. Fixes: 5163c1eede8e9 ("ASoC: omap: Kconfig: Support for omap5-uevm analog audio") Signed-off-by: Arnd Bergmann Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e7c78b0406b59c..6768e4f7d7d0e2 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -105,7 +105,7 @@ config SND_OMAP_SOC_OMAP_ABE_TWL6040 select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC - select COMMON_CLK_PALMAS if SOC_OMAP5 + select COMMON_CLK_PALMAS if MFD_PALMAS help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: From 5af76d5c0882435241841186b054262407e9eabb Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Fri, 27 Feb 2015 12:52:26 +0800 Subject: [PATCH 153/411] ASoC: rt286: correct the OR to AND Here it should be AND(&) to check the status. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 16723b167fbf0d..49c44a77b5188d 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -397,7 +397,7 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) if (jack) { /* enable IRQ */ - if (rt286->jack->status | SND_JACK_HEADPHONE) + if (rt286->jack->status & SND_JACK_HEADPHONE) snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1"); regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); /* Send an initial empty report */ From 1a5ab21c2e0f3d6b25ee9f7ca3429fac57027f76 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Fri, 27 Feb 2015 12:54:29 +0800 Subject: [PATCH 154/411] ASoC: Intel: Add suspend_pre and resume_post for Broadwell snd_soc_card For broadwell machine, we need do some machine related setting before suspend and after resume, e.g. disable/enable jack detection, here adding snd_soc_card suspend_pre and resume_post for this task. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index fba2ef5dac4280..af5d73070f60f9 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -225,6 +225,32 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { }, }; +static int broadwell_suspend(struct snd_soc_card *card){ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); + rt286_mic_detect(codec, NULL); + break; + } + } + return 0; +} + +static int broadwell_resume(struct snd_soc_card *card){ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + dev_dbg(codec->dev, "enabling jack detect for resume.\n"); + rt286_mic_detect(codec, &broadwell_headset); + break; + } + } + return 0; +} + /* broadwell audio machine driver for WPT + RT286S */ static struct snd_soc_card broadwell_rt286 = { .name = "broadwell-rt286", @@ -238,6 +264,8 @@ static struct snd_soc_card broadwell_rt286 = { .dapm_routes = broadwell_rt286_map, .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), .fully_routed = true, + .suspend_pre = broadwell_suspend, + .resume_post = broadwell_resume, }; static int broadwell_audio_probe(struct platform_device *pdev) From 0aed11244360c24c854a263eac0293acef2abd03 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 24 Feb 2015 00:54:16 +0000 Subject: [PATCH 155/411] dmaengine: export symbol of of_dma_request_slave_channel() Current DMAEngine implementation of DT bindings can't support DT subnode. This patch export symbols of of_dma_request_slave_channel() for subnode DMA DT bingings. ex) rcar_sound: rcar_sound@ec500000 { ... rcar_sound,dvc { dvc0: dvc@0 { dmas = <&audma0 0xbc>; dma-names = "tx"; }; dvc1: dvc@1 { dmas = <&audma0 0xbe>; dma-names = "tx"; }; }; ... }; Signed-off-by: Kuninori Morimoto Acked-by: Vinod Koul Signed-off-by: Mark Brown --- drivers/dma/of-dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/dma/of-dma.c b/drivers/dma/of-dma.c index ca31f1b45366d2..cbd4a8aff12087 100644 --- a/drivers/dma/of-dma.c +++ b/drivers/dma/of-dma.c @@ -194,6 +194,7 @@ struct dma_chan *of_dma_request_slave_channel(struct device_node *np, return ERR_PTR(ret_no_channel); } +EXPORT_SYMBOL_GPL(of_dma_request_slave_channel); /** * of_dma_simple_xlate - Simple DMA engine translation function From c303cf0895ad927f5e9b8a5f8ed1c6b8c96a500f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:22:37 +0000 Subject: [PATCH 156/411] ASoC: rsnd: remove SH-DMA-BASE specific implementation Renesas R-Car sound had SH-DMA-BASE specific implementation before, but, now, it is no longer needed. Let's remove it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 1 - sound/soc/sh/rcar/rsnd.h | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1b53605f715439..0b14d3762cff84 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -94,7 +94,6 @@ * */ #include -#include #include "rsnd.h" #define RSND_RATES SNDRV_PCM_RATE_8000_96000 diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e7914bd610e212..5ffde9b8955d68 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -171,7 +171,6 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod); * R-Car DMA */ struct rsnd_dma { - struct sh_dmae_slave slave; struct dma_chan *chan; enum dma_transfer_direction dir; dma_addr_t addr; From 4715219ecef50cf79d7784545bf5bb4664bb800d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:23:08 +0000 Subject: [PATCH 157/411] ASoC: rsnd: remove un-needed parameter from rsnd_dma_init() It can get DMA direction via rsnd_dai_stream. Remove un-needed is_play from rsnd_dma_init(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 6 ++++-- sound/soc/sh/rcar/rsnd.h | 16 +--------------- sound/soc/sh/rcar/src.c | 1 - sound/soc/sh/rcar/ssi.c | 1 - 4 files changed, 5 insertions(+), 19 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 0b14d3762cff84..1edf4dc41c70de 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -311,13 +311,15 @@ static void rsnd_dma_of_path(struct rsnd_dma *dma, } } -int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, - int is_play, int id) +int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) { struct device *dev = rsnd_priv_to_dev(priv); struct dma_slave_config cfg; + struct rsnd_mod *mod = rsnd_dma_to_mod(dma); struct rsnd_mod *mod_from; struct rsnd_mod *mod_to; + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + int is_play = rsnd_io_is_play(io); char dma_name[DMA_NAME_SIZE]; dma_cap_mask_t mask; int ret; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 5ffde9b8955d68..93a1a256f37c13 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -179,8 +179,7 @@ struct rsnd_dma { void rsnd_dma_start(struct rsnd_dma *dma); void rsnd_dma_stop(struct rsnd_dma *dma); int rsnd_dma_available(struct rsnd_dma *dma); -int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, - int is_play, int id); +int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id); void rsnd_dma_quit(struct rsnd_priv *priv, struct rsnd_dma *dma); @@ -413,19 +412,6 @@ struct rsnd_priv { #define rsnd_lock(priv, flags) spin_lock_irqsave(&priv->lock, flags) #define rsnd_unlock(priv, flags) spin_unlock_irqrestore(&priv->lock, flags) -#define rsnd_info_is_playback(priv, type) \ -({ \ - struct rcar_snd_info *info = rsnd_priv_to_info(priv); \ - int i, is_play = 0; \ - for (i = 0; i < info->dai_info_nr; i++) { \ - if (info->dai_info[i].playback.type == (type)->info) { \ - is_play = 1; \ - break; \ - } \ - } \ - is_play; \ -}) - /* * rsnd_kctrl */ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 81c182b4bad531..7fb879871a8e34 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -713,7 +713,6 @@ static int rsnd_src_probe_gen2(struct rsnd_mod *mod, ret = rsnd_dma_init(priv, rsnd_mod_to_dma(mod), - rsnd_info_is_playback(priv, src), src->info->dma_id); if (ret) goto rsnd_src_probe_gen2_fail; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 9e7b627c08e225..57e737c7046beb 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -478,7 +478,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, ret = rsnd_dma_init( priv, rsnd_mod_to_dma(mod), - rsnd_info_is_playback(priv, ssi), dma_id); if (ret) goto rsnd_ssi_dma_probe_fail; From 9c706ab29f33b9562f570d1e99e21955d898dc85 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:23:39 +0000 Subject: [PATCH 158/411] ASoC: rsnd: remove unused rsnd_dma_available() rsnd_dma_available() is not used. Let's remove it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 5 ----- sound/soc/sh/rcar/rsnd.h | 1 - sound/soc/sh/rcar/src.c | 2 -- sound/soc/sh/rcar/ssi.c | 2 -- 4 files changed, 10 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1edf4dc41c70de..1da94bf730d2b7 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -222,11 +222,6 @@ void rsnd_dma_start(struct rsnd_dma *dma) dma_async_issue_pending(dma->chan); } -int rsnd_dma_available(struct rsnd_dma *dma) -{ - return !!dma->chan; -} - #define DMA_NAME_SIZE 16 #define MOD_MAX 4 /* MEM/SSI/SRC/DVC */ static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 93a1a256f37c13..cb12861a1f3c6b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -178,7 +178,6 @@ struct rsnd_dma { void rsnd_dma_start(struct rsnd_dma *dma); void rsnd_dma_stop(struct rsnd_dma *dma); -int rsnd_dma_available(struct rsnd_dma *dma); int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id); void rsnd_dma_quit(struct rsnd_priv *priv, struct rsnd_dma *dma); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 7fb879871a8e34..f12c8b3aa475f6 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -30,8 +30,6 @@ struct rsnd_src { #define rsnd_src_convert_rate(p) ((p)->info->convert_rate) #define rsnd_mod_to_src(_mod) \ container_of((_mod), struct rsnd_src, mod) -#define rsnd_src_dma_available(src) \ - rsnd_dma_available(rsnd_mod_to_dma(&(src)->mod)) #define for_each_rsnd_src(pos, priv, i) \ for ((i) = 0; \ diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 57e737c7046beb..fcc77ea369b9ff 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -80,8 +80,6 @@ struct rsnd_ssi { #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) #define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma)) #define rsnd_ssi_pio_available(ssi) ((ssi)->info->irq > 0) -#define rsnd_ssi_dma_available(ssi) \ - rsnd_dma_available(rsnd_mod_to_dma(&(ssi)->mod)) #define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent) #define rsnd_ssi_mode_flags(p) ((p)->info->flags) #define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id) From 8a2ff4262ca611c38b31fec0af65be656d934f52 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:24:06 +0000 Subject: [PATCH 159/411] ASoC: rsnd: remove un-needed parameter from rsnd_dma_quit() priv is not used on rsnd_dma_quit() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 5 ++--- sound/soc/sh/rcar/rsnd.h | 3 +-- sound/soc/sh/rcar/src.c | 2 +- sound/soc/sh/rcar/ssi.c | 2 +- 4 files changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1da94bf730d2b7..7db686d0cbd805 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -358,7 +358,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) return 0; rsnd_dma_init_err: - rsnd_dma_quit(priv, dma); + rsnd_dma_quit(dma); rsnd_dma_channel_err: /* @@ -370,8 +370,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) return -EAGAIN; } -void rsnd_dma_quit(struct rsnd_priv *priv, - struct rsnd_dma *dma) +void rsnd_dma_quit(struct rsnd_dma *dma) { if (dma->chan) dma_release_channel(dma->chan); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index cb12861a1f3c6b..6ee97e7f994898 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -179,8 +179,7 @@ struct rsnd_dma { void rsnd_dma_start(struct rsnd_dma *dma); void rsnd_dma_stop(struct rsnd_dma *dma); int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id); -void rsnd_dma_quit(struct rsnd_priv *priv, - struct rsnd_dma *dma); +void rsnd_dma_quit(struct rsnd_dma *dma); /* diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index f12c8b3aa475f6..e2792056ce24e0 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -730,7 +730,7 @@ static int rsnd_src_probe_gen2(struct rsnd_mod *mod, static int rsnd_src_remove_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv) { - rsnd_dma_quit(priv, rsnd_mod_to_dma(mod)); + rsnd_dma_quit(rsnd_mod_to_dma(mod)); return 0; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fcc77ea369b9ff..a0d902ad5985ea 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -499,7 +499,7 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); int irq = ssi->info->irq; - rsnd_dma_quit(priv, rsnd_mod_to_dma(mod)); + rsnd_dma_quit(rsnd_mod_to_dma(mod)); /* PIO will request IRQ again */ devm_free_irq(dev, irq, ssi); From 4ce3b17bd43b4f0136b1d0c7782d06fe9d3addfb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:24:27 +0000 Subject: [PATCH 160/411] ASoC: rsnd: tidyup rsnd_dma_to_mod() macro declaration position Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6ee97e7f994898..ec77c9f1a57ce3 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -182,6 +182,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id); void rsnd_dma_quit(struct rsnd_dma *dma); +#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) + /* * R-Car sound mod */ @@ -253,7 +255,6 @@ struct rsnd_mod { #define rsnd_mod_to_priv(mod) (rsnd_io_to_priv(rsnd_mod_to_io(mod))) #define rsnd_mod_to_dma(mod) (&(mod)->dma) -#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) #define rsnd_mod_to_io(mod) ((mod)->io) #define rsnd_mod_id(mod) ((mod)->id) #define rsnd_mod_hw_start(mod) clk_prepare_enable((mod)->clk) From 7277911c87ba85436600f5b7aab15de112416795 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:24:52 +0000 Subject: [PATCH 161/411] ASoC: rsnd: enable to get resource by name rsnd driver needs to support Audio DMAC peri peri inside sound driver. getting resource by name is useful for it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index de0685f2abaee6..d08bcd3dbfbf8f 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -118,11 +118,12 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, mask, data); } -#define rsnd_gen_regmap_init(priv, id_size, reg_id, conf) \ - _rsnd_gen_regmap_init(priv, id_size, reg_id, conf, ARRAY_SIZE(conf)) +#define rsnd_gen_regmap_init(priv, id_size, reg_id, name, conf) \ + _rsnd_gen_regmap_init(priv, id_size, reg_id, name, conf, ARRAY_SIZE(conf)) static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, int id_size, int reg_id, + const char *name, struct rsnd_regmap_field_conf *conf, int conf_size) { @@ -142,7 +143,9 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, regc.val_bits = 32; regc.reg_stride = 4; - res = platform_get_resource(pdev, IORESOURCE_MEM, reg_id); + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, name); + if (!res) + res = platform_get_resource(pdev, IORESOURCE_MEM, reg_id); if (!res) return -ENODEV; @@ -368,10 +371,10 @@ static int rsnd_gen2_probe(struct platform_device *pdev, int ret_adg; int ret_ssi; - ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, conf_ssiu); - ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, conf_scu); - ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_ADG, conf_adg); - ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, conf_ssi); + ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, "ssiu", conf_ssiu); + ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, "scu", conf_scu); + ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_ADG, "adg", conf_adg); + ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, "ssi", conf_ssi); if (ret_ssiu < 0 || ret_scu < 0 || ret_adg < 0 || @@ -440,9 +443,9 @@ static int rsnd_gen1_probe(struct platform_device *pdev, int ret_adg; int ret_ssi; - ret_sru = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SRU, conf_sru); - ret_adg = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_ADG, conf_adg); - ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, conf_ssi); + ret_sru = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SRU, "sru", conf_sru); + ret_adg = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_ADG, "adg", conf_adg); + ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, "ssi", conf_ssi); if (ret_sru < 0 || ret_adg < 0 || ret_ssi < 0) From c5212b4556b6bd120b0f4e4ae7c4a1cb9f5efe07 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:25:27 +0000 Subject: [PATCH 162/411] ASoC: rsnd: add rsnd_gen_get_phy_addr() to get physical address physical address is required from Audio DMAC peri peri. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 16 +++++++++++----- sound/soc/sh/rcar/rsnd.h | 1 + 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index d08bcd3dbfbf8f..0da04ed3aabe11 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -28,6 +28,7 @@ struct rsnd_gen { struct regmap *regmap[RSND_BASE_MAX]; struct regmap_field *regs[RSND_REG_MAX]; + phys_addr_t res[RSND_REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) @@ -118,6 +119,13 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, mask, data); } +phys_addr_t rsnd_gen_get_phy_addr(struct rsnd_priv *priv, int reg_id) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + return gen->res[reg_id]; +} + #define rsnd_gen_regmap_init(priv, id_size, reg_id, name, conf) \ _rsnd_gen_regmap_init(priv, id_size, reg_id, name, conf, ARRAY_SIZE(conf)) static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, @@ -159,6 +167,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, gen->base[reg_id] = base; gen->regmap[reg_id] = regmap; + gen->res[reg_id] = res->start; for (i = 0; i < conf_size; i++) { @@ -216,13 +225,10 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv, struct rsnd_mod *mod, int is_play, int is_from) { - struct platform_device *pdev = rsnd_priv_to_pdev(priv); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - dma_addr_t ssi_reg = platform_get_resource(pdev, - IORESOURCE_MEM, RSND_GEN2_SSI)->start; - dma_addr_t src_reg = platform_get_resource(pdev, - IORESOURCE_MEM, RSND_GEN2_SCU)->start; + phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI); + phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU); int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod); int use_src = !!rsnd_io_to_mod_src(io); int use_dvc = !!rsnd_io_to_mod_dvc(io); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ec77c9f1a57ce3..8a8a4d5d55efaf 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -331,6 +331,7 @@ void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv, struct rsnd_mod *mod, int is_play, int is_from); +phys_addr_t rsnd_gen_get_phy_addr(struct rsnd_priv *priv, int reg_id); #define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) #define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2) From bfe834be9525a82c8a40380c7df8ca3b149e9c93 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:25:55 +0000 Subject: [PATCH 163/411] ASoC: rsnd: add dma.c for Audio DMAC / Audio DMAC peri peri Renesas sound driver had been assumed that Audio DMAC / Audio DMAC peri peri are implemented by DMAEngine. But, result of DMA ML discussion, it was concluded that it is not a general purpose DMAC. And it should be moved from current DMAEngine to rsnd driver. So, Audio DMAC peri peri become non DMAEngine. OTOH, ALSA SoC has soc-generic-dmaengine-pcm implementation. but it seems difficult to use this generic implementation on rsnd driver at this point, since it needs to fallback to PIO mode if DMA can't use. and additionally it needs 2 DMAC (= Audio DMAC / Audio DMAC peri peri). These are not "generic" feature. Of course I will try to use this generic dmaengine in the future somehow, but just use current style at this point until it can formally use 2 DMACs. This patch adds new dma.c and moves current dma code to dma.c from core.c. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 218 ---------------------------------- sound/soc/sh/rcar/dma.c | 231 +++++++++++++++++++++++++++++++++++++ 3 files changed, 232 insertions(+), 219 deletions(-) create mode 100644 sound/soc/sh/rcar/dma.c diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index 9ac536429800dc..7b204925b8c5b2 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,2 @@ -snd-soc-rcar-objs := core.o gen.o src.o adg.o ssi.o dvc.o +snd-soc-rcar-objs := core.o gen.o dma.o src.o adg.o ssi.o dvc.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o \ No newline at end of file diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 7db686d0cbd805..9beea9ba338a01 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -160,224 +160,6 @@ void rsnd_mod_init(struct rsnd_mod *mod, mod->clk = clk; } -/* - * rsnd_dma functions - */ -void rsnd_dma_stop(struct rsnd_dma *dma) -{ - dmaengine_terminate_all(dma->chan); -} - -static void rsnd_dma_complete(void *data) -{ - struct rsnd_dma *dma = (struct rsnd_dma *)data; - struct rsnd_mod *mod = rsnd_dma_to_mod(dma); - struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - - /* - * Renesas sound Gen1 needs 1 DMAC, - * Gen2 needs 2 DMAC. - * In Gen2 case, it are Audio-DMAC, and Audio-DMAC-peri-peri. - * But, Audio-DMAC-peri-peri doesn't have interrupt, - * and this driver is assuming that here. - * - * If Audio-DMAC-peri-peri has interrpt, - * rsnd_dai_pointer_update() will be called twice, - * ant it will breaks io->byte_pos - */ - - rsnd_dai_pointer_update(io, io->byte_per_period); -} - -void rsnd_dma_start(struct rsnd_dma *dma) -{ - struct rsnd_mod *mod = rsnd_dma_to_mod(dma); - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - struct snd_pcm_substream *substream = io->substream; - struct device *dev = rsnd_priv_to_dev(priv); - struct dma_async_tx_descriptor *desc; - - desc = dmaengine_prep_dma_cyclic(dma->chan, - (dma->addr) ? dma->addr : - substream->runtime->dma_addr, - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), - dma->dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - - if (!desc) { - dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } - - desc->callback = rsnd_dma_complete; - desc->callback_param = dma; - - if (dmaengine_submit(desc) < 0) { - dev_err(dev, "dmaengine_submit() fail\n"); - return; - } - - dma_async_issue_pending(dma->chan); -} - -#define DMA_NAME_SIZE 16 -#define MOD_MAX 4 /* MEM/SSI/SRC/DVC */ -static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) -{ - if (mod) - return snprintf(dma_name, DMA_NAME_SIZE / 2, "%s%d", - rsnd_mod_dma_name(mod), rsnd_mod_id(mod)); - else - return snprintf(dma_name, DMA_NAME_SIZE / 2, "mem"); - -} - -static void rsnd_dma_of_name(struct rsnd_mod *mod_from, - struct rsnd_mod *mod_to, - char *dma_name) -{ - int index = 0; - - index = _rsnd_dma_of_name(dma_name + index, mod_from); - *(dma_name + index++) = '_'; - index = _rsnd_dma_of_name(dma_name + index, mod_to); -} - -static void rsnd_dma_of_path(struct rsnd_dma *dma, - int is_play, - struct rsnd_mod **mod_from, - struct rsnd_mod **mod_to) -{ - struct rsnd_mod *this = rsnd_dma_to_mod(dma); - struct rsnd_dai_stream *io = rsnd_mod_to_io(this); - struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); - struct rsnd_mod *src = rsnd_io_to_mod_src(io); - struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); - struct rsnd_mod *mod[MOD_MAX]; - int i, index; - - - for (i = 0; i < MOD_MAX; i++) - mod[i] = NULL; - - /* - * in play case... - * - * src -> dst - * - * mem -> SSI - * mem -> SRC -> SSI - * mem -> SRC -> DVC -> SSI - */ - mod[0] = NULL; /* for "mem" */ - index = 1; - for (i = 1; i < MOD_MAX; i++) { - if (!src) { - mod[i] = ssi; - } else if (!dvc) { - mod[i] = src; - src = NULL; - } else { - if ((!is_play) && (this == src)) - this = dvc; - - mod[i] = (is_play) ? src : dvc; - i++; - mod[i] = (is_play) ? dvc : src; - src = NULL; - dvc = NULL; - } - - if (mod[i] == this) - index = i; - - if (mod[i] == ssi) - break; - } - - if (is_play) { - *mod_from = mod[index - 1]; - *mod_to = mod[index]; - } else { - *mod_from = mod[index]; - *mod_to = mod[index - 1]; - } -} - -int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) -{ - struct device *dev = rsnd_priv_to_dev(priv); - struct dma_slave_config cfg; - struct rsnd_mod *mod = rsnd_dma_to_mod(dma); - struct rsnd_mod *mod_from; - struct rsnd_mod *mod_to; - struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - int is_play = rsnd_io_is_play(io); - char dma_name[DMA_NAME_SIZE]; - dma_cap_mask_t mask; - int ret; - - if (dma->chan) { - dev_err(dev, "it already has dma channel\n"); - return -EIO; - } - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - - rsnd_dma_of_path(dma, is_play, &mod_from, &mod_to); - rsnd_dma_of_name(mod_from, mod_to, dma_name); - - cfg.slave_id = id; - cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - cfg.src_addr = rsnd_gen_dma_addr(priv, mod_from, is_play, 1); - cfg.dst_addr = rsnd_gen_dma_addr(priv, mod_to, is_play, 0); - cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - - dev_dbg(dev, "dma : %s %pad -> %pad\n", - dma_name, &cfg.src_addr, &cfg.dst_addr); - - dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, - (void *)id, dev, - dma_name); - if (!dma->chan) { - dev_err(dev, "can't get dma channel\n"); - goto rsnd_dma_channel_err; - } - - ret = dmaengine_slave_config(dma->chan, &cfg); - if (ret < 0) - goto rsnd_dma_init_err; - - dma->addr = is_play ? cfg.src_addr : cfg.dst_addr; - dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - - return 0; - -rsnd_dma_init_err: - rsnd_dma_quit(dma); -rsnd_dma_channel_err: - - /* - * DMA failed. try to PIO mode - * see - * rsnd_ssi_fallback() - * rsnd_rdai_continuance_probe() - */ - return -EAGAIN; -} - -void rsnd_dma_quit(struct rsnd_dma *dma) -{ - if (dma->chan) - dma_release_channel(dma->chan); - - dma->chan = NULL; -} - /* * settting function */ diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c new file mode 100644 index 00000000000000..37acd409e88cf2 --- /dev/null +++ b/sound/soc/sh/rcar/dma.c @@ -0,0 +1,231 @@ +/* + * Renesas R-Car Audio DMAC support + * + * Copyright (C) 2015 Renesas Electronics Corp. + * Copyright (c) 2015 Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include "rsnd.h" + +static void rsnd_dma_complete(void *data) +{ + struct rsnd_dma *dma = (struct rsnd_dma *)data; + struct rsnd_mod *mod = rsnd_dma_to_mod(dma); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + + /* + * Renesas sound Gen1 needs 1 DMAC, + * Gen2 needs 2 DMAC. + * In Gen2 case, it are Audio-DMAC, and Audio-DMAC-peri-peri. + * But, Audio-DMAC-peri-peri doesn't have interrupt, + * and this driver is assuming that here. + * + * If Audio-DMAC-peri-peri has interrpt, + * rsnd_dai_pointer_update() will be called twice, + * ant it will breaks io->byte_pos + */ + + rsnd_dai_pointer_update(io, io->byte_per_period); +} + +#define DMA_NAME_SIZE 16 +#define MOD_MAX 4 /* MEM/SSI/SRC/DVC */ +static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) +{ + if (mod) + return snprintf(dma_name, DMA_NAME_SIZE / 2, "%s%d", + rsnd_mod_dma_name(mod), rsnd_mod_id(mod)); + else + return snprintf(dma_name, DMA_NAME_SIZE / 2, "mem"); + +} + +static void rsnd_dma_of_name(struct rsnd_mod *mod_from, + struct rsnd_mod *mod_to, + char *dma_name) +{ + int index = 0; + + index = _rsnd_dma_of_name(dma_name + index, mod_from); + *(dma_name + index++) = '_'; + index = _rsnd_dma_of_name(dma_name + index, mod_to); +} + +void rsnd_dma_stop(struct rsnd_dma *dma) +{ + dmaengine_terminate_all(dma->chan); +} + +void rsnd_dma_start(struct rsnd_dma *dma) +{ + struct rsnd_mod *mod = rsnd_dma_to_mod(dma); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_substream *substream = io->substream; + struct device *dev = rsnd_priv_to_dev(priv); + struct dma_async_tx_descriptor *desc; + + desc = dmaengine_prep_dma_cyclic(dma->chan, + (dma->addr) ? dma->addr : + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + dma->dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + + if (!desc) { + dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } + + desc->callback = rsnd_dma_complete; + desc->callback_param = dma; + + if (dmaengine_submit(desc) < 0) { + dev_err(dev, "dmaengine_submit() fail\n"); + return; + } + + dma_async_issue_pending(dma->chan); +} + +static void rsnd_dma_of_path(struct rsnd_dma *dma, + int is_play, + struct rsnd_mod **mod_from, + struct rsnd_mod **mod_to); + +int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct dma_slave_config cfg = {}; + struct rsnd_mod *mod = rsnd_dma_to_mod(dma); + struct rsnd_mod *mod_from; + struct rsnd_mod *mod_to; + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + int is_play = rsnd_io_is_play(io); + char dma_name[DMA_NAME_SIZE]; + dma_cap_mask_t mask; + int ret; + + if (dma->chan) { + dev_err(dev, "it already has dma channel\n"); + return -EIO; + } + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + rsnd_dma_of_path(dma, is_play, &mod_from, &mod_to); + rsnd_dma_of_name(mod_from, mod_to, dma_name); + + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + cfg.src_addr = rsnd_gen_dma_addr(priv, mod_from, is_play, 1); + cfg.dst_addr = rsnd_gen_dma_addr(priv, mod_to, is_play, 0); + cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + + dev_dbg(dev, "dma : %s %pad -> %pad\n", + dma_name, &cfg.src_addr, &cfg.dst_addr); + + dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, + (void *)id, dev, + dma_name); + if (!dma->chan) { + dev_err(dev, "can't get dma channel\n"); + goto rsnd_dma_channel_err; + } + + ret = dmaengine_slave_config(dma->chan, &cfg); + if (ret < 0) + goto rsnd_dma_init_err; + + dma->addr = is_play ? cfg.src_addr : cfg.dst_addr; + dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + return 0; + +rsnd_dma_init_err: + rsnd_dma_quit(dma); +rsnd_dma_channel_err: + + /* + * DMA failed. try to PIO mode + * see + * rsnd_ssi_fallback() + * rsnd_rdai_continuance_probe() + */ + return -EAGAIN; +} + +void rsnd_dma_quit(struct rsnd_dma *dma) +{ + if (dma->chan) + dma_release_channel(dma->chan); + + dma->chan = NULL; +} + +static void rsnd_dma_of_path(struct rsnd_dma *dma, + int is_play, + struct rsnd_mod **mod_from, + struct rsnd_mod **mod_to) +{ + struct rsnd_mod *this = rsnd_dma_to_mod(dma); + struct rsnd_dai_stream *io = rsnd_mod_to_io(this); + struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *src = rsnd_io_to_mod_src(io); + struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); + struct rsnd_mod *mod[MOD_MAX]; + int i, index; + + + for (i = 0; i < MOD_MAX; i++) + mod[i] = NULL; + + /* + * in play case... + * + * src -> dst + * + * mem -> SSI + * mem -> SRC -> SSI + * mem -> SRC -> DVC -> SSI + */ + mod[0] = NULL; /* for "mem" */ + index = 1; + for (i = 1; i < MOD_MAX; i++) { + if (!src) { + mod[i] = ssi; + } else if (!dvc) { + mod[i] = src; + src = NULL; + } else { + if ((!is_play) && (this == src)) + this = dvc; + + mod[i] = (is_play) ? src : dvc; + i++; + mod[i] = (is_play) ? dvc : src; + src = NULL; + dvc = NULL; + } + + if (mod[i] == this) + index = i; + + if (mod[i] == ssi) + break; + } + + if (is_play) { + *mod_from = mod[index - 1]; + *mod_to = mod[index]; + } else { + *mod_from = mod[index]; + *mod_to = mod[index - 1]; + } +} + From 747c71b12ee8357e73a88eb25f569e2a20e80df3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:26:29 +0000 Subject: [PATCH 164/411] ASoC: rsnd: move rsnd_gen_dma_addr() from gen.c to dma.c Now, we have dma.c for Audio DMAC / Audio DMAC peri peri. rsnd_gen_dma_addr() should go there. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 120 ++++++++++++++++++++++++++++++++++++++- sound/soc/sh/rcar/gen.c | 110 ----------------------------------- sound/soc/sh/rcar/rsnd.h | 3 - 3 files changed, 117 insertions(+), 116 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 37acd409e88cf2..188b4634939c1a 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -32,7 +32,6 @@ static void rsnd_dma_complete(void *data) } #define DMA_NAME_SIZE 16 -#define MOD_MAX 4 /* MEM/SSI/SRC/DVC */ static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) { if (mod) @@ -97,6 +96,10 @@ static void rsnd_dma_of_path(struct rsnd_dma *dma, struct rsnd_mod **mod_from, struct rsnd_mod **mod_to); +static dma_addr_t rsnd_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from); + int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) { struct device *dev = rsnd_priv_to_dev(priv); @@ -122,8 +125,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) rsnd_dma_of_name(mod_from, mod_to, dma_name); cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - cfg.src_addr = rsnd_gen_dma_addr(priv, mod_from, is_play, 1); - cfg.dst_addr = rsnd_gen_dma_addr(priv, mod_to, is_play, 0); + cfg.src_addr = rsnd_dma_addr(priv, mod_from, is_play, 1); + cfg.dst_addr = rsnd_dma_addr(priv, mod_to, is_play, 0); cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; @@ -168,6 +171,117 @@ void rsnd_dma_quit(struct rsnd_dma *dma) dma->chan = NULL; } +/* + * DMA read/write register offset + * + * RSND_xxx_I_N for Audio DMAC input + * RSND_xxx_O_N for Audio DMAC output + * RSND_xxx_I_P for Audio DMAC peri peri input + * RSND_xxx_O_P for Audio DMAC peri peri output + * + * ex) R-Car H2 case + * mod / DMAC in / DMAC out / DMAC PP in / DMAC pp out + * SSI : 0xec541000 / 0xec241008 / 0xec24100c + * SSIU: 0xec541000 / 0xec100000 / 0xec100000 / 0xec400000 / 0xec400000 + * SCU : 0xec500000 / 0xec000000 / 0xec004000 / 0xec300000 / 0xec304000 + * CMD : 0xec500000 / / 0xec008000 0xec308000 + */ +#define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) +#define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) + +#define RDMA_SSIU_I_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) +#define RDMA_SSIU_O_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) + +#define RDMA_SSIU_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) +#define RDMA_SSIU_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) + +#define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) +#define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i)) + +#define RDMA_SRC_I_P(addr, i) (addr ##_reg - 0x00200000 + (0x400 * i)) +#define RDMA_SRC_O_P(addr, i) (addr ##_reg - 0x001fc000 + (0x400 * i)) + +#define RDMA_CMD_O_N(addr, i) (addr ##_reg - 0x004f8000 + (0x400 * i)) +#define RDMA_CMD_O_P(addr, i) (addr ##_reg - 0x001f8000 + (0x400 * i)) + +static dma_addr_t +rsnd_gen2_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI); + phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU); + int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod); + int use_src = !!rsnd_io_to_mod_src(io); + int use_dvc = !!rsnd_io_to_mod_dvc(io); + int id = rsnd_mod_id(mod); + struct dma_addr { + dma_addr_t out_addr; + dma_addr_t in_addr; + } dma_addrs[3][2][3] = { + /* SRC */ + {{{ 0, 0 }, + /* Capture */ + { RDMA_SRC_O_N(src, id), RDMA_SRC_I_P(src, id) }, + { RDMA_CMD_O_N(src, id), RDMA_SRC_I_P(src, id) } }, + /* Playback */ + {{ 0, 0, }, + { RDMA_SRC_O_P(src, id), RDMA_SRC_I_N(src, id) }, + { RDMA_CMD_O_P(src, id), RDMA_SRC_I_N(src, id) } } + }, + /* SSI */ + /* Capture */ + {{{ RDMA_SSI_O_N(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 } }, + /* Playback */ + {{ 0, RDMA_SSI_I_N(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) } } + }, + /* SSIU */ + /* Capture */ + {{{ RDMA_SSIU_O_N(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 }, + { RDMA_SSIU_O_P(ssi, id), 0 } }, + /* Playback */ + {{ 0, RDMA_SSIU_I_N(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) }, + { 0, RDMA_SSIU_I_P(ssi, id) } } }, + }; + + /* it shouldn't happen */ + if (use_dvc && !use_src) + dev_err(dev, "DVC is selected without SRC\n"); + + /* use SSIU or SSI ? */ + if (is_ssi && (0 == strcmp(rsnd_mod_dma_name(mod), "ssiu"))) + is_ssi++; + + return (is_from) ? + dma_addrs[is_ssi][is_play][use_src + use_dvc].out_addr : + dma_addrs[is_ssi][is_play][use_src + use_dvc].in_addr; +} + +static dma_addr_t rsnd_dma_addr(struct rsnd_priv *priv, + struct rsnd_mod *mod, + int is_play, int is_from) +{ + /* + * gen1 uses default DMA addr + */ + if (rsnd_is_gen1(priv)) + return 0; + + if (!mod) + return 0; + + return rsnd_gen2_dma_addr(priv, mod, is_play, is_from); +} + +#define MOD_MAX 4 /* MEM/SSI/SRC/DVC */ static void rsnd_dma_of_path(struct rsnd_dma *dma, int is_play, struct rsnd_mod **mod_from, diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 0da04ed3aabe11..0273724a266885 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -187,116 +187,6 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, return 0; } -/* - * DMA read/write register offset - * - * RSND_xxx_I_N for Audio DMAC input - * RSND_xxx_O_N for Audio DMAC output - * RSND_xxx_I_P for Audio DMAC peri peri input - * RSND_xxx_O_P for Audio DMAC peri peri output - * - * ex) R-Car H2 case - * mod / DMAC in / DMAC out / DMAC PP in / DMAC pp out - * SSI : 0xec541000 / 0xec241008 / 0xec24100c - * SSIU: 0xec541000 / 0xec100000 / 0xec100000 / 0xec400000 / 0xec400000 - * SCU : 0xec500000 / 0xec000000 / 0xec004000 / 0xec300000 / 0xec304000 - * CMD : 0xec500000 / / 0xec008000 0xec308000 - */ -#define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) -#define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) - -#define RDMA_SSIU_I_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) -#define RDMA_SSIU_O_N(addr, i) (addr ##_reg - 0x00441000 + (0x1000 * i)) - -#define RDMA_SSIU_I_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) -#define RDMA_SSIU_O_P(addr, i) (addr ##_reg - 0x00141000 + (0x1000 * i)) - -#define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) -#define RDMA_SRC_O_N(addr, i) (addr ##_reg - 0x004fc000 + (0x400 * i)) - -#define RDMA_SRC_I_P(addr, i) (addr ##_reg - 0x00200000 + (0x400 * i)) -#define RDMA_SRC_O_P(addr, i) (addr ##_reg - 0x001fc000 + (0x400 * i)) - -#define RDMA_CMD_O_N(addr, i) (addr ##_reg - 0x004f8000 + (0x400 * i)) -#define RDMA_CMD_O_P(addr, i) (addr ##_reg - 0x001f8000 + (0x400 * i)) - -static dma_addr_t -rsnd_gen2_dma_addr(struct rsnd_priv *priv, - struct rsnd_mod *mod, - int is_play, int is_from) -{ - struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI); - phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU); - int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod); - int use_src = !!rsnd_io_to_mod_src(io); - int use_dvc = !!rsnd_io_to_mod_dvc(io); - int id = rsnd_mod_id(mod); - struct dma_addr { - dma_addr_t out_addr; - dma_addr_t in_addr; - } dma_addrs[3][2][3] = { - /* SRC */ - {{{ 0, 0 }, - /* Capture */ - { RDMA_SRC_O_N(src, id), RDMA_SRC_I_P(src, id) }, - { RDMA_CMD_O_N(src, id), RDMA_SRC_I_P(src, id) } }, - /* Playback */ - {{ 0, 0, }, - { RDMA_SRC_O_P(src, id), RDMA_SRC_I_N(src, id) }, - { RDMA_CMD_O_P(src, id), RDMA_SRC_I_N(src, id) } } - }, - /* SSI */ - /* Capture */ - {{{ RDMA_SSI_O_N(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 } }, - /* Playback */ - {{ 0, RDMA_SSI_I_N(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) } } - }, - /* SSIU */ - /* Capture */ - {{{ RDMA_SSIU_O_N(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 }, - { RDMA_SSIU_O_P(ssi, id), 0 } }, - /* Playback */ - {{ 0, RDMA_SSIU_I_N(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) }, - { 0, RDMA_SSIU_I_P(ssi, id) } } }, - }; - - /* it shouldn't happen */ - if (use_dvc && !use_src) - dev_err(dev, "DVC is selected without SRC\n"); - - /* use SSIU or SSI ? */ - if (is_ssi && (0 == strcmp(rsnd_mod_dma_name(mod), "ssiu"))) - is_ssi++; - - return (is_from) ? - dma_addrs[is_ssi][is_play][use_src + use_dvc].out_addr : - dma_addrs[is_ssi][is_play][use_src + use_dvc].in_addr; -} - -dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv, - struct rsnd_mod *mod, - int is_play, int is_from) -{ - /* - * gen1 uses default DMA addr - */ - if (rsnd_is_gen1(priv)) - return 0; - - if (!mod) - return 0; - - return rsnd_gen2_dma_addr(priv, mod, is_play, is_from); -} - /* * Gen2 */ diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8a8a4d5d55efaf..a73e94c1d78523 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -328,9 +328,6 @@ int rsnd_gen_probe(struct platform_device *pdev, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -dma_addr_t rsnd_gen_dma_addr(struct rsnd_priv *priv, - struct rsnd_mod *mod, - int is_play, int is_from); phys_addr_t rsnd_gen_get_phy_addr(struct rsnd_priv *priv, int reg_id); #define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) From 3c68565b6cb68b731b51eb21b59dce901002fc6e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:27:12 +0000 Subject: [PATCH 165/411] ASoC: rsnd: enable to care 1st / 2nd DMAC on rsnd_dma_xxx() rsnd driver needs to care about Audio DAMC (via DMAEngine), Audio DMAC peri peri (via local method) on rsnd driver. This patch adds new rsnd_dma_ops and care it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 78 +++++++++++++++++++++++++++------------- sound/soc/sh/rcar/rsnd.h | 12 +++++++ 2 files changed, 65 insertions(+), 25 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 188b4634939c1a..c911c079fdd094 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -10,7 +10,7 @@ */ #include "rsnd.h" -static void rsnd_dma_complete(void *data) +static void rsnd_dmaen_complete(void *data) { struct rsnd_dma *dma = (struct rsnd_dma *)data; struct rsnd_mod *mod = rsnd_dma_to_mod(dma); @@ -32,7 +32,7 @@ static void rsnd_dma_complete(void *data) } #define DMA_NAME_SIZE 16 -static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) +static int _rsnd_dmaen_of_name(char *dma_name, struct rsnd_mod *mod) { if (mod) return snprintf(dma_name, DMA_NAME_SIZE / 2, "%s%d", @@ -42,23 +42,23 @@ static int _rsnd_dma_of_name(char *dma_name, struct rsnd_mod *mod) } -static void rsnd_dma_of_name(struct rsnd_mod *mod_from, +static void rsnd_dmaen_of_name(struct rsnd_mod *mod_from, struct rsnd_mod *mod_to, char *dma_name) { int index = 0; - index = _rsnd_dma_of_name(dma_name + index, mod_from); + index = _rsnd_dmaen_of_name(dma_name + index, mod_from); *(dma_name + index++) = '_'; - index = _rsnd_dma_of_name(dma_name + index, mod_to); + index = _rsnd_dmaen_of_name(dma_name + index, mod_to); } -void rsnd_dma_stop(struct rsnd_dma *dma) +static void rsnd_dmaen_stop(struct rsnd_dma *dma) { dmaengine_terminate_all(dma->chan); } -void rsnd_dma_start(struct rsnd_dma *dma) +static void rsnd_dmaen_start(struct rsnd_dma *dma) { struct rsnd_mod *mod = rsnd_dma_to_mod(dma); struct rsnd_priv *priv = rsnd_mod_to_priv(mod); @@ -80,7 +80,7 @@ void rsnd_dma_start(struct rsnd_dma *dma) return; } - desc->callback = rsnd_dma_complete; + desc->callback = rsnd_dmaen_complete; desc->callback_param = dma; if (dmaengine_submit(desc) < 0) { @@ -91,22 +91,12 @@ void rsnd_dma_start(struct rsnd_dma *dma) dma_async_issue_pending(dma->chan); } -static void rsnd_dma_of_path(struct rsnd_dma *dma, - int is_play, - struct rsnd_mod **mod_from, - struct rsnd_mod **mod_to); - -static dma_addr_t rsnd_dma_addr(struct rsnd_priv *priv, - struct rsnd_mod *mod, - int is_play, int is_from); - -int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) +static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, + struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) { struct device *dev = rsnd_priv_to_dev(priv); struct dma_slave_config cfg = {}; struct rsnd_mod *mod = rsnd_dma_to_mod(dma); - struct rsnd_mod *mod_from; - struct rsnd_mod *mod_to; struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); int is_play = rsnd_io_is_play(io); char dma_name[DMA_NAME_SIZE]; @@ -121,12 +111,11 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - rsnd_dma_of_path(dma, is_play, &mod_from, &mod_to); - rsnd_dma_of_name(mod_from, mod_to, dma_name); + rsnd_dmaen_of_name(mod_from, mod_to, dma_name); cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - cfg.src_addr = rsnd_dma_addr(priv, mod_from, is_play, 1); - cfg.dst_addr = rsnd_dma_addr(priv, mod_to, is_play, 0); + cfg.src_addr = dma->src_addr; + cfg.dst_addr = dma->dst_addr; cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; @@ -163,7 +152,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) return -EAGAIN; } -void rsnd_dma_quit(struct rsnd_dma *dma) +static void rsnd_dmaen_quit(struct rsnd_dma *dma) { if (dma->chan) dma_release_channel(dma->chan); @@ -171,6 +160,13 @@ void rsnd_dma_quit(struct rsnd_dma *dma) dma->chan = NULL; } +static struct rsnd_dma_ops rsnd_dmaen_ops = { + .start = rsnd_dmaen_start, + .stop = rsnd_dmaen_stop, + .init = rsnd_dmaen_init, + .quit = rsnd_dmaen_quit, +}; + /* * DMA read/write register offset * @@ -343,3 +339,35 @@ static void rsnd_dma_of_path(struct rsnd_dma *dma, } } +void rsnd_dma_stop(struct rsnd_dma *dma) +{ + dma->ops->stop(dma); +} + +void rsnd_dma_start(struct rsnd_dma *dma) +{ + dma->ops->start(dma); +} + +void rsnd_dma_quit(struct rsnd_dma *dma) +{ + dma->ops->quit(dma); +} + +int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) +{ + struct rsnd_mod *mod = rsnd_dma_to_mod(dma); + struct rsnd_mod *mod_from; + struct rsnd_mod *mod_to; + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + int is_play = rsnd_io_is_play(io); + + rsnd_dma_of_path(dma, is_play, &mod_from, &mod_to); + + dma->src_addr = rsnd_dma_addr(priv, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(priv, mod_to, is_play, 0); + + dma->ops = &rsnd_dmaen_ops; + + return dma->ops->init(priv, dma, id, mod_from, mod_to); +} diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index a73e94c1d78523..c7299f74cf8362 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -170,10 +170,22 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod); /* * R-Car DMA */ +struct rsnd_dma; +struct rsnd_dma_ops { + void (*start)(struct rsnd_dma *dma); + void (*stop)(struct rsnd_dma *dma); + int (*init)(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, + struct rsnd_mod *mod_from, struct rsnd_mod *mod_to); + void (*quit)(struct rsnd_dma *dma); +}; + struct rsnd_dma { struct dma_chan *chan; + struct rsnd_dma_ops *ops; enum dma_transfer_direction dir; dma_addr_t addr; + dma_addr_t src_addr; + dma_addr_t dst_addr; }; void rsnd_dma_start(struct rsnd_dma *dma); From 288f392e729dd4d3719c2319c7c3f8d4c820488b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:27:42 +0000 Subject: [PATCH 166/411] ASoC: rsnd: add Audio DMAC peri peri support rework Renesas R-Car sound (= rsnd) needs 2 DMAC which are called as Audio DMAC (= 1st DMAC) and Audio DMAC peri peri (2nd DMAC). And rsnd had assumed that 1st / 2nd DMACs are implemented as DMAEngine. But, in result of DMA ML discussion, 2nd DMAC was concluded that it is not a general purpose DMAC (2nd DMAC is for Device to Device inside sound system). Additionally, current DMAEngine can't support Device to Device, and we don't have correct DT bindings for it at this point. So the easiest solution for it is that move it from DMAEngine to rsnd driver. Audio DMAC peri peri is controlled from sound driver without DMAEngine by this patch. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 1 + sound/soc/sh/rcar/dma.c | 211 ++++++++++++++++++++++++++++++++++++++- sound/soc/sh/rcar/rsnd.h | 11 +- 3 files changed, 221 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 9beea9ba338a01..3b6e21948c7133 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -974,6 +974,7 @@ static int rsnd_probe(struct platform_device *pdev) const struct rsnd_of_data *of_data, struct rsnd_priv *priv) = { rsnd_gen_probe, + rsnd_dma_probe, rsnd_ssi_probe, rsnd_src_probe, rsnd_dvc_probe, diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index c911c079fdd094..a01bb8c6b06808 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -8,8 +8,29 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ +#include #include "rsnd.h" +/* + * Audio DMAC peri peri register + */ +#define PDMASAR 0x00 +#define PDMADAR 0x04 +#define PDMACHCR 0x0c + +/* PDMACHCR */ +#define PDMACHCR_DE (1 << 0) + +struct rsnd_dma_ctrl { + void __iomem *base; + int dmapp_num; +}; + +#define rsnd_priv_to_dmac(p) ((struct rsnd_dma_ctrl *)(p)->dma) + +/* + * Audio DMAC + */ static void rsnd_dmaen_complete(void *data) { struct rsnd_dma *dma = (struct rsnd_dma *)data; @@ -108,6 +129,8 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, return -EIO; } + dev_dbg(dev, "Audio DMAC init\n"); + dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); @@ -167,6 +190,150 @@ static struct rsnd_dma_ops rsnd_dmaen_ops = { .quit = rsnd_dmaen_quit, }; +/* + * Audio DMAC peri peri + */ +static const u8 gen2_id_table_ssiu[] = { + 0x00, /* SSI00 */ + 0x04, /* SSI10 */ + 0x08, /* SSI20 */ + 0x0c, /* SSI3 */ + 0x0d, /* SSI4 */ + 0x0e, /* SSI5 */ + 0x0f, /* SSI6 */ + 0x10, /* SSI7 */ + 0x11, /* SSI8 */ + 0x12, /* SSI90 */ +}; +static const u8 gen2_id_table_scu[] = { + 0x2d, /* SCU_SRCI0 */ + 0x2e, /* SCU_SRCI1 */ + 0x2f, /* SCU_SRCI2 */ + 0x30, /* SCU_SRCI3 */ + 0x31, /* SCU_SRCI4 */ + 0x32, /* SCU_SRCI5 */ + 0x33, /* SCU_SRCI6 */ + 0x34, /* SCU_SRCI7 */ + 0x35, /* SCU_SRCI8 */ + 0x36, /* SCU_SRCI9 */ +}; +static const u8 gen2_id_table_cmd[] = { + 0x37, /* SCU_CMD0 */ + 0x38, /* SCU_CMD1 */ +}; + +static u32 rsnd_dmapp_get_id(struct rsnd_mod *mod) +{ + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct rsnd_mod *ssi = rsnd_io_to_mod_ssi(io); + struct rsnd_mod *src = rsnd_io_to_mod_src(io); + struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); + const u8 *entry = NULL; + int id = rsnd_mod_id(mod); + int size = 0; + + if (mod == ssi) { + entry = gen2_id_table_ssiu; + size = ARRAY_SIZE(gen2_id_table_ssiu); + } else if (mod == src) { + entry = gen2_id_table_scu; + size = ARRAY_SIZE(gen2_id_table_scu); + } else if (mod == dvc) { + entry = gen2_id_table_cmd; + size = ARRAY_SIZE(gen2_id_table_cmd); + } + + if (!entry) + return 0xFF; + + if (size <= id) + return 0xFF; + + return entry[id]; +} + +static u32 rsnd_dmapp_get_chcr(struct rsnd_mod *mod_from, + struct rsnd_mod *mod_to) +{ + return (rsnd_dmapp_get_id(mod_from) << 24) + + (rsnd_dmapp_get_id(mod_to) << 16); +} + +#define rsnd_dmapp_addr(dmac, dma, reg) \ + (dmac->base + 0x20 + (0x10 * dma->dmapp_id) + reg) +static void rsnd_dmapp_write(struct rsnd_dma *dma, u32 data, u32 reg) +{ + struct rsnd_mod *mod = rsnd_dma_to_mod(dma); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "w %p : %08x\n", rsnd_dmapp_addr(dmac, dma, reg), data); + + iowrite32(data, rsnd_dmapp_addr(dmac, dma, reg)); +} + +static u32 rsnd_dmapp_read(struct rsnd_dma *dma, u32 reg) +{ + struct rsnd_mod *mod = rsnd_dma_to_mod(dma); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + + return ioread32(rsnd_dmapp_addr(dmac, dma, reg)); +} + +static void rsnd_dmapp_stop(struct rsnd_dma *dma) +{ + int i; + + rsnd_dmapp_write(dma, 0, PDMACHCR); + + for (i = 0; i < 1024; i++) { + if (0 == rsnd_dmapp_read(dma, PDMACHCR)) + return; + udelay(1); + } +} + +static void rsnd_dmapp_start(struct rsnd_dma *dma) +{ + rsnd_dmapp_write(dma, dma->src_addr, PDMASAR); + rsnd_dmapp_write(dma, dma->dst_addr, PDMADAR); + rsnd_dmapp_write(dma, dma->chcr, PDMACHCR); +} + +static int rsnd_dmapp_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, + struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) +{ + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + struct device *dev = rsnd_priv_to_dev(priv); + + dev_dbg(dev, "Audio DMAC peri peri init\n"); + + dma->dmapp_id = dmac->dmapp_num; + dma->chcr = rsnd_dmapp_get_chcr(mod_from, mod_to) | PDMACHCR_DE; + + dmac->dmapp_num++; + + rsnd_dmapp_stop(dma); + + dev_dbg(dev, "id/src/dst/chcr = %d/%x/%x/%08x\n", + dma->dmapp_id, dma->src_addr, dma->dst_addr, dma->chcr); + + return 0; +} + +static struct rsnd_dma_ops rsnd_dmapp_ops = { + .start = rsnd_dmapp_start, + .stop = rsnd_dmapp_stop, + .init = rsnd_dmapp_init, + .quit = rsnd_dmapp_stop, +}; + +/* + * Common DMAC Interface + */ + /* * DMA read/write register offset * @@ -367,7 +534,49 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) dma->src_addr = rsnd_dma_addr(priv, mod_from, is_play, 1); dma->dst_addr = rsnd_dma_addr(priv, mod_to, is_play, 0); - dma->ops = &rsnd_dmaen_ops; + /* for Gen2 */ + if (mod_from && mod_to) + dma->ops = &rsnd_dmapp_ops; + else + dma->ops = &rsnd_dmaen_ops; + + /* for Gen1, overwrite */ + if (rsnd_is_gen1(priv)) + dma->ops = &rsnd_dmaen_ops; return dma->ops->init(priv, dma, id, mod_from, mod_to); } + +int rsnd_dma_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv) +{ + struct device *dev = rsnd_priv_to_dev(priv); + struct rsnd_dma_ctrl *dmac; + struct resource *res; + + /* + * for Gen1 + */ + if (rsnd_is_gen1(priv)) + return 0; + + /* + * for Gen2 + */ + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "audmapp"); + dmac = devm_kzalloc(dev, sizeof(*dmac), GFP_KERNEL); + if (!dmac || !res) { + dev_err(dev, "dma allocate failed\n"); + return -ENOMEM; + } + + dmac->dmapp_num = 0; + dmac->base = devm_ioremap_resource(dev, res); + if (IS_ERR(dmac->base)) + return PTR_ERR(dmac->base); + + priv->dma = dmac; + + return 0; +} diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index c7299f74cf8362..9e67142c82bd20 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -184,6 +184,8 @@ struct rsnd_dma { struct rsnd_dma_ops *ops; enum dma_transfer_direction dir; dma_addr_t addr; + int dmapp_id; + u32 chcr; dma_addr_t src_addr; dma_addr_t dst_addr; }; @@ -192,7 +194,9 @@ void rsnd_dma_start(struct rsnd_dma *dma); void rsnd_dma_stop(struct rsnd_dma *dma); int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id); void rsnd_dma_quit(struct rsnd_dma *dma); - +int rsnd_dma_probe(struct platform_device *pdev, + const struct rsnd_of_data *of_data, + struct rsnd_priv *priv); #define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) @@ -395,6 +399,11 @@ struct rsnd_priv { */ void *adg; + /* + * below value will be filled on rsnd_dma_probe() + */ + void *dma; + /* * below value will be filled on rsnd_ssi_probe() */ From 56f2906ae2d0b48b64a67feef99e3be3b40c3617 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:28:07 +0000 Subject: [PATCH 167/411] ASoC: rsnd: remove rsnd_dma::addr DMAEngine for Renesas R-Car driver is used only for Audio DMAC now. rsnd_dma::addr was added to support Audio DMAC peri peri, but it is no longer needed. Let's remove it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 2 -- sound/soc/sh/rcar/rsnd.h | 1 - 2 files changed, 3 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index a01bb8c6b06808..c407fd250d2a29 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -89,7 +89,6 @@ static void rsnd_dmaen_start(struct rsnd_dma *dma) struct dma_async_tx_descriptor *desc; desc = dmaengine_prep_dma_cyclic(dma->chan, - (dma->addr) ? dma->addr : substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), @@ -157,7 +156,6 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, if (ret < 0) goto rsnd_dma_init_err; - dma->addr = is_play ? cfg.src_addr : cfg.dst_addr; dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; return 0; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9e67142c82bd20..a2954917bfcb8f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -183,7 +183,6 @@ struct rsnd_dma { struct dma_chan *chan; struct rsnd_dma_ops *ops; enum dma_transfer_direction dir; - dma_addr_t addr; int dmapp_id; u32 chcr; dma_addr_t src_addr; From aaf4fce019ecd55a2d93b6111525deeef300f751 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:28:32 +0000 Subject: [PATCH 168/411] ASoC: rsnd: remove rsnd_dma::dir DMAEngine direction can be calculated from rsnd_dai_stream, So, rsnd_dma::dir does not make sense now. Let's remove it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 5 ++--- sound/soc/sh/rcar/rsnd.h | 1 - 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index c407fd250d2a29..3f1ea58ee14471 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -87,12 +87,13 @@ static void rsnd_dmaen_start(struct rsnd_dma *dma) struct snd_pcm_substream *substream = io->substream; struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; + int is_play = rsnd_io_is_play(io); desc = dmaengine_prep_dma_cyclic(dma->chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), - dma->dir, + is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { @@ -156,8 +157,6 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, if (ret < 0) goto rsnd_dma_init_err; - dma->dir = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; - return 0; rsnd_dma_init_err: diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index a2954917bfcb8f..5d65a4b96743dc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -182,7 +182,6 @@ struct rsnd_dma_ops { struct rsnd_dma { struct dma_chan *chan; struct rsnd_dma_ops *ops; - enum dma_transfer_direction dir; int dmapp_id; u32 chcr; dma_addr_t src_addr; From 0d00a52182be985bfae67d407ee81fefe448a0fd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:29:13 +0000 Subject: [PATCH 169/411] ASoC: rsnd: use union with rsnd_dmaen / rsnd_dmapp Renesas R-Car needs 2 DMACs.1st DMAC is DMAEngine, and 2nd DMAC is implemented as local code. These 2 DMACs are never shared. We can use union for these. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 40 +++++++++++++++++++++++++--------------- sound/soc/sh/rcar/rsnd.h | 16 ++++++++++++++-- 2 files changed, 39 insertions(+), 17 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 3f1ea58ee14471..b449763ebd4348 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -76,11 +76,14 @@ static void rsnd_dmaen_of_name(struct rsnd_mod *mod_from, static void rsnd_dmaen_stop(struct rsnd_dma *dma) { - dmaengine_terminate_all(dma->chan); + struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); + + dmaengine_terminate_all(dmaen->chan); } static void rsnd_dmaen_start(struct rsnd_dma *dma) { + struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); struct rsnd_mod *mod = rsnd_dma_to_mod(dma); struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); @@ -89,7 +92,7 @@ static void rsnd_dmaen_start(struct rsnd_dma *dma) struct dma_async_tx_descriptor *desc; int is_play = rsnd_io_is_play(io); - desc = dmaengine_prep_dma_cyclic(dma->chan, + desc = dmaengine_prep_dma_cyclic(dmaen->chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), @@ -109,12 +112,13 @@ static void rsnd_dmaen_start(struct rsnd_dma *dma) return; } - dma_async_issue_pending(dma->chan); + dma_async_issue_pending(dmaen->chan); } static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) { + struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); struct device *dev = rsnd_priv_to_dev(priv); struct dma_slave_config cfg = {}; struct rsnd_mod *mod = rsnd_dma_to_mod(dma); @@ -124,7 +128,7 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, dma_cap_mask_t mask; int ret; - if (dma->chan) { + if (dmaen->chan) { dev_err(dev, "it already has dma channel\n"); return -EIO; } @@ -145,15 +149,15 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, dev_dbg(dev, "dma : %s %pad -> %pad\n", dma_name, &cfg.src_addr, &cfg.dst_addr); - dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, + dmaen->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, (void *)id, dev, dma_name); - if (!dma->chan) { + if (!dmaen->chan) { dev_err(dev, "can't get dma channel\n"); goto rsnd_dma_channel_err; } - ret = dmaengine_slave_config(dma->chan, &cfg); + ret = dmaengine_slave_config(dmaen->chan, &cfg); if (ret < 0) goto rsnd_dma_init_err; @@ -174,10 +178,12 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, static void rsnd_dmaen_quit(struct rsnd_dma *dma) { - if (dma->chan) - dma_release_channel(dma->chan); + struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); + + if (dmaen->chan) + dma_release_channel(dmaen->chan); - dma->chan = NULL; + dmaen->chan = NULL; } static struct rsnd_dma_ops rsnd_dmaen_ops = { @@ -257,7 +263,8 @@ static u32 rsnd_dmapp_get_chcr(struct rsnd_mod *mod_from, } #define rsnd_dmapp_addr(dmac, dma, reg) \ - (dmac->base + 0x20 + (0x10 * dma->dmapp_id) + reg) + (dmac->base + 0x20 + reg + \ + (0x10 * rsnd_dma_to_dmapp(dma)->dmapp_id)) static void rsnd_dmapp_write(struct rsnd_dma *dma, u32 data, u32 reg) { struct rsnd_mod *mod = rsnd_dma_to_mod(dma); @@ -294,28 +301,31 @@ static void rsnd_dmapp_stop(struct rsnd_dma *dma) static void rsnd_dmapp_start(struct rsnd_dma *dma) { + struct rsnd_dmapp *dmapp = rsnd_dma_to_dmapp(dma); + rsnd_dmapp_write(dma, dma->src_addr, PDMASAR); rsnd_dmapp_write(dma, dma->dst_addr, PDMADAR); - rsnd_dmapp_write(dma, dma->chcr, PDMACHCR); + rsnd_dmapp_write(dma, dmapp->chcr, PDMACHCR); } static int rsnd_dmapp_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) { + struct rsnd_dmapp *dmapp = rsnd_dma_to_dmapp(dma); struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); struct device *dev = rsnd_priv_to_dev(priv); dev_dbg(dev, "Audio DMAC peri peri init\n"); - dma->dmapp_id = dmac->dmapp_num; - dma->chcr = rsnd_dmapp_get_chcr(mod_from, mod_to) | PDMACHCR_DE; + dmapp->dmapp_id = dmac->dmapp_num; + dmapp->chcr = rsnd_dmapp_get_chcr(mod_from, mod_to) | PDMACHCR_DE; dmac->dmapp_num++; rsnd_dmapp_stop(dma); dev_dbg(dev, "id/src/dst/chcr = %d/%x/%x/%08x\n", - dma->dmapp_id, dma->src_addr, dma->dst_addr, dma->chcr); + dmapp->dmapp_id, dma->src_addr, dma->dst_addr, dmapp->chcr); return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 5d65a4b96743dc..0d36e38ebbcfab 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -179,14 +179,26 @@ struct rsnd_dma_ops { void (*quit)(struct rsnd_dma *dma); }; -struct rsnd_dma { +struct rsnd_dmaen { struct dma_chan *chan; - struct rsnd_dma_ops *ops; +}; + +struct rsnd_dmapp { int dmapp_id; u32 chcr; +}; + +struct rsnd_dma { + struct rsnd_dma_ops *ops; dma_addr_t src_addr; dma_addr_t dst_addr; + union { + struct rsnd_dmaen en; + struct rsnd_dmapp pp; + } dma; }; +#define rsnd_dma_to_dmaen(dma) (&(dma)->dma.en) +#define rsnd_dma_to_dmapp(dma) (&(dma)->dma.pp) void rsnd_dma_start(struct rsnd_dma *dma); void rsnd_dma_stop(struct rsnd_dma *dma); From e879a9ddf41c47ccc83039e99e04b0d5c56cd0c5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:29:41 +0000 Subject: [PATCH 170/411] ASoC: rsnd: enable rsnd_ssi_use_busif() for DMA Renesas R-Car sound driver uses SSI, but the DMA interfaces are SSI/SSIU. This interface is based on SSI/SRC/DVC connection. And DMA function needs to know which interface is used somehow. This patch enables rsnd_ssi_use_busif() for DMA. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 2 +- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 +- 3 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index b449763ebd4348..3f34461da1e0e0 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -427,7 +427,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv, dev_err(dev, "DVC is selected without SRC\n"); /* use SSIU or SSI ? */ - if (is_ssi && (0 == strcmp(rsnd_mod_dma_name(mod), "ssiu"))) + if (is_ssi && rsnd_ssi_use_busif(mod)) is_ssi++; return (is_from) ? diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0d36e38ebbcfab..68bc3f46d70beb 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -514,6 +514,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod); int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); +int rsnd_ssi_use_busif(struct rsnd_mod *mod); /* * R-Car DVC diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index a0d902ad5985ea..7e48d562dea87b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -84,7 +84,7 @@ struct rsnd_ssi { #define rsnd_ssi_mode_flags(p) ((p)->info->flags) #define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id) -static int rsnd_ssi_use_busif(struct rsnd_mod *mod) +int rsnd_ssi_use_busif(struct rsnd_mod *mod) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); From 04e627baa68a8dc42f19b68e1b46d1c6aecebfd9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:30:02 +0000 Subject: [PATCH 171/411] ASoC: rsnd: ssi: add rsnd_ssi_of_node() This patch adds rsnd_ssi_of_node() to get SSI subnode from DT. This is prepare for new DT bindings for 1st DMAC Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 7e48d562dea87b..2133eb34ed0662 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -83,6 +83,8 @@ struct rsnd_ssi { #define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent) #define rsnd_ssi_mode_flags(p) ((p)->info->flags) #define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id) +#define rsnd_ssi_of_node(priv) \ + of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,ssi") int rsnd_ssi_use_busif(struct rsnd_mod *mod) { @@ -633,7 +635,7 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, if (!of_data) return; - node = of_get_child_by_name(dev->of_node, "rcar_sound,ssi"); + node = rsnd_ssi_of_node(priv); if (!node) return; From 82e76ed38edbdb338d64f5f2486fcd1482c8859a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:30:22 +0000 Subject: [PATCH 172/411] ASoC: rsnd: src: add rsnd_src_of_node() This patch adds rsnd_src_of_node() to get SRC subnode from DT. This is prepare for new DT bindings for 1st DMAC Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e2792056ce24e0..5a601bed415457 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -28,6 +28,9 @@ struct rsnd_src { #define RSND_SRC_NAME_SIZE 16 #define rsnd_src_convert_rate(p) ((p)->info->convert_rate) +#define rsnd_src_of_node(priv) \ + of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,src") + #define rsnd_mod_to_src(_mod) \ container_of((_mod), struct rsnd_src, mod) @@ -807,7 +810,7 @@ static void rsnd_of_parse_src(struct platform_device *pdev, if (!of_data) return; - src_node = of_get_child_by_name(dev->of_node, "rcar_sound,src"); + src_node = rsnd_src_of_node(priv); if (!src_node) return; From 93b986e246248d0587acb4f073a621179a16b763 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:30:41 +0000 Subject: [PATCH 173/411] ASoC: rsnd: dvc: add rsnd_dvc_of_node() This patch adds rsnd_dvc_of_node() to get DVC subnode from DT. This is prepare for new DT bindings for 1st DMAC Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index d7f9ed959c4e20..e0990180e1ea2b 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -24,6 +24,9 @@ struct rsnd_dvc { struct rsnd_kctrl_cfg_s rdown; /* Ramp Rate Down */ }; +#define rsnd_dvc_of_node(priv) \ + of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,dvc") + #define rsnd_mod_to_dvc(_mod) \ container_of((_mod), struct rsnd_dvc, mod) From 72adc61f4637aa3596c1db1129f84d768475a885 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:31:23 +0000 Subject: [PATCH 174/411] ASoC: rsnd: 1st DMAC dma-names cares subnode Renesas R-Car sound (= rsnd) needs 2 DMAC which are called as Audio DMAC (= 1st DMAC) and Audio DMAC peri peri (2nd DMAC). And rsnd had assumed that 1st / 2nd DMACs are implemented as DMAEngine. But, in result of DMA ML discussion, 2nd DMAC was concluded that it is not a general purpose DMAC (2nd DMAC is for Device to Device inside sound system). Additionally, current DMAEngine can't support Device to Device, and we don't have correct DT bindings for it at this point. So the easiest solution for it is that move it from DMAEngine to rsnd driver. dma-names on DT was implemented as no difference between 1st / 2nd DMAC's, since rsnd had assumed that both DMACs are implemented as DMAEngine. That style was "src_dst". But now, 2nd DMAC was implemented as non DMAEngine, and it doesn't need dma-names anymore. So, this dma-names rule is no longer needed. And additionally, dma-names was assumed that it has all (= SSI/SSIU/SRC/DVC) nodes under sound node. In upstream code, no SoC/platform is supporting DMA for rsnd driver yet. This means there is no compatible issue if this patch changes dma-names's rule of DT. This patch assumes dma-names for 1st DMAC are tx/rx base, and listed in each SSI/SRC/DVC subnode ex) rcar_sound,dvc { dvc0: dvc@0 { dmas = <&audma0 0xbc>; dma-names = "tx"; }; ... rcar_sound,src { src0: src@0 { ... dmas = <&audma0 0x85>, <&audma1 0x9a>; dma-names = "rx", "tx"; }; ... rcar_sound,ssi { ssi0: ssi@0 { ... dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; dma-names = "rx", "tx", "rxu", "txu"; }; ... Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 11 ++--- sound/soc/sh/rcar/dma.c | 88 +++++++++++++++++++++++----------------- sound/soc/sh/rcar/dvc.c | 9 ++++ sound/soc/sh/rcar/rsnd.h | 6 ++- sound/soc/sh/rcar/src.c | 13 ++++++ sound/soc/sh/rcar/ssi.c | 17 ++++++-- 6 files changed, 95 insertions(+), 49 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 3b6e21948c7133..7b995f025e2276 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -137,15 +137,12 @@ char *rsnd_mod_name(struct rsnd_mod *mod) return mod->ops->name; } -char *rsnd_mod_dma_name(struct rsnd_mod *mod) +struct dma_chan *rsnd_mod_dma_req(struct rsnd_mod *mod) { - if (!mod || !mod->ops) - return "unknown"; - - if (!mod->ops->dma_name) - return mod->ops->name; + if (!mod || !mod->ops || !mod->ops->dma_req) + return NULL; - return mod->ops->dma_name(mod); + return mod->ops->dma_req(mod); } void rsnd_mod_init(struct rsnd_mod *mod, diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 3f34461da1e0e0..92fd55044ee6b4 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -9,6 +9,7 @@ * published by the Free Software Foundation. */ #include +#include #include "rsnd.h" /* @@ -52,28 +53,6 @@ static void rsnd_dmaen_complete(void *data) rsnd_dai_pointer_update(io, io->byte_per_period); } -#define DMA_NAME_SIZE 16 -static int _rsnd_dmaen_of_name(char *dma_name, struct rsnd_mod *mod) -{ - if (mod) - return snprintf(dma_name, DMA_NAME_SIZE / 2, "%s%d", - rsnd_mod_dma_name(mod), rsnd_mod_id(mod)); - else - return snprintf(dma_name, DMA_NAME_SIZE / 2, "mem"); - -} - -static void rsnd_dmaen_of_name(struct rsnd_mod *mod_from, - struct rsnd_mod *mod_to, - char *dma_name) -{ - int index = 0; - - index = _rsnd_dmaen_of_name(dma_name + index, mod_from); - *(dma_name + index++) = '_'; - index = _rsnd_dmaen_of_name(dma_name + index, mod_to); -} - static void rsnd_dmaen_stop(struct rsnd_dma *dma) { struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); @@ -115,6 +94,40 @@ static void rsnd_dmaen_start(struct rsnd_dma *dma) dma_async_issue_pending(dmaen->chan); } +struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, + struct rsnd_mod *mod, char *name) +{ + struct dma_chan *chan; + struct device_node *np; + int i = 0; + + for_each_child_of_node(of_node, np) { + if (i == rsnd_mod_id(mod)) + break; + i++; + } + + chan = of_dma_request_slave_channel(np, name); + + of_node_put(np); + of_node_put(of_node); + + return chan; +} + +static struct dma_chan *rsnd_dmaen_request_channel(struct rsnd_mod *mod_from, + struct rsnd_mod *mod_to) +{ + if ((!mod_from && !mod_to) || + (mod_from && mod_to)) + return NULL; + + if (mod_from) + return rsnd_mod_dma_req(mod_from); + else + return rsnd_mod_dma_req(mod_to); +} + static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) { @@ -124,8 +137,6 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, struct rsnd_mod *mod = rsnd_dma_to_mod(dma); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); int is_play = rsnd_io_is_play(io); - char dma_name[DMA_NAME_SIZE]; - dma_cap_mask_t mask; int ret; if (dmaen->chan) { @@ -135,10 +146,21 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, dev_dbg(dev, "Audio DMAC init\n"); - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + if (dev->of_node) { + dmaen->chan = rsnd_dmaen_request_channel(mod_from, mod_to); + } else { + dma_cap_mask_t mask; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - rsnd_dmaen_of_name(mod_from, mod_to, dma_name); + dmaen->chan = dma_request_channel(mask, shdma_chan_filter, + (void *)id); + } + if (IS_ERR_OR_NULL(dmaen->chan)) { + dev_err(dev, "can't get dma channel\n"); + goto rsnd_dma_channel_err; + } cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; cfg.src_addr = dma->src_addr; @@ -146,16 +168,8 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - dev_dbg(dev, "dma : %s %pad -> %pad\n", - dma_name, &cfg.src_addr, &cfg.dst_addr); - - dmaen->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, - (void *)id, dev, - dma_name); - if (!dmaen->chan) { - dev_err(dev, "can't get dma channel\n"); - goto rsnd_dma_channel_err; - } + dev_dbg(dev, "dma : %pad -> %pad\n", + &cfg.src_addr, &cfg.dst_addr); ret = dmaengine_slave_config(dmaen->chan, &cfg); if (ret < 0) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index e0990180e1ea2b..aeeef1352eeee8 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -272,8 +272,17 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, return 0; } +static struct dma_chan *rsnd_dvc_dma_req(struct rsnd_mod *mod) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + + return rsnd_dma_request_channel(rsnd_dvc_of_node(priv), + mod, "tx"); +} + static struct rsnd_mod_ops rsnd_dvc_ops = { .name = DVC_NAME, + .dma_req = rsnd_dvc_dma_req, .probe = rsnd_dvc_probe_gen2, .remove = rsnd_dvc_remove_gen2, .init = rsnd_dvc_init, diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 68bc3f46d70beb..52c401c9eeef62 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -207,6 +207,8 @@ void rsnd_dma_quit(struct rsnd_dma *dma); int rsnd_dma_probe(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv); +struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, + struct rsnd_mod *mod, char *name); #define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) @@ -222,7 +224,7 @@ enum rsnd_mod_type { struct rsnd_mod_ops { char *name; - char* (*dma_name)(struct rsnd_mod *mod); + struct dma_chan* (*dma_req)(struct rsnd_mod *mod); int (*probe)(struct rsnd_mod *mod, struct rsnd_priv *priv); int (*remove)(struct rsnd_mod *mod, @@ -292,7 +294,7 @@ void rsnd_mod_init(struct rsnd_mod *mod, enum rsnd_mod_type type, int id); char *rsnd_mod_name(struct rsnd_mod *mod); -char *rsnd_mod_dma_name(struct rsnd_mod *mod); +struct dma_chan *rsnd_mod_dma_req(struct rsnd_mod *mod); /* * R-Car sound DAI diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 5a601bed415457..6ce8985757c12b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -114,6 +114,17 @@ struct rsnd_src { /* * Gen1/Gen2 common functions */ +static struct dma_chan *rsnd_src_dma_req(struct rsnd_mod *mod) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + int is_play = rsnd_io_is_play(io); + + return rsnd_dma_request_channel(rsnd_src_of_node(priv), + mod, + is_play ? "rx" : "tx"); +} + int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, int use_busif) { @@ -506,6 +517,7 @@ static int rsnd_src_stop_gen1(struct rsnd_mod *mod, static struct rsnd_mod_ops rsnd_src_gen1_ops = { .name = SRC_NAME, + .dma_req = rsnd_src_dma_req, .probe = rsnd_src_probe_gen1, .init = rsnd_src_init_gen1, .quit = rsnd_src_quit, @@ -780,6 +792,7 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, static struct rsnd_mod_ops rsnd_src_gen2_ops = { .name = SRC_NAME, + .dma_req = rsnd_src_dma_req, .probe = rsnd_src_probe_gen2, .remove = rsnd_src_remove_gen2, .init = rsnd_src_init_gen2, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2133eb34ed0662..fea4aa53918ab8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -553,14 +553,25 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, return 0; } -static char *rsnd_ssi_dma_name(struct rsnd_mod *mod) +static struct dma_chan *rsnd_ssi_dma_req(struct rsnd_mod *mod) { - return rsnd_ssi_use_busif(mod) ? "ssiu" : SSI_NAME; + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + int is_play = rsnd_io_is_play(io); + char *name; + + if (rsnd_ssi_use_busif(mod)) + name = is_play ? "rxu" : "txu"; + else + name = is_play ? "rx" : "tx"; + + return rsnd_dma_request_channel(rsnd_ssi_of_node(priv), + mod, name); } static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .name = SSI_NAME, - .dma_name = rsnd_ssi_dma_name, + .dma_req = rsnd_ssi_dma_req, .probe = rsnd_ssi_dma_probe, .remove = rsnd_ssi_dma_remove, .init = rsnd_ssi_init, From 5cf4f68672856dcbca883b460aee4ee5bfbeafb0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:31:55 +0000 Subject: [PATCH 175/411] ASoC: rsnd: add sample code of rcar_sound,src irq cfcefe01265cbaf5ca7209226d043b07bfa8b587 (ASoC: rsnd: add recovery support for under/over flow error on SRC) added SRC recovery support which needs IRQ. This patch adds SRC irq sample code on DT bingdings text Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/renesas,rsnd.txt | 40 ++++++++++++++----- 1 file changed, 30 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 2dd690bc19cc64..87e0fc2ce399f5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -53,16 +53,36 @@ rcar_sound: rcar_sound@ec500000 { }; rcar_sound,src { - src0: src@0 { }; - src1: src@1 { }; - src2: src@2 { }; - src3: src@3 { }; - src4: src@4 { }; - src5: src@5 { }; - src6: src@6 { }; - src7: src@7 { }; - src8: src@8 { }; - src9: src@9 { }; + src0: src@0 { + interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; + }; + src1: src@1 { + interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; + }; + src2: src@2 { + interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; + }; + src3: src@3 { + interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; + }; + src4: src@4 { + interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; + }; + src5: src@5 { + interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; + }; + src6: src@6 { + interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; + }; + src7: src@7 { + interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; + }; + src8: src@8 { + interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; + }; + src9: src@9 { + interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; + }; }; rcar_sound,ssi { From bb02714fe5f198c0f07a8039ebcc636c1bffbc03 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:32:15 +0000 Subject: [PATCH 176/411] ASoC: rsnd: add sample code of missing clocks Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/renesas,rsnd.txt | 21 +++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 87e0fc2ce399f5..503967ba39dba3 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -47,6 +47,27 @@ rcar_sound: rcar_sound@ec500000 { <0 0xec540000 0 0x1000>, /* SSIU */ <0 0xec541000 0 0x1280>; /* SSI */ + clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, + <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, + <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, + <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, + <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, + <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, + <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, + <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, + <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, + <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, + <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, + <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, + <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; + clock-names = "ssi-all", + "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", + "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", + "src.9", "src.8", "src.7", "src.6", "src.5", + "src.4", "src.3", "src.2", "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", "clk_c", "clk_i"; + rcar_sound,dvc { dvc0: dvc@0 { }; dvc1: dvc@1 { }; From e80a2fb18bc7dca116ec623dc909f07c5e505349 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:32:49 +0000 Subject: [PATCH 177/411] ASoC: rsnd: add sample code of dma entry rsnd driver supports Audio DMAC (via DMAEngine) / Audio DMAC peri peri (via rsnd driver) now. This patch adds DT binding sample code on DT binging text Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/renesas,rsnd.txt | 60 ++++++++++++++++++- 1 file changed, 57 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 503967ba39dba3..6e660d4f3631bc 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -29,9 +29,17 @@ SSI subnode properties: - shared-pin : if shared clock pin - pio-transfer : use PIO transfer mode - no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case +- dma : Should contain Audio DMAC entry +- dma-names : SSI case "rx" (=playback), "tx" (=capture) + SSIU case "rxu" (=playback), "txu" (=capture) SRC subnode properties: -no properties at this point +- dma : Should contain Audio DMAC entry +- dma-names : "rx" (=playback), "tx" (=capture) + +DVC subnode properties: +- dma : Should contain Audio DMAC entry +- dma-names : "tx" (=playback/capture) DAI subnode properties: - playback : list of playback modules @@ -69,73 +77,119 @@ rcar_sound: rcar_sound@ec500000 { "clk_a", "clk_b", "clk_c", "clk_i"; rcar_sound,dvc { - dvc0: dvc@0 { }; - dvc1: dvc@1 { }; + dvc0: dvc@0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc@1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; }; rcar_sound,src { src0: src@0 { interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x85>, <&audma1 0x9a>; + dma-names = "rx", "tx"; }; src1: src@1 { interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; }; src2: src@2 { interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x89>, <&audma1 0x9e>; + dma-names = "rx", "tx"; }; src3: src@3 { interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8b>, <&audma1 0xa0>; + dma-names = "rx", "tx"; }; src4: src@4 { interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8d>, <&audma1 0xb0>; + dma-names = "rx", "tx"; }; src5: src@5 { interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x8f>, <&audma1 0xb2>; + dma-names = "rx", "tx"; }; src6: src@6 { interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x91>, <&audma1 0xb4>; + dma-names = "rx", "tx"; }; src7: src@7 { interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x93>, <&audma1 0xb6>; + dma-names = "rx", "tx"; }; src8: src@8 { interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x95>, <&audma1 0xb8>; + dma-names = "rx", "tx"; }; src9: src@9 { interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x97>, <&audma1 0xba>; + dma-names = "rx", "tx"; }; }; rcar_sound,ssi { ssi0: ssi@0 { interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi1: ssi@1 { interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi2: ssi@2 { interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi3: ssi@3 { interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi4: ssi@4 { interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi5: ssi@5 { interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi6: ssi@6 { interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi7: ssi@7 { interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi8: ssi@8 { interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; + dma-names = "rx", "tx", "rxu", "txu"; }; ssi9: ssi@9 { interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; + dma-names = "rx", "tx", "rxu", "txu"; }; }; From d3b1c0badbc86c034af55704ea88c1e656b402fe Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 20 Feb 2015 10:33:09 +0000 Subject: [PATCH 178/411] ASoC: rsnd: add sample code of reg-names Current rsnd driver supports reg-names, and Audio DMAC peri peri. This patch adds sample code for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 6e660d4f3631bc..f316ce1f214a03 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -53,7 +53,9 @@ rcar_sound: rcar_sound@ec500000 { reg = <0 0xec500000 0 0x1000>, /* SCU */ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>; /* SSI */ + <0 0xec541000 0 0x1280>, /* SSI */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, From 778952598e53f09251bebe1655177b355cd9e836 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 7 Mar 2015 19:35:08 +0100 Subject: [PATCH 179/411] ASoC: Remove unnecessary device_remove_file() Since commit d29697dc3b92 ("ASoC: Add sysfs entries via static attribute groups") the sysfs attributes of the rtd are manged by the device core and there is no need to manually call device_remove_file() anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca5bacb98..acf99f1250e5f2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1410,7 +1410,6 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } From 967beb2e87771411e08467152e3d9f1c3ae73a67 Mon Sep 17 00:00:00 2001 From: Zubair Lutfullah Kakakhel Date: Mon, 9 Mar 2015 12:11:08 +0000 Subject: [PATCH 180/411] ASoC: jz4740: Add jz4780 support The jz4780 and jz4740 have very similar i2s blocks. The slight difference is in Rx/Tx fifos. And the bitclocks for input/output are different. This patch adds jz4780 support to the driver Signed-off-by: Zubair Lutfullah Kakakhel Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- .../bindings/sound/ingenic,jz4740-i2s.txt | 2 +- sound/soc/jz4740/jz4740-i2s.c | 84 ++++++++++++++++--- 2 files changed, 75 insertions(+), 11 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt index b41433386e2fe5..b623d50004fb01 100644 --- a/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt @@ -1,7 +1,7 @@ Ingenic JZ4740 I2S controller Required properties: -- compatible : "ingenic,jz4740-i2s" +- compatible : "ingenic,jz4740-i2s" or "ingenic,jz4780-i2s" - reg : I2S registers location and length - clocks : AIC and I2S PLL clock specifiers. - clock-names: "aic" and "i2s" diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 07f77815a586fe..b05fb1c1a8483e 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -58,6 +58,12 @@ #define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12 #define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8 +#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24 +#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16 +#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_MASK \ + (0xf << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) +#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_MASK \ + (0x1f << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) #define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19) #define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16) @@ -79,6 +85,7 @@ #define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET 16 #define JZ_AIC_I2S_FMT_DISABLE_BIT_CLK BIT(12) +#define JZ_AIC_I2S_FMT_DISABLE_BIT_ICLK BIT(13) #define JZ_AIC_I2S_FMT_ENABLE_SYS_CLK BIT(4) #define JZ_AIC_I2S_FMT_MSB BIT(0) @@ -87,6 +94,13 @@ #define JZ_AIC_CLK_DIV_MASK 0xf #define I2SDIV_DV_SHIFT 8 #define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT) +#define I2SDIV_IDV_SHIFT 8 +#define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT) + +enum jz47xx_i2s_version { + JZ_I2S_JZ4740, + JZ_I2S_JZ4780, +}; struct jz4740_i2s { struct resource *mem; @@ -98,6 +112,8 @@ struct jz4740_i2s { struct snd_dmaengine_dai_dma_data playback_dma_data; struct snd_dmaengine_dai_dma_data capture_dma_data; + + enum jz47xx_i2s_version version; }; static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s, @@ -267,13 +283,22 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO; else ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; + + div_reg &= ~I2SDIV_DV_MASK; + div_reg |= (div - 1) << I2SDIV_DV_SHIFT; } else { ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; + + if (i2s->version >= JZ_I2S_JZ4780) { + div_reg &= ~I2SDIV_IDV_MASK; + div_reg |= (div - 1) << I2SDIV_IDV_SHIFT; + } else { + div_reg &= ~I2SDIV_DV_MASK; + div_reg |= (div - 1) << I2SDIV_DV_SHIFT; + } } - div_reg &= ~I2SDIV_DV_MASK; - div_reg |= (div - 1) << I2SDIV_DV_SHIFT; jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); jz4740_i2s_write(i2s, JZ_REG_AIC_CLK_DIV, div_reg); @@ -369,11 +394,19 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, &i2s->capture_dma_data); - conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | - (8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | - JZ_AIC_CONF_OVERFLOW_PLAY_LAST | - JZ_AIC_CONF_I2S | - JZ_AIC_CONF_INTERNAL_CODEC; + if (i2s->version >= JZ_I2S_JZ4780) { + conf = (7 << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | + (8 << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | + JZ_AIC_CONF_OVERFLOW_PLAY_LAST | + JZ_AIC_CONF_I2S | + JZ_AIC_CONF_INTERNAL_CODEC; + } else { + conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | + (8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | + JZ_AIC_CONF_OVERFLOW_PLAY_LAST | + JZ_AIC_CONF_I2S | + JZ_AIC_CONF_INTERNAL_CODEC; + } jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, JZ_AIC_CONF_RESET); jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); @@ -422,13 +455,34 @@ static struct snd_soc_dai_driver jz4740_i2s_dai = { .resume = jz4740_i2s_resume, }; +static struct snd_soc_dai_driver jz4780_i2s_dai = { + .probe = jz4740_i2s_dai_probe, + .remove = jz4740_i2s_dai_remove, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = JZ4740_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = JZ4740_I2S_FMTS, + }, + .ops = &jz4740_i2s_dai_ops, + .suspend = jz4740_i2s_suspend, + .resume = jz4740_i2s_resume, +}; + static const struct snd_soc_component_driver jz4740_i2s_component = { .name = "jz4740-i2s", }; #ifdef CONFIG_OF static const struct of_device_id jz4740_of_matches[] = { - { .compatible = "ingenic,jz4740-i2s" }, + { .compatible = "ingenic,jz4740-i2s", .data = (void *)JZ_I2S_JZ4740 }, + { .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 }, { /* sentinel */ } }; #endif @@ -438,11 +492,16 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) struct jz4740_i2s *i2s; struct resource *mem; int ret; + const struct of_device_id *match; i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; + match = of_match_device(jz4740_of_matches, &pdev->dev); + if (match) + i2s->version = (enum jz47xx_i2s_version)match->data; + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); i2s->base = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(i2s->base)) @@ -460,8 +519,13 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); - ret = devm_snd_soc_register_component(&pdev->dev, - &jz4740_i2s_component, &jz4740_i2s_dai, 1); + if (i2s->version == JZ_I2S_JZ4780) + ret = devm_snd_soc_register_component(&pdev->dev, + &jz4740_i2s_component, &jz4780_i2s_dai, 1); + else + ret = devm_snd_soc_register_component(&pdev->dev, + &jz4740_i2s_component, &jz4740_i2s_dai, 1); + if (ret) return ret; From bd22f9d405eb14cc074d294919d0b909e0bc6170 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 9 Mar 2015 16:55:22 +0800 Subject: [PATCH 181/411] ASoC: rt5670: Revert Keep sysclk on patch The "Keep sysclk on if JD func is used" patch force enable/disable pin in rt5670_set_dai_sysclk. But some machine driver call it in dapm widget event. It will cause kernel crash. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 592f961b5de581..32cd26678bae5d 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2273,13 +2273,6 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) return 0; - if (rt5670->pdata.jd_mode) { - if (clk_id == RT5670_SCLK_S_PLL1) - snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); - else - snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); - snd_soc_dapm_sync(&codec->dapm); - } switch (clk_id) { case RT5670_SCLK_S_MCLK: reg_val |= RT5670_SCLK_SRC_MCLK; @@ -2724,10 +2717,6 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } if (rt5670->pdata.jd_mode) { - regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK, - RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); - rt5670->sysclk = 0; - rt5670->sysclk_src = RT5670_SCLK_S_RCCLK; regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, RT5670_PWR_MB, RT5670_PWR_MB); regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, From 485372dc24ca2eaac18ce41a51b9dd017bc11400 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 9 Mar 2015 16:55:23 +0800 Subject: [PATCH 182/411] ASoC: rt5670: Check sysclk source by private data Currently, is_sys_clk_from_pll check sysclk source by reading codec register value. And it will be invoked before updating dapm widget power. In some machine driver, snd_soc_dai_set_sysclk is called in dapm event to switch codec sysclk to RC clock in idle mode. And in some use cases (such as syspend/resume) hw_params will not be called when the dapm widget is powered up. As a result, is_sys_clk_from_pll will return 0 although it is supposed to be 1. To solve this, we let is_sys_clk_from_pll check sysclk sysclk_src which is stored in private data and don't change the value of sysclk_src when codec sysclk is switched to internal clock. The internal clock can only be used in idle mode, so it sould be fine if we don't set sysclk_src to internal clock. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 32cd26678bae5d..9e3bc43178f116 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -517,11 +517,10 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - unsigned int val; + struct snd_soc_codec *codec = source->codec; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); - val = snd_soc_read(source->codec, RT5670_GLB_CLK); - val &= RT5670_SCLK_SRC_MASK; - if (val == RT5670_SCLK_SRC_PLL1) + if (rt5670->sysclk_src == RT5670_SCLK_S_PLL1) return 1; else return 0; @@ -2270,9 +2269,6 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); unsigned int reg_val = 0; - if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) - return 0; - switch (clk_id) { case RT5670_SCLK_S_MCLK: reg_val |= RT5670_SCLK_SRC_MCLK; @@ -2290,7 +2286,8 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_update_bits(codec, RT5670_GLB_CLK, RT5670_SCLK_SRC_MASK, reg_val); rt5670->sysclk = freq; - rt5670->sysclk_src = clk_id; + if (clk_id != RT5670_SCLK_S_RCCLK) + rt5670->sysclk_src = clk_id; dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); From 62c76fe2e580f6a975679e8711bade09e24c204b Mon Sep 17 00:00:00 2001 From: Nicholas Mc Guire Date: Sun, 8 Mar 2015 06:02:15 -0400 Subject: [PATCH 183/411] ASoC: wm8996: match wait_for_completion_timeout return type return type of wait_for_completion_timeout is unsigned long not int. An appropriately named unsigned long is added and the assignment fixed up in case of completion occurring the remaining time is >=1 so ret is set to 1 if no timeout occurred. Signed-off-by: Nicholas Mc Guire Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index dc92d5e4e9426e..7a2b96959589cd 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2009,7 +2009,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); struct _fll_div fll_div; - unsigned long timeout; + unsigned long timeout, time_left; int ret, reg, retry; /* Any change? */ @@ -2113,10 +2113,11 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, timeout /= 2; for (retry = 0; retry < 10; retry++) { - ret = wait_for_completion_timeout(&wm8996->fll_lock, - timeout); - if (ret != 0) { + time_left = wait_for_completion_timeout(&wm8996->fll_lock, + timeout); + if (time_left != 0) { WARN_ON(!i2c->irq); + ret = 1; break; } From 159366ea38706402e8ebd0f55eef52931333deac Mon Sep 17 00:00:00 2001 From: Nicholas Mc Guire Date: Sun, 8 Mar 2015 06:02:38 -0400 Subject: [PATCH 184/411] ASoC: wm8996: ensure lower bounds of 1 for timeout wait_for_completion_timeout can be called with timeout == 0 due to msecs_to_jiffies(2) == 1 for HZ < 1000 and usecs_to_jiffies(300) == 1 for all reasonable values of HZ, thus the following timeout /= 2; sets timeout to 0. This patch simply adds a lower-bounds of 1. Signed-off-by: Nicholas Mc Guire Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 7a2b96959589cd..308748a022c552 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2110,7 +2110,8 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (i2c->irq) timeout *= 10; else - timeout /= 2; + /* ensure timeout of atleast 1 jiffies */ + timeout = timeout/2 ? : 1; for (retry = 0; retry < 10; retry++) { time_left = wait_for_completion_timeout(&wm8996->fll_lock, From 17f4ad601d2753be7f15cd2928e89309759e4936 Mon Sep 17 00:00:00 2001 From: Nicholas Mc Guire Date: Sun, 8 Mar 2015 09:16:49 -0400 Subject: [PATCH 185/411] ASoC: arizona: match wait_for_completion_timeout return type return type of wait_for_completion_timeout is unsigned long not int. An appropriately named unsigned long is added and the assignment fixed up. Signed-off-by: Nicholas Mc Guire Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29202610dd0dd9..9015b44a9e1101 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1901,7 +1901,7 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll) static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; - int ret; + unsigned long time_left; bool use_sync = false; int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; @@ -1977,9 +1977,9 @@ static int arizona_enable_fll(struct arizona_fll *fll) regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); - ret = wait_for_completion_timeout(&fll->ok, + time_left = wait_for_completion_timeout(&fll->ok, msecs_to_jiffies(250)); - if (ret == 0) + if (time_left == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); return 0; From 905a808664402dec0ac11376833e79da4ae7b2fd Mon Sep 17 00:00:00 2001 From: Nicholas Mc Guire Date: Sun, 8 Mar 2015 06:57:07 -0400 Subject: [PATCH 186/411] ASoC: wm5100: match wait_for_completion_timeout return type return type of wait_for_completion_timeout is unsigned long not int. An appropriately named unsigned long is added and the assignment fixed up. Signed-off-by: Nicholas Mc Guire Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index ea09db585aa1fd..96740379b711bf 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1762,6 +1762,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct _fll_div factors; struct wm5100_fll *fll; int ret, base, lock, i, timeout; + unsigned long time_left; switch (fll_id) { case WM5100_FLL1: @@ -1842,9 +1843,9 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Poll for the lock; will use interrupt when we can test */ for (i = 0; i < timeout; i++) { if (i2c->irq) { - ret = wait_for_completion_timeout(&fll->lock, - msecs_to_jiffies(25)); - if (ret > 0) + time_left = wait_for_completion_timeout(&fll->lock, + msecs_to_jiffies(25)); + if (time_left > 0) break; } else { msleep(1); From 78bb997ace926451ec8d1ed15178b161dc61b805 Mon Sep 17 00:00:00 2001 From: Nicholas Mc Guire Date: Sun, 8 Mar 2015 07:06:05 -0400 Subject: [PATCH 187/411] ASoC: wm2200: match wait_for_completion_timeout return type return type of wait_for_completion_timeout is unsigned long not int. An appropriately named unsigned long is added and the assignment fixed up. Signed-off-by: Nicholas Mc Guire Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 15599845a66021..b48694a8d21370 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1942,6 +1942,7 @@ static int wm2200_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct wm2200_priv *wm2200 = snd_soc_codec_get_drvdata(codec); struct _fll_div factors; int ret, i, timeout; + unsigned long time_left; if (!Fout) { dev_dbg(codec->dev, "FLL disabled"); @@ -2021,9 +2022,10 @@ static int wm2200_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Poll for the lock; will use the interrupt to exit quickly */ for (i = 0; i < timeout; i++) { if (i2c->irq) { - ret = wait_for_completion_timeout(&wm2200->fll_lock, - msecs_to_jiffies(25)); - if (ret > 0) + time_left = wait_for_completion_timeout( + &wm2200->fll_lock, + msecs_to_jiffies(25)); + if (time_left > 0) break; } else { msleep(1); From 32556394501a27c02e7185c4d11a51b636b02f4b Mon Sep 17 00:00:00 2001 From: Dmitry Torokhov Date: Mon, 9 Mar 2015 11:15:28 -0700 Subject: [PATCH 188/411] ASoC: cx20442: remove incorerct __exit markups Even if bus is not hot-pluggable, the devices can be unbound from the driver via sysfs, so we should not be using __exit annotations on remove() methods. The only exception is drivers registered with platform_driver_probe() which specifically disables sysfs bind/unbind attributes. Signed-off-by: Dmitry Torokhov Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 0b10979513c4a9..0f334bc1b63c78 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -420,7 +420,7 @@ static int cx20442_platform_probe(struct platform_device *pdev) &cx20442_codec_dev, &cx20442_dai, 1); } -static int __exit cx20442_platform_remove(struct platform_device *pdev) +static int cx20442_platform_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; @@ -431,7 +431,7 @@ static struct platform_driver cx20442_platform_driver = { .name = "cx20442-codec", }, .probe = cx20442_platform_probe, - .remove = __exit_p(cx20442_platform_remove), + .remove = cx20442_platform_remove, }; module_platform_driver(cx20442_platform_driver); From f580f8afd0d81c3f04d8b393c9d675ef289e4d40 Mon Sep 17 00:00:00 2001 From: Dmitry Torokhov Date: Mon, 9 Mar 2015 11:18:14 -0700 Subject: [PATCH 189/411] ASoC: tlv320aic23: remove incorrect __exit markups Even if bus is not hot-pluggable, the devices can be unbound from the driver via sysfs, so we should not be using __exit annotations on remove() methods. The only exception is drivers registered with platform_driver_probe() which specifically disables sysfs bind/unbind attributes. Signed-off-by: Dmitry Torokhov Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23-i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c index f137019954828f..78a94af6551859 100644 --- a/sound/soc/codecs/tlv320aic23-i2c.c +++ b/sound/soc/codecs/tlv320aic23-i2c.c @@ -31,7 +31,7 @@ static int tlv320aic23_i2c_probe(struct i2c_client *i2c, return tlv320aic23_probe(&i2c->dev, regmap); } -static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +static int tlv320aic23_i2c_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); return 0; @@ -56,7 +56,7 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic23_of_match), }, .probe = tlv320aic23_i2c_probe, - .remove = __exit_p(tlv320aic23_i2c_remove), + .remove = tlv320aic23_i2c_remove, .id_table = tlv320aic23_id, }; From ce991981311e0ae258982b600564226ad6cb24ea Mon Sep 17 00:00:00 2001 From: Yannick Guerrini Date: Mon, 9 Mar 2015 22:13:03 +0100 Subject: [PATCH 190/411] ALSA: firewire: Fix trivial typos in comments Change 'propper' to 'proper' Change 'paramters' to 'parameters' Change 'SYT_INTEVAL' to 'SYT_INTERVAL' Change 'aligh'/'alighed' to 'align'/'aligned' Signed-off-by: Yannick Guerrini Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.c | 8 ++++---- sound/firewire/fireworks/fireworks_transaction.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 5cc356db5351d9..e061355f535f07 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -166,10 +166,10 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * One AMDTP packet can include some frames. In blocking mode, the * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's - * preferrable to aligh period/buffer sizes to current SYT_INTERVAL. + * preferrable to align period/buffer sizes to current SYT_INTERVAL. * - * TODO: These constraints can be improved with propper rules. - * Currently apply LCM of SYT_INTEVALs. + * TODO: These constraints can be improved with proper rules. + * Currently apply LCM of SYT_INTERVALs. */ err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); @@ -270,7 +270,7 @@ static void amdtp_read_s32(struct amdtp_stream *s, * @s: the AMDTP stream to configure * @format: the format of the ALSA PCM device * - * The sample format must be set after the other paramters (rate/PCM channels/ + * The sample format must be set after the other parameters (rate/PCM channels/ * MIDI) and before the stream is started, and must not be changed while the * stream is running. */ diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 2a85e4209f0b74..f550808d178416 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -13,7 +13,7 @@ * * Transaction substance: * At first, 6 data exist. Following to the data, parameters for each command - * exist. All of the parameters are 32 bit alighed to big endian. + * exist. All of the parameters are 32 bit aligned to big endian. * data[0]: Length of transaction substance * data[1]: Transaction version * data[2]: Sequence number. This is incremented by the device From 4ed56666b7fc98c750a23b5263350b75e742b534 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 22:13:30 +0900 Subject: [PATCH 191/411] ALSA: core: use precomputed table to check userspace control params The parameters can be decided in compile time. This commit adds precomputed table to reduce calculating time. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 60 +++++++++++++++++++++----------------------- 1 file changed, 28 insertions(+), 32 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8e83c867..0b85cbc27e4d2e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1161,6 +1161,23 @@ static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol) static int snd_ctl_elem_add(struct snd_ctl_file *file, struct snd_ctl_elem_info *info, int replace) { + /* The capacity of struct snd_ctl_elem_value.value.*/ + static const unsigned int value_sizes[] = { + [SNDRV_CTL_ELEM_TYPE_BOOLEAN] = sizeof(long), + [SNDRV_CTL_ELEM_TYPE_INTEGER] = sizeof(long), + [SNDRV_CTL_ELEM_TYPE_ENUMERATED] = sizeof(unsigned int), + [SNDRV_CTL_ELEM_TYPE_BYTES] = sizeof(unsigned char), + [SNDRV_CTL_ELEM_TYPE_IEC958] = sizeof(struct snd_aes_iec958), + [SNDRV_CTL_ELEM_TYPE_INTEGER64] = sizeof(long long), + }; + static const unsigned int max_value_counts[] = { + [SNDRV_CTL_ELEM_TYPE_BOOLEAN] = 128, + [SNDRV_CTL_ELEM_TYPE_INTEGER] = 128, + [SNDRV_CTL_ELEM_TYPE_ENUMERATED] = 128, + [SNDRV_CTL_ELEM_TYPE_BYTES] = 512, + [SNDRV_CTL_ELEM_TYPE_IEC958] = 1, + [SNDRV_CTL_ELEM_TYPE_INTEGER64] = 64, + }; struct snd_card *card = file->card; struct snd_kcontrol kctl, *_kctl; unsigned int access; @@ -1168,8 +1185,6 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, struct user_element *ue; int idx, err; - if (info->count < 1) - return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| @@ -1201,37 +1216,18 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, kctl.tlv.c = snd_ctl_elem_user_tlv; access |= SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; } - switch (info->type) { - case SNDRV_CTL_ELEM_TYPE_BOOLEAN: - case SNDRV_CTL_ELEM_TYPE_INTEGER: - private_size = sizeof(long); - if (info->count > 128) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_INTEGER64: - private_size = sizeof(long long); - if (info->count > 64) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_ENUMERATED: - private_size = sizeof(unsigned int); - if (info->count > 128 || info->value.enumerated.items == 0) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_BYTES: - private_size = sizeof(unsigned char); - if (info->count > 512) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_IEC958: - private_size = sizeof(struct snd_aes_iec958); - if (info->count != 1) - return -EINVAL; - break; - default: + + if (info->type < SNDRV_CTL_ELEM_TYPE_BOOLEAN || + info->type > SNDRV_CTL_ELEM_TYPE_INTEGER64) return -EINVAL; - } - private_size *= info->count; + if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED && + info->value.enumerated.items == 0) + return -EINVAL; + if (info->count < 1 || + info->count > max_value_counts[info->type]) + return -EINVAL; + + private_size = value_sizes[info->type] * info->count; ue = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); if (ue == NULL) return -ENOMEM; From 2225e79b9b0370bc179f44756bee809b5e7b4d06 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 22:13:31 +0900 Subject: [PATCH 192/411] ALSA: core: reduce stack usage related to snd_ctl_new() The callers of snd_ctl_new() need to have 'struct snd_kcontrol' data, and pass the data as template. Then, the function allocates the structure data again and copy from the template. This is a waste of resources. Especially, the callers use large stack for the template. This commit removes a need of template for the function, thus, changes the prototype of snd_ctl_new(). Furthermore, this commit changes the code of callers, snd_ctl_new1() and snd_ctl_elem_add() for better shape. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 213 ++++++++++++++++++++++++++----------------- 1 file changed, 130 insertions(+), 83 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 0b85cbc27e4d2e..e1d8e0c816f081 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -192,36 +192,43 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, EXPORT_SYMBOL(snd_ctl_notify); /** - * snd_ctl_new - create a control instance from the template - * @control: the control template - * @access: the default control access + * snd_ctl_new - create a new control instance with some elements + * @kctl: the pointer to store new control instance + * @count: the number of elements in this control + * @access: the default access flags for elements in this control + * @file: given when locking these elements * - * Allocates a new struct snd_kcontrol instance and copies the given template - * to the new instance. It does not copy volatile data (access). + * Allocates a memory object for a new control instance. The instance has + * elements as many as the given number (@count). Each element has given + * access permissions (@access). Each element is locked when @file is given. * - * Return: The pointer of the new instance, or %NULL on failure. + * Return: 0 on success, error code on failure */ -static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, - unsigned int access) +static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, + unsigned int access, struct snd_ctl_file *file) { - struct snd_kcontrol *kctl; + unsigned int size; unsigned int idx; - if (snd_BUG_ON(!control || !control->count)) - return NULL; + if (count == 0 || count > MAX_CONTROL_COUNT) + return -EINVAL; - if (control->count > MAX_CONTROL_COUNT) - return NULL; + size = sizeof(struct snd_kcontrol); + size += sizeof(struct snd_kcontrol_volatile) * count; - kctl = kzalloc(sizeof(*kctl) + sizeof(struct snd_kcontrol_volatile) * control->count, GFP_KERNEL); - if (kctl == NULL) { + *kctl = kzalloc(size, GFP_KERNEL); + if (*kctl == NULL) { pr_err("ALSA: Cannot allocate control instance\n"); - return NULL; + return -ENOMEM; } - *kctl = *control; - for (idx = 0; idx < kctl->count; idx++) - kctl->vd[idx].access = access; - return kctl; + + for (idx = 0; idx < count; idx++) { + (*kctl)->vd[idx].access = access; + (*kctl)->vd[idx].owner = file; + } + (*kctl)->count = count; + + return 0; } /** @@ -238,37 +245,53 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, void *private_data) { - struct snd_kcontrol kctl; + struct snd_kcontrol *kctl; + unsigned int count; unsigned int access; + int err; if (snd_BUG_ON(!ncontrol || !ncontrol->info)) return NULL; - memset(&kctl, 0, sizeof(kctl)); - kctl.id.iface = ncontrol->iface; - kctl.id.device = ncontrol->device; - kctl.id.subdevice = ncontrol->subdevice; + + count = ncontrol->count; + if (count == 0) + count = 1; + + access = ncontrol->access; + if (access == 0) + access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + access &= (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE | + SNDRV_CTL_ELEM_ACCESS_INACTIVE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); + + err = snd_ctl_new(&kctl, count, access, NULL); + if (err < 0) + return NULL; + + /* The 'numid' member is decided when calling snd_ctl_add(). */ + kctl->id.iface = ncontrol->iface; + kctl->id.device = ncontrol->device; + kctl->id.subdevice = ncontrol->subdevice; if (ncontrol->name) { - strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); - if (strcmp(ncontrol->name, kctl.id.name) != 0) + strlcpy(kctl->id.name, ncontrol->name, sizeof(kctl->id.name)); + if (strcmp(ncontrol->name, kctl->id.name) != 0) pr_warn("ALSA: Control name '%s' truncated to '%s'\n", - ncontrol->name, kctl.id.name); + ncontrol->name, kctl->id.name); } - kctl.id.index = ncontrol->index; - kctl.count = ncontrol->count ? ncontrol->count : 1; - access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : - (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| - SNDRV_CTL_ELEM_ACCESS_VOLATILE| - SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| - SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); - kctl.info = ncontrol->info; - kctl.get = ncontrol->get; - kctl.put = ncontrol->put; - kctl.tlv.p = ncontrol->tlv.p; - kctl.private_value = ncontrol->private_value; - kctl.private_data = private_data; - return snd_ctl_new(&kctl, access); + kctl->id.index = ncontrol->index; + + kctl->info = ncontrol->info; + kctl->get = ncontrol->get; + kctl->put = ncontrol->put; + kctl->tlv.p = ncontrol->tlv.p; + + kctl->private_value = ncontrol->private_value; + kctl->private_data = private_data; + + return kctl; } EXPORT_SYMBOL(snd_ctl_new1); @@ -1179,44 +1202,48 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, [SNDRV_CTL_ELEM_TYPE_INTEGER64] = 64, }; struct snd_card *card = file->card; - struct snd_kcontrol kctl, *_kctl; + struct snd_kcontrol *kctl; + unsigned int count; unsigned int access; long private_size; struct user_element *ue; - int idx, err; - - access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : - (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| - SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)); - info->id.numid = 0; - memset(&kctl, 0, sizeof(kctl)); + int err; + /* Delete a control to replace them if needed. */ if (replace) { + info->id.numid = 0; err = snd_ctl_remove_user_ctl(file, &info->id); if (err) return err; } - if (card->user_ctl_count >= MAX_USER_CONTROLS) + /* + * The number of userspace controls are counted control by control, + * not element by element. + */ + if (card->user_ctl_count + 1 > MAX_USER_CONTROLS) return -ENOMEM; - memcpy(&kctl.id, &info->id, sizeof(info->id)); - kctl.count = info->owner ? info->owner : 1; - access |= SNDRV_CTL_ELEM_ACCESS_USER; - if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) - kctl.info = snd_ctl_elem_user_enum_info; - else - kctl.info = snd_ctl_elem_user_info; - if (access & SNDRV_CTL_ELEM_ACCESS_READ) - kctl.get = snd_ctl_elem_user_get; - if (access & SNDRV_CTL_ELEM_ACCESS_WRITE) - kctl.put = snd_ctl_elem_user_put; - if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) { - kctl.tlv.c = snd_ctl_elem_user_tlv; + /* Check the number of elements for this userspace control. */ + count = info->owner; + if (count == 0) + count = 1; + + /* Arrange access permissions if needed. */ + access = info->access; + if (access == 0) + access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + access &= (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_INACTIVE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE); + if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) access |= SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; - } + access |= SNDRV_CTL_ELEM_ACCESS_USER; + /* + * Check information and calculate the size of data specific to + * this userspace control. + */ if (info->type < SNDRV_CTL_ELEM_TYPE_BOOLEAN || info->type > SNDRV_CTL_ELEM_TYPE_INTEGER64) return -EINVAL; @@ -1226,11 +1253,27 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1 || info->count > max_value_counts[info->type]) return -EINVAL; - private_size = value_sizes[info->type] * info->count; - ue = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); - if (ue == NULL) + + /* + * Keep memory object for this userspace control. After passing this + * code block, the instance should be freed by snd_ctl_free_one(). + * + * Note that these elements in this control are locked. + */ + err = snd_ctl_new(&kctl, count, access, file); + if (err < 0) + return err; + kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, + GFP_KERNEL); + if (kctl->private_data == NULL) { + kfree(kctl); return -ENOMEM; + } + kctl->private_free = snd_ctl_elem_user_free; + + /* Set private data for this userspace control. */ + ue = (struct user_element *)kctl->private_data; ue->card = card; ue->info = *info; ue->info.access = 0; @@ -1239,21 +1282,25 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (ue->info.type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) { err = snd_ctl_elem_init_enum_names(ue); if (err < 0) { - kfree(ue); + snd_ctl_free_one(kctl); return err; } } - kctl.private_free = snd_ctl_elem_user_free; - _kctl = snd_ctl_new(&kctl, access); - if (_kctl == NULL) { - kfree(ue->priv_data); - kfree(ue); - return -ENOMEM; - } - _kctl->private_data = ue; - for (idx = 0; idx < _kctl->count; idx++) - _kctl->vd[idx].owner = file; - err = snd_ctl_add(card, _kctl); + + /* Set callback functions. */ + if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) + kctl->info = snd_ctl_elem_user_enum_info; + else + kctl->info = snd_ctl_elem_user_info; + if (access & SNDRV_CTL_ELEM_ACCESS_READ) + kctl->get = snd_ctl_elem_user_get; + if (access & SNDRV_CTL_ELEM_ACCESS_WRITE) + kctl->put = snd_ctl_elem_user_put; + if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) + kctl->tlv.c = snd_ctl_elem_user_tlv; + + /* This function manage to free the instance on failure. */ + err = snd_ctl_add(card, kctl); if (err < 0) return err; From 8d98a0673f761f9b7be51a293ca9142ec0c037ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 15:39:55 +0100 Subject: [PATCH 193/411] ALSA: seq_oss: Drop superfluous error/debug messages after malloc failures The kernel memory allocators already report the errors when the requested allocation fails, thus we don't need to warn it again in each caller side. Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 4 +--- sound/core/seq/oss/seq_oss_midi.c | 5 ++--- sound/core/seq/oss/seq_oss_readq.c | 9 ++++----- sound/core/seq/oss/seq_oss_synth.c | 6 ++---- 4 files changed, 9 insertions(+), 15 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b0e32e161dd187..2de3feff70d06d 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -188,10 +188,8 @@ snd_seq_oss_open(struct file *file, int level) struct seq_oss_devinfo *dp; dp = kzalloc(sizeof(*dp), GFP_KERNEL); - if (!dp) { - pr_err("ALSA: seq_oss: can't malloc device info\n"); + if (!dp) return -ENOMEM; - } dp->cseq = system_client; dp->port = -1; diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index e79cc44b1394e6..96e8395ae5863e 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -173,10 +173,9 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) /* * allocate midi info record */ - if ((mdev = kzalloc(sizeof(*mdev), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc midi info\n"); + mdev = kzalloc(sizeof(*mdev), GFP_KERNEL); + if (!mdev) return -ENOMEM; - } /* copy the port information */ mdev->client = pinfo->addr.client; diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c index 654d17a5023c8a..c080c73cea04a2 100644 --- a/sound/core/seq/oss/seq_oss_readq.c +++ b/sound/core/seq/oss/seq_oss_readq.c @@ -47,13 +47,12 @@ snd_seq_oss_readq_new(struct seq_oss_devinfo *dp, int maxlen) { struct seq_oss_readq *q; - if ((q = kzalloc(sizeof(*q), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue\n"); + q = kzalloc(sizeof(*q), GFP_KERNEL); + if (!q) return NULL; - } - if ((q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue buffer\n"); + q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL); + if (!q->q) { kfree(q); return NULL; } diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 835edc80f9186f..48e4fe1b68abba 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -106,10 +106,9 @@ snd_seq_oss_synth_probe(struct device *_dev) struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev); unsigned long flags; - if ((rec = kzalloc(sizeof(*rec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc synth info\n"); + rec = kzalloc(sizeof(*rec), GFP_KERNEL); + if (!rec) return -ENOMEM; - } rec->seq_device = -1; rec->synth_type = reg->type; rec->synth_subtype = reg->subtype; @@ -249,7 +248,6 @@ snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp) if (info->nr_voices > 0) { info->ch = kcalloc(info->nr_voices, sizeof(struct seq_oss_chinfo), GFP_KERNEL); if (!info->ch) { - pr_err("ALSA: seq_oss: Cannot malloc voices\n"); rec->oper.close(&info->arg); module_put(rec->oper.owner); snd_use_lock_free(&rec->use_lock); From 24db8bbaa3fcfaf0c2faccbff5864b58088ac1f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 15:41:18 +0100 Subject: [PATCH 194/411] ALSA: seq: Drop superfluous error/debug messages after malloc failures The kernel memory allocators already report the errors when the requested allocation fails, thus we don't need to warn it again in each caller side. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_fifo.c | 4 +--- sound/core/seq/seq_memory.c | 8 ++------ sound/core/seq/seq_ports.c | 4 +--- sound/core/seq/seq_prioq.c | 4 +--- sound/core/seq/seq_queue.c | 4 +--- sound/core/seq/seq_timer.c | 4 +--- 6 files changed, 7 insertions(+), 21 deletions(-) diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 53a403e17c5bba..1d5acbe0c08bdf 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -33,10 +33,8 @@ struct snd_seq_fifo *snd_seq_fifo_new(int poolsize) struct snd_seq_fifo *f; f = kzalloc(sizeof(*f), GFP_KERNEL); - if (f == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_fifo_new() \n"); + if (!f) return NULL; - } f->pool = snd_seq_pool_new(poolsize); if (f->pool == NULL) { diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index ba8e4a64e13ee3..801076687bb16f 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -387,10 +387,8 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) return 0; pool->ptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size); - if (pool->ptr == NULL) { - pr_debug("ALSA: seq: malloc for sequencer events failed\n"); + if (!pool->ptr) return -ENOMEM; - } /* add new cells to the free cell list */ spin_lock_irqsave(&pool->lock, flags); @@ -463,10 +461,8 @@ struct snd_seq_pool *snd_seq_pool_new(int poolsize) /* create pool block */ pool = kzalloc(sizeof(*pool), GFP_KERNEL); - if (pool == NULL) { - pr_debug("ALSA: seq: malloc failed for pool\n"); + if (!pool) return NULL; - } spin_lock_init(&pool->lock); pool->ptr = NULL; pool->free = NULL; diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 46ff593f618dc0..55170a20ae7237 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -141,10 +141,8 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, /* create a new port */ new_port = kzalloc(sizeof(*new_port), GFP_KERNEL); - if (! new_port) { - pr_debug("ALSA: seq: malloc failed for registering client port\n"); + if (!new_port) return NULL; /* failure, out of memory */ - } /* init port data */ new_port->addr.client = client->number; new_port->addr.port = -1; diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index 021b02bc9330f5..bc1c8488fc2a15 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -59,10 +59,8 @@ struct snd_seq_prioq *snd_seq_prioq_new(void) struct snd_seq_prioq *f; f = kzalloc(sizeof(*f), GFP_KERNEL); - if (f == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_prioq_new()\n"); + if (!f) return NULL; - } spin_lock_init(&f->lock); f->head = NULL; diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index aad4878cee5579..a0cda38205b976 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -111,10 +111,8 @@ static struct snd_seq_queue *queue_new(int owner, int locked) struct snd_seq_queue *q; q = kzalloc(sizeof(*q), GFP_KERNEL); - if (q == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_queue_new()\n"); + if (!q) return NULL; - } spin_lock_init(&q->owner_lock); spin_lock_init(&q->check_lock); diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index e73605393eee53..186f1611103c52 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -56,10 +56,8 @@ struct snd_seq_timer *snd_seq_timer_new(void) struct snd_seq_timer *tmr; tmr = kzalloc(sizeof(*tmr), GFP_KERNEL); - if (tmr == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_timer_new() \n"); + if (!tmr) return NULL; - } spin_lock_init(&tmr->lock); /* reset setup to defaults */ From ec0e9937aaa8b0a4b0633711c4d70d622acd9a7f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 15:42:14 +0100 Subject: [PATCH 195/411] ALSA: core: Drop superfluous error/debug messages after malloc failures The kernel memory allocators already report the errors when the requested allocation fails, thus we don't need to warn it again in each caller side. Signed-off-by: Takashi Iwai --- sound/core/control.c | 4 +--- sound/core/device.c | 4 +--- sound/core/hwdep.c | 4 +--- sound/core/oss/mixer_oss.c | 4 +--- sound/core/oss/pcm_oss.c | 1 - sound/core/pcm.c | 13 +++---------- sound/core/rawmidi.c | 8 ++------ sound/core/timer.c | 4 +--- 8 files changed, 10 insertions(+), 32 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index e1d8e0c816f081..833b223a363a03 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -217,10 +217,8 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, size += sizeof(struct snd_kcontrol_volatile) * count; *kctl = kzalloc(size, GFP_KERNEL); - if (*kctl == NULL) { - pr_err("ALSA: Cannot allocate control instance\n"); + if (!*kctl) return -ENOMEM; - } for (idx = 0; idx < count; idx++) { (*kctl)->vd[idx].access = access; diff --git a/sound/core/device.c b/sound/core/device.c index 41bec3075ae5b8..c1a845b42a8bd4 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -50,10 +50,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, if (snd_BUG_ON(!card || !device_data || !ops)) return -ENXIO; dev = kzalloc(sizeof(*dev), GFP_KERNEL); - if (dev == NULL) { - dev_err(card->dev, "Cannot allocate device, type=%d\n", type); + if (!dev) return -ENOMEM; - } INIT_LIST_HEAD(&dev->list); dev->card = card; dev->type = type; diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 84244a5143cfe9..51692c8a39ea1b 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -378,10 +378,8 @@ int snd_hwdep_new(struct snd_card *card, char *id, int device, if (rhwdep) *rhwdep = NULL; hwdep = kzalloc(sizeof(*hwdep), GFP_KERNEL); - if (hwdep == NULL) { - dev_err(card->dev, "hwdep: cannot allocate\n"); + if (!hwdep) return -ENOMEM; - } init_waitqueue_head(&hwdep->open_wait); mutex_init(&hwdep->open_mutex); diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 5e6349f00ecd77..056f8e27485181 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1212,10 +1212,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, /* not changed */ goto __unlock; tbl = kmalloc(sizeof(*tbl), GFP_KERNEL); - if (! tbl) { - pr_err("ALSA: mixer_oss: no memory\n"); + if (!tbl) goto __unlock; - } tbl->oss_id = ch; tbl->name = kstrdup(str, GFP_KERNEL); if (! tbl->name) { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 80423a4ccab621..58550cc93f2805 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -854,7 +854,6 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) params = kmalloc(sizeof(*params), GFP_KERNEL); sparams = kmalloc(sizeof(*sparams), GFP_KERNEL); if (!sw_params || !params || !sparams) { - pcm_dbg(substream->pcm, "No memory\n"); err = -ENOMEM; goto failure; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index e9b87465c73da6..b25bcf5b864463 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -343,11 +343,8 @@ static void snd_pcm_proc_info_read(struct snd_pcm_substream *substream, return; info = kmalloc(sizeof(*info), GFP_KERNEL); - if (! info) { - pcm_dbg(substream->pcm, - "snd_pcm_proc_info_read: cannot malloc\n"); + if (!info) return; - } err = snd_pcm_info(substream, info); if (err < 0) { @@ -717,10 +714,8 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) prev = NULL; for (idx = 0, prev = NULL; idx < substream_count; idx++) { substream = kzalloc(sizeof(*substream), GFP_KERNEL); - if (substream == NULL) { - pcm_err(pcm, "Cannot allocate PCM substream\n"); + if (!substream) return -ENOMEM; - } substream->pcm = pcm; substream->pstr = pstr; substream->number = idx; @@ -774,10 +769,8 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, if (rpcm) *rpcm = NULL; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); - if (pcm == NULL) { - dev_err(card->dev, "Cannot allocate PCM\n"); + if (!pcm) return -ENOMEM; - } pcm->card = card; pcm->device = device; pcm->internal = internal; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b5a748596fc40d..a7759846fbaadf 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1429,10 +1429,8 @@ static int snd_rawmidi_alloc_substreams(struct snd_rawmidi *rmidi, for (idx = 0; idx < count; idx++) { substream = kzalloc(sizeof(*substream), GFP_KERNEL); - if (substream == NULL) { - rmidi_err(rmidi, "rawmidi: cannot allocate substream\n"); + if (!substream) return -ENOMEM; - } substream->stream = direction; substream->number = idx; substream->rmidi = rmidi; @@ -1479,10 +1477,8 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, if (rrawmidi) *rrawmidi = NULL; rmidi = kzalloc(sizeof(*rmidi), GFP_KERNEL); - if (rmidi == NULL) { - dev_err(card->dev, "rawmidi: cannot allocate\n"); + if (!rmidi) return -ENOMEM; - } rmidi->card = card; rmidi->device = device; mutex_init(&rmidi->open_mutex); diff --git a/sound/core/timer.c b/sound/core/timer.c index 490b489d713d4c..a9a1a047c521ff 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -774,10 +774,8 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, if (rtimer) *rtimer = NULL; timer = kzalloc(sizeof(*timer), GFP_KERNEL); - if (timer == NULL) { - pr_err("ALSA: timer: cannot allocate\n"); + if (!timer) return -ENOMEM; - } timer->tmr_class = tid->dev_class; timer->card = card; timer->tmr_device = tid->device; From 8d0c38a3f2a6bb70e952f127ed817fc7a08db52c Mon Sep 17 00:00:00 2001 From: Jin Yao Date: Tue, 10 Mar 2015 09:05:38 +0800 Subject: [PATCH 196/411] ASoC: Intel: move sysclk source setting to platform_clock_control for balance. A playback noise happens after suspend/resume on Braswell. The issue is due to the codec PLL and codec ASRC are not enabled correctly due to the incorrect sysclk setting after resume. This patch resets the sysclk source setting in platform clock control widget handler. Signed-off-by: Bard Liao Signed-off-by: Jin Yao Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 40 ++++++++++++++++++++++---------- 1 file changed, 28 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index bc8dcacd5e6a08..279df4c43de16c 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -50,6 +50,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; + int ret; codec_dai = cht_get_codec_dai(card); if (!codec_dai) { @@ -57,17 +58,31 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (!SND_SOC_DAPM_EVENT_OFF(event)) - return 0; - - /* Set codec sysclk source to its internal clock because codec PLL will - * be off when idle and MCLK will also be off by ACPI when codec is - * runtime suspended. Codec needs clock for jack detection and button - * press. - */ - snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, - 0, SND_SOC_CLOCK_IN); - + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, 48000 * 512); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* set codec sysclk source to PLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + 48000 * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + } else { + /* Set codec sysclk source to its internal clock because codec + * PLL will be off when idle and MCLK will also be off by ACPI + * when codec is runtime suspended. Codec needs clock for jack + * detection and button press. + */ + snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, + 48000 * 512, SND_SOC_CLOCK_IN); + } return 0; } @@ -77,7 +92,8 @@ static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_POST_PMD), + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cht_audio_map[] = { From 4945f1fdc14ef090abe50d1b5682bfc1e4763c06 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2015 12:50:15 +0100 Subject: [PATCH 197/411] ALSA: seq: Fix init order of snd_seq_device stuff When the sequencer driver is built in kernel, it may panic at boot because of the uninitialized snd_seq_bus_type. Initialize it properly via subsys_initcall() instead of module_init() to assure that the bus is registered beforehand. Reported-by: Fengguang Wu Fixes: 7c37ae5c625a ('ALSA: seq: Rewrite sequencer device binding with standard bus') Signed-off-by: Takashi Iwai --- sound/core/seq/seq_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 355b34269bd19f..d99f99d61983a2 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) bus_unregister(&snd_seq_bus_type); } -module_init(alsa_seq_device_init) +subsys_initcall(alsa_seq_device_init) module_exit(alsa_seq_device_exit) From 6ec6fb6f231547834925ff802deb4415f49f708e Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Tue, 10 Mar 2015 11:48:05 +0100 Subject: [PATCH 198/411] ASoC: rsnd: Use %pad to print dma_addr_t in rsnd_dmapp_init() sound/soc/sh/rcar/dma.c: In function 'rsnd_dmapp_init': sound/soc/sh/rcar/dma.c:341:2: warning: format '%x' expects argument of type 'unsigned int', but argument 5 has type 'dma_addr_t' [-Wformat=] dev_dbg(dev, "id/src/dst/chcr = %d/%x/%x/%08x\n", ^ sound/soc/sh/rcar/dma.c:341:2: warning: format '%x' expects argument of type 'unsigned int', but argument 6 has type 'dma_addr_t' [-Wformat=] Fixes: 288f392e729dd4d3 ("ASoC: rsnd: add Audio DMAC peri peri support rework") Signed-off-by: Geert Uytterhoeven Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 92fd55044ee6b4..2b73df84a76904 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -338,8 +338,8 @@ static int rsnd_dmapp_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, rsnd_dmapp_stop(dma); - dev_dbg(dev, "id/src/dst/chcr = %d/%x/%x/%08x\n", - dmapp->dmapp_id, dma->src_addr, dma->dst_addr, dmapp->chcr); + dev_dbg(dev, "id/src/dst/chcr = %d/%pad/%pad/%08x\n", + dmapp->dmapp_id, &dma->src_addr, &dma->dst_addr, dmapp->chcr); return 0; } From 8537483a17038f08c68c1f0a561f0fe686e5706f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Mar 2015 01:25:01 +0000 Subject: [PATCH 199/411] ASoC: rsnd: recover PIO mode for new dma interface Renesas sound driver needs 1st/2nd DMA interface, and 1st DMA is using DMAEngine, and 2nd is using local method now. 2nd DMA had been DMAEngine, but it was moved to local method by previous patchset. But then, it lost PIO mode fallback when probe. this patch recovers it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 2b73df84a76904..cd7b79a01ce239 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -539,6 +539,13 @@ void rsnd_dma_start(struct rsnd_dma *dma) void rsnd_dma_quit(struct rsnd_dma *dma) { + struct rsnd_mod *mod = rsnd_dma_to_mod(dma); + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + + if (!dmac) + return; + dma->ops->quit(dma); } @@ -548,8 +555,18 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id) struct rsnd_mod *mod_from; struct rsnd_mod *mod_to; struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); int is_play = rsnd_io_is_play(io); + /* + * DMA failed. try to PIO mode + * see + * rsnd_ssi_fallback() + * rsnd_rdai_continuance_probe() + */ + if (!dmac) + return -EAGAIN; + rsnd_dma_of_path(dma, is_play, &mod_from, &mod_to); dma->src_addr = rsnd_dma_addr(priv, mod_from, is_play, 1); @@ -589,7 +606,7 @@ int rsnd_dma_probe(struct platform_device *pdev, dmac = devm_kzalloc(dev, sizeof(*dmac), GFP_KERNEL); if (!dmac || !res) { dev_err(dev, "dma allocate failed\n"); - return -ENOMEM; + return 0; /* it will be PIO mode */ } dmac->dmapp_num = 0; From 530b7b4a52365ce2a250669b87e964a843bdbedb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Mar 2015 01:25:36 +0000 Subject: [PATCH 200/411] ASoC: rsnd: add regmap_config::name for debugfs Renesas sound driver needs SSI/SRC/DVC regmaps, but it didn't have regmap_config::name for devm_regmap_init_mmio(). Thus, debugfs initialization code tried to use same driver name many times, and failed. This patch adds eacy own name for regmap_config::name Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 0273724a266885..a17a504d93b533 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -150,6 +150,7 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, regc.reg_bits = 32; regc.val_bits = 32; regc.reg_stride = 4; + regc.name = name; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, name); if (!res) From d3ef70543429b754dacc87fc68c30c2c34502337 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 11 Mar 2015 11:42:44 +0800 Subject: [PATCH 201/411] ASoC: rt5670: Add IRQ function This patch adds the IRQ function support of rt5670. We use a flag named dev_gpio in platform data to inform codec driver if the IRQ function is used or not. Also, we export rt5670_set_jack_detect for machine driver to pass the jack point. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- include/sound/rt5670.h | 1 + sound/soc/codecs/rt5670.c | 176 +++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5670.h | 4 + 3 files changed, 179 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h index bd311197a3b575..b7d60510819b02 100644 --- a/include/sound/rt5670.h +++ b/include/sound/rt5670.h @@ -14,6 +14,7 @@ struct rt5670_platform_data { int jd_mode; bool in2_diff; + bool dev_gpio; bool dmic_en; unsigned int dmic1_data_pin; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 1a6a9c4dc87983..a900db5fb1d917 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -403,6 +403,171 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg) } } +/** + * rt5670_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int val; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_DET_MODE | RT5670_CBJ_MN_JD, + RT5670_CBJ_MN_JD); + snd_soc_write(codec, RT5670_GPIO_CTRL2, 0x0004); + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + snd_soc_update_bits(codec, RT5670_CJ_CTRL1, + RT5670_CBJ_BST1_EN, RT5670_CBJ_BST1_EN); + snd_soc_write(codec, RT5670_JD_CTRL3, 0x00f0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, RT5670_CBJ_MN_JD); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, 0); + msleep(300); + val = snd_soc_read(codec, RT5670_CJ_CTRL3) & 0x7; + if (val == 0x1 || val == 0x2) { + rt5670->jack_type = SND_JACK_HEADSET; + /* for push button */ + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5670_IL_CMD, 0x40, 0x40); + snd_soc_read(codec, RT5670_IL_CMD); + } else { + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = SND_JACK_HEADPHONE; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + } else { + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x0); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = 0; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + return rt5670->jack_type; +} + +static int rt5670_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5670_IL_CMD); + btn_type = val & 0xff80; + snd_soc_write(codec, RT5670_IL_CMD, val); + if (btn_type != 0) { + msleep(20); + val = snd_soc_read(codec, RT5670_IL_CMD); + snd_soc_write(codec, RT5670_IL_CMD, val); + } + + return btn_type; +} + +static int rt5670_irq_detection(void *data) +{ + struct rt5670_priv *rt5670 = (struct rt5670_priv *)data; + struct snd_soc_jack_gpio *gpio = &rt5670->hp_gpio; + struct snd_soc_jack *jack = rt5670->jack; + int val, btn_type, report = jack->status; + + if (rt5670->pdata.jd_mode == 1) /* 2 port */ + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0070; + else + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0020; + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5670->jack_type == 0) { + report = rt5670_headset_detect(rt5670->codec, 1); + /* for push button and jack out */ + gpio->debounce_time = 25; + break; + } + btn_type = 0; + if (snd_soc_read(rt5670->codec, RT5670_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5670_button_detect(rt5670->codec); + switch (btn_type) { + case 0x2000: /* up */ + report |= SND_JACK_BTN_1; + break; + case 0x0400: /* center */ + report |= SND_JACK_BTN_0; + break; + case 0x0080: /* down */ + report |= SND_JACK_BTN_2; + break; + default: + dev_err(rt5670->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5670->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5670->codec, RT5670_INT_IRQ_ST, 0x1, 0x0); + rt5670_headset_detect(rt5670->codec, 0); + gpio->debounce_time = 150; /* for jack in */ + break; + default: + break; + } + + return report; +} + +int rt5670_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5670->jack = jack; + rt5670->hp_gpio.gpiod_dev = codec->dev; + rt5670->hp_gpio.name = "headphone detect"; + rt5670->hp_gpio.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2; + rt5670->hp_gpio.debounce_time = 150; + rt5670->hp_gpio.wake = true; + rt5670->hp_gpio.data = (struct rt5670_priv *)rt5670; + rt5670->hp_gpio.jack_status_check = rt5670_irq_detection; + + ret = snd_soc_jack_add_gpios(rt5670->jack, 1, + &rt5670->hp_gpio); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_set_jack_detect); + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -2506,6 +2671,7 @@ static int rt5670_remove(struct snd_soc_codec *codec) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5670->regmap, RT5670_RESET, 0); + snd_soc_jack_free_gpios(rt5670->jack, 1, &rt5670->hp_gpio); return 0; } @@ -2665,6 +2831,7 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (dmi_check_system(dmi_platform_intel_braswell)) { rt5670->pdata.dmic_en = true; rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.dev_gpio = true; rt5670->pdata.jd_mode = 1; } @@ -2706,12 +2873,17 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5670->regmap, RT5670_IN2, RT5670_IN_DF2, RT5670_IN_DF2); - if (i2c->irq) { + if (rt5670->pdata.dev_gpio) { + /* for push button */ + regmap_write(rt5670->regmap, RT5670_IL_CMD, 0x0000); + regmap_write(rt5670->regmap, RT5670_IL_CMD2, 0x0010); + regmap_write(rt5670->regmap, RT5670_IL_CMD3, 0x0014); + /* for irq */ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); - + regmap_update_bits(rt5670->regmap, RT5670_DIG_MISC, 0x8, 0x8); } if (rt5670->pdata.jd_mode) { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 0a67adbcfbc37f..0f3255aeeb9b73 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1988,6 +1988,8 @@ struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; struct regmap *regmap; + struct snd_soc_jack *jack; + struct snd_soc_jack_gpio hp_gpio; int sysclk; int sysclk_src; @@ -2004,4 +2006,6 @@ struct rt5670_priv { int jack_type; }; +int rt5670_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); #endif /* __RT5670_H__ */ From cc3c340d28b9f730fdc6bc5caa77e3bbd1e2377c Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 11 Mar 2015 11:42:45 +0800 Subject: [PATCH 202/411] ASoC: rt5670: export jack suspend/resume APIs We force enable "Mic Det Power" when a jack is inserted. Also, we set codec idle_bias_off = true. As a result, codec driver will not suspend as we expect. On Braswell, we don't need the jack detection when suspend but need it after resume, so export the jack suspend/resume APIs which are provided for machine driver to control during suspend/resume. Signed-off-by: Jie Yang Signed-off-by: Jin Yao Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ sound/soc/codecs/rt5670.h | 3 +++ 2 files changed, 21 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a900db5fb1d917..91d2069a931364 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -461,6 +461,24 @@ static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert) return rt5670->jack_type; } +void rt5670_jack_suspend(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + rt5670->jack_type_saved = rt5670->jack_type; + rt5670_headset_detect(codec, 0); +} +EXPORT_SYMBOL_GPL(rt5670_jack_suspend); + +void rt5670_jack_resume(struct snd_soc_codec *codec) +{ + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (rt5670->jack_type_saved) + rt5670_headset_detect(codec, 1); +} +EXPORT_SYMBOL_GPL(rt5670_jack_resume); + static int rt5670_button_detect(struct snd_soc_codec *codec) { int btn_type, val; diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 0f3255aeeb9b73..dc2b46236c5cb5 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -2004,8 +2004,11 @@ struct rt5670_priv { int dsp_sw; /* expected parameter setting */ int dsp_rate; int jack_type; + int jack_type_saved; }; +void rt5670_jack_suspend(struct snd_soc_codec *codec); +void rt5670_jack_resume(struct snd_soc_codec *codec); int rt5670_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); #endif /* __RT5670_H__ */ From 9449d39b990d6d3d6386fbb92f3b86c808157b47 Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Tue, 10 Mar 2015 10:41:20 +0800 Subject: [PATCH 203/411] ASoC: Intel: add function to load firmware image Add a general method to load firmware image, and apply to base firmware image loading. With the method, the driver will support loading multiple different modules in order to support different features. Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp-priv.h | 13 +++++ sound/soc/intel/sst-firmware.c | 1 + sound/soc/intel/sst-haswell-dsp.c | 1 + sound/soc/intel/sst-haswell-ipc.c | 83 ++++++++++++++++++++++++++++--- sound/soc/intel/sst-haswell-ipc.h | 6 +++ 5 files changed, 98 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index b9da030e312dc6..396d54510350ed 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -172,6 +172,16 @@ struct sst_module_runtime_context { u32 *buffer; }; +/* + * Audio DSP Module State + */ +enum sst_module_state { + SST_MODULE_STATE_UNLOADED = 0, /* default state */ + SST_MODULE_STATE_LOADED, + SST_MODULE_STATE_INITIALIZED, /* and inactive */ + SST_MODULE_STATE_ACTIVE, +}; + /* * Audio DSP Generic Module. * @@ -203,6 +213,9 @@ struct sst_module { struct list_head list; /* DSP list of modules */ struct list_head list_fw; /* FW list of modules */ struct list_head runtime_list; /* list of runtime module objects*/ + + /* state */ + enum sst_module_state state; }; /* diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5f71ef607a57e4..5e5800897da220 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -498,6 +498,7 @@ struct sst_module *sst_module_new(struct sst_fw *sst_fw, sst_module->scratch_size = template->scratch_size; sst_module->persistent_size = template->persistent_size; sst_module->entry = template->entry; + sst_module->state = SST_MODULE_STATE_UNLOADED; INIT_LIST_HEAD(&sst_module->block_list); INIT_LIST_HEAD(&sst_module->runtime_list); diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 402b728c0a0647..8ad733befbbd97 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -169,6 +169,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, block = (void *)block + sizeof(*block) + block->size; } + mod->state = SST_MODULE_STATE_LOADED; return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 863a9ca34b8e37..ec688f598320bf 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1844,6 +1844,8 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) if (ret < 0) dev_err(dev, "error: audio DSP boot failure\n"); + sst_hsw_init_module_state(hsw); + ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { @@ -1886,6 +1888,74 @@ struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw) return hsw->dsp; } +void sst_hsw_init_module_state(struct sst_hsw *hsw) +{ + struct sst_module *module; + enum sst_hsw_module_id id; + + /* the base fw contains several modules */ + for (id = SST_HSW_MODULE_BASE_FW; id < SST_HSW_MAX_MODULE_ID; id++) { + module = sst_module_get_from_id(hsw->dsp, id); + if (module) + module->state = SST_MODULE_STATE_ACTIVE; + } +} + +int sst_hsw_module_load(struct sst_hsw *hsw, + u32 module_id, u32 instance_id, char *name) +{ + int ret = 0; + const struct firmware *fw = NULL; + struct sst_fw *hsw_sst_fw; + struct sst_module *module; + struct device *dev = hsw->dev; + struct sst_dsp *dsp = hsw->dsp; + + dev_dbg(dev, "sst_hsw_module_load id=%d, name='%s'", module_id, name); + + module = sst_module_get_from_id(dsp, module_id); + if (module == NULL) { + /* loading for the first time */ + if (module_id == SST_HSW_MODULE_BASE_FW) { + /* for base module: use fw requested in acpi probe */ + fw = dsp->pdata->fw; + if (!fw) { + dev_err(dev, "request Base fw failed\n"); + return -ENODEV; + } + } else { + /* try and load any other optional modules if they are + * available. Use dev_info instead of dev_err in case + * request firmware failed */ + ret = request_firmware(&fw, name, dev); + if (ret) { + dev_info(dev, "fw image %s not available(%d)\n", + name, ret); + return ret; + } + } + hsw_sst_fw = sst_fw_new(dsp, fw, hsw); + if (hsw_sst_fw == NULL) { + dev_err(dev, "error: failed to load firmware\n"); + ret = -ENOMEM; + goto out; + } + module = sst_module_get_from_id(dsp, module_id); + if (module == NULL) { + dev_err(dev, "error: no module %d in firmware %s\n", + module_id, name); + } + } else + dev_info(dev, "module %d (%s) already loaded\n", + module_id, name); +out: + /* release fw, but base fw should be released by acpi driver */ + if (fw && module_id != SST_HSW_MODULE_BASE_FW) + release_firmware(fw); + + return ret; +} + static struct sst_dsp_device hsw_dev = { .thread = hsw_irq_thread, .ops = &haswell_ops, @@ -1947,12 +2017,10 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* keep the DSP in reset state for base FW loading */ sst_dsp_reset(hsw->dsp); - hsw->sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw); - if (hsw->sst_fw == NULL) { - ret = -ENODEV; - dev_err(dev, "error: failed to load firmware\n"); + /* load base module and other modules in base firmware image */ + ret = sst_hsw_module_load(hsw, SST_HSW_MODULE_BASE_FW, 0, "Base"); + if (ret < 0) goto fw_err; - } /* allocate scratch mem regions */ ret = sst_block_alloc_scratch(hsw->dsp); @@ -1971,6 +2039,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) goto boot_err; } + /* init module state after boot */ + sst_hsw_init_module_state(hsw); + /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); @@ -1986,7 +2057,7 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) boot_err: sst_dsp_reset(hsw->dsp); - sst_fw_free(hsw->sst_fw); + sst_fw_free_all(hsw->dsp); fw_err: dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 858096041cb1bb..e071b3a54eaecb 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -467,6 +467,12 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata); struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); +/* fw module function */ +void sst_hsw_init_module_state(struct sst_hsw *hsw); + +int sst_hsw_module_load(struct sst_hsw *hsw, + u32 module_id, u32 instance_id, char *name); + /* runtime module management */ struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw, int mod_id, int offset); From 8c43fc2fdda0858879ee4dd7d7ed8e67890f699f Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Tue, 10 Mar 2015 10:41:21 +0800 Subject: [PATCH 204/411] ASoC: Intel: add function to load sound effect module waves Try to load module waves and allocate runtime blocks for it if the firmware image of module waves exists. Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 2 ++ sound/soc/intel/sst-haswell-ipc.c | 23 +++++++++++++++++++++-- sound/soc/intel/sst-haswell-ipc.h | 1 + sound/soc/intel/sst-haswell-pcm.c | 21 +++++++++++++++++++-- 4 files changed, 43 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 8ad733befbbd97..b3e957d4693365 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -100,6 +100,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, && module->type != SST_HSW_MODULE_PCM && module->type != SST_HSW_MODULE_PCM_REFERENCE && module->type != SST_HSW_MODULE_PCM_CAPTURE + && module->type != SST_HSW_MODULE_WAVES && module->type != SST_HSW_MODULE_LPAL) return 0; @@ -139,6 +140,7 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: + case SST_HSW_REGS: ram = dsp->addr.lpe; mod->offset = block->ram_offset; mod->type = SST_MEM_DRAM; diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index ec688f598320bf..63740e36dd2620 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1896,11 +1896,27 @@ void sst_hsw_init_module_state(struct sst_hsw *hsw) /* the base fw contains several modules */ for (id = SST_HSW_MODULE_BASE_FW; id < SST_HSW_MAX_MODULE_ID; id++) { module = sst_module_get_from_id(hsw->dsp, id); - if (module) - module->state = SST_MODULE_STATE_ACTIVE; + if (module) { + /* module waves is active only after being enabled */ + if (id == SST_HSW_MODULE_WAVES) + module->state = SST_MODULE_STATE_INITIALIZED; + else + module->state = SST_MODULE_STATE_ACTIVE; + } } } +bool sst_hsw_is_module_loaded(struct sst_hsw *hsw, u32 module_id) +{ + struct sst_module *module; + + module = sst_module_get_from_id(hsw->dsp, module_id); + if (module == NULL || module->state == SST_MODULE_STATE_UNLOADED) + return false; + else + return true; +} + int sst_hsw_module_load(struct sst_hsw *hsw, u32 module_id, u32 instance_id, char *name) { @@ -2022,6 +2038,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (ret < 0) goto fw_err; + /* try to load module waves */ + sst_hsw_module_load(hsw, SST_HSW_MODULE_WAVES, 0, "intel/IntcPP01.bin"); + /* allocate scratch mem regions */ ret = sst_block_alloc_scratch(hsw->dsp); if (ret < 0) diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index e071b3a54eaecb..208724b74cf7b5 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -469,6 +469,7 @@ struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); /* fw module function */ void sst_hsw_init_module_state(struct sst_hsw *hsw); +bool sst_hsw_is_module_loaded(struct sst_hsw *hsw, u32 module_id); int sst_hsw_module_load(struct sst_hsw *hsw, u32 module_id, u32 instance_id, char *name); diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 7e21e8f85885e4..a604cc4421110a 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -807,6 +807,17 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) pcm_data->runtime->persistent_offset; } + /* create runtime blocks for module waves */ + if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { + pcm_data = &pdata->pcm[HSW_PCM_COUNT-1][0]; + pcm_data->runtime = sst_hsw_runtime_module_create(hsw, + SST_HSW_MODULE_WAVES, pcm_data->persistent_offset); + if (pcm_data->runtime == NULL) + goto err; + pcm_data->persistent_offset = + pcm_data->runtime->persistent_offset; + } + return 0; err: @@ -820,12 +831,16 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) { + struct sst_hsw *hsw = pdata->hsw; struct hsw_pcm_data *pcm_data; int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; - + sst_hsw_runtime_module_free(pcm_data->runtime); + } + if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { + pcm_data = &pdata->pcm[HSW_PCM_COUNT-1][0]; sst_hsw_runtime_module_free(pcm_data->runtime); } } @@ -984,7 +999,9 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) } /* allocate runtime modules */ - hsw_pcm_create_modules(priv_data); + ret = hsw_pcm_create_modules(priv_data); + if (ret < 0) + goto err; /* enable runtime PM with auto suspend */ pm_runtime_set_autosuspend_delay(platform->dev, From e8e79ede44ec99e09f8604c23ee99dc25065a343 Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Tue, 10 Mar 2015 10:41:22 +0800 Subject: [PATCH 205/411] ASoC: Intel: add function to enable/disable sound effect module waves Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 175 ++++++++++++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 11 ++ 2 files changed, 186 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 63740e36dd2620..265d754a40907a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -79,6 +79,15 @@ #define IPC_LOG_ID_MASK (0xf << IPC_LOG_ID_SHIFT) #define IPC_LOG_ID(x) (x << IPC_LOG_ID_SHIFT) +/* Module Message */ +#define IPC_MODULE_OPERATION_SHIFT 20 +#define IPC_MODULE_OPERATION_MASK (0xf << IPC_MODULE_OPERATION_SHIFT) +#define IPC_MODULE_OPERATION(x) (x << IPC_MODULE_OPERATION_SHIFT) + +#define IPC_MODULE_ID_SHIFT 16 +#define IPC_MODULE_ID_MASK (0xf << IPC_MODULE_ID_SHIFT) +#define IPC_MODULE_ID(x) (x << IPC_MODULE_ID_SHIFT) + /* IPC message timeout (msecs) */ #define IPC_TIMEOUT_MSECS 300 #define IPC_BOOT_MSECS 200 @@ -115,6 +124,7 @@ enum ipc_glb_type { IPC_GLB_ENTER_DX_STATE = 12, IPC_GLB_GET_MIXER_STREAM_INFO = 13, /* Request mixer stream params */ IPC_GLB_DEBUG_LOG_MESSAGE = 14, /* Message to or from the debug logger. */ + IPC_GLB_MODULE_OPERATION = 15, /* Message to loadable fw module */ IPC_GLB_REQUEST_TRANSFER = 16, /* < Request Transfer for host */ IPC_GLB_MAX_IPC_MESSAGE_TYPE = 17, /* Maximum message number */ }; @@ -133,6 +143,16 @@ enum ipc_glb_reply { IPC_GLB_REPLY_SOURCE_NOT_STARTED = 10, /* Source was not started. */ }; +enum ipc_module_operation { + IPC_MODULE_NOTIFICATION = 0, + IPC_MODULE_ENABLE = 1, + IPC_MODULE_DISABLE = 2, + IPC_MODULE_GET_PARAMETER = 3, + IPC_MODULE_SET_PARAMETER = 4, + IPC_MODULE_GET_INFO = 5, + IPC_MODULE_MAX_MESSAGE +}; + /* Stream Message - Types */ enum ipc_str_operation { IPC_STR_RESET = 0, @@ -352,6 +372,16 @@ static inline u32 msg_get_notify_reason(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } +static inline u32 msg_get_module_operation(u32 msg) +{ + return (msg & IPC_MODULE_OPERATION_MASK) >> IPC_MODULE_OPERATION_SHIFT; +} + +static inline u32 msg_get_module_id(u32 msg) +{ + return (msg & IPC_MODULE_ID_MASK) >> IPC_MODULE_ID_SHIFT; +} + u32 create_channel_map(enum sst_hsw_channel_config config) { switch (config) { @@ -795,6 +825,31 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) return 1; } +static int hsw_module_message(struct sst_hsw *hsw, u32 header) +{ + u32 operation, module_id; + int handled = 0; + + operation = msg_get_module_operation(header); + module_id = msg_get_module_id(header); + dev_dbg(hsw->dev, "received module message header: 0x%8.8x\n", + header); + dev_dbg(hsw->dev, "operation: 0x%8.8x module_id: 0x%8.8x\n", + operation, module_id); + + switch (operation) { + case IPC_MODULE_NOTIFICATION: + dev_dbg(hsw->dev, "module notification received"); + handled = 1; + break; + default: + handled = hsw_process_reply(hsw, header); + break; + } + + return handled; +} + static int hsw_stream_message(struct sst_hsw *hsw, u32 header) { u32 stream_msg, stream_id, stage_type; @@ -890,6 +945,9 @@ static int hsw_process_notification(struct sst_hsw *hsw) case IPC_GLB_DEBUG_LOG_MESSAGE: handled = hsw_log_message(hsw, header); break; + case IPC_GLB_MODULE_OPERATION: + handled = hsw_module_message(hsw, header); + break; default: dev_err(hsw->dev, "error: unexpected type %d hdr 0x%8.8x\n", type, header); @@ -1917,6 +1975,17 @@ bool sst_hsw_is_module_loaded(struct sst_hsw *hsw, u32 module_id) return true; } +bool sst_hsw_is_module_active(struct sst_hsw *hsw, u32 module_id) +{ + struct sst_module *module; + + module = sst_module_get_from_id(hsw->dsp, module_id); + if (module != NULL && module->state == SST_MODULE_STATE_ACTIVE) + return true; + else + return false; +} + int sst_hsw_module_load(struct sst_hsw *hsw, u32 module_id, u32 instance_id, char *name) { @@ -1972,6 +2041,112 @@ int sst_hsw_module_load(struct sst_hsw *hsw, return ret; } +int sst_hsw_module_enable(struct sst_hsw *hsw, + u32 module_id, u32 instance_id) +{ + int ret; + u32 header = 0; + struct sst_hsw_ipc_module_config config; + struct sst_module *module; + struct sst_module_runtime *runtime; + struct device *dev = hsw->dev; + struct sst_dsp *dsp = hsw->dsp; + + if (!sst_hsw_is_module_loaded(hsw, module_id)) { + dev_dbg(dev, "module %d not loaded\n", module_id); + return 0; + } + + if (sst_hsw_is_module_active(hsw, module_id)) { + dev_info(dev, "module %d already enabled\n", module_id); + return 0; + } + + module = sst_module_get_from_id(dsp, module_id); + if (module == NULL) { + dev_err(dev, "module %d not valid\n", module_id); + return -ENXIO; + } + + runtime = sst_module_runtime_get_from_id(module, module_id); + if (runtime == NULL) { + dev_err(dev, "runtime %d not valid", module_id); + return -ENXIO; + } + + header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) | + IPC_MODULE_OPERATION(IPC_MODULE_ENABLE) | + IPC_MODULE_ID(module_id); + dev_dbg(dev, "module enable header: %x\n", header); + + config.map.module_entries_count = 1; + config.map.module_entries[0].module_id = module->id; + config.map.module_entries[0].entry_point = module->entry; + + config.persistent_mem.offset = + sst_dsp_get_offset(dsp, + runtime->persistent_offset, SST_MEM_DRAM); + config.persistent_mem.size = module->persistent_size; + + config.scratch_mem.offset = + sst_dsp_get_offset(dsp, + dsp->scratch_offset, SST_MEM_DRAM); + config.scratch_mem.size = module->scratch_size; + dev_dbg(dev, "mod %d enable p:%d @ %x, s:%d @ %x, ep: %x", + config.map.module_entries[0].module_id, + config.persistent_mem.size, + config.persistent_mem.offset, + config.scratch_mem.size, config.scratch_mem.offset, + config.map.module_entries[0].entry_point); + + ret = ipc_tx_message_wait(hsw, header, + &config, sizeof(config), NULL, 0); + if (ret < 0) + dev_err(dev, "ipc: module enable failed - %d\n", ret); + else + module->state = SST_MODULE_STATE_ACTIVE; + + return ret; +} + +int sst_hsw_module_disable(struct sst_hsw *hsw, + u32 module_id, u32 instance_id) +{ + int ret; + u32 header; + struct sst_module *module; + struct device *dev = hsw->dev; + struct sst_dsp *dsp = hsw->dsp; + + if (!sst_hsw_is_module_loaded(hsw, module_id)) { + dev_dbg(dev, "module %d not loaded\n", module_id); + return 0; + } + + if (!sst_hsw_is_module_active(hsw, module_id)) { + dev_info(dev, "module %d already disabled\n", module_id); + return 0; + } + + module = sst_module_get_from_id(dsp, module_id); + if (module == NULL) { + dev_err(dev, "module %d not valid\n", module_id); + return -ENXIO; + } + + header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) | + IPC_MODULE_OPERATION(IPC_MODULE_DISABLE) | + IPC_MODULE_ID(module_id); + + ret = ipc_tx_message_wait(hsw, header, NULL, 0, NULL, 0); + if (ret < 0) + dev_err(dev, "module disable failed - %d\n", ret); + else + module->state = SST_MODULE_STATE_INITIALIZED; + + return ret; +} + static struct sst_dsp_device hsw_dev = { .thread = hsw_irq_thread, .ops = &haswell_ops, diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 208724b74cf7b5..30c65b28fa6072 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -215,6 +215,12 @@ struct sst_hsw_fx_enable { struct sst_hsw_memory_info persistent_mem; } __attribute__((packed)); +struct sst_hsw_ipc_module_config { + struct sst_hsw_module_map map; + struct sst_hsw_memory_info persistent_mem; + struct sst_hsw_memory_info scratch_mem; +} __attribute__((packed)); + struct sst_hsw_get_fx_param { u32 parameter_id; u32 param_size; @@ -470,9 +476,14 @@ struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); /* fw module function */ void sst_hsw_init_module_state(struct sst_hsw *hsw); bool sst_hsw_is_module_loaded(struct sst_hsw *hsw, u32 module_id); +bool sst_hsw_is_module_active(struct sst_hsw *hsw, u32 module_id); int sst_hsw_module_load(struct sst_hsw *hsw, u32 module_id, u32 instance_id, char *name); +int sst_hsw_module_enable(struct sst_hsw *hsw, + u32 module_id, u32 instance_id); +int sst_hsw_module_disable(struct sst_hsw *hsw, + u32 module_id, u32 instance_id); /* runtime module management */ struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw, From b3f5dbec2f077be7ee392da3365d11e393a50038 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Wed, 11 Mar 2015 17:56:33 +0100 Subject: [PATCH 206/411] ASoC: ab8500-codec: don't export static symbol The semantic patch that fixes this problem is as follows: (http://coccinelle.lip6.fr/) // @r@ type T; identifier f; @@ static T f (...) { ... } @@ identifier r.f; declarer name EXPORT_SYMBOL_GPL; @@ -EXPORT_SYMBOL_GPL(f); // Signed-off-by: Julia Lawall Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 7895689588daa8..88ca9cb0ce7937 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2003,7 +2003,6 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, return 0; } -EXPORT_SYMBOL_GPL(ab8500_audio_setup_mics); static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec, enum ear_cm_voltage ear_cmv) @@ -2036,7 +2035,6 @@ static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec, return 0; } -EXPORT_SYMBOL_GPL(ab8500_audio_set_ear_cmv); static int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai, unsigned int delay) From 1ff2765182d1969947dfb50f4e42ebd7e83699fd Mon Sep 17 00:00:00 2001 From: Anish Kumar Date: Mon, 9 Mar 2015 15:50:34 -0700 Subject: [PATCH 207/411] ASoC: Add max98925 codec driver Signed-off-by: Anish Kumar Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/max98925.txt | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98925.c | 655 ++++++++++++++ sound/soc/codecs/max98925.h | 832 ++++++++++++++++++ 5 files changed, 1515 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98925.txt create mode 100644 sound/soc/codecs/max98925.c create mode 100644 sound/soc/codecs/max98925.h diff --git a/Documentation/devicetree/bindings/sound/max98925.txt b/Documentation/devicetree/bindings/sound/max98925.txt new file mode 100644 index 00000000000000..27be63e2aa0de0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98925.txt @@ -0,0 +1,22 @@ +max98925 audio CODEC + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98925" + + - vmon-slot-no : slot number used to send voltage information + + - imon-slot-no : slot number used to send current information + + - reg : the I2C address of the device for I2C + +Example: + +codec: max98925@1a { + compatible = "maxim,max98925"; + vmon-slot-no = <0>; + imon-slot-no = <2>; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea9f0e31f9d40d..0ccce5a1f4fac5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -70,6 +70,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98925 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C @@ -460,6 +461,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98925 + tristate + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666d187a0e..333ee3a4efa11d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o @@ -247,6 +248,7 @@ obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o +obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c new file mode 100644 index 00000000000000..74f4f0b60108fd --- /dev/null +++ b/sound/soc/codecs/max98925.c @@ -0,0 +1,655 @@ +/* + * max98925.c -- ALSA SoC Stereo MAX98925 driver + * Copyright 2013-15 Maxim Integrated Products + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "max98925.h" + +static const char *const dai_text[] = { + "Left", "Right", "LeftRight", "LeftRightDiv2", +}; + +static const char * const max98925_boost_voltage_text[] = { + "8.5V", "8.25V", "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V" +}; + +static SOC_ENUM_SINGLE_DECL(max98925_boost_voltage, + MAX98925_CONFIGURATION, M98925_BST_VOUT_SHIFT, + max98925_boost_voltage_text); + +static const char *const hpf_text[] = { + "Disable", "DC Block", "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static struct reg_default max98925_reg[] = { + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x06 }, /* DAI Clock Mode 1 */ + { 0x1B, 0xC0 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ +}; + +static const struct soc_enum max98925_dai_enum = + SOC_ENUM_SINGLE(MAX98925_GAIN, 5, ARRAY_SIZE(dai_text), dai_text); + +static const struct soc_enum max98925_hpf_enum = + SOC_ENUM_SINGLE(MAX98925_FILTERS, 0, ARRAY_SIZE(hpf_text), hpf_text); + +static const struct snd_kcontrol_new max98925_hpf_sel_mux = + SOC_DAPM_ENUM("Rc Filter MUX Mux", max98925_hpf_enum); + +static const struct snd_kcontrol_new max98925_dai_sel_mux = + SOC_DAPM_ENUM("DAI IN MUX Mux", max98925_dai_enum); + +static int max98925_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(max98925->regmap, + MAX98925_BLOCK_ENABLE, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, + M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(max98925->regmap, + MAX98925_BLOCK_ENABLE, M98925_BST_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, 0); + break; + default: + return 0; + } + return 0; +} + +static const struct snd_soc_dapm_widget max98925_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("DAI IN MUX", SND_SOC_NOPM, 0, 0, + &max98925_dai_sel_mux), + SND_SOC_DAPM_MUX("Rc Filter MUX", SND_SOC_NOPM, 0, 0, + &max98925_hpf_sel_mux), + SND_SOC_DAPM_DAC_E("Amp Enable", NULL, MAX98925_BLOCK_ENABLE, + M98925_SPK_EN_SHIFT, 0, max98925_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98925_GLOBAL_ENABLE, + M98925_EN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("BE_OUT"), +}; + +static const struct snd_soc_dapm_route max98925_audio_map[] = { + {"DAI IN MUX", "Left", "DAI_OUT"}, + {"DAI IN MUX", "Right", "DAI_OUT"}, + {"DAI IN MUX", "LeftRight", "DAI_OUT"}, + {"DAI IN MUX", "LeftRightDiv2", "DAI_OUT"}, + {"Rc Filter MUX", "Disable", "DAI IN MUX"}, + {"Rc Filter MUX", "DC Block", "DAI IN MUX"}, + {"Rc Filter MUX", "100Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "200Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "400Hz", "DAI IN MUX"}, + {"Rc Filter MUX", "800Hz", "DAI IN MUX"}, + {"Amp Enable", NULL, "Rc Filter MUX"}, + {"BE_OUT", NULL, "Amp Enable"}, + {"BE_OUT", NULL, "Global Enable"}, +}; + +static bool max98925_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_VBAT_DATA: + case MAX98925_VBST_DATA: + case MAX98925_LIVE_STATUS0: + case MAX98925_LIVE_STATUS1: + case MAX98925_LIVE_STATUS2: + case MAX98925_STATE0: + case MAX98925_STATE1: + case MAX98925_STATE2: + case MAX98925_FLAG0: + case MAX98925_FLAG1: + case MAX98925_FLAG2: + case MAX98925_REV_VERSION: + return true; + default: + return false; + } +} + +static bool max98925_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_IRQ_CLEAR0: + case MAX98925_IRQ_CLEAR1: + case MAX98925_IRQ_CLEAR2: + case MAX98925_ALC_HOLD_RLS: + return false; + default: + return true; + } +} + +DECLARE_TLV_DB_SCALE(max98925_spk_tlv, -600, 100, 0); + +static const struct snd_kcontrol_new max98925_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98925_GAIN, + M98925_SPK_GAIN_SHIFT, (1<= rate) { + *value = rate_table[i].sr; + *n = rate_table[i].divisors[clock][0]; + *m = rate_table[i].divisors[clock][1]; + ret = 0; + break; + } + } + dev_dbg(codec->dev, "%s: sample rate is %d, returning %d\n", + __func__, rate_table[i].rate, *value); + return ret; +} + +static void max98925_set_sense_data(struct max98925_priv *max98925) +{ + /* set VMON slots */ + regmap_update_bits(max98925->regmap, + MAX98925_DOUT_CFG_VMON, + M98925_DAI_VMON_EN_MASK, M98925_DAI_VMON_EN_MASK); + regmap_update_bits(max98925->regmap, + MAX98925_DOUT_CFG_VMON, + M98925_DAI_VMON_SLOT_MASK, + max98925->v_slot << M98925_DAI_VMON_SLOT_SHIFT); + /* set IMON slots */ + regmap_update_bits(max98925->regmap, + MAX98925_DOUT_CFG_IMON, + M98925_DAI_IMON_EN_MASK, M98925_DAI_IMON_EN_MASK); + regmap_update_bits(max98925->regmap, + MAX98925_DOUT_CFG_IMON, + M98925_DAI_IMON_SLOT_MASK, + max98925->i_slot << M98925_DAI_IMON_SLOT_SHIFT); +} + +static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int invert = 0; + + dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* set DAI to slave mode */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, 0); + max98925_set_sense_data(max98925); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* + * set left channel DAI to master mode, + * right channel always slave + */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, M98925_DAI_MAS_MASK); + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + invert = M98925_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + invert = M98925_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert = M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98925->regmap, MAX98925_FORMAT, + M98925_DAI_BCI_MASK | M98925_DAI_BCI_MASK, invert); + return 0; +} + +static int max98925_set_clock(struct max98925_priv *max98925, + struct snd_pcm_hw_params *params) +{ + unsigned int dai_sr = 0, clock, mdll, n, m; + struct snd_soc_codec *codec = max98925->codec; + int rate = params_rate(params); + /* BCLK/LRCLK ratio calculation */ + int blr_clk_ratio = params_channels(params) * max98925->ch_size; + + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (max98925->sysclk) { + case 6000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx16; + break; + case 11289600: + clock = 1; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12288000: + clock = 2; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + default: + dev_info(max98925->codec->dev, "unsupported sysclk %d\n", + max98925->sysclk); + return -EINVAL; + } + + if (max98925_rate_value(codec, rate, clock, &dai_sr, &n, &m)) + return -EINVAL; + + /* set DAI_SR to correct LRCLK frequency */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE2, + M98925_DAI_SR_MASK, dai_sr << M98925_DAI_SR_SHIFT); + /* set DAI m divider */ + regmap_write(max98925->regmap, + MAX98925_DAI_CLK_DIV_M_MSBS, m >> 8); + regmap_write(max98925->regmap, + MAX98925_DAI_CLK_DIV_M_LSBS, m & 0xFF); + /* set DAI n divider */ + regmap_write(max98925->regmap, + MAX98925_DAI_CLK_DIV_N_MSBS, n >> 8); + regmap_write(max98925->regmap, + MAX98925_DAI_CLK_DIV_N_LSBS, n & 0xFF); + /* set MDLL */ + regmap_update_bits(max98925->regmap, MAX98925_DAI_CLK_MODE1, + M98925_MDLL_MULT_MASK, mdll << M98925_MDLL_MULT_SHIFT); + return 0; +} + +static int max98925_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (snd_pcm_format_width(params_format(params))) { + case 16: + regmap_update_bits(max98925->regmap, + MAX98925_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_16); + max98925->ch_size = 16; + break; + case 24: + regmap_update_bits(max98925->regmap, + MAX98925_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + max98925->ch_size = 32; + break; + case 32: + regmap_update_bits(max98925->regmap, + MAX98925_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + max98925->ch_size = 32; + break; + default: + pr_err("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: format supported %d", + __func__, params_format(params)); + return max98925_set_clock(max98925, params); +} + +static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case 0: + /* use MCLK for Left channel, right channel always BCLK */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, 0); + break; + case 1: + /* configure dai clock source to BCLK instead of MCLK */ + regmap_update_bits(max98925->regmap, + MAX98925_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, + M98925_DAI_CLK_SOURCE_MASK); + break; + default: + return -EINVAL; + } + max98925->sysclk = freq; + return 0; +} + +#define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops max98925_dai_ops = { + .set_sysclk = max98925_dai_set_sysclk, + .set_fmt = max98925_dai_set_fmt, + .hw_params = max98925_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98925_dai[] = { + { + .name = "max98925-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .ops = &max98925_dai_ops, + } +}; + +static int max98925_probe(struct snd_soc_codec *codec) +{ + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + max98925->codec = codec; + codec->control_data = max98925->regmap; + regmap_write(max98925->regmap, MAX98925_GLOBAL_ENABLE, 0x00); + /* It's not the default but we need to set DAI_DLY */ + regmap_write(max98925->regmap, + MAX98925_FORMAT, M98925_DAI_DLY_MASK); + regmap_write(max98925->regmap, MAX98925_TDM_SLOT_SELECT, 0xC8); + regmap_write(max98925->regmap, MAX98925_DOUT_HIZ_CFG1, 0xFF); + regmap_write(max98925->regmap, MAX98925_DOUT_HIZ_CFG2, 0xFF); + regmap_write(max98925->regmap, MAX98925_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98925->regmap, MAX98925_DOUT_HIZ_CFG4, 0xF0); + regmap_write(max98925->regmap, MAX98925_FILTERS, 0xD8); + regmap_write(max98925->regmap, MAX98925_ALC_CONFIGURATION, 0xF8); + regmap_write(max98925->regmap, MAX98925_CONFIGURATION, 0xF0); + /* Disable ALC muting */ + regmap_write(max98925->regmap, MAX98925_BOOST_LIMITER, 0xF8); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98925 = { + .probe = max98925_probe, + .controls = max98925_snd_controls, + .num_controls = ARRAY_SIZE(max98925_snd_controls), + .dapm_routes = max98925_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98925_audio_map), + .dapm_widgets = max98925_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98925_dapm_widgets), +}; + +static struct regmap_config max98925_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98925_REV_VERSION, + .reg_defaults = max98925_reg, + .num_reg_defaults = ARRAY_SIZE(max98925_reg), + .volatile_reg = max98925_volatile_register, + .readable_reg = max98925_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98925_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + int ret, reg; + u32 value; + struct max98925_priv *max98925; + + max98925 = devm_kzalloc(&i2c->dev, + sizeof(*max98925), GFP_KERNEL); + if (!max98925) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98925); + max98925->regmap = devm_regmap_init_i2c(i2c, &max98925_regmap); + if (IS_ERR(max98925->regmap)) { + ret = PTR_ERR(max98925->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + goto err_out; + } + + if (!of_property_read_u32(i2c->dev.of_node, "vmon-slot-no", &value)) { + if (value > M98925_DAI_VMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "vmon slot number is wrong:\n"); + return -EINVAL; + } + max98925->v_slot = value; + } + if (!of_property_read_u32(i2c->dev.of_node, "imon-slot-no", &value)) { + if (value > M98925_DAI_IMON_SLOT_1E_1F) { + dev_err(&i2c->dev, "imon slot number is wrong:\n"); + return -EINVAL; + } + max98925->i_slot = value; + } + ret = regmap_read(max98925->regmap, + MAX98925_REV_VERSION, ®); + if ((ret < 0) || + ((reg != MAX98925_VERSION) && + (reg != MAX98925_VERSION1))) { + dev_err(&i2c->dev, + "device initialization error (%d 0x%02X)\n", + ret, reg); + goto err_out; + } + dev_info(&i2c->dev, "device version 0x%02X\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98925, + max98925_dai, ARRAY_SIZE(max98925_dai)); + if (ret < 0) + dev_err(&i2c->dev, + "Failed to register codec: %d\n", ret); +err_out: + return ret; +} + +static int max98925_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98925_i2c_id[] = { + { "max98925", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98925_i2c_id); + +static const struct of_device_id max98925_of_match[] = { + { .compatible = "maxim,max98925", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98925_of_match); + +static struct i2c_driver max98925_i2c_driver = { + .driver = { + .name = "max98925", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98925_of_match), + .pm = NULL, + }, + .probe = max98925_i2c_probe, + .remove = max98925_i2c_remove, + .id_table = max98925_i2c_id, +}; + +module_i2c_driver(max98925_i2c_driver) + +MODULE_DESCRIPTION("ALSA SoC MAX98925 driver"); +MODULE_AUTHOR("Ralph Birt , Anish kumar "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98925.h b/sound/soc/codecs/max98925.h new file mode 100644 index 00000000000000..3783248f2780f3 --- /dev/null +++ b/sound/soc/codecs/max98925.h @@ -0,0 +1,832 @@ +/* + * max98925.h -- MAX98925 ALSA SoC Audio driver + * + * Copyright 2013-2015 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98925_H +#define _MAX98925_H + +#define MAX98925_VERSION 0x51 +#define MAX98925_VERSION1 0x80 +#define MAX98925_VBAT_DATA 0x00 +#define MAX98925_VBST_DATA 0x01 +#define MAX98925_LIVE_STATUS0 0x02 +#define MAX98925_LIVE_STATUS1 0x03 +#define MAX98925_LIVE_STATUS2 0x04 +#define MAX98925_STATE0 0x05 +#define MAX98925_STATE1 0x06 +#define MAX98925_STATE2 0x07 +#define MAX98925_FLAG0 0x08 +#define MAX98925_FLAG1 0x09 +#define MAX98925_FLAG2 0x0A +#define MAX98925_IRQ_ENABLE0 0x0B +#define MAX98925_IRQ_ENABLE1 0x0C +#define MAX98925_IRQ_ENABLE2 0x0D +#define MAX98925_IRQ_CLEAR0 0x0E +#define MAX98925_IRQ_CLEAR1 0x0F +#define MAX98925_IRQ_CLEAR2 0x10 +#define MAX98925_MAP0 0x11 +#define MAX98925_MAP1 0x12 +#define MAX98925_MAP2 0x13 +#define MAX98925_MAP3 0x14 +#define MAX98925_MAP4 0x15 +#define MAX98925_MAP5 0x16 +#define MAX98925_MAP6 0x17 +#define MAX98925_MAP7 0x18 +#define MAX98925_MAP8 0x19 +#define MAX98925_DAI_CLK_MODE1 0x1A +#define MAX98925_DAI_CLK_MODE2 0x1B +#define MAX98925_DAI_CLK_DIV_M_MSBS 0x1C +#define MAX98925_DAI_CLK_DIV_M_LSBS 0x1D +#define MAX98925_DAI_CLK_DIV_N_MSBS 0x1E +#define MAX98925_DAI_CLK_DIV_N_LSBS 0x1F +#define MAX98925_FORMAT 0x20 +#define MAX98925_TDM_SLOT_SELECT 0x21 +#define MAX98925_DOUT_CFG_VMON 0x22 +#define MAX98925_DOUT_CFG_IMON 0x23 +#define MAX98925_DOUT_CFG_VBAT 0x24 +#define MAX98925_DOUT_CFG_VBST 0x25 +#define MAX98925_DOUT_CFG_FLAG 0x26 +#define MAX98925_DOUT_HIZ_CFG1 0x27 +#define MAX98925_DOUT_HIZ_CFG2 0x28 +#define MAX98925_DOUT_HIZ_CFG3 0x29 +#define MAX98925_DOUT_HIZ_CFG4 0x2A +#define MAX98925_DOUT_DRV_STRENGTH 0x2B +#define MAX98925_FILTERS 0x2C +#define MAX98925_GAIN 0x2D +#define MAX98925_GAIN_RAMPING 0x2E +#define MAX98925_SPK_AMP 0x2F +#define MAX98925_THRESHOLD 0x30 +#define MAX98925_ALC_ATTACK 0x31 +#define MAX98925_ALC_ATTEN_RLS 0x32 +#define MAX98925_ALC_HOLD_RLS 0x33 +#define MAX98925_ALC_CONFIGURATION 0x34 +#define MAX98925_BOOST_CONVERTER 0x35 +#define MAX98925_BLOCK_ENABLE 0x36 +#define MAX98925_CONFIGURATION 0x37 +#define MAX98925_GLOBAL_ENABLE 0x38 +#define MAX98925_BOOST_LIMITER 0x3A +#define MAX98925_REV_VERSION 0xFF + +#define MAX98925_REG_CNT (MAX98925_R03A_BOOST_LIMITER+1) + +/* MAX98925 Register Bit Fields */ + +/* MAX98925_R002_LIVE_STATUS0 */ +#define M98925_THERMWARN_STATUS_MASK (1<<3) +#define M98925_THERMWARN_STATUS_SHIFT 3 +#define M98925_THERMWARN_STATUS_WIDTH 1 +#define M98925_THERMSHDN_STATUS_MASK (1<<1) +#define M98925_THERMSHDN_STATUS_SHIFT 1 +#define M98925_THERMSHDN_STATUS_WIDTH 1 + +/* MAX98925_R003_LIVE_STATUS1 */ +#define M98925_SPKCURNT_STATUS_MASK (1<<5) +#define M98925_SPKCURNT_STATUS_SHIFT 5 +#define M98925_SPKCURNT_STATUS_WIDTH 1 +#define M98925_WATCHFAIL_STATUS_MASK (1<<4) +#define M98925_WATCHFAIL_STATUS_SHIFT 4 +#define M98925_WATCHFAIL_STATUS_WIDTH 1 +#define M98925_ALCINFH_STATUS_MASK (1<<3) +#define M98925_ALCINFH_STATUS_SHIFT 3 +#define M98925_ALCINFH_STATUS_WIDTH 1 +#define M98925_ALCACT_STATUS_MASK (1<<2) +#define M98925_ALCACT_STATUS_SHIFT 2 +#define M98925_ALCACT_STATUS_WIDTH 1 +#define M98925_ALCMUT_STATUS_MASK (1<<1) +#define M98925_ALCMUT_STATUS_SHIFT 1 +#define M98925_ALCMUT_STATUS_WIDTH 1 +#define M98925_ACLP_STATUS_MASK (1<<0) +#define M98925_ACLP_STATUS_SHIFT 0 +#define M98925_ACLP_STATUS_WIDTH 1 + +/* MAX98925_R004_LIVE_STATUS2 */ +#define M98925_SLOTOVRN_STATUS_MASK (1<<6) +#define M98925_SLOTOVRN_STATUS_SHIFT 6 +#define M98925_SLOTOVRN_STATUS_WIDTH 1 +#define M98925_INVALSLOT_STATUS_MASK (1<<5) +#define M98925_INVALSLOT_STATUS_SHIFT 5 +#define M98925_INVALSLOT_STATUS_WIDTH 1 +#define M98925_SLOTCNFLT_STATUS_MASK (1<<4) +#define M98925_SLOTCNFLT_STATUS_SHIFT 4 +#define M98925_SLOTCNFLT_STATUS_WIDTH 1 +#define M98925_VBSTOVFL_STATUS_MASK (1<<3) +#define M98925_VBSTOVFL_STATUS_SHIFT 3 +#define M98925_VBSTOVFL_STATUS_WIDTH 1 +#define M98925_VBATOVFL_STATUS_MASK (1<<2) +#define M98925_VBATOVFL_STATUS_SHIFT 2 +#define M98925_VBATOVFL_STATUS_WIDTH 1 +#define M98925_IMONOVFL_STATUS_MASK (1<<1) +#define M98925_IMONOVFL_STATUS_SHIFT 1 +#define M98925_IMONOVFL_STATUS_WIDTH 1 +#define M98925_VMONOVFL_STATUS_MASK (1<<0) +#define M98925_VMONOVFL_STATUS_SHIFT 0 +#define M98925_VMONOVFL_STATUS_WIDTH 1 + +/* MAX98925_R005_STATE0 */ +#define M98925_THERMWARN_END_STATE_MASK (1<<3) +#define M98925_THERMWARN_END_STATE_SHIFT 3 +#define M98925_THERMWARN_END_STATE_WIDTH 1 +#define M98925_THERMWARN_BGN_STATE_MASK (1<<2) +#define M98925_THERMWARN_BGN_STATE_SHIFT 1 +#define M98925_THERMWARN_BGN_STATE_WIDTH 1 +#define M98925_THERMSHDN_END_STATE_MASK (1<<1) +#define M98925_THERMSHDN_END_STATE_SHIFT 1 +#define M98925_THERMSHDN_END_STATE_WIDTH 1 +#define M98925_THERMSHDN_BGN_STATE_MASK (1<<0) +#define M98925_THERMSHDN_BGN_STATE_SHIFT 0 +#define M98925_THERMSHDN_BGN_STATE_WIDTH 1 + +/* MAX98925_R006_STATE1 */ +#define M98925_SPRCURNT_STATE_MASK (1<<5) +#define M98925_SPRCURNT_STATE_SHIFT 5 +#define M98925_SPRCURNT_STATE_WIDTH 1 +#define M98925_WATCHFAIL_STATE_MASK (1<<4) +#define M98925_WATCHFAIL_STATE_SHIFT 4 +#define M98925_WATCHFAIL_STATE_WIDTH 1 +#define M98925_ALCINFH_STATE_MASK (1<<3) +#define M98925_ALCINFH_STATE_SHIFT 3 +#define M98925_ALCINFH_STATE_WIDTH 1 +#define M98925_ALCACT_STATE_MASK (1<<2) +#define M98925_ALCACT_STATE_SHIFT 2 +#define M98925_ALCACT_STATE_WIDTH 1 +#define M98925_ALCMUT_STATE_MASK (1<<1) +#define M98925_ALCMUT_STATE_SHIFT 1 +#define M98925_ALCMUT_STATE_WIDTH 1 +#define M98925_ALCP_STATE_MASK (1<<0) +#define M98925_ALCP_STATE_SHIFT 0 +#define M98925_ALCP_STATE_WIDTH 1 + +/* MAX98925_R007_STATE2 */ +#define M98925_SLOTOVRN_STATE_MASK (1<<6) +#define M98925_SLOTOVRN_STATE_SHIFT 6 +#define M98925_SLOTOVRN_STATE_WIDTH 1 +#define M98925_INVALSLOT_STATE_MASK (1<<5) +#define M98925_INVALSLOT_STATE_SHIFT 5 +#define M98925_INVALSLOT_STATE_WIDTH 1 +#define M98925_SLOTCNFLT_STATE_MASK (1<<4) +#define M98925_SLOTCNFLT_STATE_SHIFT 4 +#define M98925_SLOTCNFLT_STATE_WIDTH 1 +#define M98925_VBSTOVFL_STATE_MASK (1<<3) +#define M98925_VBSTOVFL_STATE_SHIFT 3 +#define M98925_VBSTOVFL_STATE_WIDTH 1 +#define M98925_VBATOVFL_STATE_MASK (1<<2) +#define M98925_VBATOVFL_STATE_SHIFT 2 +#define M98925_VBATOVFL_STATE_WIDTH 1 +#define M98925_IMONOVFL_STATE_MASK (1<<1) +#define M98925_IMONOVFL_STATE_SHIFT 1 +#define M98925_IMONOVFL_STATE_WIDTH 1 +#define M98925_VMONOVFL_STATE_MASK (1<<0) +#define M98925_VMONOVFL_STATE_SHIFT 0 +#define M98925_VMONOVFL_STATE_WIDTH 1 + +/* MAX98925_R008_FLAG0 */ +#define M98925_THERMWARN_END_FLAG_MASK (1<<3) +#define M98925_THERMWARN_END_FLAG_SHIFT 3 +#define M98925_THERMWARN_END_FLAG_WIDTH 1 +#define M98925_THERMWARN_BGN_FLAG_MASK (1<<2) +#define M98925_THERMWARN_BGN_FLAG_SHIFT 2 +#define M98925_THERMWARN_BGN_FLAG_WIDTH 1 +#define M98925_THERMSHDN_END_FLAG_MASK (1<<1) +#define M98925_THERMSHDN_END_FLAG_SHIFT 1 +#define M98925_THERMSHDN_END_FLAG_WIDTH 1 +#define M98925_THERMSHDN_BGN_FLAG_MASK (1<<0) +#define M98925_THERMSHDN_BGN_FLAG_SHIFT 0 +#define M98925_THERMSHDN_BGN_FLAG_WIDTH 1 + +/* MAX98925_R009_FLAG1 */ +#define M98925_SPKCURNT_FLAG_MASK (1<<5) +#define M98925_SPKCURNT_FLAG_SHIFT 5 +#define M98925_SPKCURNT_FLAG_WIDTH 1 +#define M98925_WATCHFAIL_FLAG_MASK (1<<4) +#define M98925_WATCHFAIL_FLAG_SHIFT 4 +#define M98925_WATCHFAIL_FLAG_WIDTH 1 +#define M98925_ALCINFH_FLAG_MASK (1<<3) +#define M98925_ALCINFH_FLAG_SHIFT 3 +#define M98925_ALCINFH_FLAG_WIDTH 1 +#define M98925_ALCACT_FLAG_MASK (1<<2) +#define M98925_ALCACT_FLAG_SHIFT 2 +#define M98925_ALCACT_FLAG_WIDTH 1 +#define M98925_ALCMUT_FLAG_MASK (1<<1) +#define M98925_ALCMUT_FLAG_SHIFT 1 +#define M98925_ALCMUT_FLAG_WIDTH 1 +#define M98925_ALCP_FLAG_MASK (1<<0) +#define M98925_ALCP_FLAG_SHIFT 0 +#define M98925_ALCP_FLAG_WIDTH 1 + +/* MAX98925_R00A_FLAG2 */ +#define M98925_SLOTOVRN_FLAG_MASK (1<<6) +#define M98925_SLOTOVRN_FLAG_SHIFT 6 +#define M98925_SLOTOVRN_FLAG_WIDTH 1 +#define M98925_INVALSLOT_FLAG_MASK (1<<5) +#define M98925_INVALSLOT_FLAG_SHIFT 5 +#define M98925_INVALSLOT_FLAG_WIDTH 1 +#define M98925_SLOTCNFLT_FLAG_MASK (1<<4) +#define M98925_SLOTCNFLT_FLAG_SHIFT 4 +#define M98925_SLOTCNFLT_FLAG_WIDTH 1 +#define M98925_VBSTOVFL_FLAG_MASK (1<<3) +#define M98925_VBSTOVFL_FLAG_SHIFT 3 +#define M98925_VBSTOVFL_FLAG_WIDTH 1 +#define M98925_VBATOVFL_FLAG_MASK (1<<2) +#define M98925_VBATOVFL_FLAG_SHIFT 2 +#define M98925_VBATOVFL_FLAG_WIDTH 1 +#define M98925_IMONOVFL_FLAG_MASK (1<<1) +#define M98925_IMONOVFL_FLAG_SHIFT 1 +#define M98925_IMONOVFL_FLAG_WIDTH 1 +#define M98925_VMONOVFL_FLAG_MASK (1<<0) +#define M98925_VMONOVFL_FLAG_SHIFT 0 +#define M98925_VMONOVFL_FLAG_WIDTH 1 + +/* MAX98925_R00B_IRQ_ENABLE0 */ +#define M98925_THERMWARN_END_EN_MASK (1<<3) +#define M98925_THERMWARN_END_EN_SHIFT 3 +#define M98925_THERMWARN_END_EN_WIDTH 1 +#define M98925_THERMWARN_BGN_EN_MASK (1<<2) +#define M98925_THERMWARN_BGN_EN_SHIFT 2 +#define M98925_THERMWARN_BGN_EN_WIDTH 1 +#define M98925_THERMSHDN_END_EN_MASK (1<<1) +#define M98925_THERMSHDN_END_EN_SHIFT 1 +#define M98925_THERMSHDN_END_EN_WIDTH 1 +#define M98925_THERMSHDN_BGN_EN_MASK (1<<0) +#define M98925_THERMSHDN_BGN_EN_SHIFT 0 +#define M98925_THERMSHDN_BGN_EN_WIDTH 1 + +/* MAX98925_R00C_IRQ_ENABLE1 */ +#define M98925_SPKCURNT_EN_MASK (1<<5) +#define M98925_SPKCURNT_EN_SHIFT 5 +#define M98925_SPKCURNT_EN_WIDTH 1 +#define M98925_WATCHFAIL_EN_MASK (1<<4) +#define M98925_WATCHFAIL_EN_SHIFT 4 +#define M98925_WATCHFAIL_EN_WIDTH 1 +#define M98925_ALCINFH_EN_MASK (1<<3) +#define M98925_ALCINFH_EN_SHIFT 3 +#define M98925_ALCINFH_EN_WIDTH 1 +#define M98925_ALCACT_EN_MASK (1<<2) +#define M98925_ALCACT_EN_SHIFT 2 +#define M98925_ALCACT_EN_WIDTH 1 +#define M98925_ALCMUT_EN_MASK (1<<1) +#define M98925_ALCMUT_EN_SHIFT 1 +#define M98925_ALCMUT_EN_WIDTH 1 +#define M98925_ALCP_EN_MASK (1<<0) +#define M98925_ALCP_EN_SHIFT 0 +#define M98925_ALCP_EN_WIDTH 1 + +/* MAX98925_R00D_IRQ_ENABLE2 */ +#define M98925_SLOTOVRN_EN_MASK (1<<6) +#define M98925_SLOTOVRN_EN_SHIFT 6 +#define M98925_SLOTOVRN_EN_WIDTH 1 +#define M98925_INVALSLOT_EN_MASK (1<<5) +#define M98925_INVALSLOT_EN_SHIFT 5 +#define M98925_INVALSLOT_EN_WIDTH 1 +#define M98925_SLOTCNFLT_EN_MASK (1<<4) +#define M98925_SLOTCNFLT_EN_SHIFT 4 +#define M98925_SLOTCNFLT_EN_WIDTH 1 +#define M98925_VBSTOVFL_EN_MASK (1<<3) +#define M98925_VBSTOVFL_EN_SHIFT 3 +#define M98925_VBSTOVFL_EN_WIDTH 1 +#define M98925_VBATOVFL_EN_MASK (1<<2) +#define M98925_VBATOVFL_EN_SHIFT 2 +#define M98925_VBATOVFL_EN_WIDTH 1 +#define M98925_IMONOVFL_EN_MASK (1<<1) +#define M98925_IMONOVFL_EN_SHIFT 1 +#define M98925_IMONOVFL_EN_WIDTH 1 +#define M98925_VMONOVFL_EN_MASK (1<<0) +#define M98925_VMONOVFL_EN_SHIFT 0 +#define M98925_VMONOVFL_EN_WIDTH 1 + +/* MAX98925_R00E_IRQ_CLEAR0 */ +#define M98925_THERMWARN_END_CLR_MASK (1<<3) +#define M98925_THERMWARN_END_CLR_SHIFT 3 +#define M98925_THERMWARN_END_CLR_WIDTH 1 +#define M98925_THERMWARN_BGN_CLR_MASK (1<<2) +#define M98925_THERMWARN_BGN_CLR_SHIFT 2 +#define M98925_THERMWARN_BGN_CLR_WIDTH 1 +#define M98925_THERMSHDN_END_CLR_MASK (1<<1) +#define M98925_THERMSHDN_END_CLR_SHIFT 1 +#define M98925_THERMSHDN_END_CLR_WIDTH 1 +#define M98925_THERMSHDN_BGN_CLR_MASK (1<<0) +#define M98925_THERMSHDN_BGN_CLR_SHIFT 0 +#define M98925_THERMSHDN_BGN_CLR_WIDTH 1 + +/* MAX98925_R00F_IRQ_CLEAR1 */ +#define M98925_SPKCURNT_CLR_MASK (1<<5) +#define M98925_SPKCURNT_CLR_SHIFT 5 +#define M98925_SPKCURNT_CLR_WIDTH 1 +#define M98925_WATCHFAIL_CLR_MASK (1<<4) +#define M98925_WATCHFAIL_CLR_SHIFT 4 +#define M98925_WATCHFAIL_CLR_WIDTH 1 +#define M98925_ALCINFH_CLR_MASK (1<<3) +#define M98925_ALCINFH_CLR_SHIFT 3 +#define M98925_ALCINFH_CLR_WIDTH 1 +#define M98925_ALCACT_CLR_MASK (1<<2) +#define M98925_ALCACT_CLR_SHIFT 2 +#define M98925_ALCACT_CLR_WIDTH 1 +#define M98925_ALCMUT_CLR_MASK (1<<1) +#define M98925_ALCMUT_CLR_SHIFT 1 +#define M98925_ALCMUT_CLR_WIDTH 1 +#define M98925_ALCP_CLR_MASK (1<<0) +#define M98925_ALCP_CLR_SHIFT 0 +#define M98925_ALCP_CLR_WIDTH 1 + +/* MAX98925_R010_IRQ_CLEAR2 */ +#define M98925_SLOTOVRN_CLR_MASK (1<<6) +#define M98925_SLOTOVRN_CLR_SHIFT 6 +#define M98925_SLOTOVRN_CLR_WIDTH 1 +#define M98925_INVALSLOT_CLR_MASK (1<<5) +#define M98925_INVALSLOT_CLR_SHIFT 5 +#define M98925_INVALSLOT_CLR_WIDTH 1 +#define M98925_SLOTCNFLT_CLR_MASK (1<<4) +#define M98925_SLOTCNFLT_CLR_SHIFT 4 +#define M98925_SLOTCNFLT_CLR_WIDTH 1 +#define M98925_VBSTOVFL_CLR_MASK (1<<3) +#define M98925_VBSTOVFL_CLR_SHIFT 3 +#define M98925_VBSTOVFL_CLR_WIDTH 1 +#define M98925_VBATOVFL_CLR_MASK (1<<2) +#define M98925_VBATOVFL_CLR_SHIFT 2 +#define M98925_VBATOVFL_CLR_WIDTH 1 +#define M98925_IMONOVFL_CLR_MASK (1<<1) +#define M98925_IMONOVFL_CLR_SHIFT 1 +#define M98925_IMONOVFL_CLR_WIDTH 1 +#define M98925_VMONOVFL_CLR_MASK (1<<0) +#define M98925_VMONOVFL_CLR_SHIFT 0 +#define M98925_VMONOVFL_CLR_WIDTH 1 + +/* MAX98925_R011_MAP0 */ +#define M98925_ER_THERMWARN_EN_MASK (1<<7) +#define M98925_ER_THERMWARN_EN_SHIFT 7 +#define M98925_ER_THERMWARN_EN_WIDTH 1 +#define M98925_ER_THERMWARN_MAP_MASK (0x07<<4) +#define M98925_ER_THERMWARN_MAP_SHIFT 4 +#define M98925_ER_THERMWARN_MAP_WIDTH 3 + +/* MAX98925_R012_MAP1 */ +#define M98925_ER_ALCMUT_EN_MASK (1<<7) +#define M98925_ER_ALCMUT_EN_SHIFT 7 +#define M98925_ER_ALCMUT_EN_WIDTH 1 +#define M98925_ER_ALCMUT_MAP_MASK (0x07<<4) +#define M98925_ER_ALCMUT_MAP_SHIFT 4 +#define M98925_ER_ALCMUT_MAP_WIDTH 3 +#define M98925_ER_ALCP_EN_MASK (1<<3) +#define M98925_ER_ALCP_EN_SHIFT 3 +#define M98925_ER_ALCP_EN_WIDTH 1 +#define M98925_ER_ALCP_MAP_MASK (0x07<<0) +#define M98925_ER_ALCP_MAP_SHIFT 0 +#define M98925_ER_ALCP_MAP_WIDTH 3 + +/* MAX98925_R013_MAP2 */ +#define M98925_ER_ALCINFH_EN_MASK (1<<7) +#define M98925_ER_ALCINFH_EN_SHIFT 7 +#define M98925_ER_ALCINFH_EN_WIDTH 1 +#define M98925_ER_ALCINFH_MAP_MASK (0x07<<4) +#define M98925_ER_ALCINFH_MAP_SHIFT 4 +#define M98925_ER_ALCINFH_MAP_WIDTH 3 +#define M98925_ER_ALCACT_EN_MASK (1<<3) +#define M98925_ER_ALCACT_EN_SHIFT 3 +#define M98925_ER_ALCACT_EN_WIDTH 1 +#define M98925_ER_ALCACT_MAP_MASK (0x07<<0) +#define M98925_ER_ALCACT_MAP_SHIFT 0 +#define M98925_ER_ALCACT_MAP_WIDTH 3 + +/* MAX98925_R014_MAP3 */ +#define M98925_ER_SPKCURNT_EN_MASK (1<<7) +#define M98925_ER_SPKCURNT_EN_SHIFT 7 +#define M98925_ER_SPKCURNT_EN_WIDTH 1 +#define M98925_ER_SPKCURNT_MAP_MASK (0x07<<4) +#define M98925_ER_SPKCURNT_MAP_SHIFT 4 +#define M98925_ER_SPKCURNT_MAP_WIDTH 3 + +/* MAX98925_R015_MAP4 */ +/* RESERVED */ + +/* MAX98925_R016_MAP5 */ +#define M98925_ER_IMONOVFL_EN_MASK (1<<7) +#define M98925_ER_IMONOVFL_EN_SHIFT 7 +#define M98925_ER_IMONOVFL_EN_WIDTH 1 +#define M98925_ER_IMONOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_IMONOVFL_MAP_SHIFT 4 +#define M98925_ER_IMONOVFL_MAP_WIDTH 3 +#define M98925_ER_VMONOVFL_EN_MASK (1<<3) +#define M98925_ER_VMONOVFL_EN_SHIFT 3 +#define M98925_ER_VMONOVFL_EN_WIDTH 1 +#define M98925_ER_VMONOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VMONOVFL_MAP_SHIFT 0 +#define M98925_ER_VMONOVFL_MAP_WIDTH 3 + +/* MAX98925_R017_MAP6 */ +#define M98925_ER_VBSTOVFL_EN_MASK (1<<7) +#define M98925_ER_VBSTOVFL_EN_SHIFT 7 +#define M98925_ER_VBSTOVFL_EN_WIDTH 1 +#define M98925_ER_VBSTOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_VBSTOVFL_MAP_SHIFT 4 +#define M98925_ER_VBSTOVFL_MAP_WIDTH 3 +#define M98925_ER_VBATOVFL_EN_MASK (1<<3) +#define M98925_ER_VBATOVFL_EN_SHIFT 3 +#define M98925_ER_VBATOVFL_EN_WIDTH 1 +#define M98925_ER_VBATOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VBATOVFL_MAP_SHIFT 0 +#define M98925_ER_VBATOVFL_MAP_WIDTH 3 + +/* MAX98925_R018_MAP7 */ +#define M98925_ER_INVALSLOT_EN_MASK (1<<7) +#define M98925_ER_INVALSLOT_EN_SHIFT 7 +#define M98925_ER_INVALSLOT_EN_WIDTH 1 +#define M98925_ER_INVALSLOT_MAP_MASK (0x07<<4) +#define M98925_ER_INVALSLOT_MAP_SHIFT 4 +#define M98925_ER_INVALSLOT_MAP_WIDTH 3 +#define M98925_ER_SLOTCNFLT_EN_MASK (1<<3) +#define M98925_ER_SLOTCNFLT_EN_SHIFT 3 +#define M98925_ER_SLOTCNFLT_EN_WIDTH 1 +#define M98925_ER_SLOTCNFLT_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTCNFLT_MAP_SHIFT 0 +#define M98925_ER_SLOTCNFLT_MAP_WIDTH 3 + +/* MAX98925_R019_MAP8 */ +#define M98925_ER_SLOTOVRN_EN_MASK (1<<3) +#define M98925_ER_SLOTOVRN_EN_SHIFT 3 +#define M98925_ER_SLOTOVRN_EN_WIDTH 1 +#define M98925_ER_SLOTOVRN_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTOVRN_MAP_SHIFT 0 +#define M98925_ER_SLOTOVRN_MAP_WIDTH 3 + +/* MAX98925_R01A_DAI_CLK_MODE1 */ +#define M98925_DAI_CLK_SOURCE_MASK (1<<6) +#define M98925_DAI_CLK_SOURCE_SHIFT 6 +#define M98925_DAI_CLK_SOURCE_WIDTH 1 +#define M98925_MDLL_MULT_MASK (0x0F<<0) +#define M98925_MDLL_MULT_SHIFT 0 +#define M98925_MDLL_MULT_WIDTH 4 + +#define M98925_MDLL_MULT_MCLKx8 6 +#define M98925_MDLL_MULT_MCLKx16 8 + +/* MAX98925_R01B_DAI_CLK_MODE2 */ +#define M98925_DAI_SR_MASK (0x0F<<4) +#define M98925_DAI_SR_SHIFT 4 +#define M98925_DAI_SR_WIDTH 4 +#define M98925_DAI_MAS_MASK (1<<3) +#define M98925_DAI_MAS_SHIFT 3 +#define M98925_DAI_MAS_WIDTH 1 +#define M98925_DAI_BSEL_MASK (0x07<<0) +#define M98925_DAI_BSEL_SHIFT 0 +#define M98925_DAI_BSEL_WIDTH 3 + +#define M98925_DAI_BSEL_32 (0 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_48 (1 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_64 (2 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_256 (6 << M98925_DAI_BSEL_SHIFT) + +/* MAX98925_R01C_DAI_CLK_DIV_M_MSBS */ +#define M98925_DAI_M_MSBS_MASK (0xFF<<0) +#define M98925_DAI_M_MSBS_SHIFT 0 +#define M98925_DAI_M_MSBS_WIDTH 8 + +/* MAX98925_R01D_DAI_CLK_DIV_M_LSBS */ +#define M98925_DAI_M_LSBS_MASK (0xFF<<0) +#define M98925_DAI_M_LSBS_SHIFT 0 +#define M98925_DAI_M_LSBS_WIDTH 8 + +/* MAX98925_R01E_DAI_CLK_DIV_N_MSBS */ +#define M98925_DAI_N_MSBS_MASK (0x7F<<0) +#define M98925_DAI_N_MSBS_SHIFT 0 +#define M98925_DAI_N_MSBS_WIDTH 7 + +/* MAX98925_R01F_DAI_CLK_DIV_N_LSBS */ +#define M98925_DAI_N_LSBS_MASK (0xFF<<0) +#define M98925_DAI_N_LSBS_SHIFT 0 +#define M98925_DAI_N_LSBS_WIDTH 8 + +/* MAX98925_R020_FORMAT */ +#define M98925_DAI_CHANSZ_MASK (0x03<<6) +#define M98925_DAI_CHANSZ_SHIFT 6 +#define M98925_DAI_CHANSZ_WIDTH 2 +#define M98925_DAI_EXTBCLK_HIZ_MASK (1<<4) +#define M98925_DAI_EXTBCLK_HIZ_SHIFT 4 +#define M98925_DAI_EXTBCLK_HIZ_WIDTH 1 +#define M98925_DAI_WCI_MASK (1<<3) +#define M98925_DAI_WCI_SHIFT 3 +#define M98925_DAI_WCI_WIDTH 1 +#define M98925_DAI_BCI_MASK (1<<2) +#define M98925_DAI_BCI_SHIFT 2 +#define M98925_DAI_BCI_WIDTH 1 +#define M98925_DAI_DLY_MASK (1<<1) +#define M98925_DAI_DLY_SHIFT 1 +#define M98925_DAI_DLY_WIDTH 1 +#define M98925_DAI_TDM_MASK (1<<0) +#define M98925_DAI_TDM_SHIFT 0 +#define M98925_DAI_TDM_WIDTH 1 + +#define M98925_DAI_CHANSZ_16 (1 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_24 (2 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_32 (3 << M98925_DAI_CHANSZ_SHIFT) + +/* MAX98925_R021_TDM_SLOT_SELECT */ +#define M98925_DAI_DO_EN_MASK (1<<7) +#define M98925_DAI_DO_EN_SHIFT 7 +#define M98925_DAI_DO_EN_WIDTH 1 +#define M98925_DAI_DIN_EN_MASK (1<<6) +#define M98925_DAI_DIN_EN_SHIFT 6 +#define M98925_DAI_DIN_EN_WIDTH 1 +#define M98925_DAI_INR_SOURCE_MASK (0x07<<3) +#define M98925_DAI_INR_SOURCE_SHIFT 3 +#define M98925_DAI_INR_SOURCE_WIDTH 3 +#define M98925_DAI_INL_SOURCE_MASK (0x07<<0) +#define M98925_DAI_INL_SOURCE_SHIFT 0 +#define M98925_DAI_INL_SOURCE_WIDTH 3 + +/* MAX98925_R022_DOUT_CFG_VMON */ +#define M98925_DAI_VMON_EN_MASK (1<<5) +#define M98925_DAI_VMON_EN_SHIFT 5 +#define M98925_DAI_VMON_EN_WIDTH 1 +#define M98925_DAI_VMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VMON_SLOT_SHIFT 0 +#define M98925_DAI_VMON_SLOT_WIDTH 5 + +#define M98925_DAI_VMON_SLOT_00_01 (0 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_01_02 (1 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_02_03 (2 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_03_04 (3 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_04_05 (4 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_05_06 (5 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_06_07 (6 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_07_08 (7 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_08_09 (8 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_09_0A (9 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0A_0B (10 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0B_0C (11 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0C_0D (12 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0D_0E (13 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0E_0F (14 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0F_10 (15 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_10_11 (16 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_11_12 (17 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_12_13 (18 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_13_14 (19 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_14_15 (20 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_15_16 (21 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_16_17 (22 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_17_18 (23 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_18_19 (24 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_19_1A (25 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1A_1B (26 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1B_1C (27 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1C_1D (28 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1D_1E (29 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1E_1F (30 << M98925_DAI_VMON_SLOT_SHIFT) + +/* MAX98925_R023_DOUT_CFG_IMON */ +#define M98925_DAI_IMON_EN_MASK (1<<5) +#define M98925_DAI_IMON_EN_SHIFT 5 +#define M98925_DAI_IMON_EN_WIDTH 1 +#define M98925_DAI_IMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_IMON_SLOT_SHIFT 0 +#define M98925_DAI_IMON_SLOT_WIDTH 5 + +#define M98925_DAI_IMON_SLOT_00_01 (0 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_01_02 (1 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_02_03 (2 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_03_04 (3 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_04_05 (4 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_05_06 (5 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_06_07 (6 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_07_08 (7 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_08_09 (8 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_09_0A (9 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0A_0B (10 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0B_0C (11 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0C_0D (12 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0D_0E (13 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0E_0F (14 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0F_10 (15 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_10_11 (16 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_11_12 (17 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_12_13 (18 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_13_14 (19 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_14_15 (20 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_15_16 (21 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_16_17 (22 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_17_18 (23 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_18_19 (24 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_19_1A (25 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1A_1B (26 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1B_1C (27 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1C_1D (28 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1D_1E (29 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1E_1F (30 << M98925_DAI_IMON_SLOT_SHIFT) + +/* MAX98925_R024_DOUT_CFG_VBAT */ +#define M98925_DAI_VBAT_EN_MASK (1<<5) +#define M98925_DAI_VBAT_EN_SHIFT 5 +#define M98925_DAI_VBAT_EN_WIDTH 1 +#define M98925_DAI_VBAT_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBAT_SLOT_SHIFT 0 +#define M98925_DAI_VBAT_SLOT_WIDTH 5 + +/* MAX98925_R025_DOUT_CFG_VBST */ +#define M98925_DAI_VBST_EN_MASK (1<<5) +#define M98925_DAI_VBST_EN_SHIFT 5 +#define M98925_DAI_VBST_EN_WIDTH 1 +#define M98925_DAI_VBST_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBST_SLOT_SHIFT 0 +#define M98925_DAI_VBST_SLOT_WIDTH 5 + +/* MAX98925_R026_DOUT_CFG_FLAG */ +#define M98925_DAI_FLAG_EN_MASK (1<<5) +#define M98925_DAI_FLAG_EN_SHIFT 5 +#define M98925_DAI_FLAG_EN_WIDTH 1 +#define M98925_DAI_FLAG_SLOT_MASK (0x1F<<0) +#define M98925_DAI_FLAG_SLOT_SHIFT 0 +#define M98925_DAI_FLAG_SLOT_WIDTH 5 + +/* MAX98925_R027_DOUT_HIZ_CFG1 */ +#define M98925_DAI_SLOT_HIZ_CFG1_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG1_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG1_WIDTH 8 + +/* MAX98925_R028_DOUT_HIZ_CFG2 */ +#define M98925_DAI_SLOT_HIZ_CFG2_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG2_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG2_WIDTH 8 + +/* MAX98925_R029_DOUT_HIZ_CFG3 */ +#define M98925_DAI_SLOT_HIZ_CFG3_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG3_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG3_WIDTH 8 + +/* MAX98925_R02A_DOUT_HIZ_CFG4 */ +#define M98925_DAI_SLOT_HIZ_CFG4_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG4_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG4_WIDTH 8 + +/* MAX98925_R02B_DOUT_DRV_STRENGTH */ +#define M98925_DAI_OUT_DRIVE_MASK (0x03<<0) +#define M98925_DAI_OUT_DRIVE_SHIFT 0 +#define M98925_DAI_OUT_DRIVE_WIDTH 2 + +/* MAX98925_R02C_FILTERS */ +#define M98925_ADC_DITHER_EN_MASK (1<<7) +#define M98925_ADC_DITHER_EN_SHIFT 7 +#define M98925_ADC_DITHER_EN_WIDTH 1 +#define M98925_IV_DCB_EN_MASK (1<<6) +#define M98925_IV_DCB_EN_SHIFT 6 +#define M98925_IV_DCB_EN_WIDTH 1 +#define M98925_DAC_DITHER_EN_MASK (1<<4) +#define M98925_DAC_DITHER_EN_SHIFT 4 +#define M98925_DAC_DITHER_EN_WIDTH 1 +#define M98925_DAC_FILTER_MODE_MASK (1<<3) +#define M98925_DAC_FILTER_MODE_SHIFT 3 +#define M98925_DAC_FILTER_MODE_WIDTH 1 +#define M98925_DAC_HPF_MASK (0x07<<0) +#define M98925_DAC_HPF_SHIFT 0 +#define M98925_DAC_HPF_WIDTH 3 +#define M98925_DAC_HPF_DISABLE (0 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_DC_BLOCK (1 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_100 (2 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_200 (3 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_400 (4 << M98925_DAC_HPF_SHIFT) +#define M98925_DAC_HPF_EN_800 (5 << M98925_DAC_HPF_SHIFT) + +/* MAX98925_R02D_GAIN */ +#define M98925_DAC_IN_SEL_MASK (0x03<<5) +#define M98925_DAC_IN_SEL_SHIFT 5 +#define M98925_DAC_IN_SEL_WIDTH 2 +#define M98925_SPK_GAIN_MASK (0x1F<<0) +#define M98925_SPK_GAIN_SHIFT 0 +#define M98925_SPK_GAIN_WIDTH 5 + +#define M98925_DAC_IN_SEL_LEFT_DAI (0 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_RIGHT_DAI (1 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_SUMMED_DAI (2 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_DIV2_SUMMED_DAI (3 << M98925_DAC_IN_SEL_SHIFT) + +/* MAX98925_R02E_GAIN_RAMPING */ +#define M98925_SPK_RMP_EN_MASK (1<<1) +#define M98925_SPK_RMP_EN_SHIFT 1 +#define M98925_SPK_RMP_EN_WIDTH 1 +#define M98925_SPK_ZCD_EN_MASK (1<<0) +#define M98925_SPK_ZCD_EN_SHIFT 0 +#define M98925_SPK_ZCD_EN_WIDTH 1 + +/* MAX98925_R02F_SPK_AMP */ +#define M98925_SPK_MODE_MASK (1<<0) +#define M98925_SPK_MODE_SHIFT 0 +#define M98925_SPK_MODE_WIDTH 1 + +/* MAX98925_R030_THRESHOLD */ +#define M98925_ALC_EN_MASK (1<<5) +#define M98925_ALC_EN_SHIFT 5 +#define M98925_ALC_EN_WIDTH 1 +#define M98925_ALC_TH_MASK (0x1F<<0) +#define M98925_ALC_TH_SHIFT 0 +#define M98925_ALC_TH_WIDTH 5 + +/* MAX98925_R031_ALC_ATTACK */ +#define M98925_ALC_ATK_STEP_MASK (0x0F<<4) +#define M98925_ALC_ATK_STEP_SHIFT 4 +#define M98925_ALC_ATK_STEP_WIDTH 4 +#define M98925_ALC_ATK_RATE_MASK (0x7<<0) +#define M98925_ALC_ATK_RATE_SHIFT 0 +#define M98925_ALC_ATK_RATE_WIDTH 3 + +/* MAX98925_R032_ALC_ATTEN_RLS */ +#define M98925_ALC_MAX_ATTEN_MASK (0x0F<<4) +#define M98925_ALC_MAX_ATTEN_SHIFT 4 +#define M98925_ALC_MAX_ATTEN_WIDTH 4 +#define M98925_ALC_RLS_RATE_MASK (0x7<<0) +#define M98925_ALC_RLS_RATE_SHIFT 0 +#define M98925_ALC_RLS_RATE_WIDTH 3 + +/* MAX98925_R033_ALC_HOLD_RLS */ +#define M98925_ALC_RLS_TGR_MASK (1<<0) +#define M98925_ALC_RLS_TGR_SHIFT 0 +#define M98925_ALC_RLS_TGR_WIDTH 1 + +/* MAX98925_R034_ALC_CONFIGURATION */ +#define M98925_ALC_MUTE_EN_MASK (1<<7) +#define M98925_ALC_MUTE_EN_SHIFT 7 +#define M98925_ALC_MUTE_EN_WIDTH 1 +#define M98925_ALC_MUTE_DLY_MASK (0x07<<4) +#define M98925_ALC_MUTE_DLY_SHIFT 4 +#define M98925_ALC_MUTE_DLY_WIDTH 3 +#define M98925_ALC_RLS_DBT_MASK (0x07<<0) +#define M98925_ALC_RLS_DBT_SHIFT 0 +#define M98925_ALC_RLS_DBT_WIDTH 3 + +/* MAX98925_R035_BOOST_CONVERTER */ +#define M98925_BST_SYNC_MASK (1<<7) +#define M98925_BST_SYNC_SHIFT 7 +#define M98925_BST_SYNC_WIDTH 1 +#define M98925_BST_PHASE_MASK (0x03<<4) +#define M98925_BST_PHASE_SHIFT 4 +#define M98925_BST_PHASE_WIDTH 2 +#define M98925_BST_SKIP_MODE_MASK (0x03<<0) +#define M98925_BST_SKIP_MODE_SHIFT 0 +#define M98925_BST_SKIP_MODE_WIDTH 2 + +/* MAX98925_R036_BLOCK_ENABLE */ +#define M98925_BST_EN_MASK (1<<7) +#define M98925_BST_EN_SHIFT 7 +#define M98925_BST_EN_WIDTH 1 +#define M98925_WATCH_EN_MASK (1<<6) +#define M98925_WATCH_EN_SHIFT 6 +#define M98925_WATCH_EN_WIDTH 1 +#define M98925_CLKMON_EN_MASK (1<<5) +#define M98925_CLKMON_EN_SHIFT 5 +#define M98925_CLKMON_EN_WIDTH 1 +#define M98925_SPK_EN_MASK (1<<4) +#define M98925_SPK_EN_SHIFT 4 +#define M98925_SPK_EN_WIDTH 1 +#define M98925_ADC_VBST_EN_MASK (1<<3) +#define M98925_ADC_VBST_EN_SHIFT 3 +#define M98925_ADC_VBST_EN_WIDTH 1 +#define M98925_ADC_VBAT_EN_MASK (1<<2) +#define M98925_ADC_VBAT_EN_SHIFT 2 +#define M98925_ADC_VBAT_EN_WIDTH 1 +#define M98925_ADC_IMON_EN_MASK (1<<1) +#define M98925_ADC_IMON_EN_SHIFT 1 +#define M98925_ADC_IMON_EN_WIDTH 1 +#define M98925_ADC_VMON_EN_MASK (1<<0) +#define M98925_ADC_VMON_EN_SHIFT 0 +#define M98925_ADC_VMON_EN_WIDTH 1 + +/* MAX98925_R037_CONFIGURATION */ +#define M98925_BST_VOUT_MASK (0x0F<<4) +#define M98925_BST_VOUT_SHIFT 4 +#define M98925_BST_VOUT_WIDTH 4 +#define M98925_THERMWARN_LEVEL_MASK (0x03<<2) +#define M98925_THERMWARN_LEVEL_SHIFT 2 +#define M98925_THERMWARN_LEVEL_WIDTH 2 +#define M98925_WATCH_TIME_MASK (0x03<<0) +#define M98925_WATCH_TIME_SHIFT 0 +#define M98925_WATCH_TIME_WIDTH 2 + +/* MAX98925_R038_GLOBAL_ENABLE */ +#define M98925_EN_MASK (1<<7) +#define M98925_EN_SHIFT 7 +#define M98925_EN_WIDTH 1 + +/* MAX98925_R03A_BOOST_LIMITER */ +#define M98925_BST_ILIM_MASK (0x1F<<3) +#define M98925_BST_ILIM_SHIFT 3 +#define M98925_BST_ILIM_WIDTH 5 + +/* MAX98925_R0FF_VERSION */ +#define M98925_REV_ID_MASK (0xFF<<0) +#define M98925_REV_ID_SHIFT 0 +#define M98925_REV_ID_WIDTH 8 + +struct max98925_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct max98925_pdata *pdata; + unsigned int sysclk; + unsigned int v_slot; + unsigned int i_slot; + unsigned int spk_gain; + unsigned int ch_size; +}; +#endif From 10dcc448d13071ed0d3797233187e79be258d90c Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Thu, 12 Mar 2015 03:05:44 +0800 Subject: [PATCH 208/411] ASoC: max98925_spk_tlv can be static Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/codecs/max98925.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 74f4f0b60108fd..34fc4d0441fd54 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -186,7 +186,7 @@ static bool max98925_readable_register(struct device *dev, unsigned int reg) } } -DECLARE_TLV_DB_SCALE(max98925_spk_tlv, -600, 100, 0); +static DECLARE_TLV_DB_SCALE(max98925_spk_tlv, -600, 100, 0); static const struct snd_kcontrol_new max98925_snd_controls[] = { SOC_SINGLE_TLV("Speaker Volume", MAX98925_GAIN, From 66454b3eb31e8109387606c054ae02f3cfd6543e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 15 Jan 2015 12:52:15 +0100 Subject: [PATCH 209/411] ASoC: rt5670: Replace w->codec snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 91d2069a931364..cc7f84a150a7f0 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -699,7 +699,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); if (rt5670->sysclk_src == RT5670_SCLK_S_PLL1) From 76c07b8265c68d9a89fb4c0a634e373a087f11be Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Thu, 12 Mar 2015 13:53:00 +0800 Subject: [PATCH 210/411] ASoC: Intel: add kcontrol to enable/disable sound effect module waves Add kcontrol to enable/disable module waves. IPC is valid only when module is loaded. Also track module state over suspend so it's state can be restored on resume. Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 19 +++++++++ sound/soc/intel/sst-haswell-ipc.h | 3 ++ sound/soc/intel/sst-haswell-pcm.c | 68 +++++++++++++++++++++++++++++++ 3 files changed, 90 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 265d754a40907a..ebca9035efcec8 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -337,6 +337,10 @@ struct sst_hsw { /* FW log stream */ struct sst_hsw_log_stream log_stream; + + /* flags bit field to track module state when resume from RTD3, + * each bit represent state (enabled/disabled) of single module */ + u32 enabled_modules_rtd3; }; #define CREATE_TRACE_POINTS @@ -1986,6 +1990,21 @@ bool sst_hsw_is_module_active(struct sst_hsw *hsw, u32 module_id) return false; } +void sst_hsw_set_module_enabled_rtd3(struct sst_hsw *hsw, u32 module_id) +{ + hsw->enabled_modules_rtd3 |= (1 << module_id); +} + +void sst_hsw_set_module_disabled_rtd3(struct sst_hsw *hsw, u32 module_id) +{ + hsw->enabled_modules_rtd3 &= ~(1 << module_id); +} + +bool sst_hsw_is_module_enabled_rtd3(struct sst_hsw *hsw, u32 module_id) +{ + return hsw->enabled_modules_rtd3 & (1 << module_id); +} + int sst_hsw_module_load(struct sst_hsw *hsw, u32 module_id, u32 instance_id, char *name) { diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 30c65b28fa6072..48290a1cfe5d57 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -477,6 +477,9 @@ struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); void sst_hsw_init_module_state(struct sst_hsw *hsw); bool sst_hsw_is_module_loaded(struct sst_hsw *hsw, u32 module_id); bool sst_hsw_is_module_active(struct sst_hsw *hsw, u32 module_id); +void sst_hsw_set_module_enabled_rtd3(struct sst_hsw *hsw, u32 module_id); +void sst_hsw_set_module_disabled_rtd3(struct sst_hsw *hsw, u32 module_id); +bool sst_hsw_is_module_enabled_rtd3(struct sst_hsw *hsw, u32 module_id); int sst_hsw_module_load(struct sst_hsw *hsw, u32 module_id, u32 instance_id, char *name); diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index a604cc4421110a..b3de87aac37360 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -318,6 +318,54 @@ static int hsw_volume_get(struct snd_kcontrol *kcontrol, return 0; } +static int hsw_waves_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct sst_hsw *hsw = pdata->hsw; + enum sst_hsw_module_id id = SST_HSW_MODULE_WAVES; + + ucontrol->value.integer.value[0] = + (sst_hsw_is_module_active(hsw, id) || + sst_hsw_is_module_enabled_rtd3(hsw, id)); + return 0; +} + +static int hsw_waves_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct sst_hsw *hsw = pdata->hsw; + int ret = 0; + enum sst_hsw_module_id id = SST_HSW_MODULE_WAVES; + bool switch_on = (bool)ucontrol->value.integer.value[0]; + + /* if module is in RAM on the DSP, apply user settings to module through + * ipc. If module is not in RAM on the DSP, store user setting for + * track */ + if (sst_hsw_is_module_loaded(hsw, id)) { + if (switch_on == sst_hsw_is_module_active(hsw, id)) + return 0; + + if (switch_on) + ret = sst_hsw_module_enable(hsw, id, 0); + else + ret = sst_hsw_module_disable(hsw, id, 0); + } else { + if (switch_on == sst_hsw_is_module_enabled_rtd3(hsw, id)) + return 0; + + if (switch_on) + sst_hsw_set_module_enabled_rtd3(hsw, id); + else + sst_hsw_set_module_disabled_rtd3(hsw, id); + } + + return ret; +} + /* TLV used by both global and stream volumes */ static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1); @@ -339,6 +387,9 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), + /* enable/disable module waves */ + SOC_SINGLE_BOOL_EXT("Waves Switch", 0, + hsw_waves_switch_get, hsw_waves_switch_put), }; /* Create DMA buffer page table for DSP */ @@ -1118,10 +1169,18 @@ static int hsw_pcm_runtime_suspend(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; + int ret; if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; + /* fw modules will be unloaded on RTD3, set flag to track */ + if (sst_hsw_is_module_active(hsw, SST_HSW_MODULE_WAVES)) { + ret = sst_hsw_module_disable(hsw, SST_HSW_MODULE_WAVES, 0); + if (ret < 0) + return ret; + sst_hsw_set_module_enabled_rtd3(hsw, SST_HSW_MODULE_WAVES); + } sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); pdata->pm_state = HSW_PM_STATE_RTD3; @@ -1156,6 +1215,15 @@ static int hsw_pcm_runtime_resume(struct device *dev) else if (ret == 1) /* no action required */ return 0; + /* check flag when resume */ + if (sst_hsw_is_module_enabled_rtd3(hsw, SST_HSW_MODULE_WAVES)) { + ret = sst_hsw_module_enable(hsw, SST_HSW_MODULE_WAVES, 0); + if (ret < 0) + return ret; + /* unset flag */ + sst_hsw_set_module_disabled_rtd3(hsw, SST_HSW_MODULE_WAVES); + } + pdata->pm_state = HSW_PM_STATE_D0; return ret; } From 201892268b8335ae8c9dc7cc9a0d4bf6e1336f0e Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Thu, 12 Mar 2015 13:53:01 +0800 Subject: [PATCH 211/411] ASoC: Intel: add function to set parameter to sound effect module waves Add function to set parameters to module waves. The parameters can be set only when module is enabled, and parameter size is limited to 500 Bytes. Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 59 +++++++++++++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 26 ++++++++++++++ 2 files changed, 85 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index ebca9035efcec8..a97324dff8fa47 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -2166,6 +2166,65 @@ int sst_hsw_module_disable(struct sst_hsw *hsw, return ret; } +int sst_hsw_module_set_param(struct sst_hsw *hsw, + u32 module_id, u32 instance_id, u32 parameter_id, + u32 param_size, char *param) +{ + int ret; + unsigned char *data = NULL; + u32 header = 0; + u32 payload_size = 0, transfer_parameter_size = 0; + dma_addr_t dma_addr = 0; + struct sst_hsw_transfer_parameter *parameter; + struct device *dev = hsw->dev; + + header = IPC_GLB_TYPE(IPC_GLB_MODULE_OPERATION) | + IPC_MODULE_OPERATION(IPC_MODULE_SET_PARAMETER) | + IPC_MODULE_ID(module_id); + dev_dbg(dev, "sst_hsw_module_set_param header=%x\n", header); + + payload_size = param_size + + sizeof(struct sst_hsw_transfer_parameter) - + sizeof(struct sst_hsw_transfer_list); + dev_dbg(dev, "parameter size : %d\n", param_size); + dev_dbg(dev, "payload size : %d\n", payload_size); + + if (payload_size <= SST_HSW_IPC_MAX_SHORT_PARAMETER_SIZE) { + /* short parameter, mailbox can contain data */ + dev_dbg(dev, "transfer parameter size : %d\n", + transfer_parameter_size); + + transfer_parameter_size = ALIGN(payload_size, 4); + dev_dbg(dev, "transfer parameter aligned size : %d\n", + transfer_parameter_size); + + parameter = kzalloc(transfer_parameter_size, GFP_KERNEL); + if (parameter == NULL) + return -ENOMEM; + + memcpy(parameter->data, param, param_size); + } else { + dev_warn(dev, "transfer parameter size too large!"); + return 0; + } + + parameter->parameter_id = parameter_id; + parameter->data_size = param_size; + + ret = ipc_tx_message_wait(hsw, header, + parameter, transfer_parameter_size , NULL, 0); + if (ret < 0) + dev_err(dev, "ipc: module set parameter failed - %d\n", ret); + + kfree(parameter); + + if (data) + dma_free_coherent(hsw->dsp->dma_dev, + param_size, (void *)data, dma_addr); + + return ret; +} + static struct sst_dsp_device hsw_dev = { .thread = hsw_irq_thread, .ops = &haswell_ops, diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 48290a1cfe5d57..16bec433265ceb 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -37,6 +37,7 @@ #define SST_HSW_IPC_MAX_PAYLOAD_SIZE 400 #define SST_HSW_MAX_INFO_SIZE 64 #define SST_HSW_BUILD_HASH_LENGTH 40 +#define SST_HSW_IPC_MAX_SHORT_PARAMETER_SIZE 500 struct sst_hsw; struct sst_hsw_stream; @@ -187,6 +188,28 @@ enum sst_hsw_performance_action { SST_HSW_PERF_STOP = 1, }; +struct sst_hsw_transfer_info { + uint32_t destination; /* destination address */ + uint32_t reverse:1; /* if 1 data flows from destination */ + uint32_t size:31; /* transfer size in bytes.*/ + uint16_t first_page_offset; /* offset to data in the first page. */ + uint8_t packed_pages; /* page addresses. Each occupies 20 bits */ +} __attribute__((packed)); + +struct sst_hsw_transfer_list { + uint32_t transfers_count; + struct sst_hsw_transfer_info transfers; +} __attribute__((packed)); + +struct sst_hsw_transfer_parameter { + uint32_t parameter_id; + uint32_t data_size; + union { + uint8_t data[1]; + struct sst_hsw_transfer_list transfer_list; + }; +} __attribute__((packed)); + /* SST firmware module info */ struct sst_hsw_module_info { u8 name[SST_HSW_MAX_INFO_SIZE]; @@ -487,6 +510,9 @@ int sst_hsw_module_enable(struct sst_hsw *hsw, u32 module_id, u32 instance_id); int sst_hsw_module_disable(struct sst_hsw *hsw, u32 module_id, u32 instance_id); +int sst_hsw_module_set_param(struct sst_hsw *hsw, + u32 module_id, u32 instance_id, u32 parameter_id, + u32 param_size, char *param); /* runtime module management */ struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw, From 3814c204446822cd3c82ec4e8616600732c1f94e Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Thu, 12 Mar 2015 13:53:02 +0800 Subject: [PATCH 212/411] ASoC: Intel: add kcontrol to set parameter to sound effect module waves Each kcontrol command includes a line of parameters up to 128 bytes. kcontrol command to set param: cset "name='Waves Set Param' <0x01,0xff,...>" or cset-bin-file "name='Waves Set Param' " The parameter lines are stored in a buffer array, so can be read back from buffer rather than from DSP, and be relaunched to DSP when resume from RTD3. The buffer size is 160 parameter lines. kcontrol command to reset the buffer: cset "name='Waves Set Param' 0xff" alsa-lib v1.0.29 or commit 6ea14c36 and f47480af are required to support the kcontrol commands. Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 64 +++++++++++++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 6 +++ sound/soc/intel/sst-haswell-pcm.c | 50 ++++++++++++++++++++++++ 3 files changed, 120 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index a97324dff8fa47..43fb5f33916836 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -341,6 +341,11 @@ struct sst_hsw { /* flags bit field to track module state when resume from RTD3, * each bit represent state (enabled/disabled) of single module */ u32 enabled_modules_rtd3; + + /* buffer to store parameter lines */ + u32 param_idx_w; /* write index */ + u32 param_idx_r; /* read index */ + u8 param_buf[WAVES_PARAM_LINES][WAVES_PARAM_COUNT]; }; #define CREATE_TRACE_POINTS @@ -2005,6 +2010,62 @@ bool sst_hsw_is_module_enabled_rtd3(struct sst_hsw *hsw, u32 module_id) return hsw->enabled_modules_rtd3 & (1 << module_id); } +void sst_hsw_reset_param_buf(struct sst_hsw *hsw) +{ + hsw->param_idx_w = 0; + hsw->param_idx_r = 0; + memset((void *)hsw->param_buf, 0, sizeof(hsw->param_buf)); +} + +int sst_hsw_store_param_line(struct sst_hsw *hsw, u8 *buf) +{ + /* save line to the first available position of param buffer */ + if (hsw->param_idx_w > WAVES_PARAM_LINES - 1) { + dev_warn(hsw->dev, "warning: param buffer overflow!\n"); + return -EPERM; + } + memcpy(hsw->param_buf[hsw->param_idx_w], buf, WAVES_PARAM_COUNT); + hsw->param_idx_w++; + return 0; +} + +int sst_hsw_load_param_line(struct sst_hsw *hsw, u8 *buf) +{ + u8 id = 0; + + /* read the first matching line from param buffer */ + while (hsw->param_idx_r < WAVES_PARAM_LINES) { + id = hsw->param_buf[hsw->param_idx_r][0]; + hsw->param_idx_r++; + if (buf[0] == id) { + memcpy(buf, hsw->param_buf[hsw->param_idx_r], + WAVES_PARAM_COUNT); + break; + } + } + if (hsw->param_idx_r > WAVES_PARAM_LINES - 1) { + dev_dbg(hsw->dev, "end of buffer, roll to the beginning\n"); + hsw->param_idx_r = 0; + return 0; + } + return 0; +} + +int sst_hsw_launch_param_buf(struct sst_hsw *hsw) +{ + int ret, idx; + + /* put all param lines to DSP through ipc */ + for (idx = 0; idx < hsw->param_idx_w; idx++) { + ret = sst_hsw_module_set_param(hsw, + SST_HSW_MODULE_WAVES, 0, hsw->param_buf[idx][0], + WAVES_PARAM_COUNT, hsw->param_buf[idx]); + if (ret < 0) + return ret; + } + return 0; +} + int sst_hsw_module_load(struct sst_hsw *hsw, u32 module_id, u32 instance_id, char *name) { @@ -2299,6 +2360,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (ret < 0) goto boot_err; + /* init param buffer */ + sst_hsw_reset_param_buf(hsw); + /* wait for DSP boot completion */ sst_dsp_boot(hsw->dsp); ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 16bec433265ceb..06d71aefa1fe79 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -38,6 +38,8 @@ #define SST_HSW_MAX_INFO_SIZE 64 #define SST_HSW_BUILD_HASH_LENGTH 40 #define SST_HSW_IPC_MAX_SHORT_PARAMETER_SIZE 500 +#define WAVES_PARAM_COUNT 128 +#define WAVES_PARAM_LINES 160 struct sst_hsw; struct sst_hsw_stream; @@ -503,6 +505,10 @@ bool sst_hsw_is_module_active(struct sst_hsw *hsw, u32 module_id); void sst_hsw_set_module_enabled_rtd3(struct sst_hsw *hsw, u32 module_id); void sst_hsw_set_module_disabled_rtd3(struct sst_hsw *hsw, u32 module_id); bool sst_hsw_is_module_enabled_rtd3(struct sst_hsw *hsw, u32 module_id); +void sst_hsw_reset_param_buf(struct sst_hsw *hsw); +int sst_hsw_store_param_line(struct sst_hsw *hsw, u8 *buf); +int sst_hsw_load_param_line(struct sst_hsw *hsw, u8 *buf); +int sst_hsw_launch_param_buf(struct sst_hsw *hsw); int sst_hsw_module_load(struct sst_hsw *hsw, u32 module_id, u32 instance_id, char *name); diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index b3de87aac37360..b40ec746bc19f5 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -366,6 +366,49 @@ static int hsw_waves_switch_put(struct snd_kcontrol *kcontrol, return ret; } +static int hsw_waves_param_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct sst_hsw *hsw = pdata->hsw; + + /* return a matching line from param buffer */ + return sst_hsw_load_param_line(hsw, ucontrol->value.bytes.data); +} + +static int hsw_waves_param_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct sst_hsw *hsw = pdata->hsw; + int ret; + enum sst_hsw_module_id id = SST_HSW_MODULE_WAVES; + int param_id = ucontrol->value.bytes.data[0]; + int param_size = WAVES_PARAM_COUNT; + + /* clear param buffer and reset buffer index */ + if (param_id == 0xFF) { + sst_hsw_reset_param_buf(hsw); + return 0; + } + + /* store params into buffer */ + ret = sst_hsw_store_param_line(hsw, ucontrol->value.bytes.data); + if (ret < 0) + return ret; + + if (sst_hsw_is_module_loaded(hsw, id)) { + if (!sst_hsw_is_module_active(hsw, id)) + return 0; + + ret = sst_hsw_module_set_param(hsw, id, 0, param_id, + param_size, ucontrol->value.bytes.data); + } + return ret; +} + /* TLV used by both global and stream volumes */ static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1); @@ -390,6 +433,9 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { /* enable/disable module waves */ SOC_SINGLE_BOOL_EXT("Waves Switch", 0, hsw_waves_switch_get, hsw_waves_switch_put), + /* set parameters to module waves */ + SND_SOC_BYTES_EXT("Waves Set Param", WAVES_PARAM_COUNT, + hsw_waves_param_get, hsw_waves_param_put), }; /* Create DMA buffer page table for DSP */ @@ -1218,6 +1264,10 @@ static int hsw_pcm_runtime_resume(struct device *dev) /* check flag when resume */ if (sst_hsw_is_module_enabled_rtd3(hsw, SST_HSW_MODULE_WAVES)) { ret = sst_hsw_module_enable(hsw, SST_HSW_MODULE_WAVES, 0); + if (ret < 0) + return ret; + /* put parameters from buffer to dsp */ + ret = sst_hsw_launch_param_buf(hsw); if (ret < 0) return ret; /* unset flag */ From 42ce5b8ab81e94089c79791fc682a7f46af9790a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 12 Mar 2015 20:25:07 +0800 Subject: [PATCH 213/411] ASoC: rt5645: Add TDM support for rt5650 rt5650 and rt5645 use different register bits for TDM configuration. This patch modifies rt5645_set_tdm_slot to support both codecs. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 41 ++++++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c9a4c5be083b45..b79347688873ba 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2285,23 +2285,42 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; - unsigned int val = 0; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int i_slot_sft, o_slot_sft, i_width_sht, o_width_sht, en_sft; + unsigned int mask, val = 0; + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + en_sft = 15; + i_slot_sft = 10; + o_slot_sft = 8; + i_width_sht = 6; + o_width_sht = 4; + mask = 0x8ff0; + break; + default: + en_sft = 14; + i_slot_sft = o_slot_sft = 12; + i_width_sht = o_width_sht = 10; + mask = 0x7c00; + break; + } if (rx_mask || tx_mask) { - val |= (1 << 14); - snd_soc_update_bits(codec, RT5645_BASS_BACK, - RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); + val |= (1 << en_sft); + if (rt5645->codec_type == CODEC_TYPE_RT5645) + snd_soc_update_bits(codec, RT5645_BASS_BACK, + RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); } switch (slots) { case 4: - val |= (1 << 12); + val |= (1 << i_slot_sft) | (1 << o_slot_sft); break; case 6: - val |= (2 << 12); + val |= (2 << i_slot_sft) | (2 << o_slot_sft); break; case 8: - val |= (3 << 12); + val |= (3 << i_slot_sft) | (3 << o_slot_sft); break; case 2: default: @@ -2310,20 +2329,20 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (slot_width) { case 20: - val |= (1 << 10); + val |= (1 << i_width_sht) | (1 << o_width_sht); break; case 24: - val |= (2 << 10); + val |= (2 << i_width_sht) | (2 << o_width_sht); break; case 32: - val |= (3 << 10); + val |= (3 << i_width_sht) | (3 << o_width_sht); break; case 16: default: break; } - snd_soc_update_bits(codec, RT5645_TDM_CTRL_1, 0x7c00, val); + snd_soc_update_bits(codec, RT5645_TDM_CTRL_1, mask, val); return 0; } From 718b25c803df8c1f8f2090c115911d123e34c78a Mon Sep 17 00:00:00 2001 From: Anish Kumar Date: Wed, 11 Mar 2015 15:13:16 -0700 Subject: [PATCH 214/411] ASoC: max98925: trivial duplicate typo fix in set_fmt Signed-off-by: Anish Kumar Signed-off-by: Mark Brown --- sound/soc/codecs/max98925.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 34fc4d0441fd54..1f8e8031574982 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -346,7 +346,7 @@ static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, } regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_BCI_MASK | M98925_DAI_BCI_MASK, invert); + M98925_DAI_BCI_MASK, invert); return 0; } From e79d74ab25339437447478e4dfe2b35c5b560512 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 16:57:51 +0100 Subject: [PATCH 215/411] ALSA: control: Fix breakage of user ctl element addition In the commit [2225e79b9b03: 'ALSA: core: reduce stack usage related to snd_ctl_new()'], the id field of the newly added kctl is untouched, thus all attribute like name string remain empty. The fix is just to add the forgotten memcpy of the id field. Fixes: 2225e79b9b03 ('ALSA: core: reduce stack usage related to snd_ctl_new()') Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 54a412af32241c..d677c27746e989 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1267,6 +1267,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, err = snd_ctl_new(&kctl, count, access, file); if (err < 0) return err; + memcpy(&kctl->id, &info->id, sizeof(kctl->id)); kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); if (kctl->private_data == NULL) { From b52551e0d0e659db43f5cfa813ae09d4c3744761 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 13 Mar 2015 10:50:26 +0800 Subject: [PATCH 216/411] ASoC: rt5645: Remove adc stereo2 filter Remove adc stereo2 filter since it is not in rt5645/rt5650 codec. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 3 --- sound/soc/codecs/rt5645.h | 2 -- 2 files changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index b79347688873ba..4c384a14de1d26 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1538,8 +1538,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("adc stereo1 filter", 1, RT5645_PWR_DIG2, RT5645_PWR_ADC_S1F_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY_S("adc stereo2 filter", 1, RT5645_PWR_DIG2, - RT5645_PWR_ADC_S2F_BIT, 0, NULL, 0), SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, rt5645_sto1_adc_l_mix, ARRAY_SIZE(rt5645_sto1_adc_l_mix), NULL, 0), @@ -1729,7 +1727,6 @@ static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, - { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc }, { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc }, { "dac mono left filter", NULL, "DAC MONO L ASRC", is_using_asrc }, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index dbfd98c22f4dea..db78e946287697 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -804,8 +804,6 @@ #define RT5645_PWR_DAC_MF_L_BIT 10 #define RT5645_PWR_DAC_MF_R (0x1 << 9) #define RT5645_PWR_DAC_MF_R_BIT 9 -#define RT5645_PWR_ADC_S2F (0x1 << 8) -#define RT5645_PWR_ADC_S2F_BIT 8 #define RT5645_PWR_PDM1 (0x1 << 7) #define RT5645_PWR_PDM1_BIT 7 #define RT5645_PWR_PDM2 (0x1 << 6) From b2a0bafa758256442e04d1f34d6d0746b846d23d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Mar 2015 17:21:32 +0100 Subject: [PATCH 217/411] ALSA: hda - Use shutdown driver ops instead of reboot notifier The driver shutdown ops is simpler than registering reboot notifier manually. There should be no functional change by this -- the codec driver calls its own callback while the bus driver just calls azx_stop() like before. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 10 ++++++++++ sound/pci/hda/hda_codec.c | 18 ------------------ sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_controller.c | 26 -------------------------- sound/pci/hda/hda_controller.h | 6 ------ sound/pci/hda/hda_intel.c | 16 +++++++++++++--- sound/pci/hda/hda_tegra.c | 16 +++++++++++++--- 7 files changed, 36 insertions(+), 57 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index a49bc45c2ea5ac..1f40ce3c16969f 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -9,6 +9,7 @@ #include #include #include +#include #include #include "hda_codec.h" #include "hda_local.h" @@ -142,6 +143,14 @@ static int hda_codec_driver_remove(struct device *dev) return 0; } +static void hda_codec_driver_shutdown(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + + if (!pm_runtime_suspended(dev) && codec->patch_ops.reboot_notify) + codec->patch_ops.reboot_notify(codec); +} + int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, struct module *owner) { @@ -150,6 +159,7 @@ int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, drv->driver.bus = &snd_hda_bus_type; drv->driver.probe = hda_codec_driver_probe; drv->driver.remove = hda_codec_driver_remove; + drv->driver.shutdown = hda_codec_driver_shutdown; drv->driver.pm = &hda_codec_driver_pm; return driver_register(&drv->driver); } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6fecf57c8d7c5a..3e4fb7a8fdcbb8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4941,24 +4941,6 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) } } -/** - * snd_hda_bus_reboot_notify - call the reboot notifier of each codec - * @bus: HD-audio bus - */ -void snd_hda_bus_reboot_notify(struct hda_bus *bus) -{ - struct hda_codec *codec; - - if (!bus) - return; - list_for_each_entry(codec, &bus->codec_list, list) { - if (hda_codec_is_power_on(codec) && - codec->patch_ops.reboot_notify) - codec->patch_ops.reboot_notify(codec); - } -} -EXPORT_SYMBOL_GPL(snd_hda_bus_reboot_notify); - /** * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index bf9efb7e1b9a3f..70851e6d5f109e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -563,7 +563,6 @@ extern const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[]; * Misc */ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); -void snd_hda_bus_reboot_notify(struct hda_bus *bus); void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index cae50d5ffb814f..b1143f22a0c251 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include "hda_controller.h" @@ -1972,30 +1971,5 @@ int azx_init_stream(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_init_stream); -/* - * reboot notifier for hang-up problem at power-down - */ -static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf) -{ - struct azx *chip = container_of(nb, struct azx, reboot_notifier); - snd_hda_bus_reboot_notify(chip->bus); - azx_stop_chip(chip); - return NOTIFY_OK; -} - -void azx_notifier_register(struct azx *chip) -{ - chip->reboot_notifier.notifier_call = azx_halt; - register_reboot_notifier(&chip->reboot_notifier); -} -EXPORT_SYMBOL_GPL(azx_notifier_register); - -void azx_notifier_unregister(struct azx *chip) -{ - if (chip->reboot_notifier.notifier_call) - unregister_reboot_notifier(&chip->reboot_notifier); -} -EXPORT_SYMBOL_GPL(azx_notifier_unregister); - MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Common HDA driver functions"); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 94c1a4719f7fbd..be1b7ded8d8202 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -362,9 +362,6 @@ struct azx { /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; - /* reboot notifier (for mysterious hangup problem at power-down) */ - struct notifier_block reboot_notifier; - #ifdef CONFIG_SND_HDA_DSP_LOADER struct azx_dev saved_azx_dev; #endif @@ -437,7 +434,4 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots); int azx_codec_configure(struct azx *chip); int azx_init_stream(struct azx *chip); -void azx_notifier_register(struct azx *chip); -void azx_notifier_unregister(struct azx *chip); - #endif /* __SOUND_HDA_CONTROLLER_H */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index dbc5a593da46af..25668fde848017 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1066,8 +1066,6 @@ static int azx_free(struct azx *chip) azx_del_card_list(chip); - azx_notifier_unregister(chip); - hda->init_failed = 1; /* to be sure */ complete_all(&hda->probe_wait); @@ -1900,7 +1898,6 @@ static int azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; - azx_notifier_register(chip); azx_add_card_list(chip); snd_hda_set_power_save(chip->bus, power_save * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) @@ -1921,6 +1918,18 @@ static void azx_remove(struct pci_dev *pci) snd_card_free(card); } +static void azx_shutdown(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip; + + if (!card) + return; + chip = card->private_data; + if (chip && chip->running) + azx_stop_chip(chip); +} + /* PCI IDs */ static const struct pci_device_id azx_ids[] = { /* CPT */ @@ -2143,6 +2152,7 @@ static struct pci_driver azx_driver = { .id_table = azx_ids, .probe = azx_probe, .remove = azx_remove, + .shutdown = azx_shutdown, .driver = { .pm = AZX_PM_OPS, }, diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 7586abe91dfb3f..2e4fd5c56d3b96 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -290,8 +290,6 @@ static int hda_tegra_dev_free(struct snd_device *device) int i; struct azx *chip = device->device_data; - azx_notifier_unregister(chip); - if (chip->initialized) { for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); @@ -502,7 +500,6 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; chip->running = 1; - azx_notifier_register(chip); snd_hda_set_power_save(chip->bus, power_save * 1000); return 0; @@ -517,6 +514,18 @@ static int hda_tegra_remove(struct platform_device *pdev) return snd_card_free(dev_get_drvdata(&pdev->dev)); } +static void hda_tegra_shutdown(struct platform_device *pdev) +{ + struct snd_card *card = dev_get_drvdata(&pdev->dev); + struct azx *chip; + + if (!card) + return; + chip = card->private_data; + if (chip && chip->running) + azx_stop_chip(chip); +} + static struct platform_driver tegra_platform_hda = { .driver = { .name = "tegra-hda", @@ -525,6 +534,7 @@ static struct platform_driver tegra_platform_hda = { }, .probe = hda_tegra_probe, .remove = hda_tegra_remove, + .shutdown = hda_tegra_shutdown, }; module_platform_driver(tegra_platform_hda); From 97a33ced310ab9bdb16699c2c64b28f29de0a23d Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Fri, 13 Mar 2015 00:54:16 -0700 Subject: [PATCH 218/411] ASoC: qcom: Change qcom,adsp in LPASS CPU bindings Change the representation of the audio DSP, in the LPASS CPU bindings description, from a required subnode to an optional phandle. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt index e7c6e9321863f0..e00732dac9393e 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -22,13 +22,9 @@ Required properties: - reg-names : A list which must include the following entries: * "lpass-lpaif" -Required subnodes: +Optional properties: -- qcom,adsp : Audio DSP sub-node - -Optional Audio DSP subnode properties: - -- status : "disabled" indicates the adsp is not available. +- qcom,adsp : Phandle for the audio DSP node Example: @@ -43,7 +39,5 @@ lpass@28100000 { pinctrl-1 = <&mi2s_idle>; reg = <0x28100000 0x10000>; reg-names = "lpass-lpaif"; - qcom,adsp { - status = "disabled"; - }; + qcom,adsp = <&adsp>; }; From 8ebe148be9aa12641c62a3c99c65859bf95445fe Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Fri, 13 Mar 2015 00:54:17 -0700 Subject: [PATCH 219/411] ASoC: qcom: Modify test for DSP in LPASS driver As the representation of the DSP in the device tree has changed from a required subnode to an optional phandle, modify the test for DSP existence in the LPASS CPU DAI driver, accordingly. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 33 +++++++-------------------------- 1 file changed, 7 insertions(+), 26 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index d5167131787f3a..6698d058de2960 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -359,45 +359,26 @@ static const struct regmap_config lpass_cpu_regmap_config = { .cache_type = REGCACHE_FLAT, }; -static int lpass_cpu_parse_of(struct device *dev) +static int lpass_cpu_platform_probe(struct platform_device *pdev) { + struct lpass_data *drvdata; struct device_node *dsp_of_node; + struct resource *res; + int ret; - dsp_of_node = of_get_child_by_name(dev->of_node, "qcom,adsp"); - if (!dsp_of_node) { - dev_err(dev, "%s() error getting qcom,adsp sub-node\n", - __func__); - return -EINVAL; - } - - if (of_device_is_available(dsp_of_node)) { - dev_err(dev, "%s() DSP exists and holds audio resources\n", + dsp_of_node = of_parse_phandle(pdev->dev.of_node, "qcom,adsp", 0); + if (dsp_of_node) { + dev_err(&pdev->dev, "%s() DSP exists and holds audio resources\n", __func__); return -EBUSY; } - return 0; -} - -static int lpass_cpu_platform_probe(struct platform_device *pdev) -{ - struct lpass_data *drvdata; - struct resource *res; - int ret; - drvdata = devm_kzalloc(&pdev->dev, sizeof(struct lpass_data), GFP_KERNEL); if (!drvdata) return -ENOMEM; platform_set_drvdata(pdev, drvdata); - ret = lpass_cpu_parse_of(&pdev->dev); - if (ret) { - dev_err(&pdev->dev, "%s() error getting DT node info: %d\n", - __func__, ret); - return ret; - } - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "lpass-lpaif"); if (!res) { dev_err(&pdev->dev, "%s() error getting resource\n", __func__); From 81ddc01d56d773eed414b5c1a7f1dba1db35f538 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Fri, 13 Mar 2015 01:01:04 -0700 Subject: [PATCH 220/411] ASoC: qcom: Document Storm bindings Add documentation to the sound directory of the device-tree bindings for the soundcard of the Storm board. Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/storm.txt | 23 +++++++++++++++++++ 1 file changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/storm.txt diff --git a/Documentation/devicetree/bindings/sound/storm.txt b/Documentation/devicetree/bindings/sound/storm.txt new file mode 100644 index 00000000000000..062a4c185fa9d0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/storm.txt @@ -0,0 +1,23 @@ +* Sound complex for Storm boards + +Models a soundcard for Storm boards with the Qualcomm Technologies IPQ806x SOC +connected to a MAX98357A DAC via I2S. + +Required properties: + +- compatible : "google,storm-audio" +- cpu : Phandle of the CPU DAI +- codec : Phandle of the codec DAI + +Optional properties: + +- qcom,model : The user-visible name of this sound card. + +Example: + +sound { + compatible = "google,storm-audio"; + qcom,model = "ipq806x-storm"; + cpu = <&lpass_cpu>; + codec = <&max98357a>; +}; From 79119c798649630b3191784a708b45cea4e31daf Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Fri, 13 Mar 2015 01:01:05 -0700 Subject: [PATCH 221/411] ASoC: qcom: Add Storm machine driver Add machine driver for the Storm board with the IPQ806X SOC connected to the MAX98357A DAC. Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- sound/soc/qcom/storm.c | 162 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 162 insertions(+) create mode 100644 sound/soc/qcom/storm.c diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c new file mode 100644 index 00000000000000..b8bd296190add8 --- /dev/null +++ b/sound/soc/qcom/storm.c @@ -0,0 +1,162 @@ +/* + * Copyright (c) 2010-2011,2013-2015 The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * storm.c -- ALSA SoC machine driver for QTi ipq806x-based Storm board + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#define STORM_SYSCLK_MULT 4 + +static int storm_ops_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct snd_soc_card *card = soc_runtime->card; + snd_pcm_format_t format = params_format(params); + unsigned int rate = params_rate(params); + unsigned int sysclk_freq; + int bitwidth, ret; + + bitwidth = snd_pcm_format_width(format); + if (bitwidth < 0) { + dev_err(card->dev, "%s() invalid bit width given: %d\n", + __func__, bitwidth); + return bitwidth; + } + + /* + * as the CPU DAI is the I2S bus master and no system clock is needed by + * the MAX98357a DAC, simply set the system clock to be a constant + * multiple of the bit clock for the clock divider + */ + sysclk_freq = rate * bitwidth * 2 * STORM_SYSCLK_MULT; + + ret = snd_soc_dai_set_sysclk(soc_runtime->cpu_dai, 0, sysclk_freq, 0); + if (ret) { + dev_err(card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, sysclk_freq, ret); + return ret; + } + + return 0; +} + +static struct snd_soc_ops storm_soc_ops = { + .hw_params = storm_ops_hw_params, +}; + +static struct snd_soc_dai_link storm_dai_link = { + .name = "Primary", + .stream_name = "Primary", + .codec_dai_name = "HiFi", + .ops = &storm_soc_ops, +}; + +static struct snd_soc_card storm_soc_card = { + .name = "ipq806x-storm", + .dev = NULL, +}; + +static int storm_parse_of(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link = card->dai_link; + struct device_node *np = card->dev->of_node; + + dai_link->cpu_of_node = of_parse_phandle(np, "cpu", 0); + if (!dai_link->cpu_of_node) { + dev_err(card->dev, "%s() error getting cpu phandle\n", + __func__); + return -EINVAL; + } + dai_link->platform_of_node = dai_link->cpu_of_node; + + dai_link->codec_of_node = of_parse_phandle(np, "codec", 0); + if (!dai_link->codec_of_node) { + dev_err(card->dev, "%s() error getting codec phandle\n", + __func__); + return -EINVAL; + } + + return 0; +} + +static int storm_platform_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &storm_soc_card; + int ret; + + if (card->dev) { + dev_err(&pdev->dev, "%s() error, existing soundcard\n", + __func__); + return -ENODEV; + } + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = snd_soc_of_parse_card_name(card, "qcom,model"); + if (ret) { + dev_err(&pdev->dev, "%s() error parsing card name: %d\n", + __func__, ret); + return ret; + } + + card->dai_link = &storm_dai_link; + card->num_links = 1; + + ret = storm_parse_of(card); + if (ret) { + dev_err(&pdev->dev, "%s() error resolving dai links: %d\n", + __func__, ret); + return ret; + } + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret == -EPROBE_DEFER) { + card->dev = NULL; + return ret; + } else if (ret) { + dev_err(&pdev->dev, "%s() error registering soundcard: %d\n", + __func__, ret); + return ret; + } + + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id storm_device_id[] = { + { .compatible = "google,storm-audio" }, + {}, +}; +MODULE_DEVICE_TABLE(of, storm_device_id); +#endif + +static struct platform_driver storm_platform_driver = { + .driver = { + .name = "storm-audio", + .of_match_table = + of_match_ptr(storm_device_id), + }, + .probe = storm_platform_probe, +}; +module_platform_driver(storm_platform_driver); + +MODULE_DESCRIPTION("QTi IPQ806x-based Storm Machine Driver"); +MODULE_LICENSE("GPL v2"); From f380dd3f3cd77df3c5d297a635b635f72fb5a2b1 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Fri, 13 Mar 2015 01:01:06 -0700 Subject: [PATCH 222/411] ASoC: qcom: Add ability to build QCOM drivers Define the LPASS platform driver, the LPASS CPU DAI driver and the Storm machine driver configurations, and how to build them. Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 25 +++++++++++++++++++++++++ sound/soc/qcom/Makefile | 11 +++++++++++ 2 files changed, 36 insertions(+) create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig new file mode 100644 index 00000000000000..5f58e4f1bca98b --- /dev/null +++ b/sound/soc/qcom/Kconfig @@ -0,0 +1,25 @@ +config SND_SOC_QCOM + tristate "ASoC support for QCOM platforms" + help + Say Y or M if you want to add support to use audio devices + in Qualcomm Technologies SOC-based platforms. + +config SND_SOC_LPASS_CPU + tristate + depends on SND_SOC_QCOM + select REGMAP_MMIO + +config SND_SOC_LPASS_PLATFORM + tristate + depends on SND_SOC_QCOM + select REGMAP_MMIO + +config SND_SOC_STORM + tristate "ASoC I2S support for Storm boards" + depends on (ARCH_QCOM && SND_SOC_QCOM) || COMPILE_TEST + select SND_SOC_LPASS_CPU + select SND_SOC_LPASS_PLATFORM + select SND_SOC_MAX98357A + help + Say Y or M if you want add support for SoC audio on the + Qualcomm Technologies IPQ806X-based Storm board. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile new file mode 100644 index 00000000000000..c5ce96c761c47b --- /dev/null +++ b/sound/soc/qcom/Makefile @@ -0,0 +1,11 @@ +# Platform +snd-soc-lpass-cpu-objs := lpass-cpu.o +snd-soc-lpass-platform-objs := lpass-platform.o + +obj-$(CONFIG_SND_SOC_LPASS_CPU) += snd-soc-lpass-cpu.o +obj-$(CONFIG_SND_SOC_LPASS_PLATFORM) += snd-soc-lpass-platform.o + +# Machine +snd-soc-storm-objs := storm.o + +obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o From 6a328885896ef087a28173792ea93f4dde9782ef Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Fri, 13 Mar 2015 01:01:07 -0700 Subject: [PATCH 223/411] ASoC: Allow for building QCOM drivers Allow for the Qualcomm Technologies ASoC drivers to build. Signed-off-by: Kenneth Westfield Acked-by: Banajit Goswami Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index dcc79aa0236b54..3ba52da18bc69a 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -47,6 +47,7 @@ source "sound/soc/kirkwood/Kconfig" source "sound/soc/intel/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/qcom/Kconfig" source "sound/soc/rockchip/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/sh/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 5b3c8f67c8db7a..974ba708b4826a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -28,6 +28,7 @@ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ obj-$(CONFIG_SND_SOC) += pxa/ +obj-$(CONFIG_SND_SOC) += qcom/ obj-$(CONFIG_SND_SOC) += rockchip/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += sh/ From 6b5b042d4c675cb9d3446a1cdcaca98e715ba812 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 15 Mar 2015 10:27:20 +0100 Subject: [PATCH 224/411] ASoC: Make snd_soc_dapm_kcontrol_codec() inline snd_soc_dapm_kcontrol_codec() is a extremely simple function and inlining it typically results in less code than necessary for calling the non-inlined version of the function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 - include/sound/soc.h | 13 +++++++++++++ sound/soc/soc-dapm.c | 10 ---------- 3 files changed, 13 insertions(+), 11 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8d7416e4686174..78633efd40ee6d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -440,7 +440,6 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list); -struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade19562857..85a6853a40bbb3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1258,6 +1258,19 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( return component->dapm_ptr; } +/** + * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol + * @kcontrol: The kcontrol + * + * This function must only be used on DAPM contexts that are known to be part of + * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined. + */ +static inline struct snd_soc_codec *snd_soc_dapm_kcontrol_codec( + struct snd_kcontrol *kcontrol) +{ + return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); +} + /* codec IO */ unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6f88202b8c99a..95337c832258d2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -473,16 +473,6 @@ struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( } EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_dapm); -/** - * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol - * @kcontrol: The kcontrol - */ -struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) -{ - return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); - static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; From fa41181fe37530d78acb25b4e2c9c019241cbbf6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 15 Mar 2015 12:15:24 +0100 Subject: [PATCH 225/411] ASoC: nuc900: No need to track the dma buffer in the driver state struct The DMA buffer and address can be accessed through the snd_pcm_runtime. There is no need to manually track them in the driver's state struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/nuc900/nuc900-audio.h | 3 --- sound/soc/nuc900/nuc900-pcm.c | 31 +++++-------------------------- 2 files changed, 5 insertions(+), 29 deletions(-) diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h index 59f7e8ed1a6898..d0b725705914a1 100644 --- a/sound/soc/nuc900/nuc900-audio.h +++ b/sound/soc/nuc900/nuc900-audio.h @@ -100,10 +100,7 @@ struct nuc900_audio { void __iomem *mmio; spinlock_t lock; - dma_addr_t dma_addr[2]; - unsigned long buffersize[2]; unsigned long irq_num; - struct snd_pcm_substream *substream; struct resource *res; struct clk *clk; struct device *dev; diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index b809fa909e4d6d..5ae5ca15b6d6bf 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -42,29 +42,10 @@ static const struct snd_pcm_hardware nuc900_pcm_hardware = { static int nuc900_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct nuc900_audio *nuc900_audio = runtime->private_data; - unsigned long flags; - int ret = 0; - - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - return ret; - - spin_lock_irqsave(&nuc900_audio->lock, flags); - - nuc900_audio->substream = substream; - nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr; - nuc900_audio->buffersize[substream->stream] = - params_buffer_bytes(params); - - spin_unlock_irqrestore(&nuc900_audio->lock, flags); - - return ret; + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } -static void nuc900_update_dma_register(struct snd_pcm_substream *substream, - dma_addr_t dma_addr, size_t count) +static void nuc900_update_dma_register(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct nuc900_audio *nuc900_audio = runtime->private_data; @@ -78,8 +59,8 @@ static void nuc900_update_dma_register(struct snd_pcm_substream *substream, mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH; } - AUDIO_WRITE(mmio_addr, dma_addr); - AUDIO_WRITE(mmio_len, count); + AUDIO_WRITE(mmio_addr, runtime->dma_addr); + AUDIO_WRITE(mmio_len, runtime->dma_bytes); } static void nuc900_dma_start(struct snd_pcm_substream *substream) @@ -170,9 +151,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream) spin_lock_irqsave(&nuc900_audio->lock, flags); - nuc900_update_dma_register(substream, - nuc900_audio->dma_addr[substream->stream], - nuc900_audio->buffersize[substream->stream]); + nuc900_update_dma_register(substream); val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); From c36aa0a1929a1f0f0b8c374276e49cc663e8f957 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 16 Mar 2015 14:39:57 +0800 Subject: [PATCH 226/411] ASoC: rt5677: add API to select ASRC clock source This patch defines an API to select the clock source for specified filters. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 163 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 79 ++++++++++++++++++ 2 files changed, 242 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c2a6e409135770..af182586712d42 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1034,6 +1034,169 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } +/** + * rt5677_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5677 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc3_mask = 0, asrc3_value = 0; + unsigned int asrc4_mask = 0, asrc4_value = 0; + unsigned int asrc5_mask = 0, asrc5_value = 0; + unsigned int asrc6_mask = 0, asrc6_value = 0; + unsigned int asrc7_mask = 0, asrc7_value = 0; + + switch (clk_src) { + case RT5677_CLK_SEL_SYS: + case RT5677_CLK_SEL_I2S1_ASRC: + case RT5677_CLK_SEL_I2S2_ASRC: + case RT5677_CLK_SEL_I2S3_ASRC: + case RT5677_CLK_SEL_I2S4_ASRC: + case RT5677_CLK_SEL_I2S5_ASRC: + case RT5677_CLK_SEL_I2S6_ASRC: + case RT5677_CLK_SEL_SYS2: + case RT5677_CLK_SEL_SYS3: + case RT5677_CLK_SEL_SYS4: + case RT5677_CLK_SEL_SYS5: + case RT5677_CLK_SEL_SYS6: + case RT5677_CLK_SEL_SYS7: + break; + + default: + return -EINVAL; + } + + /* ASRC 3 */ + if (filter_mask & RT5677_DA_STEREO_FILTER) { + asrc3_mask |= RT5677_DA_STO_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5677_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO2_L_FILTER) { + asrc3_mask |= RT5677_DA_MONO2L_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_MONO2L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO2L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO2_R_FILTER) { + asrc3_mask |= RT5677_DA_MONO2R_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_MONO2R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO2R_CLK_SEL_SFT); + } + + if (asrc3_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_3, asrc3_mask, + asrc3_value); + + /* ASRC 4 */ + if (filter_mask & RT5677_DA_MONO3_L_FILTER) { + asrc4_mask |= RT5677_DA_MONO3L_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO3L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO3L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO3_R_FILTER) { + asrc4_mask |= RT5677_DA_MONO3R_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO3R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO3R_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO4_L_FILTER) { + asrc4_mask |= RT5677_DA_MONO4L_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO4L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO4L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO4_R_FILTER) { + asrc4_mask |= RT5677_DA_MONO4R_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO4R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO4R_CLK_SEL_SFT); + } + + if (asrc4_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_4, asrc4_mask, + asrc4_value); + + /* ASRC 5 */ + if (filter_mask & RT5677_AD_STEREO1_FILTER) { + asrc5_mask |= RT5677_AD_STO1_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO2_FILTER) { + asrc5_mask |= RT5677_AD_STO2_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO2_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO2_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO3_FILTER) { + asrc5_mask |= RT5677_AD_STO3_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO3_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO3_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO4_FILTER) { + asrc5_mask |= RT5677_AD_STO4_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO4_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO4_CLK_SEL_SFT); + } + + if (asrc5_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_5, asrc5_mask, + asrc5_value); + + /* ASRC 6 */ + if (filter_mask & RT5677_AD_MONO_L_FILTER) { + asrc6_mask |= RT5677_AD_MONOL_CLK_SEL_MASK; + asrc6_value = (asrc6_value & ~RT5677_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5677_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_MONO_R_FILTER) { + asrc6_mask |= RT5677_AD_MONOR_CLK_SEL_MASK; + asrc6_value = (asrc6_value & ~RT5677_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5677_AD_MONOR_CLK_SEL_SFT); + } + + if (asrc6_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_6, asrc6_mask, + asrc6_value); + + /* ASRC 7 */ + if (filter_mask & RT5677_DSP_OB_0_3_FILTER) { + asrc7_mask |= RT5677_DSP_OB_0_3_CLK_SEL_MASK; + asrc7_value = (asrc7_value & ~RT5677_DSP_OB_0_3_CLK_SEL_MASK) + | (clk_src << RT5677_DSP_OB_0_3_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DSP_OB_4_7_FILTER) { + asrc7_mask |= RT5677_DSP_OB_4_7_CLK_SEL_MASK; + asrc7_value = (asrc7_value & ~RT5677_DSP_OB_4_7_CLK_SEL_MASK) + | (clk_src << RT5677_DSP_OB_4_7_CLK_SEL_SFT); + } + + if (asrc7_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_7, asrc7_mask, + asrc7_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5677_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 07df96b43f5962..9dceb41d18ea91 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1406,6 +1406,46 @@ #define RT5677_DSP_CLK_SRC_PLL2 (0x0 << 7) #define RT5677_DSP_CLK_SRC_BYPASS (0x1 << 7) +/* ASRC Control 3 (0x85) */ +#define RT5677_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5677_DA_STO_CLK_SEL_SFT 12 +#define RT5677_DA_MONO2L_CLK_SEL_MASK (0xf << 4) +#define RT5677_DA_MONO2L_CLK_SEL_SFT 4 +#define RT5677_DA_MONO2R_CLK_SEL_MASK (0xf << 0) +#define RT5677_DA_MONO2R_CLK_SEL_SFT 0 + +/* ASRC Control 4 (0x86) */ +#define RT5677_DA_MONO3L_CLK_SEL_MASK (0xf << 12) +#define RT5677_DA_MONO3L_CLK_SEL_SFT 12 +#define RT5677_DA_MONO3R_CLK_SEL_MASK (0xf << 8) +#define RT5677_DA_MONO3R_CLK_SEL_SFT 8 +#define RT5677_DA_MONO4L_CLK_SEL_MASK (0xf << 4) +#define RT5677_DA_MONO4L_CLK_SEL_SFT 4 +#define RT5677_DA_MONO4R_CLK_SEL_MASK (0xf << 0) +#define RT5677_DA_MONO4R_CLK_SEL_SFT 0 + +/* ASRC Control 5 (0x87) */ +#define RT5677_AD_STO1_CLK_SEL_MASK (0xf << 12) +#define RT5677_AD_STO1_CLK_SEL_SFT 12 +#define RT5677_AD_STO2_CLK_SEL_MASK (0xf << 8) +#define RT5677_AD_STO2_CLK_SEL_SFT 8 +#define RT5677_AD_STO3_CLK_SEL_MASK (0xf << 4) +#define RT5677_AD_STO3_CLK_SEL_SFT 4 +#define RT5677_AD_STO4_CLK_SEL_MASK (0xf << 0) +#define RT5677_AD_STO4_CLK_SEL_SFT 0 + +/* ASRC Control 6 (0x88) */ +#define RT5677_AD_MONOL_CLK_SEL_MASK (0xf << 12) +#define RT5677_AD_MONOL_CLK_SEL_SFT 12 +#define RT5677_AD_MONOR_CLK_SEL_MASK (0xf << 8) +#define RT5677_AD_MONOR_CLK_SEL_SFT 8 + +/* ASRC Control 7 (0x89) */ +#define RT5677_DSP_OB_0_3_CLK_SEL_MASK (0xf << 12) +#define RT5677_DSP_OB_0_3_CLK_SEL_SFT 12 +#define RT5677_DSP_OB_4_7_CLK_SEL_MASK (0xf << 8) +#define RT5677_DSP_OB_4_7_CLK_SEL_SFT 8 + /* VAD Function Control 4 (0x9f) */ #define RT5677_VAD_SRC_MASK (0x7 << 8) #define RT5677_VAD_SRC_SFT 8 @@ -1670,6 +1710,42 @@ enum rt5677_type { RT5676, }; +/* ASRC clock source selection */ +enum { + RT5677_CLK_SEL_SYS, + RT5677_CLK_SEL_I2S1_ASRC, + RT5677_CLK_SEL_I2S2_ASRC, + RT5677_CLK_SEL_I2S3_ASRC, + RT5677_CLK_SEL_I2S4_ASRC, + RT5677_CLK_SEL_I2S5_ASRC, + RT5677_CLK_SEL_I2S6_ASRC, + RT5677_CLK_SEL_SYS2, + RT5677_CLK_SEL_SYS3, + RT5677_CLK_SEL_SYS4, + RT5677_CLK_SEL_SYS5, + RT5677_CLK_SEL_SYS6, + RT5677_CLK_SEL_SYS7, +}; + +/* filter mask */ +enum { + RT5677_DA_STEREO_FILTER = 0x1, + RT5677_DA_MONO2_L_FILTER = (0x1 << 1), + RT5677_DA_MONO2_R_FILTER = (0x1 << 2), + RT5677_DA_MONO3_L_FILTER = (0x1 << 3), + RT5677_DA_MONO3_R_FILTER = (0x1 << 4), + RT5677_DA_MONO4_L_FILTER = (0x1 << 5), + RT5677_DA_MONO4_R_FILTER = (0x1 << 6), + RT5677_AD_STEREO1_FILTER = (0x1 << 7), + RT5677_AD_STEREO2_FILTER = (0x1 << 8), + RT5677_AD_STEREO3_FILTER = (0x1 << 9), + RT5677_AD_STEREO4_FILTER = (0x1 << 10), + RT5677_AD_MONO_L_FILTER = (0x1 << 11), + RT5677_AD_MONO_R_FILTER = (0x1 << 12), + RT5677_DSP_OB_0_3_FILTER = (0x1 << 13), + RT5677_DSP_OB_4_7_FILTER = (0x1 << 14), +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1696,4 +1772,7 @@ struct rt5677_priv { bool is_vref_slow; }; +int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + #endif /* __RT5677_H__ */ From 99d422341376443a98ef22d0f0003b3381f32186 Mon Sep 17 00:00:00 2001 From: Songjun Wu Date: Thu, 12 Mar 2015 10:17:11 +0800 Subject: [PATCH 227/411] ASoC: wm8731: let codec to manage clock by itself Enable WM8731 to support common clock framework. Signed-off-by: Songjun Wu Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 098c143f44d653..8df1550f74b5a5 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -390,6 +392,8 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (wm8731->mclk && clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -491,6 +495,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (wm8731->mclk) + clk_prepare_enable(wm8731->mclk); break; case SND_SOC_BIAS_PREPARE: break; @@ -509,6 +515,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + if (wm8731->mclk) + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -667,6 +675,19 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&spi->dev, "Assuming static MCLK\n"); + } else { + dev_err(&spi->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -718,6 +739,19 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&i2c->dev, "Assuming static MCLK\n"); + } else { + dev_err(&i2c->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); From 0dd96b3e39df83265ef3f79170a623cebee50380 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Mar 2015 16:39:56 +0100 Subject: [PATCH 228/411] ASoC: rt286: Drop unnecessary dapm bias_level initialization The default value for the bias_level is SND_SOC_BIAS_OFF when probe is being called, there is no need to initialize it explicitly. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 49c44a77b5188d..ea967978ec6bc2 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1048,7 +1048,6 @@ static int rt286_probe(struct snd_soc_codec *codec) struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); rt286->codec = codec; - codec->dapm.bias_level = SND_SOC_BIAS_OFF; if (rt286->i2c->irq) { regmap_update_bits(rt286->regmap, From f263fa3e0f0b1d610f7a9d72a91fe67059d5564f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 16 Mar 2015 16:39:57 +0100 Subject: [PATCH 229/411] ASoC: wm2200: Drop unnecessary dapm bias_level initialization The default value for the bias_level is SND_SOC_BIAS_OFF when probe is being called, there is no need to initialize it explicitly. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index b48694a8d21370..5a9da28f4f333c 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1554,7 +1554,6 @@ static int wm2200_probe(struct snd_soc_codec *codec) int ret; wm2200->codec = codec; - codec->dapm.bias_level = SND_SOC_BIAS_OFF; ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); if (ret != 0) From 066d7b87fa11213d7eca7af8fdd8447e1c62117b Mon Sep 17 00:00:00 2001 From: Jin Yao Date: Tue, 17 Mar 2015 10:23:30 +0800 Subject: [PATCH 230/411] ASoC: Intel: Add suspend_pre and resume_post for Braswell snd_soc_card On Braswell, we need to add some machine specific setting before suspend and after resume. For example, disable/enable jack detection in codec so use snd_soc_card suspend_pre and resume_post ops for this purpose. Signed-off-by: Jin Yao Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index 279df4c43de16c..c41fae3eb9ca8d 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -267,6 +267,35 @@ static struct snd_soc_dai_link cht_dailink[] = { }, }; +static int cht_suspend_pre(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); + rt5670_jack_suspend(codec); + break; + } + } + return 0; +} + +static int cht_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + dev_dbg(codec->dev, "enabling jack detect for resume.\n"); + rt5670_jack_resume(codec); + break; + } + } + + return 0; +} + /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "cherrytrailcraudio", @@ -278,6 +307,8 @@ static struct snd_soc_card snd_soc_card_cht = { .num_dapm_routes = ARRAY_SIZE(cht_audio_map), .controls = cht_mc_controls, .num_controls = ARRAY_SIZE(cht_mc_controls), + .suspend_pre = cht_suspend_pre, + .resume_post = cht_resume_post, }; static int snd_cht_mc_probe(struct platform_device *pdev) From 6b3b58d97fb2d03f8f1d009a77baece311aa0d16 Mon Sep 17 00:00:00 2001 From: Jin Yao Date: Tue, 17 Mar 2015 10:23:31 +0800 Subject: [PATCH 231/411] ASoC: Intel: move the jack creation to Braswell machine driver The jack creation code was in rt5670 codec driver before due to the jack resources (gpio/irq) were defined under the node of codec device in ACPI on Braswell. We used the snd_soc_jack_new() to create a jack instance. But now snd_soc_jack_new() is removed from upstream and we can't use snd_soc_card_jack_new() in codec driver, so we move the jack creation code to machine driver and pass the jack instance to codec driver for further processing. Signed-off-by: Bard Liao Signed-off-by: Jin Yao Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index c41fae3eb9ca8d..4204fc4f1badfd 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -22,6 +22,7 @@ #include #include #include +#include #include "../codecs/rt5670.h" #include "sst-atom-controls.h" @@ -29,6 +30,20 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "rt5670-aif1" +static struct snd_soc_jack cht_bsw_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin cht_bsw_headset_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, +}; + static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) { int i; @@ -178,6 +193,15 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) | RT5670_AD_MONO_L_FILTER | RT5670_AD_MONO_R_FILTER, RT5670_CLK_SEL_I2S1_ASRC); + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2, &cht_bsw_headset, + cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins)); + if (ret) + return ret; + + rt5670_set_jack_detect(codec, &cht_bsw_headset); return 0; } From 379c4b05dbfee4f8e003c9bd4c92b8c4c7146277 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 16 Mar 2015 04:45:54 +0000 Subject: [PATCH 232/411] ASoC: ak4642: tidyup DAPM route for playback It needs DAC -> Playback route instead of direct settings via SND_SOC_DAPM_DAC. otherwise, it can't find correct path if sound card used prefix name Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index dde8b49c19add3..7255f69521538e 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -187,7 +187,7 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { ARRAY_SIZE(ak4642_lout_mixer_controls)), /* DAC */ - SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0), + SND_SOC_DAPM_DAC("DAC", NULL, PW_MGMT1, 2, 0), }; static const struct snd_soc_dapm_route ak4642_intercon[] = { @@ -205,6 +205,8 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = { {"DACH", NULL, "DAC"}, {"LINEOUT Mixer", "DACL", "DAC"}, + + { "DAC", NULL, "Playback" }, }; /* From fb83b6351052bf78686df2559f7ea6b10e596850 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2015 23:34:34 +0100 Subject: [PATCH 233/411] ALSA: hda - Simplify PCM setup overrides This patch does two things: - code refactoring with a local helper function, - allow codec drivers to provide the specific PCM stream info pointers only for overriding the non-NULL entries, instead of copying the whole. This simplifies the codec driver side (currently the only user is alc269's 44kHz fixed rate). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 110 ++++++++++++++++++---------------- sound/pci/hda/patch_realtek.c | 41 ------------- 2 files changed, 60 insertions(+), 91 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ebdbc023583dc2..27ce54701f0f6e 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5137,6 +5137,33 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx, strlcat(str, sfx, len); } +/* copy PCM stream info from @default_str, and override non-NULL entries + * from @spec_str and @nid + */ +static void setup_pcm_stream(struct hda_pcm_stream *str, + const struct hda_pcm_stream *default_str, + const struct hda_pcm_stream *spec_str, + hda_nid_t nid) +{ + *str = *default_str; + if (nid) + str->nid = nid; + if (spec_str) { + if (spec_str->substreams) + str->substreams = spec_str->substreams; + if (spec_str->channels_min) + str->channels_min = spec_str->channels_min; + if (spec_str->channels_max) + str->channels_max = spec_str->channels_max; + if (spec_str->rates) + str->rates = spec_str->rates; + if (spec_str->formats) + str->formats = spec_str->formats; + if (spec_str->maxbps) + str->maxbps = spec_str->maxbps; + } +} + /** * snd_hda_gen_build_pcms - build PCM streams based on the parsed results * @codec: the HDA codec @@ -5147,7 +5174,6 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; struct hda_pcm *info; - const struct hda_pcm_stream *p; bool have_multi_adcs; if (spec->no_analog) @@ -5162,11 +5188,10 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) spec->pcm_rec[0] = info; if (spec->multiout.num_dacs > 0) { - p = spec->stream_analog_playback; - if (!p) - p = &pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_analog_playback, + spec->stream_analog_playback, + spec->multiout.dac_nids[0]); info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels; if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT && @@ -5175,15 +5200,11 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) snd_pcm_2_1_chmaps; } if (spec->num_adc_nids) { - p = spec->stream_analog_capture; - if (!p) { - if (spec->dyn_adc_switch) - p = &dyn_adc_pcm_analog_capture; - else - p = &pcm_analog_capture; - } - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + (spec->dyn_adc_switch ? + &dyn_adc_pcm_analog_capture : &pcm_analog_capture), + spec->stream_analog_capture, + spec->adc_nids[0]); } skip_analog: @@ -5202,20 +5223,16 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) info->pcm_type = spec->dig_out_type; else info->pcm_type = HDA_PCM_TYPE_SPDIF; - if (spec->multiout.dig_out_nid) { - p = spec->stream_digital_playback; - if (!p) - p = &pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; - } - if (spec->dig_in_nid) { - p = spec->stream_digital_capture; - if (!p) - p = &pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; - } + if (spec->multiout.dig_out_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_digital_playback, + spec->stream_digital_playback, + spec->multiout.dig_out_nid); + if (spec->dig_in_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_digital_capture, + spec->stream_digital_capture, + spec->dig_in_nid); } if (spec->no_analog) @@ -5236,31 +5253,24 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) if (!info) return -ENOMEM; spec->pcm_rec[2] = info; - if (spec->alt_dac_nid) { - p = spec->stream_analog_alt_playback; - if (!p) - p = &pcm_analog_alt_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->alt_dac_nid; - } else { - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = - pcm_null_stream; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; - } + if (spec->alt_dac_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_analog_alt_playback, + spec->stream_analog_alt_playback, + spec->alt_dac_nid); + else + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_null_stream, NULL, 0); if (have_multi_adcs) { - p = spec->stream_analog_alt_capture; - if (!p) - p = &pcm_analog_alt_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = - spec->adc_nids[1]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_analog_alt_capture, + spec->stream_analog_alt_capture, + spec->adc_nids[1]); info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids - 1; } else { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - pcm_null_stream; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = 0; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_null_stream, NULL, 0); } } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2a61bda8115d42..124eacf67fc4dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2602,53 +2602,12 @@ static int patch_alc268(struct hda_codec *codec) * ALC269 */ -static int playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 8, .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ - /* NID is set in alc_build_pcms */ - .ops = { - .open = playback_pcm_open, - .prepare = playback_pcm_prepare, - .cleanup = playback_pcm_cleanup - }, }; static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ - /* NID is set in alc_build_pcms */ }; /* different alc269-variants */ From c66150824b8a809a502fd833fa9b18082cd89a39 Mon Sep 17 00:00:00 2001 From: Nikesh Oswal Date: Mon, 2 Feb 2015 17:06:44 +0000 Subject: [PATCH 234/411] ASoC: dapm: add code to configure dai link parameters dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 164 +++++++++++++++++++++++++++++++++++++-- 4 files changed, 166 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8d7416e4686174..eda881402dda41 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade19562857..4636a058372bc8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -941,6 +941,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca5bacb98..700ac2ffe69685 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1245,7 +1245,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1257,7 +1258,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6f88202b8c99a..6828b4ed544799 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -853,6 +853,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* create control for links with > 1 config */ + if (w->num_params <= 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. @@ -2719,6 +2749,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3193,7 +3226,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; - const struct snd_soc_pcm_stream *config = w->params; + const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; @@ -3285,8 +3318,39 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3294,14 +3358,61 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + len = strlen("Anonymous Configuration ") + 3; + w_param_text[count] = + devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + snprintf(w_param_text[count], len, + "Anonymous Configuration %d", count); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, + GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3309,6 +3420,30 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3316,15 +3451,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + devm_kfree(card->dev, w); +outfree_kcontrol_news: + devm_kfree(card->dev, (void *)template.kcontrol_news); +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_link_name: + devm_kfree(card->dev, link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, From b56df151d3b97a95ba3d3a76c2867d00f2db7782 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:26 +0100 Subject: [PATCH 235/411] ALSA: ice1724: ESI W192M: Correct copy/paste from prodigy driver MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Correct copy/paste name from prodigy driver, no behaviour change, only name. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a387b8764..f65ac193cf1a57 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -475,7 +475,7 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); From 7127744a5e14455a4b69429757fd811aa50ec4b8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:27 +0100 Subject: [PATCH 236/411] ALSA: ice1724: ESI W192M: Update eeprom structure to C99 standard MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Update eeprom structure to C99 standard to be compliant with change in alsa. It's just a notation change, no configuration change. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index f65ac193cf1a57..6e1026e0e56bd0 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,19 +485,19 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ + [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, }; From f8a8b3a835ef9e5f94163c9dc62fc2e2e375b10c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:28 +0100 Subject: [PATCH 237/411] ALSA: ice1724: ESI W192M: Enable midi i/o of port envy24 chip as available MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Enable midi i/o port of envy24 chip as their are available on ESI W192M soundcard. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6e1026e0e56bd0..59483b4d6dd0fa 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,7 +485,8 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_SYSCONF] = 0x67, /*SYSCONF: clock 192KHz, mpu401, + 4ADC, 8DAC */ [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ From 16ddbe738a5bd2afe80aa10492f762f34b09cbf0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:29 +0100 Subject: [PATCH 238/411] ALSA: ice1724: ESI W192M: Add TLV support for control value in dB scale MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add TLV support to control volume using dB scale for input and ouput on ESI W192M. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 59483b4d6dd0fa..c7ffafaa1c5cb9 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include #include #include +#include #include "ice1712.h" #include "envy24ht.h" @@ -380,17 +381,25 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/*Limits value in dB for fader*/ +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ static struct snd_kcontrol_new stac9640_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Switch", .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +428,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +448,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; From ae8a9a11256a0906831a7db39b8cbcdec51ae28a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:30 +0100 Subject: [PATCH 239/411] ALSA: ice1724: ESI W192M: Add text Line in/Mic for selecting input gain state MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add text Line in/Mic for selecting input gain state in mixer for ESI W192M. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index c7ffafaa1c5cb9..6d3412f72855af 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,14 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line In", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +360,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +376,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) @@ -411,7 +417,7 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "MIC/Line switch", + .name = "MIC/Line Input Enum", .count = 2, .info = stac9460_mic_sw_info, .get = stac9460_mic_sw_get, From 1aa9a4ea4fea5e4afe8be0229774b8f98db2e6c3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:31 +0100 Subject: [PATCH 240/411] ALSA: ice1724: ESI W192M: Add sampling rate control of the ADC/DAC MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add sampling rate control for ADC/DAC for ESI W192M. Allow to switch between 48K/96K/192K sampling rate. All DAC need to be mute when changing samplerate. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 107 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 107 insertions(+) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6d3412f72855af..9906119e0954cf 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -30,12 +30,18 @@ #include #include #include +#include #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -69,15 +75,65 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (id + 7))) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -90,6 +146,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -388,6 +446,44 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + /*Limits value in dB for fader*/ static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -487,21 +583,32 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } From e6feb5d08509be1af2ebc894dae35f32f7b92ab6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2015 21:32:11 +0100 Subject: [PATCH 241/411] ALSA: hda - Support advanced power state controls This patch enables the finer power state control of each widget depending on the jack plug state and streaming state in addition to the existing power_down_unused power optimization. The new feature is enabled only when codec->power_mgmt flag is set. Two new flags, pin_enabled and stream_enabled, are introduced in nid_path struct for marking the two individual power states: the pin plug/unplug and DAC/ADC stream, respectively. They can be set statically in case they are static routes (e.g. some mixer paths), too. The power up and down events for each pin are triggered via the standard hda_jack table. The call order is hard-coded, relying on the current implementation of jack event chain (a la FILO/stack order). One point to be dealt carefully is that DAC/ADC cannot be powered on/off while streaming. They are pinned as long as the stream is running. For controlling the power of DAC/ADC, a new patch_ops is added. The generic parser provides the default callback for that. As of this patch, only IDT/Sigmatel codec driver enables the flag. The support on other codecs will follow. An assumption we made in this code is that the widget state (e.g. amp, pinctl, connections) remains after the widget power transition (not about FG power transition). This is true for IDT codecs, at least. But if the widget state is lost at widget power transition, we'd need to implement additional code to sync the cached amp/verbs for the specific NID. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 + sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_generic.c | 301 ++++++++++++++++++++++++++++----- sound/pci/hda/hda_generic.h | 5 +- sound/pci/hda/patch_sigmatel.c | 5 + 5 files changed, 269 insertions(+), 48 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3e4fb7a8fdcbb8..7e38d6f7314b7d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1502,6 +1502,8 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, if (!p) return; + if (codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm(codec, nid, true); if (codec->pcm_format_first) update_pcm_format(codec, p, nid, format); update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); @@ -1570,6 +1572,8 @@ static void really_cleanup_stream(struct hda_codec *codec, ); memset(q, 0, sizeof(*q)); q->nid = nid; + if (codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm(codec, nid, false); } /* clean up the all conflicting obsolete streams */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 70851e6d5f109e..148e84ce61cffb 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -200,6 +200,7 @@ struct hda_codec_ops { int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif void (*reboot_notify)(struct hda_codec *codec); + void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on); }; /* record for amp information cache */ @@ -370,6 +371,7 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ + unsigned int power_mgmt:1; /* advanced PM for each widget */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ atomic_t in_pm; /* suspend/resume being performed */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 27ce54701f0f6e..8a5055d296f529 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -140,6 +140,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "single_adc_amp"); if (val >= 0) codec->single_adc_amp = !!val; + val = snd_hda_get_bool_hint(codec, "power_mgmt"); + if (val >= 0) + codec->power_mgmt = !!val; val = snd_hda_get_bool_hint(codec, "auto_mute"); if (val >= 0) @@ -648,12 +651,21 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, unsigned int dir, unsigned int idx) { struct hda_gen_spec *spec = codec->spec; + int type = get_wcaps_type(get_wcaps(codec, nid)); int i, n; for (n = 0; n < spec->paths.used; n++) { struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) continue; + if (codec->power_mgmt) { + if (!path->stream_enabled) + continue; + /* ignore unplugged paths except for DAC/ADC */ + if (!path->pin_enabled && + type != AC_WID_AUD_OUT && type != AC_WID_AUD_IN) + continue; + } for (i = 0; i < path->depth; i++) { if (path->path[i] == nid) { if (dir == HDA_OUTPUT || path->idx[i] == idx) @@ -807,6 +819,42 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } +/* sync power of each widget in the the given path */ +static hda_nid_t path_power_update(struct hda_codec *codec, + struct nid_path *path, + bool allow_powerdown) +{ + hda_nid_t nid, changed = 0; + int i, state; + + for (i = 0; i < path->depth; i++) { + nid = path->path[i]; + if (!allow_powerdown || is_active_nid_for_any(codec, nid)) + state = AC_PWRST_D0; + else + state = AC_PWRST_D3; + if (!snd_hda_check_power_state(codec, nid, state)) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_POWER_STATE, state); + changed = nid; + /* here we assume that widget attributes (e.g. amp, + * pinctl connection) don't change with local power + * state change. If not, need to sync the cache. + */ + } + } + return changed; +} + +/* do sync with the last power state change */ +static void sync_power_state_change(struct hda_codec *codec, hda_nid_t nid) +{ + if (nid) { + msleep(10); + snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); + } +} + /** * snd_hda_activate_path - activate or deactivate the given path * @codec: the HDA codec @@ -825,15 +873,13 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, if (!enable) path->active = false; + /* make sure the widget is powered up */ + if (enable && (spec->power_down_unused || codec->power_mgmt)) + path_power_update(codec, path, codec->power_mgmt); + for (i = path->depth - 1; i >= 0; i--) { hda_nid_t nid = path->path[i]; - if (enable && spec->power_down_unused) { - /* make sure the widget is powered up */ - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D0)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } + if (enable && path->multi[i]) snd_hda_codec_update_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, @@ -853,28 +899,10 @@ EXPORT_SYMBOL_GPL(snd_hda_activate_path); static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - bool changed = false; - int i; - if (!spec->power_down_unused || path->active) + if (!(spec->power_down_unused || codec->power_mgmt) || path->active) return; - - for (i = 0; i < path->depth; i++) { - hda_nid_t nid = path->path[i]; - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3) && - !is_active_nid_for_any(codec, nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - changed = true; - } - } - - if (changed) { - msleep(10); - snd_hda_codec_read(codec, path->path[0], 0, - AC_VERB_GET_POWER_STATE, 0); - } + sync_power_state_change(codec, path_power_update(codec, path, true)); } /* turn on/off EAPD on the given pin */ @@ -1574,6 +1602,7 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) return 0; /* print_nid_path(codec, "output-aamix", path); */ path->active = false; /* unused as default */ + path->pin_enabled = true; /* static route */ return snd_hda_get_path_idx(codec, path); } @@ -2998,6 +3027,7 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, } path->active = true; + path->stream_enabled = true; /* no DAC/ADC involved */ err = add_loopback_list(spec, mix_nid, idx); if (err < 0) return err; @@ -3009,6 +3039,8 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, if (path) { print_nid_path(codec, "loopback-merge", path); path->active = true; + path->pin_enabled = true; /* static route */ + path->stream_enabled = true; /* no DAC/ADC involved */ spec->loopback_merge_path = snd_hda_get_path_idx(codec, path); } @@ -3810,6 +3842,7 @@ static void parse_digital(struct hda_codec *codec) continue; print_nid_path(codec, "digout", path); path->active = true; + path->pin_enabled = true; /* no jack detection */ spec->digout_paths[i] = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_OUT, false); if (!nums) { @@ -3837,6 +3870,7 @@ static void parse_digital(struct hda_codec *codec) if (path) { print_nid_path(codec, "digin", path); path->active = true; + path->pin_enabled = true; /* no jack */ spec->dig_in_nid = dig_nid; spec->digin_path = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_IN, false); @@ -3896,6 +3930,148 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, return 1; } +/* power up/down widgets in the all paths that match with the given NID + * as terminals (either start- or endpoint) + * + * returns the last changed NID, or zero if unchanged. + */ +static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, + int pin_state, int stream_state) +{ + struct hda_gen_spec *spec = codec->spec; + hda_nid_t last, changed = 0; + struct nid_path *path; + int n; + + for (n = 0; n < spec->paths.used; n++) { + path = snd_array_elem(&spec->paths, n); + if (path->path[0] == nid || + path->path[path->depth - 1] == nid) { + bool pin_old = path->pin_enabled; + bool stream_old = path->stream_enabled; + + if (pin_state >= 0) + path->pin_enabled = pin_state; + if (stream_state >= 0) + path->stream_enabled = stream_state; + if (path->pin_enabled != pin_old || + path->stream_enabled != stream_old) { + last = path_power_update(codec, path, true); + if (last) + changed = last; + } + } + } + return changed; +} + +/* power up/down the paths of the given pin according to the jack state; + * power = 0/1 : only power up/down if it matches with the jack state, + * < 0 : force power up/down to follow the jack sate + * + * returns the last changed NID, or zero if unchanged. + */ +static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, + int power) +{ + bool on; + + if (!codec->power_mgmt) + return 0; + + on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; + if (power >= 0 && on != power) + return 0; + return set_path_power(codec, pin, on, -1); +} + +static void pin_power_callback(struct hda_codec *codec, + struct hda_jack_callback *jack, + bool on) +{ + if (jack && jack->tbl->nid) + sync_power_state_change(codec, + set_pin_power_jack(codec, jack->tbl->nid, on)); +} + +/* callback only doing power up -- called at first */ +static void pin_power_up_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + pin_power_callback(codec, jack, true); +} + +/* callback only doing power down -- called at last */ +static void pin_power_down_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + pin_power_callback(codec, jack, false); +} + +/* set up the power up/down callbacks */ +static void add_pin_power_ctls(struct hda_codec *codec, int num_pins, + const hda_nid_t *pins, bool on) +{ + int i; + hda_jack_callback_fn cb = + on ? pin_power_up_callback : pin_power_down_callback; + + for (i = 0; i < num_pins && pins[i]; i++) { + if (is_jack_detectable(codec, pins[i])) + snd_hda_jack_detect_enable_callback(codec, pins[i], cb); + else + set_path_power(codec, pins[i], true, -1); + } +} + +/* enabled power callback to each available I/O pin with jack detections; + * the digital I/O pins are excluded because of the unreliable detectsion + */ +static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + if (!codec->power_mgmt) + return; + add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + add_pin_power_ctls(codec, cfg->hp_outs, cfg->hp_pins, on); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + add_pin_power_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, on); + for (i = 0; i < cfg->num_inputs; i++) + add_pin_power_ctls(codec, 1, &cfg->inputs[i].pin, on); +} + +/* sync path power up/down with the jack states of given pins */ +static void sync_pin_power_ctls(struct hda_codec *codec, int num_pins, + const hda_nid_t *pins) +{ + int i; + + for (i = 0; i < num_pins && pins[i]; i++) + if (is_jack_detectable(codec, pins[i])) + set_pin_power_jack(codec, pins[i], -1); +} + +/* sync path power up/down with pins; called at init and resume */ +static void sync_all_pin_power_ctls(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + if (!codec->power_mgmt) + return; + sync_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + sync_pin_power_ctls(codec, cfg->hp_outs, cfg->hp_pins); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + sync_pin_power_ctls(codec, cfg->speaker_outs, cfg->speaker_pins); + for (i = 0; i < cfg->num_inputs; i++) + sync_pin_power_ctls(codec, 1, &cfg->inputs[i].pin); +} /* * Jack detections for HP auto-mute and mic-switch @@ -3933,6 +4109,10 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, if (!nid) break; + oldval = snd_hda_codec_get_pin_target(codec, nid); + if (oldval & PIN_IN) + continue; /* no mute for inputs */ + if (spec->auto_mute_via_amp) { struct nid_path *path; hda_nid_t mute_nid; @@ -3947,29 +4127,33 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, spec->mute_bits |= (1ULL << mute_nid); else spec->mute_bits &= ~(1ULL << mute_nid); - set_pin_eapd(codec, nid, !mute); continue; + } else { + /* don't reset VREF value in case it's controlling + * the amp (see alc861_fixup_asus_amp_vref_0f()) + */ + if (spec->keep_vref_in_automute) + val = oldval & ~PIN_HP; + else + val = 0; + if (!mute) + val |= oldval; + /* here we call update_pin_ctl() so that the pinctl is + * changed without changing the pinctl target value; + * the original target value will be still referred at + * the init / resume again + */ + update_pin_ctl(codec, nid, val); } - oldval = snd_hda_codec_get_pin_target(codec, nid); - if (oldval & PIN_IN) - continue; /* no mute for inputs */ - /* don't reset VREF value in case it's controlling - * the amp (see alc861_fixup_asus_amp_vref_0f()) - */ - if (spec->keep_vref_in_automute) - val = oldval & ~PIN_HP; - else - val = 0; - if (!mute) - val |= oldval; - /* here we call update_pin_ctl() so that the pinctl is changed - * without changing the pinctl target value; - * the original target value will be still referred at the - * init / resume again - */ - update_pin_ctl(codec, nid, val); set_pin_eapd(codec, nid, !mute); + if (codec->power_mgmt) { + bool on = !mute; + if (on) + on = snd_hda_jack_detect_state(codec, nid) + != HDA_JACK_NOT_PRESENT; + set_path_power(codec, nid, on, -1); + } } } @@ -4465,6 +4649,21 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) } } +/** + * snd_hda_gen_stream_pm - Stream power management callback + * @codec: the HDA codec + * @nid: audio widget + * @on: power on/off flag + * + * Set this in patch_ops.stream_pm. Only valid with power_mgmt flag. + */ +void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on) +{ + if (codec->power_mgmt) + set_path_power(codec, nid, -1, on); +} +EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm); + /** * snd_hda_gen_parse_auto_config - Parse the given BIOS configuration and * set up the hda_gen_spec @@ -4549,6 +4748,9 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; + /* add power-down pin callbacks at first */ + add_all_pin_power_ctls(codec, false); + spec->const_channel_count = spec->ext_channel_count; /* check the multiple speaker and headphone pins */ if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) @@ -4618,6 +4820,9 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, } } + /* add power-up pin callbacks at last */ + add_all_pin_power_ctls(codec, true); + /* mute all aamix input initially */ if (spec->mixer_nid) mute_all_mixer_nid(codec, spec->mixer_nid); @@ -4625,7 +4830,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused) + if (spec->power_down_unused || codec->power_mgmt) codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { @@ -5478,6 +5683,8 @@ int snd_hda_gen_init(struct hda_codec *codec) clear_unsol_on_unused_pins(codec); + sync_all_pin_power_ctls(codec); + /* call init functions of standard auto-mute helpers */ update_automute_all(codec); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index b211f889b335dd..54659b51fe16f7 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -46,7 +46,9 @@ struct nid_path { unsigned char idx[MAX_NID_PATH_DEPTH]; unsigned char multi[MAX_NID_PATH_DEPTH]; unsigned int ctls[NID_PATH_NUM_CTLS]; /* NID_PATH_XXX_CTL */ - bool active; + bool active:1; /* activated by driver */ + bool pin_enabled:1; /* pins are enabled */ + bool stream_enabled:1; /* stream is active */ }; /* mic/line-in auto switching entry */ @@ -340,5 +342,6 @@ int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid); unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state); +void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2956a6ba6bf018..86b944a6b0ed8e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4394,6 +4394,7 @@ static const struct hda_codec_ops stac_patch_ops = { #ifdef CONFIG_PM .suspend = stac_suspend, #endif + .stream_pm = snd_hda_gen_stream_pm, .reboot_notify = stac_shutup, }; @@ -4487,6 +4488,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; + codec->power_mgmt = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; @@ -4592,6 +4594,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; + codec->power_mgmt = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4641,6 +4644,7 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; + codec->power_mgmt = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4682,6 +4686,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; + codec->power_mgmt = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; From 688b12cc3ca8a5155b95ce8d01e0e43006813b27 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Mar 2015 15:56:05 +0100 Subject: [PATCH 242/411] ALSA: hda - Use the new power control for VIA codecs VIA codecs used to have the own power controls but they were disabled at transition to the generic parser due to the coding assuming the fixed routes. Now we get the proper support of equivalently fine power management in the generic parser, and the old kludges can be replaced with it. This results in the reduction of lots of dead codes. The advanced PM feature is disabled as default like before for keeping the compatible behavior. It's enabled via "Dynamic Power-Control" mixer element. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 662 +------------------------------------- 1 file changed, 12 insertions(+), 650 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 2112fbe9e57702..d5d1dca4f11b34 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -99,7 +99,6 @@ struct via_spec { /* HP mode source */ unsigned int dmic_enabled; - unsigned int no_pin_power_ctl; enum VIA_HDA_CODEC codec_type; /* analog low-power control */ @@ -108,9 +107,6 @@ struct via_spec { /* work to check hp jack state */ int hp_work_active; int vt1708_jack_detect; - - void (*set_widgets_power_state)(struct hda_codec *codec); - unsigned int dac_stream_tag[4]; }; static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); @@ -133,11 +129,12 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) /* VT1708BCE & VT1708S are almost same */ if (spec->codec_type == VT1708BCE) spec->codec_type = VT1708S; - spec->no_pin_power_ctl = 1; spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; + codec->power_mgmt = 1; + spec->gen.power_down_unused = 1; return spec; } @@ -229,90 +226,6 @@ static void vt1708_update_hp_work(struct hda_codec *codec) vt1708_stop_hp_work(codec); } -static void set_widgets_power_state(struct hda_codec *codec) -{ -#if 0 /* FIXME: the assumed connections don't match always with the - * actual routes by the generic parser, so better to disable - * the control for safety. - */ - struct via_spec *spec = codec->spec; - if (spec->set_widgets_power_state) - spec->set_widgets_power_state(codec); -#endif -} - -static void update_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int parm) -{ - if (snd_hda_check_power_state(codec, nid, parm)) - return; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); -} - -static void update_conv_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int parm, unsigned int index) -{ - struct via_spec *spec = codec->spec; - unsigned int format; - - if (snd_hda_check_power_state(codec, nid, parm)) - return; - format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (format && (spec->dac_stream_tag[index] != format)) - spec->dac_stream_tag[index] = format; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); - if (parm == AC_PWRST_D0) { - format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (!format && (spec->dac_stream_tag[index] != format)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - spec->dac_stream_tag[index]); - } -} - -static bool smart51_enabled(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - return spec->gen.ext_channel_count > 2; -} - -static bool is_smart51_pins(struct hda_codec *codec, hda_nid_t pin) -{ - struct via_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->gen.multi_ios; i++) - if (spec->gen.multi_io[i].pin == pin) - return true; - return false; -} - -static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int *affected_parm) -{ - unsigned parm; - unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); - unsigned no_presence = (def_conf & AC_DEFCFG_MISC) - >> AC_DEFCFG_MISC_SHIFT - & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ - struct via_spec *spec = codec->spec; - unsigned present = 0; - - no_presence |= spec->no_pin_power_ctl; - if (!no_presence) - present = snd_hda_jack_detect(codec, nid); - if ((smart51_enabled(codec) && is_smart51_pins(codec, nid)) - || ((no_presence || present) - && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { - *affected_parm = AC_PWRST_D0; /* if it's connected */ - parm = AC_PWRST_D0; - } else - parm = AC_PWRST_D3; - - update_power_state(codec, nid, parm); -} - static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -323,8 +236,7 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct via_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = !spec->no_pin_power_ctl; + ucontrol->value.enumerated.item[0] = codec->power_mgmt; return 0; } @@ -333,12 +245,12 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - unsigned int val = !ucontrol->value.enumerated.item[0]; + bool val = !!ucontrol->value.enumerated.item[0]; - if (val == spec->no_pin_power_ctl) + if (val == codec->power_mgmt) return 0; - spec->no_pin_power_ctl = val; - set_widgets_power_state(codec); + codec->power_mgmt = val; + spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; } @@ -383,7 +295,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) bool enable; unsigned int verb, parm; - if (spec->no_pin_power_ctl) + if (!codec->power_mgmt) enable = false; else enable = is_aa_path_mute(codec) && !spec->gen.active_streams; @@ -440,8 +352,7 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; - if (spec->set_widgets_power_state) - spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; + spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -485,7 +396,6 @@ static int via_suspend(struct hda_codec *codec) static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct via_spec *spec = codec->spec; - set_widgets_power_state(codec); analog_low_current_mode(codec); vt1708_update_hp_work(codec); return snd_hda_check_amp_list_power(codec, &spec->gen.loopback, nid); @@ -573,34 +483,6 @@ static const struct snd_kcontrol_new vt1708_jack_detect_ctl[] = { {} /* terminator */ }; -static void via_jack_powerstate_event(struct hda_codec *codec, - struct hda_jack_callback *tbl) -{ - set_widgets_power_state(codec); -} - -static void via_set_jack_unsol_events(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->gen.autocfg; - hda_nid_t pin; - int i; - - for (i = 0; i < cfg->line_outs; i++) { - pin = cfg->line_out_pins[i]; - if (pin && is_jack_detectable(codec, pin)) - snd_hda_jack_detect_enable_callback(codec, pin, - via_jack_powerstate_event); - } - - for (i = 0; i < cfg->num_inputs; i++) { - pin = cfg->line_out_pins[i]; - if (pin && is_jack_detectable(codec, pin)) - snd_hda_jack_detect_enable_callback(codec, pin, - via_jack_powerstate_event); - } -} - static const struct badness_table via_main_out_badness = { .no_primary_dac = 0x10000, .no_dac = 0x4000, @@ -634,7 +516,9 @@ static int via_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - via_set_jack_unsol_events(codec); + /* disable widget PM at start for compatibility */ + codec->power_mgmt = 0; + spec->gen.power_down_unused = 0; return 0; } @@ -647,7 +531,6 @@ static int via_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->init_verbs[i]); /* init power states */ - set_widgets_power_state(codec); __analog_low_current_mode(codec, true); snd_hda_gen_init(codec); @@ -767,78 +650,6 @@ static int patch_vt1709(struct hda_codec *codec) return 0; } -static void set_widgets_power_state_vt1708B(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - int is_8ch = 0; - if ((spec->codec_type != VT1708B_4CH) && - (codec->vendor_id != 0x11064397)) - is_8ch = 1; - - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = - (snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) - == ((spec->codec_type == VT1708S) ? 5 : 0)); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW 0/1 (13h/14h) */ - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x13, parm); - update_power_state(codec, 0x14, parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1b, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x11, parm); - - /* PW6 (22h), SW2 (26h), AOW2 (24h) */ - if (is_8ch) { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x22, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x26, parm); - update_power_state(codec, 0x24, parm); - } else if (codec->vendor_id == 0x11064397) { - /* PW7(23h), SW2(27h), AOW2(25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x27, parm); - update_power_state(codec, 0x25, parm); - } - - /* PW 3/4/7 (1ch/1dh/23h) */ - parm = AC_PWRST_D3; - /* force to D0 for internal Speaker */ - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - if (is_8ch) - set_pin_power_state(codec, 0x23, &parm); - - /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, parm); - if (is_8ch) { - update_power_state(codec, 0x25, parm); - update_power_state(codec, 0x27, parm); - } else if (codec->vendor_id == 0x11064397 && spec->gen.indep_hp_enabled) - update_power_state(codec, 0x25, parm); -} - static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B(struct hda_codec *codec) { @@ -863,9 +674,6 @@ static int patch_vt1708B(struct hda_codec *codec) } codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1708B; - return 0; } @@ -931,8 +739,6 @@ static int patch_vt1708S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -946,36 +752,6 @@ static const struct hda_verb vt1702_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1702(struct hda_codec *codec) -{ - int imux_is_smixer = - snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - unsigned int parm; - /* inputs */ - /* PW 1/2/5 (14h/15h/18h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x14, &parm); - set_pin_power_state(codec, 0x15, &parm); - set_pin_power_state(codec, 0x18, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ - /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - update_power_state(codec, 0x13, parm); - update_power_state(codec, 0x12, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x20, parm); - - /* outputs */ - /* PW 3/4 (16h/17h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x17, &parm); - set_pin_power_state(codec, 0x16, &parm); - /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x1d, parm); -} - static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; @@ -1005,8 +781,6 @@ static int patch_vt1702(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1702; return 0; } @@ -1021,71 +795,6 @@ static const struct hda_verb vt1718S_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1718S(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm, parm2; - /* MUX6 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - update_power_state(codec, 0x1a, parm); - parm2 = parm; /* for pin 0x0b */ - - /* PW2 (26h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x2b, &parm); - update_power_state(codec, 0xa, parm); - - /* PW0 (24h), AOW0 (8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (!spec->gen.indep_hp_enabled) /* check for redirected HP */ - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x8, parm); - if (!spec->gen.indep_hp_enabled && parm2 != AC_PWRST_D3) - parm = parm2; - update_power_state(codec, 0xb, parm); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm); - - /* PW1 (25h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x2a, &parm); - update_power_state(codec, 0x9, parm); - - if (spec->gen.indep_hp_enabled) { - /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x1b, parm); - update_power_state(codec, 0x34, parm); - update_power_state(codec, 0xc, parm); - } -} - /* Add a connection to the primary DAC from AA-mixer for some codecs * This isn't listed from the raw info, but the chip has a secret connection. */ @@ -1146,9 +855,6 @@ static int patch_vt1718S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1718S; - return 0; } @@ -1188,7 +894,6 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_CONNECT_SEL, index); spec->dmic_enabled = index; - set_widgets_power_state(codec); return 1; } @@ -1223,95 +928,6 @@ static const struct hda_verb vt1716S_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1716S(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - unsigned int mono_out, present; - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = - (snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_CONNECT_SEL, 0x00) == 5); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW0(13h) */ - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x13, parm); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1e, &parm); - /* PW11 (22h) */ - if (spec->dmic_enabled) - set_pin_power_state(codec, 0x22, &parm); - else - update_power_state(codec, 0x22, AC_PWRST_D3); - - /* SW2(26h), AIW1(14h) */ - update_power_state(codec, 0x26, parm); - update_power_state(codec, 0x14, parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - /* Smart 5.1 PW2(1bh) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1b, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x11, parm); - - /* PW7 (23h), SW3 (27h), AOW3 (25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - /* Smart 5.1 PW1(1ah) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x27, parm); - - /* Smart 5.1 PW5(1eh) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1e, &parm); - update_power_state(codec, 0x25, parm); - - /* Mono out */ - /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_jack_detect(codec, 0x1c); - - if (present) - mono_out = 0; - else { - present = snd_hda_jack_detect(codec, 0x1d); - if (!spec->gen.indep_hp_enabled && present) - mono_out = 0; - else - mono_out = 1; - } - parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - update_power_state(codec, 0x28, parm); - update_power_state(codec, 0x29, parm); - update_power_state(codec, 0x2a, parm); - - /* PW 3/4 (1ch/1dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - /* HP Independent Mode, power on AOW3 */ - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x25, parm); - - /* force to D0 for internal Speaker */ - /* MW0 (16h), AOW0 (10h) */ - update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm); -} - static int patch_vt1716S(struct hda_codec *codec) { struct via_spec *spec; @@ -1339,8 +955,6 @@ static int patch_vt1716S(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1716S; return 0; } @@ -1366,98 +980,6 @@ static const struct hda_verb vt1802_init_verbs[] = { { } }; -static void set_widgets_power_state_vt2002P(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - unsigned int present; - /* MUX9 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - parm = AC_PWRST_D0; - /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* AOW0 (8h)*/ - update_power_state(codec, 0x8, parm); - - if (spec->codec_type == VT1802) { - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x38, parm); - } else { - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - update_power_state(codec, 0x1c, parm); - update_power_state(codec, 0x37, parm); - } - - if (spec->codec_type == VT1802) { - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x15, parm); - update_power_state(codec, 0x35, parm); - } else { - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x19, parm); - update_power_state(codec, 0x35, parm); - } - - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x9, AC_PWRST_D0); - - /* Class-D */ - /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - if (spec->codec_type == VT1802) - update_power_state(codec, 0x14, parm); - else - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x34, parm); - - /* Mono Out */ - present = snd_hda_jack_detect(codec, 0x26); - - parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - if (spec->codec_type == VT1802) { - /* PW15 (33h), MW8(1ch), MUX8(3ch) */ - update_power_state(codec, 0x33, parm); - update_power_state(codec, 0x1c, parm); - update_power_state(codec, 0x3c, parm); - } else { - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - update_power_state(codec, 0x31, parm); - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x3b, parm); - } - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - update_power_state(codec, 0x21, AC_PWRST_D0); - else - update_power_state(codec, 0x21, AC_PWRST_D3); -} - /* * pin fix-up */ @@ -1541,8 +1063,6 @@ static int patch_vt2002P(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt2002P; return 0; } @@ -1556,81 +1076,6 @@ static const struct hda_verb vt1812_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1812(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - unsigned int parm; - unsigned int present; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* AOW0 (8h)*/ - update_power_state(codec, 0x8, AC_PWRST_D0); - - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x38, parm); - - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x15, parm); - update_power_state(codec, 0x35, parm); - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x9, AC_PWRST_D0); - - /* Internal Speaker */ - /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - update_power_state(codec, 0x14, AC_PWRST_D3); - update_power_state(codec, 0x34, AC_PWRST_D3); - } else { - update_power_state(codec, 0x14, AC_PWRST_D0); - update_power_state(codec, 0x34, AC_PWRST_D0); - } - - - /* Mono Out */ - /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_jack_detect(codec, 0x28); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - update_power_state(codec, 0x1c, AC_PWRST_D3); - update_power_state(codec, 0x3c, AC_PWRST_D3); - update_power_state(codec, 0x3e, AC_PWRST_D3); - } else { - update_power_state(codec, 0x1c, AC_PWRST_D0); - update_power_state(codec, 0x3c, AC_PWRST_D0); - update_power_state(codec, 0x3e, AC_PWRST_D0); - } - - /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x33, &parm); - update_power_state(codec, 0x1d, parm); - update_power_state(codec, 0x3d, parm); - -} - /* patch for vt1812 */ static int patch_vt1812(struct hda_codec *codec) { @@ -1657,8 +1102,6 @@ static int patch_vt1812(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1812; return 0; } @@ -1674,84 +1117,6 @@ static const struct hda_verb vt3476_init_verbs[] = { { } }; -static void set_widgets_power_state_vt3476(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm, parm2; - /* MUX10 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 4; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* PW3 (27h), MW3(37h), AOW3 (bh) */ - if (spec->codec_type == VT1705CF) { - parm = AC_PWRST_D3; - update_power_state(codec, 0x27, parm); - update_power_state(codec, 0x37, parm); - } else { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - update_power_state(codec, 0x37, parm); - } - - /* PW2 (26h), MW2(36h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - update_power_state(codec, 0x36, parm); - if (smart51_enabled(codec)) { - /* PW7(2bh), MW7(3bh), MUX7(1Bh) */ - set_pin_power_state(codec, 0x2b, &parm); - update_power_state(codec, 0x3b, parm); - update_power_state(codec, 0x1b, parm); - } - update_conv_power_state(codec, 0xa, parm, 2); - - /* PW1 (25h), MW1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x35, parm); - if (smart51_enabled(codec)) { - /* PW6(2ah), MW6(3ah), MUX6(1ah) */ - set_pin_power_state(codec, 0x2a, &parm); - update_power_state(codec, 0x3a, parm); - update_power_state(codec, 0x1a, parm); - } - update_conv_power_state(codec, 0x9, parm, 1); - - /* PW4 (28h), MW4 (38h), MUX4(18h), AOW3(bh)/AOW0(8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x38, parm); - update_power_state(codec, 0x18, parm); - if (spec->gen.indep_hp_enabled) - update_conv_power_state(codec, 0xb, parm, 3); - parm2 = parm; /* for pin 0x0b */ - - /* PW0 (24h), MW0(34h), MW9(3fh), AOW0 (8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - update_power_state(codec, 0x34, parm); - if (!spec->gen.indep_hp_enabled && parm2 != AC_PWRST_D3) - parm = parm2; - update_conv_power_state(codec, 0x8, parm, 0); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - update_power_state(codec, 0x3f, imux_is_smixer ? AC_PWRST_D0 : parm); -} - static int patch_vt3476(struct hda_codec *codec) { struct via_spec *spec; @@ -1775,9 +1140,6 @@ static int patch_vt3476(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt3476; - return 0; } From 5ccf835cc76d89bc0d426659c63d81f609050842 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Mar 2015 09:23:10 +0100 Subject: [PATCH 243/411] ALSA: hda - Adjust power of beep widget and outputs As the widget PM may turn off the pins, this might lead to the silent output for beep when no explicit paths are given. This patch adds fake output paths for the beep widget so that the output pins are dynamically powered upon beep on/off. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 29 +++++++++------- sound/pci/hda/hda_beep.h | 1 + sound/pci/hda/hda_generic.c | 69 +++++++++++++++++++++++++++++++++++++ 3 files changed, 86 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 581b7fdef0e379..4cdac3a71caea0 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -33,30 +33,36 @@ enum { DIGBEEP_HZ_MAX = 12000000, /* 12 KHz */ }; -static void snd_hda_generate_beep(struct work_struct *work) +/* generate or stop tone */ +static void generate_tone(struct hda_beep *beep, int tone) { - struct hda_beep *beep = - container_of(work, struct hda_beep, beep_work); struct hda_codec *codec = beep->codec; - int tone; - if (!beep->enabled) - return; - - tone = beep->tone; if (tone && !beep->playing) { snd_hda_power_up(codec); + if (beep->power_hook) + beep->power_hook(beep, true); beep->playing = 1; } - /* generate tone */ snd_hda_codec_write(codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, tone); if (!tone && beep->playing) { beep->playing = 0; + if (beep->power_hook) + beep->power_hook(beep, false); snd_hda_power_down(codec); } } +static void snd_hda_generate_beep(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, beep_work); + + if (beep->enabled) + generate_tone(beep, beep->tone); +} + /* (non-standard) Linear beep tone calculation for IDT/STAC codecs * * The tone frequency of beep generator on IDT/STAC codecs is @@ -130,10 +136,7 @@ static void turn_off_beep(struct hda_beep *beep) cancel_work_sync(&beep->beep_work); if (beep->playing) { /* turn off beep */ - snd_hda_codec_write(beep->codec, beep->nid, 0, - AC_VERB_SET_BEEP_CONTROL, 0); - beep->playing = 0; - snd_hda_power_down(beep->codec); + generate_tone(beep, 0); } } diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index a63b5e077332a9..46524ff7e79e68 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -40,6 +40,7 @@ struct hda_beep { unsigned int playing:1; struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; + void (*power_hook)(struct hda_beep *beep, bool on); }; #ifdef CONFIG_SND_HDA_INPUT_BEEP diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8a5055d296f529..d7ca388651daaf 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -654,6 +654,9 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, int type = get_wcaps_type(get_wcaps(codec, nid)); int i, n; + if (nid == codec->afg) + return true; + for (n = 0; n < spec->paths.used; n++) { struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) @@ -829,6 +832,8 @@ static hda_nid_t path_power_update(struct hda_codec *codec, for (i = 0; i < path->depth; i++) { nid = path->path[i]; + if (nid == codec->afg) + continue; if (!allow_powerdown || is_active_nid_for_any(codec, nid)) state = AC_PWRST_D0; else @@ -4073,6 +4078,64 @@ static void sync_all_pin_power_ctls(struct hda_codec *codec) sync_pin_power_ctls(codec, 1, &cfg->inputs[i].pin); } +/* add fake paths if not present yet */ +static int add_fake_paths(struct hda_codec *codec, hda_nid_t nid, + int num_pins, const hda_nid_t *pins) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + int i; + + for (i = 0; i < num_pins; i++) { + if (!pins[i]) + break; + if (get_nid_path(codec, nid, pins[i], 0)) + continue; + path = snd_array_new(&spec->paths); + if (!path) + return -ENOMEM; + memset(path, 0, sizeof(*path)); + path->depth = 2; + path->path[0] = nid; + path->path[1] = pins[i]; + path->active = true; + } + return 0; +} + +/* create fake paths to all outputs from beep */ +static int add_fake_beep_paths(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid = spec->beep_nid; + int err; + + if (!codec->power_mgmt || !nid) + return 0; + err = add_fake_paths(codec, nid, cfg->line_outs, cfg->line_out_pins); + if (err < 0) + return err; + if (cfg->line_out_type != AUTO_PIN_HP_OUT) { + err = add_fake_paths(codec, nid, cfg->hp_outs, cfg->hp_pins); + if (err < 0) + return err; + } + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + err = add_fake_paths(codec, nid, cfg->speaker_outs, + cfg->speaker_pins); + if (err < 0) + return err; + } + return 0; +} + +/* power up/down beep widget and its output paths */ +static void beep_power_hook(struct hda_beep *beep, bool on) +{ + set_path_power(beep->codec, beep->nid, -1, on); +} + /* * Jack detections for HP auto-mute and mic-switch */ @@ -4837,6 +4900,12 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, err = snd_hda_attach_beep_device(codec, spec->beep_nid); if (err < 0) return err; + if (codec->beep && codec->power_mgmt) { + err = add_fake_beep_paths(codec); + if (err < 0) + return err; + codec->beep->power_hook = beep_power_hook; + } } return 1; From 06ac0cd1c4e4e51fa84f866ef69b518488ffa05f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 18 Mar 2015 10:07:19 +0200 Subject: [PATCH 244/411] ASoC: Intel: Remove support for Intel MID DMA from firmware loader Intel MID DMA driver is going to be removed by the coming commit 36111da7838e ("dmaengine: intel-mid-dma: remove the driver") in spi.git tree. Since there are no users for SST_DMA_TYPE_MID type the support for it can be removed from here in advance. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 1 - sound/soc/intel/sst-firmware.c | 3 --- 2 files changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32f00777a..3412474083ffdf 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -28,7 +28,6 @@ /* Supported SST DMA Devices */ #define SST_DMA_TYPE_DW 1 -#define SST_DMA_TYPE_MID 2 /* autosuspend delay 5s*/ #define SST_RUNTIME_SUSPEND_DELAY (5 * 1000) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5e5800897da220..38881f1fb990b2 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -277,9 +277,6 @@ int sst_dma_new(struct sst_dsp *sst) case SST_DMA_TYPE_DW: dma_dev_name = "dw_dmac"; break; - case SST_DMA_TYPE_MID: - dma_dev_name = "Intel MID DMA"; - break; default: dev_err(sst->dev, "error: invalid DMA engine %d\n", sst->pdata->dma_engine); From 91b0d9aa933a2335f6f11983b19eaf9ebe3c2033 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 18 Mar 2015 10:07:20 +0200 Subject: [PATCH 245/411] ASoC: Intel: Remove vague commit about slave DMA config from firmware loader Intel MID DMA driver is going to be removed, commit should be a few lines down near to dmaengine_slave_config() call in order to not confuse and at quick look Synopsys DesignWare does seem to use some of the slave config structure fields (see drivers/dma/dw/core.c: dwc_config()). Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 38881f1fb990b2..b5659ecb80de6f 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -221,8 +221,6 @@ int sst_dsp_dma_get_channel(struct sst_dsp *dsp, int chan_id) dma_cap_mask_t mask; int ret; - /* The Intel MID DMA engine driver needs the slave config set but - * Synopsis DMA engine driver safely ignores the slave config */ dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); dma_cap_set(DMA_MEMCPY, mask); From 5d5b275d727753372f0a390b4121738d073f3e94 Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Thu, 19 Mar 2015 08:38:00 +0800 Subject: [PATCH 246/411] Intel: ASoC: Add condition check before set param to waves Check waves state before set parameter through ipc to prevent unexpected operation. Also remove redundant check. Signed-off-by: Lu, Han Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 5 +++++ sound/soc/intel/sst-haswell-pcm.c | 6 +----- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 43fb5f33916836..20b629a011de60 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -2055,6 +2055,11 @@ int sst_hsw_launch_param_buf(struct sst_hsw *hsw) { int ret, idx; + if (!sst_hsw_is_module_active(hsw, SST_HSW_MODULE_WAVES)) { + dev_dbg(hsw->dev, "module waves is not active\n"); + return 0; + } + /* put all param lines to DSP through ipc */ for (idx = 0; idx < hsw->param_idx_w; idx++) { ret = sst_hsw_module_set_param(hsw, diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index b40ec746bc19f5..6c6229ae4a0259 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -399,13 +399,9 @@ static int hsw_waves_param_put(struct snd_kcontrol *kcontrol, if (ret < 0) return ret; - if (sst_hsw_is_module_loaded(hsw, id)) { - if (!sst_hsw_is_module_active(hsw, id)) - return 0; - + if (sst_hsw_is_module_active(hsw, id)) ret = sst_hsw_module_set_param(hsw, id, 0, param_id, param_size, ucontrol->value.bytes.data); - } return ret; } From bdc455b512c880292d7f145e73aed5e37a90f6e4 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 18 Mar 2015 21:31:38 +0530 Subject: [PATCH 247/411] ASoC: Intel: acpi_probe: fix error return path Fix the sst_acpi_probe memory allocation error path by setting right error code and initiating the cleanup insteadof just returning Reported-by: Dan Carpenter Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index b3360139c41a90..98c2444dece33f 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -309,7 +309,7 @@ int sst_acpi_probe(struct platform_device *pdev) ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), GFP_KERNEL); if (!ctx->shim_regs64) { - return -ENOMEM; + ret = -ENOMEM; goto do_sst_cleanup; } From b24062bda7baba62781c2a67d126126ce0bc8899 Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Wed, 18 Mar 2015 17:48:56 +0100 Subject: [PATCH 248/411] ALSA: aoa: constify of_device_id array of_device_id is always used as const. (See driver.of_match_table and open firmware functions) Signed-off-by: Fabian Frederick Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index b9737fae656a89..1cbf210080a16c 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -31,7 +31,7 @@ module_param(force, int, 0444); MODULE_PARM_DESC(force, "Force loading i2sbus even when" " no layout-id property is present"); -static struct of_device_id i2sbus_match[] = { +static const struct of_device_id i2sbus_match[] = { { .name = "i2s" }, { } }; From 9cd974bb0f7abc83da1788a17e28890819b593b8 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 20 Mar 2015 12:35:45 +0100 Subject: [PATCH 249/411] ASoC: max98925: Constify regmap config and other codec data Constify local structures (snd_soc_dai_ops, snd_soc_codec_driver, regmap_config) and array (reg_defaults) which are not modified by the driver and passed to core as pointer to const. Signed-off-by: Krzysztof Kozlowski Signed-off-by: Mark Brown --- sound/soc/codecs/max98925.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 1f8e8031574982..0796dfaeb1394f 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -34,7 +34,7 @@ static const char *const hpf_text[] = { "Disable", "DC Block", "100Hz", "200Hz", "400Hz", "800Hz", }; -static struct reg_default max98925_reg[] = { +static const struct reg_default max98925_reg[] = { { 0x0B, 0x00 }, /* IRQ Enable0 */ { 0x0C, 0x00 }, /* IRQ Enable1 */ { 0x0D, 0x00 }, /* IRQ Enable2 */ @@ -491,7 +491,7 @@ static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, #define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops max98925_dai_ops = { +static const struct snd_soc_dai_ops max98925_dai_ops = { .set_sysclk = max98925_dai_set_sysclk, .set_fmt = max98925_dai_set_fmt, .hw_params = max98925_dai_hw_params, @@ -541,7 +541,7 @@ static int max98925_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_dev_max98925 = { +static const struct snd_soc_codec_driver soc_codec_dev_max98925 = { .probe = max98925_probe, .controls = max98925_snd_controls, .num_controls = ARRAY_SIZE(max98925_snd_controls), @@ -551,7 +551,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98925 = { .num_dapm_widgets = ARRAY_SIZE(max98925_dapm_widgets), }; -static struct regmap_config max98925_regmap = { +static const struct regmap_config max98925_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = MAX98925_REV_VERSION, From 6b275b140094b701f7ad15272f0597e9d954e5e4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Mar 2015 18:11:05 +0100 Subject: [PATCH 250/411] ALSA: hda - Fix power of pins used for mute LED with vrefs Some pins are used for controlling the LED with the VREF value. This patch changes the power behavior of such pins to be constantly up. A new state, pin_fixed, is introduced to nid_path to indicate that the path contains the fixed pin. This improves also the readability a bit for other static routes, too. Then a helper function snd_hda_gen_fix_pin_power() is called from the codec driver for such fixed pins, and it will create fake paths containing only these pins with pin_fixed=1 flag. Reported-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 37 +++++++++++++++++++++++++++------- sound/pci/hda/hda_generic.h | 2 ++ sound/pci/hda/patch_sigmatel.c | 6 ++++++ 3 files changed, 38 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index d7ca388651daaf..1cafcbb9d391b8 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -665,7 +665,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, if (!path->stream_enabled) continue; /* ignore unplugged paths except for DAC/ADC */ - if (!path->pin_enabled && + if (!(path->pin_enabled || path->pin_fixed) && type != AC_WID_AUD_OUT && type != AC_WID_AUD_IN) continue; } @@ -1607,7 +1607,7 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) return 0; /* print_nid_path(codec, "output-aamix", path); */ path->active = false; /* unused as default */ - path->pin_enabled = true; /* static route */ + path->pin_fixed = true; /* static route */ return snd_hda_get_path_idx(codec, path); } @@ -3044,7 +3044,7 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, if (path) { print_nid_path(codec, "loopback-merge", path); path->active = true; - path->pin_enabled = true; /* static route */ + path->pin_fixed = true; /* static route */ path->stream_enabled = true; /* no DAC/ADC involved */ spec->loopback_merge_path = snd_hda_get_path_idx(codec, path); @@ -3847,7 +3847,7 @@ static void parse_digital(struct hda_codec *codec) continue; print_nid_path(codec, "digout", path); path->active = true; - path->pin_enabled = true; /* no jack detection */ + path->pin_fixed = true; /* no jack detection */ spec->digout_paths[i] = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_OUT, false); if (!nums) { @@ -3875,7 +3875,7 @@ static void parse_digital(struct hda_codec *codec) if (path) { print_nid_path(codec, "digin", path); path->active = true; - path->pin_enabled = true; /* no jack */ + path->pin_fixed = true; /* no jack */ spec->dig_in_nid = dig_nid; spec->digin_path = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_IN, false); @@ -3959,8 +3959,8 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, path->pin_enabled = pin_state; if (stream_state >= 0) path->stream_enabled = stream_state; - if (path->pin_enabled != pin_old || - path->stream_enabled != stream_old) { + if ((!path->pin_fixed && path->pin_enabled != pin_old) + || path->stream_enabled != stream_old) { last = path_power_update(codec, path, true); if (last) changed = last; @@ -4136,6 +4136,29 @@ static void beep_power_hook(struct hda_beep *beep, bool on) set_path_power(beep->codec, beep->nid, -1, on); } +/** + * snd_hda_gen_fix_pin_power - Fix the power of the given pin widget to D0 + * @codec: the HDA codec + * @pin: NID of pin to fix + */ +int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + + path = snd_array_new(&spec->paths); + if (!path) + return -ENOMEM; + memset(path, 0, sizeof(*path)); + path->depth = 1; + path->path[0] = pin; + path->active = true; + path->pin_fixed = true; + path->stream_enabled = true; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_gen_fix_pin_power); + /* * Jack detections for HP auto-mute and mic-switch */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 54659b51fe16f7..56e4139b90320c 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -48,6 +48,7 @@ struct nid_path { unsigned int ctls[NID_PATH_NUM_CTLS]; /* NID_PATH_XXX_CTL */ bool active:1; /* activated by driver */ bool pin_enabled:1; /* pins are enabled */ + bool pin_fixed:1; /* path with fixed pin */ bool stream_enabled:1; /* stream is active */ }; @@ -343,5 +344,6 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state); void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); +int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 86b944a6b0ed8e..7e531d5cde51ae 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4225,6 +4225,12 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + if (spec->vref_mute_led_nid) { + err = snd_hda_gen_fix_pin_power(codec, spec->vref_mute_led_nid); + if (err < 0) + return err; + } + /* setup analog beep controls */ if (spec->anabeep_nid > 0) { err = stac_auto_create_beep_ctls(codec, From 967b1307b69b8ada8b331e01046ad1ef83742e99 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Mar 2015 18:21:03 +0100 Subject: [PATCH 251/411] ALSA: hda - Rename power_mgmt flag with power_save_node David suggested that the name "power_mgmt" is too ambiguous. Rename the flag with a bit clearer one "power_save_node". Also, add the corresponding description to HD-Audio.txt, too. Reported-by: David Henningsson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 6 +++++- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_generic.c | 30 +++++++++++++-------------- sound/pci/hda/patch_sigmatel.c | 8 +++---- sound/pci/hda/patch_via.c | 12 +++++------ 5 files changed, 31 insertions(+), 27 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39b77e6a5..e7193aac669c3a 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -466,7 +466,11 @@ The generic parser supports the following hints: - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities -- power_down_unused (bool): power down the unused widgets +- power_save_node (bool): advanced power management for each widget, + controlling the power sate (D0/D3) of each widget node depending on + the actual pin and stream states +- power_down_unused (bool): power down the unused widgets, a subset of + power_save_node, and will be dropped in future - add_hp_mic (bool): add the headphone to capture source if possible - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 148e84ce61cffb..ccf355d4a8fa09 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -371,7 +371,7 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ - unsigned int power_mgmt:1; /* advanced PM for each widget */ + unsigned int power_save_node:1; /* advanced PM for each widget */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ atomic_t in_pm; /* suspend/resume being performed */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1cafcbb9d391b8..0ef2459cd05fd6 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -140,9 +140,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "single_adc_amp"); if (val >= 0) codec->single_adc_amp = !!val; - val = snd_hda_get_bool_hint(codec, "power_mgmt"); + val = snd_hda_get_bool_hint(codec, "power_save_node"); if (val >= 0) - codec->power_mgmt = !!val; + codec->power_save_node = !!val; val = snd_hda_get_bool_hint(codec, "auto_mute"); if (val >= 0) @@ -661,7 +661,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) continue; - if (codec->power_mgmt) { + if (codec->power_save_node) { if (!path->stream_enabled) continue; /* ignore unplugged paths except for DAC/ADC */ @@ -879,8 +879,8 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, path->active = false; /* make sure the widget is powered up */ - if (enable && (spec->power_down_unused || codec->power_mgmt)) - path_power_update(codec, path, codec->power_mgmt); + if (enable && (spec->power_down_unused || codec->power_save_node)) + path_power_update(codec, path, codec->power_save_node); for (i = path->depth - 1; i >= 0; i--) { hda_nid_t nid = path->path[i]; @@ -905,7 +905,7 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - if (!(spec->power_down_unused || codec->power_mgmt) || path->active) + if (!(spec->power_down_unused || codec->power_save_node) || path->active) return; sync_power_state_change(codec, path_power_update(codec, path, true)); } @@ -3981,7 +3981,7 @@ static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, { bool on; - if (!codec->power_mgmt) + if (!codec->power_save_node) return 0; on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; @@ -4038,7 +4038,7 @@ static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4067,7 +4067,7 @@ static void sync_all_pin_power_ctls(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; sync_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4111,7 +4111,7 @@ static int add_fake_beep_paths(struct hda_codec *codec) hda_nid_t nid = spec->beep_nid; int err; - if (!codec->power_mgmt || !nid) + if (!codec->power_save_node || !nid) return 0; err = add_fake_paths(codec, nid, cfg->line_outs, cfg->line_out_pins); if (err < 0) @@ -4233,7 +4233,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } set_pin_eapd(codec, nid, !mute); - if (codec->power_mgmt) { + if (codec->power_save_node) { bool on = !mute; if (on) on = snd_hda_jack_detect_state(codec, nid) @@ -4741,11 +4741,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) * @nid: audio widget * @on: power on/off flag * - * Set this in patch_ops.stream_pm. Only valid with power_mgmt flag. + * Set this in patch_ops.stream_pm. Only valid with power_save_node flag. */ void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on) { - if (codec->power_mgmt) + if (codec->power_save_node) set_path_power(codec, nid, -1, on); } EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm); @@ -4916,14 +4916,14 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused || codec->power_mgmt) + if (spec->power_down_unused || codec->power_save_node) codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); if (err < 0) return err; - if (codec->beep && codec->power_mgmt) { + if (codec->beep && codec->power_save_node) { err = add_fake_beep_paths(codec); if (err < 0) return err; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7e531d5cde51ae..5b7c173adcb8c4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4494,7 +4494,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; @@ -4600,7 +4600,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4650,7 +4650,7 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4692,7 +4692,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d5d1dca4f11b34..485663bb91010f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -133,7 +133,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->gen.power_down_unused = 1; return spec; } @@ -236,7 +236,7 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = codec->power_mgmt; + ucontrol->value.enumerated.item[0] = codec->power_save_node; return 0; } @@ -247,9 +247,9 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; bool val = !!ucontrol->value.enumerated.item[0]; - if (val == codec->power_mgmt) + if (val == codec->power_save_node) return 0; - codec->power_mgmt = val; + codec->power_save_node = val; spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; @@ -295,7 +295,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) bool enable; unsigned int verb, parm; - if (!codec->power_mgmt) + if (!codec->power_save_node) enable = false; else enable = is_aa_path_mute(codec) && !spec->gen.active_streams; @@ -517,7 +517,7 @@ static int via_parse_auto_config(struct hda_codec *codec) return err; /* disable widget PM at start for compatibility */ - codec->power_mgmt = 0; + codec->power_save_node = 0; spec->gen.power_down_unused = 0; return 0; } From 3088ef9f5dec3d72b24dcdd33d240823ab26b432 Mon Sep 17 00:00:00 2001 From: Howard Mitchell Date: Thu, 19 Mar 2015 12:08:29 +0000 Subject: [PATCH 252/411] ALSA: ControlName.txt: Add 'Analogue' as a prefix for volume controls Currently 'Playback Volume' is the correct way to express an analogue volume control. However, this control name has initialisation defaults applied when using 'alsactl restore' and in some cases this is not appropriate. An example would be a control that has a selection of 0db and -6dB of gain that is intended to set the fullscale ouput voltage of a DAC. The TI pcm512x family of DAcs have such a control. In this case the device/driver reset defaults are preferred. Signed-off-by: Howard Mitchell Acked-by: Takashi Iwai Signed-off-by: Mark Brown --- Documentation/sound/alsa/ControlNames.txt | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index 79a6127863ca82..3fc1cf50d28e18 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -71,11 +71,11 @@ SOURCE: HDMI/DP (either HDMI or DisplayPort) Exceptions (deprecated): - [Digital] Capture Source - [Digital] Capture Switch (aka input gain switch) - [Digital] Capture Volume (aka input gain volume) - [Digital] Playback Switch (aka output gain switch) - [Digital] Playback Volume (aka output gain volume) + [Analogue|Digital] Capture Source + [Analogue|Digital] Capture Switch (aka input gain switch) + [Analogue|Digital] Capture Volume (aka input gain volume) + [Analogue|Digital] Playback Switch (aka output gain switch) + [Analogue|Digital] Playback Volume (aka output gain volume) Tone Control - Switch Tone Control - Bass Tone Control - Treble From 4d9b13c7cc803fbde59d7e998f7de2b9a2101c7e Mon Sep 17 00:00:00 2001 From: Howard Mitchell Date: Thu, 19 Mar 2015 12:08:30 +0000 Subject: [PATCH 253/411] ASoC: pcm512x: Add 'Analogue' prefix to analogue volume controls This is to ensure that 'alsactl restore' does not apply default initialisation as the chip reset defaults are preferred. Signed-off-by: Howard Mitchell Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/pcm512x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 9974f201a08f44..194f4c8c56117f 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -304,9 +304,9 @@ static const struct soc_enum pcm512x_veds = static const struct snd_kcontrol_new pcm512x_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), -SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, +SOC_DOUBLE_TLV("Analogue Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), -SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, +SOC_DOUBLE_TLV("Analogue Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, PCM512x_RQMR_SHIFT, 1, 1), From 6ffa84df2b26b1ba78f6a55b373b9513c93017d6 Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Wed, 18 Mar 2015 17:48:57 +0100 Subject: [PATCH 254/411] ASoC: fsl: constify of_device_id array of_device_id is always used as const. (See driver.of_match_table and open firmware functions) Signed-off-by: Fabian Frederick Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 08d2a8069b0abd..0bab76051fd830 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -326,7 +326,7 @@ static int psc_ac97_of_remove(struct platform_device *op) } /* Match table for of_platform binding */ -static struct of_device_id psc_ac97_match[] = { +static const struct of_device_id psc_ac97_match[] = { { .compatible = "fsl,mpc5200-psc-ac97", }, { .compatible = "fsl,mpc5200b-psc-ac97", }, {} diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 51fb0c00fe737d..d8232943ccb67c 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -217,7 +217,7 @@ static int psc_i2s_of_remove(struct platform_device *op) } /* Match table for of_platform binding */ -static struct of_device_id psc_i2s_match[] = { +static const struct of_device_id psc_i2s_match[] = { { .compatible = "fsl,mpc5200-psc-i2s", }, { .compatible = "fsl,mpc5200b-psc-i2s", }, {} diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index c44459d24c5050..ec731223cab3d7 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -113,7 +113,7 @@ static int pcm030_fabric_remove(struct platform_device *op) return ret; } -static struct of_device_id pcm030_audio_match[] = { +static const struct of_device_id pcm030_audio_match[] = { { .compatible = "phytec,pcm030-audio-fabric", }, {} }; From 7f2c52afc02554f18db27242b62d4047bbd8df4c Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Wed, 18 Mar 2015 17:48:58 +0100 Subject: [PATCH 255/411] ASoC: kirkwood: constify of_device_id array of_device_id is always used as const. (See driver.of_match_table and open firmware functions) Signed-off-by: Fabian Frederick Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index def7d8260c4e57..0a7ff0e4247b47 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -643,7 +643,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) } #ifdef CONFIG_OF -static struct of_device_id mvebu_audio_of_match[] = { +static const struct of_device_id mvebu_audio_of_match[] = { { .compatible = "marvell,kirkwood-audio" }, { .compatible = "marvell,dove-audio" }, { .compatible = "marvell,armada370-audio" }, From 261e43a35893830ff32de94443d685c8b9770f5f Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Wed, 18 Mar 2015 17:48:59 +0100 Subject: [PATCH 256/411] ASoC: rt5631: constify of_device_id array of_device_id is always used as const. (See driver.of_match_table and open firmware functions) Signed-off-by: Fabian Frederick Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index c61852742ee308..2c10d77727af5d 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1675,7 +1675,7 @@ static const struct i2c_device_id rt5631_i2c_id[] = { MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); #ifdef CONFIG_OF -static struct of_device_id rt5631_i2c_dt_ids[] = { +static const struct of_device_id rt5631_i2c_dt_ids[] = { { .compatible = "realtek,rt5631"}, { .compatible = "realtek,alc5631"}, { } From f7d4bfee667e236bc9680cfe2f17247bc007f4e3 Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Wed, 18 Mar 2015 17:49:00 +0100 Subject: [PATCH 257/411] ASoC: ak4554: constify of_device_id array of_device_id is always used as const. (See driver.of_match_table and open firmware functions) Signed-off-by: Fabian Frederick Signed-off-by: Mark Brown --- sound/soc/codecs/ak4554.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index 16ce9f9fefa175..298dedc051403b 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -84,7 +84,7 @@ static int ak4554_soc_remove(struct platform_device *pdev) return 0; } -static struct of_device_id ak4554_of_match[] = { +static const struct of_device_id ak4554_of_match[] = { { .compatible = "asahi-kasei,ak4554" }, {}, }; From c660c0a805860e3abf22b44a2508ff46a549ffa9 Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Wed, 18 Mar 2015 17:49:01 +0100 Subject: [PATCH 258/411] ASoC: fsi: constify of_device_id array of_device_id is always used as const. (See driver.of_match_table and open firmware functions) Signed-off-by: Fabian Frederick Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b87b22e88e43de..eef7083ec7d9e0 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1876,7 +1876,7 @@ static void fsi_handler_init(struct fsi_priv *fsi, } } -static struct of_device_id fsi_of_match[]; +static const struct of_device_id fsi_of_match[]; static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; @@ -2092,7 +2092,7 @@ static struct fsi_core fsi2_core = { .b_mclk = B_MST_CTLR, }; -static struct of_device_id fsi_of_match[] = { +static const struct of_device_id fsi_of_match[] = { { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, {}, From 33187fb4a203e44dec11211f2fa86a63139615bc Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Wed, 18 Mar 2015 17:49:02 +0100 Subject: [PATCH 259/411] ASoC: rsnd: constify of_device_id array of_device_id is always used as const. (See driver.of_match_table and open firmware functions) Signed-off-by: Fabian Frederick Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 7b995f025e2276..7be1602f57f3b8 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -107,7 +107,7 @@ static struct rsnd_of_data rsnd_of_data_gen2 = { .flags = RSND_GEN2, }; -static struct of_device_id rsnd_of_match[] = { +static const struct of_device_id rsnd_of_match[] = { { .compatible = "renesas,rcar_sound-gen1", .data = &rsnd_of_data_gen1 }, { .compatible = "renesas,rcar_sound-gen2", .data = &rsnd_of_data_gen2 }, {}, From 044930b4a69a6c0645b6199bec4f870e0b6e77f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Mar 2015 04:13:47 +0000 Subject: [PATCH 260/411] ASoC: rsnd: no more SSI restart when unusual situation It will be SSI interrupt endless loop f unusual situation happen. This patch adds restart limit for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fea4aa53918ab8..060d3d2052506b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -416,11 +416,14 @@ static irqreturn_t rsnd_ssi_interrupt(int irq, void *data) /* * restart SSI */ - rsnd_ssi_stop(mod, priv); - rsnd_ssi_start(mod, priv); - dev_dbg(dev, "%s[%d] restart\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + rsnd_ssi_stop(mod, priv); + if (ssi->err < 1024) + rsnd_ssi_start(mod, priv); + else + dev_warn(dev, "no more SSI restart\n"); } rsnd_ssi_record_error(ssi, status); From 639b231f866c6cc6dcefc33bcaf31e7554697186 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Mar 2015 04:14:04 +0000 Subject: [PATCH 261/411] ASoC: rsnd: no more SRC restart when unusual situation It will be SRC interrupt endless loop f unusual situation happen. This patch adds restart limit for it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6ce8985757c12b..cc93f32b0de02f 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -620,13 +620,17 @@ static irqreturn_t rsnd_src_interrupt_gen2(int irq, void *data) if (rsnd_src_error_record_gen2(mod)) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_src *src = rsnd_mod_to_src(mod); struct device *dev = rsnd_priv_to_dev(priv); - _rsnd_src_stop_gen2(mod); - _rsnd_src_start_gen2(mod); - dev_dbg(dev, "%s[%d] restart\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); + + _rsnd_src_stop_gen2(mod); + if (src->err < 1024) + _rsnd_src_start_gen2(mod); + else + dev_warn(dev, "no more SRC restart\n"); } return IRQ_HANDLED; From 072bd1e7e136e089e41c8cc2d2b2251ed60b5bcd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Mar 2015 04:14:20 +0000 Subject: [PATCH 262/411] ASoC: rsnd: tidyup error message format This driver sometimes fixups debug/error message format 30cc4faf703955cd5cd07da489bd817ae43e3fec (ASoC: rsnd: tidyup debug message format and timing) 337b0b4c5f415705f1b97df57cecfac45903449a (ASoC: rsnd: error meesage indicates its port) But, it still exist un-fomated error message. This patch fixup it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 7be1602f57f3b8..cd78a17f9cf971 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -232,7 +232,7 @@ static int rsnd_dai_connect(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - dev_err(dev, "%s%d is not empty\n", + dev_err(dev, "%s[%d] is not empty\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); return -EIO; From d2c4b80c5b392c8437f9174161a17176b9c3cd76 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Mar 2015 04:14:45 +0000 Subject: [PATCH 263/411] ASoC: rsnd: show debug info for sampling rate convert Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 7ac35c9d1cb8b0..7af374bd0849a9 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -183,6 +183,8 @@ int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod, rsnd_mod_bset(mod, DIV_EN, en, en); + dev_dbg(dev, "convert rate %d <-> %d\n", src_rate, dst_rate); + return 0; } From f8c3c3094302cb25d9720804b8100fdd37a3ace0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Mar 2015 04:15:18 +0000 Subject: [PATCH 264/411] ASoC: rsnd: add dai_link stream name This patch adds missing dai_link stream_name which is used when DPCM Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 8 ++++++++ sound/soc/sh/rcar/rsnd.h | 1 + 2 files changed, 9 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index cd78a17f9cf971..519d85692e0c61 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -648,20 +648,28 @@ static int rsnd_dai_probe(struct platform_device *pdev, drv[i].name = rdai[i].name; drv[i].ops = &rsnd_soc_dai_ops; if (pmod) { + snprintf(rdai[i].playback.name, RSND_DAI_NAME_SIZE, + "DAI%d Playback", i); + drv[i].playback.rates = RSND_RATES; drv[i].playback.formats = RSND_FMTS; drv[i].playback.channels_min = 2; drv[i].playback.channels_max = 2; + drv[i].playback.stream_name = rdai[i].playback.name; rdai[i].playback.info = &info->dai_info[i].playback; rdai[i].playback.rdai = rdai + i; rsnd_path_init(priv, &rdai[i], &rdai[i].playback); } if (cmod) { + snprintf(rdai[i].capture.name, RSND_DAI_NAME_SIZE, + "DAI%d Capture", i); + drv[i].capture.rates = RSND_RATES; drv[i].capture.formats = RSND_FMTS; drv[i].capture.channels_min = 2; drv[i].capture.channels_max = 2; + drv[i].capture.stream_name = rdai[i].capture.name; rdai[i].capture.info = &info->dai_info[i].capture; rdai[i].capture.rdai = rdai + i; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 52c401c9eeef62..5f35af7ff82126 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -301,6 +301,7 @@ struct dma_chan *rsnd_mod_dma_req(struct rsnd_mod *mod); */ #define RSND_DAI_NAME_SIZE 16 struct rsnd_dai_stream { + char name[RSND_DAI_NAME_SIZE]; struct snd_pcm_substream *substream; struct rsnd_mod *mod[RSND_MOD_MAX]; struct rsnd_dai_path_info *info; /* rcar_snd.h */ From f073faa73626f41db7050a69edd5074c53ce6d6c Mon Sep 17 00:00:00 2001 From: Howard Mitchell Date: Fri, 20 Mar 2015 21:13:45 +0000 Subject: [PATCH 265/411] ASoC: pcm512x: Fix divide by zero issue If den=1 and pllin_rate>20MHz then den and num are adjusted to 0 causing a divide by zero error a few lines further on. Therefore this patch correctly scales num and den such that pllin_rate/den < 20MHz as required in the device data sheet. Signed-off-by: Howard Mitchell Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/pcm512x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 194f4c8c56117f..0676ab8be03fa3 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -576,8 +576,8 @@ static int pcm512x_find_pll_coeff(struct snd_soc_dai *dai, /* pllin_rate / P (or here, den) cannot be greater than 20 MHz */ if (pllin_rate / den > 20000000 && num < 8) { - num *= 20000000 / (pllin_rate / den); - den *= 20000000 / (pllin_rate / den); + num *= DIV_ROUND_UP(pllin_rate / den, 20000000); + den *= DIV_ROUND_UP(pllin_rate / den, 20000000); } dev_dbg(dev, "num / den = %lu / %lu\n", num, den); From 6212755eff3171e0211bb6a9f4706e115217588c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 20 Mar 2015 15:31:34 -0500 Subject: [PATCH 266/411] ASoC: Intel: remove misleading DMA error messages on Baytrail platforms During probe, the Baytrail audio driver reports errors such as: [44.172040] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [44.172137] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed Those error messages are misleading, there is no error since the DMA is explicitly not configured for Baytrail. Add a test to remove DMA error checks when DMA is not configured and return silently. Acked-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-dsp.h | 1 + sound/soc/intel/sst-firmware.c | 4 ++++ 3 files changed, 6 insertions(+) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index b3d84560fbb5da..42f293f9c6e2dc 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -142,6 +142,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_acpi->desc = desc; sst_acpi->mach = mach; + sst_pdata->resindex_dma_base = desc->resindex_dma_base; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; sst_pdata->dma_base = desc->resindex_dma_base; diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32f00777a..148d8c589a43f6 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -206,6 +206,7 @@ struct sst_pdata { const struct firmware *fw; /* DMA */ + int resindex_dma_base; /* other fields invalid if equals to -1 */ u32 dma_base; u32 dma_size; int dma_engine; diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index b3f9489794a6ac..28beceb3f252dd 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -271,6 +271,10 @@ int sst_dma_new(struct sst_dsp *sst) const char *dma_dev_name; int ret = 0; + if (sst->pdata->resindex_dma_base == -1) + /* DMA is not used, return and squelsh error messages */ + return 0; + /* configure the correct platform data for whatever DMA engine * is attached to the ADSP IP. */ switch (sst->pdata->dma_engine) { From e3d280fc6d42017b2379503fbda83655a05294fe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 21:46:37 +0100 Subject: [PATCH 267/411] ALSA: hda - Make snd_hda_bus_type public Define the common hd-audio driver and device types to bind over snd_hda_bus_type publicly. This allows to implement other type of device and driver code over hd-audio bus. Now both struct hda_codec and struct hda_codec_driver inherit these new struct hdac_device and struct hdac_driver, respectively. The bus registration is done in subsys_initcall() to assure it before any other driver registrations. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 42 ++++++++++++++++++++++++++++++++++ sound/Kconfig | 2 ++ sound/Makefile | 2 +- sound/hda/Kconfig | 2 ++ sound/hda/Makefile | 3 +++ sound/hda/hda_bus_type.c | 42 ++++++++++++++++++++++++++++++++++ sound/pci/hda/Kconfig | 1 + sound/pci/hda/hda_bind.c | 47 ++++++++++++--------------------------- sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_codec.h | 11 +++++---- 10 files changed, 113 insertions(+), 40 deletions(-) create mode 100644 include/sound/hdaudio.h create mode 100644 sound/hda/Kconfig create mode 100644 sound/hda/Makefile create mode 100644 sound/hda/hda_bus_type.c diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h new file mode 100644 index 00000000000000..2381509bee9fc8 --- /dev/null +++ b/include/sound/hdaudio.h @@ -0,0 +1,42 @@ +/* + * HD-audio core stuff + */ + +#ifndef __SOUND_HDAUDIO_H +#define __SOUND_HDAUDIO_H + +#include + +/* + * exported bus type + */ +extern struct bus_type snd_hda_bus_type; + +/* + * HD-audio codec base device + */ +struct hdac_device { + struct device dev; + int type; +}; + +/* device/driver type used for matching */ +enum { + HDA_DEV_CORE, + HDA_DEV_LEGACY, +}; + +#define dev_to_hdac_dev(_dev) container_of(_dev, struct hdac_device, dev) + +/* + * HD-audio codec base driver + */ +struct hdac_driver { + struct device_driver driver; + int type; + int (*match)(struct hdac_device *dev, struct hdac_driver *drv); +}; + +#define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver) + +#endif /* __SOUND_HDAUDIO_H */ diff --git a/sound/Kconfig b/sound/Kconfig index c710ce2c5c3703..5a240e050ae669 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -76,6 +76,8 @@ source "sound/isa/Kconfig" source "sound/pci/Kconfig" +source "sound/hda/Kconfig" + source "sound/ppc/Kconfig" source "sound/aoa/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index ce9132b1c39522..77320709fd26b0 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -6,7 +6,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ - firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ + firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig new file mode 100644 index 00000000000000..4f428ccf64adea --- /dev/null +++ b/sound/hda/Kconfig @@ -0,0 +1,2 @@ +config SND_HDA_CORE + tristate diff --git a/sound/hda/Makefile b/sound/hda/Makefile new file mode 100644 index 00000000000000..59c8d1feb5aa3e --- /dev/null +++ b/sound/hda/Makefile @@ -0,0 +1,3 @@ +snd-hda-core-objs := hda_bus_type.o + +obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c new file mode 100644 index 00000000000000..519914a12e8aaf --- /dev/null +++ b/sound/hda/hda_bus_type.c @@ -0,0 +1,42 @@ +/* + * HD-audio bus + */ +#include +#include +#include +#include +#include + +MODULE_DESCRIPTION("HD-audio bus"); +MODULE_LICENSE("GPL"); + +static int hda_bus_match(struct device *dev, struct device_driver *drv) +{ + struct hdac_device *hdev = dev_to_hdac_dev(dev); + struct hdac_driver *hdrv = drv_to_hdac_driver(drv); + + if (hdev->type != hdrv->type) + return 0; + if (hdrv->match) + return hdrv->match(hdev, hdrv); + return 1; +} + +struct bus_type snd_hda_bus_type = { + .name = "hdaudio", + .match = hda_bus_match, +}; +EXPORT_SYMBOL_GPL(snd_hda_bus_type); + +static int __init hda_bus_init(void) +{ + return bus_register(&snd_hda_bus_type); +} + +static void __exit hda_bus_exit(void) +{ + bus_unregister(&snd_hda_bus_type); +} + +subsys_initcall(hda_bus_init); +module_exit(hda_bus_exit); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 7f0f2c5a4e9738..a5ed1c18178427 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -5,6 +5,7 @@ config SND_HDA select SND_PCM select SND_VMASTER select SND_KCTL_JACK + select SND_HDA_CORE config SND_HDA_INTEL tristate "HD Audio PCI" diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 1f40ce3c16969f..e3bd2807b64469 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -47,11 +47,11 @@ static struct hda_vendor_id hda_vendor_ids[] = { /* * find a matching codec preset */ -static int hda_bus_match(struct device *dev, struct device_driver *drv) +static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv) { - struct hda_codec *codec = container_of(dev, struct hda_codec, dev); + struct hda_codec *codec = container_of(dev, struct hda_codec, core); struct hda_codec_driver *driver = - container_of(drv, struct hda_codec_driver, driver); + container_of(drv, struct hda_codec_driver, core); const struct hda_codec_preset *preset; /* check probe_id instead of vendor_id if set */ u32 id = codec->probe_id ? codec->probe_id : codec->vendor_id; @@ -154,20 +154,22 @@ static void hda_codec_driver_shutdown(struct device *dev) int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, struct module *owner) { - drv->driver.name = name; - drv->driver.owner = owner; - drv->driver.bus = &snd_hda_bus_type; - drv->driver.probe = hda_codec_driver_probe; - drv->driver.remove = hda_codec_driver_remove; - drv->driver.shutdown = hda_codec_driver_shutdown; - drv->driver.pm = &hda_codec_driver_pm; - return driver_register(&drv->driver); + drv->core.driver.name = name; + drv->core.driver.owner = owner; + drv->core.driver.bus = &snd_hda_bus_type; + drv->core.driver.probe = hda_codec_driver_probe; + drv->core.driver.remove = hda_codec_driver_remove; + drv->core.driver.shutdown = hda_codec_driver_shutdown; + drv->core.driver.pm = &hda_codec_driver_pm; + drv->core.type = HDA_DEV_LEGACY; + drv->core.match = hda_codec_match; + return driver_register(&drv->core.driver); } EXPORT_SYMBOL_GPL(__hda_codec_driver_register); void hda_codec_driver_unregister(struct hda_codec_driver *drv) { - driver_unregister(&drv->driver); + driver_unregister(&drv->core.driver); } EXPORT_SYMBOL_GPL(hda_codec_driver_unregister); @@ -319,24 +321,3 @@ int snd_hda_codec_configure(struct hda_codec *codec) return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_configure); - -/* - * bus registration - */ -struct bus_type snd_hda_bus_type = { - .name = "hdaudio", - .match = hda_bus_match, -}; - -static int __init hda_codec_init(void) -{ - return bus_register(&snd_hda_bus_type); -} - -static void __exit hda_codec_exit(void) -{ - bus_unregister(&snd_hda_bus_type); -} - -module_init(hda_codec_init); -module_exit(hda_codec_exit); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7e38d6f7314b7d..e14f9f56287436 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1294,6 +1294,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, dev_set_name(dev, "hdaudioC%dD%d", card->number, codec_addr); dev_set_drvdata(dev, codec); /* for sysfs */ device_enable_async_suspend(dev); + codec->core.type = HDA_DEV_LEGACY; codec->bus = bus; codec->card = card; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index ccf355d4a8fa09..31a9e10e5137ca 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -26,6 +26,7 @@ #include #include #include +#include #include /* @@ -172,7 +173,7 @@ struct hda_codec_preset { #define HDA_CODEC_ID_GENERIC 0x00000201 struct hda_codec_driver { - struct device_driver driver; + struct hdac_driver core; const struct hda_codec_preset *preset; }; @@ -276,7 +277,7 @@ struct hda_pcm { /* codec information */ struct hda_codec { - struct device dev; + struct hdac_device core; struct hda_bus *bus; struct snd_card *card; unsigned int addr; /* codec addr*/ @@ -409,10 +410,8 @@ struct hda_codec { struct snd_array verbs; }; -#define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, dev) -#define hda_codec_dev(_dev) (&(_dev)->dev) - -extern struct bus_type snd_hda_bus_type; +#define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, core.dev) +#define hda_codec_dev(_dev) (&(_dev)->core.dev) /* direction */ enum { From d068ebc25e6e1360510ad8023fe7bca3dacd204e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2015 23:22:59 +0100 Subject: [PATCH 268/411] ALSA: hda - Move some codes up to hdac_bus struct A few basic codes for communicating over HD-audio bus are moved to struct hdac_bus now. It has only command and get_response ops in addition to the unsolicited event handling. Note that the codec-side tracing support is disabled temporarily during this transition due to the code shuffling. It will be re-enabled later once when all pieces are settled down. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 61 +++++++++++ sound/hda/Makefile | 2 +- sound/hda/hdac_bus.c | 181 +++++++++++++++++++++++++++++++ sound/pci/hda/hda_bind.c | 10 ++ sound/pci/hda/hda_codec.c | 191 +++++++++++---------------------- sound/pci/hda/hda_codec.h | 34 ++---- sound/pci/hda/hda_controller.c | 8 +- sound/pci/hda/hda_intel.c | 4 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/patch_conexant.c | 4 +- 10 files changed, 335 insertions(+), 162 deletions(-) create mode 100644 sound/hda/hdac_bus.c diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 2381509bee9fc8..848ab6e6809958 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -6,6 +6,11 @@ #define __SOUND_HDAUDIO_H #include +#include + +struct hdac_bus; +struct hdac_device; +struct hdac_driver; /* * exported bus type @@ -18,6 +23,9 @@ extern struct bus_type snd_hda_bus_type; struct hdac_device { struct device dev; int type; + struct hdac_bus *bus; + unsigned int addr; /* codec address */ + struct list_head list; /* list point for bus codec_list */ }; /* device/driver type used for matching */ @@ -35,8 +43,61 @@ struct hdac_driver { struct device_driver driver; int type; int (*match)(struct hdac_device *dev, struct hdac_driver *drv); + void (*unsol_event)(struct hdac_device *dev, unsigned int event); }; #define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver) +/* + * HD-audio bus base driver + */ +struct hdac_bus_ops { + /* send a single command */ + int (*command)(struct hdac_bus *bus, unsigned int cmd); + /* get a response from the last command */ + int (*get_response)(struct hdac_bus *bus, unsigned int addr, + unsigned int *res); +}; + +#define HDA_UNSOL_QUEUE_SIZE 64 + +struct hdac_bus { + struct device *dev; + const struct hdac_bus_ops *ops; + + /* codec linked list */ + struct list_head codec_list; + unsigned int num_codecs; + + /* link caddr -> codec */ + struct hdac_device *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; + + /* unsolicited event queue */ + u32 unsol_queue[HDA_UNSOL_QUEUE_SIZE * 2]; /* ring buffer */ + unsigned int unsol_rp, unsol_wp; + struct work_struct unsol_work; + + /* bit flags of powered codecs */ + unsigned long codec_powered; + + /* flags */ + bool sync_write:1; /* sync after verb write */ + + /* locks */ + struct mutex cmd_mutex; +}; + +int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, + const struct hdac_bus_ops *ops); +void snd_hdac_bus_exit(struct hdac_bus *bus); +int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr, + unsigned int cmd, unsigned int *res); +int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr, + unsigned int cmd, unsigned int *res); +void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex); + +int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec); +void snd_hdac_bus_remove_device(struct hdac_bus *bus, + struct hdac_device *codec); + #endif /* __SOUND_HDAUDIO_H */ diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 59c8d1feb5aa3e..828680b282fab1 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -1,3 +1,3 @@ -snd-hda-core-objs := hda_bus_type.o +snd-hda-core-objs := hda_bus_type.o hdac_bus.o obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c new file mode 100644 index 00000000000000..364f64c0e4a313 --- /dev/null +++ b/sound/hda/hdac_bus.c @@ -0,0 +1,181 @@ +/* + * HD-audio core bus driver + */ + +#include +#include +#include +#include +#include + +static void process_unsol_events(struct work_struct *work); + +/** + * snd_hdac_bus_init - initialize a HD-audio bas bus + * @bus: the pointer to bus object + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, + const struct hdac_bus_ops *ops) +{ + memset(bus, 0, sizeof(*bus)); + bus->dev = dev; + bus->ops = ops; + INIT_LIST_HEAD(&bus->codec_list); + INIT_WORK(&bus->unsol_work, process_unsol_events); + mutex_init(&bus->cmd_mutex); + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_init); + +/** + * snd_hdac_bus_exit - clean up a HD-audio bas bus + * @bus: the pointer to bus object + */ +void snd_hdac_bus_exit(struct hdac_bus *bus) +{ + WARN_ON(!list_empty(&bus->codec_list)); + cancel_work_sync(&bus->unsol_work); +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_exit); + +/** + * snd_hdac_bus_exec_verb - execute a HD-audio verb on the given bus + * @bus: bus object + * @cmd: HD-audio encoded verb + * @res: pointer to store the response, NULL if performing asynchronously + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr, + unsigned int cmd, unsigned int *res) +{ + int err; + + mutex_lock(&bus->cmd_mutex); + err = snd_hdac_bus_exec_verb_unlocked(bus, addr, cmd, res); + mutex_unlock(&bus->cmd_mutex); + return err; +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_exec_verb); + +/** + * snd_hdac_bus_exec_verb_unlocked - unlocked version + * @bus: bus object + * @cmd: HD-audio encoded verb + * @res: pointer to store the response, NULL if performing asynchronously + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr, + unsigned int cmd, unsigned int *res) +{ + unsigned int tmp; + int err; + + if (cmd == ~0) + return -EINVAL; + + if (res) + *res = -1; + else if (bus->sync_write) + res = &tmp; + for (;;) { + err = bus->ops->command(bus, cmd); + if (err != -EAGAIN) + break; + /* process pending verbs */ + err = bus->ops->get_response(bus, addr, &tmp); + if (err) + break; + } + if (!err && res) + err = bus->ops->get_response(bus, addr, res); + return err; +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_exec_verb_unlocked); + +/** + * snd_hdac_bus_queue_event - add an unsolicited event to queue + * @bus: the BUS + * @res: unsolicited event (lower 32bit of RIRB entry) + * @res_ex: codec addr and flags (upper 32bit or RIRB entry) + * + * Adds the given event to the queue. The events are processed in + * the workqueue asynchronously. Call this function in the interrupt + * hanlder when RIRB receives an unsolicited event. + */ +void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex) +{ + unsigned int wp; + + if (!bus) + return; + + wp = (bus->unsol_wp + 1) % HDA_UNSOL_QUEUE_SIZE; + bus->unsol_wp = wp; + + wp <<= 1; + bus->unsol_queue[wp] = res; + bus->unsol_queue[wp + 1] = res_ex; + + schedule_work(&bus->unsol_work); +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_queue_event); + +/* + * process queued unsolicited events + */ +static void process_unsol_events(struct work_struct *work) +{ + struct hdac_bus *bus = container_of(work, struct hdac_bus, unsol_work); + struct hdac_device *codec; + struct hdac_driver *drv; + unsigned int rp, caddr, res; + + while (bus->unsol_rp != bus->unsol_wp) { + rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; + bus->unsol_rp = rp; + rp <<= 1; + res = bus->unsol_queue[rp]; + caddr = bus->unsol_queue[rp + 1]; + if (!(caddr & (1 << 4))) /* no unsolicited event? */ + continue; + codec = bus->caddr_tbl[caddr & 0x0f]; + if (!codec || !codec->dev.driver) + continue; + drv = drv_to_hdac_driver(codec->dev.driver); + if (drv->unsol_event) + drv->unsol_event(codec, res); + } +} + +int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec) +{ + if (bus->caddr_tbl[codec->addr]) { + dev_err(bus->dev, "address 0x%x is already occupied\n", + codec->addr); + return -EBUSY; + } + + list_add_tail(&codec->list, &bus->codec_list); + bus->caddr_tbl[codec->addr] = codec; + set_bit(codec->addr, &bus->codec_powered); + bus->num_codecs++; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_add_device); + +void snd_hdac_bus_remove_device(struct hdac_bus *bus, + struct hdac_device *codec) +{ + WARN_ON(bus != codec->bus); + if (list_empty(&codec->list)) + return; + list_del_init(&codec->list); + bus->caddr_tbl[codec->addr] = NULL; + clear_bit(codec->addr, &bus->codec_powered); + bus->num_codecs--; +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_remove_device); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index e3bd2807b64469..0b9ea70c546b35 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -74,6 +74,15 @@ static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv) return 0; } +/* process an unsolicited event */ +static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) +{ + struct hda_codec *codec = container_of(dev, struct hda_codec, core); + + if (codec->patch_ops.unsol_event) + codec->patch_ops.unsol_event(codec, ev); +} + /* reset the codec name from the preset */ static int codec_refresh_name(struct hda_codec *codec, const char *name) { @@ -163,6 +172,7 @@ int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, drv->core.driver.pm = &hda_codec_driver_pm; drv->core.type = HDA_DEV_LEGACY; drv->core.match = hda_codec_match; + drv->core.unsol_event = hda_codec_unsol_event; return driver_register(&drv->core.driver); } EXPORT_SYMBOL_GPL(__hda_codec_driver_register); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e14f9f56287436..f96bff37c78719 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -39,9 +39,6 @@ #include "hda_jack.h" #include -#define CREATE_TRACE_POINTS -#include "hda_trace.h" - #ifdef CONFIG_PM #define codec_in_pm(codec) atomic_read(&(codec)->in_pm) #define hda_codec_is_power_on(codec) \ @@ -128,16 +125,17 @@ static inline unsigned int make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { + unsigned int addr = codec->core.addr; u32 val; - if ((codec->addr & ~0xf) || (nid & ~0x7f) || + if ((addr & ~0xf) || (nid & ~0x7f) || (verb & ~0xfff) || (parm & ~0xffff)) { codec_err(codec, "hda-codec: out of range cmd %x:%x:%x:%x\n", - codec->addr, nid, verb, parm); + addr, nid, verb, parm); return ~0; } - val = (u32)codec->addr << 28; + val = (u32)addr << 28; val |= (u32)nid << 20; val |= verb << 8; val |= parm; @@ -156,33 +154,20 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, if (cmd == ~0) return -1; - if (res) - *res = -1; again: snd_hda_power_up(codec); - mutex_lock(&bus->cmd_mutex); + mutex_lock(&bus->core.cmd_mutex); if (flags & HDA_RW_NO_RESPONSE_FALLBACK) bus->no_response_fallback = 1; - for (;;) { - trace_hda_send_cmd(codec, cmd); - err = bus->ops.command(bus, cmd); - if (err != -EAGAIN) - break; - /* process pending verbs */ - bus->ops.get_response(bus, codec->addr); - } - if (!err && res) { - *res = bus->ops.get_response(bus, codec->addr); - trace_hda_get_response(codec, *res); - } + err = snd_hdac_bus_exec_verb_unlocked(&bus->core, codec->core.addr, + cmd, res); bus->no_response_fallback = 0; - mutex_unlock(&bus->cmd_mutex); + mutex_unlock(&bus->core.cmd_mutex); snd_hda_power_down(codec); - if (!codec_in_pm(codec) && res && *res == -1 && bus->rirb_error) { + if (!codec_in_pm(codec) && res && err < 0 && bus->rirb_error) { if (bus->response_reset) { codec_dbg(codec, "resetting BUS due to fatal communication error\n"); - trace_hda_bus_reset(bus); bus->ops.bus_reset(bus); } goto again; @@ -233,9 +218,7 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { unsigned int cmd = make_codec_cmd(codec, nid, flags, verb, parm); - unsigned int res; - return codec_exec_verb(codec, cmd, flags, - codec->bus->sync_write ? &res : NULL); + return codec_exec_verb(codec, cmd, flags, NULL); } EXPORT_SYMBOL_GPL(snd_hda_codec_write); @@ -664,65 +647,6 @@ int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, return devices; } -/** - * snd_hda_queue_unsol_event - add an unsolicited event to queue - * @bus: the BUS - * @res: unsolicited event (lower 32bit of RIRB entry) - * @res_ex: codec addr and flags (upper 32bit or RIRB entry) - * - * Adds the given event to the queue. The events are processed in - * the workqueue asynchronously. Call this function in the interrupt - * hanlder when RIRB receives an unsolicited event. - * - * Returns 0 if successful, or a negative error code. - */ -int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) -{ - struct hda_bus_unsolicited *unsol; - unsigned int wp; - - if (!bus) - return 0; - - trace_hda_unsol_event(bus, res, res_ex); - unsol = &bus->unsol; - wp = (unsol->wp + 1) % HDA_UNSOL_QUEUE_SIZE; - unsol->wp = wp; - - wp <<= 1; - unsol->queue[wp] = res; - unsol->queue[wp + 1] = res_ex; - - schedule_work(&unsol->work); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_queue_unsol_event); - -/* - * process queued unsolicited events - */ -static void process_unsol_events(struct work_struct *work) -{ - struct hda_bus *bus = container_of(work, struct hda_bus, unsol.work); - struct hda_bus_unsolicited *unsol = &bus->unsol; - struct hda_codec *codec; - unsigned int rp, caddr, res; - - while (unsol->rp != unsol->wp) { - rp = (unsol->rp + 1) % HDA_UNSOL_QUEUE_SIZE; - unsol->rp = rp; - rp <<= 1; - res = unsol->queue[rp]; - caddr = unsol->queue[rp + 1]; - if (!(caddr & (1 << 4))) /* no unsolicited event? */ - continue; - codec = bus->caddr_tbl[caddr & 0x0f]; - if (codec && codec->patch_ops.unsol_event) - codec->patch_ops.unsol_event(codec, res); - } -} - /* * destructor */ @@ -730,11 +654,9 @@ static void snd_hda_bus_free(struct hda_bus *bus) { if (!bus) return; - - WARN_ON(!list_empty(&bus->codec_list)); - cancel_work_sync(&bus->unsol.work); if (bus->ops.private_free) bus->ops.private_free(bus); + snd_hdac_bus_exit(&bus->core); kfree(bus); } @@ -751,6 +673,26 @@ static int snd_hda_bus_dev_disconnect(struct snd_device *device) return 0; } +/* hdac_bus_ops translations */ +static int _hda_bus_command(struct hdac_bus *_bus, unsigned int cmd) +{ + struct hda_bus *bus = container_of(_bus, struct hda_bus, core); + return bus->ops.command(bus, cmd); +} + +static int _hda_bus_get_response(struct hdac_bus *_bus, unsigned int addr, + unsigned int *res) +{ + struct hda_bus *bus = container_of(_bus, struct hda_bus, core); + *res = bus->ops.get_response(bus, addr); + return bus->rirb_error ? -EIO : 0; +} + +static const struct hdac_bus_ops bus_ops = { + .command = _hda_bus_command, + .get_response = _hda_bus_get_response, +}; + /** * snd_hda_bus_new - create a HDA bus * @card: the card entry @@ -775,11 +717,14 @@ int snd_hda_bus_new(struct snd_card *card, if (!bus) return -ENOMEM; + err = snd_hdac_bus_init(&bus->core, card->dev, &bus_ops); + if (err < 0) { + kfree(bus); + return err; + } + bus->card = card; - mutex_init(&bus->cmd_mutex); mutex_init(&bus->prepare_mutex); - INIT_LIST_HEAD(&bus->codec_list); - INIT_WORK(&bus->unsol.work, process_unsol_events); err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops); if (err < 0) { @@ -1233,9 +1178,7 @@ static void snd_hda_codec_dev_release(struct device *dev) struct hda_codec *codec = dev_to_hda_codec(dev); free_init_pincfgs(codec); - list_del(&codec->list); - codec->bus->caddr_tbl[codec->addr] = NULL; - clear_bit(codec->addr, &codec->bus->codec_powered); + snd_hdac_bus_remove_device(&codec->bus->core, &codec->core); snd_hda_sysfs_clear(codec); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); @@ -1243,7 +1186,6 @@ static void snd_hda_codec_dev_release(struct device *dev) kfree(codec->chip_name); kfree(codec->modelname); kfree(codec->wcaps); - codec->bus->num_codecs--; kfree(codec); } @@ -1274,27 +1216,23 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; - if (bus->caddr_tbl[codec_addr]) { - dev_err(card->dev, - "address 0x%x is already occupied\n", - codec_addr); - return -EBUSY; - } - codec = kzalloc(sizeof(*codec), GFP_KERNEL); if (!codec) return -ENOMEM; + codec->core.bus = &bus->core; + codec->core.addr = codec_addr; + codec->core.type = HDA_DEV_LEGACY; + dev = hda_codec_dev(codec); device_initialize(dev); - dev->parent = card->dev; + dev->parent = bus->core.dev; dev->bus = &snd_hda_bus_type; dev->release = snd_hda_codec_dev_release; dev->groups = snd_hda_dev_attr_groups; dev_set_name(dev, "hdaudioC%dD%d", card->number, codec_addr); dev_set_drvdata(dev, codec); /* for sysfs */ device_enable_async_suspend(dev); - codec->core.type = HDA_DEV_LEGACY; codec->bus = bus; codec->card = card; @@ -1323,7 +1261,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. * it's powered down later in snd_hda_codec_dev_register(). */ - set_bit(codec->addr, &bus->codec_powered); + set_bit(codec->core.addr, &bus->core.codec_powered); pm_runtime_set_active(hda_codec_dev(codec)); pm_runtime_get_noresume(hda_codec_dev(codec)); codec->power_jiffies = jiffies; @@ -1339,10 +1277,9 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, } } - list_add_tail(&codec->list, &bus->codec_list); - bus->num_codecs++; - - bus->caddr_tbl[codec_addr] = codec; + err = snd_hdac_bus_add_device(&bus->core, &codec->core); + if (err < 0) + goto error; codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_VENDOR_ID); @@ -1516,7 +1453,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, /* make other inactive cvts with the same stream-tag dirty */ type = get_wcaps_type(get_wcaps(codec, nid)); - list_for_each_entry(c, &codec->bus->codec_list, list) { + list_for_each_codec(c, codec->bus) { for (i = 0; i < c->cvt_setups.used; i++) { p = snd_array_elem(&c->cvt_setups, i); if (!p->active && p->stream_tag == stream_tag && @@ -1583,7 +1520,7 @@ static void purify_inactive_streams(struct hda_codec *codec) struct hda_codec *c; int i; - list_for_each_entry(c, &codec->bus->codec_list, list) { + list_for_each_codec(c, codec->bus) { for (i = 0; i < c->cvt_setups.used; i++) { struct hda_cvt_setup *p; p = snd_array_elem(&c->cvt_setups, i); @@ -2436,7 +2373,7 @@ int snd_hda_lock_devices(struct hda_bus *bus) if (!list_empty(&card->ctl_files)) goto err_clear; - list_for_each_entry(codec, &bus->codec_list, list) { + list_for_each_codec(codec, bus) { struct hda_pcm *cpcm; list_for_each_entry(cpcm, &codec->pcm_list_head, list) { if (!cpcm->pcm) @@ -3607,13 +3544,13 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, verb = verb | (parm >> 8); parm &= 0xff; key = build_cmd_cache_key(nid, verb); - mutex_lock(&codec->bus->cmd_mutex); + mutex_lock(&codec->bus->core.cmd_mutex); c = get_alloc_hash(&codec->cmd_cache, key); if (c) { c->val = parm; c->dirty = cache_only; } - mutex_unlock(&codec->bus->cmd_mutex); + mutex_unlock(&codec->bus->core.cmd_mutex); return 0; } EXPORT_SYMBOL_GPL(snd_hda_codec_write_cache); @@ -3642,13 +3579,13 @@ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, verb = verb | (parm >> 8); parm &= 0xff; key = build_cmd_cache_key(nid, verb); - mutex_lock(&codec->bus->cmd_mutex); + mutex_lock(&codec->bus->core.cmd_mutex); c = get_hash(&codec->cmd_cache, key); if (c && c->val == parm) { - mutex_unlock(&codec->bus->cmd_mutex); + mutex_unlock(&codec->bus->core.cmd_mutex); return 0; } - mutex_unlock(&codec->bus->cmd_mutex); + mutex_unlock(&codec->bus->core.cmd_mutex); return snd_hda_codec_write_cache(codec, nid, flags, verb, parm); } EXPORT_SYMBOL_GPL(snd_hda_codec_update_cache); @@ -3927,7 +3864,6 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); - trace_hda_power_down(codec); update_power_acct(codec, true); atomic_dec(&codec->in_pm); return state; @@ -3956,7 +3892,6 @@ static void hda_call_codec_resume(struct hda_codec *codec) { atomic_inc(&codec->in_pm); - trace_hda_power_up(codec); hda_mark_cmd_cache_dirty(codec); codec->power_jiffies = jiffies; @@ -3992,7 +3927,7 @@ static int hda_codec_runtime_suspend(struct device *dev) snd_pcm_suspend_all(pcm->pcm); state = hda_call_codec_suspend(codec); if (codec->d3_stop_clk && codec->epss && (state & AC_PWRST_CLK_STOP_OK)) - clear_bit(codec->addr, &codec->bus->codec_powered); + clear_bit(codec->core.addr, &codec->bus->core.codec_powered); return 0; } @@ -4000,7 +3935,7 @@ static int hda_codec_runtime_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - set_bit(codec->addr, &codec->bus->codec_powered); + set_bit(codec->core.addr, &codec->bus->core.codec_powered); hda_call_codec_resume(codec); pm_runtime_mark_last_busy(dev); return 0; @@ -4582,7 +4517,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) err = codec->patch_ops.build_pcms(codec); if (err < 0) { codec_err(codec, "cannot build PCMs for #%d (error %d)\n", - codec->addr, err); + codec->core.addr, err); return err; } @@ -4638,7 +4573,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) if (err < 0) { codec_err(codec, "cannot attach PCM stream %d for codec #%d\n", - dev, codec->addr); + dev, codec->core.addr); continue; /* no fatal error */ } } @@ -4681,8 +4616,8 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, * the codec addr; if it still fails (or it's the * primary codec), then try another control index */ - if (!addr && codec->addr) - addr = codec->addr; + if (!addr && codec->core.addr) + addr = codec->core.addr; else if (!idx && !knew->index) { idx = find_empty_mixer_ctl_idx(codec, knew->name, 0); @@ -4757,7 +4692,7 @@ void snd_hda_set_power_save(struct hda_bus *bus, int delay) { struct hda_codec *c; - list_for_each_entry(c, &bus->codec_list, list) + list_for_each_codec(c, bus) codec_set_power_save(c, delay); } EXPORT_SYMBOL_GPL(snd_hda_set_power_save); @@ -5344,7 +5279,7 @@ void snd_hda_bus_reset(struct hda_bus *bus) { struct hda_codec *codec; - list_for_each_entry(codec, &bus->codec_list, list) { + list_for_each_codec(codec, bus) { /* FIXME: maybe a better way needed for forced reset */ cancel_delayed_work_sync(&codec->jackpoll_work); #ifdef CONFIG_PM diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 31a9e10e5137ca..6efcb4ad69353a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -98,16 +98,6 @@ struct hda_bus_ops { #endif }; -/* unsolicited event handler */ -#define HDA_UNSOL_QUEUE_SIZE 64 -struct hda_bus_unsolicited { - /* ring buffer */ - u32 queue[HDA_UNSOL_QUEUE_SIZE * 2]; - unsigned int rp, wp; - /* workqueue */ - struct work_struct work; -}; - /* * codec bus * @@ -115,6 +105,8 @@ struct hda_bus_unsolicited { * A hda_bus contains several codecs in the list codec_list. */ struct hda_bus { + struct hdac_bus core; + struct snd_card *card; void *private_data; @@ -122,25 +114,14 @@ struct hda_bus { const char *modelname; struct hda_bus_ops ops; - /* codec linked list */ - struct list_head codec_list; - unsigned int num_codecs; - /* link caddr -> codec */ - struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; - - struct mutex cmd_mutex; struct mutex prepare_mutex; - /* unsolicited event queue */ - struct hda_bus_unsolicited unsol; - /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); /* misc op flags */ unsigned int needs_damn_long_delay :1; unsigned int allow_bus_reset:1; /* allow bus reset at fatal error */ - unsigned int sync_write:1; /* sync after verb write */ /* status for codec/controller */ unsigned int shutdown :1; /* being unloaded */ unsigned int rirb_error:1; /* error in codec communication */ @@ -149,7 +130,6 @@ struct hda_bus { unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ - unsigned long codec_powered; /* bit flags of powered codecs */ }; /* @@ -281,7 +261,6 @@ struct hda_codec { struct hda_bus *bus; struct snd_card *card; unsigned int addr; /* codec addr*/ - struct list_head list; /* list point */ hda_nid_t afg; /* AFG node id */ hda_nid_t mfg; /* MFG node id */ @@ -413,6 +392,9 @@ struct hda_codec { #define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, core.dev) #define hda_codec_dev(_dev) (&(_dev)->core.dev) +#define list_for_each_codec(c, bus) \ + list_for_each_entry(c, &(bus)->core.codec_list, core.list) + /* direction */ enum { HDA_INPUT, HDA_OUTPUT @@ -473,7 +455,11 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq); /* unsolicited event */ -int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); +static inline void +snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) +{ + snd_hdac_bus_queue_event(&bus->core, res, res_ex); +} /* cached write */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 4fd0b2ef26e9f4..26ce990592a04d 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1764,12 +1764,12 @@ static int probe_codec(struct azx *chip, int addr) (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; unsigned int res; - mutex_lock(&chip->bus->cmd_mutex); + mutex_lock(&chip->bus->core.cmd_mutex); chip->probing = 1; azx_send_cmd(chip->bus, cmd); res = azx_get_response(chip->bus, addr); chip->probing = 0; - mutex_unlock(&chip->bus->cmd_mutex); + mutex_unlock(&chip->bus->core.cmd_mutex); if (res == -1) return -EIO; dev_dbg(chip->card->dev, "codec #%d probed OK\n", addr); @@ -1848,7 +1848,7 @@ int azx_bus_create(struct azx *chip, const char *model) */ if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - bus->sync_write = 1; + bus->core.sync_write = 1; bus->allow_bus_reset = 1; } @@ -1913,7 +1913,7 @@ EXPORT_SYMBOL_GPL(azx_probe_codecs); int azx_codec_configure(struct azx *chip) { struct hda_codec *codec; - list_for_each_entry(codec, &chip->bus->codec_list, list) { + list_for_each_codec(codec, chip->bus) { snd_hda_codec_configure(codec); } return 0; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 060f7a2b1aeb4f..feebc1dda91270 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -891,7 +891,7 @@ static int azx_runtime_resume(struct device *dev) bus = chip->bus; if (status && bus) { - list_for_each_entry(codec, &bus->codec_list, list) + list_for_each_codec(codec, bus) if (status & (1 << codec->addr)) schedule_delayed_work(&codec->jackpoll_work, codec->jackpoll_interval); @@ -919,7 +919,7 @@ static int azx_runtime_idle(struct device *dev) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || - chip->bus->codec_powered) + chip->bus->core.codec_powered) return -EBUSY; return 0; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index e13c75d6784794..3b5ed1108f9f0d 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -552,7 +552,7 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus, *codecp = NULL; if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { - list_for_each_entry(codec, &bus->codec_list, list) { + list_for_each_codec(codec, bus) { if ((vendorid <= 0 || codec->vendor_id == vendorid) && (subid <= 0 || codec->subsystem_id == subid) && codec->addr == caddr) { diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5aa466a13e430c..142a6cf786dada 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -919,10 +919,10 @@ static int patch_conexant_auto(struct hda_codec *codec) * which falls into the single-cmd mode. * Better to make reset, then. */ - if (!codec->bus->sync_write) { + if (!codec->bus->core.sync_write) { codec_info(codec, "Enable sync_write for stable communication\n"); - codec->bus->sync_write = 1; + codec->bus->core.sync_write = 1; codec->bus->allow_bus_reset = 1; } From 7639a06c23c7d4cda34c2546bd7290d8753849ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Mar 2015 10:07:24 +0100 Subject: [PATCH 269/411] ALSA: hda - Move a part of hda_codec stuff into hdac_device Now some codes and functionalities of hda_codec struct are moved to hdac_device struct. A few basic attributes like the codec address, vendor ID number, FG numbers, etc are moved to hdac_device, and they are accessed like codec->core.addr. The basic verb exec functions are moved, too. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 66 +++++ sound/hda/Makefile | 2 +- sound/hda/hdac_device.c | 471 ++++++++++++++++++++++++++++++++ sound/hda/local.h | 19 ++ sound/pci/hda/hda_auto_parser.c | 33 ++- sound/pci/hda/hda_beep.c | 4 +- sound/pci/hda/hda_bind.c | 95 +------ sound/pci/hda/hda_codec.c | 396 ++++----------------------- sound/pci/hda/hda_codec.h | 43 +-- sound/pci/hda/hda_generic.c | 25 +- sound/pci/hda/hda_local.h | 15 +- sound/pci/hda/hda_proc.c | 42 +-- sound/pci/hda/hda_sysfs.c | 58 ++-- sound/pci/hda/local.h | 39 +++ sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0132.c | 6 +- sound/pci/hda/patch_conexant.c | 16 +- sound/pci/hda/patch_hdmi.c | 20 +- sound/pci/hda/patch_realtek.c | 78 +++--- sound/pci/hda/patch_si3054.c | 6 +- sound/pci/hda/patch_sigmatel.c | 84 +++--- sound/pci/hda/patch_via.c | 23 +- sound/pci/hda/thinkpad_helper.c | 2 +- 23 files changed, 880 insertions(+), 665 deletions(-) create mode 100644 sound/hda/hdac_device.c create mode 100644 sound/hda/local.h create mode 100644 sound/pci/hda/local.h diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 848ab6e6809958..b81b4bec6f050b 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -8,6 +8,9 @@ #include #include +/* codec node id */ +typedef u16 hda_nid_t; + struct hdac_bus; struct hdac_device; struct hdac_driver; @@ -26,6 +29,30 @@ struct hdac_device { struct hdac_bus *bus; unsigned int addr; /* codec address */ struct list_head list; /* list point for bus codec_list */ + + hda_nid_t afg; /* AFG node id */ + hda_nid_t mfg; /* MFG node id */ + + /* ids */ + unsigned int vendor_id; + unsigned int subsystem_id; + unsigned int revision_id; + unsigned int afg_function_id; + unsigned int mfg_function_id; + unsigned int afg_unsol:1; + unsigned int mfg_unsol:1; + + unsigned int power_caps; /* FG power caps */ + + const char *vendor_name; /* codec vendor name */ + const char *chip_name; /* codec chip name */ + + /* widgets */ + unsigned int num_nodes; + hda_nid_t start_nid, end_nid; + + /* misc flags */ + atomic_t in_pm; /* suspend/resume being performed */ }; /* device/driver type used for matching */ @@ -34,8 +61,37 @@ enum { HDA_DEV_LEGACY, }; +/* direction */ +enum { + HDA_INPUT, HDA_OUTPUT +}; + #define dev_to_hdac_dev(_dev) container_of(_dev, struct hdac_device, dev) +int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus, + const char *name, unsigned int addr); +void snd_hdac_device_exit(struct hdac_device *dev); + +int snd_hdac_refresh_widgets(struct hdac_device *codec); + +unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, + unsigned int verb, unsigned int parm); +int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid, + unsigned int verb, unsigned int parm, unsigned int *res); +int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); +int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid, + hda_nid_t *conn_list, int max_conns); +int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, + hda_nid_t *start_id); + +#ifdef CONFIG_PM +void snd_hdac_power_up(struct hdac_device *codec); +void snd_hdac_power_down(struct hdac_device *codec); +#else +static inline void snd_hdac_power_up(struct hdac_device *codec) {} +static inline void snd_hdac_power_down(struct hdac_device *codec) {} +#endif + /* * HD-audio codec base driver */ @@ -100,4 +156,14 @@ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec); void snd_hdac_bus_remove_device(struct hdac_bus *bus, struct hdac_device *codec); +static inline void snd_hdac_codec_link_up(struct hdac_device *codec) +{ + set_bit(codec->addr, &codec->bus->codec_powered); +} + +static inline void snd_hdac_codec_link_down(struct hdac_device *codec) +{ + clear_bit(codec->addr, &codec->bus->codec_powered); +} + #endif /* __SOUND_HDAUDIO_H */ diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 828680b282fab1..3c7625e595cf55 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -1,3 +1,3 @@ -snd-hda-core-objs := hda_bus_type.o hdac_bus.o +snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c new file mode 100644 index 00000000000000..a3f52ad4de3739 --- /dev/null +++ b/sound/hda/hdac_device.c @@ -0,0 +1,471 @@ +/* + * HD-audio codec core device + */ + +#include +#include +#include +#include +#include +#include +#include +#include "local.h" + +static void setup_fg_nodes(struct hdac_device *codec); +static int get_codec_vendor_name(struct hdac_device *codec); + +static void default_release(struct device *dev) +{ + snd_hdac_device_exit(container_of(dev, struct hdac_device, dev)); +} + +/** + * snd_hdac_device_init - initialize the HD-audio codec base device + * @codec: device to initialize + * @bus: but to attach + * @name: device name string + * @addr: codec address + * + * Returns zero for success or a negative error code. + * + * This function increments the runtime PM counter and marks it active. + * The caller needs to turn it off appropriately later. + * + * The caller needs to set the device's release op properly by itself. + */ +int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, + const char *name, unsigned int addr) +{ + struct device *dev; + hda_nid_t fg; + int err; + + dev = &codec->dev; + device_initialize(dev); + dev->parent = bus->dev; + dev->bus = &snd_hda_bus_type; + dev->release = default_release; + dev_set_name(dev, "%s", name); + device_enable_async_suspend(dev); + + codec->bus = bus; + codec->addr = addr; + codec->type = HDA_DEV_CORE; + pm_runtime_set_active(&codec->dev); + pm_runtime_get_noresume(&codec->dev); + atomic_set(&codec->in_pm, 0); + + err = snd_hdac_bus_add_device(bus, codec); + if (err < 0) + goto error; + + /* fill parameters */ + codec->vendor_id = snd_hdac_read_parm(codec, AC_NODE_ROOT, + AC_PAR_VENDOR_ID); + if (codec->vendor_id == -1) { + /* read again, hopefully the access method was corrected + * in the last read... + */ + codec->vendor_id = snd_hdac_read_parm(codec, AC_NODE_ROOT, + AC_PAR_VENDOR_ID); + } + + codec->subsystem_id = snd_hdac_read_parm(codec, AC_NODE_ROOT, + AC_PAR_SUBSYSTEM_ID); + codec->revision_id = snd_hdac_read_parm(codec, AC_NODE_ROOT, + AC_PAR_REV_ID); + + setup_fg_nodes(codec); + if (!codec->afg && !codec->mfg) { + dev_err(dev, "no AFG or MFG node found\n"); + err = -ENODEV; + goto error; + } + + fg = codec->afg ? codec->afg : codec->mfg; + + err = snd_hdac_refresh_widgets(codec); + if (err < 0) + goto error; + + codec->power_caps = snd_hdac_read_parm(codec, fg, AC_PAR_POWER_STATE); + /* reread ssid if not set by parameter */ + if (codec->subsystem_id == -1) + snd_hdac_read(codec, fg, AC_VERB_GET_SUBSYSTEM_ID, 0, + &codec->subsystem_id); + + err = get_codec_vendor_name(codec); + if (err < 0) + goto error; + + codec->chip_name = kasprintf(GFP_KERNEL, "ID %x", + codec->vendor_id & 0xffff); + if (!codec->chip_name) { + err = -ENOMEM; + goto error; + } + + return 0; + + error: + pm_runtime_put_noidle(&codec->dev); + put_device(&codec->dev); + return err; +} +EXPORT_SYMBOL_GPL(snd_hdac_device_init); + +/** + * snd_hdac_device_exit - clean up the HD-audio codec base device + * @codec: device to clean up + */ +void snd_hdac_device_exit(struct hdac_device *codec) +{ + /* pm_runtime_put_noidle(&codec->dev); */ + snd_hdac_bus_remove_device(codec->bus, codec); + kfree(codec->vendor_name); + kfree(codec->chip_name); +} +EXPORT_SYMBOL_GPL(snd_hdac_device_exit); + +/** + * snd_hdac_make_cmd - compose a 32bit command word to be sent to the + * HD-audio controller + * @codec: the codec object + * @nid: NID to encode + * @verb: verb to encode + * @parm: parameter to encode + * + * Return an encoded command verb or -1 for error. + */ +unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, + unsigned int verb, unsigned int parm) +{ + u32 val, addr; + + addr = codec->addr; + if ((addr & ~0xf) || (nid & ~0x7f) || + (verb & ~0xfff) || (parm & ~0xffff)) { + dev_err(&codec->dev, "out of range cmd %x:%x:%x:%x\n", + addr, nid, verb, parm); + return -1; + } + + val = addr << 28; + val |= (u32)nid << 20; + val |= verb << 8; + val |= parm; + return val; +} +EXPORT_SYMBOL_GPL(snd_hdac_make_cmd); + +/** + * snd_hdac_read - execute a verb + * @codec: the codec object + * @nid: NID to execute a verb + * @verb: verb to execute + * @parm: parameter for a verb + * @res: the pointer to store the result, NULL if running async + * + * Returns zero if successful, or a negative error code. + */ +int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid, + unsigned int verb, unsigned int parm, unsigned int *res) +{ + unsigned int cmd = snd_hdac_make_cmd(codec, nid, verb, parm); + + return snd_hdac_bus_exec_verb(codec->bus, codec->addr, cmd, res); +} +EXPORT_SYMBOL_GPL(snd_hdac_read); + +/** + * snd_hdac_read_parm - read a codec parameter + * @codec: the codec object + * @nid: NID to read a parameter + * @parm: parameter to read + * + * Returns -1 for error. If you need to distinguish the error more + * strictly, use snd_hdac_read() directly. + */ +int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm) +{ + int val; + + if (snd_hdac_read(codec, nid, AC_VERB_PARAMETERS, parm, &val)) + return -1; + return val; +} +EXPORT_SYMBOL_GPL(snd_hdac_read_parm); + +/** + * snd_hdac_get_sub_nodes - get start NID and number of subtree nodes + * @codec: the codec object + * @nid: NID to inspect + * @start_id: the pointer to store the starting NID + * + * Returns the number of subtree nodes or zero if not found. + */ +int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, + hda_nid_t *start_id) +{ + unsigned int parm; + + parm = snd_hdac_read_parm(codec, nid, AC_PAR_NODE_COUNT); + if (parm == -1) { + *start_id = 0; + return 0; + } + *start_id = (parm >> 16) & 0x7fff; + return (int)(parm & 0x7fff); +} +EXPORT_SYMBOL_GPL(snd_hdac_get_sub_nodes); + +/* + * look for an AFG and MFG nodes + */ +static void setup_fg_nodes(struct hdac_device *codec) +{ + int i, total_nodes, function_id; + hda_nid_t nid; + + total_nodes = snd_hdac_get_sub_nodes(codec, AC_NODE_ROOT, &nid); + for (i = 0; i < total_nodes; i++, nid++) { + function_id = snd_hdac_read_parm(codec, nid, + AC_PAR_FUNCTION_TYPE); + switch (function_id & 0xff) { + case AC_GRP_AUDIO_FUNCTION: + codec->afg = nid; + codec->afg_function_id = function_id & 0xff; + codec->afg_unsol = (function_id >> 8) & 1; + break; + case AC_GRP_MODEM_FUNCTION: + codec->mfg = nid; + codec->mfg_function_id = function_id & 0xff; + codec->mfg_unsol = (function_id >> 8) & 1; + break; + default: + break; + } + } +} + +/** + * snd_hdac_refresh_widgets - Reset the widget start/end nodes + * @codec: the codec object + */ +int snd_hdac_refresh_widgets(struct hdac_device *codec) +{ + hda_nid_t start_nid; + int nums; + + nums = snd_hdac_get_sub_nodes(codec, codec->afg, &start_nid); + if (!start_nid || nums <= 0 || nums >= 0xff) { + dev_err(&codec->dev, "cannot read sub nodes for FG 0x%02x\n", + codec->afg); + return -EINVAL; + } + + codec->num_nodes = nums; + codec->start_nid = start_nid; + codec->end_nid = start_nid + nums; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_refresh_widgets); + +/* return CONNLIST_LEN parameter of the given widget */ +static unsigned int get_num_conns(struct hdac_device *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!(wcaps & AC_WCAP_CONN_LIST) && + get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + return 0; + + parm = snd_hdac_read_parm(codec, nid, AC_PAR_CONNLIST_LEN); + if (parm == -1) + parm = 0; + return parm; +} + +/** + * snd_hdac_get_connections - get a widget connection list + * @codec: the codec object + * @nid: NID + * @conn_list: the array to store the results, can be NULL + * @max_conns: the max size of the given array + * + * Returns the number of connected widgets, zero for no connection, or a + * negative error code. When the number of elements don't fit with the + * given array size, it returns -ENOSPC. + * + * When @conn_list is NULL, it just checks the number of connections. + */ +int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid, + hda_nid_t *conn_list, int max_conns) +{ + unsigned int parm; + int i, conn_len, conns, err; + unsigned int shift, num_elems, mask; + hda_nid_t prev_nid; + int null_count = 0; + + parm = get_num_conns(codec, nid); + if (!parm) + return 0; + + if (parm & AC_CLIST_LONG) { + /* long form */ + shift = 16; + num_elems = 2; + } else { + /* short form */ + shift = 8; + num_elems = 4; + } + conn_len = parm & AC_CLIST_LENGTH; + mask = (1 << (shift-1)) - 1; + + if (!conn_len) + return 0; /* no connection */ + + if (conn_len == 1) { + /* single connection */ + err = snd_hdac_read(codec, nid, AC_VERB_GET_CONNECT_LIST, 0, + &parm); + if (err < 0) + return err; + if (conn_list) + conn_list[0] = parm & mask; + return 1; + } + + /* multi connection */ + conns = 0; + prev_nid = 0; + for (i = 0; i < conn_len; i++) { + int range_val; + hda_nid_t val, n; + + if (i % num_elems == 0) { + err = snd_hdac_read(codec, nid, + AC_VERB_GET_CONNECT_LIST, i, + &parm); + if (err < 0) + return -EIO; + } + range_val = !!(parm & (1 << (shift-1))); /* ranges */ + val = parm & mask; + if (val == 0 && null_count++) { /* no second chance */ + dev_dbg(&codec->dev, + "invalid CONNECT_LIST verb %x[%i]:%x\n", + nid, i, parm); + return 0; + } + parm >>= shift; + if (range_val) { + /* ranges between the previous and this one */ + if (!prev_nid || prev_nid >= val) { + dev_warn(&codec->dev, + "invalid dep_range_val %x:%x\n", + prev_nid, val); + continue; + } + for (n = prev_nid + 1; n <= val; n++) { + if (conn_list) { + if (conns >= max_conns) + return -ENOSPC; + conn_list[conns] = n; + } + conns++; + } + } else { + if (conn_list) { + if (conns >= max_conns) + return -ENOSPC; + conn_list[conns] = val; + } + conns++; + } + prev_nid = val; + } + return conns; +} +EXPORT_SYMBOL_GPL(snd_hdac_get_connections); + +#ifdef CONFIG_PM +/** + * snd_hdac_power_up - increment the runtime pm counter + * @codec: the codec object + */ +void snd_hdac_power_up(struct hdac_device *codec) +{ + struct device *dev = &codec->dev; + + if (atomic_read(&codec->in_pm)) + return; + pm_runtime_get_sync(dev); +} +EXPORT_SYMBOL_GPL(snd_hdac_power_up); + +/** + * snd_hdac_power_up - decrement the runtime pm counter + * @codec: the codec object + */ +void snd_hdac_power_down(struct hdac_device *codec) +{ + struct device *dev = &codec->dev; + + if (atomic_read(&codec->in_pm)) + return; + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); +} +EXPORT_SYMBOL_GPL(snd_hdac_power_down); +#endif + +/* codec vendor labels */ +struct hda_vendor_id { + unsigned int id; + const char *name; +}; + +static struct hda_vendor_id hda_vendor_ids[] = { + { 0x1002, "ATI" }, + { 0x1013, "Cirrus Logic" }, + { 0x1057, "Motorola" }, + { 0x1095, "Silicon Image" }, + { 0x10de, "Nvidia" }, + { 0x10ec, "Realtek" }, + { 0x1102, "Creative" }, + { 0x1106, "VIA" }, + { 0x111d, "IDT" }, + { 0x11c1, "LSI" }, + { 0x11d4, "Analog Devices" }, + { 0x13f6, "C-Media" }, + { 0x14f1, "Conexant" }, + { 0x17e8, "Chrontel" }, + { 0x1854, "LG" }, + { 0x1aec, "Wolfson Microelectronics" }, + { 0x1af4, "QEMU" }, + { 0x434d, "C-Media" }, + { 0x8086, "Intel" }, + { 0x8384, "SigmaTel" }, + {} /* terminator */ +}; + +/* store the codec vendor name */ +static int get_codec_vendor_name(struct hdac_device *codec) +{ + const struct hda_vendor_id *c; + u16 vendor_id = codec->vendor_id >> 16; + + for (c = hda_vendor_ids; c->id; c++) { + if (c->id == vendor_id) { + codec->vendor_name = kstrdup(c->name, GFP_KERNEL); + return codec->vendor_name ? 0 : -ENOMEM; + } + } + + codec->vendor_name = kasprintf(GFP_KERNEL, "Generic %04x", vendor_id); + return codec->vendor_name ? 0 : -ENOMEM; +} diff --git a/sound/hda/local.h b/sound/hda/local.h new file mode 100644 index 00000000000000..a077d1f656f6ec --- /dev/null +++ b/sound/hda/local.h @@ -0,0 +1,19 @@ +/* + * Local helper macros and functions for HD-audio core drivers + */ + +#ifndef __HDAC_LOCAL_H +#define __HDAC_LOCAL_H + +#define get_wcaps(codec, nid) \ + snd_hdac_read_parm(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) + +/* get the widget type from widget capability bits */ +static inline int get_wcaps_type(unsigned int wcaps) +{ + if (!wcaps) + return -1; /* invalid type */ + return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; +} + +#endif /* __HDAC_LOCAL_H */ diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 3f8706bb3d1634..03b7399bb7f00a 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -172,7 +172,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, const hda_nid_t *ignore_nids, unsigned int cond_flags) { - hda_nid_t nid, end_nid; + hda_nid_t nid; short seq, assoc_line_out; struct auto_out_pin line_out[ARRAY_SIZE(cfg->line_out_pins)]; struct auto_out_pin speaker_out[ARRAY_SIZE(cfg->speaker_pins)]; @@ -189,8 +189,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, memset(hp_out, 0, sizeof(hp_out)); assoc_line_out = 0; - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) { + for_each_hda_codec_node(nid, codec) { unsigned int wid_caps = get_wcaps(codec, nid); unsigned int wid_type = get_wcaps_type(wid_caps); unsigned int def_conf; @@ -410,7 +409,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, * debug prints of the parsed results */ codec_info(codec, "autoconfig for %s: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n", - codec->chip_name, cfg->line_outs, cfg->line_out_pins[0], + codec->core.chip_name, cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], cfg->line_out_pins[2], cfg->line_out_pins[3], cfg->line_out_pins[4], cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" : @@ -836,33 +835,33 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) if (action != HDA_FIXUP_ACT_PRE_PROBE || !fix->v.pins) break; codec_dbg(codec, "%s: Apply pincfg for %s\n", - codec->chip_name, modelname); + codec->core.chip_name, modelname); snd_hda_apply_pincfgs(codec, fix->v.pins); break; case HDA_FIXUP_VERBS: if (action != HDA_FIXUP_ACT_PROBE || !fix->v.verbs) break; codec_dbg(codec, "%s: Apply fix-verbs for %s\n", - codec->chip_name, modelname); + codec->core.chip_name, modelname); snd_hda_add_verbs(codec, fix->v.verbs); break; case HDA_FIXUP_FUNC: if (!fix->v.func) break; codec_dbg(codec, "%s: Apply fix-func for %s\n", - codec->chip_name, modelname); + codec->core.chip_name, modelname); fix->v.func(codec, fix, action); break; case HDA_FIXUP_PINCTLS: if (action != HDA_FIXUP_ACT_PROBE || !fix->v.pins) break; codec_dbg(codec, "%s: Apply pinctl for %s\n", - codec->chip_name, modelname); + codec->core.chip_name, modelname); set_pin_targets(codec, fix->v.pins); break; default: codec_err(codec, "%s: Invalid fixup type %d\n", - codec->chip_name, fix->type); + codec->core.chip_name, fix->type); break; } if (!fix->chained || fix->chained_before) @@ -912,16 +911,16 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, return; for (pq = pin_quirk; pq->subvendor; pq++) { - if ((codec->subsystem_id & 0xffff0000) != (pq->subvendor << 16)) + if ((codec->core.subsystem_id & 0xffff0000) != (pq->subvendor << 16)) continue; - if (codec->vendor_id != pq->codec) + if (codec->core.vendor_id != pq->codec) continue; if (pin_config_match(codec, pq->pins)) { codec->fixup_id = pq->value; #ifdef CONFIG_SND_DEBUG_VERBOSE codec->fixup_name = pq->name; codec_dbg(codec, "%s: picked fixup %s (pin match)\n", - codec->chip_name, codec->fixup_name); + codec->core.chip_name, codec->fixup_name); #endif codec->fixup_list = fixlist; return; @@ -963,7 +962,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, codec->fixup_name = NULL; codec->fixup_id = HDA_FIXUP_ID_NO_FIXUP; codec_dbg(codec, "%s: picked no fixup (nofixup specified)\n", - codec->chip_name); + codec->core.chip_name); return; } @@ -974,7 +973,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, codec->fixup_name = models->name; codec->fixup_list = fixlist; codec_dbg(codec, "%s: picked fixup %s (model specified)\n", - codec->chip_name, codec->fixup_name); + codec->core.chip_name, codec->fixup_name); return; } models++; @@ -987,7 +986,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, #ifdef CONFIG_SND_DEBUG_VERBOSE name = q->name; codec_dbg(codec, "%s: picked fixup %s (PCI SSID%s)\n", - codec->chip_name, name, q->subdevice_mask ? "" : " - vendor generic"); + codec->core.chip_name, name, q->subdevice_mask ? "" : " - vendor generic"); #endif } } @@ -996,12 +995,12 @@ void snd_hda_pick_fixup(struct hda_codec *codec, unsigned int vendorid = q->subdevice | (q->subvendor << 16); unsigned int mask = 0xffff0000 | q->subdevice_mask; - if ((codec->subsystem_id & mask) == (vendorid & mask)) { + if ((codec->core.subsystem_id & mask) == (vendorid & mask)) { id = q->value; #ifdef CONFIG_SND_DEBUG_VERBOSE name = q->name; codec_dbg(codec, "%s: picked fixup %s (codec SSID)\n", - codec->chip_name, name); + codec->core.chip_name, name); #endif break; } diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 4cdac3a71caea0..3364dc0fdeabb2 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -165,8 +165,8 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->id.bustype = BUS_PCI; input_dev->dev.parent = &codec->card->card_dev; - input_dev->id.vendor = codec->vendor_id >> 16; - input_dev->id.product = codec->vendor_id & 0xffff; + input_dev->id.vendor = codec->core.vendor_id >> 16; + input_dev->id.product = codec->core.vendor_id & 0xffff; input_dev->id.version = 0x01; input_dev->evbit[0] = BIT_MASK(EV_SND); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 0b9ea70c546b35..ad276a9771dbeb 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -14,36 +14,6 @@ #include "hda_codec.h" #include "hda_local.h" -/* codec vendor labels */ -struct hda_vendor_id { - unsigned int id; - const char *name; -}; - -static struct hda_vendor_id hda_vendor_ids[] = { - { 0x1002, "ATI" }, - { 0x1013, "Cirrus Logic" }, - { 0x1057, "Motorola" }, - { 0x1095, "Silicon Image" }, - { 0x10de, "Nvidia" }, - { 0x10ec, "Realtek" }, - { 0x1102, "Creative" }, - { 0x1106, "VIA" }, - { 0x111d, "IDT" }, - { 0x11c1, "LSI" }, - { 0x11d4, "Analog Devices" }, - { 0x13f6, "C-Media" }, - { 0x14f1, "Conexant" }, - { 0x17e8, "Chrontel" }, - { 0x1854, "LG" }, - { 0x1aec, "Wolfson Microelectronics" }, - { 0x1af4, "QEMU" }, - { 0x434d, "C-Media" }, - { 0x8086, "Intel" }, - { 0x8384, "SigmaTel" }, - {} /* terminator */ -}; - /* * find a matching codec preset */ @@ -54,19 +24,19 @@ static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv) container_of(drv, struct hda_codec_driver, core); const struct hda_codec_preset *preset; /* check probe_id instead of vendor_id if set */ - u32 id = codec->probe_id ? codec->probe_id : codec->vendor_id; + u32 id = codec->probe_id ? codec->probe_id : codec->core.vendor_id; for (preset = driver->preset; preset->id; preset++) { u32 mask = preset->mask; - if (preset->afg && preset->afg != codec->afg) + if (preset->afg && preset->afg != codec->core.afg) continue; - if (preset->mfg && preset->mfg != codec->mfg) + if (preset->mfg && preset->mfg != codec->core.mfg) continue; if (!mask) mask = ~0; if (preset->id == (id & mask) && - (!preset->rev || preset->rev == codec->revision_id)) { + (!preset->rev || preset->rev == codec->core.revision_id)) { codec->preset = preset; return 1; } @@ -86,15 +56,11 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) /* reset the codec name from the preset */ static int codec_refresh_name(struct hda_codec *codec, const char *name) { - char tmp[16]; - - kfree(codec->chip_name); - if (!name) { - sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); - name = tmp; + if (name) { + kfree(codec->core.chip_name); + codec->core.chip_name = kstrdup(name, GFP_KERNEL); } - codec->chip_name = kstrdup(name, GFP_KERNEL); - return codec->chip_name ? 0 : -ENOMEM; + return codec->core.chip_name ? 0 : -ENOMEM; } static int hda_codec_driver_probe(struct device *dev) @@ -192,48 +158,23 @@ static inline bool codec_probed(struct hda_codec *codec) static void codec_bind_module(struct hda_codec *codec) { #ifdef MODULE - request_module("snd-hda-codec-id:%08x", codec->vendor_id); + request_module("snd-hda-codec-id:%08x", codec->core.vendor_id); if (codec_probed(codec)) return; request_module("snd-hda-codec-id:%04x*", - (codec->vendor_id >> 16) & 0xffff); + (codec->core.vendor_id >> 16) & 0xffff); if (codec_probed(codec)) return; #endif } -/* store the codec vendor name */ -static int get_codec_vendor_name(struct hda_codec *codec) -{ - const struct hda_vendor_id *c; - const char *vendor = NULL; - u16 vendor_id = codec->vendor_id >> 16; - char tmp[16]; - - for (c = hda_vendor_ids; c->id; c++) { - if (c->id == vendor_id) { - vendor = c->name; - break; - } - } - if (!vendor) { - sprintf(tmp, "Generic %04x", vendor_id); - vendor = tmp; - } - codec->vendor_name = kstrdup(vendor, GFP_KERNEL); - if (!codec->vendor_name) - return -ENOMEM; - return 0; -} - #if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) /* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ static bool is_likely_hdmi_codec(struct hda_codec *codec) { - hda_nid_t nid = codec->start_nid; - int i; + hda_nid_t nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { + for_each_hda_codec_node(nid, codec) { unsigned int wcaps = get_wcaps(codec, nid); switch (get_wcaps_type(wcaps)) { case AC_WID_AUD_IN: @@ -294,12 +235,6 @@ int snd_hda_codec_configure(struct hda_codec *codec) { int err; - if (!codec->vendor_name) { - err = get_codec_vendor_name(codec); - if (err < 0) - return err; - } - if (is_generic_config(codec)) codec->probe_id = HDA_CODEC_ID_GENERIC; else @@ -320,10 +255,10 @@ int snd_hda_codec_configure(struct hda_codec *codec) } /* audio codec should override the mixer name */ - if (codec->afg || !*codec->card->mixername) + if (codec->core.afg || !*codec->card->mixername) snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->vendor_name, codec->chip_name); + sizeof(codec->card->mixername), "%s %s", + codec->core.vendor_name, codec->core.chip_name); return 0; error: diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f96bff37c78719..ddfc0fbbee2324 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -40,7 +40,7 @@ #include #ifdef CONFIG_PM -#define codec_in_pm(codec) atomic_read(&(codec)->in_pm) +#define codec_in_pm(codec) atomic_read(&(codec)->core.in_pm) #define hda_codec_is_power_on(codec) \ (!pm_runtime_suspended(hda_codec_dev(codec))) #else @@ -48,6 +48,11 @@ #define hda_codec_is_power_on(codec) 1 #endif +#define codec_has_epss(codec) \ + ((codec)->core.power_caps & AC_PWRST_EPSS) +#define codec_has_clkstop(codec) \ + ((codec)->core.power_caps & AC_PWRST_CLKSTOP) + /** * snd_hda_get_jack_location - Give a location string of the jack * @cfg: pin default config value @@ -118,30 +123,6 @@ const char *snd_hda_get_jack_type(u32 cfg) } EXPORT_SYMBOL_GPL(snd_hda_get_jack_type); -/* - * Compose a 32bit command word to be sent to the HD-audio controller - */ -static inline unsigned int -make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int flags, - unsigned int verb, unsigned int parm) -{ - unsigned int addr = codec->core.addr; - u32 val; - - if ((addr & ~0xf) || (nid & ~0x7f) || - (verb & ~0xfff) || (parm & ~0xffff)) { - codec_err(codec, "hda-codec: out of range cmd %x:%x:%x:%x\n", - addr, nid, verb, parm); - return ~0; - } - - val = (u32)addr << 28; - val |= (u32)nid << 20; - val |= verb << 8; - val |= parm; - return val; -} - /* * Send and receive a verb */ @@ -194,7 +175,7 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { - unsigned cmd = make_codec_cmd(codec, nid, flags, verb, parm); + unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm); unsigned int res; if (codec_exec_verb(codec, cmd, flags, &res)) return -1; @@ -217,7 +198,7 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_read); int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { - unsigned int cmd = make_codec_cmd(codec, nid, flags, verb, parm); + unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm); return codec_exec_verb(codec, cmd, flags, NULL); } EXPORT_SYMBOL_GPL(snd_hda_codec_write); @@ -237,30 +218,6 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq) } EXPORT_SYMBOL_GPL(snd_hda_sequence_write); -/** - * snd_hda_get_sub_nodes - get the range of sub nodes - * @codec: the HDA codec - * @nid: NID to parse - * @start_id: the pointer to store the start NID - * - * Parse the NID and store the start NID of its sub-nodes. - * Returns the number of sub-nodes. - */ -int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *start_id) -{ - unsigned int parm; - - parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT); - if (parm == -1) { - *start_id = 0; - return 0; - } - *start_id = (parm >> 16) & 0x7fff; - return (int)(parm & 0x7fff); -} -EXPORT_SYMBOL_GPL(snd_hda_get_sub_nodes); - /* connection list element */ struct hda_conn_list { struct list_head list; @@ -401,128 +358,6 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hda_get_connections); -/* return CONNLIST_LEN parameter of the given widget */ -static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int wcaps = get_wcaps(codec, nid); - unsigned int parm; - - if (!(wcaps & AC_WCAP_CONN_LIST) && - get_wcaps_type(wcaps) != AC_WID_VOL_KNB) - return 0; - - parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); - if (parm == -1) - parm = 0; - return parm; -} - -int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid) -{ - return snd_hda_get_raw_connections(codec, nid, NULL, 0); -} - -/** - * snd_hda_get_raw_connections - copy connection list without cache - * @codec: the HDA codec - * @nid: NID to parse - * @conn_list: connection list array - * @max_conns: max. number of connections to store - * - * Like snd_hda_get_connections(), copy the connection list but without - * checking through the connection-list cache. - * Currently called only from hda_proc.c, so not exported. - */ -int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns) -{ - unsigned int parm; - int i, conn_len, conns; - unsigned int shift, num_elems, mask; - hda_nid_t prev_nid; - int null_count = 0; - - parm = get_num_conns(codec, nid); - if (!parm) - return 0; - - if (parm & AC_CLIST_LONG) { - /* long form */ - shift = 16; - num_elems = 2; - } else { - /* short form */ - shift = 8; - num_elems = 4; - } - conn_len = parm & AC_CLIST_LENGTH; - mask = (1 << (shift-1)) - 1; - - if (!conn_len) - return 0; /* no connection */ - - if (conn_len == 1) { - /* single connection */ - parm = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, 0); - if (parm == -1 && codec->bus->rirb_error) - return -EIO; - if (conn_list) - conn_list[0] = parm & mask; - return 1; - } - - /* multi connection */ - conns = 0; - prev_nid = 0; - for (i = 0; i < conn_len; i++) { - int range_val; - hda_nid_t val, n; - - if (i % num_elems == 0) { - parm = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, i); - if (parm == -1 && codec->bus->rirb_error) - return -EIO; - } - range_val = !!(parm & (1 << (shift-1))); /* ranges */ - val = parm & mask; - if (val == 0 && null_count++) { /* no second chance */ - codec_dbg(codec, - "invalid CONNECT_LIST verb %x[%i]:%x\n", - nid, i, parm); - return 0; - } - parm >>= shift; - if (range_val) { - /* ranges between the previous and this one */ - if (!prev_nid || prev_nid >= val) { - codec_warn(codec, - "invalid dep_range_val %x:%x\n", - prev_nid, val); - continue; - } - for (n = prev_nid + 1; n <= val; n++) { - if (conn_list) { - if (conns >= max_conns) - return -ENOSPC; - conn_list[conns] = n; - } - conns++; - } - } else { - if (conn_list) { - if (conns >= max_conns) - return -ENOSPC; - conn_list[conns] = val; - } - conns++; - } - prev_nid = val; - } - return conns; -} - /** * snd_hda_override_conn_list - add/modify the connection-list to cache * @codec: the HDA codec @@ -737,35 +572,6 @@ int snd_hda_bus_new(struct snd_card *card, } EXPORT_SYMBOL_GPL(snd_hda_bus_new); -/* - * look for an AFG and MFG nodes - */ -static void setup_fg_nodes(struct hda_codec *codec) -{ - int i, total_nodes, function_id; - hda_nid_t nid; - - total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); - for (i = 0; i < total_nodes; i++, nid++) { - function_id = snd_hda_param_read(codec, nid, - AC_PAR_FUNCTION_TYPE); - switch (function_id & 0xff) { - case AC_GRP_AUDIO_FUNCTION: - codec->afg = nid; - codec->afg_function_id = function_id & 0xff; - codec->afg_unsol = (function_id >> 8) & 1; - break; - case AC_GRP_MODEM_FUNCTION: - codec->mfg = nid; - codec->mfg_function_id = function_id & 0xff; - codec->mfg_unsol = (function_id >> 8) & 1; - break; - default: - break; - } - } -} - /* * read widget caps for each widget and store in cache */ @@ -774,13 +580,11 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) int i; hda_nid_t nid; - codec->num_nodes = snd_hda_get_sub_nodes(codec, fg_node, - &codec->start_nid); - codec->wcaps = kmalloc(codec->num_nodes * 4, GFP_KERNEL); + codec->wcaps = kmalloc(codec->core.num_nodes * 4, GFP_KERNEL); if (!codec->wcaps) return -ENOMEM; - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) + nid = codec->core.start_nid; + for (i = 0; i < codec->core.num_nodes; i++, nid++) codec->wcaps[i] = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); return 0; @@ -789,10 +593,9 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) /* read all pin default configurations and save codec->init_pins */ static int read_pin_defaults(struct hda_codec *codec) { - int i; - hda_nid_t nid = codec->start_nid; + hda_nid_t nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { + for_each_hda_codec_node(nid, codec) { struct hda_pincfg *pin; unsigned int wcaps = get_wcaps(codec, nid); unsigned int wid_type = get_wcaps_type(wcaps); @@ -1136,9 +939,6 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) remove_conn_list(codec); } -static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, - hda_nid_t fg, unsigned int power_state); - static unsigned int hda_set_power_state(struct hda_codec *codec, unsigned int power_state); @@ -1178,12 +978,10 @@ static void snd_hda_codec_dev_release(struct device *dev) struct hda_codec *codec = dev_to_hda_codec(dev); free_init_pincfgs(codec); - snd_hdac_bus_remove_device(&codec->bus->core, &codec->core); + snd_hdac_device_exit(&codec->core); snd_hda_sysfs_clear(codec); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); - kfree(codec->vendor_name); - kfree(codec->chip_name); kfree(codec->modelname); kfree(codec->wcaps); kfree(codec); @@ -1201,7 +999,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp) { struct hda_codec *codec; - struct device *dev; char component[31]; hda_nid_t fg; int err; @@ -1220,19 +1017,16 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (!codec) return -ENOMEM; - codec->core.bus = &bus->core; - codec->core.addr = codec_addr; - codec->core.type = HDA_DEV_LEGACY; + sprintf(component, "hdaudioC%dD%d", card->number, codec_addr); + err = snd_hdac_device_init(&codec->core, &bus->core, component, + codec_addr); + if (err < 0) { + kfree(codec); + return err; + } - dev = hda_codec_dev(codec); - device_initialize(dev); - dev->parent = bus->core.dev; - dev->bus = &snd_hda_bus_type; - dev->release = snd_hda_codec_dev_release; - dev->groups = snd_hda_dev_attr_groups; - dev_set_name(dev, "hdaudioC%dD%d", card->number, codec_addr); - dev_set_drvdata(dev, codec); /* for sysfs */ - device_enable_async_suspend(dev); + codec->core.dev.release = snd_hda_codec_dev_release; + codec->core.type = HDA_DEV_LEGACY; codec->bus = bus; codec->card = card; @@ -1258,12 +1052,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM - /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. - * it's powered down later in snd_hda_codec_dev_register(). - */ - set_bit(codec->core.addr, &bus->core.codec_powered); - pm_runtime_set_active(hda_codec_dev(codec)); - pm_runtime_get_noresume(hda_codec_dev(codec)); codec->power_jiffies = jiffies; #endif @@ -1277,31 +1065,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, } } - err = snd_hdac_bus_add_device(&bus->core, &codec->core); - if (err < 0) - goto error; - - codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, - AC_PAR_VENDOR_ID); - if (codec->vendor_id == -1) - /* read again, hopefully the access method was corrected - * in the last read... - */ - codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, - AC_PAR_VENDOR_ID); - codec->subsystem_id = snd_hda_param_read(codec, AC_NODE_ROOT, - AC_PAR_SUBSYSTEM_ID); - codec->revision_id = snd_hda_param_read(codec, AC_NODE_ROOT, - AC_PAR_REV_ID); - - setup_fg_nodes(codec); - if (!codec->afg && !codec->mfg) { - codec_err(codec, "no AFG or MFG node found\n"); - err = -ENODEV; - goto error; - } - - fg = codec->afg ? codec->afg : codec->mfg; + fg = codec->core.afg ? codec->core.afg : codec->core.mfg; err = read_widget_caps(codec, fg); if (err < 0) goto error; @@ -1309,19 +1073,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, if (err < 0) goto error; - if (!codec->subsystem_id) { - codec->subsystem_id = - snd_hda_codec_read(codec, fg, 0, - AC_VERB_GET_SUBSYSTEM_ID, 0); - } - -#ifdef CONFIG_PM - codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, fg, - AC_PWRST_CLKSTOP); -#endif - codec->epss = snd_hda_codec_get_supported_ps(codec, fg, - AC_PWRST_EPSS); - /* power-up all before initialization */ hda_set_power_state(codec, AC_PWRST_D0); @@ -1329,8 +1080,8 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, snd_hda_create_hwdep(codec); - sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, - codec->subsystem_id, codec->revision_id); + sprintf(component, "HDA:%08x,%08x,%08x", codec->core.vendor_id, + codec->core.subsystem_id, codec->core.revision_id); snd_component_add(card, component); err = snd_device_new(card, SNDRV_DEV_CODEC, codec, &dev_ops); @@ -1342,6 +1093,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, return 0; error: + pm_runtime_put_noidle(hda_codec_dev(codec)); put_device(hda_codec_dev(codec)); return err; } @@ -1359,11 +1111,15 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) hda_nid_t fg; int err; + err = snd_hdac_refresh_widgets(&codec->core); + if (err < 0) + return err; + /* Assume the function group node does not change, * only the widget nodes may change. */ kfree(codec->wcaps); - fg = codec->afg ? codec->afg : codec->mfg; + fg = codec->core.afg ? codec->core.afg : codec->core.mfg; err = read_widget_caps(codec, fg); if (err < 0) return err; @@ -1663,7 +1419,7 @@ static unsigned int read_amp_cap(struct hda_codec *codec, hda_nid_t nid, int direction) { if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) - nid = codec->afg; + nid = codec->core.afg; return snd_hda_param_read(codec, nid, direction == HDA_OUTPUT ? AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); @@ -3664,10 +3420,9 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_flush_cache); void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { - hda_nid_t nid = codec->start_nid; - int i; + hda_nid_t nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { + for_each_hda_codec_node(nid, codec) { unsigned int wcaps = get_wcaps(codec, nid); unsigned int state = power_state; if (!(wcaps & AC_WCAP_POWER)) @@ -3683,22 +3438,6 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, } EXPORT_SYMBOL_GPL(snd_hda_codec_set_power_to_all); -/* - * supported power states check - */ -static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) -{ - int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - - if (sup == -1) - return false; - if (sup & power_state) - return true; - else - return false; -} - /* * wait until the state is reached, returns the current state */ @@ -3738,7 +3477,7 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) { - if (nid == codec->afg || nid == codec->mfg) + if (nid == codec->core.afg || nid == codec->core.mfg) return power_state; if (power_state == AC_PWRST_D3 && get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_PIN && @@ -3758,7 +3497,7 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_eapd_power_filter); static unsigned int hda_set_power_state(struct hda_codec *codec, unsigned int power_state) { - hda_nid_t fg = codec->afg ? codec->afg : codec->mfg; + hda_nid_t fg = codec->core.afg ? codec->core.afg : codec->core.mfg; int count; unsigned int state; int flags = 0; @@ -3766,7 +3505,7 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { if (codec->depop_delay < 0) - msleep(codec->epss ? 10 : 100); + msleep(codec_has_epss(codec) ? 10 : 100); else if (codec->depop_delay > 0) msleep(codec->depop_delay); flags = HDA_RW_NO_RESPONSE_FALLBACK; @@ -3800,14 +3539,13 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, */ static void sync_power_up_states(struct hda_codec *codec) { - hda_nid_t nid = codec->start_nid; - int i; + hda_nid_t nid; /* don't care if no filter is used */ if (!codec->power_filter) return; - for (i = 0; i < codec->num_nodes; i++, nid++) { + for_each_hda_codec_node(nid, codec) { unsigned int wcaps = get_wcaps(codec, nid); unsigned int target; if (!(wcaps & AC_WCAP_POWER)) @@ -3858,14 +3596,14 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) { unsigned int state; - atomic_inc(&codec->in_pm); + atomic_inc(&codec->core.in_pm); if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); update_power_acct(codec, true); - atomic_dec(&codec->in_pm); + atomic_dec(&codec->core.in_pm); return state; } @@ -3890,7 +3628,7 @@ static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { - atomic_inc(&codec->in_pm); + atomic_inc(&codec->core.in_pm); hda_mark_cmd_cache_dirty(codec); @@ -3913,7 +3651,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_jackpoll_work(&codec->jackpoll_work.work); else snd_hda_jack_report_sync(codec); - atomic_dec(&codec->in_pm); + atomic_dec(&codec->core.in_pm); } static int hda_codec_runtime_suspend(struct device *dev) @@ -3926,8 +3664,9 @@ static int hda_codec_runtime_suspend(struct device *dev) list_for_each_entry(pcm, &codec->pcm_list_head, list) snd_pcm_suspend_all(pcm->pcm); state = hda_call_codec_suspend(codec); - if (codec->d3_stop_clk && codec->epss && (state & AC_PWRST_CLK_STOP_OK)) - clear_bit(codec->core.addr, &codec->bus->core.codec_powered); + if (codec_has_clkstop(codec) && codec_has_epss(codec) && + (state & AC_PWRST_CLK_STOP_OK)) + snd_hdac_codec_link_down(&codec->core); return 0; } @@ -3935,7 +3674,7 @@ static int hda_codec_runtime_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - set_bit(codec->core.addr, &codec->bus->core.codec_powered); + snd_hdac_codec_link_up(&codec->core); hda_call_codec_resume(codec); pm_runtime_mark_last_busy(dev); return 0; @@ -4129,11 +3868,11 @@ static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid, int dir) { unsigned int val = 0; - if (nid != codec->afg && + if (nid != codec->core.afg && (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) val = snd_hda_param_read(codec, nid, AC_PAR_PCM); if (!val || val == -1) - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); + val = snd_hda_param_read(codec, codec->core.afg, AC_PAR_PCM); if (!val || val == -1) return 0; return val; @@ -4150,7 +3889,7 @@ static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid, { unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (!streams || streams == -1) - streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); + streams = snd_hda_param_read(codec, codec->core.afg, AC_PAR_STREAM); if (!streams || streams == -1) return 0; return streams; @@ -4632,39 +4371,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls); #ifdef CONFIG_PM -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the usage counter and resume the device if not done yet. - */ -void snd_hda_power_up(struct hda_codec *codec) -{ - struct device *dev = hda_codec_dev(codec); - - if (codec_in_pm(codec)) - return; - pm_runtime_get_sync(dev); -} -EXPORT_SYMBOL_GPL(snd_hda_power_up); - -/** - * snd_hda_power_down - Power-down the codec - * @codec: HD-audio codec - * - * Decrement the usage counter and schedules the autosuspend if none used. - */ -void snd_hda_power_down(struct hda_codec *codec) -{ - struct device *dev = hda_codec_dev(codec); - - if (codec_in_pm(codec)) - return; - pm_runtime_mark_last_busy(dev); - pm_runtime_put_autosuspend(dev); -} -EXPORT_SYMBOL_GPL(snd_hda_power_down); - static void codec_set_power_save(struct hda_codec *codec, int delay) { struct device *dev = hda_codec_dev(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6efcb4ad69353a..e7c47a43976280 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -261,24 +261,10 @@ struct hda_codec { struct hda_bus *bus; struct snd_card *card; unsigned int addr; /* codec addr*/ - - hda_nid_t afg; /* AFG node id */ - hda_nid_t mfg; /* MFG node id */ - - /* ids */ - u8 afg_function_id; - u8 mfg_function_id; - u8 afg_unsol; - u8 mfg_unsol; - u32 vendor_id; - u32 subsystem_id; - u32 revision_id; u32 probe_id; /* overridden id for probing */ /* detected preset */ const struct hda_codec_preset *preset; - const char *vendor_name; /* codec vendor name */ - const char *chip_name; /* codec chip name */ const char *modelname; /* model name for preset */ /* set by patch */ @@ -295,8 +281,6 @@ struct hda_codec { unsigned int beep_mode; /* widget capabilities cache */ - unsigned int num_nodes; - hda_nid_t start_nid; u32 *wcaps; struct snd_array mixers; /* list of assigned mixer elements */ @@ -347,14 +331,11 @@ struct hda_codec { unsigned int inv_eapd:1; /* broken h/w: inverted EAPD control */ unsigned int inv_jack_detect:1; /* broken h/w: inverted detection bit */ unsigned int pcm_format_first:1; /* PCM format must be set first */ - unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ unsigned int power_save_node:1; /* advanced PM for each widget */ #ifdef CONFIG_PM - unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - atomic_t in_pm; /* suspend/resume being performed */ unsigned long power_on_acct; unsigned long power_off_acct; unsigned long power_jiffies; @@ -395,11 +376,6 @@ struct hda_codec { #define list_for_each_codec(c, bus) \ list_for_each_entry(c, &(bus)->core.codec_list, core.list) -/* direction */ -enum { - HDA_INPUT, HDA_OUTPUT -}; - /* snd_hda_codec_read/write optional flags */ #define HDA_RW_NO_RESPONSE_FALLBACK (1 << 0) @@ -422,8 +398,8 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm); #define snd_hda_param_read(codec, nid, param) \ snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param) -int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *start_id); +#define snd_hda_get_sub_nodes(codec, nid, start_nid) \ + snd_hdac_get_sub_nodes(&(codec)->core, nid, start_nid) int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); static inline int @@ -431,9 +407,12 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) { return snd_hda_get_connections(codec, nid, NULL, 0); } -int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns); + +#define snd_hda_get_raw_connections(codec, nid, list, max_conns) \ + snd_hdac_get_connections(&(codec)->core, nid, list, max_conns) +#define snd_hda_get_num_raw_conns(codec, nid) \ + snd_hdac_get_connections(&(codec)->core, nid, NULL, 0); + int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, const hda_nid_t **listp); int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, @@ -582,14 +561,12 @@ const char *snd_hda_get_jack_location(u32 cfg); /* * power saving */ +#define snd_hda_power_up(codec) snd_hdac_power_up(&(codec)->core) +#define snd_hda_power_down(codec) snd_hdac_power_down(&(codec)->core) #ifdef CONFIG_PM -void snd_hda_power_up(struct hda_codec *codec); -void snd_hda_power_down(struct hda_codec *codec); void snd_hda_set_power_save(struct hda_bus *bus, int delay); void snd_hda_update_power_acct(struct hda_codec *codec); #else -static inline void snd_hda_power_up(struct hda_codec *codec) {} -static inline void snd_hda_power_down(struct hda_codec *codec) {} static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 0ef2459cd05fd6..4850f92c89c4c7 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -654,7 +654,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, int type = get_wcaps_type(get_wcaps(codec, nid)); int i, n; - if (nid == codec->afg) + if (nid == codec->core.afg) return true; for (n = 0; n < spec->paths.used; n++) { @@ -832,7 +832,7 @@ static hda_nid_t path_power_update(struct hda_codec *codec, for (i = 0; i < path->depth; i++) { nid = path->path[i]; - if (nid == codec->afg) + if (nid == codec->core.afg) continue; if (!allow_powerdown || is_active_nid_for_any(codec, nid)) state = AC_PWRST_D0; @@ -1897,12 +1897,11 @@ static void debug_show_configs(struct hda_codec *codec, static void fill_all_dac_nids(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; - int i; - hda_nid_t nid = codec->start_nid; + hda_nid_t nid; spec->num_all_dacs = 0; memset(spec->all_dacs, 0, sizeof(spec->all_dacs)); - for (i = 0; i < codec->num_nodes; i++, nid++) { + for_each_hda_codec_node(nid, codec) { if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_AUD_OUT) continue; if (spec->num_all_dacs >= ARRAY_SIZE(spec->all_dacs)) { @@ -3067,10 +3066,9 @@ static int fill_adc_nids(struct hda_codec *codec) hda_nid_t nid; hda_nid_t *adc_nids = spec->adc_nids; int max_nums = ARRAY_SIZE(spec->adc_nids); - int i, nums = 0; + int nums = 0; - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { + for_each_hda_codec_node(nid, codec) { unsigned int caps = get_wcaps(codec, nid); int type = get_wcaps_type(caps); @@ -3864,8 +3862,7 @@ static void parse_digital(struct hda_codec *codec) if (spec->autocfg.dig_in_pin) { pin = spec->autocfg.dig_in_pin; - dig_nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, dig_nid++) { + for_each_hda_codec_node(dig_nid, codec) { unsigned int wcaps = get_wcaps(codec, dig_nid); if (get_wcaps_type(wcaps) != AC_WID_AUD_IN) continue; @@ -4706,7 +4703,7 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) { - if (power_state != AC_PWRST_D0 || nid == codec->afg) + if (power_state != AC_PWRST_D0 || nid == codec->core.afg) return power_state; if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER) return power_state; @@ -5478,7 +5475,7 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) fill_pcm_stream_name(spec->stream_name_analog, sizeof(spec->stream_name_analog), - " Analog", codec->chip_name); + " Analog", codec->core.chip_name); info = snd_hda_codec_pcm_new(codec, "%s", spec->stream_name_analog); if (!info) return -ENOMEM; @@ -5509,7 +5506,7 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) if (spec->multiout.dig_out_nid || spec->dig_in_nid) { fill_pcm_stream_name(spec->stream_name_digital, sizeof(spec->stream_name_digital), - " Digital", codec->chip_name); + " Digital", codec->core.chip_name); info = snd_hda_codec_pcm_new(codec, "%s", spec->stream_name_digital); if (!info) @@ -5544,7 +5541,7 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) if (spec->alt_dac_nid || have_multi_adcs) { fill_pcm_stream_name(spec->stream_name_alt_analog, sizeof(spec->stream_name_alt_analog), - " Alt Analog", codec->chip_name); + " Alt Analog", codec->core.chip_name); info = snd_hda_codec_pcm_new(codec, "%s", spec->stream_name_alt_analog); if (!info) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1d001647fc47bb..e0db30c66e5fcc 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -515,15 +515,18 @@ int snd_hda_codec_get_pin_target(struct hda_codec *codec, hda_nid_t nid); int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid, unsigned int val); +#define for_each_hda_codec_node(nid, codec) \ + for ((nid) = (codec)->core.start_nid; (nid) < (codec)->core.end_nid; (nid)++) + /* * get widget capabilities */ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) { - if (nid < codec->start_nid || - nid >= codec->start_nid + codec->num_nodes) + if (nid < codec->core.start_nid || + nid >= codec->core.start_nid + codec->core.num_nodes) return 0; - return codec->wcaps[nid - codec->start_nid]; + return codec->wcaps[nid - codec->core.start_nid]; } /* get the widget type from widget capability bits */ @@ -547,9 +550,9 @@ static inline unsigned int get_wcaps_channels(u32 wcaps) static inline void snd_hda_override_wcaps(struct hda_codec *codec, hda_nid_t nid, u32 val) { - if (nid >= codec->start_nid && - nid < codec->start_nid + codec->num_nodes) - codec->wcaps[nid - codec->start_nid] = val; + if (nid >= codec->core.start_nid && + nid < codec->core.start_nid + codec->core.num_nodes) + codec->wcaps[nid - codec->core.start_nid] = val; } u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index dacfe74a2a1fef..a4f5a30f1d41f2 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -289,7 +289,7 @@ static void print_pin_caps(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Balanced"); if (caps & AC_PINCAP_HDMI) { /* Realtek uses this bit as a different meaning */ - if ((codec->vendor_id >> 16) == 0x10ec) + if ((codec->core.vendor_id >> 16) == 0x10ec) snd_iprintf(buffer, " R/L"); else { if (caps & AC_PINCAP_HBR) @@ -597,7 +597,7 @@ static void print_gpio(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { unsigned int gpio = - snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + snd_hda_param_read(codec, codec->core.afg, AC_PAR_GPIO_CAP); unsigned int enable, direction, wake, unsol, sticky, data; int i, max; snd_iprintf(buffer, "GPIO: io=%d, o=%d, i=%d, " @@ -667,13 +667,9 @@ static void print_device_list(struct snd_info_buffer *buffer, } } -static void print_codec_info(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) +static void print_codec_core_info(struct hdac_device *codec, + struct snd_info_buffer *buffer) { - struct hda_codec *codec = entry->private_data; - hda_nid_t nid; - int i, nodes; - snd_iprintf(buffer, "Codec: "); if (codec->vendor_name && codec->chip_name) snd_iprintf(buffer, "%s %s\n", @@ -695,29 +691,39 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg); else snd_iprintf(buffer, "No Modem Function Group found\n"); +} + +static void print_codec_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct hda_codec *codec = entry->private_data; + hda_nid_t nid, fg; + int i, nodes; - if (! codec->afg) + print_codec_core_info(&codec->core, buffer); + fg = codec->core.afg; + if (!fg) return; snd_hda_power_up(codec); snd_iprintf(buffer, "Default PCM:\n"); - print_pcm_caps(buffer, codec, codec->afg); + print_pcm_caps(buffer, codec, fg); snd_iprintf(buffer, "Default Amp-In caps: "); - print_amp_caps(buffer, codec, codec->afg, HDA_INPUT); + print_amp_caps(buffer, codec, fg, HDA_INPUT); snd_iprintf(buffer, "Default Amp-Out caps: "); - print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT); - snd_iprintf(buffer, "State of AFG node 0x%02x:\n", codec->afg); - print_power_state(buffer, codec, codec->afg); + print_amp_caps(buffer, codec, fg, HDA_OUTPUT); + snd_iprintf(buffer, "State of AFG node 0x%02x:\n", fg); + print_power_state(buffer, codec, fg); - nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + nodes = snd_hda_get_sub_nodes(codec, fg, &nid); if (! nid || nodes < 0) { snd_iprintf(buffer, "Invalid AFG subtree\n"); snd_hda_power_down(codec); return; } - print_gpio(buffer, codec, codec->afg); + print_gpio(buffer, codec, fg); if (codec->proc_widget_hook) - codec->proc_widget_hook(buffer, codec, codec->afg); + codec->proc_widget_hook(buffer, codec, fg); for (i = 0; i < nodes; i++, nid++) { unsigned int wid_caps = @@ -860,7 +866,7 @@ int snd_hda_codec_proc_new(struct hda_codec *codec) struct snd_info_entry *entry; int err; - snprintf(name, sizeof(name), "codec#%d", codec->addr); + snprintf(name, sizeof(name), "codec#%d", codec->core.addr); err = snd_card_proc_new(codec->card, name, &entry); if (err < 0) return err; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 3b5ed1108f9f0d..a6e3d9b511ab5f 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -48,33 +48,33 @@ static DEVICE_ATTR_RO(power_on_acct); static DEVICE_ATTR_RO(power_off_acct); #endif /* CONFIG_PM */ -#define CODEC_INFO_SHOW(type) \ +#define CODEC_INFO_SHOW(type, field) \ static ssize_t type##_show(struct device *dev, \ struct device_attribute *attr, \ char *buf) \ { \ struct hda_codec *codec = dev_get_drvdata(dev); \ - return sprintf(buf, "0x%x\n", codec->type); \ + return sprintf(buf, "0x%x\n", codec->field); \ } -#define CODEC_INFO_STR_SHOW(type) \ +#define CODEC_INFO_STR_SHOW(type, field) \ static ssize_t type##_show(struct device *dev, \ struct device_attribute *attr, \ char *buf) \ { \ struct hda_codec *codec = dev_get_drvdata(dev); \ return sprintf(buf, "%s\n", \ - codec->type ? codec->type : ""); \ + codec->field ? codec->field : ""); \ } -CODEC_INFO_SHOW(vendor_id); -CODEC_INFO_SHOW(subsystem_id); -CODEC_INFO_SHOW(revision_id); -CODEC_INFO_SHOW(afg); -CODEC_INFO_SHOW(mfg); -CODEC_INFO_STR_SHOW(vendor_name); -CODEC_INFO_STR_SHOW(chip_name); -CODEC_INFO_STR_SHOW(modelname); +CODEC_INFO_SHOW(vendor_id, core.vendor_id); +CODEC_INFO_SHOW(subsystem_id, core.subsystem_id); +CODEC_INFO_SHOW(revision_id, core.revision_id); +CODEC_INFO_SHOW(afg, core.afg); +CODEC_INFO_SHOW(mfg, core.mfg); +CODEC_INFO_STR_SHOW(vendor_name, core.vendor_name); +CODEC_INFO_STR_SHOW(chip_name, core.chip_name); +CODEC_INFO_STR_SHOW(modelname, modelname); static ssize_t pin_configs_show(struct hda_codec *codec, struct snd_array *list, @@ -170,7 +170,7 @@ static char *kstrndup_noeol(const char *src, size_t len) return s; } -#define CODEC_INFO_STORE(type) \ +#define CODEC_INFO_STORE(type, field) \ static ssize_t type##_store(struct device *dev, \ struct device_attribute *attr, \ const char *buf, size_t count) \ @@ -180,11 +180,11 @@ static ssize_t type##_store(struct device *dev, \ int err = kstrtoul(buf, 0, &val); \ if (err < 0) \ return err; \ - codec->type = val; \ + codec->field = val; \ return count; \ } -#define CODEC_INFO_STR_STORE(type) \ +#define CODEC_INFO_STR_STORE(type, field) \ static ssize_t type##_store(struct device *dev, \ struct device_attribute *attr, \ const char *buf, size_t count) \ @@ -193,17 +193,17 @@ static ssize_t type##_store(struct device *dev, \ char *s = kstrndup_noeol(buf, 64); \ if (!s) \ return -ENOMEM; \ - kfree(codec->type); \ - codec->type = s; \ + kfree(codec->field); \ + codec->field = s; \ return count; \ } -CODEC_INFO_STORE(vendor_id); -CODEC_INFO_STORE(subsystem_id); -CODEC_INFO_STORE(revision_id); -CODEC_INFO_STR_STORE(vendor_name); -CODEC_INFO_STR_STORE(chip_name); -CODEC_INFO_STR_STORE(modelname); +CODEC_INFO_STORE(vendor_id, core.vendor_id); +CODEC_INFO_STORE(subsystem_id, core.subsystem_id); +CODEC_INFO_STORE(revision_id, core.revision_id); +CODEC_INFO_STR_STORE(vendor_name, core.vendor_name); +CODEC_INFO_STR_STORE(chip_name, core.chip_name); +CODEC_INFO_STR_STORE(modelname, modelname); #define CODEC_ACTION_STORE(type) \ static ssize_t type##_store(struct device *dev, \ @@ -553,9 +553,9 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus, *codecp = NULL; if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { list_for_each_codec(codec, bus) { - if ((vendorid <= 0 || codec->vendor_id == vendorid) && - (subid <= 0 || codec->subsystem_id == subid) && - codec->addr == caddr) { + if ((vendorid <= 0 || codec->core.vendor_id == vendorid) && + (subid <= 0 || codec->core.subsystem_id == subid) && + codec->core.addr == caddr) { *codecp = codec; break; } @@ -595,8 +595,8 @@ static void parse_model_mode(char *buf, struct hda_bus *bus, static void parse_chip_name_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - kfree((*codecp)->chip_name); - (*codecp)->chip_name = kstrdup(buf, GFP_KERNEL); + kfree((*codecp)->core.chip_name); + (*codecp)->core.chip_name = kstrdup(buf, GFP_KERNEL); } #define DEFINE_PARSE_ID_MODE(name) \ @@ -605,7 +605,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \ { \ unsigned long val; \ if (!kstrtoul(buf, 0, &val)) \ - (*codecp)->name = val; \ + (*codecp)->core.name = val; \ } DEFINE_PARSE_ID_MODE(vendor_id); diff --git a/sound/pci/hda/local.h b/sound/pci/hda/local.h new file mode 100644 index 00000000000000..28cb7f98982ec7 --- /dev/null +++ b/sound/pci/hda/local.h @@ -0,0 +1,39 @@ +/* + */ + +#ifndef __HDAC_LOCAL_H +#define __HDAC_LOCAL_H + +int hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); + +#define get_wcaps(codec, nid) \ + hdac_read_parm(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) +/* get the widget type from widget capability bits */ +static inline int get_wcaps_type(unsigned int wcaps) +{ + if (!wcaps) + return -1; /* invalid type */ + return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; +} + +#define get_pin_caps(codec, nid) \ + hdac_read_parm(codec, nid, AC_PAR_PIN_CAP) + +static inline +unsigned int get_pin_cfg(struct hdac_device *codec, hda_nid_t nid) +{ + unsigned int val; + + if (snd_hdac_read(codec, nid, AC_VERB_GET_CONFIG_DEFAULT, 0, &val)) + return -1; + return val; +} + +#define get_amp_caps(codec, nid, dir) \ + hdac_read_parm(codec, nid, (dir) == HDA_OUTPUT ? \ + AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP) + +#define get_power_caps(codec, nid) \ + hdac_read_parm(codec, nid, AC_PAR_POWER_STATE) + +#endif /* __HDAC_LOCAL_H */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index af4c7be86c27b8..2278e83234b578 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -99,7 +99,7 @@ static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front, static void ad198x_power_eapd(struct hda_codec *codec) { /* We currently only handle front, HP */ - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x11d41882: case 0x11d4882a: case 0x11d41884: diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 72d20652df504b..5aff35a09fd47b 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4243,13 +4243,9 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; int i; - hda_nid_t nid; codec_dbg(codec, "ca0132_refresh_widget_caps.\n"); - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) - codec->wcaps[i] = snd_hda_param_read(codec, nid, - AC_PAR_AUDIO_WIDGET_CAP); + snd_hda_codec_update_widgets(codec); for (i = 0; i < spec->multiout.num_dacs; i++) refresh_amp_caps(codec, spec->dacs[i], HDA_OUTPUT); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 142a6cf786dada..1e21f9fbd54b93 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -103,10 +103,9 @@ static int add_beep_ctls(struct hda_codec *codec) static void cx_auto_parse_beep(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - hda_nid_t nid, end_nid; + hda_nid_t nid; - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) + for_each_hda_codec_node(nid, codec) if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) { set_beep_amp(spec, nid, 0, HDA_OUTPUT); break; @@ -120,10 +119,9 @@ static void cx_auto_parse_beep(struct hda_codec *codec) static void cx_auto_parse_eapd(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - hda_nid_t nid, end_nid; + hda_nid_t nid; - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) { + for_each_hda_codec_node(nid, codec) { if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) continue; if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)) @@ -848,7 +846,7 @@ static int patch_conexant_auto(struct hda_codec *codec) struct conexant_spec *spec; int err; - codec_info(codec, "%s: BIOS auto-probing.\n", codec->chip_name); + codec_info(codec, "%s: BIOS auto-probing.\n", codec->core.chip_name); spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) @@ -862,7 +860,7 @@ static int patch_conexant_auto(struct hda_codec *codec) if (spec->dynamic_eapd) spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x14f15045: codec->single_adc_amp = 1; spec->gen.mixer_nid = 0x17; @@ -896,7 +894,7 @@ static int patch_conexant_auto(struct hda_codec *codec) * others may use EAPD really as an amp switch, so it might be * not good to expose it blindly. */ - switch (codec->subsystem_id >> 16) { + switch (codec->core.subsystem_id >> 16) { case 0x103c: spec->gen.vmaster_mute_enum = 1; break; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7e9ff7b16e5658..35d92a8a99ce98 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -45,14 +45,14 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); -#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) -#define is_broadwell(codec) ((codec)->vendor_id == 0x80862808) -#define is_skylake(codec) ((codec)->vendor_id == 0x80862809) +#define is_haswell(codec) ((codec)->core.vendor_id == 0x80862807) +#define is_broadwell(codec) ((codec)->core.vendor_id == 0x80862808) +#define is_skylake(codec) ((codec)->core.vendor_id == 0x80862809) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec)) -#define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) -#define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) +#define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) +#define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) #define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) struct hdmi_spec_per_cvt { @@ -1391,13 +1391,12 @@ static void intel_not_share_assigned_cvt(struct hda_codec *codec, hda_nid_t pin_nid, int mux_idx) { struct hdmi_spec *spec = codec->spec; - hda_nid_t nid, end_nid; + hda_nid_t nid; int cvt_idx, curr; struct hdmi_spec_per_cvt *per_cvt; /* configure all pins, including "no physical connection" ones */ - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) { + for_each_hda_codec_node(nid, codec) { unsigned int wid_caps = get_wcaps(codec, nid); unsigned int wid_type = get_wcaps_type(wid_caps); @@ -1728,7 +1727,7 @@ static int hdmi_parse_codec(struct hda_codec *codec) hda_nid_t nid; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); if (!nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; @@ -2928,7 +2927,8 @@ static int patch_nvhdmi(struct hda_codec *codec) */ #define is_amdhdmi_rev3_or_later(codec) \ - ((codec)->vendor_id == 0x1002aa01 && ((codec)->revision_id & 0xff00) >= 0x0300) + ((codec)->core.vendor_id == 0x1002aa01 && \ + ((codec)->core.revision_id & 0xff00) >= 0x0300) #define has_amd_full_remap_support(codec) is_amdhdmi_rev3_or_later(codec) /* ATI/AMD specific HDA pin verbs, see the AMD HDA Verbs specification */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 124eacf67fc4dc..eee4532ab5f6db 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -299,7 +299,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) coef = alc_get_coef0(codec); - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0262: alc_update_coef_idx(codec, 0x7, 0, 1<<5); break; @@ -432,7 +432,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) snd_hda_sequence_write(codec, alc_gpio3_init_verbs); break; case ALC_INIT_DEFAULT: - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0260: alc_update_coefex_idx(codec, 0x1a, 7, 0, 0x2010); break; @@ -498,18 +498,18 @@ static int alc_auto_parse_customize_define(struct hda_codec *codec) if (!codec->bus->pci) return -1; - ass = codec->subsystem_id & 0xffff; + ass = codec->core.subsystem_id & 0xffff; if (ass != codec->bus->pci->subsystem_device && (ass & 1)) goto do_sku; nid = 0x1d; - if (codec->vendor_id == 0x10ec0260) + if (codec->core.vendor_id == 0x10ec0260) nid = 0x17; ass = snd_hda_codec_get_pincfg(codec, nid); if (!(ass & 1)) { codec_info(codec, "%s: SKU not ready 0x%08x\n", - codec->chip_name, ass); + codec->core.chip_name, ass); return -1; } @@ -585,7 +585,7 @@ static int alc_subsystem_id(struct hda_codec *codec, const hda_nid_t *ports) goto do_sku; } - ass = codec->subsystem_id & 0xffff; + ass = codec->core.subsystem_id & 0xffff; if (codec->bus->pci && ass != codec->bus->pci->subsystem_device && (ass & 1)) goto do_sku; @@ -600,7 +600,7 @@ static int alc_subsystem_id(struct hda_codec *codec, const hda_nid_t *ports) * 0 : override */ nid = 0x1d; - if (codec->vendor_id == 0x10ec0260) + if (codec->core.vendor_id == 0x10ec0260) nid = 0x17; ass = snd_hda_codec_get_pincfg(codec, nid); codec_dbg(codec, @@ -621,7 +621,7 @@ static int alc_subsystem_id(struct hda_codec *codec, const hda_nid_t *ports) return 0; do_sku: codec_dbg(codec, "realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n", - ass & 0xffff, codec->vendor_id); + ass & 0xffff, codec->core.vendor_id); /* * 0 : override * 1 : Swap Jack @@ -826,9 +826,9 @@ static const struct hda_codec_ops alc_patch_ops = { /* replace the codec chip_name with the given string */ static int alc_codec_rename(struct hda_codec *codec, const char *name) { - kfree(codec->chip_name); - codec->chip_name = kstrdup(name, GFP_KERNEL); - if (!codec->chip_name) { + kfree(codec->core.chip_name); + codec->core.chip_name = kstrdup(name, GFP_KERNEL); + if (!codec->core.chip_name) { alc_free(codec); return -ENOMEM; } @@ -904,7 +904,7 @@ static int alc_codec_rename_from_preset(struct hda_codec *codec) const struct alc_codec_rename_pci_table *q; for (p = rename_tbl; p->vendor_id; p++) { - if (p->vendor_id != codec->vendor_id) + if (p->vendor_id != codec->core.vendor_id) continue; if ((alc_get_coef0(codec) & p->coef_mask) == p->coef_bits) return alc_codec_rename(codec, p->name); @@ -913,7 +913,7 @@ static int alc_codec_rename_from_preset(struct hda_codec *codec) if (!codec->bus->pci) return 0; for (q = rename_pci_tbl; q->codec_vendor_id; q++) { - if (q->codec_vendor_id != codec->vendor_id) + if (q->codec_vendor_id != codec->core.vendor_id) continue; if (q->pci_subvendor != codec->bus->pci->subsystem_vendor) continue; @@ -1785,7 +1785,7 @@ static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) { unsigned int gpiostate, gpiomask, gpiodir; - gpiostate = snd_hda_codec_read(codec, codec->afg, 0, + gpiostate = snd_hda_codec_read(codec, codec->core.afg, 0, AC_VERB_GET_GPIO_DATA, 0); if (!muted) @@ -1793,23 +1793,23 @@ static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted) else gpiostate &= ~(1 << pin); - gpiomask = snd_hda_codec_read(codec, codec->afg, 0, + gpiomask = snd_hda_codec_read(codec, codec->core.afg, 0, AC_VERB_GET_GPIO_MASK, 0); gpiomask |= (1 << pin); - gpiodir = snd_hda_codec_read(codec, codec->afg, 0, + gpiodir = snd_hda_codec_read(codec, codec->core.afg, 0, AC_VERB_GET_GPIO_DIRECTION, 0); gpiodir |= (1 << pin); - snd_hda_codec_write(codec, codec->afg, 0, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_write(codec, codec->afg, 0, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DIRECTION, gpiodir); msleep(1); - snd_hda_codec_write(codec, codec->afg, 0, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DATA, gpiostate); } @@ -2269,7 +2269,7 @@ static int patch_alc882(struct hda_codec *codec) spec = codec->spec; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0882: case 0x10ec0885: case 0x10ec0900: @@ -3067,7 +3067,7 @@ static int alc269_resume(struct hda_codec *codec) * in the driver. */ if (spec->gpio_led) - snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); if (spec->has_alc5505_dsp) @@ -3112,8 +3112,8 @@ static void alc271_fixup_dmic(struct hda_codec *codec, }; unsigned int cfg; - if (strcmp(codec->chip_name, "ALC271X") && - strcmp(codec->chip_name, "ALC269VB")) + if (strcmp(codec->core.chip_name, "ALC271X") && + strcmp(codec->core.chip_name, "ALC269VB")) return; cfg = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED) @@ -3479,9 +3479,9 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, } snd_hda_add_verbs(codec, gpio_init); - snd_hda_codec_write_cache(codec, codec->afg, 0, + snd_hda_codec_write_cache(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x04); - snd_hda_jack_detect_enable_callback(codec, codec->afg, + snd_hda_jack_detect_enable_callback(codec, codec->core.afg, gpio2_mic_hotkey_event); spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; @@ -3564,7 +3564,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) {} }; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; @@ -3619,7 +3619,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, {} }; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0255: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); @@ -3688,7 +3688,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) {} }; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; @@ -3742,7 +3742,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) {} }; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; @@ -3796,7 +3796,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) {} }; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; @@ -3841,7 +3841,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) {} }; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0255: alc_process_coef_fw(codec, coef0255); msleep(300); @@ -4078,7 +4078,7 @@ static unsigned int alc_power_filter_xps13(struct hda_codec *codec, /* Avoid pop noises when headphones are plugged in */ if (spec->gen.hp_jack_present) - if (nid == codec->afg || nid == 0x02 || nid == 0x15) + if (nid == codec->core.afg || nid == 0x02 || nid == 0x15) return AC_PWRST_D0; return power_state; } @@ -5428,7 +5428,7 @@ static int patch_alc269(struct hda_codec *codec) if (has_cdefine_beep(codec)) spec->gen.beep_nid = 0x01; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0269: spec->codec_variant = ALC269_TYPE_ALC269VA; switch (alc_get_coef0(codec) & 0x00f0) { @@ -5772,9 +5772,9 @@ static int alc662_parse_auto_config(struct hda_codec *codec) static const hda_nid_t alc662_ssids[] = { 0x15, 0x1b, 0x14, 0 }; const hda_nid_t *ssids; - if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 || - codec->vendor_id == 0x10ec0671) + if (codec->core.vendor_id == 0x10ec0272 || codec->core.vendor_id == 0x10ec0663 || + codec->core.vendor_id == 0x10ec0665 || codec->core.vendor_id == 0x10ec0670 || + codec->core.vendor_id == 0x10ec0671) ssids = alc663_ssids; else ssids = alc662_ssids; @@ -5819,7 +5819,7 @@ static unsigned int gpio_led_power_filter(struct hda_codec *codec, unsigned int power_state) { struct alc_spec *spec = codec->spec; - if (nid == codec->afg && power_state == AC_PWRST_D3 && spec->gpio_led) + if (nid == codec->core.afg && power_state == AC_PWRST_D3 && spec->gpio_led) return AC_PWRST_D0; return power_state; } @@ -6317,7 +6317,7 @@ static int patch_alc662(struct hda_codec *codec) alc_fix_pll_init(codec, 0x20, 0x04, 15); - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0668: spec->init_hook = alc668_restore_default_value; break; @@ -6347,7 +6347,7 @@ static int patch_alc662(struct hda_codec *codec) goto error; if (!spec->gen.no_analog && spec->gen.beep_nid) { - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x10ec0662: set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); break; diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index df243134baa883..49b4868797a59b 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -205,8 +205,8 @@ static int si3054_build_pcms(struct hda_codec *codec) return -ENOMEM; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = codec->mfg; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = codec->mfg; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = codec->core.mfg; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = codec->core.mfg; info->pcm_type = HDA_PCM_TYPE_MODEM; return 0; } @@ -223,7 +223,7 @@ static int si3054_init(struct hda_codec *codec) u16 val; snd_hda_codec_write(codec, AC_NODE_ROOT, 0, AC_VERB_SET_CODEC_RESET, 0); - snd_hda_codec_write(codec, codec->mfg, 0, AC_VERB_SET_STREAM_FORMAT, 0); + snd_hda_codec_write(codec, codec->core.mfg, 0, AC_VERB_SET_STREAM_FORMAT, 0); SET_REG(codec, SI3054_LINE_RATE, 9600); SET_REG(codec, SI3054_LINE_LEVEL, SI3054_DTAG_MASK|SI3054_ATAG_MASK); SET_REG(codec, SI3054_EXTENDED_MID, 0); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 5b7c173adcb8c4..b314551749f179 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -299,32 +299,33 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, unsigned int dir_mask, unsigned int data) { unsigned int gpiostate, gpiomask, gpiodir; + hda_nid_t fg = codec->core.afg; codec_dbg(codec, "%s msk %x dir %x gpio %x\n", __func__, mask, dir_mask, data); - gpiostate = snd_hda_codec_read(codec, codec->afg, 0, + gpiostate = snd_hda_codec_read(codec, fg, 0, AC_VERB_GET_GPIO_DATA, 0); gpiostate = (gpiostate & ~dir_mask) | (data & dir_mask); - gpiomask = snd_hda_codec_read(codec, codec->afg, 0, + gpiomask = snd_hda_codec_read(codec, fg, 0, AC_VERB_GET_GPIO_MASK, 0); gpiomask |= mask; - gpiodir = snd_hda_codec_read(codec, codec->afg, 0, + gpiodir = snd_hda_codec_read(codec, fg, 0, AC_VERB_GET_GPIO_DIRECTION, 0); gpiodir |= dir_mask; /* Configure GPIOx as CMOS */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); + snd_hda_codec_write(codec, fg, 0, 0x7e7, 0); - snd_hda_codec_write(codec, codec->afg, 0, + snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_read(codec, codec->afg, 0, + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_GPIO_DIRECTION, gpiodir); /* sync */ msleep(1); - snd_hda_codec_read(codec, codec->afg, 0, + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } @@ -387,7 +388,7 @@ static unsigned int stac_vref_led_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) { - if (nid == codec->afg && power_state == AC_PWRST_D3) + if (nid == codec->core.afg && power_state == AC_PWRST_D3) return AC_PWRST_D1; return snd_hda_gen_path_power_filter(codec, nid, power_state); } @@ -432,7 +433,7 @@ static void stac_update_outputs(struct hda_codec *codec) if (spec->gpio_mute) spec->gen.master_mute = - !(snd_hda_codec_read(codec, codec->afg, 0, + !(snd_hda_codec_read(codec, codec->core.afg, 0, AC_VERB_GET_GPIO_DATA, 0) & spec->gpio_mute); snd_hda_gen_update_outputs(codec); @@ -476,7 +477,7 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, if (val != spec->power_map_bits) { spec->power_map_bits = val; if (do_write) - snd_hda_codec_write(codec, codec->afg, 0, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_IDT_SET_POWER_MAP, val); } } @@ -508,7 +509,8 @@ static void jack_update_power(struct hda_codec *codec, false); } - snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_IDT_SET_POWER_MAP, + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_IDT_SET_POWER_MAP, spec->power_map_bits); } @@ -517,10 +519,10 @@ static void stac_vref_event(struct hda_codec *codec, { unsigned int data; - data = snd_hda_codec_read(codec, codec->afg, 0, + data = snd_hda_codec_read(codec, codec->core.afg, 0, AC_VERB_GET_GPIO_DATA, 0); /* toggle VREF state based on GPIOx status */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, + snd_hda_codec_write(codec, codec->core.afg, 0, 0x7e0, !!(data & (1 << event->private_data))); } @@ -622,7 +624,7 @@ static int stac_aloopback_put(struct snd_kcontrol *kcontrol, /* Only return the bits defined by the shift value of the * first two bytes of the mask */ - dac_mode = snd_hda_codec_read(codec, codec->afg, 0, + dac_mode = snd_hda_codec_read(codec, codec->core.afg, 0, kcontrol->private_value & 0xFFFF, 0x0); dac_mode >>= spec->aloopback_shift; @@ -634,7 +636,7 @@ static int stac_aloopback_put(struct snd_kcontrol *kcontrol, dac_mode &= ~idx_val; } - snd_hda_codec_write_cache(codec, codec->afg, 0, + snd_hda_codec_write_cache(codec, codec->core.afg, 0, kcontrol->private_value >> 16, dac_mode); return 1; @@ -658,11 +660,11 @@ static int stac_aloopback_put(struct snd_kcontrol *kcontrol, /* check whether it's a HP laptop with a docking port */ static bool hp_bnb2011_with_dock(struct hda_codec *codec) { - if (codec->vendor_id != 0x111d7605 && - codec->vendor_id != 0x111d76d1) + if (codec->core.vendor_id != 0x111d7605 && + codec->core.vendor_id != 0x111d76d1) return false; - switch (codec->subsystem_id) { + switch (codec->core.subsystem_id) { case 0x103c1618: case 0x103c1619: case 0x103c161a: @@ -733,7 +735,7 @@ static void set_hp_led_gpio(struct hda_codec *codec) if (spec->gpio_led) return; - gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + gpio = snd_hda_param_read(codec, codec->core.afg, AC_PAR_GPIO_CAP); gpio &= AC_GPIO_IO_COUNT; if (gpio > 3) spec->gpio_led = 0x08; /* GPIO 3 */ @@ -777,7 +779,7 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) &spec->gpio_led_polarity, &spec->gpio_led) == 2) { unsigned int max_gpio; - max_gpio = snd_hda_param_read(codec, codec->afg, + max_gpio = snd_hda_param_read(codec, codec->core.afg, AC_PAR_GPIO_CAP); max_gpio &= AC_GPIO_IO_COUNT; if (spec->gpio_led < max_gpio) @@ -807,7 +809,7 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) * we statically set the GPIO - if not a B-series system * and default polarity is provided */ - if (!hp_blike_system(codec->subsystem_id) && + if (!hp_blike_system(codec->core.subsystem_id) && (default_polarity == 0 || default_polarity == 1)) { set_hp_led_gpio(codec); spec->gpio_led_polarity = default_polarity; @@ -2134,7 +2136,7 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ #ifdef CONFIG_PM /* resetting controller clears GPIO, so we need to keep on */ - codec->d3_stop_clk = 0; + codec->core.power_caps &= ~AC_PWRST_CLKSTOP; #endif } } @@ -3031,9 +3033,9 @@ static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, return; /* Enable VREF power saving on GPIO1 detect */ - snd_hda_codec_write_cache(codec, codec->afg, 0, + snd_hda_codec_write_cache(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); - jack = snd_hda_jack_detect_enable_callback(codec, codec->afg, + jack = snd_hda_jack_detect_enable_callback(codec, codec->core.afg, stac_vref_event); if (!IS_ERR(jack)) jack->private_data = 0x02; @@ -3093,7 +3095,7 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, if (action != HDA_FIXUP_ACT_PRE_PROBE) return; - if (hp_blike_system(codec->subsystem_id)) { + if (hp_blike_system(codec->core.subsystem_id)) { unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || @@ -3792,7 +3794,7 @@ static void stac927x_fixup_dell_dmic(struct hda_codec *codec, if (action != HDA_FIXUP_ACT_PRE_PROBE) return; - if (codec->subsystem_id != 0x1028022f) { + if (codec->core.subsystem_id != 0x1028022f) { /* GPIO2 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = 0x04; spec->gpio_dir = spec->gpio_data = 0x04; @@ -4053,9 +4055,9 @@ static void stac9205_fixup_dell_m43(struct hda_codec *codec, snd_hda_apply_pincfgs(codec, dell_9205_m43_pin_configs); /* Enable unsol response for GPIO4/Dock HP connection */ - snd_hda_codec_write_cache(codec, codec->afg, 0, + snd_hda_codec_write_cache(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); - jack = snd_hda_jack_detect_enable_callback(codec, codec->afg, + jack = snd_hda_jack_detect_enable_callback(codec, codec->core.afg, stac_vref_event); if (!IS_ERR(jack)) jack->private_data = 0x01; @@ -4302,7 +4304,7 @@ static int stac_init(struct hda_codec *codec) /* sync the power-map */ if (spec->num_pwrs) - snd_hda_codec_write(codec, codec->afg, 0, + snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_IDT_SET_POWER_MAP, spec->power_map_bits); @@ -4338,7 +4340,7 @@ static void stac_shutup(struct hda_codec *codec) static void stac92hd_proc_hook(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { - if (nid == codec->afg) + if (nid == codec->core.afg) snd_iprintf(buffer, "Power-Map: 0x%02x\n", snd_hda_codec_read(codec, nid, 0, AC_VERB_IDT_GET_POWER_MAP, 0)); @@ -4349,7 +4351,7 @@ static void analog_loop_proc_hook(struct snd_info_buffer *buffer, unsigned int verb) { snd_iprintf(buffer, "Analog Loopback: 0x%02x\n", - snd_hda_codec_read(codec, codec->afg, 0, verb, 0)); + snd_hda_codec_read(codec, codec->core.afg, 0, verb, 0)); } /* stac92hd71bxx, stac92hd73xx */ @@ -4357,21 +4359,21 @@ static void stac92hd7x_proc_hook(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { stac92hd_proc_hook(buffer, codec, nid); - if (nid == codec->afg) + if (nid == codec->core.afg) analog_loop_proc_hook(buffer, codec, 0xfa0); } static void stac9205_proc_hook(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { - if (nid == codec->afg) + if (nid == codec->core.afg) analog_loop_proc_hook(buffer, codec, 0xfe0); } static void stac927x_proc_hook(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { - if (nid == codec->afg) + if (nid == codec->core.afg) analog_loop_proc_hook(buffer, codec, 0xfeb); } #else @@ -4597,7 +4599,8 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) if (err < 0) return err; - codec->epss = 0; /* longer delay needed for D3 */ + /* longer delay needed for D3 */ + codec->core.power_caps &= ~AC_PWRST_EPSS; spec = codec->spec; codec->power_save_node = 1; @@ -4647,7 +4650,8 @@ static int patch_stac92hd95(struct hda_codec *codec) if (err < 0) return err; - codec->epss = 0; /* longer delay needed for D3 */ + /* longer delay needed for D3 */ + codec->core.power_caps &= ~AC_PWRST_EPSS; spec = codec->spec; codec->power_save_node = 1; @@ -4706,14 +4710,14 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->gpio_dir = 0x01; spec->gpio_data = 0x01; - switch (codec->vendor_id) { + switch (codec->core.vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: unmute_init++; break; case 0x111d7608: /* 5 Port with Analog Mixer */ - if ((codec->revision_id & 0xf) == 0 || - (codec->revision_id & 0xf) == 1) + if ((codec->core.revision_id & 0xf) == 0 || + (codec->core.revision_id & 0xf) == 1) spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ @@ -4722,7 +4726,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); break; case 0x111d7603: /* 6 Port with Analog Mixer */ - if ((codec->revision_id & 0xf) == 1) + if ((codec->core.revision_id & 0xf) == 1) spec->stream_delay = 40; /* 40 milliseconds */ break; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 485663bb91010f..a34d7671937f67 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -140,7 +140,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { - u32 vendor_id = codec->vendor_id; + u32 vendor_id = codec->core.vendor_id; u16 ven_id = vendor_id >> 16; u16 dev_id = vendor_id & 0xffff; enum VIA_HDA_CODEC codec_type; @@ -335,7 +335,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) return; /* other codecs are not supported */ } /* send verb */ - snd_hda_codec_write(codec, codec->afg, 0, verb, parm); + snd_hda_codec_write(codec, codec->core.afg, 0, verb, parm); } static void analog_low_current_mode(struct hda_codec *codec) @@ -558,7 +558,7 @@ static int vt1708_build_pcms(struct hda_codec *codec) int i, err; err = snd_hda_gen_build_pcms(codec); - if (err < 0 || codec->vendor_id != 0x11061708) + if (err < 0 || codec->core.vendor_id != 0x11061708) return err; /* We got noisy outputs on the right channel on VT1708 when @@ -714,19 +714,19 @@ static int patch_vt1708S(struct hda_codec *codec) /* correct names for VT1708BCE */ if (get_codec_type(codec) == VT1708BCE) { - kfree(codec->chip_name); - codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); + kfree(codec->core.chip_name); + codec->core.chip_name = kstrdup("VT1708BCE", GFP_KERNEL); snprintf(codec->card->mixername, sizeof(codec->card->mixername), - "%s %s", codec->vendor_name, codec->chip_name); + "%s %s", codec->core.vendor_name, codec->core.chip_name); } /* correct names for VT1705 */ - if (codec->vendor_id == 0x11064397) { - kfree(codec->chip_name); - codec->chip_name = kstrdup("VT1705", GFP_KERNEL); + if (codec->core.vendor_id == 0x11064397) { + kfree(codec->core.chip_name); + codec->core.chip_name = kstrdup("VT1705", GFP_KERNEL); snprintf(codec->card->mixername, sizeof(codec->card->mixername), - "%s %s", codec->vendor_name, codec->chip_name); + "%s %s", codec->core.vendor_name, codec->core.chip_name); } /* automatic parse from the BIOS config */ @@ -815,8 +815,7 @@ static int add_secret_dac_path(struct hda_codec *codec) } /* find the primary DAC and add to the connection list */ - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { + for_each_hda_codec_node(nid, codec) { unsigned int caps = get_wcaps(codec, nid); if (get_wcaps_type(caps) == AC_WID_AUD_OUT && !(caps & AC_WCAP_DIGITAL)) { diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 6ba0b5517c407f..0a4ad5feb82e78 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -21,7 +21,7 @@ static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, static bool is_thinkpad(struct hda_codec *codec) { bool found = false; - if (codec->subsystem_id >> 16 != 0x17aa) + if (codec->core.subsystem_id >> 16 != 0x17aa) return false; if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) return true; From 3256be6537751f65c76b3ecfbb4e667f87525a2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Feb 2015 14:59:42 +0100 Subject: [PATCH 270/411] ALSA: hda - Add widget sysfs tree This patch changes the sysfs files assigned to the codec device on the bus which were formerly identical with hwdep sysfs files. Now it shows only a few core parameter, vendor_id, subsystem_id, revision_id, afg, mfg, vendor_name and chip_name. In addition, now a widget tree is added to the bus device sysfs directory for showing the widget topology and attributes. It's just a flat tree consisting of subdirectories named as the widget NID including various attributes like widget capability bits. The AFG (usually NID 0x01) is always found there, and it contains always amp_in_caps, amp_out_caps and power_caps files. Each of these attributes show a single value. The rest are the widget nodes belonging to that AFG. Note that the child node might not start from 0x02 but from another value like 0x0a. Each child node may contain caps, pin_caps, amp_in_caps, amp_out_caps, power_caps and connections files. The caps (representing the widget capability bits) always contain a value. The rest may contain value(s) if the attribute exists on the node. Only connections file show multiple values while other attributes have zero or one single value. An example of ls -R output is like below: % ls -R /sys/bus/hdaudio/devices/hdaudioC0D0/ /sys/bus/hdaudio/devices/hdaudioC0D0/widgets/: 01/ 04/ 07/ 0a/ 0d/ 10/ 13/ 16/ 19/ 1c/ 1f/ 22/ 02/ 05/ 08/ 0b/ 0e/ 11/ 14/ 17/ 1a/ 1d/ 20/ 23/ 03/ 06/ 09/ 0c/ 0f/ 12/ 15/ 18/ 1b/ 1e/ 21/ /sys/bus/hdaudio/devices/hdaudioC0D0/widgets/01: amp_in_caps amp_out_caps power_caps /sys/bus/hdaudio/devices/hdaudioC0D0/widgets/02: amp_in_caps amp_out_caps caps connections pin_caps pin_cfg power_caps /sys/bus/hdaudio/devices/hdaudioC0D0/widgets/03: ..... Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 6 + sound/hda/Makefile | 2 +- sound/hda/hdac_device.c | 35 ++++ sound/hda/hdac_sysfs.c | 404 ++++++++++++++++++++++++++++++++++++++ sound/hda/local.h | 4 + sound/pci/hda/hda_bind.c | 4 +- sound/pci/hda/hda_codec.c | 6 +- 7 files changed, 454 insertions(+), 7 deletions(-) create mode 100644 sound/hda/hdac_sysfs.c diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b81b4bec6f050b..6ed2b421e29ebf 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -14,6 +14,7 @@ typedef u16 hda_nid_t; struct hdac_bus; struct hdac_device; struct hdac_driver; +struct hdac_widget_tree; /* * exported bus type @@ -53,6 +54,9 @@ struct hdac_device { /* misc flags */ atomic_t in_pm; /* suspend/resume being performed */ + + /* sysfs */ + struct hdac_widget_tree *widgets; }; /* device/driver type used for matching */ @@ -71,6 +75,8 @@ enum { int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus, const char *name, unsigned int addr); void snd_hdac_device_exit(struct hdac_device *dev); +int snd_hdac_device_register(struct hdac_device *codec); +void snd_hdac_device_unregister(struct hdac_device *codec); int snd_hdac_refresh_widgets(struct hdac_device *codec); diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 3c7625e595cf55..ae8b5128b5c320 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -1,3 +1,3 @@ -snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o +snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index a3f52ad4de3739..1470ecc354dbc1 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -45,6 +45,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, dev->parent = bus->dev; dev->bus = &snd_hda_bus_type; dev->release = default_release; + dev->groups = hdac_dev_attr_groups; dev_set_name(dev, "%s", name); device_enable_async_suspend(dev); @@ -127,6 +128,40 @@ void snd_hdac_device_exit(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_device_exit); +/** + * snd_hdac_device_register - register the hd-audio codec base device + * codec: the device to register + */ +int snd_hdac_device_register(struct hdac_device *codec) +{ + int err; + + err = device_add(&codec->dev); + if (err < 0) + return err; + err = hda_widget_sysfs_init(codec); + if (err < 0) { + device_del(&codec->dev); + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_device_register); + +/** + * snd_hdac_device_unregister - unregister the hd-audio codec base device + * codec: the device to unregister + */ +void snd_hdac_device_unregister(struct hdac_device *codec) +{ + if (device_is_registered(&codec->dev)) { + hda_widget_sysfs_exit(codec); + device_del(&codec->dev); + } +} +EXPORT_SYMBOL_GPL(snd_hdac_device_unregister); + /** * snd_hdac_make_cmd - compose a 32bit command word to be sent to the * HD-audio controller diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c new file mode 100644 index 00000000000000..b358d515780238 --- /dev/null +++ b/sound/hda/hdac_sysfs.c @@ -0,0 +1,404 @@ +/* + * sysfs support for HD-audio core device + */ + +#include +#include +#include +#include +#include +#include "local.h" + +struct hdac_widget_tree { + struct kobject *root; + struct kobject *afg; + struct kobject **nodes; +}; + +#define CODEC_ATTR(type) \ +static ssize_t type##_show(struct device *dev, \ + struct device_attribute *attr, \ + char *buf) \ +{ \ + struct hdac_device *codec = dev_to_hdac_dev(dev); \ + return sprintf(buf, "0x%x\n", codec->type); \ +} \ +static DEVICE_ATTR_RO(type) + +#define CODEC_ATTR_STR(type) \ +static ssize_t type##_show(struct device *dev, \ + struct device_attribute *attr, \ + char *buf) \ +{ \ + struct hdac_device *codec = dev_to_hdac_dev(dev); \ + return sprintf(buf, "%s\n", \ + codec->type ? codec->type : ""); \ +} \ +static DEVICE_ATTR_RO(type) + +CODEC_ATTR(vendor_id); +CODEC_ATTR(subsystem_id); +CODEC_ATTR(revision_id); +CODEC_ATTR(afg); +CODEC_ATTR(mfg); +CODEC_ATTR_STR(vendor_name); +CODEC_ATTR_STR(chip_name); + +static struct attribute *hdac_dev_attrs[] = { + &dev_attr_vendor_id.attr, + &dev_attr_subsystem_id.attr, + &dev_attr_revision_id.attr, + &dev_attr_afg.attr, + &dev_attr_mfg.attr, + &dev_attr_vendor_name.attr, + &dev_attr_chip_name.attr, + NULL +}; + +static struct attribute_group hdac_dev_attr_group = { + .attrs = hdac_dev_attrs, +}; + +const struct attribute_group *hdac_dev_attr_groups[] = { + &hdac_dev_attr_group, + NULL +}; + +/* + * Widget tree sysfs + * + * This is a tree showing the attributes of each widget. It appears like + * /sys/bus/hdaudioC0D0/widgets/04/caps + */ + +struct widget_attribute; + +struct widget_attribute { + struct attribute attr; + ssize_t (*show)(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf); + ssize_t (*store)(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, + const char *buf, size_t count); +}; + +static int get_codec_nid(struct kobject *kobj, struct hdac_device **codecp) +{ + struct device *dev = kobj_to_dev(kobj->parent->parent); + int nid; + ssize_t ret; + + ret = kstrtoint(kobj->name, 16, &nid); + if (ret < 0) + return ret; + *codecp = dev_to_hdac_dev(dev); + return nid; +} + +static ssize_t widget_attr_show(struct kobject *kobj, struct attribute *attr, + char *buf) +{ + struct widget_attribute *wid_attr = + container_of(attr, struct widget_attribute, attr); + struct hdac_device *codec; + int nid; + + if (!wid_attr->show) + return -EIO; + nid = get_codec_nid(kobj, &codec); + if (nid < 0) + return nid; + return wid_attr->show(codec, nid, wid_attr, buf); +} + +static ssize_t widget_attr_store(struct kobject *kobj, struct attribute *attr, + const char *buf, size_t count) +{ + struct widget_attribute *wid_attr = + container_of(attr, struct widget_attribute, attr); + struct hdac_device *codec; + int nid; + + if (!wid_attr->store) + return -EIO; + nid = get_codec_nid(kobj, &codec); + if (nid < 0) + return nid; + return wid_attr->store(codec, nid, wid_attr, buf, count); +} + +static const struct sysfs_ops widget_sysfs_ops = { + .show = widget_attr_show, + .store = widget_attr_store, +}; + +static void widget_release(struct kobject *kobj) +{ + kfree(kobj); +} + +static struct kobj_type widget_ktype = { + .release = widget_release, + .sysfs_ops = &widget_sysfs_ops, +}; + +#define WIDGET_ATTR_RO(_name) \ + struct widget_attribute wid_attr_##_name = __ATTR_RO(_name) +#define WIDGET_ATTR_RW(_name) \ + struct widget_attribute wid_attr_##_name = __ATTR_RW(_name) + +static ssize_t caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return sprintf(buf, "0x%08x\n", get_wcaps(codec, nid)); +} + +static ssize_t pin_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + return 0; + return sprintf(buf, "0x%08x\n", + snd_hdac_read_parm(codec, nid, AC_PAR_PIN_CAP)); +} + +static ssize_t pin_cfg_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + unsigned int val; + + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + return 0; + if (snd_hdac_read(codec, nid, AC_VERB_GET_CONFIG_DEFAULT, 0, &val)) + return 0; + return sprintf(buf, "0x%08x\n", val); +} + +static bool has_pcm_cap(struct hdac_device *codec, hda_nid_t nid) +{ + if (nid == codec->afg || nid == codec->mfg) + return true; + switch (get_wcaps_type(get_wcaps(codec, nid))) { + case AC_WID_AUD_OUT: + case AC_WID_AUD_IN: + return true; + default: + return false; + } +} + +static ssize_t pcm_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (!has_pcm_cap(codec, nid)) + return 0; + return sprintf(buf, "0x%08x\n", + snd_hdac_read_parm(codec, nid, AC_PAR_PCM)); +} + +static ssize_t pcm_formats_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (!has_pcm_cap(codec, nid)) + return 0; + return sprintf(buf, "0x%08x\n", + snd_hdac_read_parm(codec, nid, AC_PAR_STREAM)); +} + +static ssize_t amp_in_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (nid != codec->afg && !(get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) + return 0; + return sprintf(buf, "0x%08x\n", + snd_hdac_read_parm(codec, nid, AC_PAR_AMP_IN_CAP)); +} + +static ssize_t amp_out_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (nid != codec->afg && !(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) + return 0; + return sprintf(buf, "0x%08x\n", + snd_hdac_read_parm(codec, nid, AC_PAR_AMP_OUT_CAP)); +} + +static ssize_t power_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (nid != codec->afg && !(get_wcaps(codec, nid) & AC_WCAP_POWER)) + return 0; + return sprintf(buf, "0x%08x\n", + snd_hdac_read_parm(codec, nid, AC_PAR_POWER_STATE)); +} + +static ssize_t gpio_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return sprintf(buf, "0x%08x\n", + snd_hdac_read_parm(codec, nid, AC_PAR_GPIO_CAP)); +} + +static ssize_t connections_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + hda_nid_t list[32]; + int i, nconns; + ssize_t ret = 0; + + nconns = snd_hdac_get_connections(codec, nid, list, ARRAY_SIZE(list)); + if (nconns <= 0) + return nconns; + for (i = 0; i < nconns; i++) + ret += sprintf(buf + ret, "%s0x%02x", i ? " " : "", list[i]); + ret += sprintf(buf + ret, "\n"); + return ret; +} + +static WIDGET_ATTR_RO(caps); +static WIDGET_ATTR_RO(pin_caps); +static WIDGET_ATTR_RO(pin_cfg); +static WIDGET_ATTR_RO(pcm_caps); +static WIDGET_ATTR_RO(pcm_formats); +static WIDGET_ATTR_RO(amp_in_caps); +static WIDGET_ATTR_RO(amp_out_caps); +static WIDGET_ATTR_RO(power_caps); +static WIDGET_ATTR_RO(gpio_caps); +static WIDGET_ATTR_RO(connections); + +static struct attribute *widget_node_attrs[] = { + &wid_attr_caps.attr, + &wid_attr_pin_caps.attr, + &wid_attr_pin_cfg.attr, + &wid_attr_pcm_caps.attr, + &wid_attr_pcm_formats.attr, + &wid_attr_amp_in_caps.attr, + &wid_attr_amp_out_caps.attr, + &wid_attr_power_caps.attr, + &wid_attr_connections.attr, + NULL, +}; + +static struct attribute *widget_afg_attrs[] = { + &wid_attr_pcm_caps.attr, + &wid_attr_pcm_formats.attr, + &wid_attr_amp_in_caps.attr, + &wid_attr_amp_out_caps.attr, + &wid_attr_power_caps.attr, + &wid_attr_gpio_caps.attr, + NULL, +}; + +static const struct attribute_group widget_node_group = { + .attrs = widget_node_attrs, +}; + +static const struct attribute_group widget_afg_group = { + .attrs = widget_afg_attrs, +}; + +static void free_widget_node(struct kobject *kobj, + const struct attribute_group *group) +{ + if (kobj) { + sysfs_remove_group(kobj, group); + kobject_put(kobj); + } +} + +static void widget_tree_free(struct hdac_device *codec) +{ + struct hdac_widget_tree *tree = codec->widgets; + struct kobject **p; + + if (!tree) + return; + if (tree->nodes) { + for (p = tree->nodes; *p; p++) + free_widget_node(*p, &widget_node_group); + kfree(tree->nodes); + } + free_widget_node(tree->afg, &widget_afg_group); + if (tree->root) + kobject_put(tree->root); + kfree(tree); + codec->widgets = NULL; +} + +static int add_widget_node(struct kobject *parent, hda_nid_t nid, + const struct attribute_group *group, + struct kobject **res) +{ + struct kobject *kobj = kzalloc(sizeof(*kobj), GFP_KERNEL); + int err; + + if (!kobj) + return -ENOMEM; + kobject_init(kobj, &widget_ktype); + err = kobject_add(kobj, parent, "%02x", nid); + if (err < 0) + return err; + err = sysfs_create_group(kobj, group); + if (err < 0) { + kobject_put(kobj); + return err; + } + + *res = kobj; + return 0; +} + +static int widget_tree_create(struct hdac_device *codec) +{ + struct hdac_widget_tree *tree; + int i, err; + hda_nid_t nid; + + tree = codec->widgets = kzalloc(sizeof(*tree), GFP_KERNEL); + if (!tree) + return -ENOMEM; + + tree->root = kobject_create_and_add("widgets", &codec->dev.kobj); + if (!tree->root) + return -ENOMEM; + + if (codec->afg) { + err = add_widget_node(tree->root, codec->afg, + &widget_afg_group, &tree->afg); + if (err < 0) + return err; + } + + tree->nodes = kcalloc(codec->num_nodes + 1, sizeof(*tree->nodes), + GFP_KERNEL); + if (!tree->nodes) + return -ENOMEM; + + for (i = 0, nid = codec->start_nid; i < codec->num_nodes; i++, nid++) { + err = add_widget_node(tree->root, nid, &widget_node_group, + &tree->nodes[i]); + if (err < 0) + return err; + } + + kobject_uevent(tree->root, KOBJ_CHANGE); + return 0; +} + +int hda_widget_sysfs_init(struct hdac_device *codec) +{ + int err; + + err = widget_tree_create(codec); + if (err < 0) { + widget_tree_free(codec); + return err; + } + + return 0; +} + +void hda_widget_sysfs_exit(struct hdac_device *codec) +{ + widget_tree_free(codec); +} diff --git a/sound/hda/local.h b/sound/hda/local.h index a077d1f656f6ec..d692f417ddc0a2 100644 --- a/sound/hda/local.h +++ b/sound/hda/local.h @@ -16,4 +16,8 @@ static inline int get_wcaps_type(unsigned int wcaps) return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; } +extern const struct attribute_group *hdac_dev_attr_groups[]; +int hda_widget_sysfs_init(struct hdac_device *codec); +void hda_widget_sysfs_exit(struct hdac_device *codec); + #endif /* __HDAC_LOCAL_H */ diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index ad276a9771dbeb..130f672e6f37b2 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -240,7 +240,7 @@ int snd_hda_codec_configure(struct hda_codec *codec) else codec->probe_id = 0; - err = device_add(hda_codec_dev(codec)); + err = snd_hdac_device_register(&codec->core); if (err < 0) return err; @@ -262,7 +262,7 @@ int snd_hda_codec_configure(struct hda_codec *codec) return 0; error: - device_del(hda_codec_dev(codec)); + snd_hdac_device_unregister(&codec->core); return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_configure); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ddfc0fbbee2324..b162fc40348f4a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -967,8 +967,7 @@ static int snd_hda_codec_dev_free(struct snd_device *device) struct hda_codec *codec = device->device_data; codec->in_freeing = 1; - if (device_is_registered(hda_codec_dev(codec))) - device_del(hda_codec_dev(codec)); + snd_hdac_device_unregister(&codec->core); put_device(hda_codec_dev(codec)); return 0; } @@ -2182,8 +2181,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) return -EBUSY; /* OK, let it free */ - if (device_is_registered(hda_codec_dev(codec))) - device_del(hda_codec_dev(codec)); + snd_hdac_device_unregister(&codec->core); /* allow device access again */ snd_hda_unlock_devices(bus); From 05852448690d7d810175f8ceccefba083525aa89 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Mar 2015 15:40:08 +0100 Subject: [PATCH 271/411] ALSA: hda - Support indirect execution of verbs Add an overriding exec_verb op to struct hdac_device so that the call via snd_hdac_exec_verb() can switch to a different route depending on the setup. The codec driver sets this field so that it can handle the errors or applying quirks appropriately. Furthermore, this mechanism will be used for smooth transition for the regmap support in later patches. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 6 ++++++ sound/hda/hdac_device.c | 24 +++++++++++++++++++++++- sound/pci/hda/hda_codec.c | 12 +++++++----- 3 files changed, 36 insertions(+), 6 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 6ed2b421e29ebf..675614dc2b8803 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -48,6 +48,10 @@ struct hdac_device { const char *vendor_name; /* codec vendor name */ const char *chip_name; /* codec chip name */ + /* verb exec op override */ + int (*exec_verb)(struct hdac_device *dev, unsigned int cmd, + unsigned int flags, unsigned int *res); + /* widgets */ unsigned int num_nodes; hda_nid_t start_nid, end_nid; @@ -82,6 +86,8 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec); unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, unsigned int verb, unsigned int parm); +int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd, + unsigned int flags, unsigned int *res); int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid, unsigned int verb, unsigned int parm, unsigned int *res); int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 1470ecc354dbc1..aaece36247e772 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -193,6 +193,28 @@ unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hdac_make_cmd); +/** + * snd_hdac_exec_verb - execute an encoded verb + * @codec: the codec object + * @cmd: encoded verb to execute + * @flags: optional flags, pass zero for default + * @res: the pointer to store the result, NULL if running async + * + * Returns zero if successful, or a negative error code. + * + * This calls the exec_verb op when set in hdac_codec. If not, + * call the default snd_hdac_bus_exec_verb(). + */ +int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd, + unsigned int flags, unsigned int *res) +{ + if (codec->exec_verb) + return codec->exec_verb(codec, cmd, flags, res); + return snd_hdac_bus_exec_verb(codec->bus, codec->addr, cmd, res); +} +EXPORT_SYMBOL_GPL(snd_hdac_exec_verb); + + /** * snd_hdac_read - execute a verb * @codec: the codec object @@ -208,7 +230,7 @@ int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid, { unsigned int cmd = snd_hdac_make_cmd(codec, nid, verb, parm); - return snd_hdac_bus_exec_verb(codec->bus, codec->addr, cmd, res); + return snd_hdac_exec_verb(codec, cmd, 0, res); } EXPORT_SYMBOL_GPL(snd_hdac_read); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b162fc40348f4a..36483f7dd3ce7e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -124,11 +124,12 @@ const char *snd_hda_get_jack_type(u32 cfg) EXPORT_SYMBOL_GPL(snd_hda_get_jack_type); /* - * Send and receive a verb + * Send and receive a verb - passed to exec_verb override for hdac_device */ -static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, - int flags, unsigned int *res) +static int codec_exec_verb(struct hdac_device *dev, unsigned int cmd, + unsigned int flags, unsigned int *res) { + struct hda_codec *codec = container_of(dev, struct hda_codec, core); struct hda_bus *bus = codec->bus; int err; @@ -177,7 +178,7 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, { unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm); unsigned int res; - if (codec_exec_verb(codec, cmd, flags, &res)) + if (snd_hdac_exec_verb(&codec->core, cmd, flags, &res)) return -1; return res; } @@ -199,7 +200,7 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm); - return codec_exec_verb(codec, cmd, flags, NULL); + return snd_hdac_exec_verb(&codec->core, cmd, flags, NULL); } EXPORT_SYMBOL_GPL(snd_hda_codec_write); @@ -1026,6 +1027,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, codec->core.dev.release = snd_hda_codec_dev_release; codec->core.type = HDA_DEV_LEGACY; + codec->core.exec_verb = codec_exec_verb; codec->bus = bus; codec->card = card; From c4c2533f802d6877803c4d778def43d8a122f27b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Mar 2015 17:22:12 +0100 Subject: [PATCH 272/411] ALSA: hda - Fix possible runtime PM refcount unbalance When the driver is unloaded before the codec is bound, it still keeps the runtime PM refcount up, and results in the unbalance. This patch covers these cases by introducing a flag indicating the runtime PM initialization and handling the codec registration procedure more properly. It also fixes the missing input beep device as a gratis, too. Signed-off-by: Takashi Iwai --- sound/hda/hdac_device.c | 3 +-- sound/pci/hda/hda_bind.c | 1 + sound/pci/hda/hda_codec.c | 30 ++++++++++++++++++++++-------- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_local.h | 1 + 5 files changed, 26 insertions(+), 10 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index aaece36247e772..6e8ee1d6974a23 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -109,7 +109,6 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, return 0; error: - pm_runtime_put_noidle(&codec->dev); put_device(&codec->dev); return err; } @@ -121,7 +120,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); */ void snd_hdac_device_exit(struct hdac_device *codec) { - /* pm_runtime_put_noidle(&codec->dev); */ + pm_runtime_put_noidle(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 130f672e6f37b2..7b269c3237e3cd 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -95,6 +95,7 @@ static int hda_codec_driver_probe(struct device *dev) err = snd_card_register(codec->card); if (err < 0) goto error_module; + snd_hda_codec_register(codec); } return 0; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 36483f7dd3ce7e..145cae7903b619 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -912,6 +912,13 @@ static void codec_release_pcms(struct hda_codec *codec) void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) { + if (codec->registered) { + /* pm_runtime_put() is called in snd_hdac_device_exit() */ + pm_runtime_get_noresume(hda_codec_dev(codec)); + pm_runtime_disable(hda_codec_dev(codec)); + codec->registered = 0; + } + cancel_delayed_work_sync(&codec->jackpoll_work); if (!codec->in_freeing) snd_hda_ctls_clear(codec); @@ -943,15 +950,23 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) static unsigned int hda_set_power_state(struct hda_codec *codec, unsigned int power_state); -static int snd_hda_codec_dev_register(struct snd_device *device) +/* also called from hda_bind.c */ +void snd_hda_codec_register(struct hda_codec *codec) { - struct hda_codec *codec = device->device_data; - - snd_hda_register_beep_device(codec); - if (device_is_registered(hda_codec_dev(codec))) + if (codec->registered) + return; + if (device_is_registered(hda_codec_dev(codec))) { + snd_hda_register_beep_device(codec); pm_runtime_enable(hda_codec_dev(codec)); - /* it was powered up in snd_hda_codec_new(), now all done */ - snd_hda_power_down(codec); + /* it was powered up in snd_hda_codec_new(), now all done */ + snd_hda_power_down(codec); + codec->registered = 1; + } +} + +static int snd_hda_codec_dev_register(struct snd_device *device) +{ + snd_hda_codec_register(device->device_data); return 0; } @@ -1094,7 +1109,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, return 0; error: - pm_runtime_put_noidle(hda_codec_dev(codec)); put_device(hda_codec_dev(codec)); return err; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e7c47a43976280..76776164623de0 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -314,6 +314,7 @@ struct hda_codec { /* misc flags */ unsigned int in_freeing:1; /* being released */ + unsigned int registered:1; /* codec was registered */ unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each * status change * (e.g. Realtek codecs) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index e0db30c66e5fcc..8a83775e0e2735 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -150,6 +150,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, #define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL) int snd_hda_codec_reset(struct hda_codec *codec); +void snd_hda_codec_register(struct hda_codec *codec); void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); enum { From e311782acd196d17d25b323d115709c50c8f7d3f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 15:16:28 +0100 Subject: [PATCH 273/411] ALSA: hda - Re-add tracepoints to HD-audio core driver Now let's take the basic tracepoints back to the HD-audio driver. The three bus tracepoints, hda_send_cmd, hda_get_response and hda_unsol_event are revived but in a slightly different form. Since we don't assign the card number there, print the bus device name instead. Signed-off-by: Takashi Iwai --- sound/hda/Makefile | 3 + sound/hda/hdac_bus.c | 7 ++- sound/hda/trace.c | 6 ++ sound/hda/trace.h | 62 ++++++++++++++++++++ sound/pci/hda/Makefile | 1 - sound/pci/hda/hda_trace.h | 119 -------------------------------------- 6 files changed, 77 insertions(+), 121 deletions(-) create mode 100644 sound/hda/trace.c create mode 100644 sound/hda/trace.h delete mode 100644 sound/pci/hda/hda_trace.h diff --git a/sound/hda/Makefile b/sound/hda/Makefile index ae8b5128b5c320..eec5da03b41f55 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -1,3 +1,6 @@ snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o +snd-hda-core-objs += trace.o +CFLAGS_trace.o := -I$(src) + obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 364f64c0e4a313..8e262da74f6a4b 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -7,6 +7,7 @@ #include #include #include +#include "trace.h" static void process_unsol_events(struct work_struct *work); @@ -82,6 +83,7 @@ int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr, else if (bus->sync_write) res = &tmp; for (;;) { + trace_hda_send_cmd(bus, cmd); err = bus->ops->command(bus, cmd); if (err != -EAGAIN) break; @@ -90,8 +92,10 @@ int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr, if (err) break; } - if (!err && res) + if (!err && res) { err = bus->ops->get_response(bus, addr, res); + trace_hda_get_response(bus, addr, *res); + } return err; } EXPORT_SYMBOL_GPL(snd_hdac_bus_exec_verb_unlocked); @@ -113,6 +117,7 @@ void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex) if (!bus) return; + trace_hda_unsol_event(bus, res, res_ex); wp = (bus->unsol_wp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_wp = wp; diff --git a/sound/hda/trace.c b/sound/hda/trace.c new file mode 100644 index 00000000000000..ca2d6bd94518e0 --- /dev/null +++ b/sound/hda/trace.c @@ -0,0 +1,6 @@ +/* + * tracepoint definitions for HD-audio core drivers + */ + +#define CREATE_TRACE_POINTS +#include "trace.h" diff --git a/sound/hda/trace.h b/sound/hda/trace.h new file mode 100644 index 00000000000000..33a7eb5573d4ce --- /dev/null +++ b/sound/hda/trace.h @@ -0,0 +1,62 @@ +#undef TRACE_SYSTEM +#define TRACE_SYSTEM hda + +#if !defined(__HDAC_TRACE_H) || defined(TRACE_HEADER_MULTI_READ) +#define __HDAC_TRACE_H + +#include +#include +#include + +#ifndef HDAC_MSG_MAX +#define HDAC_MSG_MAX 500 +#endif + +struct hdac_bus; +struct hdac_codec; + +TRACE_EVENT(hda_send_cmd, + TP_PROTO(struct hdac_bus *bus, unsigned int cmd), + TP_ARGS(bus, cmd), + TP_STRUCT__entry(__dynamic_array(char, msg, HDAC_MSG_MAX)), + TP_fast_assign( + snprintf(__get_str(msg), HDAC_MSG_MAX, + "[%s:%d] val=0x%08x", + dev_name((bus)->dev), (cmd) >> 28, cmd); + ), + TP_printk("%s", __get_str(msg)) +); + +TRACE_EVENT(hda_get_response, + TP_PROTO(struct hdac_bus *bus, unsigned int addr, unsigned int res), + TP_ARGS(bus, addr, res), + TP_STRUCT__entry(__dynamic_array(char, msg, HDAC_MSG_MAX)), + TP_fast_assign( + snprintf(__get_str(msg), HDAC_MSG_MAX, + "[%s:%d] val=0x%08x", + dev_name((bus)->dev), addr, res); + ), + TP_printk("%s", __get_str(msg)) +); + +TRACE_EVENT(hda_unsol_event, + TP_PROTO(struct hdac_bus *bus, u32 res, u32 res_ex), + TP_ARGS(bus, res, res_ex), + TP_STRUCT__entry(__dynamic_array(char, msg, HDAC_MSG_MAX)), + TP_fast_assign( + snprintf(__get_str(msg), HDAC_MSG_MAX, + "[%s:%d] res=0x%08x, res_ex=0x%08x", + dev_name((bus)->dev), res_ex & 0x0f, res, res_ex); + ), + TP_printk("%s", __get_str(msg)) +); +#endif /* __HDAC_TRACE_H */ + +/* This part must be outside protection */ +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . + +#undef TRACE_INCLUDE_FILE +#define TRACE_INCLUDE_FILE trace + +#include diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 96caaebfc19d25..af78fb33a4fd8d 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -10,7 +10,6 @@ snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o # for trace-points -CFLAGS_hda_codec.o := -I$(src) CFLAGS_hda_controller.o := -I$(src) snd-hda-codec-generic-objs := hda_generic.o diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h deleted file mode 100644 index 7fedfa86241970..00000000000000 --- a/sound/pci/hda/hda_trace.h +++ /dev/null @@ -1,119 +0,0 @@ -#undef TRACE_SYSTEM -#define TRACE_SYSTEM hda -#define TRACE_INCLUDE_FILE hda_trace - -#if !defined(_TRACE_HDA_H) || defined(TRACE_HEADER_MULTI_READ) -#define _TRACE_HDA_H - -#include - -struct hda_bus; -struct hda_codec; - -DECLARE_EVENT_CLASS(hda_cmd, - - TP_PROTO(struct hda_codec *codec, unsigned int val), - - TP_ARGS(codec, val), - - TP_STRUCT__entry( - __field( unsigned int, card ) - __field( unsigned int, addr ) - __field( unsigned int, val ) - ), - - TP_fast_assign( - __entry->card = (codec)->card->number; - __entry->addr = (codec)->addr; - __entry->val = (val); - ), - - TP_printk("[%d:%d] val=%x", __entry->card, __entry->addr, __entry->val) -); - -DEFINE_EVENT(hda_cmd, hda_send_cmd, - TP_PROTO(struct hda_codec *codec, unsigned int val), - TP_ARGS(codec, val) -); - -DEFINE_EVENT(hda_cmd, hda_get_response, - TP_PROTO(struct hda_codec *codec, unsigned int val), - TP_ARGS(codec, val) -); - -TRACE_EVENT(hda_bus_reset, - - TP_PROTO(struct hda_bus *bus), - - TP_ARGS(bus), - - TP_STRUCT__entry( - __field( unsigned int, card ) - ), - - TP_fast_assign( - __entry->card = (bus)->card->number; - ), - - TP_printk("[%d]", __entry->card) -); - -#ifdef CONFIG_PM -DECLARE_EVENT_CLASS(hda_power, - - TP_PROTO(struct hda_codec *codec), - - TP_ARGS(codec), - - TP_STRUCT__entry( - __field( unsigned int, card ) - __field( unsigned int, addr ) - ), - - TP_fast_assign( - __entry->card = (codec)->card->number; - __entry->addr = (codec)->addr; - ), - - TP_printk("[%d:%d]", __entry->card, __entry->addr) -); - -DEFINE_EVENT(hda_power, hda_power_down, - TP_PROTO(struct hda_codec *codec), - TP_ARGS(codec) -); - -DEFINE_EVENT(hda_power, hda_power_up, - TP_PROTO(struct hda_codec *codec), - TP_ARGS(codec) -); -#endif /* CONFIG_PM */ - -TRACE_EVENT(hda_unsol_event, - - TP_PROTO(struct hda_bus *bus, u32 res, u32 res_ex), - - TP_ARGS(bus, res, res_ex), - - TP_STRUCT__entry( - __field( unsigned int, card ) - __field( u32, res ) - __field( u32, res_ex ) - ), - - TP_fast_assign( - __entry->card = (bus)->card->number; - __entry->res = res; - __entry->res_ex = res_ex; - ), - - TP_printk("[%d] res=%x, res_ex=%x", __entry->card, - __entry->res, __entry->res_ex) -); - -#endif /* _TRACE_HDA_H */ - -/* This part must be outside protection */ -#undef TRACE_INCLUDE_PATH -#define TRACE_INCLUDE_PATH . -#include From 71fc4c7ef5ef2d0ddd22f0545ede4c135b554b84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Mar 2015 17:33:10 +0100 Subject: [PATCH 274/411] ALSA: hda - Move generic array helpers to core lib This will be used by the regmap support. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 33 ++++++++++++++++++++++++++ sound/hda/Makefile | 3 ++- sound/hda/array.c | 49 +++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.c | 46 ------------------------------------ sound/pci/hda/hda_codec.h | 30 ------------------------ 5 files changed, 84 insertions(+), 77 deletions(-) create mode 100644 sound/hda/array.c diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 675614dc2b8803..3abdd3f1652807 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -21,6 +21,17 @@ struct hdac_widget_tree; */ extern struct bus_type snd_hda_bus_type; +/* + * generic arrays + */ +struct snd_array { + unsigned int used; + unsigned int alloced; + unsigned int elem_size; + unsigned int alloc_align; + void *list; +}; + /* * HD-audio codec base device */ @@ -178,4 +189,26 @@ static inline void snd_hdac_codec_link_down(struct hdac_device *codec) clear_bit(codec->addr, &codec->bus->codec_powered); } +/* + * generic array helpers + */ +void *snd_array_new(struct snd_array *array); +void snd_array_free(struct snd_array *array); +static inline void snd_array_init(struct snd_array *array, unsigned int size, + unsigned int align) +{ + array->elem_size = size; + array->alloc_align = align; +} + +static inline void *snd_array_elem(struct snd_array *array, unsigned int idx) +{ + return array->list + idx * array->elem_size; +} + +static inline unsigned int snd_array_index(struct snd_array *array, void *ptr) +{ + return (unsigned long)(ptr - array->list) / array->elem_size; +} + #endif /* __SOUND_HDAUDIO_H */ diff --git a/sound/hda/Makefile b/sound/hda/Makefile index eec5da03b41f55..e508ba1102cb44 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -1,4 +1,5 @@ -snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o +snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \ + array.o snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) diff --git a/sound/hda/array.c b/sound/hda/array.c new file mode 100644 index 00000000000000..516795baa7db60 --- /dev/null +++ b/sound/hda/array.c @@ -0,0 +1,49 @@ +/* + * generic arrays + */ + +#include +#include +#include + +/** + * snd_array_new - get a new element from the given array + * @array: the array object + * + * Get a new element from the given array. If it exceeds the + * pre-allocated array size, re-allocate the array. + * + * Returns NULL if allocation failed. + */ +void *snd_array_new(struct snd_array *array) +{ + if (snd_BUG_ON(!array->elem_size)) + return NULL; + if (array->used >= array->alloced) { + int num = array->alloced + array->alloc_align; + int size = (num + 1) * array->elem_size; + void *nlist; + if (snd_BUG_ON(num >= 4096)) + return NULL; + nlist = krealloc(array->list, size, GFP_KERNEL | __GFP_ZERO); + if (!nlist) + return NULL; + array->list = nlist; + array->alloced = num; + } + return snd_array_elem(array, array->used++); +} +EXPORT_SYMBOL_GPL(snd_array_new); + +/** + * snd_array_free - free the given array elements + * @array: the array object + */ +void snd_array_free(struct snd_array *array) +{ + kfree(array->list); + array->used = 0; + array->alloced = 0; + array->list = NULL; +} +EXPORT_SYMBOL_GPL(snd_array_free); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 145cae7903b619..10e257ff908422 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5012,52 +5012,6 @@ void snd_hda_bus_reset(struct hda_bus *bus) } EXPORT_SYMBOL_GPL(snd_hda_bus_reset); -/* - * generic arrays - */ - -/** - * snd_array_new - get a new element from the given array - * @array: the array object - * - * Get a new element from the given array. If it exceeds the - * pre-allocated array size, re-allocate the array. - * - * Returns NULL if allocation failed. - */ -void *snd_array_new(struct snd_array *array) -{ - if (snd_BUG_ON(!array->elem_size)) - return NULL; - if (array->used >= array->alloced) { - int num = array->alloced + array->alloc_align; - int size = (num + 1) * array->elem_size; - void *nlist; - if (snd_BUG_ON(num >= 4096)) - return NULL; - nlist = krealloc(array->list, size, GFP_KERNEL | __GFP_ZERO); - if (!nlist) - return NULL; - array->list = nlist; - array->alloced = num; - } - return snd_array_elem(array, array->used++); -} -EXPORT_SYMBOL_GPL(snd_array_new); - -/** - * snd_array_free - free the given array elements - * @array: the array object - */ -void snd_array_free(struct snd_array *array) -{ - kfree(array->list); - array->used = 0; - array->alloced = 0; - array->list = NULL; -} -EXPORT_SYMBOL_GPL(snd_array_free); - /** * snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer * @pcm: PCM caps bits diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 76776164623de0..3068163b3db2b4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -29,36 +29,6 @@ #include #include -/* - * generic arrays - */ -struct snd_array { - unsigned int used; - unsigned int alloced; - unsigned int elem_size; - unsigned int alloc_align; - void *list; -}; - -void *snd_array_new(struct snd_array *array); -void snd_array_free(struct snd_array *array); -static inline void snd_array_init(struct snd_array *array, unsigned int size, - unsigned int align) -{ - array->elem_size = size; - array->alloc_align = align; -} - -static inline void *snd_array_elem(struct snd_array *array, unsigned int idx) -{ - return array->list + idx * array->elem_size; -} - -static inline unsigned int snd_array_index(struct snd_array *array, void *ptr) -{ - return (unsigned long)(ptr - array->list) / array->elem_size; -} - /* * Structures */ From 4d75faa0448a6bb2fd6d56051a3675a3937cbada Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Feb 2015 14:42:38 +0100 Subject: [PATCH 275/411] ALSA: hda - Add regmap support This patch adds an infrastructure to support regmap-based verb accesses. Because o the asymmetric nature of HD-audio verbs, especially the amp verbs, we need to translate the verbs as a sort of pseudo registers to be mapped uniquely in regmap. In this patch, a pseudo register is built from the NID, the AC_VERB_GET_* and 8bit parameters, i.e. almost in the form to be sent to HD-audio bus but without codec address field. OTOH, for writing, the same pseudo register is translated to AC_VERB_SET_* automatically. The AC_VERB_SET_AMP_* verb is re-encoded from the corresponding AC_VERB_GET_AMP_* verb and parameter at writing. Some verbs has a single command for read but multiple for writes. A write for such a verb is split automatically to multiple verbs. The patch provides also a few handy helper functions. They are designed to be accessible even without regmap. When no regmap is set up (e.g. before the codec device instantiation), the direct hardware access is used. Also, it tries to avoid the unnecessary power-up. The power up/down sequence is performed only on demand. The codec driver needs to call snd_hdac_regmap_exit() and snd_hdac_regmap_exit() at probe and remove if it wants the regmap access. There is one flag added to hdac_device. When the flag lazy_cache is set, regmap helper ignores a write for a suspended device and returns as if it was actually written. It reduces the hardware access pretty much, e.g. when adjusting the mixer volume while in idle. This assumes that the driver will sync the cache later at resume properly, so use it carefully. Signed-off-by: Takashi Iwai --- include/sound/hda_regmap.h | 145 ++++++++++++++++++ include/sound/hdaudio.h | 4 + sound/hda/Kconfig | 1 + sound/hda/Makefile | 2 +- sound/hda/hdac_regmap.c | 304 +++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_bind.c | 4 + sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_codec.h | 1 + 8 files changed, 461 insertions(+), 1 deletion(-) create mode 100644 include/sound/hda_regmap.h create mode 100644 sound/hda/hdac_regmap.c diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h new file mode 100644 index 00000000000000..95651d26437d47 --- /dev/null +++ b/include/sound/hda_regmap.h @@ -0,0 +1,145 @@ +/* + * HD-audio regmap helpers + */ + +#ifndef __SOUND_HDA_REGMAP_H +#define __SOUND_HDA_REGMAP_H + +#include +#include +#include + +int snd_hdac_regmap_init(struct hdac_device *codec); +void snd_hdac_regmap_exit(struct hdac_device *codec); + +int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, + unsigned int *val); +int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, + unsigned int val); +int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, + unsigned int mask, unsigned int val); + +/** + * snd_hdac_regmap_encode_verb - encode the verb to a pseudo register + * @nid: widget NID + * @verb: codec verb + * + * Returns an encoded pseudo register. + */ +#define snd_hdac_regmap_encode_verb(nid, verb) \ + (((verb) << 8) | 0x80000 | ((unsigned int)(nid) << 20)) + +/** + * snd_hdac_regmap_encode_amp - encode the AMP verb to a pseudo register + * @nid: widget NID + * @ch: channel (left = 0, right = 1) + * @dir: direction (#HDA_INPUT, #HDA_OUTPUT) + * @idx: input index value + * + * Returns an encoded pseudo register. + */ +#define snd_hdac_regmap_encode_amp(nid, ch, dir, idx) \ + (snd_hdac_regmap_encode_verb(nid, AC_VERB_GET_AMP_GAIN_MUTE) | \ + ((ch) ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT) | \ + ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \ + (idx)) + +/** + * snd_hdac_regmap_write - Write a verb with caching + * @nid: codec NID + * @reg: verb to write + * @val: value to write + * + * For writing an amp value, use snd_hda_regmap_amp_update(). + */ +static inline int +snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, + unsigned int verb, unsigned int val) +{ + unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb); + + return snd_hdac_regmap_write_raw(codec, cmd, val); +} + +/** + * snd_hda_regmap_update - Update a verb value with caching + * @nid: codec NID + * @verb: verb to update + * @mask: bit mask to update + * @val: value to update + * + * For updating an amp value, use snd_hda_regmap_amp_update(). + */ +static inline int +snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid, + unsigned int verb, unsigned int mask, + unsigned int val) +{ + unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb); + + return snd_hdac_regmap_update_raw(codec, cmd, mask, val); +} + +/** + * snd_hda_regmap_read - Read a verb with caching + * @nid: codec NID + * @verb: verb to read + * @val: pointer to store the value + * + * For reading an amp value, use snd_hda_regmap_get_amp(). + */ +static inline int +snd_hdac_regmap_read(struct hdac_device *codec, hda_nid_t nid, + unsigned int verb, unsigned int *val) +{ + unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb); + + return snd_hdac_regmap_read_raw(codec, cmd, val); +} + +/** + * snd_hdac_regmap_get_amp - Read AMP value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel (left=0 or right=1) + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @index: the index value (only for input direction) + * @val: the pointer to store the value + * + * Read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. + * Returns the value or a negative error. + */ +static inline int +snd_hdac_regmap_get_amp(struct hdac_device *codec, hda_nid_t nid, + int ch, int dir, int idx) +{ + unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx); + int err, val; + + err = snd_hdac_regmap_read_raw(codec, cmd, &val); + return err < 0 ? err : val; +} + +/** + * snd_hdac_regmap_update_amp - update the AMP value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel (left=0 or right=1) + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Update the AMP value with a bit mask. + * Returns 0 if the value is unchanged, 1 if changed, or a negative error. + */ +static inline int +snd_hdac_regmap_update_amp(struct hdac_device *codec, hda_nid_t nid, + int ch, int dir, int idx, int mask, int val) +{ + unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx); + + return snd_hdac_regmap_update_raw(codec, cmd, mask, val); +} + +#endif /* __SOUND_HDA_REGMAP_H */ diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 3abdd3f1652807..47e20b741c51e1 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -72,6 +72,10 @@ struct hdac_device { /* sysfs */ struct hdac_widget_tree *widgets; + + /* regmap */ + struct regmap *regmap; + bool lazy_cache:1; /* don't wake up for writes */ }; /* device/driver type used for matching */ diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 4f428ccf64adea..001c6588a5ffbb 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -1,2 +1,3 @@ config SND_HDA_CORE tristate + select REGMAP diff --git a/sound/hda/Makefile b/sound/hda/Makefile index e508ba1102cb44..7a359f5b7e25fe 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -1,5 +1,5 @@ snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \ - array.o + hdac_regmap.o array.o snd-hda-core-objs += trace.o CFLAGS_trace.o := -I$(src) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c new file mode 100644 index 00000000000000..db03d60d9c9917 --- /dev/null +++ b/sound/hda/hdac_regmap.c @@ -0,0 +1,304 @@ +/* + * Regmap support for HD-audio verbs + * + * A virtual register is translated to one or more hda verbs for write, + * vice versa for read. + * + * A few limitations: + * - Provided for not all verbs but only subset standard non-volatile verbs. + * - For reading, only AC_VERB_GET_* variants can be used. + * - For writing, mapped to the *corresponding* AC_VERB_SET_* variants, + * so can't handle asymmetric verbs for read and write + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#ifdef CONFIG_PM +#define codec_is_running(codec) \ + (atomic_read(&(codec)->in_pm) || \ + !pm_runtime_suspended(&(codec)->dev)) +#else +#define codec_is_running(codec) true +#endif + +#define get_verb(reg) (((reg) >> 8) & 0xfff) + +static bool hda_volatile_reg(struct device *dev, unsigned int reg) +{ + unsigned int verb = get_verb(reg); + + switch (verb) { + case AC_VERB_GET_PROC_COEF: + case AC_VERB_GET_COEF_INDEX: + case AC_VERB_GET_PROC_STATE: + case AC_VERB_GET_POWER_STATE: + case AC_VERB_GET_PIN_SENSE: + case AC_VERB_GET_HDMI_DIP_SIZE: + case AC_VERB_GET_HDMI_ELDD: + case AC_VERB_GET_HDMI_DIP_INDEX: + case AC_VERB_GET_HDMI_DIP_DATA: + case AC_VERB_GET_HDMI_DIP_XMIT: + case AC_VERB_GET_HDMI_CP_CTRL: + case AC_VERB_GET_HDMI_CHAN_SLOT: + case AC_VERB_GET_DEVICE_SEL: + case AC_VERB_GET_DEVICE_LIST: /* read-only volatile */ + return true; + } + + return false; +} + +static bool hda_writeable_reg(struct device *dev, unsigned int reg) +{ + unsigned int verb = get_verb(reg); + + switch (verb & 0xf00) { + case AC_VERB_GET_STREAM_FORMAT: + case AC_VERB_GET_AMP_GAIN_MUTE: + return true; + case 0xf00: + break; + default: + return false; + } + + switch (verb) { + case AC_VERB_GET_CONNECT_SEL: + case AC_VERB_GET_SDI_SELECT: + case AC_VERB_GET_CONV: + case AC_VERB_GET_PIN_WIDGET_CONTROL: + case AC_VERB_GET_UNSOLICITED_RESPONSE: /* only as SET_UNSOLICITED_ENABLE */ + case AC_VERB_GET_BEEP_CONTROL: + case AC_VERB_GET_EAPD_BTLENABLE: + case AC_VERB_GET_DIGI_CONVERT_1: + case AC_VERB_GET_DIGI_CONVERT_2: /* only for beep control */ + case AC_VERB_GET_VOLUME_KNOB_CONTROL: + case AC_VERB_GET_CONFIG_DEFAULT: + case AC_VERB_GET_GPIO_MASK: + case AC_VERB_GET_GPIO_DIRECTION: + case AC_VERB_GET_GPIO_DATA: /* not for volatile read */ + case AC_VERB_GET_GPIO_WAKE_MASK: + case AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK: + case AC_VERB_GET_GPIO_STICKY_MASK: + case AC_VERB_GET_CVT_CHAN_COUNT: + return true; + } + + return false; +} + +static bool hda_readable_reg(struct device *dev, unsigned int reg) +{ + unsigned int verb = get_verb(reg); + + switch (verb) { + case AC_VERB_PARAMETERS: + case AC_VERB_GET_CONNECT_LIST: + case AC_VERB_GET_SUBSYSTEM_ID: + return true; + } + + return hda_writeable_reg(dev, reg); +} + +static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) +{ + struct hdac_device *codec = context; + + if (!codec_is_running(codec)) + return -EAGAIN; + reg |= (codec->addr << 28); + return snd_hdac_exec_verb(codec, reg, 0, val); +} + +static int hda_reg_write(void *context, unsigned int reg, unsigned int val) +{ + struct hdac_device *codec = context; + unsigned int verb; + int i, bytes, err; + + if (!codec_is_running(codec)) + return codec->lazy_cache ? 0 : -EAGAIN; + + reg &= ~0x00080000U; /* drop GET bit */ + reg |= (codec->addr << 28); + verb = get_verb(reg); + + switch (verb & 0xf00) { + case AC_VERB_SET_AMP_GAIN_MUTE: + verb = AC_VERB_SET_AMP_GAIN_MUTE; + if (reg & AC_AMP_GET_LEFT) + verb |= AC_AMP_SET_LEFT >> 8; + else + verb |= AC_AMP_SET_RIGHT >> 8; + if (reg & AC_AMP_GET_OUTPUT) { + verb |= AC_AMP_SET_OUTPUT >> 8; + } else { + verb |= AC_AMP_SET_INPUT >> 8; + verb |= reg & 0xf; + } + break; + } + + switch (verb) { + case AC_VERB_SET_DIGI_CONVERT_1: + bytes = 2; + break; + case AC_VERB_SET_CONFIG_DEFAULT_BYTES_0: + bytes = 4; + break; + default: + bytes = 1; + break; + } + + for (i = 0; i < bytes; i++) { + reg &= ~0xfffff; + reg |= (verb + i) << 8 | ((val >> (8 * i)) & 0xff); + err = snd_hdac_exec_verb(codec, reg, 0, NULL); + if (err < 0) + return err; + } + + return 0; +} + +static const struct regmap_config hda_regmap_cfg = { + .name = "hdaudio", + .reg_bits = 32, + .val_bits = 32, + .max_register = 0xfffffff, + .writeable_reg = hda_writeable_reg, + .readable_reg = hda_readable_reg, + .volatile_reg = hda_volatile_reg, + .cache_type = REGCACHE_RBTREE, + .reg_read = hda_reg_read, + .reg_write = hda_reg_write, +}; + +int snd_hdac_regmap_init(struct hdac_device *codec) +{ + struct regmap *regmap; + + regmap = regmap_init(&codec->dev, NULL, codec, &hda_regmap_cfg); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + codec->regmap = regmap; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); + +void snd_hdac_regmap_exit(struct hdac_device *codec) +{ + if (codec->regmap) { + regmap_exit(codec->regmap); + codec->regmap = NULL; + } +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_exit); + +/* + * helper functions + */ + +/* write a pseudo-register value (w/o power sequence) */ +static int reg_raw_write(struct hdac_device *codec, unsigned int reg, + unsigned int val) +{ + if (!codec->regmap) + return hda_reg_write(codec, reg, val); + else + return regmap_write(codec->regmap, reg, val); +} + +/** + * snd_hdac_regmap_write_raw - write a pseudo register with power mgmt + * @codec: the codec object + * @reg: pseudo register + * @val: value to write + * + * Returns zero if successful or a negative error code. + */ +int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, + unsigned int val) +{ + int err; + + err = reg_raw_write(codec, reg, val); + if (err == -EAGAIN) { + snd_hdac_power_up(codec); + err = reg_raw_write(codec, reg, val); + snd_hdac_power_down(codec); + } + return err; +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_write_raw); + +static int reg_raw_read(struct hdac_device *codec, unsigned int reg, + unsigned int *val) +{ + if (!codec->regmap) + return hda_reg_read(codec, reg, val); + else + return regmap_read(codec->regmap, reg, val); +} + +/** + * snd_hdac_regmap_read_raw - read a pseudo register with power mgmt + * @codec: the codec object + * @reg: pseudo register + * @val: pointer to store the read value + * + * Returns zero if successful or a negative error code. + */ +int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, + unsigned int *val) +{ + int err; + + err = reg_raw_read(codec, reg, val); + if (err == -EAGAIN) { + snd_hdac_power_up(codec); + err = reg_raw_read(codec, reg, val); + snd_hdac_power_down(codec); + } + return err; +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_read_raw); + +/** + * snd_hdac_regmap_update_raw - update a pseudo register with power mgmt + * @codec: the codec object + * @reg: pseudo register + * @mask: bit mask to udpate + * @val: value to update + * + * Returns zero if successful or a negative error code. + */ +int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, + unsigned int mask, unsigned int val) +{ + unsigned int orig; + int err; + + val &= mask; + err = snd_hdac_regmap_read_raw(codec, reg, &orig); + if (err < 0) + return err; + val |= orig & ~mask; + if (val == orig) + return 0; + err = snd_hdac_regmap_write_raw(codec, reg, val); + if (err < 0) + return err; + return 1; +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 7b269c3237e3cd..00aa31c5f08e3b 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -73,6 +73,9 @@ static int hda_codec_driver_probe(struct device *dev) return -EINVAL; err = codec_refresh_name(codec, codec->preset->name); + if (err < 0) + goto error; + err = snd_hdac_regmap_init(&codec->core); if (err < 0) goto error; @@ -98,6 +101,7 @@ static int hda_codec_driver_probe(struct device *dev) snd_hda_codec_register(codec); } + codec->core.lazy_cache = true; return 0; error_module: diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 10e257ff908422..8eb42f4226ff38 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -945,6 +945,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) snd_array_free(&codec->mixers); snd_array_free(&codec->nids); remove_conn_list(codec); + snd_hdac_regmap_exit(&codec->core); } static unsigned int hda_set_power_state(struct hda_codec *codec, diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 3068163b3db2b4..0f5749ca1600c5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -28,6 +28,7 @@ #include #include #include +#include /* * Structures From 01ed3c06c6d5e7e861650ae76117dd4194d87316 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2015 13:57:47 +0100 Subject: [PATCH 276/411] ALSA: hda - Use regmap for codec parameter reads Let's start converting the access functions to regmap. The first one is the simplest, just converting the codec parameter read helper function snd_hda_param_read(). Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 20 +++++++++++++++++++- sound/hda/hdac_device.c | 21 +++++++++------------ sound/pci/hda/hda_codec.h | 2 +- 3 files changed, 29 insertions(+), 14 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 47e20b741c51e1..ddfcc44970fac2 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -105,12 +105,30 @@ int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd, unsigned int flags, unsigned int *res); int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid, unsigned int verb, unsigned int parm, unsigned int *res); -int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); +int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm, + unsigned int *res); int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *start_id); +/** + * snd_hdac_read_parm - read a codec parameter + * @codec: the codec object + * @nid: NID to read a parameter + * @parm: parameter to read + * + * Returns -1 for error. If you need to distinguish the error more + * strictly, use _snd_hdac_read_parm() directly. + */ +static inline int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, + int parm) +{ + unsigned int val; + + return _snd_hdac_read_parm(codec, nid, parm, &val) < 0 ? -1 : val; +} + #ifdef CONFIG_PM void snd_hdac_power_up(struct hdac_device *codec); void snd_hdac_power_down(struct hdac_device *codec); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 6e8ee1d6974a23..ba9c1fc6e3ea1b 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -9,6 +9,7 @@ #include #include #include +#include #include "local.h" static void setup_fg_nodes(struct hdac_device *codec); @@ -234,23 +235,19 @@ int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid, EXPORT_SYMBOL_GPL(snd_hdac_read); /** - * snd_hdac_read_parm - read a codec parameter - * @codec: the codec object - * @nid: NID to read a parameter - * @parm: parameter to read + * _snd_hdac_read_parm - read a parmeter * - * Returns -1 for error. If you need to distinguish the error more - * strictly, use snd_hdac_read() directly. + * This function returns zero or an error unlike snd_hdac_read_parm(). */ -int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm) +int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm, + unsigned int *res) { - int val; + unsigned int cmd; - if (snd_hdac_read(codec, nid, AC_VERB_PARAMETERS, parm, &val)) - return -1; - return val; + cmd = snd_hdac_regmap_encode_verb(nid, AC_VERB_PARAMETERS) | parm; + return snd_hdac_regmap_read_raw(codec, cmd, res); } -EXPORT_SYMBOL_GPL(snd_hdac_read_parm); +EXPORT_SYMBOL_GPL(_snd_hdac_read_parm); /** * snd_hdac_get_sub_nodes - get start NID and number of subtree nodes diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0f5749ca1600c5..135b70f066f1ad 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -369,7 +369,7 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm); #define snd_hda_param_read(codec, nid, param) \ - snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param) + snd_hdac_read_parm(&(codec)->core, nid, param) #define snd_hda_get_sub_nodes(codec, nid, start_nid) \ snd_hdac_get_sub_nodes(&(codec)->core, nid, start_nid) int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, From 9ba17b4d132f56a680fa1ba0bc2a8f98b6263d93 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Mar 2015 23:29:47 +0100 Subject: [PATCH 277/411] ALSA: hda - Implement uncached version of parameter reads Sometimes we need the uncached reads, e.g. for refreshing the tree. This patch provides the helper function for that and uses it for refreshing widgets, reading subtrees and the whole proc reads. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 2 ++ sound/hda/hdac_device.c | 26 +++++++++++++++++++++++++- sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/hda_proc.c | 28 ++++++++++++++-------------- 4 files changed, 43 insertions(+), 17 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index ddfcc44970fac2..65ea6429f3a7fe 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -107,6 +107,8 @@ int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid, unsigned int verb, unsigned int parm, unsigned int *res); int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm, unsigned int *res); +int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid, + int parm); int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index ba9c1fc6e3ea1b..0dac746df7da9c 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -249,6 +249,29 @@ int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm, } EXPORT_SYMBOL_GPL(_snd_hdac_read_parm); +/** + * snd_hdac_read_parm_uncached - read a codec parameter without caching + * @codec: the codec object + * @nid: NID to read a parameter + * @parm: parameter to read + * + * Returns -1 for error. If you need to distinguish the error more + * strictly, use snd_hdac_read() directly. + */ +int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid, + int parm) +{ + int val; + + if (codec->regmap) + regcache_cache_bypass(codec->regmap, true); + val = snd_hdac_read_parm(codec, nid, parm); + if (codec->regmap) + regcache_cache_bypass(codec->regmap, false); + return val; +} +EXPORT_SYMBOL_GPL(snd_hdac_read_parm_uncached); + /** * snd_hdac_get_sub_nodes - get start NID and number of subtree nodes * @codec: the codec object @@ -256,13 +279,14 @@ EXPORT_SYMBOL_GPL(_snd_hdac_read_parm); * @start_id: the pointer to store the starting NID * * Returns the number of subtree nodes or zero if not found. + * This function reads parameters always without caching. */ int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *start_id) { unsigned int parm; - parm = snd_hdac_read_parm(codec, nid, AC_PAR_NODE_COUNT); + parm = snd_hdac_read_parm_uncached(codec, nid, AC_PAR_NODE_COUNT); if (parm == -1) { *start_id = 0; return 0; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8eb42f4226ff38..39b5660653f0f4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -586,8 +586,8 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) return -ENOMEM; nid = codec->core.start_nid; for (i = 0; i < codec->core.num_nodes; i++, nid++) - codec->wcaps[i] = snd_hda_param_read(codec, nid, - AC_PAR_AUDIO_WIDGET_CAP); + codec->wcaps[i] = snd_hdac_read_parm_uncached(&codec->core, + nid, AC_PAR_AUDIO_WIDGET_CAP); return 0; } diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index a4f5a30f1d41f2..ee6230767c641d 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -32,6 +32,10 @@ static int dump_coef = -1; module_param(dump_coef, int, 0644); MODULE_PARM_DESC(dump_coef, "Dump processing coefficients in codec proc file (-1=auto, 0=disable, 1=enable)"); +/* always use noncached version */ +#define param_read(codec, nid, parm) \ + snd_hdac_read_parm_uncached(&(codec)->core, nid, parm) + static char *bits_names(unsigned int bits, char *names[], int size) { int i, n; @@ -119,9 +123,8 @@ static void print_amp_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, int dir) { unsigned int caps; - caps = snd_hda_param_read(codec, nid, - dir == HDA_OUTPUT ? - AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); + caps = param_read(codec, nid, dir == HDA_OUTPUT ? + AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); if (caps == -1 || caps == 0) { snd_iprintf(buffer, "N/A\n"); return; @@ -225,8 +228,8 @@ static void print_pcm_formats(struct snd_info_buffer *buffer, static void print_pcm_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { - unsigned int pcm = snd_hda_param_read(codec, nid, AC_PAR_PCM); - unsigned int stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM); + unsigned int pcm = param_read(codec, nid, AC_PAR_PCM); + unsigned int stream = param_read(codec, nid, AC_PAR_STREAM); if (pcm == -1 || stream == -1) { snd_iprintf(buffer, "N/A\n"); return; @@ -273,7 +276,7 @@ static void print_pin_caps(struct snd_info_buffer *buffer, static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; unsigned int caps, val; - caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + caps = param_read(codec, nid, AC_PAR_PIN_CAP); snd_iprintf(buffer, " Pincap 0x%08x:", caps); if (caps & AC_PINCAP_IN) snd_iprintf(buffer, " IN"); @@ -401,8 +404,7 @@ static void print_pin_ctls(struct snd_info_buffer *buffer, static void print_vol_knob(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { - unsigned int cap = snd_hda_param_read(codec, nid, - AC_PAR_VOL_KNB_CAP); + unsigned int cap = param_read(codec, nid, AC_PAR_VOL_KNB_CAP); snd_iprintf(buffer, " Volume-Knob: delta=%d, steps=%d, ", (cap >> 7) & 1, cap & 0x7f); cap = snd_hda_codec_read(codec, nid, 0, @@ -487,7 +489,7 @@ static void print_power_state(struct snd_info_buffer *buffer, [ilog2(AC_PWRST_EPSS)] = "EPSS", }; - int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE); + int sup = param_read(codec, nid, AC_PAR_POWER_STATE); int pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); if (sup != -1) @@ -531,8 +533,7 @@ static void print_proc_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { unsigned int i, ncoeff, oldindex; - unsigned int proc_caps = snd_hda_param_read(codec, nid, - AC_PAR_PROC_CAP); + unsigned int proc_caps = param_read(codec, nid, AC_PAR_PROC_CAP); ncoeff = (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT; snd_iprintf(buffer, " Processing caps: benign=%d, ncoeff=%d\n", proc_caps & AC_PCAP_BENIGN, ncoeff); @@ -597,7 +598,7 @@ static void print_gpio(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { unsigned int gpio = - snd_hda_param_read(codec, codec->core.afg, AC_PAR_GPIO_CAP); + param_read(codec, codec->core.afg, AC_PAR_GPIO_CAP); unsigned int enable, direction, wake, unsol, sticky, data; int i, max; snd_iprintf(buffer, "GPIO: io=%d, o=%d, i=%d, " @@ -727,8 +728,7 @@ static void print_codec_info(struct snd_info_entry *entry, for (i = 0; i < nodes; i++, nid++) { unsigned int wid_caps = - snd_hda_param_read(codec, nid, - AC_PAR_AUDIO_WIDGET_CAP); + param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = get_wcaps_type(wid_caps); hda_nid_t *conn = NULL; int conn_len = 0; From eeecd9d10d3da0b9efd6f58fad55c4355dcc246a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Feb 2015 15:18:50 +0100 Subject: [PATCH 278/411] ALSA: hda - Use regmap for amp accesses This patch converts the amp access functions to the regmap helpers. The amp values were formerly cached in the own hash table. Now it's dropped by the regmap's cache. The only tricky conversion is snd_hda_codec_amp_init(). This function shouldn't do anything if the amp was already initialized. For achieving this behavior, a value is read once at first temporarily in the cache-only mode. Only if it returns an error, i.e. the item still doesn't exist in the cache, it proceeds to the update. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 215 ++++------------------------------ sound/pci/hda/hda_local.h | 10 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 4 +- 4 files changed, 32 insertions(+), 199 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 39b5660653f0f4..52962f697825b4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1543,139 +1543,6 @@ int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hda_override_pin_caps); -/* read or sync the hash value with the current value; - * call within hash_mutex - */ -static struct hda_amp_info * -update_amp_hash(struct hda_codec *codec, hda_nid_t nid, int ch, - int direction, int index, bool init_only) -{ - struct hda_amp_info *info; - unsigned int parm, val = 0; - bool val_read = false; - - retry: - info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index)); - if (!info) - return NULL; - if (!(info->head.val & INFO_AMP_VOL(ch))) { - if (!val_read) { - mutex_unlock(&codec->hash_mutex); - parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT; - parm |= direction == HDA_OUTPUT ? - AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT; - parm |= index; - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, parm); - val &= 0xff; - val_read = true; - mutex_lock(&codec->hash_mutex); - goto retry; - } - info->vol[ch] = val; - info->head.val |= INFO_AMP_VOL(ch); - } else if (init_only) - return NULL; - return info; -} - -/* - * write the current volume in info to the h/w - */ -static void put_vol_mute(struct hda_codec *codec, unsigned int amp_caps, - hda_nid_t nid, int ch, int direction, int index, - int val) -{ - u32 parm; - - parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT; - parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT; - parm |= index << AC_AMP_SET_INDEX_SHIFT; - if ((val & HDA_AMP_MUTE) && !(amp_caps & AC_AMPCAP_MUTE) && - (amp_caps & AC_AMPCAP_MIN_MUTE)) - ; /* set the zero value as a fake mute */ - else - parm |= val; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm); -} - -/** - * snd_hda_codec_amp_read - Read AMP value - * @codec: HD-audio codec - * @nid: NID to read the AMP value - * @ch: channel (left=0 or right=1) - * @direction: #HDA_INPUT or #HDA_OUTPUT - * @index: the index value (only for input direction) - * - * Read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. - */ -int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, - int direction, int index) -{ - struct hda_amp_info *info; - unsigned int val = 0; - - mutex_lock(&codec->hash_mutex); - info = update_amp_hash(codec, nid, ch, direction, index, false); - if (info) - val = info->vol[ch]; - mutex_unlock(&codec->hash_mutex); - return val; -} -EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read); - -static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, - int direction, int idx, int mask, int val, - bool init_only, bool cache_only) -{ - struct hda_amp_info *info; - unsigned int caps; - - if (snd_BUG_ON(mask & ~0xff)) - mask &= 0xff; - val &= mask; - - mutex_lock(&codec->hash_mutex); - info = update_amp_hash(codec, nid, ch, direction, idx, init_only); - if (!info) { - mutex_unlock(&codec->hash_mutex); - return 0; - } - val |= info->vol[ch] & ~mask; - if (info->vol[ch] == val) { - mutex_unlock(&codec->hash_mutex); - return 0; - } - info->vol[ch] = val; - info->head.dirty |= cache_only; - caps = info->amp_caps; - mutex_unlock(&codec->hash_mutex); - if (!cache_only) - put_vol_mute(codec, caps, nid, ch, direction, idx, val); - return 1; -} - -/** - * snd_hda_codec_amp_update - update the AMP value - * @codec: HD-audio codec - * @nid: NID to read the AMP value - * @ch: channel (left=0 or right=1) - * @direction: #HDA_INPUT or #HDA_OUTPUT - * @idx: the index value (only for input direction) - * @mask: bit mask to set - * @val: the bits value to set - * - * Update the AMP value with a bit mask. - * Returns 0 if the value is unchanged, 1 if changed. - */ -int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, - int direction, int idx, int mask, int val) -{ - return codec_amp_update(codec, nid, ch, direction, idx, mask, val, - false, codec->cached_write); -} -EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); - /** * snd_hda_codec_amp_stereo - update the AMP stereo values * @codec: HD-audio codec @@ -1719,8 +1586,16 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true, - codec->cached_write); + int orig; + + if (!codec->core.regmap) + return -EINVAL; + regcache_cache_only(codec->core.regmap, true); + orig = snd_hda_codec_amp_read(codec, nid, ch, dir, idx); + regcache_cache_only(codec->core.regmap, false); + if (orig >= 0) + return 0; + return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, mask, val); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); @@ -1749,49 +1624,6 @@ int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init_stereo); -/** - * snd_hda_codec_resume_amp - Resume all AMP commands from the cache - * @codec: HD-audio codec - * - * Resume the all amp commands from the cache. - */ -void snd_hda_codec_resume_amp(struct hda_codec *codec) -{ - int i; - - mutex_lock(&codec->hash_mutex); - codec->cached_write = 0; - for (i = 0; i < codec->amp_cache.buf.used; i++) { - struct hda_amp_info *buffer; - u32 key; - hda_nid_t nid; - unsigned int idx, dir, ch; - struct hda_amp_info info; - - buffer = snd_array_elem(&codec->amp_cache.buf, i); - if (!buffer->head.dirty) - continue; - buffer->head.dirty = 0; - info = *buffer; - key = info.head.key; - if (!key) - continue; - nid = key & 0xff; - idx = (key >> 16) & 0xff; - dir = (key >> 24) & 0xff; - for (ch = 0; ch < 2; ch++) { - if (!(info.head.val & INFO_AMP_VOL(ch))) - continue; - mutex_unlock(&codec->hash_mutex); - put_vol_mute(codec, info.amp_caps, nid, ch, dir, idx, - info.vol[ch]); - mutex_lock(&codec->hash_mutex); - } - } - mutex_unlock(&codec->hash_mutex); -} -EXPORT_SYMBOL_GPL(snd_hda_codec_resume_amp); - static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int ofs) { @@ -1862,8 +1694,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, maxval = get_amp_max_value(codec, nid, dir, 0); if (val > maxval) val = maxval; - return codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val, - false, !hda_codec_is_power_on(codec)); + return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, + HDA_AMP_VOLMASK, val); } /** @@ -2546,17 +2378,15 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, int change = 0; if (chs & 1) { - change = codec_amp_update(codec, nid, 0, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE, false, - !hda_codec_is_power_on(codec)); + change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE); valp++; } if (chs & 2) - change |= codec_amp_update(codec, nid, 1, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE, false, - !hda_codec_is_power_on(codec)); + change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE); hda_call_check_power_status(codec, nid); return change; } @@ -3417,7 +3247,8 @@ EXPORT_SYMBOL_GPL(snd_hda_sequence_write_cache); */ void snd_hda_codec_flush_cache(struct hda_codec *codec) { - snd_hda_codec_resume_amp(codec); + if (codec->core.regmap) + regcache_sync(codec->core.regmap); snd_hda_codec_resume_cache(codec); } EXPORT_SYMBOL_GPL(snd_hda_codec_flush_cache); @@ -3645,6 +3476,9 @@ static void hda_call_codec_resume(struct hda_codec *codec) { atomic_inc(&codec->core.in_pm); + if (codec->core.regmap) + regcache_mark_dirty(codec->core.regmap); + hda_mark_cmd_cache_dirty(codec); codec->power_jiffies = jiffies; @@ -3658,7 +3492,8 @@ static void hda_call_codec_resume(struct hda_codec *codec) else { if (codec->patch_ops.init) codec->patch_ops.init(codec); - snd_hda_codec_resume_amp(codec); + if (codec->core.regmap) + regcache_sync(codec->core.regmap); snd_hda_codec_resume_cache(codec); } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 8a83775e0e2735..7023eeee8b9d2d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -127,18 +127,16 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); #endif /* lowlevel accessor with caching; use carefully */ -int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, - int direction, int index); -int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, - int direction, int idx, int mask, int val); +#define snd_hda_codec_amp_read(codec, nid, ch, dir, idx) \ + snd_hdac_regmap_get_amp(&(codec)->core, nid, ch, dir, idx) +#define snd_hda_codec_amp_update(codec, nid, ch, dir, idx, mask, val) \ + snd_hdac_regmap_update_amp(&(codec)->core, nid, ch, dir, idx, mask, val) int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, int mask, int val); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val); int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, int mask, int val); -void snd_hda_codec_resume_amp(struct hda_codec *codec); - void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int *tlv); struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 35d92a8a99ce98..04c5ab20eb7676 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2211,7 +2211,7 @@ static int generic_hdmi_resume(struct hda_codec *codec) int pin_idx; codec->patch_ops.init(codec); - snd_hda_codec_resume_amp(codec); + regcache_sync(codec->core.regmap); snd_hda_codec_resume_cache(codec); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eee4532ab5f6db..a440e539230fc0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -799,7 +799,7 @@ static int alc_resume(struct hda_codec *codec) if (!spec->no_depop_delay) msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); - snd_hda_codec_resume_amp(codec); + regcache_sync(codec->core.regmap); snd_hda_codec_resume_cache(codec); hda_call_check_power_status(codec, 0x01); return 0; @@ -3058,7 +3058,7 @@ static int alc269_resume(struct hda_codec *codec) msleep(200); } - snd_hda_codec_resume_amp(codec); + regcache_sync(codec->core.regmap); snd_hda_codec_resume_cache(codec); hda_call_check_power_status(codec, 0x01); From faa75f8a2edf005a5caf43be4875ffeeb9bcb498 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2015 08:54:56 +0100 Subject: [PATCH 279/411] ALSA: hda - Use regmap for parameter caches, too The amp hash table was used for recording the cached reads of some capability values like pin caps or amp caps. Now all these are moved to regmap as well. One addition to the regmap helper is codec->caps_overwriting flag. This is set in snd_hdac_override_parm(), and the regmap helper accepts any register while this flag is set, so that it can overwrite even the read-only verb like AC_VERB_PARAMETERS. The flag is cleared immediately in snd_hdac_override_parm(), as it's a once-off flag. Along with these changes, the no longer needed amp hash and relevant fields are removed from hda_codec struct now. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 3 + sound/hda/hdac_device.c | 23 ++++++ sound/hda/hdac_regmap.c | 8 +++ sound/pci/hda/hda_codec.c | 144 ++++---------------------------------- sound/pci/hda/hda_codec.h | 7 -- sound/pci/hda/hda_local.h | 38 +++++++++- 6 files changed, 82 insertions(+), 141 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 65ea6429f3a7fe..ce7d8d1f59c63c 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -76,6 +76,7 @@ struct hdac_device { /* regmap */ struct regmap *regmap; bool lazy_cache:1; /* don't wake up for writes */ + bool caps_overwriting:1; /* caps overwrite being in process */ }; /* device/driver type used for matching */ @@ -109,6 +110,8 @@ int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm, unsigned int *res); int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid, int parm); +int snd_hdac_override_parm(struct hdac_device *codec, hda_nid_t nid, + unsigned int parm, unsigned int val); int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 0dac746df7da9c..72c584eb011b97 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -272,6 +272,29 @@ int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hdac_read_parm_uncached); +/** + * snd_hdac_override_parm - override read-only parameters + * @codec: the codec object + * @nid: NID for the parameter + * @parm: the parameter to change + * @val: the parameter value to overwrite + */ +int snd_hdac_override_parm(struct hdac_device *codec, hda_nid_t nid, + unsigned int parm, unsigned int val) +{ + unsigned int verb = (AC_VERB_PARAMETERS << 8) | (nid << 20) | parm; + int err; + + if (!codec->regmap) + return -EINVAL; + + codec->caps_overwriting = true; + err = snd_hdac_regmap_write_raw(codec, verb, val); + codec->caps_overwriting = false; + return err; +} +EXPORT_SYMBOL_GPL(snd_hdac_override_parm); + /** * snd_hdac_get_sub_nodes - get start NID and number of subtree nodes * @codec: the codec object diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index db03d60d9c9917..933907b164574c 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -58,8 +58,12 @@ static bool hda_volatile_reg(struct device *dev, unsigned int reg) static bool hda_writeable_reg(struct device *dev, unsigned int reg) { + struct hdac_device *codec = dev_to_hdac_dev(dev); unsigned int verb = get_verb(reg); + if (codec->caps_overwriting) + return true; + switch (verb & 0xf00) { case AC_VERB_GET_STREAM_FORMAT: case AC_VERB_GET_AMP_GAIN_MUTE: @@ -97,8 +101,12 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg) static bool hda_readable_reg(struct device *dev, unsigned int reg) { + struct hdac_device *codec = dev_to_hdac_dev(dev); unsigned int verb = get_verb(reg); + if (codec->caps_overwriting) + return true; + switch (verb) { case AC_VERB_PARAMETERS: case AC_VERB_GET_CONNECT_LIST: diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 52962f697825b4..b27f250088c15d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -929,9 +929,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) codec->proc_widget_hook = NULL; codec->spec = NULL; - free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); - init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); /* free only driver_pins so that init_pins + user_pins are restored */ @@ -996,7 +994,6 @@ static void snd_hda_codec_dev_release(struct device *dev) free_init_pincfgs(codec); snd_hdac_device_exit(&codec->core); snd_hda_sysfs_clear(codec); - free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); kfree(codec->modelname); kfree(codec->wcaps); @@ -1051,7 +1048,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); mutex_init(&codec->hash_mutex); - init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); @@ -1380,67 +1376,6 @@ static struct hda_cache_head *get_alloc_hash(struct hda_cache_rec *cache, return info; } -/* query and allocate an amp hash entry */ -static inline struct hda_amp_info * -get_alloc_amp_hash(struct hda_codec *codec, u32 key) -{ - return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key); -} - -/* overwrite the value with the key in the caps hash */ -static int write_caps_hash(struct hda_codec *codec, u32 key, unsigned int val) -{ - struct hda_amp_info *info; - - mutex_lock(&codec->hash_mutex); - info = get_alloc_amp_hash(codec, key); - if (!info) { - mutex_unlock(&codec->hash_mutex); - return -EINVAL; - } - info->amp_caps = val; - info->head.val |= INFO_AMP_CAPS; - mutex_unlock(&codec->hash_mutex); - return 0; -} - -/* query the value from the caps hash; if not found, fetch the current - * value from the given function and store in the hash - */ -static unsigned int -query_caps_hash(struct hda_codec *codec, hda_nid_t nid, int dir, u32 key, - unsigned int (*func)(struct hda_codec *, hda_nid_t, int)) -{ - struct hda_amp_info *info; - unsigned int val; - - mutex_lock(&codec->hash_mutex); - info = get_alloc_amp_hash(codec, key); - if (!info) { - mutex_unlock(&codec->hash_mutex); - return 0; - } - if (!(info->head.val & INFO_AMP_CAPS)) { - mutex_unlock(&codec->hash_mutex); /* for reentrance */ - val = func(codec, nid, dir); - write_caps_hash(codec, key, val); - } else { - val = info->amp_caps; - mutex_unlock(&codec->hash_mutex); - } - return val; -} - -static unsigned int read_amp_cap(struct hda_codec *codec, hda_nid_t nid, - int direction) -{ - if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) - nid = codec->core.afg; - return snd_hda_param_read(codec, nid, - direction == HDA_OUTPUT ? - AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); -} - /** * query_amp_caps - query AMP capabilities * @codec: the HD-auio codec @@ -1455,9 +1390,11 @@ static unsigned int read_amp_cap(struct hda_codec *codec, hda_nid_t nid, */ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) { - return query_caps_hash(codec, nid, direction, - HDA_HASH_KEY(nid, direction, 0), - read_amp_cap); + if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD)) + nid = codec->core.afg; + return snd_hda_param_read(codec, nid, + direction == HDA_OUTPUT ? + AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP); } EXPORT_SYMBOL_GPL(query_amp_caps); @@ -1498,50 +1435,14 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_caps); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps) { - return write_caps_hash(codec, HDA_HASH_KEY(nid, dir, 0), caps); -} -EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps); - -static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid, - int dir) -{ - return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); -} - -/** - * snd_hda_query_pin_caps - Query PIN capabilities - * @codec: the HD-auio codec - * @nid: the NID to query - * - * Query PIN capabilities for the given widget. - * Returns the obtained capability bits. - * - * When cap bits have been already read, this doesn't read again but - * returns the cached value. - */ -u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) -{ - return query_caps_hash(codec, nid, 0, HDA_HASH_PINCAP_KEY(nid), - read_pin_cap); -} -EXPORT_SYMBOL_GPL(snd_hda_query_pin_caps); + unsigned int parm; -/** - * snd_hda_override_pin_caps - Override the pin capabilities - * @codec: the CODEC - * @nid: the NID to override - * @caps: the capability bits to set - * - * Override the cached PIN capabilitiy bits value by the given one. - * - * Returns zero if successful or a negative error code. - */ -int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, - unsigned int caps) -{ - return write_caps_hash(codec, HDA_HASH_PINCAP_KEY(nid), caps); + snd_hda_override_wcaps(codec, nid, + get_wcaps(codec, nid) | AC_WCAP_AMP_OVRD); + parm = dir == HDA_OUTPUT ? AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP; + return snd_hdac_override_parm(&codec->core, nid, parm, caps); } -EXPORT_SYMBOL_GPL(snd_hda_override_pin_caps); +EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps); /** * snd_hda_codec_amp_stereo - update the AMP stereo values @@ -3462,11 +3363,6 @@ static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) cmd = snd_array_elem(&codec->cmd_cache.buf, i); cmd->dirty = 1; } - for (i = 0; i < codec->amp_cache.buf.used; i++) { - struct hda_amp_info *amp; - amp = snd_array_elem(&codec->amp_cache.buf, i); - amp->head.dirty = 1; - } } /* @@ -3714,8 +3610,7 @@ unsigned int snd_hda_calc_stream_format(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_calc_stream_format); -static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid, - int dir) +static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid) { unsigned int val = 0; if (nid != codec->core.afg && @@ -3728,14 +3623,7 @@ static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid, return val; } -static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid) -{ - return query_caps_hash(codec, nid, 0, HDA_HASH_PARPCM_KEY(nid), - get_pcm_param); -} - -static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid, - int dir) +static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid) { unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (!streams || streams == -1) @@ -3745,12 +3633,6 @@ static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid, return streams; } -static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid) -{ - return query_caps_hash(codec, nid, 0, HDA_HASH_PARSTR_KEY(nid), - get_stream_param); -} - /** * snd_hda_query_supported_pcm - query the supported PCM rates and formats * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 135b70f066f1ad..6af801a5bf89f2 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -163,12 +163,6 @@ struct hda_cache_head { u16 next; }; -struct hda_amp_info { - struct hda_cache_head head; - u32 amp_caps; /* amp capabilities */ - u16 vol[2]; /* current volume & mute */ -}; - struct hda_cache_rec { u16 hash[64]; /* hash table for index */ struct snd_array buf; /* record entries */ @@ -257,7 +251,6 @@ struct hda_codec { struct snd_array mixers; /* list of assigned mixer elements */ struct snd_array nids; /* list of mapped mixer elements */ - struct hda_cache_rec amp_cache; /* cache for amp access */ struct hda_cache_rec cmd_cache; /* cache for other commands */ struct list_head conn_list; /* linked-list of connection-list */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 7023eeee8b9d2d..3b567f42296b9d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -557,9 +557,41 @@ static inline void snd_hda_override_wcaps(struct hda_codec *codec, u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); -u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, - unsigned int caps); +/** + * snd_hda_query_pin_caps - Query PIN capabilities + * @codec: the HD-auio codec + * @nid: the NID to query + * + * Query PIN capabilities for the given widget. + * Returns the obtained capability bits. + * + * When cap bits have been already read, this doesn't read again but + * returns the cached value. + */ +static inline u32 +snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + +} + +/** + * snd_hda_override_pin_caps - Override the pin capabilities + * @codec: the CODEC + * @nid: the NID to override + * @caps: the capability bits to set + * + * Override the cached PIN capabilitiy bits value by the given one. + * + * Returns zero if successful or a negative error code. + */ +static inline int +snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid, + unsigned int caps) +{ + return snd_hdac_override_parm(&codec->core, nid, AC_PAR_PIN_CAP, caps); +} + bool snd_hda_check_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int bits); From 5e56bcea5017b7b7808df60f21ef01738b6e1a25 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2015 12:29:03 +0100 Subject: [PATCH 280/411] ALSA: hda - Allow driver to add vendor-specific verbs for regmap Codecs may have own vendor-specific verbs, and we need to allow each driver to give such verbs for cached accesses. Here a verb can be put into a single array and looked through it at readable and writeable callbacks. Signed-off-by: Takashi Iwai --- include/sound/hda_regmap.h | 3 ++- include/sound/hdaudio.h | 1 + sound/hda/hdac_regmap.c | 28 ++++++++++++++++++++++++++++ 3 files changed, 31 insertions(+), 1 deletion(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index 95651d26437d47..a6a4f3ddb46922 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -11,7 +11,8 @@ int snd_hdac_regmap_init(struct hdac_device *codec); void snd_hdac_regmap_exit(struct hdac_device *codec); - +int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec, + unsigned int verb); int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, unsigned int *val); int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index ce7d8d1f59c63c..702032598bea01 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -75,6 +75,7 @@ struct hdac_device { /* regmap */ struct regmap *regmap; + struct snd_array vendor_verbs; bool lazy_cache:1; /* don't wake up for writes */ bool caps_overwriting:1; /* caps overwrite being in process */ }; diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 933907b164574c..486ef720cbff89 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -60,6 +60,13 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg) { struct hdac_device *codec = dev_to_hdac_dev(dev); unsigned int verb = get_verb(reg); + int i; + + for (i = 0; i < codec->vendor_verbs.used; i++) { + unsigned int *v = snd_array_elem(&codec->vendor_verbs, i); + if (verb == *v) + return true; + } if (codec->caps_overwriting) return true; @@ -200,6 +207,7 @@ int snd_hdac_regmap_init(struct hdac_device *codec) if (IS_ERR(regmap)) return PTR_ERR(regmap); codec->regmap = regmap; + snd_array_init(&codec->vendor_verbs, sizeof(unsigned int), 8); return 0; } EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); @@ -209,10 +217,30 @@ void snd_hdac_regmap_exit(struct hdac_device *codec) if (codec->regmap) { regmap_exit(codec->regmap); codec->regmap = NULL; + snd_array_free(&codec->vendor_verbs); } } EXPORT_SYMBOL_GPL(snd_hdac_regmap_exit); +/** + * snd_hdac_regmap_add_vendor_verb - add a vendor-specific verb to regmap + * @codec: the codec object + * @verb: verb to allow accessing via regmap + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec, + unsigned int verb) +{ + unsigned int *p = snd_array_new(&codec->vendor_verbs); + + if (!p) + return -ENOMEM; + *p = verb; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_regmap_add_vendor_verb); + /* * helper functions */ From a551d91473e5e3a591f6fe86ac5a5fb460c3f96a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2015 12:34:49 +0100 Subject: [PATCH 281/411] ALSA: hda - Use regmap for command verb caches, too Like the previous patches, this patch converts also to the regmap, at this time, the cached verb writes are the target. But this conversion needs a bit more caution than before. - In the old code, we just record any verbs as is, and restore them at resume. For the regmap scheme, this doesn't work, since a few verbs like AMP or DIGI_CONVERT are asymmetrical. Such verbs are converted either to the dedicated function (snd_hda_regmap_xxx_amp()) or changed to the unified verb. - Some verbs have to be declared as vendor-specific ones before accessing via regmap. Also, the minor optimization with codec->cached_write flag is dropped in a few places, as this would confuse the operation. Further optimizations will be brought in the later patches, if any. This conversion ends up with a drop of significant amount of codes, mostly the helper codes that are no longer used. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 271 ++++----------------------------- sound/pci/hda/hda_codec.h | 34 ++--- sound/pci/hda/hda_generic.c | 14 +- sound/pci/hda/patch_analog.c | 16 +- sound/pci/hda/patch_conexant.c | 5 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 - sound/pci/hda/patch_si3054.c | 4 + sound/pci/hda/patch_sigmatel.c | 25 +-- 9 files changed, 64 insertions(+), 309 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b27f250088c15d..41851f9b48c113 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -807,10 +807,6 @@ static void hda_jackpoll_work(struct work_struct *work) codec->jackpoll_interval); } -static void init_hda_cache(struct hda_cache_rec *cache, - unsigned int record_size); -static void free_hda_cache(struct hda_cache_rec *cache); - /* release all pincfg lists */ static void free_init_pincfgs(struct hda_codec *codec) { @@ -929,9 +925,6 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) codec->proc_widget_hook = NULL; codec->spec = NULL; - free_hda_cache(&codec->cmd_cache); - init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - /* free only driver_pins so that init_pins + user_pins are restored */ snd_array_free(&codec->driver_pins); snd_array_free(&codec->cvt_setups); @@ -994,7 +987,6 @@ static void snd_hda_codec_dev_release(struct device *dev) free_init_pincfgs(codec); snd_hdac_device_exit(&codec->core); snd_hda_sysfs_clear(codec); - free_hda_cache(&codec->cmd_cache); kfree(codec->modelname); kfree(codec->wcaps); kfree(codec); @@ -1047,8 +1039,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); - mutex_init(&codec->hash_mutex); - init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); @@ -1316,66 +1306,6 @@ static void hda_cleanup_all_streams(struct hda_codec *codec) * amp access functions */ -/* FIXME: more better hash key? */ -#define HDA_HASH_KEY(nid, dir, idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) -#define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) -#define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) -#define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) -#define INFO_AMP_CAPS (1<<0) -#define INFO_AMP_VOL(ch) (1 << (1 + (ch))) - -/* initialize the hash table */ -static void init_hda_cache(struct hda_cache_rec *cache, - unsigned int record_size) -{ - memset(cache, 0, sizeof(*cache)); - memset(cache->hash, 0xff, sizeof(cache->hash)); - snd_array_init(&cache->buf, record_size, 64); -} - -static void free_hda_cache(struct hda_cache_rec *cache) -{ - snd_array_free(&cache->buf); -} - -/* query the hash. allocate an entry if not found. */ -static struct hda_cache_head *get_hash(struct hda_cache_rec *cache, u32 key) -{ - u16 idx = key % (u16)ARRAY_SIZE(cache->hash); - u16 cur = cache->hash[idx]; - struct hda_cache_head *info; - - while (cur != 0xffff) { - info = snd_array_elem(&cache->buf, cur); - if (info->key == key) - return info; - cur = info->next; - } - return NULL; -} - -/* query the hash. allocate an entry if not found. */ -static struct hda_cache_head *get_alloc_hash(struct hda_cache_rec *cache, - u32 key) -{ - struct hda_cache_head *info = get_hash(cache, key); - if (!info) { - u16 idx, cur; - /* add a new hash entry */ - info = snd_array_new(&cache->buf); - if (!info) - return NULL; - cur = snd_array_index(&cache->buf, info); - info->key = key; - info->val = 0; - info->dirty = 0; - idx = key % (u16)ARRAY_SIZE(cache->hash); - info->next = cache->hash[idx]; - cache->hash[idx] = cur; - } - return info; -} - /** * query_amp_caps - query AMP capabilities * @codec: the HD-auio codec @@ -2589,25 +2519,35 @@ static unsigned int convert_to_spdif_status(unsigned short val) /* set digital convert verbs both for the given NID and its slaves */ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid, - int verb, int val) + int mask, int val) { const hda_nid_t *d; - snd_hda_codec_write_cache(codec, nid, 0, verb, val); + snd_hdac_regmap_update(&codec->core, nid, AC_VERB_SET_DIGI_CONVERT_1, + mask, val); d = codec->slave_dig_outs; if (!d) return; for (; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, verb, val); + snd_hdac_regmap_update(&codec->core, nid, + AC_VERB_SET_DIGI_CONVERT_1, mask, val); } static inline void set_dig_out_convert(struct hda_codec *codec, hda_nid_t nid, int dig1, int dig2) { - if (dig1 != -1) - set_dig_out(codec, nid, AC_VERB_SET_DIGI_CONVERT_1, dig1); - if (dig2 != -1) - set_dig_out(codec, nid, AC_VERB_SET_DIGI_CONVERT_2, dig2); + unsigned int mask = 0; + unsigned int val = 0; + + if (dig1 != -1) { + mask |= 0xff; + val = dig1; + } + if (dig2 != -1) { + mask |= 0xff00; + val |= dig2 << 8; + } + set_dig_out(codec, nid, mask, val); } static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, @@ -2740,6 +2680,7 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec, struct snd_kcontrol *kctl; struct snd_kcontrol_new *dig_mix; int idx = 0; + int val = 0; const int spdif_index = 16; struct hda_spdif_out *spdif; struct hda_bus *bus = codec->bus; @@ -2780,8 +2721,9 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec, return err; } spdif->nid = cvt_nid; - spdif->ctls = snd_hda_codec_read(codec, cvt_nid, 0, - AC_VERB_GET_DIGI_CONVERT_1, 0); + snd_hdac_regmap_read(&codec->core, cvt_nid, + AC_VERB_GET_DIGI_CONVERT_1, &val); + spdif->ctls = val; spdif->status = convert_to_spdif_status(spdif->ctls); return 0; } @@ -2925,8 +2867,8 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, change = codec->spdif_in_enable != val; if (change) { codec->spdif_in_enable = val; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_DIGI_CONVERT_1, val); + snd_hdac_regmap_write(&codec->core, nid, + AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); return change; @@ -2937,10 +2879,11 @@ static int snd_hda_spdif_in_status_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = kcontrol->private_value; - unsigned short val; + unsigned int val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + snd_hdac_regmap_read(&codec->core, nid, + AC_VERB_GET_DIGI_CONVERT_1, &val); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -3006,154 +2949,6 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) } EXPORT_SYMBOL_GPL(snd_hda_create_spdif_in_ctls); -/* - * command cache - */ - -/* build a 31bit cache key with the widget id and the command parameter */ -#define build_cmd_cache_key(nid, verb) ((verb << 8) | nid) -#define get_cmd_cache_nid(key) ((key) & 0xff) -#define get_cmd_cache_cmd(key) (((key) >> 8) & 0xffff) - -/** - * snd_hda_codec_write_cache - send a single command with caching - * @codec: the HDA codec - * @nid: NID to send the command - * @flags: optional bit flags - * @verb: the verb to send - * @parm: the parameter for the verb - * - * Send a single command without waiting for response. - * - * Returns 0 if successful, or a negative error code. - */ -int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int flags, unsigned int verb, unsigned int parm) -{ - int err; - struct hda_cache_head *c; - u32 key; - unsigned int cache_only; - - cache_only = codec->cached_write; - if (!cache_only) { - err = snd_hda_codec_write(codec, nid, flags, verb, parm); - if (err < 0) - return err; - } - - /* parm may contain the verb stuff for get/set amp */ - verb = verb | (parm >> 8); - parm &= 0xff; - key = build_cmd_cache_key(nid, verb); - mutex_lock(&codec->bus->core.cmd_mutex); - c = get_alloc_hash(&codec->cmd_cache, key); - if (c) { - c->val = parm; - c->dirty = cache_only; - } - mutex_unlock(&codec->bus->core.cmd_mutex); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_codec_write_cache); - -/** - * snd_hda_codec_update_cache - check cache and write the cmd only when needed - * @codec: the HDA codec - * @nid: NID to send the command - * @flags: optional bit flags - * @verb: the verb to send - * @parm: the parameter for the verb - * - * This function works like snd_hda_codec_write_cache(), but it doesn't send - * command if the parameter is already identical with the cached value. - * If not, it sends the command and refreshes the cache. - * - * Returns 0 if successful, or a negative error code. - */ -int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int flags, unsigned int verb, unsigned int parm) -{ - struct hda_cache_head *c; - u32 key; - - /* parm may contain the verb stuff for get/set amp */ - verb = verb | (parm >> 8); - parm &= 0xff; - key = build_cmd_cache_key(nid, verb); - mutex_lock(&codec->bus->core.cmd_mutex); - c = get_hash(&codec->cmd_cache, key); - if (c && c->val == parm) { - mutex_unlock(&codec->bus->core.cmd_mutex); - return 0; - } - mutex_unlock(&codec->bus->core.cmd_mutex); - return snd_hda_codec_write_cache(codec, nid, flags, verb, parm); -} -EXPORT_SYMBOL_GPL(snd_hda_codec_update_cache); - -/** - * snd_hda_codec_resume_cache - Resume the all commands from the cache - * @codec: HD-audio codec - * - * Execute all verbs recorded in the command caches to resume. - */ -void snd_hda_codec_resume_cache(struct hda_codec *codec) -{ - int i; - - mutex_lock(&codec->hash_mutex); - codec->cached_write = 0; - for (i = 0; i < codec->cmd_cache.buf.used; i++) { - struct hda_cache_head *buffer; - u32 key; - - buffer = snd_array_elem(&codec->cmd_cache.buf, i); - key = buffer->key; - if (!key) - continue; - if (!buffer->dirty) - continue; - buffer->dirty = 0; - mutex_unlock(&codec->hash_mutex); - snd_hda_codec_write(codec, get_cmd_cache_nid(key), 0, - get_cmd_cache_cmd(key), buffer->val); - mutex_lock(&codec->hash_mutex); - } - mutex_unlock(&codec->hash_mutex); -} -EXPORT_SYMBOL_GPL(snd_hda_codec_resume_cache); - -/** - * snd_hda_sequence_write_cache - sequence writes with caching - * @codec: the HDA codec - * @seq: VERB array to send - * - * Send the commands sequentially from the given array. - * Thte commands are recorded on cache for power-save and resume. - * The array must be terminated with NID=0. - */ -void snd_hda_sequence_write_cache(struct hda_codec *codec, - const struct hda_verb *seq) -{ - for (; seq->nid; seq++) - snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb, - seq->param); -} -EXPORT_SYMBOL_GPL(snd_hda_sequence_write_cache); - -/** - * snd_hda_codec_flush_cache - Execute all pending (cached) amps / verbs - * @codec: HD-audio codec - */ -void snd_hda_codec_flush_cache(struct hda_codec *codec) -{ - if (codec->core.regmap) - regcache_sync(codec->core.regmap); - snd_hda_codec_resume_cache(codec); -} -EXPORT_SYMBOL_GPL(snd_hda_codec_flush_cache); - /** * snd_hda_codec_set_power_to_all - Set the power state to all widgets * @codec: the HDA codec @@ -3354,17 +3149,6 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) return state; } -/* mark all entries of cmd and amp caches dirty */ -static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) -{ - int i; - for (i = 0; i < codec->cmd_cache.buf.used; i++) { - struct hda_cache_head *cmd; - cmd = snd_array_elem(&codec->cmd_cache.buf, i); - cmd->dirty = 1; - } -} - /* * kick up codec; used both from PM and power-save */ @@ -3375,8 +3159,6 @@ static void hda_call_codec_resume(struct hda_codec *codec) if (codec->core.regmap) regcache_mark_dirty(codec->core.regmap); - hda_mark_cmd_cache_dirty(codec); - codec->power_jiffies = jiffies; hda_set_power_state(codec, AC_PWRST_D0); @@ -3390,7 +3172,6 @@ static void hda_call_codec_resume(struct hda_codec *codec) codec->patch_ops.init(codec); if (codec->core.regmap) regcache_sync(codec->core.regmap); - snd_hda_codec_resume_cache(codec); } if (codec->jackpoll_interval) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6af801a5bf89f2..26cbb1fa97294d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -155,19 +155,6 @@ struct hda_codec_ops { void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on); }; -/* record for amp information cache */ -struct hda_cache_head { - u32 key:31; /* hash key */ - u32 dirty:1; - u16 val; /* assigned value */ - u16 next; -}; - -struct hda_cache_rec { - u16 hash[64]; /* hash table for index */ - struct snd_array buf; /* record entries */ -}; - /* PCM callbacks */ struct hda_pcm_ops { int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec, @@ -251,13 +238,10 @@ struct hda_codec { struct snd_array mixers; /* list of assigned mixer elements */ struct snd_array nids; /* list of mapped mixer elements */ - struct hda_cache_rec cmd_cache; /* cache for other commands */ - struct list_head conn_list; /* linked-list of connection-list */ struct mutex spdif_mutex; struct mutex control_mutex; - struct mutex hash_mutex; struct snd_array spdif_out; unsigned int spdif_in_enable; /* SPDIF input enable? */ const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ @@ -406,15 +390,15 @@ snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) } /* cached write */ -int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int flags, unsigned int verb, unsigned int parm); -void snd_hda_sequence_write_cache(struct hda_codec *codec, - const struct hda_verb *seq); -int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int flags, unsigned int verb, unsigned int parm); -void snd_hda_codec_resume_cache(struct hda_codec *codec); -/* both for cmd & amp caches */ -void snd_hda_codec_flush_cache(struct hda_codec *codec); +static inline int +snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + return snd_hdac_regmap_write(&codec->core, nid, verb, parm); +} + +#define snd_hda_codec_update_cache(codec, nid, flags, verb, parm) \ + snd_hda_codec_write_cache(codec, nid, flags, verb, parm) /* the struct for codec->pin_configs */ struct hda_pincfg { diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 4850f92c89c4c7..f7ccef5559deac 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3381,11 +3381,6 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, imux = &spec->input_mux; adc_idx = kcontrol->id.index; mutex_lock(&codec->control_mutex); - /* we use the cache-only update at first since multiple input paths - * may shared the same amp; by updating only caches, the redundant - * writes to hardware can be reduced. - */ - codec->cached_write = 1; for (i = 0; i < imux->num_items; i++) { path = get_input_path(codec, adc_idx, i); if (!path || !path->ctls[type]) @@ -3393,12 +3388,9 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, kcontrol->private_value = path->ctls[type]; err = func(kcontrol, ucontrol); if (err < 0) - goto error; + break; } - error: - codec->cached_write = 0; mutex_unlock(&codec->control_mutex); - snd_hda_codec_flush_cache(codec); /* flush the updates */ if (err >= 0 && spec->cap_sync_hook) spec->cap_sync_hook(codec, kcontrol, ucontrol); return err; @@ -5760,8 +5752,6 @@ int snd_hda_gen_init(struct hda_codec *codec) snd_hda_apply_verbs(codec); - codec->cached_write = 1; - init_multi_out(codec); init_extra_out(codec); init_multi_io(codec); @@ -5777,7 +5767,7 @@ int snd_hda_gen_init(struct hda_codec *codec) /* call init functions of standard auto-mute helpers */ update_automute_all(codec); - snd_hda_codec_flush_cache(codec); + regcache_sync(codec->core.regmap); if (spec->vmaster_mute.sw_kctl && spec->vmaster_mute.hook) snd_hda_sync_vmaster_hook(&spec->vmaster_mute); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2278e83234b578..231f89029779a0 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -777,7 +777,6 @@ static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol, return 0; mutex_lock(&codec->control_mutex); - codec->cached_write = 1; path = snd_hda_get_path_from_idx(codec, spec->smux_paths[spec->cur_smux]); if (path) @@ -786,9 +785,7 @@ static int ad1988_auto_smux_enum_put(struct snd_kcontrol *kcontrol, if (path) snd_hda_activate_path(codec, path, true, true); spec->cur_smux = val; - codec->cached_write = 0; mutex_unlock(&codec->control_mutex); - snd_hda_codec_flush_cache(codec); /* flush the updates */ return 1; } @@ -1004,18 +1001,17 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct ad198x_spec *spec = codec->spec; - static const struct hda_verb gpio_init_verbs[] = { - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, - {}, - }; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook; spec->gen.own_eapd_ctl = 1; - snd_hda_sequence_write_cache(codec, gpio_init_verbs); + snd_hda_codec_write_cache(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, 0x02); + snd_hda_codec_write_cache(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x02); + snd_hda_codec_write_cache(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x02); break; case HDA_FIXUP_ACT_PROBE: if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1e21f9fbd54b93..f8f0dfbef14945 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -302,6 +302,7 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC; + snd_hdac_regmap_add_vendor_verb(&codec->core, 0x410); break; case HDA_FIXUP_ACT_PROBE: spec->gen.cap_sync_hook = cxt_update_headset_mode_hook; @@ -409,15 +410,11 @@ static void olpc_xo_automic(struct hda_codec *codec, struct hda_jack_callback *jack) { struct conexant_spec *spec = codec->spec; - int saved_cached_write = codec->cached_write; - codec->cached_write = 1; /* in DC mode, we don't handle automic */ if (!spec->dc_enable) snd_hda_gen_mic_autoswitch(codec, jack); olpc_xo_update_mic_pins(codec); - snd_hda_codec_flush_cache(codec); - codec->cached_write = saved_cached_write; if (spec->dc_enable) olpc_xo_update_mic_boost(codec); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 04c5ab20eb7676..ca0c05e1c42ede 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2212,7 +2212,6 @@ static int generic_hdmi_resume(struct hda_codec *codec) codec->patch_ops.init(codec); regcache_sync(codec->core.regmap); - snd_hda_codec_resume_cache(codec); for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); @@ -2299,6 +2298,7 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) /* enable DP1.2 mode */ vendor_param |= INTEL_EN_DP12; + snd_hdac_regmap_add_vendor_verb(&codec->core, INTEL_SET_VENDOR_VERB); snd_hda_codec_write_cache(codec, INTEL_VENDOR_NID, 0, INTEL_SET_VENDOR_VERB, vendor_param); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a440e539230fc0..d44cb7e37094f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -800,7 +800,6 @@ static int alc_resume(struct hda_codec *codec) msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); regcache_sync(codec->core.regmap); - snd_hda_codec_resume_cache(codec); hda_call_check_power_status(codec, 0x01); return 0; } @@ -3059,7 +3058,6 @@ static int alc269_resume(struct hda_codec *codec) } regcache_sync(codec->core.regmap); - snd_hda_codec_resume_cache(codec); hda_call_check_power_status(codec, 0x01); /* on some machine, the BIOS will clear the codec gpio data when enter diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 49b4868797a59b..5104bebb228699 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -222,6 +222,10 @@ static int si3054_init(struct hda_codec *codec) unsigned wait_count; u16 val; + if (snd_hdac_regmap_add_vendor_verb(&codec->core, + SI3054_VERB_WRITE_NODE)) + return -ENOMEM; + snd_hda_codec_write(codec, AC_NODE_ROOT, 0, AC_VERB_SET_CODEC_RESET, 0); snd_hda_codec_write(codec, codec->core.mfg, 0, AC_VERB_SET_STREAM_FORMAT, 0); SET_REG(codec, SI3054_LINE_RATE, 9600); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b314551749f179..43c99ce4a520c3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1050,12 +1050,9 @@ static const struct hda_verb stac92hd71bxx_core_init[] = { {} }; -static const struct hda_verb stac92hd71bxx_unmute_core_init[] = { +static const hda_nid_t stac92hd71bxx_unmute_nids[] = { /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {} + 0x0f, 0x0a, 0x0d, 0 }; static const struct hda_verb stac925x_core_init[] = { @@ -4269,6 +4266,10 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (spec->aloopback_ctl && snd_hda_get_bool_hint(codec, "loopback") == 1) { + unsigned int wr_verb = + spec->aloopback_ctl->private_value >> 16; + if (snd_hdac_regmap_add_vendor_verb(&codec->core, wr_verb)) + return -ENOMEM; if (!snd_hda_gen_add_kctl(&spec->gen, NULL, spec->aloopback_ctl)) return -ENOMEM; } @@ -4688,7 +4689,7 @@ static int patch_stac92hd95(struct hda_codec *codec) static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; - const struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; + const hda_nid_t *unmute_nids = stac92hd71bxx_unmute_nids; int err; err = alloc_stac_spec(codec); @@ -4713,7 +4714,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: - unmute_init++; + unmute_nids++; break; case 0x111d7608: /* 5 Port with Analog Mixer */ if ((codec->core.revision_id & 0xf) == 0 || @@ -4721,7 +4722,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - unmute_init++; + unmute_nids++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); break; @@ -4735,8 +4736,12 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) snd_hda_add_verbs(codec, stac92hd71bxx_core_init); - if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) - snd_hda_sequence_write_cache(codec, unmute_init); + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) { + const hda_nid_t *p; + for (p = unmute_nids; *p; p++) + snd_hda_codec_amp_init_stereo(codec, *p, HDA_INPUT, 0, + 0xff, 0x00); + } spec->aloopback_ctl = &stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; From d313e0a88d1b29d17198ef659af042a633a2d3de Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Mar 2015 20:43:20 +0100 Subject: [PATCH 282/411] ALSA: hda - Add a fake stereo amp register support HD-audio spec is inconvenient regarding the handling of stereo volume controls. It can set and get only single channel at once (although there is a special option to set the same value to both channels). This patch provides a fake pseudo-register via the regmap access so that the stereo channels can be read and written by a single call. It'd be useful, for example, for implementing DAPM widgets. A stereo amp pseudo register consists of the encoding like the normal amp verbs but it has both SET_LEFT (bit 13) and SET_RIGHT (bit 12) bits set. The regmap reads and writes a 16bit value for this pseudo register where the upper 8bit is for the right chanel and the lower 8bit for the left channel. Note that the driver doesn't recognize conflicts when both stereo and mono channel registers are mixed. Mixing them would certainly confuse the operation. So, use carefully. Signed-off-by: Takashi Iwai --- include/sound/hda_regmap.h | 59 +++++++++++++++++++++++++++++++ sound/hda/hdac_regmap.c | 71 +++++++++++++++++++++++++++++++++++++- 2 files changed, 129 insertions(+), 1 deletion(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index a6a4f3ddb46922..76648ccfbbf898 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -45,6 +45,20 @@ int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \ (idx)) +/** + * snd_hdac_regmap_encode_amp_stereo - encode a pseudo register for stereo AMPs + * @nid: widget NID + * @dir: direction (#HDA_INPUT, #HDA_OUTPUT) + * @idx: input index value + * + * Returns an encoded pseudo register. + */ +#define snd_hdac_regmap_encode_amp_stereo(nid, dir, idx) \ + (snd_hdac_regmap_encode_verb(nid, AC_VERB_GET_AMP_GAIN_MUTE) | \ + AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT | /* both bits set! */ \ + ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \ + (idx)) + /** * snd_hdac_regmap_write - Write a verb with caching * @nid: codec NID @@ -143,4 +157,49 @@ snd_hdac_regmap_update_amp(struct hdac_device *codec, hda_nid_t nid, return snd_hdac_regmap_update_raw(codec, cmd, mask, val); } +/** + * snd_hdac_regmap_get_amp_stereo - Read stereo AMP values + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel (left=0 or right=1) + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @index: the index value (only for input direction) + * @val: the pointer to store the value + * + * Read stereo AMP values. The lower byte is left, the upper byte is right. + * Returns the value or a negative error. + */ +static inline int +snd_hdac_regmap_get_amp_stereo(struct hdac_device *codec, hda_nid_t nid, + int dir, int idx) +{ + unsigned int cmd = snd_hdac_regmap_encode_amp_stereo(nid, dir, idx); + int err, val; + + err = snd_hdac_regmap_read_raw(codec, cmd, &val); + return err < 0 ? err : val; +} + +/** + * snd_hdac_regmap_update_amp_stereo - update the stereo AMP value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Update the stereo AMP value with a bit mask. + * The lower byte is left, the upper byte is right. + * Returns 0 if the value is unchanged, 1 if changed, or a negative error. + */ +static inline int +snd_hdac_regmap_update_amp_stereo(struct hdac_device *codec, hda_nid_t nid, + int dir, int idx, int mask, int val) +{ + unsigned int cmd = snd_hdac_regmap_encode_amp_stereo(nid, dir, idx); + + return snd_hdac_regmap_update_raw(codec, cmd, mask, val); +} + #endif /* __SOUND_HDA_REGMAP_H */ diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 486ef720cbff89..fb4a02e0319f7a 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -124,6 +124,70 @@ static bool hda_readable_reg(struct device *dev, unsigned int reg) return hda_writeable_reg(dev, reg); } +/* + * Stereo amp pseudo register: + * for making easier to handle the stereo volume control, we provide a + * fake register to deal both left and right channels by a single + * (pseudo) register access. A verb consisting of SET_AMP_GAIN with + * *both* SET_LEFT and SET_RIGHT bits takes a 16bit value, the lower 8bit + * for the left and the upper 8bit for the right channel. + */ +static bool is_stereo_amp_verb(unsigned int reg) +{ + if (((reg >> 8) & 0x700) != AC_VERB_SET_AMP_GAIN_MUTE) + return false; + return (reg & (AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT)) == + (AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT); +} + +/* read a pseudo stereo amp register (16bit left+right) */ +static int hda_reg_read_stereo_amp(struct hdac_device *codec, + unsigned int reg, unsigned int *val) +{ + unsigned int left, right; + int err; + + reg &= ~(AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT); + err = snd_hdac_exec_verb(codec, reg | AC_AMP_GET_LEFT, 0, &left); + if (err < 0) + return err; + err = snd_hdac_exec_verb(codec, reg | AC_AMP_GET_RIGHT, 0, &right); + if (err < 0) + return err; + *val = left | (right << 8); + return 0; +} + +/* write a pseudo stereo amp register (16bit left+right) */ +static int hda_reg_write_stereo_amp(struct hdac_device *codec, + unsigned int reg, unsigned int val) +{ + int err; + unsigned int verb, left, right; + + verb = AC_VERB_SET_AMP_GAIN_MUTE << 8; + if (reg & AC_AMP_GET_OUTPUT) + verb |= AC_AMP_SET_OUTPUT; + else + verb |= AC_AMP_SET_INPUT | ((reg & 0xf) << 8); + reg = (reg & ~0xfffff) | verb; + + left = val & 0xff; + right = (val >> 8) & 0xff; + if (left == right) { + reg |= AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT; + return snd_hdac_exec_verb(codec, reg | left, 0, NULL); + } + + err = snd_hdac_exec_verb(codec, reg | AC_AMP_SET_LEFT | left, 0, NULL); + if (err < 0) + return err; + err = snd_hdac_exec_verb(codec, reg | AC_AMP_SET_RIGHT | right, 0, NULL); + if (err < 0) + return err; + return 0; +} + static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) { struct hdac_device *codec = context; @@ -131,6 +195,8 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) if (!codec_is_running(codec)) return -EAGAIN; reg |= (codec->addr << 28); + if (is_stereo_amp_verb(reg)) + return hda_reg_read_stereo_amp(codec, reg, val); return snd_hdac_exec_verb(codec, reg, 0, val); } @@ -145,8 +211,11 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) reg &= ~0x00080000U; /* drop GET bit */ reg |= (codec->addr << 28); - verb = get_verb(reg); + if (is_stereo_amp_verb(reg)) + return hda_reg_write_stereo_amp(codec, reg, val); + + verb = get_verb(reg); switch (verb & 0xf00) { case AC_VERB_SET_AMP_GAIN_MUTE: verb = AC_VERB_SET_AMP_GAIN_MUTE; From 33f819400659da9ff9f636b78f33ff4f1f08cbd4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Mar 2015 22:19:47 +0100 Subject: [PATCH 283/411] ALSA: hda - Handle get/set power verb symmetrically via regmap HD-audio has quite a few asymmetrical ways of accessing verbs, and one of typical ones is GET/SET_POWER_STATE verbs. While it takes only the power state for setting, it returns a combination of states for getting. For making the state handling simpler, this patch adds a code to translate the value returned from GET_POWER_STATE to return only the actual state or -1 for error. In that way, the driver can simplify the power state management. Signed-off-by: Takashi Iwai --- sound/hda/hdac_regmap.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index fb4a02e0319f7a..2eea8d4e6a7fe0 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -191,13 +191,24 @@ static int hda_reg_write_stereo_amp(struct hdac_device *codec, static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) { struct hdac_device *codec = context; + int err; if (!codec_is_running(codec)) return -EAGAIN; reg |= (codec->addr << 28); if (is_stereo_amp_verb(reg)) return hda_reg_read_stereo_amp(codec, reg, val); - return snd_hdac_exec_verb(codec, reg, 0, val); + err = snd_hdac_exec_verb(codec, reg, 0, val); + if (err < 0) + return err; + /* special handling for asymmetric reads */ + if (get_verb(reg) == AC_VERB_GET_POWER_STATE) { + if (*val & AC_PWRST_ERROR) + *val = -1; + else /* take only the actual state */ + *val = (*val >> 4) & 0x0f; + } + return 0; } static int hda_reg_write(void *context, unsigned int reg, unsigned int val) From 40ba66a702b83f46c53456eaaac692fc12f82cb0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Mar 2015 15:56:25 +0100 Subject: [PATCH 284/411] ALSA: hda - Add cache support for COEF read/write The 16bit COEF read/write is pretty standard for many codecs, and they can be cached in most cases -- more importantly, they need to be restored at resume. For making this easier, add the cache support to regmap. If the codec driver wants to cache the COEF access, set codec->cache_coef flag and issue AC_VERB_GET_PROC_COEF with the coef index in LSB 8 bits. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + sound/hda/hdac_regmap.c | 49 ++++++++++++++++++++++++++++++++++++++++- 2 files changed, 49 insertions(+), 1 deletion(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 702032598bea01..95acc337aea5af 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -78,6 +78,7 @@ struct hdac_device { struct snd_array vendor_verbs; bool lazy_cache:1; /* don't wake up for writes */ bool caps_overwriting:1; /* caps overwrite being in process */ + bool cache_coef:1; /* cache COEF read/write too */ }; /* device/driver type used for matching */ diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 2eea8d4e6a7fe0..e1dcf104d273bd 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -33,10 +33,12 @@ static bool hda_volatile_reg(struct device *dev, unsigned int reg) { + struct hdac_device *codec = dev_to_hdac_dev(dev); unsigned int verb = get_verb(reg); switch (verb) { case AC_VERB_GET_PROC_COEF: + return !codec->cache_coef; case AC_VERB_GET_COEF_INDEX: case AC_VERB_GET_PROC_STATE: case AC_VERB_GET_POWER_STATE: @@ -75,6 +77,8 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg) case AC_VERB_GET_STREAM_FORMAT: case AC_VERB_GET_AMP_GAIN_MUTE: return true; + case AC_VERB_GET_PROC_COEF: + return codec->cache_coef; case 0xf00: break; default: @@ -188,9 +192,47 @@ static int hda_reg_write_stereo_amp(struct hdac_device *codec, return 0; } +/* read a pseudo coef register (16bit) */ +static int hda_reg_read_coef(struct hdac_device *codec, unsigned int reg, + unsigned int *val) +{ + unsigned int verb; + int err; + + if (!codec->cache_coef) + return -EINVAL; + /* LSB 8bit = coef index */ + verb = (reg & ~0xfff00) | (AC_VERB_SET_COEF_INDEX << 8); + err = snd_hdac_exec_verb(codec, verb, 0, NULL); + if (err < 0) + return err; + verb = (reg & ~0xfffff) | (AC_VERB_GET_COEF_INDEX << 8); + return snd_hdac_exec_verb(codec, verb, 0, val); +} + +/* write a pseudo coef register (16bit) */ +static int hda_reg_write_coef(struct hdac_device *codec, unsigned int reg, + unsigned int val) +{ + unsigned int verb; + int err; + + if (!codec->cache_coef) + return -EINVAL; + /* LSB 8bit = coef index */ + verb = (reg & ~0xfff00) | (AC_VERB_SET_COEF_INDEX << 8); + err = snd_hdac_exec_verb(codec, verb, 0, NULL); + if (err < 0) + return err; + verb = (reg & ~0xfffff) | (AC_VERB_GET_COEF_INDEX << 8) | + (val & 0xffff); + return snd_hdac_exec_verb(codec, verb, 0, NULL); +} + static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) { struct hdac_device *codec = context; + int verb = get_verb(reg); int err; if (!codec_is_running(codec)) @@ -198,11 +240,13 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) reg |= (codec->addr << 28); if (is_stereo_amp_verb(reg)) return hda_reg_read_stereo_amp(codec, reg, val); + if (verb == AC_VERB_GET_PROC_COEF) + return hda_reg_read_coef(codec, reg, val); err = snd_hdac_exec_verb(codec, reg, 0, val); if (err < 0) return err; /* special handling for asymmetric reads */ - if (get_verb(reg) == AC_VERB_GET_POWER_STATE) { + if (verb == AC_VERB_GET_POWER_STATE) { if (*val & AC_PWRST_ERROR) *val = -1; else /* take only the actual state */ @@ -227,6 +271,9 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) return hda_reg_write_stereo_amp(codec, reg, val); verb = get_verb(reg); + if (verb == AC_VERB_SET_PROC_COEF) + return hda_reg_write_coef(codec, reg, val); + switch (verb & 0xf00) { case AC_VERB_SET_AMP_GAIN_MUTE: verb = AC_VERB_SET_AMP_GAIN_MUTE; From 034f90b393a3d5c99654f048c4b22b3d05e3a345 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 23 Mar 2015 12:34:46 +0300 Subject: [PATCH 285/411] ALSA: ak411x: simplify snd_ak4113_create() a bit "err" is always a negative error code here, so there is no point in checking. Removing the check silences a static checker warning and makes the code a bit more clear. Also we don't need to initialize "err". Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4113.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index 88844881cbff99..2183e9ebaa6df1 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -73,7 +73,7 @@ int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, void *private_data, struct ak4113 **r_ak4113) { struct ak4113 *chip; - int err = 0; + int err; unsigned char reg; static struct snd_device_ops ops = { .dev_free = snd_ak4113_dev_free, @@ -109,7 +109,7 @@ int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, __fail: snd_ak4113_free(chip); - return err < 0 ? err : -EIO; + return err; } EXPORT_SYMBOL_GPL(snd_ak4113_create); From 77008b70fe0a9ffe354580d9dfda329cdde7f20b Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 23 Mar 2015 12:41:31 +0300 Subject: [PATCH 286/411] ALSA: echoaudio: read past end of array We need to cap "ucontrol->id.index / num_busses_in(chip)" so the we don't read beyond the end of the array. I also adding a check on "in" and changing the type in snd_echo_mixer_put() from short to unsigned int. Those changes are done for symmetry and are cosmetic. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index a962de03ebb68c..862db9a0b04112 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1283,12 +1283,14 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol, static int snd_echo_mixer_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct echoaudio *chip; + struct echoaudio *chip = snd_kcontrol_chip(kcontrol); + unsigned int out = ucontrol->id.index / num_busses_in(chip); + unsigned int in = ucontrol->id.index % num_busses_in(chip); - chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = - chip->monitor_gain[ucontrol->id.index / num_busses_in(chip)] - [ucontrol->id.index % num_busses_in(chip)]; + if (out >= ECHO_MAXAUDIOOUTPUTS || in >= ECHO_MAXAUDIOINPUTS) + return -EINVAL; + + ucontrol->value.integer.value[0] = chip->monitor_gain[out][in]; return 0; } @@ -1297,12 +1299,14 @@ static int snd_echo_mixer_put(struct snd_kcontrol *kcontrol, { struct echoaudio *chip; int changed, gain; - short out, in; + unsigned int out, in; changed = 0; chip = snd_kcontrol_chip(kcontrol); out = ucontrol->id.index / num_busses_in(chip); in = ucontrol->id.index % num_busses_in(chip); + if (out >= ECHO_MAXAUDIOOUTPUTS || in >= ECHO_MAXAUDIOINPUTS) + return -EINVAL; gain = ucontrol->value.integer.value[0]; if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT) return -EINVAL; From cd02e3df529eb3fbf8904ba60a617f6217ac7629 Mon Sep 17 00:00:00 2001 From: Howard Mitchell Date: Mon, 23 Mar 2015 21:17:01 +0000 Subject: [PATCH 287/411] ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4 Currently GPIO4 is hardcoded to output the pll-lock signal. Unfortunately this is after the pll-out GPIO is configured which is selectable in the device tree. Therefore it is not possible to use GPIO4 for pll-out. Therefore this patch removes the configuration of GPIO4. Signed-off-by: Howard Mitchell Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 0676ab8be03fa3..8c09e3ffdcaa5c 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1156,25 +1156,6 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, ret, pcm512x->pll_out); return ret; } - - gpio = PCM512x_G1OE << (4 - 1); - ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN, - gpio, gpio); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable gpio %d: %d\n", - 4, ret); - return ret; - } - - gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1; - ret = regmap_update_bits(pcm512x->regmap, gpio, - PCM512x_GxSL, PCM512x_GxSL_PLLLK); - if (ret != 0) { - dev_err(codec->dev, - "Failed to output pll lock on %d: %d\n", - ret, 4); - return ret; - } } ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE, From 9a303dc7ba93769e96471158892b264042ddc3fc Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Mon, 23 Mar 2015 12:44:49 -0700 Subject: [PATCH 288/411] sound: Deparenthesize negative error returns Make the returns a bit more kernel standard style. Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/isa/wavefront/wavefront_fx.c | 6 +++--- sound/isa/wavefront/wavefront_synth.c | 26 +++++++++++++------------- sound/oss/dev_table.c | 6 +++--- sound/oss/v_midi.c | 4 ++-- 4 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index b5a19708473aaf..0608a5a4289df8 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -79,13 +79,13 @@ wavefront_fx_memset (snd_wavefront_t *dev, if (page < 0 || page > 7) { snd_printk ("FX memset: " "page must be >= 0 and <= 7\n"); - return -(EINVAL); + return -EINVAL; } if (addr < 0 || addr > 0x7f) { snd_printk ("FX memset: " "addr must be >= 0 and <= 7f\n"); - return -(EINVAL); + return -EINVAL; } if (cnt == 1) { @@ -118,7 +118,7 @@ wavefront_fx_memset (snd_wavefront_t *dev, snd_printk ("FX memset " "(0x%x, 0x%x, 0x%lx, %d) incomplete\n", page, addr, (unsigned long) data, cnt); - return -(EIO); + return -EIO; } } diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 33f5ec14fcfa27..69f76ff5693d4e 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -793,7 +793,7 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PATCH, NULL, buf)) { snd_printk ("download patch failed\n"); - return -(EIO); + return -EIO; } return (0); @@ -831,7 +831,7 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PROGRAM, NULL, buf)) { snd_printk ("download patch failed\n"); - return -(EIO); + return -EIO; } return (0); @@ -952,7 +952,7 @@ wavefront_send_sample (snd_wavefront_t *dev, if (skip > 0 && header->hdr.s.SampleResolution != LINEAR_16BIT) { snd_printk ("channel selection only " "possible on 16-bit samples"); - return -(EINVAL); + return -EINVAL; } switch (skip) { @@ -1049,7 +1049,7 @@ wavefront_send_sample (snd_wavefront_t *dev, NULL, sample_hdr)) { snd_printk ("sample %sdownload refused.\n", header->size ? "" : "header "); - return -(EIO); + return -EIO; } if (header->size == 0) { @@ -1075,7 +1075,7 @@ wavefront_send_sample (snd_wavefront_t *dev, if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_BLOCK, NULL, NULL)) { snd_printk ("download block " "request refused.\n"); - return -(EIO); + return -EIO; } for (i = 0; i < blocksize; i++) { @@ -1135,12 +1135,12 @@ wavefront_send_sample (snd_wavefront_t *dev, if (dma_ack == -1) { snd_printk ("upload sample " "DMA ack timeout\n"); - return -(EIO); + return -EIO; } else { snd_printk ("upload sample " "DMA ack error 0x%x\n", dma_ack); - return -(EIO); + return -EIO; } } } @@ -1181,7 +1181,7 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_SAMPLE_ALIAS, NULL, alias_hdr)) { snd_printk ("download alias failed.\n"); - return -(EIO); + return -EIO; } dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_ALIAS); @@ -1232,7 +1232,7 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) msample_hdr)) { snd_printk ("download of multisample failed.\n"); kfree(msample_hdr); - return -(EIO); + return -EIO; } dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_MULTISAMPLE); @@ -1254,7 +1254,7 @@ wavefront_fetch_multisample (snd_wavefront_t *dev, if (snd_wavefront_cmd (dev, WFC_UPLOAD_MULTISAMPLE, log_ns, number)) { snd_printk ("upload multisample failed.\n"); - return -(EIO); + return -EIO; } DPRINT (WF_DEBUG_DATA, "msample %d has %d samples\n", @@ -1273,14 +1273,14 @@ wavefront_fetch_multisample (snd_wavefront_t *dev, if ((val = wavefront_read (dev)) == -1) { snd_printk ("upload multisample failed " "during sample loop.\n"); - return -(EIO); + return -EIO; } d[0] = val; if ((val = wavefront_read (dev)) == -1) { snd_printk ("upload multisample failed " "during sample loop.\n"); - return -(EIO); + return -EIO; } d[1] = val; @@ -1315,7 +1315,7 @@ wavefront_send_drum (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_EDRUM_PROGRAM, NULL, drumbuf)) { snd_printk ("download drum failed.\n"); - return -(EIO); + return -EIO; } return (0); diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c index d8cf3e58dc76aa..6dad51596b701d 100644 --- a/sound/oss/dev_table.c +++ b/sound/oss/dev_table.c @@ -58,13 +58,13 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, if (vers != AUDIO_DRIVER_VERSION || driver_size > sizeof(struct audio_driver)) { printk(KERN_ERR "Sound: Incompatible audio driver for %s\n", name); - return -(EINVAL); + return -EINVAL; } num = sound_alloc_audiodev(); if (num == -1) { printk(KERN_ERR "sound: Too many audio drivers\n"); - return -(EBUSY); + return -EBUSY; } d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver))); sound_nblocks++; @@ -79,7 +79,7 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, if (d == NULL || op == NULL) { printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name); sound_unload_audiodev(num); - return -(ENOMEM); + return -ENOMEM; } init_waitqueue_head(&op->in_sleeper); init_waitqueue_head(&op->out_sleeper); diff --git a/sound/oss/v_midi.c b/sound/oss/v_midi.c index f0b4151d9b17c4..fc0ba276cc8fd1 100644 --- a/sound/oss/v_midi.c +++ b/sound/oss/v_midi.c @@ -49,13 +49,13 @@ static int v_midi_open (int dev, int mode, unsigned long flags; if (devc == NULL) - return -(ENXIO); + return -ENXIO; spin_lock_irqsave(&devc->lock,flags); if (devc->opened) { spin_unlock_irqrestore(&devc->lock,flags); - return -(EBUSY); + return -EBUSY; } devc->opened = 1; spin_unlock_irqrestore(&devc->lock,flags); From 46d212cbe4111351741b85b29dbb106b50d68532 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 24 Mar 2015 11:48:24 +0100 Subject: [PATCH 289/411] ALSA: asihpi: Fix duplicate const for clock sources Replace duplicated const keyword for 'sampleclock_sources' with proper array of const pointers to const strings. Signed-off-by: Krzysztof Kozlowski Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e5cd7be8535519..1039eccbb895f2 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2376,7 +2376,7 @@ static int snd_asihpi_cmode_add(struct snd_card_asihpi *asihpi, /*------------------------------------------------------------ Sampleclock source controls ------------------------------------------------------------*/ -static const char const *sampleclock_sources[] = { +static const char * const sampleclock_sources[] = { "N/A", "Local PLL", "Digital Sync", "Word External", "Word Header", "SMPTE", "Digital1", "Auto", "Network", "Invalid", "Prev Module", "BLU-Link", From 143526ee94a295ed33b9cc19e9532ab6d14a1cc0 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 24 Mar 2015 09:51:12 +0800 Subject: [PATCH 290/411] ASoC: rt286: check regmap_read result for ID check It is worth to check the regmap_read result for ID check since it is the first regmap_read. And we can check if there is any i2c issue. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index ea967978ec6bc2..842cfb9fa19169 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1219,7 +1219,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret; + int i, ret, val; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1234,11 +1234,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { + ret = regmap_read(rt286->regmap, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret != 0) { + dev_err(&i2c->dev, "I2C error %d\n", ret); + return ret; + } + if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %x is not rt286\n", ret); + "Device with ID register %x is not rt286\n", val); return -ENODEV; } From 39c26180641da983269d5f24f124f8624f2587b4 Mon Sep 17 00:00:00 2001 From: Takeshi Kihara Date: Tue, 24 Mar 2015 05:15:22 +0000 Subject: [PATCH 291/411] ASoC: ak4642: enable stereo line output power-save mode ak4642 has power-save mode for stereo line to reduce pop noise. This patch enables it. Signed-off-by: Takeshi Kihara Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 33 +++++++++++++++++++++++++++++++-- 1 file changed, 31 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7255f69521538e..24ceb36bf489ae 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -97,6 +97,9 @@ #define PMMP (1 << 2) /* MPWR pin Power Management */ #define MGAIN0 (1 << 0) /* MIC amp gain*/ +/* SG_SL2 */ +#define LOPS (1 << 6) /* Stero Line-out Power Save Mode */ + /* TIMER */ #define ZTM(param) ((param & 0x3) << 4) /* ALC Zero Crossing TimeOut */ #define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) @@ -168,6 +171,29 @@ static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), }; +/* event handlers */ +static int ak4642_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_PRE_PMU: + /* Power save mode ON */ + snd_soc_update_bits(codec, SG_SL2, LOPS, LOPS); + break; + case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_POST_PMD: + /* Power save mode OFF */ + mdelay(300); + snd_soc_update_bits(codec, SG_SL2, LOPS, 0); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { /* Outputs */ @@ -182,9 +208,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, + SND_SOC_DAPM_MIXER_E("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], - ARRAY_SIZE(ak4642_lout_mixer_controls)), + ARRAY_SIZE(ak4642_lout_mixer_controls), + ak4642_lout_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), /* DAC */ SND_SOC_DAPM_DAC("DAC", NULL, PW_MGMT1, 2, 0), From 2f4b1e6bb25899e7d21e1764abcfb23f14250535 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 24 Mar 2015 11:47:43 +0100 Subject: [PATCH 292/411] ASoC: rsnd: Fix duplicate const for DVC ramp rates Replace duplicated const keyword for 'dvc_ramp_rate' with proper array of const pointers to const strings. Signed-off-by: Krzysztof Kozlowski Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index aeeef1352eeee8..6d85e36c6c2bc2 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -36,7 +36,7 @@ struct rsnd_dvc { ((pos) = (struct rsnd_dvc *)(priv)->dvc + i); \ i++) -static const char const *dvc_ramp_rate[] = { +static const char * const dvc_ramp_rate[] = { "128 dB/1 step", /* 00000 */ "64 dB/1 step", /* 00001 */ "32 dB/1 step", /* 00010 */ From a75a053f1eefbbbbae0f7d6bf1ed12ce012112b7 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 20 Mar 2015 13:31:08 +0200 Subject: [PATCH 293/411] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used Set rule constraints to allow only combinations of sample-rate, sample-format, and channels counts that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 207 ++++++++++++++++++++++++++++-- 1 file changed, 197 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d40b392b3da2d8..76156d18ed4640 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include @@ -65,6 +66,11 @@ struct davinci_mcasp_context { bool pm_state; }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; @@ -99,6 +105,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -868,6 +876,30 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, return 0; } +static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, + unsigned int bclk_freq, + int *error_ppm) +{ + int div = mcasp->sysclk_freq / bclk_freq; + int rem = mcasp->sysclk_freq % bclk_freq; + + if (rem != 0) { + if (div == 0 || + ((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + div++; + rem = rem - bclk_freq; + } + } + if (error_ppm) + *error_ppm = + (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) + /div - 1000000; + + return div; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -883,16 +915,20 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, * the machine driver, we need to calculate the ratio. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - unsigned int bclk_freq = snd_soc_params_to_bclk(params); - unsigned int div = mcasp->sysclk_freq / bclk_freq; - if (mcasp->sysclk_freq % bclk_freq != 0) { - if (((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) - div++; - dev_warn(mcasp->dev, - "Inaccurate BCLK: %u Hz / %u != %u Hz\n", - mcasp->sysclk_freq, div, bclk_freq); - } + int channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &ppm); + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -974,6 +1010,120 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static const unsigned int davinci_mcasp_dai_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, +}; + +#define DAVINCI_MAX_RATE_ERROR_PPM 1000 + +static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ri = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + if (ri->min <= davinci_mcasp_dai_rates[i] && + ri->max >= davinci_mcasp_dai_rates[i]) { + uint bclk_freq = sbits*channels* + davinci_mcasp_dai_rates[i]; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + } + } + dev_dbg(rd->mcasp->dev, + "%d frequencies (%d-%d) for %d sbits and %d channels\n", + count, ri->min, ri->max, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask nfmt; + int rate = params_rate(params); + int channels = params_channels(params); + int i, count = 0; + + snd_mask_none(&nfmt); + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + if (snd_mask_test(fmt, i)) { + uint bclk_freq = snd_pcm_format_width(i)*channels*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + snd_mask_set(&nfmt, i); + count++; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible sample format for %d Hz and %d channels\n", + count, rate, channels); + + return snd_mask_refine(fmt, &nfmt); +} + +static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts (%d-%d) for %d Hz and %d sbits\n", + count, ci->min, ci->max, rate, sbits); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -999,6 +1149,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1013,6 +1164,42 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); + + /* + * If we rely on implicit BCLK divider setting we should + * set constraints based on what we can provide. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + davinci_mcasp_hw_rule_format, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret) + return ret; + } + return 0; } From 4e2576bd36a12e78ac3786d05b99a820dffe687f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 24 Mar 2015 09:01:00 +0000 Subject: [PATCH 294/411] ASoC: soc-core: initialize debugfs in snd_soc_instantiate_card() Current soc_init_card_debugfs() is called from snd_soc_register_card() but, soc_cleanup_card_debugfs() is called from soc_cleanup_card_resources(), not from paired function. This differences don't matter for now. But if anyone wants to implement a proper hotplug/unplug, this difference would become clearer. Now, we can assume that snd_soc_instantiate_card() and soc_cleanup_card_resources() are paired function. soc_init_card_debugfs() / soc_cleanup_card_debugfs() paired function should be called from these. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index acf99f1250e5f2..4443581d69f285 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1662,6 +1662,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); + soc_init_card_debugfs(card); + return 0; probe_aux_dev_err: @@ -2352,8 +2354,6 @@ int snd_soc_register_card(struct snd_soc_card *card) snd_soc_initialize_card_lists(card); - soc_init_card_debugfs(card); - card->rtd = devm_kzalloc(card->dev, sizeof(struct snd_soc_pcm_runtime) * (card->num_links + card->num_aux_devs), @@ -2384,7 +2384,7 @@ int snd_soc_register_card(struct snd_soc_card *card) ret = snd_soc_instantiate_card(card); if (ret != 0) - soc_cleanup_card_debugfs(card); + return ret; /* deactivate pins to sleep state */ for (i = 0; i < card->num_rtd; i++) { From 44e39b985393e506d82b669c9765e850daed9dc5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Mar 2015 07:40:33 +0100 Subject: [PATCH 295/411] ALSA: hda - Remove superfluous hda_nid_t definition in hda_codec.h Just forgotten to remove. It's now in sound/hdaudio.h. Reported-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 76776164623de0..21632174744bed 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -69,9 +69,6 @@ struct hda_codec; struct hda_pcm; struct hda_pcm_stream; -/* NID type */ -typedef u16 hda_nid_t; - /* bus operators */ struct hda_bus_ops { /* send a single command */ From 4738465c37298cd9228750f7e16aad88af8495c5 Mon Sep 17 00:00:00 2001 From: "W. Trevor King" Date: Wed, 25 Mar 2015 00:43:42 -0700 Subject: [PATCH 296/411] ALSA: hda/via - Add beep controls to VIA codecs My codec has a beep-generating node: $ cat /proc/asound/card1/codec#0 Codec: VIA VT1802 ... Vendor Id: 0x11068446 Subsystem Id: 0x15587410 Revision Id: 0x100000 ... Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1 Amp-Out vals: [0x0a] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 ... But I was missing the: Control: name=... entries that I need to manage this widget from alsamixer. With this patch (based on the similar Mono Amp-Out handling in patch_conexant.c), I get a new: input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card1/hdaudioC1D0/input15 entry in dmesg and controls to manage that beep: $ cat /proc/asound/card1/codec#0 | grep -A5 Beep Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1 Amp-Out vals: [0x12] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 [rebased and modified for the latest tree by tiwai] Signed-off-by: W. Trevor King Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 61 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 61 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a34d7671937f67..742087ef378fd5 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -107,6 +107,8 @@ struct via_spec { /* work to check hp jack state */ int hp_work_active; int vt1708_jack_detect; + + unsigned int beep_amp; }; static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); @@ -266,6 +268,59 @@ static const struct snd_kcontrol_new via_pin_power_ctl_enum[] = { {} /* terminator */ }; +#ifdef CONFIG_SND_HDA_INPUT_BEEP +static inline void set_beep_amp(struct via_spec *spec, hda_nid_t nid, + int idx, int dir) +{ + spec->gen.beep_nid = nid; + spec->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); +} + +/* additional beep mixers; the actual parameters are overwritten at build */ +static const struct snd_kcontrol_new cxt_beep_mixer[] = { + HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT), + { } /* end */ +}; + +/* create beep controls if needed */ +static int add_beep_ctls(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + if (spec->beep_amp) { + const struct snd_kcontrol_new *knew; + for (knew = cxt_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + } + } + return 0; +} + +static void auto_parse_beep(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid; + + for_each_hda_codec_node(nid, codec) + if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) { + set_beep_amp(spec, nid, 0, HDA_OUTPUT); + break; + } +} +#else +#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#define add_beep_ctls(codec) 0 +#define auto_parse_beep(codec) +#endif /* check AA path's mute status */ static bool is_aa_path_mute(struct hda_codec *codec) @@ -352,6 +407,10 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; + err = add_beep_ctls(codec); + if (err < 0) + return err; + spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; for (i = 0; i < spec->num_mixers; i++) { @@ -512,6 +571,8 @@ static int via_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + auto_parse_beep(codec); + err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) return err; From 46172b6c26667133b9945b916b6223cc87bbf10c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 25 Mar 2015 11:22:35 +0000 Subject: [PATCH 297/411] ASoC: dapm: Fix build warning commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by switching to using devm_kasprintf. This also saves a couple of lines of code. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6828b4ed544799..e5ec9d7133ae1b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3356,7 +3356,6 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, { struct snd_soc_dapm_widget template; struct snd_soc_dapm_widget *w; - size_t len; char *link_name; int ret, count; unsigned long private_value; @@ -3376,28 +3375,26 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w_param_text) return -ENOMEM; - len = strlen(source->name) + strlen(sink->name) + 2; - link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); if (!link_name) { ret = -ENOMEM; goto outfree_w_param; } - snprintf(link_name, len, "%s-%s", source->name, sink->name); for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); - len = strlen("Anonymous Configuration ") + 3; w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); + devm_kasprintf(card->dev, GFP_KERNEL, + "Anonymous Configuration %d", + count); if (!w_param_text[count]) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, - "Anonymous Configuration %d", count); } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, From 8787041d9bb832b9449b1eb878cedcebce42c61a Mon Sep 17 00:00:00 2001 From: Sergej Sawazki Date: Tue, 24 Mar 2015 21:13:22 +0100 Subject: [PATCH 298/411] ASoC: wm8741: Fix rates constraints values The WM8741 DAC supports the following typical audio sampling rates: 44.1kHz, 88.2kHz, 176.4kHz (eg: with a master clock of 22.5792MHz) 32kHz, 48kHz, 96kHz, 192kHz (eg: with a master clock of 24.576MHz) For the rates lists, we should use 82000 instead of 88235, 176400 instead of 1764000 and 192000 instead of 19200 (seems to be a typo). Signed-off-by: Sergej Sawazki Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8741.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 31bb4801a00564..9e71c768966f05 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -123,7 +123,7 @@ static struct { }; static const unsigned int rates_11289[] = { - 44100, 88235, + 44100, 88200, }; static const struct snd_pcm_hw_constraint_list constraints_11289 = { @@ -150,7 +150,7 @@ static const struct snd_pcm_hw_constraint_list constraints_16384 = { }; static const unsigned int rates_16934[] = { - 44100, 88235, + 44100, 88200, }; static const struct snd_pcm_hw_constraint_list constraints_16934 = { @@ -168,7 +168,7 @@ static const struct snd_pcm_hw_constraint_list constraints_18432 = { }; static const unsigned int rates_22579[] = { - 44100, 88235, 1764000 + 44100, 88200, 176400 }; static const struct snd_pcm_hw_constraint_list constraints_22579 = { @@ -186,7 +186,7 @@ static const struct snd_pcm_hw_constraint_list constraints_24576 = { }; static const unsigned int rates_36864[] = { - 48000, 96000, 19200 + 48000, 96000, 192000 }; static const struct snd_pcm_hw_constraint_list constraints_36864 = { From 8bc174e9e3079b2475fb09e244f71fd57de7a802 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Mar 2015 14:18:34 +0100 Subject: [PATCH 299/411] ALSA: hda - Handle a few verbs as read-only Although they can be written, handle a few verbs as read-only in regmap interface: CONFIG_DEFAULT, CONV and CVT_CHAN_COUNT. These are either updated in PCM or HDMI management code in a volatile manner, or just needed only as parameter, thus they don't need to be written at resume sync. Signed-off-by: Takashi Iwai --- sound/hda/hdac_regmap.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index e1dcf104d273bd..d401e5c69fe3b6 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -88,7 +88,6 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg) switch (verb) { case AC_VERB_GET_CONNECT_SEL: case AC_VERB_GET_SDI_SELECT: - case AC_VERB_GET_CONV: case AC_VERB_GET_PIN_WIDGET_CONTROL: case AC_VERB_GET_UNSOLICITED_RESPONSE: /* only as SET_UNSOLICITED_ENABLE */ case AC_VERB_GET_BEEP_CONTROL: @@ -96,14 +95,12 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg) case AC_VERB_GET_DIGI_CONVERT_1: case AC_VERB_GET_DIGI_CONVERT_2: /* only for beep control */ case AC_VERB_GET_VOLUME_KNOB_CONTROL: - case AC_VERB_GET_CONFIG_DEFAULT: case AC_VERB_GET_GPIO_MASK: case AC_VERB_GET_GPIO_DIRECTION: case AC_VERB_GET_GPIO_DATA: /* not for volatile read */ case AC_VERB_GET_GPIO_WAKE_MASK: case AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK: case AC_VERB_GET_GPIO_STICKY_MASK: - case AC_VERB_GET_CVT_CHAN_COUNT: return true; } @@ -123,6 +120,13 @@ static bool hda_readable_reg(struct device *dev, unsigned int reg) case AC_VERB_GET_CONNECT_LIST: case AC_VERB_GET_SUBSYSTEM_ID: return true; + /* below are basically writable, but disabled for reducing unnecessary + * writes at sync + */ + case AC_VERB_GET_CONFIG_DEFAULT: /* usually just read */ + case AC_VERB_GET_CONV: /* managed in PCM code */ + case AC_VERB_GET_CVT_CHAN_COUNT: /* managed in HDMI CA code */ + return true; } return hda_writeable_reg(dev, reg); From a57069e33fbc6625f39e1b09c88ea44629a35206 Mon Sep 17 00:00:00 2001 From: Manish Badarkhe Date: Thu, 26 Mar 2015 15:38:25 +0200 Subject: [PATCH 300/411] ASoC: davinci-evm: drop un-necessary remove function As davinci card gets registered using 'devm_' api there is no need to unregister the card in 'remove' function. Hence drop the 'remove' function. Fixes: ee2f615d6e59c (ASoC: davinci-evm: Add device tree binding) Signed-off-by: Manish Badarkhe Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/davinci/davinci-evm.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b6bb5947a8a843..8c2b9be80a9a24 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -425,18 +425,8 @@ static int davinci_evm_probe(struct platform_device *pdev) return ret; } -static int davinci_evm_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static struct platform_driver davinci_evm_driver = { .probe = davinci_evm_probe, - .remove = davinci_evm_remove, .driver = { .name = "davinci_evm", .pm = &snd_soc_pm_ops, From dd38c1d4a17cda2486883cef7ec1bd84b5095260 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 24 Mar 2015 01:06:01 +0000 Subject: [PATCH 301/411] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format The DAI link format should be specified for the whole link rather than just one component on the link. So move the format specification for the HDMI audio link from the CPU component to the link itself. Since the sh-mobile-hdmi DAI driver doesn't implement the set_fmt() callback in this case there is no functional difference between only specifying the the format for the CPU side or for the whole link, but the later it will allow us to remove support for just specifying the format for one component. Signed-off-by: Lars-Peter Clausen Signed-off-by: Kuninori Morimoto Acked-by: Simon Horman Signed-off-by: Mark Brown --- arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 6d949f1c850bda..75de26c43d67ed 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1040,9 +1040,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", From 947a37cd38796f5b196a934353165a001cbcb0a9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 24 Mar 2015 01:06:40 +0000 Subject: [PATCH 302/411] ARM: shmobile: armadillo800eva: fix clock inversion When operating in left-justfied mode both the frame-clock and the bit-clock need to be inverted to be standards compliant. This means that the exta clock inversion setting in the armadillo800eva machine driver for CPU component should now be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Kuninori Morimoto Acked-by: Simon Horman Signed-off-by: Mark Brown --- arch/arm/mach-shmobile/board-armadillo800eva.c | 1 - 1 file changed, 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 75de26c43d67ed..36aaeb12e1a558 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1015,7 +1015,6 @@ static struct asoc_simple_card_info fsi_wm8978_info = { .platform = "sh_fsi2", .daifmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, .cpu_dai = { - .fmt = SND_SOC_DAIFMT_IB_NF, .name = "fsia-dai", }, .codec_dai = { From 1efb53a220b78fdfdbb97b726a2156713e75bdab Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 24 Mar 2015 01:07:08 +0000 Subject: [PATCH 303/411] ASoC: simple-card: Remove support for setting differing DAI formats Having to set different formats on the CPU side and the CODEC side of a DAI link is usually indication that something is terribly wrong and in most cases is a result of a broken driver that implements a set_fmt() callback which does not follow the specification. In the past this feature has been used to work around broken drivers, rather than fixing them. We don't really want to encourage this, so remove support for setting different formats on both ends of the link. Along the way switch to static DAI format setup by setting the the dai_fmt field of the snd_soc_dai_link rather than calling snd_soc_dai_fmt(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card.h | 1 - sound/soc/generic/simple-card.c | 30 ++++++++---------------------- 2 files changed, 8 insertions(+), 23 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 1255ddb1d3e2c2..b9b4f289fe6b32 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -16,7 +16,6 @@ struct asoc_simple_dai { const char *name; - unsigned int fmt; unsigned int sysclk; int slots; int slot_width; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5daee..3efd9472b8a359 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -125,14 +125,6 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, { int ret; - if (set->fmt) { - ret = snd_soc_dai_set_fmt(dai, set->fmt); - if (ret && ret != -ENOTSUPP) { - dev_err(dai->dev, "simple-card: set_fmt error\n"); - goto err; - } - } - if (set->sysclk) { ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); if (ret && ret != -ENOTSUPP) { @@ -269,12 +261,10 @@ static int asoc_simple_card_parse_daifmt(struct device_node *node, struct device_node *codec, char *prefix, int idx) { + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); struct device *dev = simple_priv_to_dev(priv); struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; - struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); - struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; - struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; unsigned int daifmt; daifmt = snd_soc_of_parse_daifmt(node, prefix, @@ -289,8 +279,7 @@ static int asoc_simple_card_parse_daifmt(struct device_node *node, */ dev_dbg(dev, "Revert to legacy daifmt parsing\n"); - cpu_dai->fmt = codec_dai->fmt = - snd_soc_of_parse_daifmt(codec, NULL, NULL, NULL) | + daifmt = snd_soc_of_parse_daifmt(codec, NULL, NULL, NULL) | (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK); } else { if (codec == bitclkmaster) @@ -299,11 +288,10 @@ static int asoc_simple_card_parse_daifmt(struct device_node *node, else daifmt |= (codec == framemaster) ? SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; - - cpu_dai->fmt = daifmt; - codec_dai->fmt = daifmt; } + dai_link->dai_fmt = daifmt; + of_node_put(bitclkmaster); of_node_put(framemaster); @@ -379,13 +367,12 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, dai_link->init = asoc_simple_card_dai_init; dev_dbg(dev, "\tname : %s\n", dai_link->stream_name); - dev_dbg(dev, "\tcpu : %s / %04x / %d\n", + dev_dbg(dev, "\tformat : %04x\n", dai_link->dai_fmt); + dev_dbg(dev, "\tcpu : %s / %d\n", dai_link->cpu_dai_name, - dai_props->cpu_dai.fmt, dai_props->cpu_dai.sysclk); - dev_dbg(dev, "\tcodec : %s / %04x / %d\n", + dev_dbg(dev, "\tcodec : %s / %d\n", dai_link->codec_dai_name, - dai_props->codec_dai.fmt, dai_props->codec_dai.sysclk); /* @@ -572,14 +559,13 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link->codec_name = cinfo->codec; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->codec_dai_name = cinfo->codec_dai.name; + dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_card_dai_init; memcpy(&priv->dai_props->cpu_dai, &cinfo->cpu_dai, sizeof(priv->dai_props->cpu_dai)); memcpy(&priv->dai_props->codec_dai, &cinfo->codec_dai, sizeof(priv->dai_props->codec_dai)); - priv->dai_props->cpu_dai.fmt |= cinfo->daifmt; - priv->dai_props->codec_dai.fmt |= cinfo->daifmt; } snd_soc_card_set_drvdata(&priv->snd_card, priv); From 9d82f9272ddd8492afdd721c9999171741be835b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Mar 2015 14:07:26 +0100 Subject: [PATCH 304/411] ALSA: hda - Set use_single_rw flag for regmap HD-audio doesn't support the bulk access. Currently it works even without this flag as implicitly assumed, but it's safer to set explicitly. Signed-off-by: Takashi Iwai --- sound/hda/hdac_regmap.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index d401e5c69fe3b6..1eb43209fe2c59 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -328,6 +328,7 @@ static const struct regmap_config hda_regmap_cfg = { .cache_type = REGCACHE_RBTREE, .reg_read = hda_reg_read, .reg_write = hda_reg_write, + .use_single_rw = true, }; int snd_hdac_regmap_init(struct hdac_device *codec) From 2c0ed6349287a15f7be73bba00e520106087cd1b Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Fri, 27 Mar 2015 15:03:57 +0800 Subject: [PATCH 305/411] ASoC: Intel: fix warning reported by static check tool smatch The smatch tool report warning: ... CHECK sound/soc/intel/sst-haswell-pcm.c sound/soc/intel/sst-haswell-pcm.c:1110 hsw_pcm_probe() error: buffer overflow\ 'hsw_dais' 4 <= 4 sound/soc/intel/sst-haswell-pcm.c:1112 hsw_pcm_probe() error: buffer overflow\ 'hsw_dais' 4 <= 4 ... fix it by use its own struct member for post-process module, rather than sharing unused pcm member. Signed-off-by: Lu, Han Acked-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 6c6229ae4a0259..31ffc0f0498f8e 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -137,6 +137,7 @@ struct hsw_priv_data { struct device *dev; enum hsw_pm_state pm_state; struct snd_soc_card *soc_card; + struct sst_module_runtime *runtime_waves; /* sound effect module */ /* page tables */ struct snd_dma_buffer dmab[HSW_PCM_COUNT][2]; @@ -902,13 +903,10 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) /* create runtime blocks for module waves */ if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { - pcm_data = &pdata->pcm[HSW_PCM_COUNT-1][0]; - pcm_data->runtime = sst_hsw_runtime_module_create(hsw, - SST_HSW_MODULE_WAVES, pcm_data->persistent_offset); - if (pcm_data->runtime == NULL) + pdata->runtime_waves = sst_hsw_runtime_module_create(hsw, + SST_HSW_MODULE_WAVES, 0); + if (pdata->runtime_waves == NULL) goto err; - pcm_data->persistent_offset = - pcm_data->runtime->persistent_offset; } return 0; @@ -933,8 +931,7 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) sst_hsw_runtime_module_free(pcm_data->runtime); } if (sst_hsw_is_module_loaded(hsw, SST_HSW_MODULE_WAVES)) { - pcm_data = &pdata->pcm[HSW_PCM_COUNT-1][0]; - sst_hsw_runtime_module_free(pcm_data->runtime); + sst_hsw_runtime_module_free(pdata->runtime_waves); } } From 57bf27365c56b547097253004768904419df6c69 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 27 Mar 2015 20:19:06 +0800 Subject: [PATCH 306/411] ASoC: rt5645: Redefine format config for rt5650 rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a14de1d26..0133c8c58592b2 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: From afefc12801e501fea90f1d9a678e0985f47dc1bf Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 27 Mar 2015 20:19:07 +0800 Subject: [PATCH 307/411] ASoC: rt5645: Set use_single_rw flag for regmap RT5645 doesn't support auto incrementing writes so driver should set the use_single_rw flag for regmap. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 0133c8c58592b2..f9edf09253d9f4 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2633,7 +2633,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5645 = { static const struct regmap_config rt5645_regmap = { .reg_bits = 8, .val_bits = 16, - + .use_single_rw = true, .max_register = RT5645_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5645_ranges) * RT5645_PR_SPACING), .volatile_reg = rt5645_volatile_register, From 1b5d0160e8f17db0714016a2550d3b1d65c70c3e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 27 Mar 2015 20:19:08 +0800 Subject: [PATCH 308/411] ASoC: rt5645: Use update_bits for bit control In codec bias level off, we need to disable gate mode with MCLK for power saving. It is set by one bit. We don't need to write while register for that. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f9edf09253d9f4..b6d5b9570efb59 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2396,7 +2396,8 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); - snd_soc_write(codec, RT5645_GEN_CTRL1, 0x0128); + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, 0); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2 | From 373225510f9608150a18b3491e756fbf3f58ff24 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 27 Mar 2015 20:19:09 +0800 Subject: [PATCH 309/411] ASoC: rt5645: Restore HP depop setting in HP off This driver will set RT5645_DEPOP_MAN bit in headphone power up depop process. We need to restore it in headphone power down process. Otherwise, we will get headphone noise when push button function is enabled. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index b6d5b9570efb59..69528ae5410c99 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1270,6 +1270,8 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on) snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_HP_L | RT5645_PWR_HP_R | RT5645_PWR_HA, 0); + snd_soc_update_bits(codec, RT5645_DEPOP_M2, + RT5645_DEPOP_MASK, 0); } } } From 5116ede10dc92d7a11a38fc59f545c24a33809cd Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 27 Mar 2015 17:25:21 +0800 Subject: [PATCH 310/411] ASoC: max98925: Fix bit-width 24 settings in max98925_dai_hw_params Trivial typo fix. Signed-off-by: Axel Lin Anish Kumar Signed-off-by: Mark Brown --- sound/soc/codecs/max98925.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 0796dfaeb1394f..9b5a17de469091 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -442,8 +442,8 @@ static int max98925_dai_hw_params(struct snd_pcm_substream *substream, case 24: regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); - max98925->ch_size = 32; + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_24); + max98925->ch_size = 24; break; case 32: regmap_update_bits(max98925->regmap, From 415f1cb29d3be865b034b528058c7115bc262f43 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 26 Mar 2015 04:01:27 +0000 Subject: [PATCH 311/411] ASoC: rsrc-card: add Renesas sampling rate convert sound card support Renesas sound card has "sampling rate convert" feature which should be implemented via DPCM. But, sound card driver point of view, it is difficult to add this DPCM feature on simple-card driver. Especially, DT binding support is very difficult. This patch implements DPCM feature on DT as Renesas specific sound card. This new driver is copied from current simple-card driver. Main difference between simple-card and this driver are... 1. removed unused feature from simple-card 2. removed driver named prefix from DT property 3. CPU will be FE, CODEC will be BE with snd-soc-dummy 4. it supports sampling rate convert via .be_hw_params_fixup 5. board specific routing is implemented in driver Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/renesas,rsrc-card.txt | 66 +++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/rsrc-card.c | 489 ++++++++++++++++++ 4 files changed, 564 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt create mode 100644 sound/soc/sh/rcar/rsrc-card.c diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt new file mode 100644 index 00000000000000..12e287ed4dceed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt @@ -0,0 +1,66 @@ +Renesas Sampling Rate Convert Sound Card: + +Renesas Sampling Rate Convert Sound Card specifies audio DAI connections of SoC <-> codec. + +Required properties: + +- compatible : "renesas,rsrc-card," + Examples with soctypes are: + - "renesas,rsrc-card,lager" + - "renesas,rsrc-card,koelsch" +Optional properties: + +- card_name : User specified audio sound card name, one string + property. +- cpu : CPU sub-node +- codec : CODEC sub-node + +Optional subnode properties: + +- format : CPU/CODEC common audio format. + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" +- frame-master : Indicates dai-link frame master. + phandle to a cpu or codec subnode. +- bitclock-master : Indicates dai-link bit clock master. + phandle to a cpu or codec subnode. +- bitclock-inversion : bool property. Add this if the + dai-link uses bit clock inversion. +- frame-inversion : bool property. Add this if the + dai-link uses frame clock inversion. + +Required CPU/CODEC subnodes properties: + +- sound-dai : phandle and port of CPU/CODEC + +Optional CPU/CODEC subnodes properties: + +- clocks / system-clock-frequency : specify subnode's clock if needed. + it can be specified via "clocks" if system has + clock node (= common clock), or "system-clock-frequency" + (if system doens't support common clock) + If a clock is specified, it is + enabled with clk_prepare_enable() + in dai startup() and disabled with + clk_disable_unprepare() in dai + shutdown(). + +Example + +sound { + compatible = "renesas,rsrc-card,lager"; + + card-name = "rsnd-ak4643"; + format = "left_j"; + bitclock-master = <&sndcodec>; + frame-master = <&sndcodec>; + + sndcpu: cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; +}; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 80245b6eebd653..2b30304155734f 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -41,6 +41,11 @@ config SND_SOC_RCAR help This option enables R-Car SUR/SCU/SSIU/SSI sound support +config SND_SOC_RSRC_CARD + tristate "Renesas Sampling Rate Convert Sound Card" + help + This option enables simple sound if you need sampling rate convert + ## ## Boards ## diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index 7b204925b8c5b2..f1b445173fba78 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,2 +1,5 @@ snd-soc-rcar-objs := core.o gen.o dma.o src.o adg.o ssi.o dvc.o -obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o \ No newline at end of file +obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o + +snd-soc-rsrc-card-objs := rsrc-card.o +obj-$(CONFIG_SND_SOC_RSRC_CARD) += snd-soc-rsrc-card.o diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c new file mode 100644 index 00000000000000..3baeab726bc33d --- /dev/null +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -0,0 +1,489 @@ +/* + * Renesas Sampling Rate Convert Sound Card for DPCM + * + * Copyright (C) 2015 Renesas Solutions Corp. + * Kuninori Morimoto + * + * based on ${LINUX}/sound/soc/generic/simple-card.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +struct rsrc_card_of_data { + const char *prefix; + const struct snd_soc_dapm_route *routes; + int num_routes; +}; + +static const struct snd_soc_dapm_route routes_ssi0_ak4642[] = { + {"ak4642 Playback", NULL, "DAI0 Playback"}, + {"DAI0 Capture", NULL, "ak4642 Capture"}, +}; + +static const struct rsrc_card_of_data routes_of_ssi0_ak4642 = { + .prefix = "ak4642", + .routes = routes_ssi0_ak4642, + .num_routes = ARRAY_SIZE(routes_ssi0_ak4642), +}; + +static const struct of_device_id rsrc_card_of_match[] = { + { .compatible = "renesas,rsrc-card,lager", .data = &routes_of_ssi0_ak4642 }, + { .compatible = "renesas,rsrc-card,koelsch", .data = &routes_of_ssi0_ak4642 }, + {}, +}; +MODULE_DEVICE_TABLE(of, rsrc_card_of_match); + +struct rsrc_card_dai { + const char *name; + unsigned int fmt; + unsigned int sysclk; + struct clk *clk; +}; + +#define RSRC_FB_NUM 2 /* FE/BE */ +#define IDX_CPU 0 +#define IDX_CODEC 1 +struct rsrc_card_priv { + struct snd_soc_card snd_card; + struct rsrc_card_dai_props { + struct rsrc_card_dai cpu_dai; + struct rsrc_card_dai codec_dai; + } dai_props[RSRC_FB_NUM]; + struct snd_soc_codec_conf codec_conf; + struct snd_soc_dai_link dai_link[RSRC_FB_NUM]; +}; + +#define rsrc_priv_to_dev(priv) ((priv)->snd_card.dev) +#define rsrc_priv_to_link(priv, i) ((priv)->snd_card.dai_link + i) +#define rsrc_priv_to_props(priv, i) ((priv)->dai_props + i) +#define rsrc_dev_to_of_data(dev) (of_match_device(rsrc_card_of_match, (dev))->data) + +static int rsrc_card_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rsrc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct rsrc_card_dai_props *dai_props = + &priv->dai_props[rtd - rtd->card->rtd]; + int ret; + + ret = clk_prepare_enable(dai_props->cpu_dai.clk); + if (ret) + return ret; + + ret = clk_prepare_enable(dai_props->codec_dai.clk); + if (ret) + clk_disable_unprepare(dai_props->cpu_dai.clk); + + return ret; +} + +static void rsrc_card_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rsrc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct rsrc_card_dai_props *dai_props = + &priv->dai_props[rtd - rtd->card->rtd]; + + clk_disable_unprepare(dai_props->cpu_dai.clk); + + clk_disable_unprepare(dai_props->codec_dai.clk); +} + +static struct snd_soc_ops rsrc_card_ops = { + .startup = rsrc_card_startup, + .shutdown = rsrc_card_shutdown, +}; + +static int __rsrc_card_dai_init(struct snd_soc_dai *dai, + struct rsrc_card_dai *set) +{ + int ret; + + if (set->fmt) { + ret = snd_soc_dai_set_fmt(dai, set->fmt); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "set_fmt error\n"); + goto err; + } + } + + if (set->sysclk) { + ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "set_sysclk error\n"); + goto err; + } + } + + ret = 0; + +err: + return ret; +} + +static int rsrc_card_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct rsrc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *codec = rtd->codec_dai; + struct snd_soc_dai *cpu = rtd->cpu_dai; + struct rsrc_card_dai_props *dai_props; + int num, ret; + + num = rtd - rtd->card->rtd; + dai_props = &priv->dai_props[num]; + ret = __rsrc_card_dai_init(codec, &dai_props->codec_dai); + if (ret < 0) + return ret; + + ret = __rsrc_card_dai_init(cpu, &dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; +} + +static int +rsrc_card_sub_parse_of(struct rsrc_card_priv *priv, + struct device_node *np, + struct rsrc_card_dai *dai, + struct snd_soc_dai_link *dai_link, + int *args_count) +{ + struct device *dev = rsrc_priv_to_dev(priv); + const struct rsrc_card_of_data *of_data = rsrc_dev_to_of_data(dev); + struct of_phandle_args args; + struct device_node **p_node; + struct clk *clk; + const char **dai_name; + const char **name; + u32 val; + int ret; + + if (args_count) { + p_node = &dai_link->cpu_of_node; + dai_name = &dai_link->cpu_dai_name; + name = &dai_link->cpu_name; + } else { + p_node = &dai_link->codec_of_node; + dai_name = &dai_link->codec_dai_name; + name = &dai_link->codec_name; + } + + if (!np) { + /* use snd-soc-dummy */ + *p_node = NULL; + *dai_name = "snd-soc-dummy-dai"; + *name = "snd-soc-dummy"; + return 0; + } + + /* + * Get node via "sound-dai = <&phandle port>" + * it will be used as xxx_of_node on soc_bind_dai_link() + */ + ret = of_parse_phandle_with_args(np, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) + return ret; + + *p_node = args.np; + + /* Get dai->name */ + ret = snd_soc_of_get_dai_name(np, dai_name); + if (ret < 0) + return ret; + + /* + * FIXME + * + * rsrc assumes DPCM playback/capture + */ + dai_link->dpcm_playback = 1; + dai_link->dpcm_capture = 1; + + if (args_count) { + *args_count = args.args_count; + dai_link->dynamic = 1; + } else { + dai_link->no_pcm = 1; + priv->codec_conf.of_node = (*p_node); + priv->codec_conf.name_prefix = of_data->prefix; + } + + /* + * Parse dai->sysclk come from "clocks = <&xxx>" + * (if system has common clock) + * or "system-clock-frequency = " + * or device's module clock. + */ + if (of_property_read_bool(np, "clocks")) { + clk = of_clk_get(np, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + return ret; + } + + dai->sysclk = clk_get_rate(clk); + dai->clk = clk; + } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { + dai->sysclk = val; + } else { + clk = of_clk_get(args.np, 0); + if (!IS_ERR(clk)) + dai->sysclk = clk_get_rate(clk); + } + + return 0; +} + +static int rsrc_card_parse_daifmt(struct device_node *node, + struct rsrc_card_priv *priv, + struct device_node *codec, + int idx) +{ + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + struct rsrc_card_dai_props *dai_props = rsrc_priv_to_props(priv, idx); + struct rsrc_card_dai *cpu_dai = &dai_props->cpu_dai; + struct rsrc_card_dai *codec_dai = &dai_props->codec_dai; + unsigned int daifmt; + + daifmt = snd_soc_of_parse_daifmt(node, NULL, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + + if (!bitclkmaster && !framemaster) + return -EINVAL; + + if (codec == bitclkmaster) + daifmt |= (codec == framemaster) ? + SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; + else + daifmt |= (codec == framemaster) ? + SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; + + cpu_dai->fmt = daifmt; + codec_dai->fmt = daifmt; + + of_node_put(bitclkmaster); + of_node_put(framemaster); + + return 0; +} + +static int rsrc_card_dai_link_of(struct device_node *node, + struct rsrc_card_priv *priv, + int idx) +{ + struct device *dev = rsrc_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = rsrc_priv_to_link(priv, idx); + struct rsrc_card_dai_props *dai_props = rsrc_priv_to_props(priv, idx); + struct device_node *cpu = NULL; + struct device_node *codec = NULL; + char *name; + char prop[128]; + int ret, cpu_args; + + cpu = of_get_child_by_name(node, "cpu"); + codec = of_get_child_by_name(node, "codec"); + + if (!cpu || !codec) { + ret = -EINVAL; + dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); + goto dai_link_of_err; + } + + ret = rsrc_card_parse_daifmt(node, priv, codec, idx); + if (ret < 0) + goto dai_link_of_err; + + ret = rsrc_card_sub_parse_of(priv, (idx == IDX_CPU) ? cpu : NULL, + &dai_props->cpu_dai, + dai_link, + &cpu_args); + if (ret < 0) + goto dai_link_of_err; + + ret = rsrc_card_sub_parse_of(priv, (idx == IDX_CODEC) ? codec : NULL, + &dai_props->codec_dai, + dai_link, + NULL); + if (ret < 0) + goto dai_link_of_err; + + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) { + ret = -EINVAL; + goto dai_link_of_err; + } + + /* Simple Card assumes platform == cpu */ + dai_link->platform_of_node = dai_link->cpu_of_node; + + /* DAI link name is created from CPU/CODEC dai name */ + name = devm_kzalloc(dev, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + 2, + GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto dai_link_of_err; + } + + sprintf(name, "%s-%s", dai_link->cpu_dai_name, + dai_link->codec_dai_name); + dai_link->name = dai_link->stream_name = name; + dai_link->ops = &rsrc_card_ops; + dai_link->init = rsrc_card_dai_init; + + dev_dbg(dev, "\tname : %s\n", dai_link->stream_name); + dev_dbg(dev, "\tcpu : %s / %04x / %d\n", + dai_link->cpu_dai_name, + dai_props->cpu_dai.fmt, + dai_props->cpu_dai.sysclk); + dev_dbg(dev, "\tcodec : %s / %04x / %d\n", + dai_link->codec_dai_name, + dai_props->codec_dai.fmt, + dai_props->codec_dai.sysclk); + + /* + * In soc_bind_dai_link() will check cpu name after + * of_node matching if dai_link has cpu_dai_name. + * but, it will never match if name was created by + * fmt_single_name() remove cpu_dai_name if cpu_args + * was 0. See: + * fmt_single_name() + * fmt_multiple_name() + */ + if (!cpu_args) + dai_link->cpu_dai_name = NULL; + +dai_link_of_err: + of_node_put(cpu); + of_node_put(codec); + + return ret; +} + +static int rsrc_card_parse_of(struct device_node *node, + struct rsrc_card_priv *priv) +{ + struct device *dev = rsrc_priv_to_dev(priv); + const struct rsrc_card_of_data *of_data = rsrc_dev_to_of_data(dev); + int ret; + int i; + + if (!node) + return -EINVAL; + + /* Parse the card name from DT */ + snd_soc_of_parse_card_name(&priv->snd_card, "card-name"); + + /* DAPM routes */ + priv->snd_card.of_dapm_routes = of_data->routes; + priv->snd_card.num_of_dapm_routes = of_data->num_routes; + + dev_dbg(dev, "New rsrc-audio-card: %s\n", priv->snd_card.name ? + priv->snd_card.name : ""); + + /* FE/BE */ + for (i = 0; i < RSRC_FB_NUM; i++) { + ret = rsrc_card_dai_link_of(node, priv, i); + if (ret < 0) + return ret; + } + + if (!priv->snd_card.name) + priv->snd_card.name = priv->snd_card.dai_link->name; + + return 0; +} + +/* Decrease the reference count of the device nodes */ +static int rsrc_card_unref(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + int num_links; + + for (num_links = 0, dai_link = card->dai_link; + num_links < card->num_links; + num_links++, dai_link++) { + of_node_put(dai_link->cpu_of_node); + of_node_put(dai_link->codec_of_node); + } + return 0; +} + +static int rsrc_card_probe(struct platform_device *pdev) +{ + struct rsrc_card_priv *priv; + struct snd_soc_dai_link *dai_link; + struct device_node *np = pdev->dev.of_node; + struct device *dev = &pdev->dev; + int ret; + + /* Allocate the private data */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + /* Init snd_soc_card */ + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = dev; + dai_link = priv->dai_link; + priv->snd_card.dai_link = dai_link; + priv->snd_card.num_links = RSRC_FB_NUM; + priv->snd_card.codec_conf = &priv->codec_conf; + priv->snd_card.num_configs = 1; + + ret = rsrc_card_parse_of(np, priv); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + goto err; + } + + snd_soc_card_set_drvdata(&priv->snd_card, priv); + + ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); + if (ret >= 0) + return ret; +err: + rsrc_card_unref(&priv->snd_card); + + return ret; +} + +static int rsrc_card_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + return rsrc_card_unref(card); +} + +static struct platform_driver rsrc_card = { + .driver = { + .name = "renesas-src-audio-card", + .of_match_table = rsrc_card_of_match, + }, + .probe = rsrc_card_probe, + .remove = rsrc_card_remove, +}; + +module_platform_driver(rsrc_card); + +MODULE_ALIAS("platform:renesas-src-audio-card"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Renesas Sampling Rate Convert Sound Card"); +MODULE_AUTHOR("Kuninori Morimoto "); From af7e2be96623785816c1a6e521307b7af11ad016 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 26 Mar 2015 04:01:46 +0000 Subject: [PATCH 312/411] ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate Current rsnd-dpcm-card is supporting DPCM FE/BE sound card. This patch adds .be_hw_params_fixup and enabled sampling convert rate. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- .../bindings/sound/renesas,rsrc-card.txt | 1 + sound/soc/sh/rcar/rsrc-card.c | 27 +++++++++++++++++-- 2 files changed, 26 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt index 12e287ed4dceed..c64155027288e2 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt @@ -28,6 +28,7 @@ Optional subnode properties: dai-link uses bit clock inversion. - frame-inversion : bool property. Add this if the dai-link uses frame clock inversion. +- convert-rate : platform specified sampling rate convert Required CPU/CODEC subnodes properties: diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index 3baeab726bc33d..a68517afe61596 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -63,6 +63,7 @@ struct rsrc_card_priv { } dai_props[RSRC_FB_NUM]; struct snd_soc_codec_conf codec_conf; struct snd_soc_dai_link dai_link[RSRC_FB_NUM]; + u32 convert_rate; }; #define rsrc_priv_to_dev(priv) ((priv)->snd_card.dev) @@ -154,6 +155,21 @@ static int rsrc_card_dai_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int rsrc_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct rsrc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + + if (!priv->convert_rate) + return 0; + + rate->min = rate->max = priv->convert_rate; + + return 0; +} + static int rsrc_card_sub_parse_of(struct rsrc_card_priv *priv, struct device_node *np, @@ -347,6 +363,9 @@ static int rsrc_card_dai_link_of(struct device_node *node, dai_link->ops = &rsrc_card_ops; dai_link->init = rsrc_card_dai_init; + if (idx == IDX_CODEC) + dai_link->be_hw_params_fixup = rsrc_card_be_hw_params_fixup; + dev_dbg(dev, "\tname : %s\n", dai_link->stream_name); dev_dbg(dev, "\tcpu : %s / %04x / %d\n", dai_link->cpu_dai_name, @@ -394,8 +413,12 @@ static int rsrc_card_parse_of(struct device_node *node, priv->snd_card.of_dapm_routes = of_data->routes; priv->snd_card.num_of_dapm_routes = of_data->num_routes; - dev_dbg(dev, "New rsrc-audio-card: %s\n", priv->snd_card.name ? - priv->snd_card.name : ""); + /* sampling rate convert */ + of_property_read_u32(node, "convert-rate", &priv->convert_rate); + + dev_dbg(dev, "New rsrc-audio-card: %s (%d)\n", + priv->snd_card.name ? priv->snd_card.name : "", + priv->convert_rate); /* FE/BE */ for (i = 0; i < RSRC_FB_NUM; i++) { From 2f78dd7f40264697afed4c2ac0890df8f0588e49 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 26 Mar 2015 04:02:09 +0000 Subject: [PATCH 313/411] ASoC: rsnd: call clk_prepare/unprepare() in probe/remove clk_prepare_enable()/clk_disable_unprepare() uses mutex inside, in concretely clk_prepare()/clk_unprepare().And it uses __schedule(). Then, raw_spin_lock/unlock_irq() is called, and it breaks Renesas sound driver's spin lock irq. This patch separates thesse into clk_prepare()/clk_unprepare() and clk_enable/clk_disable. And call clk_prepare()/clk_unprepare() from probe/remove function. Special thanks to Das Biju. Reported-by: Das Biju Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 24 +++++++++++++++++++++++- sound/soc/sh/rcar/dvc.c | 17 +++++++++++++++-- sound/soc/sh/rcar/rsnd.h | 11 ++++++++--- sound/soc/sh/rcar/src.c | 17 +++++++++++++++-- sound/soc/sh/rcar/ssi.c | 17 +++++++++++++++-- 5 files changed, 76 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1b53605f715439..6046c10ef3c7f9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -149,16 +149,29 @@ char *rsnd_mod_dma_name(struct rsnd_mod *mod) return mod->ops->dma_name(mod); } -void rsnd_mod_init(struct rsnd_mod *mod, +int rsnd_mod_init(struct rsnd_mod *mod, struct rsnd_mod_ops *ops, struct clk *clk, enum rsnd_mod_type type, int id) { + int ret = clk_prepare(clk); + + if (ret) + return ret; + mod->id = id; mod->ops = ops; mod->type = type; mod->clk = clk; + + return ret; +} + +void rsnd_mod_quit(struct rsnd_mod *mod) +{ + if (mod->clk) + clk_unprepare(mod->clk); } /* @@ -1290,6 +1303,12 @@ static int rsnd_remove(struct platform_device *pdev) { struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev); struct rsnd_dai *rdai; + void (*remove_func[])(struct platform_device *pdev, + struct rsnd_priv *priv) = { + rsnd_ssi_remove, + rsnd_src_remove, + rsnd_dvc_remove, + }; int ret = 0, i; pm_runtime_disable(&pdev->dev); @@ -1299,6 +1318,9 @@ static int rsnd_remove(struct platform_device *pdev) ret |= rsnd_dai_call(remove, &rdai->capture, priv); } + for (i = 0; i < ARRAY_SIZE(remove_func); i++) + remove_func[i](pdev, priv); + snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index d7f9ed959c4e20..261997a3f5899d 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -333,7 +333,7 @@ int rsnd_dvc_probe(struct platform_device *pdev, struct rsnd_dvc *dvc; struct clk *clk; char name[RSND_DVC_NAME_SIZE]; - int i, nr; + int i, nr, ret; rsnd_of_parse_dvc(pdev, of_data, priv); @@ -366,11 +366,24 @@ int rsnd_dvc_probe(struct platform_device *pdev, dvc->info = &info->dvc_info[i]; - rsnd_mod_init(&dvc->mod, &rsnd_dvc_ops, + ret = rsnd_mod_init(&dvc->mod, &rsnd_dvc_ops, clk, RSND_MOD_DVC, i); + if (ret) + return ret; dev_dbg(dev, "CMD%d probed\n", i); } return 0; } + +void rsnd_dvc_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + struct rsnd_dvc *dvc; + int i; + + for_each_rsnd_dvc(dvc, priv, i) { + rsnd_mod_quit(&dvc->mod); + } +} diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e7914bd610e212..1bccc5515b5a9b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -260,14 +260,15 @@ struct rsnd_mod { #define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma) #define rsnd_mod_to_io(mod) ((mod)->io) #define rsnd_mod_id(mod) ((mod)->id) -#define rsnd_mod_hw_start(mod) clk_prepare_enable((mod)->clk) -#define rsnd_mod_hw_stop(mod) clk_disable_unprepare((mod)->clk) +#define rsnd_mod_hw_start(mod) clk_enable((mod)->clk) +#define rsnd_mod_hw_stop(mod) clk_disable((mod)->clk) -void rsnd_mod_init(struct rsnd_mod *mod, +int rsnd_mod_init(struct rsnd_mod *mod, struct rsnd_mod_ops *ops, struct clk *clk, enum rsnd_mod_type type, int id); +void rsnd_mod_quit(struct rsnd_mod *mod); char *rsnd_mod_name(struct rsnd_mod *mod); char *rsnd_mod_dma_name(struct rsnd_mod *mod); @@ -480,6 +481,8 @@ int rsnd_kctrl_new_e(struct rsnd_mod *mod, int rsnd_src_probe(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv); +void rsnd_src_remove(struct platform_device *pdev, + struct rsnd_priv *priv); struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id); unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, @@ -498,6 +501,8 @@ int rsnd_src_ssi_irq_disable(struct rsnd_mod *ssi_mod); int rsnd_ssi_probe(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv); +void rsnd_ssi_remove(struct platform_device *pdev, + struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod); int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 81c182b4bad531..c77d059edc84ce 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -850,7 +850,7 @@ int rsnd_src_probe(struct platform_device *pdev, struct rsnd_mod_ops *ops; struct clk *clk; char name[RSND_SRC_NAME_SIZE]; - int i, nr; + int i, nr, ret; ops = NULL; if (rsnd_is_gen1(priv)) @@ -890,10 +890,23 @@ int rsnd_src_probe(struct platform_device *pdev, src->info = &info->src_info[i]; - rsnd_mod_init(&src->mod, ops, clk, RSND_MOD_SRC, i); + ret = rsnd_mod_init(&src->mod, ops, clk, RSND_MOD_SRC, i); + if (ret) + return ret; dev_dbg(dev, "SRC%d probed\n", i); } return 0; } + +void rsnd_src_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + struct rsnd_src *src; + int i; + + for_each_rsnd_src(src, priv, i) { + rsnd_mod_quit(&src->mod); + } +} diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 9e7b627c08e225..f7cb1fd635a0eb 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -697,7 +697,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, struct clk *clk; struct rsnd_ssi *ssi; char name[RSND_SSI_NAME_SIZE]; - int i, nr; + int i, nr, ret; rsnd_of_parse_ssi(pdev, of_data, priv); @@ -732,10 +732,23 @@ int rsnd_ssi_probe(struct platform_device *pdev, else if (rsnd_ssi_pio_available(ssi)) ops = &rsnd_ssi_pio_ops; - rsnd_mod_init(&ssi->mod, ops, clk, RSND_MOD_SSI, i); + ret = rsnd_mod_init(&ssi->mod, ops, clk, RSND_MOD_SSI, i); + if (ret) + return ret; rsnd_ssi_parent_clk_setup(priv, ssi); } return 0; } + +void rsnd_ssi_remove(struct platform_device *pdev, + struct rsnd_priv *priv) +{ + struct rsnd_ssi *ssi; + int i; + + for_each_rsnd_ssi(ssi, priv, i) { + rsnd_mod_quit(&ssi->mod); + } +} From b543b52a44c4e45283cd17721af1299049405136 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 26 Mar 2015 04:02:32 +0000 Subject: [PATCH 314/411] ASoC: rsnd: remove useless debug message This patch removes useless debug message. especially some kind of "probed" message will be printed from core.c if it has #define DEBUG Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 2 -- sound/soc/sh/rcar/dma.c | 4 ---- sound/soc/sh/rcar/dvc.c | 14 -------------- sound/soc/sh/rcar/gen.c | 6 ------ sound/soc/sh/rcar/src.c | 27 +-------------------------- sound/soc/sh/rcar/ssi.c | 19 +------------------ 6 files changed, 2 insertions(+), 70 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 7af374bd0849a9..fefc881dbac24a 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -434,7 +434,5 @@ int rsnd_adg_probe(struct platform_device *pdev, priv->adg = adg; - dev_dbg(dev, "adg probed\n"); - return 0; } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index cd7b79a01ce239..ac3756f6af603e 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -144,8 +144,6 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, return -EIO; } - dev_dbg(dev, "Audio DMAC init\n"); - if (dev->of_node) { dmaen->chan = rsnd_dmaen_request_channel(mod_from, mod_to); } else { @@ -329,8 +327,6 @@ static int rsnd_dmapp_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "Audio DMAC peri peri init\n"); - dmapp->dmapp_id = dmac->dmapp_num; dmapp->chcr = rsnd_dmapp_get_chcr(mod_from, mod_to) | PDMACHCR_DE; diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index aab45267e508c5..e5fcb062ad7723 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -119,17 +119,6 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) rsnd_mod_write(mod, DVC_DVUER, 1); } -static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod, - struct rsnd_priv *priv) -{ - struct device *dev = rsnd_priv_to_dev(priv); - - dev_dbg(dev, "%s[%d] (Gen2) is probed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); - - return 0; -} - static int rsnd_dvc_remove_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv) { @@ -283,7 +272,6 @@ static struct dma_chan *rsnd_dvc_dma_req(struct rsnd_mod *mod) static struct rsnd_mod_ops rsnd_dvc_ops = { .name = DVC_NAME, .dma_req = rsnd_dvc_dma_req, - .probe = rsnd_dvc_probe_gen2, .remove = rsnd_dvc_remove_gen2, .init = rsnd_dvc_init, .quit = rsnd_dvc_quit, @@ -382,8 +370,6 @@ int rsnd_dvc_probe(struct platform_device *pdev, clk, RSND_MOD_DVC, i); if (ret) return ret; - - dev_dbg(dev, "CMD%d probed\n", i); } return 0; diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index a17a504d93b533..8c7dc51b1c4fd8 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -194,7 +194,6 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, static int rsnd_gen2_probe(struct platform_device *pdev, struct rsnd_priv *priv) { - struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_regmap_field_conf conf_ssiu[] = { RSND_GEN_S_REG(SSI_MODE0, 0x800), RSND_GEN_S_REG(SSI_MODE1, 0x804), @@ -278,8 +277,6 @@ static int rsnd_gen2_probe(struct platform_device *pdev, ret_ssi < 0) return ret_ssiu | ret_scu | ret_adg | ret_ssi; - dev_dbg(dev, "Gen2 is probed\n"); - return 0; } @@ -290,7 +287,6 @@ static int rsnd_gen2_probe(struct platform_device *pdev, static int rsnd_gen1_probe(struct platform_device *pdev, struct rsnd_priv *priv) { - struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_regmap_field_conf conf_sru[] = { RSND_GEN_S_REG(SRC_ROUTE_SEL, 0x00), RSND_GEN_S_REG(SRC_TMG_SEL0, 0x08), @@ -348,8 +344,6 @@ static int rsnd_gen1_probe(struct platform_device *pdev, ret_ssi < 0) return ret_sru | ret_adg | ret_ssi; - dev_dbg(dev, "Gen1 is probed\n"); - return 0; } diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6099a8ee000718..83611fa450bf02 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -460,17 +460,6 @@ static int rsnd_src_set_convert_rate_gen1(struct rsnd_mod *mod) return 0; } -static int rsnd_src_probe_gen1(struct rsnd_mod *mod, - struct rsnd_priv *priv) -{ - struct device *dev = rsnd_priv_to_dev(priv); - - dev_dbg(dev, "%s[%d] (Gen1) is probed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); - - return 0; -} - static int rsnd_src_init_gen1(struct rsnd_mod *mod, struct rsnd_priv *priv) { @@ -518,7 +507,6 @@ static int rsnd_src_stop_gen1(struct rsnd_mod *mod, static struct rsnd_mod_ops rsnd_src_gen1_ops = { .name = SRC_NAME, .dma_req = rsnd_src_dma_req, - .probe = rsnd_src_probe_gen1, .init = rsnd_src_init_gen1, .quit = rsnd_src_quit, .start = rsnd_src_start_gen1, @@ -725,23 +713,12 @@ static int rsnd_src_probe_gen2(struct rsnd_mod *mod, IRQF_SHARED, dev_name(dev), mod); if (ret) - goto rsnd_src_probe_gen2_fail; + return ret; } ret = rsnd_dma_init(priv, rsnd_mod_to_dma(mod), src->info->dma_id); - if (ret) - goto rsnd_src_probe_gen2_fail; - - dev_dbg(dev, "%s[%d] (Gen2) is probed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); - - return ret; - -rsnd_src_probe_gen2_fail: - dev_err(dev, "%s[%d] (Gen2) failed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } @@ -910,8 +887,6 @@ int rsnd_src_probe(struct platform_device *pdev, ret = rsnd_mod_init(&src->mod, ops, clk, RSND_MOD_SRC, i); if (ret) return ret; - - dev_dbg(dev, "SRC%d probed\n", i); } return 0; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 79dc7a3c78e6de..7bb9c087f3dc45 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -445,12 +445,6 @@ static int rsnd_ssi_pio_probe(struct rsnd_mod *mod, rsnd_ssi_interrupt, IRQF_SHARED, dev_name(dev), ssi); - if (ret) - dev_err(dev, "%s[%d] (PIO) request interrupt failed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); - else - dev_dbg(dev, "%s[%d] (PIO) is probed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } @@ -477,22 +471,11 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, IRQF_SHARED, dev_name(dev), ssi); if (ret) - goto rsnd_ssi_dma_probe_fail; + return ret; ret = rsnd_dma_init( priv, rsnd_mod_to_dma(mod), dma_id); - if (ret) - goto rsnd_ssi_dma_probe_fail; - - dev_dbg(dev, "%s[%d] (DMA) is probed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); - - return ret; - -rsnd_ssi_dma_probe_fail: - dev_err(dev, "%s[%d] (DMA) is failed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } From 3b7843ff618f63d1776abd71de7eea9130987037 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 26 Mar 2015 04:02:51 +0000 Subject: [PATCH 315/411] ASoC: rsnd: add DPCM based sampling rate convert This patch supports DPCM based sampling rate convert on Renesas sound driver. It assumes... 1. SRC is implemented as FE 2. BE dai_link supports .be_hw_params_fixup Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 11 ++++++++- sound/soc/sh/rcar/rsnd.h | 8 +++++++ sound/soc/sh/rcar/src.c | 49 ++++++++++++++++++++++++++++++++++++---- 3 files changed, 63 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 770247cddb6c44..9b0de428c45b4d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -203,7 +203,7 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod) ({ \ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \ struct device *dev = rsnd_priv_to_dev(priv); \ - u32 mask = 1 << __rsnd_mod_shift_##func; \ + u32 mask = (1 << __rsnd_mod_shift_##func) & ~(1 << 31); \ u32 call = __rsnd_mod_call_##func << __rsnd_mod_shift_##func; \ int ret = 0; \ if ((mod->status & mask) == call) { \ @@ -728,6 +728,15 @@ static int rsnd_pcm_open(struct snd_pcm_substream *substream) static int rsnd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct snd_soc_dai *dai = rsnd_substream_to_dai(substream); + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + struct rsnd_dai_stream *io = rsnd_rdai_to_io(rdai, substream); + int ret; + + ret = rsnd_dai_call(hw_params, io, substream, hw_params); + if (ret) + return ret; + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index f7af0be1155838..4e6de6804cfb17 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -239,6 +239,9 @@ struct rsnd_mod_ops { struct rsnd_priv *priv); int (*pcm_new)(struct rsnd_mod *mod, struct snd_soc_pcm_runtime *rtd); + int (*hw_params)(struct rsnd_mod *mod, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); int (*fallback)(struct rsnd_mod *mod, struct rsnd_priv *priv); }; @@ -262,6 +265,9 @@ struct rsnd_mod { * 2 0: start 1: stop * 3 0: pcm_new * 4 0: fallback + * + * 31 bit is always called (see __rsnd_mod_call) + * 31 0: hw_params */ #define __rsnd_mod_shift_probe 0 #define __rsnd_mod_shift_remove 0 @@ -271,6 +277,7 @@ struct rsnd_mod { #define __rsnd_mod_shift_stop 2 #define __rsnd_mod_shift_pcm_new 3 #define __rsnd_mod_shift_fallback 4 +#define __rsnd_mod_shift_hw_params 31 /* always called */ #define __rsnd_mod_call_probe 0 #define __rsnd_mod_call_remove 1 @@ -280,6 +287,7 @@ struct rsnd_mod { #define __rsnd_mod_call_stop 1 #define __rsnd_mod_call_pcm_new 0 #define __rsnd_mod_call_fallback 0 +#define __rsnd_mod_call_hw_params 0 #define rsnd_mod_to_priv(mod) (rsnd_io_to_priv(rsnd_mod_to_io(mod))) #define rsnd_mod_to_dma(mod) (&(mod)->dma) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 83611fa450bf02..a0a2bdac09d9ac 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -22,12 +22,13 @@ struct rsnd_src { struct rsnd_src_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; + u32 convert_rate; /* sampling rate convert */ int err; }; #define RSND_SRC_NAME_SIZE 16 -#define rsnd_src_convert_rate(p) ((p)->info->convert_rate) +#define rsnd_src_convert_rate(s) ((s)->convert_rate) #define rsnd_src_of_node(priv) \ of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,src") @@ -288,7 +289,43 @@ static int rsnd_src_set_convert_rate(struct rsnd_mod *mod) return 0; } -static int rsnd_src_init(struct rsnd_mod *mod) +static int rsnd_src_hw_params(struct rsnd_mod *mod, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *fe_params) +{ + struct rsnd_src *src = rsnd_mod_to_src(mod); + struct snd_soc_pcm_runtime *fe = substream->private_data; + + /* default value (mainly for non-DT) */ + src->convert_rate = src->info->convert_rate; + + /* + * SRC assumes that it is used under DPCM if user want to use + * sampling rate convert. Then, SRC should be FE. + * And then, this function will be called *after* BE settings. + * this means, each BE already has fixuped hw_params. + * see + * dpcm_fe_dai_hw_params() + * dpcm_be_dai_hw_params() + */ + if (fe->dai_link->dynamic) { + int stream = substream->stream; + struct snd_soc_dpcm *dpcm; + struct snd_pcm_hw_params *be_params; + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + be_params = &dpcm->hw_params; + + if (params_rate(fe_params) != params_rate(be_params)) + src->convert_rate = params_rate(be_params); + } + } + + return 0; +} + +static int rsnd_src_init(struct rsnd_mod *mod, + struct rsnd_priv *priv) { struct rsnd_src *src = rsnd_mod_to_src(mod); @@ -317,6 +354,8 @@ static int rsnd_src_quit(struct rsnd_mod *mod, dev_warn(dev, "%s[%d] under/over flow err = %d\n", rsnd_mod_name(mod), rsnd_mod_id(mod), src->err); + src->convert_rate = 0; + return 0; } @@ -465,7 +504,7 @@ static int rsnd_src_init_gen1(struct rsnd_mod *mod, { int ret; - ret = rsnd_src_init(mod); + ret = rsnd_src_init(mod, priv); if (ret < 0) return ret; @@ -511,6 +550,7 @@ static struct rsnd_mod_ops rsnd_src_gen1_ops = { .quit = rsnd_src_quit, .start = rsnd_src_start_gen1, .stop = rsnd_src_stop_gen1, + .hw_params = rsnd_src_hw_params, }; /* @@ -736,7 +776,7 @@ static int rsnd_src_init_gen2(struct rsnd_mod *mod, { int ret; - ret = rsnd_src_init(mod); + ret = rsnd_src_init(mod, priv); if (ret < 0) return ret; @@ -780,6 +820,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = { .quit = rsnd_src_quit, .start = rsnd_src_start_gen2, .stop = rsnd_src_stop_gen2, + .hw_params = rsnd_src_hw_params, }; struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) From 1c6ae56c5d26d22e8ba9ea6d3a0afc8b22b4e207 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Sun, 29 Mar 2015 21:47:16 +0200 Subject: [PATCH 316/411] ASoC: fsi: fix license specification MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit According to the file header only GPL v2 applies to it. Fix the MODULE_LICENSE parameter accordingly. Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index eef7083ec7d9e0..936c02d4e3859c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2119,7 +2119,7 @@ static struct platform_driver fsi_driver = { module_platform_driver(fsi_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); MODULE_AUTHOR("Kuninori Morimoto "); MODULE_ALIAS("platform:fsi-pcm-audio"); From 3a82002c7cbb166b891de515b5463aec41035c75 Mon Sep 17 00:00:00 2001 From: Nicolas Ferre Date: Tue, 31 Mar 2015 15:34:51 +0200 Subject: [PATCH 317/411] MAINTAINERS: change the Atmel audio alsa driver entry I take over the the maintainship of Atmel alsa drivers from Voice. Thanks for your work! Signed-off-by: Nicolas Ferre Acked-by: Bo Shen Signed-off-by: Mark Brown --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index ddc5a8cf9a8ac0..679b2d0cb2d109 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1751,7 +1751,7 @@ S: Supported F: drivers/tty/serial/atmel_serial.c ATMEL Audio ALSA driver -M: Bo Shen +M: Nicolas Ferre L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/atmel From 3b6281cf2893a3c140a37be817b0802c46af292b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 31 Mar 2015 16:48:56 +0200 Subject: [PATCH 318/411] ASoC: fsi: reorder code to make a forward declaration superfluous MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 69 +++++++++++++++++++++++----------------------- 1 file changed, 34 insertions(+), 35 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 936c02d4e3859c..5d26f8c9865033 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1876,7 +1876,40 @@ static void fsi_handler_init(struct fsi_priv *fsi, } } -static const struct of_device_id fsi_of_match[]; +static struct fsi_core fsi1_core = { + .ver = 1, + + /* Interrupt */ + .int_st = INT_ST, + .iemsk = IEMSK, + .imsk = IMSK, +}; + +static struct fsi_core fsi2_core = { + .ver = 2, + + /* Interrupt */ + .int_st = CPU_INT_ST, + .iemsk = CPU_IEMSK, + .imsk = CPU_IMSK, + .a_mclk = A_MST_CTLR, + .b_mclk = B_MST_CTLR, +}; + +static const struct of_device_id fsi_of_match[] = { + { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, + { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, + {}, +}; +MODULE_DEVICE_TABLE(of, fsi_of_match); + +static struct platform_device_id fsi_id_table[] = { + { "sh_fsi", (kernel_ulong_t)&fsi1_core }, + { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, + {}, +}; +MODULE_DEVICE_TABLE(platform, fsi_id_table); + static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; @@ -2072,40 +2105,6 @@ static struct dev_pm_ops fsi_pm_ops = { .resume = fsi_resume, }; -static struct fsi_core fsi1_core = { - .ver = 1, - - /* Interrupt */ - .int_st = INT_ST, - .iemsk = IEMSK, - .imsk = IMSK, -}; - -static struct fsi_core fsi2_core = { - .ver = 2, - - /* Interrupt */ - .int_st = CPU_INT_ST, - .iemsk = CPU_IEMSK, - .imsk = CPU_IMSK, - .a_mclk = A_MST_CTLR, - .b_mclk = B_MST_CTLR, -}; - -static const struct of_device_id fsi_of_match[] = { - { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, - { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, - {}, -}; -MODULE_DEVICE_TABLE(of, fsi_of_match); - -static struct platform_device_id fsi_id_table[] = { - { "sh_fsi", (kernel_ulong_t)&fsi1_core }, - { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, - {}, -}; -MODULE_DEVICE_TABLE(platform, fsi_id_table); - static struct platform_driver fsi_driver = { .driver = { .name = "fsi-pcm-audio", From 9a42ab04aae96d47cd86e065b5127e472fd9eab9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 31 Mar 2015 16:48:57 +0200 Subject: [PATCH 319/411] ASoC: fsi: mark several data structures as const MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A driver's platform_device_id and device data should and can be const. Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5d26f8c9865033..0c2af21b0b8251 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1876,7 +1876,7 @@ static void fsi_handler_init(struct fsi_priv *fsi, } } -static struct fsi_core fsi1_core = { +static const struct fsi_core fsi1_core = { .ver = 1, /* Interrupt */ @@ -1885,7 +1885,7 @@ static struct fsi_core fsi1_core = { .imsk = IMSK, }; -static struct fsi_core fsi2_core = { +static const struct fsi_core fsi2_core = { .ver = 2, /* Interrupt */ @@ -1903,7 +1903,7 @@ static const struct of_device_id fsi_of_match[] = { }; MODULE_DEVICE_TABLE(of, fsi_of_match); -static struct platform_device_id fsi_id_table[] = { +static const struct platform_device_id fsi_id_table[] = { { "sh_fsi", (kernel_ulong_t)&fsi1_core }, { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, {}, From 969b8619069f0e4da767c54481dcc10812f949a5 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Tue, 31 Mar 2015 20:35:09 +0200 Subject: [PATCH 320/411] ASoC: rcar: mark device data as constant MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A driver's device data should and can be const. This is a follow-up on commit 33187fb4a203 (ASoC: rsnd: constify of_device_id array) which marked the of_device_id as const. Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 9b0de428c45b4d..6f60149acdbffb 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -99,11 +99,11 @@ #define RSND_RATES SNDRV_PCM_RATE_8000_96000 #define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -static struct rsnd_of_data rsnd_of_data_gen1 = { +static const struct rsnd_of_data rsnd_of_data_gen1 = { .flags = RSND_GEN1, }; -static struct rsnd_of_data rsnd_of_data_gen2 = { +static const struct rsnd_of_data rsnd_of_data_gen2 = { .flags = RSND_GEN2, }; From 43cb6954f8c8a68fdc354226fa045ff43c7e4d39 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 1 Apr 2015 04:15:16 +0000 Subject: [PATCH 321/411] ASoC: rsnd: add Synchronous SRC mode Renesas R-Car sound SRC (= Sampling Rate Converter) has Asynchronous/Synchronous SRC mode. Asynchronous mode is already supported via DPCM. This patch adds Synchronous mode on it. The condition of enabling Synchronous mode are - SoC is clock master - Sound uses SRC - Sound doesn't use DVC - Sound card uses DPCM (= rsrc-card card) amixer set "SRC Out Rate" on aplay xxx.wav & amixer set "SRC Out Rate" 48000 Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 126 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a0a2bdac09d9ac..3beb32eb412a10 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -22,13 +22,15 @@ struct rsnd_src { struct rsnd_src_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; + struct rsnd_kctrl_cfg_s sen; /* sync convert enable */ + struct rsnd_kctrl_cfg_s sync; /* sync convert */ u32 convert_rate; /* sampling rate convert */ int err; }; #define RSND_SRC_NAME_SIZE 16 -#define rsnd_src_convert_rate(s) ((s)->convert_rate) +#define rsnd_enable_sync_convert(src) ((src)->sen.val) #define rsnd_src_of_node(priv) \ of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,src") @@ -233,6 +235,30 @@ int rsnd_src_ssi_irq_disable(struct rsnd_mod *ssi_mod) return 0; } +static u32 rsnd_src_convert_rate(struct rsnd_src *src) +{ + struct rsnd_mod *mod = &src->mod; + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 convert_rate; + + if (!runtime) + return 0; + + if (!rsnd_enable_sync_convert(src)) + return src->convert_rate; + + convert_rate = src->sync.val; + + if (!convert_rate) + convert_rate = src->convert_rate; + + if (!convert_rate) + convert_rate = runtime->rate; + + return convert_rate; +} + unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime) @@ -333,6 +359,9 @@ static int rsnd_src_init(struct rsnd_mod *mod, src->err = 0; + /* reset sync convert_rate */ + src->sync.val = 0; + /* * Initialize the operation of the SRC internal circuits * see rsnd_src_start() @@ -356,6 +385,9 @@ static int rsnd_src_quit(struct rsnd_mod *mod, src->convert_rate = 0; + /* reset sync convert_rate */ + src->sync.val = 0; + return 0; } @@ -672,6 +704,7 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod) struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_src *src = rsnd_mod_to_src(mod); u32 convert_rate = rsnd_src_convert_rate(src); + u32 cr, route; uint ratio; int ret; @@ -692,13 +725,21 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod) if (ret < 0) return ret; - rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); - + cr = 0x00011110; + route = 0x0; if (convert_rate) { - /* Gen1/Gen2 are not compatible */ - rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); + route = 0x1; + + if (rsnd_enable_sync_convert(src)) { + cr |= 0x1; + route |= rsnd_io_is_play(io) ? + (0x1 << 24) : (0x1 << 25); + } } + rsnd_mod_write(mod, SRC_SRCCR, cr); + rsnd_mod_write(mod, SRC_ROUTE_MODE0, route); + switch (rsnd_mod_id(mod)) { case 5: case 6: @@ -811,6 +852,80 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, return ret; } +static void rsnd_src_reconvert_update(struct rsnd_mod *mod) +{ + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + u32 convert_rate = rsnd_src_convert_rate(src); + u32 fsrate; + + if (!runtime) + return; + + if (!convert_rate) + convert_rate = runtime->rate; + + fsrate = 0x0400000 / convert_rate * runtime->rate; + + /* update IFS */ + rsnd_mod_write(mod, SRC_IFSVR, fsrate); +} + +static int rsnd_src_pcm_new(struct rsnd_mod *mod, + struct snd_soc_pcm_runtime *rtd) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + int ret; + + /* + * enable SRC sync convert if possible + */ + + /* + * Gen1 is not supported + */ + if (rsnd_is_gen1(priv)) + return 0; + + /* + * SRC sync convert needs clock master + */ + if (!rsnd_rdai_is_clk_master(rdai)) + return 0; + + /* + * We can't use SRC sync convert + * if it has DVC + */ + if (rsnd_io_to_mod_dvc(io)) + return 0; + + /* + * enable sync convert + */ + ret = rsnd_kctrl_new_s(mod, rtd, + rsnd_io_is_play(io) ? + "SRC Out Rate Switch" : + "SRC In Rate Switch", + rsnd_src_reconvert_update, + &src->sen, 1); + if (ret < 0) + return ret; + + ret = rsnd_kctrl_new_s(mod, rtd, + rsnd_io_is_play(io) ? + "SRC Out Rate" : + "SRC In Rate", + rsnd_src_reconvert_update, + &src->sync, 192000); + + return ret; +} + static struct rsnd_mod_ops rsnd_src_gen2_ops = { .name = SRC_NAME, .dma_req = rsnd_src_dma_req, @@ -821,6 +936,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = { .start = rsnd_src_start_gen2, .stop = rsnd_src_stop_gen2, .hw_params = rsnd_src_hw_params, + .pcm_new = rsnd_src_pcm_new, }; struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) From 488cb533911b96fe41af68a0206878aa46b799bc Mon Sep 17 00:00:00 2001 From: Alexandre Belloni Date: Mon, 30 Mar 2015 19:54:54 +0200 Subject: [PATCH 322/411] ASoC: atmel-pcm-pdc: merge atmel-pcm back in atmel-pcm.c was split into two files to create a generic framework for both PDC and DMA. atmel-pcm-dma.c is using the generic dmaengine framework since 95e0e07e710e (ASoC: atmel-pcm: use generic dmaengine framework). Merge atmel-pcm.c in atmel-pcm-pdc.c as this is now the only user. Signed-off-by: Alexandre Belloni Acked-by: Nicolas Ferre Signed-off-by: Mark Brown --- sound/soc/atmel/Makefile | 2 - sound/soc/atmel/atmel-pcm-pdc.c | 79 +++++++++++++++++++++ sound/soc/atmel/atmel-pcm.c | 121 -------------------------------- sound/soc/atmel/atmel-pcm.h | 5 -- 4 files changed, 79 insertions(+), 128 deletions(-) delete mode 100644 sound/soc/atmel/atmel-pcm.c diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 466a821da98ca9..b327e5cc8de352 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -1,10 +1,8 @@ # AT91 Platform Support -snd-soc-atmel-pcm-objs := atmel-pcm.o snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o -obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index a366b3503c282c..da861b44413f7f 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -47,6 +47,85 @@ #include "atmel-pcm.h" +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = ATMEL_SSC_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", + (void *)buf->area, (void *)(long)buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + /*--------------------------------------------------------------------------*\ * Hardware definition \*--------------------------------------------------------------------------*/ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c deleted file mode 100644 index 8ae3fa5ac60a5f..00000000000000 --- a/sound/soc/atmel/atmel-pcm.c +++ /dev/null @@ -1,121 +0,0 @@ -/* - * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. - * - * Copyright (C) 2005 SAN People - * Copyright (C) 2008 Atmel - * - * Authors: Sedji Gaouaou - * - * Based on at91-pcm. by: - * Frank Mandarino - * Copyright 2006 Endrelia Technologies Inc. - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -#include -#include -#include -#include -#include "atmel-pcm.h" - -static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = ATMEL_SSC_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", - (void *)buf->area, (void *)(long)buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -int atmel_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} -EXPORT_SYMBOL_GPL(atmel_pcm_mmap); - -int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} -EXPORT_SYMBOL_GPL(atmel_pcm_new); - -void atmel_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} -EXPORT_SYMBOL_GPL(atmel_pcm_free); - diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 12ae814eff214f..6eaf081cad5083 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -83,11 +83,6 @@ struct atmel_pcm_dma_params { #define ssc_readx(base, reg) (__raw_readl((base) + (reg))) #define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) -int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd); -void atmel_pcm_free(struct snd_pcm *pcm); -int atmel_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma); - #if defined(CONFIG_SND_ATMEL_SOC_PDC) || \ defined(CONFIG_SND_ATMEL_SOC_PDC_MODULE) int atmel_pcm_pdc_platform_register(struct device *dev); From ab87ce1d9bb0501fccfc00d5e5ce7c16cd2bcc3e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Mar 2015 21:04:45 +0200 Subject: [PATCH 323/411] ASoC: wm8971: Use system_power_efficient_wq instead of custom workqueue The delayed work used by the wm8971 driver to manage the caps charging doesn't have any special requirements that would justify using a custom workqueue, just use the generic system_power_efficient_wq instead. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b8834ccb..44baacd3325218 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -31,8 +31,6 @@ #define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; - /* codec private data */ struct wm8971_priv { unsigned int sysclk; @@ -636,7 +634,8 @@ static int wm8971_resume(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); } @@ -649,9 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) u16 reg; INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); - wm8971_workq = create_workqueue("wm8971"); - if (wm8971_workq == NULL) - return -ENOMEM; wm8971_reset(codec); @@ -659,7 +655,8 @@ static int wm8971_probe(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -681,8 +678,6 @@ static int wm8971_remove(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8971_workq) - destroy_workqueue(wm8971_workq); return 0; } From 643518403c3fdecd40d9edfd50f4bafcebeda79b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Mar 2015 21:04:46 +0200 Subject: [PATCH 324/411] ASoC: wm8971: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 71 +++++++++++++++++---------------------- 1 file changed, 31 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 44baacd3325218..4ab034d484742c 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -34,6 +34,8 @@ /* codec private data */ struct wm8971_priv { unsigned int sysclk; + struct delayed_work charge_work; + struct regmap *regmap; }; /* @@ -550,9 +552,19 @@ static int wm8971_mute(struct snd_soc_dai *dai, int mute) return 0; } +static void wm8971_charge_work(struct work_struct *work) +{ + struct wm8971_priv *wm8971 = + container_of(work, struct wm8971_priv, charge_work.work); + + /* Set to 500k */ + regmap_update_bits(wm8971->regmap, WM8971_PWR1, 0x0180, 0x0100); +} + static int wm8971_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; switch (level) { @@ -561,15 +573,24 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); break; case SND_SOC_BIAS_PREPARE: + /* Wait until fully charged */ + flush_delayed_work(&wm8971->charge_work); break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { snd_soc_cache_sync(codec); + /* charge output caps - set vmid to 5k for quick power up */ + snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x01c0); + queue_delayed_work(system_power_efficient_wq, + &wm8971->charge_work, msecs_to_jiffies(1000)); + } else { + /* mute dac and set vmid to 500k, enable VREF */ + snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140); + } - /* mute dac and set vmid to 500k, enable VREF */ - snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140); break; case SND_SOC_BIAS_OFF: + cancel_delayed_work_sync(&wm8971->charge_work); snd_soc_write(codec, WM8971_PWR1, 0x0001); break; } @@ -608,15 +629,6 @@ static struct snd_soc_dai_driver wm8971_dai = { .ops = &wm8971_dai_ops, }; -static void wm8971_work(struct work_struct *work) -{ - struct snd_soc_dapm_context *dapm = - container_of(work, struct snd_soc_dapm_context, - delayed_work.work); - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); - wm8971_set_bias_level(codec, codec->dapm.bias_level); -} - static int wm8971_suspend(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -625,39 +637,19 @@ static int wm8971_suspend(struct snd_soc_codec *codec) static int wm8971_resume(struct snd_soc_codec *codec) { - u16 reg; - wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* charge wm8971 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; - snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(system_power_efficient_wq, - &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); - } - return 0; } static int wm8971_probe(struct snd_soc_codec *codec) { - int ret = 0; - u16 reg; + struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); - INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); + INIT_DELAYED_WORK(&wm8971->charge_work, wm8971_charge_work); wm8971_reset(codec); - /* charge output caps - set vmid to 5k for quick power up */ - reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; - snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); - codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - queue_delayed_work(system_power_efficient_wq, - &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set the update bits */ snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); @@ -669,7 +661,7 @@ static int wm8971_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8971_LINVOL, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8971_RINVOL, 0x0100, 0x0100); - return ret; + return 0; } @@ -710,7 +702,6 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8971_priv *wm8971; - struct regmap *regmap; int ret; wm8971 = devm_kzalloc(&i2c->dev, sizeof(struct wm8971_priv), @@ -718,9 +709,9 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, if (wm8971 == NULL) return -ENOMEM; - regmap = devm_regmap_init_i2c(i2c, &wm8971_regmap); - if (IS_ERR(regmap)) - return PTR_ERR(regmap); + wm8971->regmap = devm_regmap_init_i2c(i2c, &wm8971_regmap); + if (IS_ERR(wm8971->regmap)) + return PTR_ERR(wm8971->regmap); i2c_set_clientdata(i2c, wm8971); From c59e6abba9dd7bc273c3dd389ae9927d1da88f35 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Mar 2015 21:04:47 +0200 Subject: [PATCH 325/411] ASoC: wm8971: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 27 +-------------------------- 1 file changed, 1 insertion(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 4ab034d484742c..f9cbabdc623889 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -629,18 +629,6 @@ static struct snd_soc_dai_driver wm8971_dai = { .ops = &wm8971_dai_ops, }; -static int wm8971_suspend(struct snd_soc_codec *codec) -{ - wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8971_resume(struct snd_soc_codec *codec) -{ - wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int wm8971_probe(struct snd_soc_codec *codec) { struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -649,8 +637,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) wm8971_reset(codec); - wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* set the update bits */ snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8971_RDAC, 0x0100, 0x0100); @@ -664,21 +650,10 @@ static int wm8971_probe(struct snd_soc_codec *codec) return 0; } - -/* power down chip */ -static int wm8971_remove(struct snd_soc_codec *codec) -{ - wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8971 = { .probe = wm8971_probe, - .remove = wm8971_remove, - .suspend = wm8971_suspend, - .resume = wm8971_resume, .set_bias_level = wm8971_set_bias_level, + .suspend_bias_off = true, .controls = wm8971_snd_controls, .num_controls = ARRAY_SIZE(wm8971_snd_controls), From 35afd9221b301d1959eadab2d45a2cb94dcb7d30 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Mar 2015 21:04:48 +0200 Subject: [PATCH 326/411] ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 54 +++++++++++++++++---------------------- 1 file changed, 24 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 21ca3a94fc96ea..176fcb1530c343 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -153,6 +153,7 @@ struct wm8753_priv { unsigned int hifi_fmt; int dai_func; + struct delayed_work charge_work; }; #define wm8753_reset(c) snd_soc_write(c, WM8753_RESET, 0) @@ -1326,9 +1327,19 @@ static int wm8753_mute(struct snd_soc_dai *dai, int mute) return 0; } +static void wm8753_charge_work(struct work_struct *work) +{ + struct wm8753_priv *wm8753 = + container_of(work, struct wm8753_priv, charge_work.work); + + /* Set to 500k */ + regmap_update_bits(wm8753->regmap, WM8753_PWR1, 0x0180, 0x0100); +} + static int wm8753_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 pwr_reg = snd_soc_read(codec, WM8753_PWR1) & 0xfe3e; switch (level) { @@ -1337,14 +1348,22 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8753_PWR1, pwr_reg | 0x00c0); break; case SND_SOC_BIAS_PREPARE: - /* set vmid to 5k for quick power up */ - snd_soc_write(codec, WM8753_PWR1, pwr_reg | 0x01c1); + /* Wait until fully charged */ + flush_delayed_work(&wm8753->charge_work); break; case SND_SOC_BIAS_STANDBY: - /* mute dac and set vmid to 500k, enable VREF */ - snd_soc_write(codec, WM8753_PWR1, pwr_reg | 0x0141); + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* set vmid to 5k for quick power up */ + snd_soc_write(codec, WM8753_PWR1, pwr_reg | 0x01c1); + schedule_delayed_work(&wm8753->charge_work, + msecs_to_jiffies(caps_charge)); + } else { + /* mute dac and set vmid to 500k, enable VREF */ + snd_soc_write(codec, WM8753_PWR1, pwr_reg | 0x0141); + } break; case SND_SOC_BIAS_OFF: + cancel_delayed_work_sync(&wm8753->charge_work); snd_soc_write(codec, WM8753_PWR1, 0x0001); break; } @@ -1428,15 +1447,6 @@ static struct snd_soc_dai_driver wm8753_dai[] = { }, }; -static void wm8753_work(struct work_struct *work) -{ - struct snd_soc_dapm_context *dapm = - container_of(work, struct snd_soc_dapm_context, - delayed_work.work); - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); - wm8753_set_bias_level(codec, dapm->bias_level); -} - static int wm8753_suspend(struct snd_soc_codec *codec) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -1450,16 +1460,6 @@ static int wm8753_resume(struct snd_soc_codec *codec) regcache_sync(wm8753->regmap); wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* charge wm8753 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(system_power_efficient_wq, - &codec->dapm.delayed_work, - msecs_to_jiffies(caps_charge)); - } - return 0; } @@ -1468,7 +1468,7 @@ static int wm8753_probe(struct snd_soc_codec *codec) struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); int ret; - INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work); + INIT_DELAYED_WORK(&wm8753->charge_work, wm8753_charge_work); ret = wm8753_reset(codec); if (ret < 0) { @@ -1479,11 +1479,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8753->dai_func = 0; - /* charge output caps */ - wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - schedule_delayed_work(&codec->dapm.delayed_work, - msecs_to_jiffies(caps_charge)); - /* set the update bits */ snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100); @@ -1502,7 +1497,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) /* power down chip */ static int wm8753_remove(struct snd_soc_codec *codec) { - flush_delayed_work(&codec->dapm.delayed_work); wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; From a1f0b9674936bf55d5d49813de01547de2667690 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Mar 2015 21:04:49 +0200 Subject: [PATCH 327/411] ASoC: wm8753: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 19 +------------------ 1 file changed, 1 insertion(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 176fcb1530c343..c50a5959345fcf 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1447,19 +1447,12 @@ static struct snd_soc_dai_driver wm8753_dai[] = { }, }; -static int wm8753_suspend(struct snd_soc_codec *codec) -{ - wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int wm8753_resume(struct snd_soc_codec *codec) { struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); regcache_sync(wm8753->regmap); - wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -1476,7 +1469,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) return ret; } - wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8753->dai_func = 0; /* set the update bits */ @@ -1494,20 +1486,11 @@ static int wm8753_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int wm8753_remove(struct snd_soc_codec *codec) -{ - wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .probe = wm8753_probe, - .remove = wm8753_remove, - .suspend = wm8753_suspend, .resume = wm8753_resume, .set_bias_level = wm8753_set_bias_level, + .suspend_bias_off = true, .controls = wm8753_snd_controls, .num_controls = ARRAY_SIZE(wm8753_snd_controls), From 37660b6daf6d28bb2206c95ec75c8063f2db1606 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Mar 2015 21:04:50 +0200 Subject: [PATCH 328/411] ASoC: Remove suspend_bias_level from DAPM context struct The only two users of the suspend_bias_level field were two rather old drivers which weren't exactly doing things by the book. Those drivers have been updated and field is now unused and can be removed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 - sound/soc/soc-core.c | 10 ++-------- 2 files changed, 2 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8d7416e4686174..9e35203653ff1c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -586,7 +586,6 @@ struct snd_soc_dapm_update { /* DAPM context */ struct snd_soc_dapm_context { enum snd_soc_bias_level bias_level; - enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ /* Go to BIAS_OFF in suspend if the DAPM context is idle */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca5bacb98..72601a86a74839 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -583,15 +583,9 @@ int snd_soc_suspend(struct device *dev) cpu_dai->driver->suspend(cpu_dai); } - /* close any waiting streams and save state */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai **codec_dais = card->rtd[i].codec_dais; + /* close any waiting streams */ + for (i = 0; i < card->num_rtd; i++) flush_delayed_work(&card->rtd[i].delayed_work); - for (j = 0; j < card->rtd[i].num_codecs; j++) { - codec_dais[j]->codec->dapm.suspend_bias_level = - codec_dais[j]->codec->dapm.bias_level; - } - } for (i = 0; i < card->num_rtd; i++) { From cd5d822688f3b32af286a76f7078dfe49c716282 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Mar 2015 21:04:51 +0200 Subject: [PATCH 329/411] ASoC: wm8350: Move delayed work struct from DAPM context to driver state The wm8350 driver is the last driver that still uses the delayed_work field from the snd_soc_dapm_context struct. Moving this over to the driver's private data struct will allow us to remove the field from the DAPM context, which will drastically reduce its size. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 25 +++++++++++-------------- 1 file changed, 11 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index c81a9eab3e3e51..c65e5a75fc1afd 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -69,14 +69,14 @@ struct wm8350_data { struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; int fll_freq_out; int fll_freq_in; + struct delayed_work pga_work; }; /* * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown. */ -static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec) +static inline int wm8350_out1_ramp_step(struct wm8350_data *wm8350_data) { - struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1; struct wm8350 *wm8350 = wm8350_data->wm8350; int left_complete = 0, right_complete = 0; @@ -140,9 +140,8 @@ static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec) /* * Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown. */ -static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec) +static inline int wm8350_out2_ramp_step(struct wm8350_data *wm8350_data) { - struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out2 = &wm8350_data->out2; struct wm8350 *wm8350 = wm8350_data->wm8350; int left_complete = 0, right_complete = 0; @@ -210,10 +209,8 @@ static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec) */ static void wm8350_pga_work(struct work_struct *work) { - struct snd_soc_dapm_context *dapm = - container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); - struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); + struct wm8350_data *wm8350_data = + container_of(work, struct wm8350_data, pga_work.work); struct wm8350_output *out1 = &wm8350_data->out1, *out2 = &wm8350_data->out2; int i, out1_complete, out2_complete; @@ -226,9 +223,9 @@ static void wm8350_pga_work(struct work_struct *work) for (i = 0; i <= 63; i++) { out1_complete = 1, out2_complete = 1; if (out1->ramp != WM8350_RAMP_NONE) - out1_complete = wm8350_out1_ramp_step(codec); + out1_complete = wm8350_out1_ramp_step(wm8350_data); if (out2->ramp != WM8350_RAMP_NONE) - out2_complete = wm8350_out2_ramp_step(codec); + out2_complete = wm8350_out2_ramp_step(wm8350_data); /* ramp finished ? */ if (out1_complete && out2_complete) @@ -283,7 +280,7 @@ static int pga_event(struct snd_soc_dapm_widget *w, out->ramp = WM8350_RAMP_UP; out->active = 1; - schedule_delayed_work(&codec->dapm.delayed_work, + schedule_delayed_work(&wm8350_data->pga_work, msecs_to_jiffies(1)); break; @@ -291,7 +288,7 @@ static int pga_event(struct snd_soc_dapm_widget *w, out->ramp = WM8350_RAMP_DOWN; out->active = 0; - schedule_delayed_work(&codec->dapm.delayed_work, + schedule_delayed_work(&wm8350_data->pga_work, msecs_to_jiffies(1)); break; } @@ -1492,7 +1489,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); - INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8350_pga_work); + INIT_DELAYED_WORK(&priv->pga_work, wm8350_pga_work); INIT_DELAYED_WORK(&priv->hpl.work, wm8350_hpl_work); INIT_DELAYED_WORK(&priv->hpr.work, wm8350_hpr_work); @@ -1578,7 +1575,7 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) /* if there was any work waiting then we run it now and * wait for its completion */ - flush_delayed_work(&codec->dapm.delayed_work); + flush_delayed_work(&priv->pga_work); wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); From 7c0e3facf39a68f30c2eae69e969918eca1ff9d6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 30 Mar 2015 21:04:52 +0200 Subject: [PATCH 330/411] ASoC: dapm: Remove delayed_work from dapm context struct The delayed_work field in the snd_soc_dapm_context struct is now unused and can be removed. Removing it reduces the size of the snd_soc_dapm_context struct by ~50% from 100 bytes to 48 bytes. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9e35203653ff1c..485fc9d1a7bc6c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -586,7 +586,6 @@ struct snd_soc_dapm_update { /* DAPM context */ struct snd_soc_dapm_context { enum snd_soc_bias_level bias_level; - struct delayed_work delayed_work; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ /* Go to BIAS_OFF in suspend if the DAPM context is idle */ unsigned int suspend_bias_off:1; From 7b425f264fc2572bea8b15fe97948659507c6a45 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 1 Apr 2015 20:35:57 +0300 Subject: [PATCH 331/411] ASoC: omap-hdmi-audio: No not use IEC958_AES1_PRO_MODE_NOTID No IEC958_AES?_PRO_* macros should be used in HDMI consumer audio mode and IEC958_AES1_PRO_MODE_NOTID should be applied to byte 1 when applicable. However IEC958_AES1_PRO_MODE_NOTID is defined as 0 so this fix does not affect the functionality in any way. Reported-by: Russell King Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/omap/omap-hdmi-audio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index ccfb41c22e53b1..1822578bbc2ca8 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -142,8 +142,6 @@ static int hdmi_dai_hw_params(struct snd_pcm_substream *substream, iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE; - iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID; - iec->status[1] = IEC958_AES1_CON_GENERAL; iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC; From 7b3d165a282145e605247148d3dec034814e0a78 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 27 Mar 2015 11:47:51 +0200 Subject: [PATCH 332/411] ASoC: davinci-mcasp: Index ruledata in drvdata with substream->stream The serializer direction definitions runs from 1 to 2, which does not suite the purpose. The substream->stream is perfect for the purpose and should have been used from the beginning. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 76156d18ed4640..0b6b1b286201d6 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1149,7 +1149,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } - mcasp->ruledata[dir].serializers = max_channels; + mcasp->ruledata[substream->stream].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1172,7 +1172,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { int ret; - mcasp->ruledata[dir].mcasp = mcasp; + mcasp->ruledata[substream->stream].mcasp = mcasp; ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, From c14e2591bf54c45c9f80cf728fb90976c4e10384 Mon Sep 17 00:00:00 2001 From: Alexandre Belloni Date: Mon, 30 Mar 2015 21:40:37 +0200 Subject: [PATCH 333/411] ASoC: atmel-pcm-dma: increase buffer_bytes_max atmel-pcm-dma is not limited to a buffer size of 64kB like atmel-pcm-pdc. Increase buffer_bytes_max to 512kB to allow for higher bit rates (i.e. 32bps at 192kHz) to work correctly. By default, keep the prealloc at 64kB. Signed-off-by: Alexandre Belloni Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm-dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index b8e7bad05eb14a..b6625c8c411b8a 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -54,7 +54,7 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { .period_bytes_max = 2 * 0xffff, /* if 2 bytes format */ .periods_min = 8, .periods_max = 1024, /* no limit */ - .buffer_bytes_max = ATMEL_SSC_DMABUF_SIZE, + .buffer_bytes_max = 512 * 1024, }; /** @@ -119,7 +119,7 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = { .prepare_slave_config = atmel_pcm_configure_dma, .pcm_hardware = &atmel_pcm_dma_hardware, - .prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE, + .prealloc_buffer_size = 64 * 1024, }; int atmel_pcm_dma_platform_register(struct device *dev) From 74ff960222d90999508b4ba0d3449f796695b6d5 Mon Sep 17 00:00:00 2001 From: Pascal Huerst Date: Thu, 2 Apr 2015 10:17:40 +0200 Subject: [PATCH 334/411] ASoC: cs4271: Increase delay time after reset The delay time after a reset in the codec probe callback was too short, and did not work on certain hw because the codec needs more time to power on. This increases the delay time from 1us to 1ms. Signed-off-by: Pascal Huerst Acked-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 79a4efcb894c19..55ca6ecf9f4d08 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -561,10 +561,10 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ gpio_direction_output(cs4271->gpio_nreset, 0); - udelay(1); + mdelay(1); gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ - udelay(1); + mdelay(1); } ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, From ffda568e8b4979c6a04bbdd92acfd93b5dc5e163 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 1 Apr 2015 12:43:00 +0200 Subject: [PATCH 335/411] ALSA: hda - Fix subsystem ID read regression A regression was introduced in 7639a06c23c7d4cda3: if AC_PAR_SUBSYSTEM_ID reads as zero, one should retry using AC_VERB_GET_SUBSYSTEM_ID. This seems to hit many codecs (my own laptop included), and causes quirks for some machines not to apply correctly. Reported-by: TienFu Chen Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/hda/hdac_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 6e8ee1d6974a23..53b6b95ff8cda9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -91,7 +91,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->power_caps = snd_hdac_read_parm(codec, fg, AC_PAR_POWER_STATE); /* reread ssid if not set by parameter */ - if (codec->subsystem_id == -1) + if (codec->subsystem_id == -1 || codec->subsystem_id == 0) snd_hdac_read(codec, fg, AC_VERB_GET_SUBSYSTEM_ID, 0, &codec->subsystem_id); From d545a57c5f84c01b50187177921e232a450858c7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 4 Apr 2015 12:17:28 +0200 Subject: [PATCH 336/411] ALSA: hda - Sync node attributes at resume from widget power saving So far we assumed that the node attributes like amp values remain during the power state transition of the node itself. While this is true for IDT/STAC codecs I've tested, but some other codecs don't seem behaving in that way. This patch implements a partial sync mechanism specific to the given widget node. Now we've merged the regmap support, and it can be easily written with regcache_sync_region(). Tested-by: Hui Wang Signed-off-by: Takashi Iwai --- include/sound/hda_regmap.h | 12 ++++++++++++ sound/pci/hda/hda_generic.c | 6 ++---- 2 files changed, 14 insertions(+), 4 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index 76648ccfbbf898..53a18b3635e24a 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -202,4 +202,16 @@ snd_hdac_regmap_update_amp_stereo(struct hdac_device *codec, hda_nid_t nid, return snd_hdac_regmap_update_raw(codec, cmd, mask, val); } +/** + * snd_hdac_regmap_sync_node - sync the widget node attributes + * @codec: HD-audio codec + * @nid: NID to sync + */ +static inline void +snd_hdac_regmap_sync_node(struct hdac_device *codec, hda_nid_t nid) +{ + regcache_mark_dirty(codec->regmap); + regcache_sync_region(codec->regmap, nid << 20, ((nid + 1) << 20) - 1); +} + #endif /* __SOUND_HDA_REGMAP_H */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f7ccef5559deac..1f2ca7be146860 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -842,10 +842,8 @@ static hda_nid_t path_power_update(struct hda_codec *codec, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, state); changed = nid; - /* here we assume that widget attributes (e.g. amp, - * pinctl connection) don't change with local power - * state change. If not, need to sync the cache. - */ + if (state == AC_PWRST_D0) + snd_hdac_regmap_sync_node(&codec->core, nid); } } return changed; From 382fd7becc409be9cc18dea8e3d53f6d184f9a5c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 4 Apr 2015 12:24:00 +0200 Subject: [PATCH 337/411] ALSA: hda - Enable widget power saving for Realtek codecs Recent Realtek codecs support the finer power state control on each widget. Let's enable the new feature. Tested-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d44cb7e37094f2..e0c06f9a0e8030 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5413,6 +5413,7 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; + codec->power_save_node = 1; snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); From 751e2216899cb143fe1d5909fe762870faa945f6 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Thu, 2 Apr 2015 16:51:43 +0530 Subject: [PATCH 338/411] ALSA: hda: fix possible null dereference we are dereferencing pcm first then checking pcm. instead now lets put them in same if condition so that pcm is checked first. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 41851f9b48c113..16dfa1ed10dd1d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3230,9 +3230,8 @@ static int add_std_chmaps(struct hda_codec *codec) struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (pcm->own_chmap) - continue; - if (!pcm || !hinfo->substreams) + if (!pcm || pcm->own_chmap || + !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, From eab0fbfa41040f4f76b173cad17c0c8ed40cba33 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sat, 4 Apr 2015 13:38:25 -0700 Subject: [PATCH 339/411] ALSA: Use const struct ac97_quirk Use const to reduce data by ~3Kb. Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 4 +++- sound/pci/ac97/ac97_codec.c | 3 ++- sound/pci/ad1889.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/via82xx.c | 2 +- 7 files changed, 10 insertions(+), 7 deletions(-) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index d315a08d6c6d4e..0e9d75b49bede1 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -608,7 +608,9 @@ struct ac97_quirk { int type; /* quirk type above */ }; -int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override); +int snd_ac97_tune_hardware(struct snd_ac97 *ac97, + const struct ac97_quirk *quirk, + const char *override); int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate); /* diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 5bca1a33fed69e..82259ca61e6442 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2902,7 +2902,8 @@ static int apply_quirk_str(struct snd_ac97 *ac97, const char *typestr) * Return: Zero if successful, or a negative error code on failure. */ -int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override) +int snd_ac97_tune_hardware(struct snd_ac97 *ac97, + const struct ac97_quirk *quirk, const char *override) { int result; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 850a8c984c2500..66ddd981d1d5d4 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -747,7 +747,7 @@ snd_ad1889_proc_init(struct snd_ad1889 *chip) snd_info_set_text_ops(entry, chip, snd_ad1889_proc_read); } -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x11d4, /* AD */ .subdevice = 0x1889, /* AD1889 */ diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d5f15c9bbeda32..42a20c806b3926 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1390,7 +1390,7 @@ static irqreturn_t snd_atiixp_interrupt(int irq, void *dev_id) * ac97 mixer section */ -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x103c, .subdevice = 0x006b, diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 802c33f1cc5957..963b912550d477 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -43,7 +43,7 @@ static char *ac97_quirk; module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 board specific workarounds."); -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { #if 0 /* Not yet confirmed if all 5536 boards are HP only */ { .subvendor = PCI_VENDOR_ID_AMD, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 2c5484eeb96386..749069aa6997be 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1795,7 +1795,7 @@ static struct ac97_pcm ac97_pcm_defs[] = { }, }; -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x0e11, .subdevice = 0x000e, diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8622283e89f3e0..3dd038bdb20421 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1812,7 +1812,7 @@ static void snd_via82xx_mixer_free_ac97(struct snd_ac97 *ac97) chip->ac97 = NULL; } -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x1106, .subdevice = 0x4161, From 646cb6dae4994eb332ddf7376520eed255c5a8b4 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 5 Apr 2015 14:06:33 +0200 Subject: [PATCH 340/411] ALSA: au1x00: fix error return code Return a negative error code on failure. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ identifier ret; expression e1,e2; @@ ( if (\(ret < 0\|ret != 0\)) { ... return ret; } | ret = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/mips/au1x00.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index fbcaa5434fd83d..1e30e8475431ed 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -633,19 +633,25 @@ static int au1000_ac97_probe(struct platform_device *pdev) au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); - if (!au1000->stream[PLAYBACK]) + if (!au1000->stream[PLAYBACK]) { + err = -ENOMEM; goto out; + } au1000->stream[PLAYBACK]->dma = -1; au1000->stream[CAPTURE] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); - if (!au1000->stream[CAPTURE]) + if (!au1000->stream[CAPTURE]) { + err = -ENOMEM; goto out; + } au1000->stream[CAPTURE]->dma = -1; r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) + if (!r) { + err = -ENODEV; goto out; + } err = -EBUSY; au1000->ac97_res_port = request_mem_region(r->start, resource_size(r), From f32c1c1b46c2bd5fda77b49e28b9f9329cd830d1 Mon Sep 17 00:00:00 2001 From: Michael Gernoth Date: Sat, 28 Mar 2015 19:20:35 +0100 Subject: [PATCH 341/411] ALSA: emu10k1: add Audigy 5/Rx The Audigy 5/Rx is essentially an Audigy 4 behind a PLX PCIe- bridge with an additional TOSLINK output. Signed-off-by: Michael Gernoth Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index b4458a630a7c26..4887299011efcc 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1325,6 +1325,22 @@ static int snd_emu10k1_dev_free(struct snd_device *device) } static struct snd_emu_chip_details emu_chip_details[] = { + /* Audigy 5/Rx SB1550 */ + /* Tested by michael@gernoth.net 28 Mar 2015 */ + /* DSP: CA10300-IAT LF + * DAC: Cirrus Logic CS4382-KQZ + * ADC: Philips 1361T + * AC97: Sigmatel STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10241102, + .driver = "Audigy2", .name = "SB Audigy 5/Rx [SB1550]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit */ + .ac97_chip = 1}, /* Audigy4 (Not PRO) SB0610 */ /* Tested by James@superbug.co.uk 4th April 2006 */ /* A_IOCFG bits From 158bf4ed7f686f92ea6c046c5d9d04afad92eb17 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 3 Apr 2015 18:28:32 +0100 Subject: [PATCH 342/411] ASoC: wm5102: Remove set of volume update bits for output 3R The earpiece on wm5102 is mono, thus there is no output 3R. Don't toggle the volume update bits for this output, although worth noting that doing so had no negative effects it is just redundant. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6d0fe0ac95a3f6..0c6d1bc0526eff 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1861,7 +1861,6 @@ static unsigned int wm5102_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_DAC_DIGITAL_VOLUME_3L, - ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_5L, From 2106241a680397f6f49da796a4ce11eb5cf2698e Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 2 Apr 2015 15:37:00 +0800 Subject: [PATCH 343/411] ASoC: Intel: create common folder and move common files in Restructure the sound/soc/intel/ directory: create common folder, and move sst common files here. Signed-off-by: Jie Yang Acked-by: Jarkko Nikula Tested-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 6 +----- sound/soc/intel/common/Makefile | 6 ++++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/sst/sst.c | 2 +- sound/soc/intel/sst/sst_acpi.c | 2 +- sound/soc/intel/sst/sst_drv_interface.c | 2 +- sound/soc/intel/sst/sst_ipc.c | 2 +- sound/soc/intel/sst/sst_loader.c | 2 +- sound/soc/intel/sst/sst_pvt.c | 2 +- sound/soc/intel/sst/sst_stream.c | 2 +- 14 files changed, 14 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index a8e53c45c6b655..28de8cd6f321ee 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,6 +1,5 @@ # Core support -snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o -snd-soc-sst-acpi-objs := sst-acpi.o +obj-$(CONFIG_SND_SOC_INTEL_SST) += common/ snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \ sst-mfld-platform-compress.o sst-atom-controls.o @@ -9,9 +8,6 @@ snd-soc-mfld-machine-objs := mfld_machine.o obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o -obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o -obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o - # Platform Support snd-soc-sst-haswell-pcm-objs := \ sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile new file mode 100644 index 00000000000000..3df0e1ca76c042 --- /dev/null +++ b/sound/soc/intel/common/Makefile @@ -0,0 +1,6 @@ +snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o +snd-soc-sst-acpi-objs := sst-acpi.o + +obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o +obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o + diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c similarity index 100% rename from sound/soc/intel/sst-acpi.c rename to sound/soc/intel/common/sst-acpi.c diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h similarity index 100% rename from sound/soc/intel/sst-dsp-priv.h rename to sound/soc/intel/common/sst-dsp-priv.h diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c similarity index 100% rename from sound/soc/intel/sst-dsp.c rename to sound/soc/intel/common/sst-dsp.c diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h similarity index 100% rename from sound/soc/intel/sst-dsp.h rename to sound/soc/intel/common/sst-dsp.h diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c similarity index 100% rename from sound/soc/intel/sst-firmware.c rename to sound/soc/intel/common/sst-firmware.c diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 1a7eeec444b14b..26b1e31c500345 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -32,7 +32,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../sst-dsp.h" +#include "../common/sst-dsp.h" MODULE_AUTHOR("Vinod Koul "); MODULE_AUTHOR("Harsha Priya "); diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index b782dfdcdbba10..2a19cbcac811c8 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -39,7 +39,7 @@ #include #include #include "../sst-mfld-platform.h" -#include "../sst-dsp.h" +#include "../common/sst-dsp.h" #include "sst.h" struct sst_machines { diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index f0e4b99b3aeb13..36d68b8dfd288c 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -32,7 +32,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../sst-dsp.h" +#include "../common/sst-dsp.h" diff --git a/sound/soc/intel/sst/sst_ipc.c b/sound/soc/intel/sst/sst_ipc.c index 484e60978477bd..3943ae856c3195 100644 --- a/sound/soc/intel/sst/sst_ipc.c +++ b/sound/soc/intel/sst/sst_ipc.c @@ -32,7 +32,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../sst-dsp.h" +#include "../common/sst-dsp.h" struct sst_block *sst_create_block(struct intel_sst_drv *ctx, u32 msg_id, u32 drv_id) diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index e88907ae8b154f..6622e66e17962e 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -37,7 +37,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../sst-dsp.h" +#include "../common/sst-dsp.h" void memcpy32_toio(void __iomem *dst, const void *src, int count) { diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/sst/sst_pvt.c index 4b7720864492ad..2bb0e9e0677d82 100644 --- a/sound/soc/intel/sst/sst_pvt.c +++ b/sound/soc/intel/sst/sst_pvt.c @@ -34,7 +34,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../sst-dsp.h" +#include "../common/sst-dsp.h" int sst_shim_write(void __iomem *addr, int offset, int value) { diff --git a/sound/soc/intel/sst/sst_stream.c b/sound/soc/intel/sst/sst_stream.c index dae2a41997aaa3..7638fca02de078 100644 --- a/sound/soc/intel/sst/sst_stream.c +++ b/sound/soc/intel/sst/sst_stream.c @@ -31,7 +31,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../sst-dsp.h" +#include "../common/sst-dsp.h" int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) { From ba57f68235cf6e9105bf649b01cf9eafc321ea7b Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 2 Apr 2015 15:37:01 +0800 Subject: [PATCH 344/411] ASoC: Intel: create haswell folder and move haswell platform files in Restructure the sound/soc/intel/ directory: create haswell folder, and move haswell platform files here. Signed-off-by: Jie Yang Reviewed-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 4 +--- sound/soc/intel/haswell/Makefile | 4 ++++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 +++--- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 +++--- 6 files changed, 13 insertions(+), 11 deletions(-) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 28de8cd6f321ee..eb3efce4ec24b9 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -9,12 +9,10 @@ obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o # Platform Support -snd-soc-sst-haswell-pcm-objs := \ - sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o +obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/ snd-soc-sst-baytrail-pcm-objs := \ sst-baytrail-ipc.o sst-baytrail-pcm.o sst-baytrail-dsp.o -obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += snd-soc-sst-haswell-pcm.o obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o # Machine support diff --git a/sound/soc/intel/haswell/Makefile b/sound/soc/intel/haswell/Makefile new file mode 100644 index 00000000000000..9c1723112d226c --- /dev/null +++ b/sound/soc/intel/haswell/Makefile @@ -0,0 +1,4 @@ +snd-soc-sst-haswell-pcm-objs := \ + sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o + +obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += snd-soc-sst-haswell-pcm.o diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c similarity index 99% rename from sound/soc/intel/sst-haswell-dsp.c rename to sound/soc/intel/haswell/sst-haswell-dsp.c index b3e957d4693365..7f94920c8a4d85 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -28,9 +28,9 @@ #include #include -#include "sst-dsp.h" -#include "sst-dsp-priv.h" -#include "sst-haswell-ipc.h" +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "../haswell/sst-haswell-ipc.h" #include diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c similarity index 99% rename from sound/soc/intel/sst-haswell-ipc.c rename to sound/soc/intel/haswell/sst-haswell-ipc.c index 20b629a011de60..28667d8e200506 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -34,8 +34,8 @@ #include #include "sst-haswell-ipc.h" -#include "sst-dsp.h" -#include "sst-dsp-priv.h" +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" /* Global Message - Generic */ #define IPC_GLB_TYPE_SHIFT 24 diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/haswell/sst-haswell-ipc.h similarity index 100% rename from sound/soc/intel/sst-haswell-ipc.h rename to sound/soc/intel/haswell/sst-haswell-ipc.h diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c similarity index 99% rename from sound/soc/intel/sst-haswell-pcm.c rename to sound/soc/intel/haswell/sst-haswell-pcm.c index 31ffc0f0498f8e..157b3a6c509ecc 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -29,9 +29,9 @@ #include #include -#include "sst-haswell-ipc.h" -#include "sst-dsp-priv.h" -#include "sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" +#include "../common/sst-dsp-priv.h" +#include "../common/sst-dsp.h" #define HSW_PCM_COUNT 6 #define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */ From e56c72d5f201044b14191c5b83a25e17f2d68ccf Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 2 Apr 2015 15:37:02 +0800 Subject: [PATCH 345/411] ASoC: Intel: create boards folder and move sst boards files in Restructure the sound/soc/intel/ directory: create boards folder, and move sst boards files here. Signed-off-by: Jie Yang Acked-by: Vinod Koul Acked-by: Jarkko Nikula Tested-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 16 +--------------- sound/soc/intel/boards/Makefile | 15 +++++++++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 +++--- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 ++-- .../bytcr_rt5640.c} | 4 ++-- sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 4 ++-- sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 4 ++-- sound/soc/intel/{ => boards}/haswell.c | 6 +++--- sound/soc/intel/{ => boards}/mfld_machine.c | 0 10 files changed, 31 insertions(+), 30 deletions(-) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (98%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (99%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (99%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index eb3efce4ec24b9..ac0248f100ff0f 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -16,21 +16,7 @@ snd-soc-sst-baytrail-pcm-objs := \ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o # Machine support -snd-soc-sst-haswell-objs := haswell.o -snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o -snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o -snd-soc-sst-broadwell-objs := broadwell.o -snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o -snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o -snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o - -obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o -obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o -obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o -obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o -obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o -obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o -obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o +obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile new file mode 100644 index 00000000000000..f8237f0044eb0c --- /dev/null +++ b/sound/soc/intel/boards/Makefile @@ -0,0 +1,15 @@ +snd-soc-sst-haswell-objs := haswell.o +snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o +snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o +snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o + +obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/boards/broadwell.c similarity index 98% rename from sound/soc/intel/broadwell.c rename to sound/soc/intel/boards/broadwell.c index 6c75b6bd004928..8bafaf6ceab126 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -22,10 +22,10 @@ #include #include -#include "sst-dsp.h" -#include "sst-haswell-ipc.h" +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" -#include "../codecs/rt286.h" +#include "../../codecs/rt286.h" static struct snd_soc_jack broadwell_headset; /* Headset jack detection DAPM pins */ diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c similarity index 99% rename from sound/soc/intel/byt-max98090.c rename to sound/soc/intel/boards/byt-max98090.c index d8b1f038da1c41..7ab8cc9fbfd53b 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -24,7 +24,7 @@ #include #include #include -#include "../codecs/max98090.h" +#include "../../codecs/max98090.h" struct byt_max98090_private { struct snd_soc_jack jack; diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c similarity index 98% rename from sound/soc/intel/byt-rt5640.c rename to sound/soc/intel/boards/byt-rt5640.c index 354eaad886e1c0..ae89b9b966d9ff 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -23,9 +23,9 @@ #include #include #include -#include "../codecs/rt5640.h" +#include "../../codecs/rt5640.h" -#include "sst-dsp.h" +#include "../common/sst-dsp.h" static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c similarity index 98% rename from sound/soc/intel/bytcr_dpcm_rt5640.c rename to sound/soc/intel/boards/bytcr_rt5640.c index 3b262d01c1b370..5c2d8fabb5ede4 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -26,8 +26,8 @@ #include #include #include -#include "../codecs/rt5640.h" -#include "sst-atom-controls.h" +#include "../../codecs/rt5640.h" +#include "../sst-atom-controls.h" static const struct snd_soc_dapm_widget byt_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c similarity index 99% rename from sound/soc/intel/cht_bsw_rt5645.c rename to sound/soc/intel/boards/cht_bsw_rt5645.c index 012227997ed9ca..93bb6711ba3d36 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -27,8 +27,8 @@ #include #include #include -#include "../codecs/rt5645.h" -#include "sst-atom-controls.h" +#include "../../codecs/rt5645.h" +#include "../sst-atom-controls.h" #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "rt5645-aif1" diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c similarity index 99% rename from sound/soc/intel/cht_bsw_rt5672.c rename to sound/soc/intel/boards/cht_bsw_rt5672.c index 4204fc4f1badfd..2cea002a31bb91 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -23,8 +23,8 @@ #include #include #include -#include "../codecs/rt5670.h" -#include "sst-atom-controls.h" +#include "../../codecs/rt5670.h" +#include "../sst-atom-controls.h" /* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */ #define CHT_PLAT_CLK_3_HZ 19200000 diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/boards/haswell.c similarity index 98% rename from sound/soc/intel/haswell.c rename to sound/soc/intel/boards/haswell.c index 00fddd3f5dfb8f..22558572cb9ca1 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -21,10 +21,10 @@ #include #include -#include "sst-dsp.h" -#include "sst-haswell-ipc.h" +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" -#include "../codecs/rt5640.h" +#include "../../codecs/rt5640.h" /* Haswell ULT platforms have a Headphone and Mic jack */ static const struct snd_soc_dapm_widget haswell_widgets[] = { diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c similarity index 100% rename from sound/soc/intel/mfld_machine.c rename to sound/soc/intel/boards/mfld_machine.c From 66a6fd9846f0aecdbab9324b792b319fd8e95e77 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 2 Apr 2015 15:37:03 +0800 Subject: [PATCH 346/411] ASoC: Intel: create baytrail folder and move baytrail platform files in Restructure the sound/soc/intel/ directory: create baytrail folder, and move sst baytrail platform files here. Signed-off-by: Jie Yang Acked-by: Jarkko Nikula Tested-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 5 +---- sound/soc/intel/baytrail/Makefile | 4 ++++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 ++-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 ++-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 ++-- 6 files changed, 11 insertions(+), 10 deletions(-) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index ac0248f100ff0f..62de82af670367 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -10,10 +10,7 @@ obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o # Platform Support obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/ -snd-soc-sst-baytrail-pcm-objs := \ - sst-baytrail-ipc.o sst-baytrail-pcm.o sst-baytrail-dsp.o - -obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o +obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ # Machine support obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ diff --git a/sound/soc/intel/baytrail/Makefile b/sound/soc/intel/baytrail/Makefile new file mode 100644 index 00000000000000..488408cadf6d56 --- /dev/null +++ b/sound/soc/intel/baytrail/Makefile @@ -0,0 +1,4 @@ +snd-soc-sst-baytrail-pcm-objs := \ + sst-baytrail-ipc.o sst-baytrail-pcm.o sst-baytrail-dsp.o + +obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/baytrail/sst-baytrail-dsp.c similarity index 99% rename from sound/soc/intel/sst-baytrail-dsp.c rename to sound/soc/intel/baytrail/sst-baytrail-dsp.c index 5a9e56700f315e..01d023cc05dd0a 100644 --- a/sound/soc/intel/sst-baytrail-dsp.c +++ b/sound/soc/intel/baytrail/sst-baytrail-dsp.c @@ -22,8 +22,8 @@ #include #include -#include "sst-dsp.h" -#include "sst-dsp-priv.h" +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" #include "sst-baytrail-ipc.h" #define SST_BYT_FW_SIGNATURE_SIZE 4 diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c similarity index 99% rename from sound/soc/intel/sst-baytrail-ipc.c rename to sound/soc/intel/baytrail/sst-baytrail-ipc.c index b4ad98c43e5c6d..aabb9b0f48b8c4 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -29,8 +29,8 @@ #include #include "sst-baytrail-ipc.h" -#include "sst-dsp.h" -#include "sst-dsp-priv.h" +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" /* IPC message timeout */ #define IPC_TIMEOUT_MSECS 300 diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/baytrail/sst-baytrail-ipc.h similarity index 100% rename from sound/soc/intel/sst-baytrail-ipc.h rename to sound/soc/intel/baytrail/sst-baytrail-ipc.h diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c similarity index 99% rename from sound/soc/intel/sst-baytrail-pcm.c rename to sound/soc/intel/baytrail/sst-baytrail-pcm.c index 224c49c9f13542..79547bec558bc2 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -20,8 +20,8 @@ #include #include #include "sst-baytrail-ipc.h" -#include "sst-dsp-priv.h" -#include "sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "../common/sst-dsp.h" #define BYT_PCM_COUNT 2 From b97169da06992ef04081e66ed22bbdb23dbf6610 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 2 Apr 2015 15:37:04 +0800 Subject: [PATCH 347/411] ASoC: Intel: create atom folder and move atom platform files in Restructure the sound/soc/intel/ directory: create atom folder, and move sst atom platform files here. Signed-off-by: Jie Yang Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 11 +---------- sound/soc/intel/atom/Makefile | 7 +++++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../soc/intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 2 +- sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_ipc.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_loader.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_stream.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- 21 files changed, 18 insertions(+), 20 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (99%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (99%) diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 62de82af670367..cd9aee9871a36a 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,19 +1,10 @@ # Core support obj-$(CONFIG_SND_SOC_INTEL_SST) += common/ -snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \ - sst-mfld-platform-compress.o sst-atom-controls.o -snd-soc-mfld-machine-objs := mfld_machine.o - -obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o -obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o - # Platform Support obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ +obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += atom/ # Machine support obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ - -# DSP driver -obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile new file mode 100644 index 00000000000000..ce8074fa6d667c --- /dev/null +++ b/sound/soc/intel/atom/Makefile @@ -0,0 +1,7 @@ +snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \ + sst-mfld-platform-compress.o sst-atom-controls.o + +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o + +# DSP driver +obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c similarity index 100% rename from sound/soc/intel/sst-atom-controls.c rename to sound/soc/intel/atom/sst-atom-controls.c diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h similarity index 100% rename from sound/soc/intel/sst-atom-controls.h rename to sound/soc/intel/atom/sst-atom-controls.h diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/atom/sst-mfld-dsp.h similarity index 100% rename from sound/soc/intel/sst-mfld-dsp.h rename to sound/soc/intel/atom/sst-mfld-dsp.h diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/atom/sst-mfld-platform-compress.c similarity index 100% rename from sound/soc/intel/sst-mfld-platform-compress.c rename to sound/soc/intel/atom/sst-mfld-platform-compress.c diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c similarity index 100% rename from sound/soc/intel/sst-mfld-platform-pcm.c rename to sound/soc/intel/atom/sst-mfld-platform-pcm.c diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/atom/sst-mfld-platform.h similarity index 100% rename from sound/soc/intel/sst-mfld-platform.h rename to sound/soc/intel/atom/sst-mfld-platform.h diff --git a/sound/soc/intel/sst/Makefile b/sound/soc/intel/atom/sst/Makefile similarity index 100% rename from sound/soc/intel/sst/Makefile rename to sound/soc/intel/atom/sst/Makefile diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/atom/sst/sst.c similarity index 99% rename from sound/soc/intel/sst/sst.c rename to sound/soc/intel/atom/sst/sst.c index 26b1e31c500345..96c2e420cce61f 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -32,7 +32,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../common/sst-dsp.h" +#include "../../common/sst-dsp.h" MODULE_AUTHOR("Vinod Koul "); MODULE_AUTHOR("Harsha Priya "); diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/atom/sst/sst.h similarity index 100% rename from sound/soc/intel/sst/sst.h rename to sound/soc/intel/atom/sst/sst.h diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c similarity index 99% rename from sound/soc/intel/sst/sst_acpi.c rename to sound/soc/intel/atom/sst/sst_acpi.c index 2a19cbcac811c8..678f36ed97a5e9 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -39,7 +39,7 @@ #include #include #include "../sst-mfld-platform.h" -#include "../common/sst-dsp.h" +#include "../../common/sst-dsp.h" #include "sst.h" struct sst_machines { diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c similarity index 99% rename from sound/soc/intel/sst/sst_drv_interface.c rename to sound/soc/intel/atom/sst/sst_drv_interface.c index 36d68b8dfd288c..718838b3fc24aa 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -32,7 +32,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../common/sst-dsp.h" +#include "../../common/sst-dsp.h" diff --git a/sound/soc/intel/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c similarity index 99% rename from sound/soc/intel/sst/sst_ipc.c rename to sound/soc/intel/atom/sst/sst_ipc.c index 3943ae856c3195..5a278618466c5a 100644 --- a/sound/soc/intel/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -32,7 +32,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../common/sst-dsp.h" +#include "../../common/sst-dsp.h" struct sst_block *sst_create_block(struct intel_sst_drv *ctx, u32 msg_id, u32 drv_id) diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c similarity index 99% rename from sound/soc/intel/sst/sst_loader.c rename to sound/soc/intel/atom/sst/sst_loader.c index 6622e66e17962e..33917146d9c445 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -37,7 +37,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../common/sst-dsp.h" +#include "../../common/sst-dsp.h" void memcpy32_toio(void __iomem *dst, const void *src, int count) { diff --git a/sound/soc/intel/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c similarity index 100% rename from sound/soc/intel/sst/sst_pci.c rename to sound/soc/intel/atom/sst/sst_pci.c diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c similarity index 99% rename from sound/soc/intel/sst/sst_pvt.c rename to sound/soc/intel/atom/sst/sst_pvt.c index 2bb0e9e0677d82..3c178444638bd9 100644 --- a/sound/soc/intel/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -34,7 +34,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../common/sst-dsp.h" +#include "../../common/sst-dsp.h" int sst_shim_write(void __iomem *addr, int offset, int value) { diff --git a/sound/soc/intel/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c similarity index 99% rename from sound/soc/intel/sst/sst_stream.c rename to sound/soc/intel/atom/sst/sst_stream.c index 7638fca02de078..a74c64c7053cfa 100644 --- a/sound/soc/intel/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -31,7 +31,7 @@ #include #include "../sst-mfld-platform.h" #include "sst.h" -#include "../common/sst-dsp.h" +#include "../../common/sst-dsp.h" int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5c2d8fabb5ede4..7f55d59024a88b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -27,7 +27,7 @@ #include #include #include "../../codecs/rt5640.h" -#include "../sst-atom-controls.h" +#include "../atom/sst-atom-controls.h" static const struct snd_soc_dapm_widget byt_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 93bb6711ba3d36..20a28b22e30fea 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -28,7 +28,7 @@ #include #include #include "../../codecs/rt5645.h" -#include "../sst-atom-controls.h" +#include "../atom/sst-atom-controls.h" #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "rt5645-aif1" diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2cea002a31bb91..2c9cc5be439e65 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -24,7 +24,7 @@ #include #include #include "../../codecs/rt5670.h" -#include "../sst-atom-controls.h" +#include "../atom/sst-atom-controls.h" /* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */ #define CHT_PLAT_CLK_3_HZ 19200000 From f34c4bc7e599bb895f77381c4d91ccc77635d68f Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Tue, 7 Apr 2015 03:06:06 +0800 Subject: [PATCH 348/411] ASoC: Intel: read_shim_data() can be static Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_pvt.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index 3c178444638bd9..2d7424956d051f 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -111,7 +111,7 @@ int sst_wait_interruptible(struct intel_sst_drv *sst_drv_ctx, } -unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) +static unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) { unsigned long long val = 0; @@ -124,7 +124,7 @@ unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) return val; } -void write_shim_data(struct intel_sst_drv *sst, int addr, +static void write_shim_data(struct intel_sst_drv *sst, int addr, unsigned long long data) { switch (sst->dev_id) { From 4cd9db08598454e4d051b818c5f1d61ac7539a47 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 7 Apr 2015 14:03:53 +0300 Subject: [PATCH 349/411] ASoC: davinci-mcasp: Fix ruledata setup in davinci_mcasp_startup Passing &mcasp->ruledata[dir] to snd_pcm_hw_rule_add() is not correct since commit: 7b3d165a2821 ASoC: davinci-mcasp: Index ruledata in drvdata with substream->stream now sets up the struct based on the substream->stream (0 or 1) while we pass a pointer which we take with dir (1 or 2). This will lead kernel crash. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0b6b1b286201d6..bb4b78eada586d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1128,6 +1128,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + struct davinci_mcasp_ruledata *ruledata = + &mcasp->ruledata[substream->stream]; u32 max_channels = 0; int i, dir; @@ -1149,7 +1151,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } - mcasp->ruledata[substream->stream].serializers = max_channels; + ruledata->serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1172,12 +1174,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { int ret; - mcasp->ruledata[substream->stream].mcasp = mcasp; + ruledata->mcasp = mcasp; ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, davinci_mcasp_hw_rule_rate, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (ret) @@ -1185,7 +1187,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, davinci_mcasp_hw_rule_format, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (ret) @@ -1193,7 +1195,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, davinci_mcasp_hw_rule_channels, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (ret) From 7e5ee1c33e9ce4e4be0a6b8955c760e9a41a9e84 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 7 Apr 2015 11:34:50 +0100 Subject: [PATCH 350/411] ASoC: wm8804: Add support for hardware reset line It is best to use the physical reset if it is available. This patch adds support for a GPIO controlled physical reset for the chip. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index f44da83f50dc85..cc168c4a4be022 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -61,6 +62,8 @@ struct wm8804_priv { struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; int mclk_div; + + struct gpio_desc *reset; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -182,7 +185,7 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct wm8804_priv *wm8804) +static int wm8804_soft_reset(struct wm8804_priv *wm8804) { return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } @@ -586,6 +589,14 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + wm8804->reset = devm_gpiod_get_optional(dev, "wlf,reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm8804->reset)) { + ret = PTR_ERR(wm8804->reset); + dev_err(dev, "Failed to get reset line: %d\n", ret); + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; @@ -620,6 +631,9 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) return ret; } + if (wm8804->reset) + gpiod_set_value_cansleep(wm8804->reset, 1); + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); if (ret < 0) { dev_err(dev, "Failed to read device ID: %d\n", ret); @@ -648,10 +662,12 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) } dev_info(dev, "revision %c\n", id1 + 'A'); - ret = wm8804_reset(wm8804); - if (ret < 0) { - dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; + if (!wm8804->reset) { + ret = wm8804_soft_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } } ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, From 9b5c352e5425622b4371387d4718060da5d3ae32 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 7 Apr 2015 11:34:51 +0100 Subject: [PATCH 351/411] ASoC: wm8804: Update binding documentation to include reset GPIO Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/wm8804.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8804.txt b/Documentation/devicetree/bindings/sound/wm8804.txt index 10ef07606b84c0..6fd124b1649652 100644 --- a/Documentation/devicetree/bindings/sound/wm8804.txt +++ b/Documentation/devicetree/bindings/sound/wm8804.txt @@ -13,6 +13,10 @@ Required properties: - PVDD-supply, DVDD-supply : Power supplies for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt +Optional properties: + + - wlf,reset-gpio: A GPIO specifier for the GPIO controlling the reset pin + Example: codec: wm8804@1a { From 2d846c74027231264e57e4e34faf9576394e3ce0 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Tue, 7 Apr 2015 20:32:20 +0800 Subject: [PATCH 352/411] ALSA: hda_intel: add AZX_DCAPS_I915_POWERWELL for SKL and BSW HDMI/DP codec on SKL/BSW is in the power well. The power well must be turned on before probing the HDMI/DP codec. This is a temporary patch, which will power on the powerwell by adding AZX_DCAPS_I915_POWERWELL for SKL and BSW. After restructuring and new flag is added, this patch will be reverted. Signed-off-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index feebc1dda91270..9bcc5457a83e9a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,8 +297,12 @@ enum { AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) +#define AZX_DCAPS_INTEL_BRASWELL \ + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL) + #define AZX_DCAPS_INTEL_SKYLAKE \ - (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG) + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + AZX_DCAPS_I915_POWERWELL) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -1991,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL }, /* ICH6 */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, From f3b703326541d0c1ce85f5e570f6d2b6bd4296ec Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 8 Apr 2015 15:01:17 +0800 Subject: [PATCH 353/411] ALSA: hda/realtek - Support headset mode for ALC286/288 Support headset mode for ALC286 and ALC288 platforms. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 77 +++++++++++++++++++++++++++++++++++ 1 file changed, 77 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7f46d063af573e..ceb599a8780398 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3542,6 +3542,14 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) WRITE_COEF(0x32, 0x42a3), {} }; + static struct coef_fw coef0288[] = { + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), + UPDATE_COEF(0x50, 0x2000, 0x2000), + UPDATE_COEF(0x56, 0x0006, 0x0006), + UPDATE_COEF(0x66, 0x0008, 0), + UPDATE_COEF(0x67, 0x2000, 0), + {} + }; static struct coef_fw coef0292[] = { WRITE_COEF(0x76, 0x000e), WRITE_COEF(0x6c, 0x2400), @@ -3573,6 +3581,10 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0283: alc_process_coef_fw(codec, coef0233); break; + case 0x10ec0286: + case 0x10ec0288: + alc_process_coef_fw(codec, coef0288); + break; case 0x10ec0292: alc_process_coef_fw(codec, coef0292); break; @@ -3602,6 +3614,14 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, WRITE_COEF(0x26, 0x008c), {} }; + static struct coef_fw coef0288[] = { + UPDATE_COEF(0x50, 0x2000, 0), + UPDATE_COEF(0x56, 0x0006, 0), + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), + UPDATE_COEF(0x66, 0x0008, 0x0008), + UPDATE_COEF(0x67, 0x2000, 0x2000), + {} + }; static struct coef_fw coef0292[] = { WRITE_COEF(0x19, 0xa208), WRITE_COEF(0x2e, 0xacf0), @@ -3635,6 +3655,13 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, alc_process_coef_fw(codec, coef0233); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0286: + case 0x10ec0288: + alc_update_coef_idx(codec, 0x4f, 0x000c, 0); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_process_coef_fw(codec, coef0288); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; case 0x10ec0292: snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0292); @@ -3670,6 +3697,14 @@ static void alc_headset_mode_default(struct hda_codec *codec) WRITE_COEF(0x32, 0x4ea3), {} }; + static struct coef_fw coef0288[] = { + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), /* Set to TRS type */ + UPDATE_COEF(0x50, 0x2000, 0x2000), + UPDATE_COEF(0x56, 0x0006, 0x0006), + UPDATE_COEF(0x66, 0x0008, 0), + UPDATE_COEF(0x67, 0x2000, 0), + {} + }; static struct coef_fw coef0292[] = { WRITE_COEF(0x76, 0x000e), WRITE_COEF(0x6c, 0x2400), @@ -3699,6 +3734,11 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0283: alc_process_coef_fw(codec, coef0233); break; + case 0x10ec0286: + case 0x10ec0288: + alc_process_coef_fw(codec, coef0288); + break; + break; case 0x10ec0292: alc_process_coef_fw(codec, coef0292); break; @@ -3727,6 +3767,13 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) WRITE_COEF(0x32, 0x4ea3), {} }; + static struct coef_fw coef0288[] = { + UPDATE_COEF(0x50, 0x2000, 0x2000), + UPDATE_COEF(0x56, 0x0006, 0x0006), + UPDATE_COEF(0x66, 0x0008, 0), + UPDATE_COEF(0x67, 0x2000, 0), + {} + }; static struct coef_fw coef0292[] = { WRITE_COEF(0x6b, 0xd429), WRITE_COEF(0x76, 0x0008), @@ -3754,6 +3801,12 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0283: alc_process_coef_fw(codec, coef0233); break; + case 0x10ec0286: + case 0x10ec0288: + alc_update_coef_idx(codec, 0x4f, 0xfcc0, 0xd400); + msleep(300); + alc_process_coef_fw(codec, coef0288); + break; case 0x10ec0292: alc_process_coef_fw(codec, coef0292); break; @@ -3782,6 +3835,13 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) WRITE_COEF(0x32, 0x4ea3), {} }; + static struct coef_fw coef0288[] = { + UPDATE_COEF(0x50, 0x2000, 0x2000), + UPDATE_COEF(0x56, 0x0006, 0x0006), + UPDATE_COEF(0x66, 0x0008, 0), + UPDATE_COEF(0x67, 0x2000, 0), + {} + }; static struct coef_fw coef0292[] = { WRITE_COEF(0x6b, 0xe429), WRITE_COEF(0x76, 0x0008), @@ -3809,6 +3869,12 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0283: alc_process_coef_fw(codec, coef0233); break; + case 0x10ec0286: + case 0x10ec0288: + alc_update_coef_idx(codec, 0x4f, 0xfcc0, 0xe400); + msleep(300); + alc_process_coef_fw(codec, coef0288); + break; case 0x10ec0292: alc_process_coef_fw(codec, coef0292); break; @@ -3833,6 +3899,10 @@ static void alc_determine_headset_type(struct hda_codec *codec) conteol) */ {} }; + static struct coef_fw coef0288[] = { + UPDATE_COEF(0x4f, 0xfcc0, 0xd400), /* Check Type */ + {} + }; static struct coef_fw coef0293[] = { UPDATE_COEF(0x4a, 0x000f, 0x0008), /* Combo Jack auto detect */ WRITE_COEF(0x45, 0xD429), /* Set to ctia type */ @@ -3861,6 +3931,13 @@ static void alc_determine_headset_type(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x0070) == 0x0070; break; + case 0x10ec0286: + case 0x10ec0288: + alc_process_coef_fw(codec, coef0288); + msleep(350); + val = alc_read_coef_idx(codec, 0x50); + is_ctia = (val & 0x0070) == 0x0070; + break; case 0x10ec0292: alc_write_coef_idx(codec, 0x6b, 0xd429); msleep(300); From e1e62b98ebddc3234f3259019d3236f66fc667f8 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 8 Apr 2015 16:01:22 +0800 Subject: [PATCH 354/411] ALSA: hda/realtek - Support Dell headset mode for ALC288 Dell create new platform with ALC288 codec. This patch will enable headset mode for Dino platform. [slight code refactoring and compile fix by tiwai] Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 67 +++++++++++++++++++++++++++++++++++ 1 file changed, 67 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ceb599a8780398..7b5c93e0e78cdf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4121,6 +4121,29 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc288_update_headset_jack_cb(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + struct alc_spec *spec = codec->spec; + int present; + + alc_update_headset_jack_cb(codec, jack); + /* Headset Mic enable or disable, only for Dell Dino */ + present = spec->gen.hp_jack_present ? 0x40 : 0; + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + present); +} + +static void alc_fixup_headset_mode_dell_alc288(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_headset_mode(codec, fix, action); + if (action == HDA_FIXUP_ACT_PROBE) { + struct alc_spec *spec = codec->spec; + spec->gen.hp_automute_hook = alc288_update_headset_jack_cb; + } +} + static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4460,6 +4483,9 @@ enum { ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, ALC280_FIXUP_HP_DOCK_PINS, + ALC288_FIXUP_DELL_HEADSET_MODE, + ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC288_FIXUP_DELL_XPS_13_GPIO6, }; static const struct hda_fixup alc269_fixups[] = { @@ -4948,6 +4974,33 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC280_FIXUP_HP_GPIO4 }, + [ALC288_FIXUP_DELL_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_dell_alc288, + .chained = true, + .chain_id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED + }, + [ALC288_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC288_FIXUP_DELL_HEADSET_MODE + }, + [ALC288_FIXUP_DELL_XPS_13_GPIO6] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x01, AC_VERB_SET_GPIO_MASK, 0x40}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x40}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + { } + }, + .chained = true, + .chain_id = ALC288_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5174,6 +5227,13 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x1b, 0x411111f0}, \ {0x1e, 0x411111f0} +#define ALC288_STANDARD_PINS \ + {0x17, 0x411111f0}, \ + {0x18, 0x411111f0}, \ + {0x19, 0x411111f0}, \ + {0x1a, 0x411111f0}, \ + {0x1e, 0x411111f0} + #define ALC290_STANDARD_PINS \ {0x12, 0x99a30130}, \ {0x13, 0x40000000}, \ @@ -5369,6 +5429,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x19, 0x03a11020}, {0x1d, 0x40e00001}, {0x21, 0x0321101f}), + SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL_XPS_13_GPIO6, + ALC288_STANDARD_PINS, + {0x12, 0x90a60120}, + {0x13, 0x40000000}, + {0x14, 0x90170110}, + {0x1d, 0x4076832d}, + {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x14, 0x411111f0}, From 142267c9e026827ca5fa622f1f13780b6db26cf8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Apr 2015 11:41:59 +0200 Subject: [PATCH 355/411] ALSA: hda - Create AFG sysfs node at last ... so that user-space can know that the whole nodes have been created. Unfortunately, this can't be implemented easily in race-free way, so it's a kind of compromise. Signed-off-by: Takashi Iwai --- sound/hda/hdac_sysfs.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index b358d515780238..18aea43d230d9f 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -313,12 +313,12 @@ static void widget_tree_free(struct hdac_device *codec) if (!tree) return; + free_widget_node(tree->afg, &widget_afg_group); if (tree->nodes) { for (p = tree->nodes; *p; p++) free_widget_node(*p, &widget_node_group); kfree(tree->nodes); } - free_widget_node(tree->afg, &widget_afg_group); if (tree->root) kobject_put(tree->root); kfree(tree); @@ -362,13 +362,6 @@ static int widget_tree_create(struct hdac_device *codec) if (!tree->root) return -ENOMEM; - if (codec->afg) { - err = add_widget_node(tree->root, codec->afg, - &widget_afg_group, &tree->afg); - if (err < 0) - return err; - } - tree->nodes = kcalloc(codec->num_nodes + 1, sizeof(*tree->nodes), GFP_KERNEL); if (!tree->nodes) @@ -381,6 +374,13 @@ static int widget_tree_create(struct hdac_device *codec) return err; } + if (codec->afg) { + err = add_widget_node(tree->root, codec->afg, + &widget_afg_group, &tree->afg); + if (err < 0) + return err; + } + kobject_uevent(tree->root, KOBJ_CHANGE); return 0; } From 8e64aedf80ae14b852abc0d7ca262530b69e9a18 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 7 Apr 2015 20:14:59 +0800 Subject: [PATCH 356/411] ASoC: Intel: Fix a buffer overflow issue 0day robot reported a buffer overflow issue: ... sound/soc/intel/haswell/sst-haswell-pcm.c:1107 hsw_pcm_probe() error: buffer\ overflow 'hsw_dais' 4 <= 4 sound/soc/intel/haswell/sst-haswell-pcm.c:1109 hsw_pcm_probe() error: buffer\ overflow 'hsw_dais' 4 <= 4 ... Fix it by initializing the index(i) to correct value. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 157b3a6c509ecc..23ae0400d6db99 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1103,7 +1103,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) return 0; err: - for (;i >= 0; i--) { + for (--i; i >= 0; i--) { if (hsw_dais[i].playback.channels_min) snd_dma_free_pages(&priv_data->dmab[i][0]); if (hsw_dais[i].capture.channels_min) From 664c715573c2c116c2d8f5de7d59ce85a98a1751 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Apr 2015 11:43:14 +0200 Subject: [PATCH 357/411] ALSA: hda - Work around races of power up/down with runtime PM Currently, snd_hdac_power_up()/down() helpers checks whether the codec is being in pm (suspend/resume), and skips the call of runtime get/put during it. This is needed as there are lots of power up/down sequences called in the paths that are also used in the PM itself. An example is found in hda_codec.c::codec_exec_verb(), where this can power up the codec while it may be called again in its power up sequence, too. The above works in most cases, but sometimes we really want to wait for the real power up. For example, the control element get/put may want explicit power up so that the value change is assured to reach to the hardware. Using the current snd_hdac_power_up(), however, results in a race, e.g. when it's called during the runtime suspend is being performed. In the worst case, as found in patch_ca0132.c, it can even lead to the deadlock because the code assumes the power up while it was skipped due to the check above. For dealing with such cases, this patch makes snd_hdac_power_up() and _down() to two variants: with and without in_pm flag check. The version with pm flag check is named as snd_hdac_power_up_pm() while the version without pm flag check is still kept as snd_hdac_power_up(). (Just because the usage of the former is fewer.) Then finally, the patch replaces each call potentially done in PM with the new _pm() variant. In theory, we can implement a unified version -- if we can distinguish the current context whether it's in the pm path. But such an implementation is cumbersome, so leave the code like this a bit messy way for now... Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96271 Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 28 ++++++++++++++++++++++++++++ sound/hda/hdac_device.c | 16 +++++++--------- sound/hda/hdac_regmap.c | 8 ++++---- sound/pci/hda/hda_codec.c | 8 ++++---- sound/pci/hda/hda_codec.h | 2 ++ sound/pci/hda/patch_ca0132.c | 12 ++++++------ sound/pci/hda/patch_hdmi.c | 4 ++-- 7 files changed, 53 insertions(+), 25 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 95acc337aea5af..30446f17c6a65c 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -144,6 +144,34 @@ static inline void snd_hdac_power_up(struct hdac_device *codec) {} static inline void snd_hdac_power_down(struct hdac_device *codec) {} #endif +/** + * snd_hdac_power_up_pm - power up the codec + * @codec: the codec object + * + * This function can be called in a recursive code path like init code + * which may be called by PM suspend/resume again. OTOH, if a power-up + * call must wake up the sleeper (e.g. in a kctl callback), use + * snd_hdac_power_up() instead. + */ +static inline void snd_hdac_power_up_pm(struct hdac_device *codec) +{ + if (!atomic_read(&codec->in_pm)) + snd_hdac_power_up(codec); +} + +/** + * snd_hdac_power_down_pm - power down the codec + * @codec: the codec object + * + * Like snd_hdac_power_up_pm(), this function is used in a recursive + * code path like init code which may be called by PM suspend/resume again. + */ +static inline void snd_hdac_power_down_pm(struct hdac_device *codec) +{ + if (!atomic_read(&codec->in_pm)) + snd_hdac_power_down(codec); +} + /* * HD-audio codec base driver */ diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index d4a0e723af2ca6..92604bbcee5ff1 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -494,29 +494,27 @@ EXPORT_SYMBOL_GPL(snd_hdac_get_connections); #ifdef CONFIG_PM /** - * snd_hdac_power_up - increment the runtime pm counter + * snd_hdac_power_up - power up the codec * @codec: the codec object + * + * This function calls the runtime PM helper to power up the given codec. + * Unlike snd_hdac_power_up_pm(), you should call this only for the code + * path that isn't included in PM path. Otherwise it gets stuck. */ void snd_hdac_power_up(struct hdac_device *codec) { - struct device *dev = &codec->dev; - - if (atomic_read(&codec->in_pm)) - return; - pm_runtime_get_sync(dev); + pm_runtime_get_sync(&codec->dev); } EXPORT_SYMBOL_GPL(snd_hdac_power_up); /** - * snd_hdac_power_up - decrement the runtime pm counter + * snd_hdac_power_down - power down the codec * @codec: the codec object */ void snd_hdac_power_down(struct hdac_device *codec) { struct device *dev = &codec->dev; - if (atomic_read(&codec->in_pm)) - return; pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); } diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 1eb43209fe2c59..64876fa357c9f7 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -402,9 +402,9 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_write(codec, reg, val); if (err == -EAGAIN) { - snd_hdac_power_up(codec); + snd_hdac_power_up_pm(codec); err = reg_raw_write(codec, reg, val); - snd_hdac_power_down(codec); + snd_hdac_power_down_pm(codec); } return err; } @@ -434,9 +434,9 @@ int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_read(codec, reg, val); if (err == -EAGAIN) { - snd_hdac_power_up(codec); + snd_hdac_power_up_pm(codec); err = reg_raw_read(codec, reg, val); - snd_hdac_power_down(codec); + snd_hdac_power_down_pm(codec); } return err; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 16dfa1ed10dd1d..e70a7fb393dd5d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -137,7 +137,7 @@ static int codec_exec_verb(struct hdac_device *dev, unsigned int cmd, return -1; again: - snd_hda_power_up(codec); + snd_hda_power_up_pm(codec); mutex_lock(&bus->core.cmd_mutex); if (flags & HDA_RW_NO_RESPONSE_FALLBACK) bus->no_response_fallback = 1; @@ -145,7 +145,7 @@ static int codec_exec_verb(struct hdac_device *dev, unsigned int cmd, cmd, res); bus->no_response_fallback = 0; mutex_unlock(&bus->core.cmd_mutex); - snd_hda_power_down(codec); + snd_hda_power_down_pm(codec); if (!codec_in_pm(codec) && res && err < 0 && bus->rirb_error) { if (bus->response_reset) { codec_dbg(codec, @@ -3951,7 +3951,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, if (!(v & HDA_AMP_MUTE) && v > 0) { if (!check->power_on) { check->power_on = 1; - snd_hda_power_up(codec); + snd_hda_power_up_pm(codec); } return 1; } @@ -3959,7 +3959,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, } if (check->power_on) { check->power_on = 0; - snd_hda_power_down(codec); + snd_hda_power_down_pm(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index acf868c6a785c1..9075ac28dc4b03 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -508,7 +508,9 @@ const char *snd_hda_get_jack_location(u32 cfg); * power saving */ #define snd_hda_power_up(codec) snd_hdac_power_up(&(codec)->core) +#define snd_hda_power_up_pm(codec) snd_hdac_power_up_pm(&(codec)->core) #define snd_hda_power_down(codec) snd_hdac_power_down(&(codec)->core) +#define snd_hda_power_down_pm(codec) snd_hdac_power_down_pm(&(codec)->core) #ifdef CONFIG_PM void snd_hda_set_power_save(struct hda_bus *bus, int delay); void snd_hda_update_power_acct(struct hda_codec *codec); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 5aff35a09fd47b..4a4e7b282e4f87 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3131,7 +3131,7 @@ static int ca0132_select_out(struct hda_codec *codec) codec_dbg(codec, "ca0132_select_out\n"); - snd_hda_power_up(codec); + snd_hda_power_up_pm(codec); auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; @@ -3215,7 +3215,7 @@ static int ca0132_select_out(struct hda_codec *codec) } exit: - snd_hda_power_down(codec); + snd_hda_power_down_pm(codec); return err < 0 ? err : 0; } @@ -3293,7 +3293,7 @@ static int ca0132_select_mic(struct hda_codec *codec) codec_dbg(codec, "ca0132_select_mic\n"); - snd_hda_power_up(codec); + snd_hda_power_up_pm(codec); auto_jack = spec->vnode_lswitch[VNID_AMIC1_ASEL - VNODE_START_NID]; @@ -3326,7 +3326,7 @@ static int ca0132_select_mic(struct hda_codec *codec) ca0132_effects_set(codec, VOICE_FOCUS, 0); } - snd_hda_power_down(codec); + snd_hda_power_down_pm(codec); return 0; } @@ -4546,7 +4546,7 @@ static int ca0132_init(struct hda_codec *codec) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; - snd_hda_power_up(codec); + snd_hda_power_up_pm(codec); ca0132_init_unsol(codec); @@ -4577,7 +4577,7 @@ static int ca0132_init(struct hda_codec *codec) snd_hda_jack_report_sync(codec); - snd_hda_power_down(codec); + snd_hda_power_down_pm(codec); return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ca0c05e1c42ede..5f44f60a638976 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1545,7 +1545,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) bool eld_changed = false; bool ret; - snd_hda_power_up(codec); + snd_hda_power_up_pm(codec); present = snd_hda_pin_sense(codec, pin_nid); mutex_lock(&per_pin->lock); @@ -1631,7 +1631,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) jack->block_report = !ret; mutex_unlock(&per_pin->lock); - snd_hda_power_down(codec); + snd_hda_power_down_pm(codec); return ret; } From 1f544fd8ff377127a512e20358045cc9b92c245c Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Wed, 8 Apr 2015 18:34:25 +0530 Subject: [PATCH 358/411] ASoC: Intel: remove unused functions these functions were never called by anyone. Signed-off-by: Sudip Mukherjee Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_pvt.c | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index 2d7424956d051f..adb32fefd693a0 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -111,30 +111,6 @@ int sst_wait_interruptible(struct intel_sst_drv *sst_drv_ctx, } -static unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) -{ - unsigned long long val = 0; - - switch (sst->dev_id) { - case SST_MRFLD_PCI_ID: - case SST_BYT_ACPI_ID: - val = sst_shim_read64(sst->shim, addr); - break; - } - return val; -} - -static void write_shim_data(struct intel_sst_drv *sst, int addr, - unsigned long long data) -{ - switch (sst->dev_id) { - case SST_MRFLD_PCI_ID: - case SST_BYT_ACPI_ID: - sst_shim_write64(sst->shim, addr, (u64) data); - break; - } -} - /* * sst_wait_timeout - wait on event for timeout * From c6b424fee7511407107403d3a7c50e0756ae282e Mon Sep 17 00:00:00 2001 From: Caesar Wang Date: Wed, 8 Apr 2015 19:05:56 +0800 Subject: [PATCH 359/411] ASoC: max98090: add shutdown callback for max98090 To fix pop noise when shutdown,the pop noise during shutdown is the pmic cutoff power of codec without any notice. Signed-off-by: jay.xu Signed-off-by: zhengxing Signed-off-by: Caesar Wang Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b112b1c2c394a1..3e33ef2acf3c99 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2605,8 +2605,24 @@ static int max98090_i2c_probe(struct i2c_client *i2c, return ret; } +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + /* + * Enable volume smoothing, disable zero cross. This will cause + * a quick 40ms ramp to mute on shutdown. + */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + static int max98090_i2c_remove(struct i2c_client *client) { + max98090_i2c_shutdown(client); snd_soc_unregister_codec(&client->dev); return 0; } @@ -2696,6 +2712,7 @@ static struct i2c_driver max98090_i2c_driver = { .acpi_match_table = ACPI_PTR(max98090_acpi_match), }, .probe = max98090_i2c_probe, + .shutdown = max98090_i2c_shutdown, .remove = max98090_i2c_remove, .id_table = max98090_i2c_id, }; From 5631f18763f1b0989cec7cbf8f3badcb74dfe469 Mon Sep 17 00:00:00 2001 From: Sapthagiri Baratam Date: Tue, 7 Apr 2015 12:55:09 +0100 Subject: [PATCH 360/411] ASoC: wm8804: Add DAPM widgets for SPDIF/AIF This change converts the driver to use DAPM to control the power for the various blocks on the chip. As part of this change the existing controls "TX Playback Switch" (controlled power for the SPDIF TX block) and "AIF Playback Switch" (controlled power for the AIF block) are both removed, as they are now redundant since the power state of those blocks is controlled automatically by DAPM. There are several benefits of this change, the most important of which is this change adds support for powering down the SPDIF RX block. The RX block will automatically assume control of the PLL on the chip when it is receiving a signal, so leaving this enabled all the time as was currently done in the driver can be problematic. An incoming SPDIF signal that is not being used can completely destroy the clocking for an in use TX signal. But this change ensures that the RX block will only be powered when the user intends to be receiving data, thus avoiding this issue. Additional benefits include the chip being simpler to operate as the power no longer needs to be manually controlled between use-cases and a small power saving (although it is acknowledged that this is likely unimportant in the typical use-cases for this chip). Signed-off-by: Sapthagiri Baratam Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 140 ++++++++++++++++++++++---------------- 1 file changed, 83 insertions(+), 57 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index cc168c4a4be022..cff34be61f8879 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -25,6 +25,7 @@ #include #include #include +#include #include "wm8804.h" @@ -64,14 +65,16 @@ struct wm8804_priv { int mclk_div; struct gpio_desc *reset; -}; -static int txsrc_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); + int aif_pwr; +}; static int txsrc_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +static int wm8804_aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); + /* * We can't use the same notifier block for more than one supply and * there's no way I can see to get from a callback to the caller @@ -93,26 +96,62 @@ WM8804_REGULATOR_EVENT(0) WM8804_REGULATOR_EVENT(1) static const char *txsrc_text[] = { "S/PDIF RX", "AIF" }; -static SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text); +static const SOC_ENUM_SINGLE_DECL(txsrc, WM8804_SPDTX4, 6, txsrc_text); -static const struct snd_kcontrol_new wm8804_snd_controls[] = { - SOC_ENUM_EXT("Input Source", txsrc, txsrc_get, txsrc_put), - SOC_SINGLE("TX Playback Switch", WM8804_PWRDN, 2, 1, 1), - SOC_SINGLE("AIF Playback Switch", WM8804_PWRDN, 4, 1, 1) +static const struct snd_kcontrol_new wm8804_tx_source_mux[] = { + SOC_DAPM_ENUM_EXT("Input Source", txsrc, + snd_soc_dapm_get_enum_double, txsrc_put), }; -static int txsrc_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec; - unsigned int src; +static const struct snd_soc_dapm_widget wm8804_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("SPDIF Out"), +SND_SOC_DAPM_INPUT("SPDIF In"), + +SND_SOC_DAPM_PGA("SPDIFTX", WM8804_PWRDN, 2, 1, NULL, 0), +SND_SOC_DAPM_PGA("SPDIFRX", WM8804_PWRDN, 1, 1, NULL, 0), + +SND_SOC_DAPM_MUX("Tx Source", SND_SOC_NOPM, 6, 0, wm8804_tx_source_mux), + +SND_SOC_DAPM_AIF_OUT_E("AIFTX", NULL, 0, SND_SOC_NOPM, 0, 0, wm8804_aif_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_AIF_IN_E("AIFRX", NULL, 0, SND_SOC_NOPM, 0, 0, wm8804_aif_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route wm8804_dapm_routes[] = { + { "AIFRX", NULL, "Playback" }, + { "Tx Source", "AIF", "AIFRX" }, + + { "SPDIFRX", NULL, "SPDIF In" }, + { "Tx Source", "S/PDIF RX", "SPDIFRX" }, + + { "SPDIFTX", NULL, "Tx Source" }, + { "SPDIF Out", NULL, "SPDIFTX" }, - codec = snd_soc_kcontrol_codec(kcontrol); - src = snd_soc_read(codec, WM8804_SPDTX4); - if (src & 0x40) - ucontrol->value.integer.value[0] = 1; - else - ucontrol->value.integer.value[0] = 0; + { "AIFTX", NULL, "SPDIFRX" }, + { "Capture", NULL, "AIFTX" }, +}; + +static int wm8804_aif_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct wm8804_priv *wm8804 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* power up the aif */ + if (!wm8804->aif_pwr) + snd_soc_update_bits(codec, WM8804_PWRDN, 0x10, 0x0); + wm8804->aif_pwr++; + break; + case SND_SOC_DAPM_POST_PMD: + /* power down only both paths are disabled */ + wm8804->aif_pwr--; + if (!wm8804->aif_pwr) + snd_soc_update_bits(codec, WM8804_PWRDN, 0x10, 0x10); + break; + } return 0; } @@ -120,48 +159,33 @@ static int txsrc_get(struct snd_kcontrol *kcontrol, static int txsrc_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec; - unsigned int src, txpwr; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val = ucontrol->value.enumerated.item[0] << e->shift_l; + unsigned int mask = 1 << e->shift_l; + unsigned int txpwr; + + if (val != 0 && val != mask) + return -EINVAL; - codec = snd_soc_kcontrol_codec(kcontrol); + snd_soc_dapm_mutex_lock(dapm); - if (ucontrol->value.integer.value[0] != 0 - && ucontrol->value.integer.value[0] != 1) - return -EINVAL; + if (snd_soc_test_bits(codec, e->reg, mask, val)) { + /* save the current power state of the transmitter */ + txpwr = snd_soc_read(codec, WM8804_PWRDN) & 0x4; - src = snd_soc_read(codec, WM8804_SPDTX4); - switch ((src & 0x40) >> 6) { - case 0: - if (!ucontrol->value.integer.value[0]) - return 0; - break; - case 1: - if (ucontrol->value.integer.value[1]) - return 0; - break; - } + /* power down the transmitter */ + snd_soc_update_bits(codec, WM8804_PWRDN, 0x4, 0x4); - /* save the current power state of the transmitter */ - txpwr = snd_soc_read(codec, WM8804_PWRDN) & 0x4; - /* power down the transmitter */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x4, 0x4); - /* set the tx source */ - snd_soc_update_bits(codec, WM8804_SPDTX4, 0x40, - ucontrol->value.integer.value[0] << 6); - - if (ucontrol->value.integer.value[0]) { - /* power down the receiver */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x2, 0x2); - /* power up the AIF */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x10, 0); - } else { - /* don't power down the AIF -- may be used as an output */ - /* power up the receiver */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x2, 0); + /* set the tx source */ + snd_soc_update_bits(codec, e->reg, mask, val); + + /* restore the transmitter's configuration */ + snd_soc_update_bits(codec, WM8804_PWRDN, 0x4, txpwr); } - /* restore the transmitter's configuration */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x4, txpwr); + snd_soc_dapm_mutex_unlock(dapm); return 0; } @@ -558,8 +582,10 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, - .controls = wm8804_snd_controls, - .num_controls = ARRAY_SIZE(wm8804_snd_controls), + .dapm_widgets = wm8804_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8804_dapm_widgets), + .dapm_routes = wm8804_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8804_dapm_routes), }; const struct regmap_config wm8804_regmap_config = { From 1a60667fc81fdab3733a1d41480da3a5d0ccecea Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 7 Apr 2015 12:55:10 +0100 Subject: [PATCH 361/411] ASoC: wm8804: Enable runtime PM Currently both the oscillator and the PLL are powered up in set_bias_level. This can be problematic when using output clocks from the wm8804 for other devices. The snd_soc_codec_set_pll API defines that a clock should be available once the call returns, however, with all the clocking controlled in set_bias_level this is not currently the case. This patch enables pm_runtime for the wm8804, enabling both the regulators and the oscillator when the chip resumes, and enabling the PLL in the snd_soc_codec_set_pll call. Naturally the enabling the PLL will also cause the chip to resume. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804-i2c.c | 1 + sound/soc/codecs/wm8804-spi.c | 1 + sound/soc/codecs/wm8804.c | 109 ++++++++++++++++++---------------- sound/soc/codecs/wm8804.h | 1 + 4 files changed, 62 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c index 5bd4af2b4059c4..6596f5f3a0c351 100644 --- a/sound/soc/codecs/wm8804-i2c.c +++ b/sound/soc/codecs/wm8804-i2c.c @@ -50,6 +50,7 @@ static struct i2c_driver wm8804_i2c_driver = { .driver = { .name = "wm8804", .owner = THIS_MODULE, + .pm = &wm8804_pm, .of_match_table = wm8804_of_match, }, .probe = wm8804_i2c_probe, diff --git a/sound/soc/codecs/wm8804-spi.c b/sound/soc/codecs/wm8804-spi.c index 287e11e9079428..407a3cf391e505 100644 --- a/sound/soc/codecs/wm8804-spi.c +++ b/sound/soc/codecs/wm8804-spi.c @@ -43,6 +43,7 @@ static struct spi_driver wm8804_spi_driver = { .driver = { .name = "wm8804", .owner = THIS_MODULE, + .pm = &wm8804_pm, .of_match_table = wm8804_of_match, }, .probe = wm8804_spi_probe, diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index cff34be61f8879..1e403f67cf161c 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -59,6 +60,7 @@ static const struct reg_default wm8804_reg_defaults[] = { }; struct wm8804_priv { + struct device *dev; struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; @@ -403,19 +405,19 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec; + struct snd_soc_codec *codec = dai->codec; + struct wm8804_priv *wm8804 = snd_soc_codec_get_drvdata(codec); + bool change; - codec = dai->codec; if (!freq_in || !freq_out) { /* disable the PLL */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x1, 0x1); - return 0; + regmap_update_bits_check(wm8804->regmap, WM8804_PWRDN, + 0x1, 0x1, &change); + if (change) + pm_runtime_put(wm8804->dev); } else { int ret; struct pll_div pll_div; - struct wm8804_priv *wm8804; - - wm8804 = snd_soc_codec_get_drvdata(codec); ret = pll_factors(&pll_div, freq_out, freq_in, wm8804->mclk_div); @@ -423,7 +425,10 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, return ret; /* power down the PLL before reprogramming it */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x1, 0x1); + regmap_update_bits_check(wm8804->regmap, WM8804_PWRDN, + 0x1, 0x1, &change); + if (!change) + pm_runtime_get_sync(wm8804->dev); /* set PLLN and PRESCALE */ snd_soc_update_bits(codec, WM8804_PLL4, 0xf | 0x10, @@ -501,47 +506,6 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, return 0; } -static int wm8804_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - int ret; - struct wm8804_priv *wm8804; - - wm8804 = snd_soc_codec_get_drvdata(codec); - switch (level) { - case SND_SOC_BIAS_ON: - break; - case SND_SOC_BIAS_PREPARE: - /* power up the OSC and the PLL */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x9, 0); - break; - case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, - "Failed to enable supplies: %d\n", - ret); - return ret; - } - regcache_sync(wm8804->regmap); - } - /* power down the OSC and the PLL */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x9, 0x9); - break; - case SND_SOC_BIAS_OFF: - /* power down the OSC and the PLL */ - snd_soc_update_bits(codec, WM8804_PWRDN, 0x9, 0x9); - regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - break; - } - - codec->dapm.bias_level = level; - return 0; -} - static const struct snd_soc_dai_ops wm8804_dai_ops = { .hw_params = wm8804_hw_params, .set_fmt = wm8804_set_fmt, @@ -579,7 +543,6 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, .dapm_widgets = wm8804_dapm_widgets, @@ -613,6 +576,7 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, wm8804); + wm8804->dev = dev; wm8804->regmap = regmap; wm8804->reset = devm_gpiod_get_optional(dev, "wlf,reset", @@ -703,6 +667,10 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) goto err_reg_enable; } + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + return 0; err_reg_enable: @@ -713,10 +681,51 @@ EXPORT_SYMBOL_GPL(wm8804_probe); void wm8804_remove(struct device *dev) { + pm_runtime_disable(dev); snd_soc_unregister_codec(dev); } EXPORT_SYMBOL_GPL(wm8804_remove); +#if IS_ENABLED(CONFIG_PM) +static int wm8804_runtime_resume(struct device *dev) +{ + struct wm8804_priv *wm8804 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(wm8804->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_sync(wm8804->regmap); + + /* Power up OSCCLK */ + regmap_update_bits(wm8804->regmap, WM8804_PWRDN, 0x8, 0x0); + + return 0; +} + +static int wm8804_runtime_suspend(struct device *dev) +{ + struct wm8804_priv *wm8804 = dev_get_drvdata(dev); + + /* Power down OSCCLK */ + regmap_update_bits(wm8804->regmap, WM8804_PWRDN, 0x8, 0x8); + + regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + + return 0; +} +#endif + +const struct dev_pm_ops wm8804_pm = { + SET_RUNTIME_PM_OPS(wm8804_runtime_suspend, wm8804_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(wm8804_pm); + MODULE_DESCRIPTION("ASoC WM8804 driver"); MODULE_AUTHOR("Dimitris Papastamos "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index a39a2563dc6704..aa72fa66c932a8 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -65,6 +65,7 @@ #define WM8804_MCLKDIV_128FS 1 extern const struct regmap_config wm8804_regmap_config; +extern const struct dev_pm_ops wm8804_pm; int wm8804_probe(struct device *dev, struct regmap *regmap); void wm8804_remove(struct device *dev); From 884c0f5b2ad542037b6f04de84a3fd3f82b15828 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 3 Apr 2015 13:04:05 +0200 Subject: [PATCH 362/411] ASoC: tegra_alc5632: Use card DAPM context to access widgets The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 6dcd06a966c7c6..ba272e21a6fa31 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -101,9 +101,6 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, @@ -118,7 +115,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) &tegra_alc5632_hp_jack_gpio); } - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "MICBIAS1"); return 0; } From c67a443b11d96f35f8514283334c50302253f2b4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 3 Apr 2015 13:04:06 +0200 Subject: [PATCH 363/411] ASoC: tegra_rt5677: Use card DAPM context to access widgets The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5677.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index d082882547ddac..1470873ecde680 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -141,9 +141,6 @@ static const struct snd_kcontrol_new tegra_rt5677_controls[] = { static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, @@ -167,7 +164,7 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) &tegra_rt5677_mic_jack_gpio); } - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "MICBIAS1"); return 0; } From 14e954f5ddc56c0dc9c32fadf188f44f5855e0fa Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 3 Apr 2015 13:04:07 +0200 Subject: [PATCH 364/411] ASoC: tegra_wm8903: Use card DAPM context to access widgets The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 4a95b70f0cf082..21604009bc1a2d 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -171,7 +171,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); @@ -193,7 +192,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, 0); - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(&card->dapm, "MICBIAS"); return 0; } From 1b13fe718b6ebca2b12cfa3d6cd32b8b68a03e38 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 3 Apr 2015 13:04:08 +0200 Subject: [PATCH 365/411] ASoC: tegra_wm9712: Use card DAPM context to access widgets The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm9712.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 2868b4839bc0a0..6492f8143ff1f1 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -46,11 +46,7 @@ static const struct snd_soc_dapm_widget tegra_wm9712_dapm_widgets[] = { static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - return snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + return snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias"); } static struct snd_soc_dai_link tegra_wm9712_dai = { From 9efe2731db3a8944d26233bb47532011411810fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2015 07:58:46 +0200 Subject: [PATCH 366/411] ALSA: hda - Always allow access for POWER_STATE verbs via regmap The hdac regmap code checks whether the codec is powered on while accessing, but there must be an exception -- the verbs to control the power state. Currently HD-audio driver doesn't access them via regmap, so this patch doesn't fix any current behavior, but it's just for future. Signed-off-by: Takashi Iwai --- sound/hda/hdac_regmap.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 64876fa357c9f7..51f1b5c8a91ccc 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -239,7 +239,7 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) int verb = get_verb(reg); int err; - if (!codec_is_running(codec)) + if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE) return -EAGAIN; reg |= (codec->addr << 28); if (is_stereo_amp_verb(reg)) @@ -265,16 +265,16 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) unsigned int verb; int i, bytes, err; - if (!codec_is_running(codec)) - return codec->lazy_cache ? 0 : -EAGAIN; - reg &= ~0x00080000U; /* drop GET bit */ reg |= (codec->addr << 28); + verb = get_verb(reg); + + if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE) + return codec->lazy_cache ? 0 : -EAGAIN; if (is_stereo_amp_verb(reg)) return hda_reg_write_stereo_amp(codec, reg, val); - verb = get_verb(reg); if (verb == AC_VERB_SET_PROC_COEF) return hda_reg_write_coef(codec, reg, val); From 2206dc949286fe0010c69213d3d8b4c53e6a2295 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2015 10:18:31 +0200 Subject: [PATCH 367/411] ALSA: hda/generic - Check power state cap at updating the widget power The new widget power-saving tries to apply the power change no matter whether the node has a power cap or not. It's bad (although most of codecs chip just ignore it). Check the capability properly beforehand. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1f2ca7be146860..afc6b1b0898cab 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -832,6 +832,8 @@ static hda_nid_t path_power_update(struct hda_codec *codec, for (i = 0; i < path->depth; i++) { nid = path->path[i]; + if (!(get_wcaps(codec, nid) & AC_WCAP_POWER)) + continue; if (nid == codec->core.afg) continue; if (!allow_powerdown || is_active_nid_for_any(codec, nid)) From d5ac0100a9027bdf488cf20247b1041f26f796f3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2015 10:21:30 +0200 Subject: [PATCH 368/411] ALSA: hda/generic - Fix wrong initial power state for fixed pins When the widget power-saving is enabled, the first automute hook invocation checks through the whole pins and it also tries to synchronize the power state. However, this results in a wrong state because it calls unconditionally snd_hda_jack_detect_state(). This patch adds a check of jack detectability before the actual jack detection call. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index afc6b1b0898cab..46b559832d2cb9 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3959,6 +3959,14 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, return changed; } +/* check the jack status for power control */ +static bool detect_pin_state(struct hda_codec *codec, hda_nid_t pin) +{ + if (!is_jack_detectable(codec, pin)) + return true; + return snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; +} + /* power up/down the paths of the given pin according to the jack state; * power = 0/1 : only power up/down if it matches with the jack state, * < 0 : force power up/down to follow the jack sate @@ -3973,7 +3981,8 @@ static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, if (!codec->power_save_node) return 0; - on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; + on = detect_pin_state(codec, pin); + if (power >= 0 && on != power) return 0; return set_path_power(codec, pin, on, -1); @@ -4225,8 +4234,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, if (codec->power_save_node) { bool on = !mute; if (on) - on = snd_hda_jack_detect_state(codec, nid) - != HDA_JACK_NOT_PRESENT; + on = detect_pin_state(codec, nid); set_path_power(codec, nid, on, -1); } } From b6c09b3c7b54743cbc58af8c27bbae8299c4f8cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2015 10:25:03 +0200 Subject: [PATCH 369/411] ALSA: hda/generic - Make snd_hda_gen_path_power_filter() always applicable Add the check of power_save_node flag at the beginning of the function so that it skips the rest if the flag isn't set. In this way, we can call this function safely no matter whether the widget power-saving is really used or not. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 46b559832d2cb9..f0475a19fad747 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4703,6 +4703,10 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) { + struct hda_gen_spec *spec = codec->spec; + + if (!spec->power_down_unused && !codec->power_save_node) + return power_state; if (power_state != AC_PWRST_D0 || nid == codec->core.afg) return power_state; if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER) From 24fef9022a83d789ee372a393ea4782dc22b9b51 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2015 10:28:15 +0200 Subject: [PATCH 370/411] ALSA: hda/generic - Don't override power_filter when power_save_node is set Currently the generic parser sets codec->power_filter when power_save_node flag is set. But this overrides the existing filter that has been already set by the codec driver, thus it looses some features. Instead, set the default power_filter only when it's not set yet. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f0475a19fad747..3d2597b7037bbc 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4918,7 +4918,8 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, parse_digital(codec); if (spec->power_down_unused || codec->power_save_node) - codec->power_filter = snd_hda_gen_path_power_filter; + if (!codec->power_filter) + codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); From 75afbd052b3675e9b812f9327e19be63f3e7b5de Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 9 Apr 2015 12:02:39 +0300 Subject: [PATCH 371/411] ASoC: Intel: do cast earlier in sst_cdev_tstamp() My static checker complains about these because it looks like the multiply can overflow and then we cast to a larger data type. I don't think this is a problem, but it's also harmless to do the cast earlier so let's silence the static checker warning. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst_drv_interface.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 718838b3fc24aa..7b50a9d17ec1bc 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -381,7 +381,7 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, tstamp->copied_total = fw_tstamp.ring_buffer_counter; tstamp->pcm_frames = fw_tstamp.frames_decoded; tstamp->pcm_io_frames = div_u64(fw_tstamp.hardware_counter, - (u64)((stream->num_ch) * SST_GET_BYTES_PER_SAMPLE(24))); + (u64)stream->num_ch * SST_GET_BYTES_PER_SAMPLE(24)); tstamp->sampling_rate = fw_tstamp.sampling_frequency; dev_dbg(dev, "PCM = %u\n", tstamp->pcm_io_frames); From cffd39668177afe5566104a2f289242c3c5ce3d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2015 10:30:25 +0200 Subject: [PATCH 372/411] ALSA: hda/realtek - Fix the regression by widget power-saving While enabling the widget power-saving for ALC269 & co, the important setup was forgotten -- stream_pm ops. Without this setup, the paths for PCM won't be powered up at all. Also, the power_filter callbacks used in ALC269 & co need to chain to the default snd_hda_gen_path_power_filter(). Tested-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b5c93e0e78cdf..acccedf8a4be3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3223,7 +3223,7 @@ static unsigned int led_power_filter(struct hda_codec *codec, snd_hda_set_pin_ctl(codec, nid, snd_hda_codec_get_pin_target(codec, nid)); - return AC_PWRST_D0; + return snd_hda_gen_path_power_filter(codec, nid, power_state); } static void alc269_fixup_hp_mute_led(struct hda_codec *codec, @@ -4186,7 +4186,7 @@ static unsigned int alc_power_filter_xps13(struct hda_codec *codec, if (spec->gen.hp_jack_present) if (nid == codec->core.afg || nid == 0x02 || nid == 0x15) return AC_PWRST_D0; - return power_state; + return snd_hda_gen_path_power_filter(codec, nid, power_state); } static void alc_fixup_dell_xps13(struct hda_codec *codec, @@ -5673,6 +5673,7 @@ static int patch_alc269(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; + codec->patch_ops.stream_pm = snd_hda_gen_stream_pm, #ifdef CONFIG_PM codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; From 0757d834eb7482e5763fb9ee014abb50789f906a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 9 Apr 2015 10:52:36 +0200 Subject: [PATCH 373/411] ASoC: Create card debugfs directory earlier Create the card debugfs directory at the begining of the initilization rather then the end as various steps in the initilization sequence will try to register files and sub-directories in the card directory. Fixes: 4e2576bd36a1 ("ASoC: soc-core: initialize debugfs in snd_soc_instantiate_card()") Reported-by: Fabio Estevam Reported-by: Nicolin Chen Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 71585d0562fa9c..3f18fa7f090dec 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1559,6 +1559,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto base_error; } + soc_init_card_debugfs(card); + card->dapm.bias_level = SND_SOC_BIAS_OFF; card->dapm.dev = card->dev; card->dapm.card = card; @@ -1680,8 +1682,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); - soc_init_card_debugfs(card); - return 0; probe_aux_dev_err: @@ -1695,6 +1695,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (card->remove) card->remove(card); + soc_cleanup_card_debugfs(card); snd_card_free(card->snd_card); base_error: From d53d59ecad74f3e90c6eefedd2186abbadd64d7c Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 Apr 2015 11:20:32 +0800 Subject: [PATCH 374/411] ASoC: rt286: Restore default in probe RT286 can't do register reset. If the hardware power is still existing in power off, rt286 will keep the register settings. So, we need to restore the default register value in probe to make sure the cache value is the same as the real register value. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 842cfb9fa19169..87af81b9e971f0 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1250,6 +1250,14 @@ static int rt286_i2c_probe(struct i2c_client *i2c, rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); + /* restore codec default */ + for (i = 0; i < INDEX_CACHE_SIZE; i++) + regmap_write(rt286->regmap, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + for (i = 0; i < ARRAY_SIZE(rt286_reg); i++) + regmap_write(rt286->regmap, rt286_reg[i].reg, + rt286_reg[i].def); + if (pdata) rt286->pdata = *pdata; From 2e55b90a5e234524f8195c657006ec4f71103dff Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 9 Apr 2015 10:52:37 +0200 Subject: [PATCH 375/411] ASoC: Make soc_dpcm_debugfs_add() non-fatal Failing to register the debugfs entries is not fatal and will not affect normal operation of the sound card. Don't abort the card registration if soc_dpcm_debugfs_add() fails. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 2 +- sound/soc/soc-core.c | 11 ++--------- sound/soc/soc-pcm.c | 8 +++----- 3 files changed, 6 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 98f2ade0266eb9..806059052bfcca 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -135,7 +135,7 @@ void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream, /* internal use only */ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); -int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); +void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); int dpcm_path_get(struct snd_soc_pcm_runtime *fe, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3f18fa7f090dec..d29c68ad83c728 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1328,15 +1328,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) #ifdef CONFIG_DEBUG_FS /* add DPCM sysfs entries */ - if (dai_link->dynamic) { - ret = soc_dpcm_debugfs_add(rtd); - if (ret < 0) { - dev_err(rtd->dev, - "ASoC: failed to add dpcm sysfs entries: %d\n", - ret); - return ret; - } - } + if (dai_link->dynamic) + soc_dpcm_debugfs_add(rtd); #endif if (cpu_dai->driver->compress_dai) { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e7cb88c2..9c514fde71542f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2802,10 +2802,10 @@ static const struct file_operations dpcm_state_fops = { .llseek = default_llseek, }; -int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) +void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) { if (!rtd->dai_link) - return 0; + return; rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, rtd->card->debugfs_card_root); @@ -2813,13 +2813,11 @@ int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) dev_dbg(rtd->dev, "ASoC: Failed to create dpcm debugfs directory %s\n", rtd->dai_link->name); - return -EINVAL; + return; } rtd->debugfs_dpcm_state = debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, rtd, &dpcm_state_fops); - - return 0; } #endif From 6553bf06a369683408895b87e5172aa99a9266bd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 9 Apr 2015 10:52:38 +0200 Subject: [PATCH 376/411] ASoC: Don't try to register debugfs entries if the parent does not exist If the registration of a debugfs directory fails this is treated as a non-fatal error in ASoC and operation continues as normal. This means we need to be careful and check if the parent debugfs directory exists if we try to register a debugfs file or sub-directory. Otherwise we might end up passing NULL for the parent and the file or directory will be registered in the top-level debugfs directory. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 66 ++++++++++++++++++++++++++++++-------------- sound/soc/soc-dapm.c | 3 ++ sound/soc/soc-pcm.c | 8 ++++-- 3 files changed, 54 insertions(+), 23 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d29c68ad83c728..9dfa2e2418650e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -292,6 +292,9 @@ static const struct file_operations codec_reg_fops = { static void soc_init_component_debugfs(struct snd_soc_component *component) { + if (!component->card->debugfs_card_root) + return; + if (component->debugfs_prefix) { char *name; @@ -455,6 +458,9 @@ static const struct file_operations platform_list_fops = { static void soc_init_card_debugfs(struct snd_soc_card *card) { + if (!snd_soc_debugfs_root) + return; + card->debugfs_card_root = debugfs_create_dir(card->name, snd_soc_debugfs_root); if (!card->debugfs_card_root) { @@ -476,6 +482,34 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card) debugfs_remove_recursive(card->debugfs_card_root); } + +static void snd_soc_debugfs_init(void) +{ + snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); + if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) { + pr_warn("ASoC: Failed to create debugfs directory\n"); + snd_soc_debugfs_root = NULL; + return; + } + + if (!debugfs_create_file("codecs", 0444, snd_soc_debugfs_root, NULL, + &codec_list_fops)) + pr_warn("ASoC: Failed to create CODEC list debugfs file\n"); + + if (!debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL, + &dai_list_fops)) + pr_warn("ASoC: Failed to create DAI list debugfs file\n"); + + if (!debugfs_create_file("platforms", 0444, snd_soc_debugfs_root, NULL, + &platform_list_fops)) + pr_warn("ASoC: Failed to create platform list debugfs file\n"); +} + +static void snd_soc_debugfs_exit(void) +{ + debugfs_remove_recursive(snd_soc_debugfs_root); +} + #else #define soc_init_codec_debugfs NULL @@ -497,6 +531,15 @@ static inline void soc_init_card_debugfs(struct snd_soc_card *card) static inline void soc_cleanup_card_debugfs(struct snd_soc_card *card) { } + +static inline void snd_soc_debugfs_init(void) +{ +} + +static inline void snd_soc_debugfs_exit(void) +{ +} + #endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, @@ -3580,26 +3623,7 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_link_codecs); static int __init snd_soc_init(void) { -#ifdef CONFIG_DEBUG_FS - snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); - if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) { - pr_warn("ASoC: Failed to create debugfs directory\n"); - snd_soc_debugfs_root = NULL; - } - - if (!debugfs_create_file("codecs", 0444, snd_soc_debugfs_root, NULL, - &codec_list_fops)) - pr_warn("ASoC: Failed to create CODEC list debugfs file\n"); - - if (!debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL, - &dai_list_fops)) - pr_warn("ASoC: Failed to create DAI list debugfs file\n"); - - if (!debugfs_create_file("platforms", 0444, snd_soc_debugfs_root, NULL, - &platform_list_fops)) - pr_warn("ASoC: Failed to create platform list debugfs file\n"); -#endif - + snd_soc_debugfs_init(); snd_soc_util_init(); return platform_driver_register(&soc_driver); @@ -3609,9 +3633,9 @@ module_init(snd_soc_init); static void __exit snd_soc_exit(void) { snd_soc_util_exit(); + snd_soc_debugfs_exit(); #ifdef CONFIG_DEBUG_FS - debugfs_remove_recursive(snd_soc_debugfs_root); #endif platform_driver_unregister(&soc_driver); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6f88202b8c99a..1fd2d458824edd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1898,6 +1898,9 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, { struct dentry *d; + if (!parent) + return; + dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); if (!dapm->debugfs_dapm) { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9c514fde71542f..b0d61e6531e615 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1097,8 +1097,9 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, stream ? "<-" : "->", be->dai_link->name); #ifdef CONFIG_DEBUG_FS - dpcm->debugfs_state = debugfs_create_u32(be->dai_link->name, 0644, - fe->debugfs_dpcm_root, &dpcm->state); + if (fe->debugfs_dpcm_root) + dpcm->debugfs_state = debugfs_create_u32(be->dai_link->name, 0644, + fe->debugfs_dpcm_root, &dpcm->state); #endif return 1; } @@ -2807,6 +2808,9 @@ void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) if (!rtd->dai_link) return; + if (!rtd->card->debugfs_card_root) + return; + rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, rtd->card->debugfs_card_root); if (!rtd->debugfs_dpcm_root) { From cfd3113e7cc358b68bc13a2ca46714b0445dca56 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Apr 2015 15:23:59 +0200 Subject: [PATCH 377/411] ALSA: hda/via - Add missing stream_pm ops setup Similar like the case for Realtek, VIA codec driver needs this ops as well for making the widget power-save working. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 742087ef378fd5..31a95cca015d4d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -472,6 +472,7 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = snd_hda_jack_unsol_event, + .stream_pm = snd_hda_gen_stream_pm, #ifdef CONFIG_PM .suspend = via_suspend, .check_power_status = via_check_power_status, From 39d118677baa531cd9ee4c025a34f243746a3d18 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 10 Apr 2015 08:43:00 +0900 Subject: [PATCH 378/411] ALSA: ctl: evaluate macro instead of numerical value SNDRV_CTL_TLV_OP_XXX is defined but not used in core code. Instead, raw numerical value is evaluated. This commit replaces these values to these macros for better looking. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index d677c27746e989..00fcaa0ca647e3 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1099,7 +1099,7 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol, int change = 0; void *new_data; - if (op_flag > 0) { + if (op_flag == SNDRV_CTL_TLV_OP_WRITE) { if (size > 1024 * 128) /* sane value */ return -EINVAL; @@ -1381,9 +1381,12 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, goto __kctl_end; } vd = &kctl->vd[tlv.numid - kctl->id.numid]; - if ((op_flag == 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) == 0) || - (op_flag > 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) == 0) || - (op_flag < 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) == 0)) { + if ((op_flag == SNDRV_CTL_TLV_OP_READ && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) == 0) || + (op_flag == SNDRV_CTL_TLV_OP_WRITE && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) == 0) || + (op_flag == SNDRV_CTL_TLV_OP_CMD && + (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) == 0)) { err = -ENXIO; goto __kctl_end; } @@ -1400,7 +1403,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, return 0; } } else { - if (op_flag) { + if (op_flag != SNDRV_CTL_ELEM_ACCESS_TLV_READ) { err = -ENXIO; goto __kctl_end; } From f4d3129c2b580ad2b82aa3211a6216bbf7ad007c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Apr 2015 09:53:04 +0200 Subject: [PATCH 379/411] ALSA: hda/realtek - Fix a typo A comma was used instead of a semicolon, which may lead to a build error. Fixes: cffd39668177 ('ALSA: hda/realtek - Fix the regression by widget power-saving') Reported-by: kbuild test robot Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index acccedf8a4be3c..12f62af507d94f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5673,7 +5673,7 @@ static int patch_alc269(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; - codec->patch_ops.stream_pm = snd_hda_gen_stream_pm, + codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; #ifdef CONFIG_PM codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; From c01673e0b7236140eb5bbe1c7b30aa262f142d7e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 10 Apr 2015 08:46:22 +0000 Subject: [PATCH 380/411] ASoC: ak4642: fixup channels_min ak4642 doesn't have Mono record, ak4643 have it, but not supported. This patch fixes channel mismatch Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index dde8b49c19add3..fba80f30de4de2 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -468,13 +468,13 @@ static struct snd_soc_dai_driver ak4642_dai = { .name = "ak4642-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, From 299e7e97cc33d2d8894250ae2a3101bfb5670141 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 9 Apr 2015 14:56:41 -0300 Subject: [PATCH 381/411] ASoC: fsl_ssi: Use devm_snd_soc_register_component() Using devm_snd_soc_register_component() can make the code shorter and cleaner. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2595611e8a6ded..4201bfe2e9b9e4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1390,8 +1390,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, - &ssi_private->cpu_dai_drv, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component, + &ssi_private->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); goto error_asoc_register; @@ -1404,13 +1404,13 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); - goto error_irq; + goto error_asoc_register; } } ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); if (ret) - goto error_irq; + goto error_asoc_register; /* * If codec-handle property is missing from SSI node, we assume @@ -1451,9 +1451,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); -error_irq: - snd_soc_unregister_component(&pdev->dev); - error_asoc_register: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); @@ -1469,7 +1466,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->pdev) platform_device_unregister(ssi_private->pdev); - snd_soc_unregister_component(&pdev->dev); if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); From 91bf0c2dcb935a87e5c0795f5047456b965fd143 Mon Sep 17 00:00:00 2001 From: Michael Gernoth Date: Thu, 9 Apr 2015 23:42:15 +0200 Subject: [PATCH 382/411] ALSA: emu10k1: don't deadlock in proc-functions The functions snd_emu10k1_proc_spdif_read and snd_emu1010_fpga_read acquire the emu_lock before accessing the FPGA. The function used to access the FPGA (snd_emu1010_fpga_read) also tries to take the emu_lock which causes a deadlock. Remove the outer locking in the proc-functions (guarding only the already safe fpga read) to prevent this deadlock. [removed superfluous flags variables too -- tiwai] Signed-off-by: Michael Gernoth Cc: Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emuproc.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 2ca9f2e93139e1..53745f4c2bf59f 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -241,31 +241,22 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, struct snd_emu10k1 *emu = entry->private_data; u32 value; u32 value2; - unsigned long flags; u32 rate; if (emu->card_capabilities->emu_model) { - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x38, &value); - spin_unlock_irqrestore(&emu->emu_lock, flags); if ((value & 0x1) == 0) { - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x2a, &value); snd_emu1010_fpga_read(emu, 0x2b, &value2); - spin_unlock_irqrestore(&emu->emu_lock, flags); rate = 0x1770000 / (((value << 5) | value2)+1); snd_iprintf(buffer, "ADAT Locked : %u\n", rate); } else { snd_iprintf(buffer, "ADAT Unlocked\n"); } - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x20, &value); - spin_unlock_irqrestore(&emu->emu_lock, flags); if ((value & 0x4) == 0) { - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x28, &value); snd_emu1010_fpga_read(emu, 0x29, &value2); - spin_unlock_irqrestore(&emu->emu_lock, flags); rate = 0x1770000 / (((value << 5) | value2)+1); snd_iprintf(buffer, "SPDIF Locked : %d\n", rate); } else { @@ -410,14 +401,11 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry, { struct snd_emu10k1 *emu = entry->private_data; u32 value; - unsigned long flags; int i; snd_iprintf(buffer, "EMU1010 Registers:\n\n"); for(i = 0; i < 0x40; i+=1) { - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, i, &value); - spin_unlock_irqrestore(&emu->emu_lock, flags); snd_iprintf(buffer, "%02X: %08X, %02X\n", i, value, (value >> 8) & 0x7f); } } From ca2641891d8f0503f166502d168690c1e7d38e49 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 10 Apr 2015 07:12:29 -0300 Subject: [PATCH 383/411] ASoC: fsl_ssi: Use devm_ioremap_resource() Using platform_get_resource() and devm_ioremap_resource() can make the code a bit simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4201bfe2e9b9e4..4f643c45068fd2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1285,7 +1285,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) const struct of_device_id *of_id; const char *p, *sprop; const uint32_t *iprop; - struct resource res; + struct resource *res; void __iomem *iomem; char name[64]; @@ -1332,19 +1332,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->cpu_dai_drv.name = dev_name(&pdev->dev); - /* Get the addresses and IRQ */ - ret = of_address_to_resource(np, 0, &res); - if (ret) { - dev_err(&pdev->dev, "could not determine device resources\n"); - return ret; - } - ssi_private->ssi_phys = res.start; - - iomem = devm_ioremap(&pdev->dev, res.start, resource_size(&res)); - if (!iomem) { - dev_err(&pdev->dev, "could not map device resources\n"); - return -ENOMEM; - } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + iomem = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(iomem)) + return PTR_ERR(iomem); + ssi_private->ssi_phys = res->start; ret = of_property_match_string(np, "clock-names", "ipg"); if (ret < 0) { From a5053a8e200e865ab786384df3f985a3cbb346fe Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 10 Apr 2015 09:47:00 +0000 Subject: [PATCH 384/411] ASoC: core: call snd_soc_runtime_set_dai_fmt() before soc_new_pcm() Current snd_soc_runtime_set_dai_fmt() is called after soc_probe_link_dais(). this means snd_soc_dai_set_fmt() will be called after soc_new_pcm(). Before appling 1efb53a220b78fdfdbb97b726a2156713e75bdab (ASoC: simple-card: Remove support for setting differing DAI formats) simple-card user had (1) snd_soc_dai_set_fmt() -> soc_new_pcm(), but, after that it is (2) soc_new_pcm() -> snd_soc_dai_set_fmt(). At least rsnd driver is assuming (1) pattern. This patch move snd_soc_dai_set_fmt() into soc_probe_link_dais() after the dai_link->init section to solve this issue. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 25fcd80cb108e5..2fb3bf738b5b23 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1359,6 +1359,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } } + if (dai_link->dai_fmt) + snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); + ret = soc_post_component_init(rtd, dai_link->name); if (ret) return ret; @@ -1672,12 +1675,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, card->num_of_dapm_routes); - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].dai_fmt) - snd_soc_runtime_set_dai_fmt(&card->rtd[i], - card->dai_link[i].dai_fmt); - } - snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), From 1169006b054ed98f6056b67fd7f18231b65794a0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 9 Apr 2015 15:32:09 -0300 Subject: [PATCH 385/411] ASoC: fsl: Add the audio interface acronyms in Kconfig text To keep consistency with the other Kconfig entries, use the audio interface acronyms (SSI and SPDIF) in the Kconfig menu text. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 081e406b3713aa..19c302b0d76397 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,7 +24,7 @@ config SND_SOC_FSL_SAI in-tree drivers select it automatically. config SND_SOC_FSL_SSI - tristate "Synchronous Serial Interface module support" + tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO @@ -35,7 +35,7 @@ config SND_SOC_FSL_SSI in-tree drivers select it automatically. config SND_SOC_FSL_SPDIF - tristate "Sony/Philips Digital Interface module support" + tristate "Sony/Philips Digital Interface (S/PDIF) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) From a33c1ec5cf82efb76f0e7339b13f11cfb53a2a2f Mon Sep 17 00:00:00 2001 From: Jin Yao Date: Tue, 7 Apr 2015 09:33:30 +0800 Subject: [PATCH 386/411] ASoC: Intel: Refactor common IPC/mailbox code into generic APIs Currently in Intel SST driver, some similar IPC/mailbox processing code are used in different platforms (e.g. in baytrail/broadwell). This patch extracts the common code and creates new files (sst-ipc.c/sst-ipc.h) to contain the common code and provide the generic APIs for IPC/mailbox processing. Signed-off-by: Jin Yao Acked-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/sst-ipc.c | 294 +++++++++++++++++++++++++++++++ sound/soc/intel/common/sst-ipc.h | 91 ++++++++++ 3 files changed, 387 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/common/sst-ipc.c create mode 100644 sound/soc/intel/common/sst-ipc.h diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 3df0e1ca76c042..f24154ca4e983e 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,6 +1,7 @@ snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o snd-soc-sst-acpi-objs := sst-acpi.o +snd-soc-sst-ipc-objs := sst-ipc.o -obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o +obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c new file mode 100644 index 00000000000000..4b62a553823ccd --- /dev/null +++ b/sound/soc/intel/common/sst-ipc.c @@ -0,0 +1,294 @@ +/* + * Intel SST generic IPC Support + * + * Copyright (C) 2015, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" +#include "sst-ipc.h" + +/* IPC message timeout (msecs) */ +#define IPC_TIMEOUT_MSECS 300 + +#define IPC_EMPTY_LIST_SIZE 8 + +/* locks held by caller */ +static struct ipc_message *msg_get_empty(struct sst_generic_ipc *ipc) +{ + struct ipc_message *msg = NULL; + + if (!list_empty(&ipc->empty_list)) { + msg = list_first_entry(&ipc->empty_list, struct ipc_message, + list); + list_del(&msg->list); + } + + return msg; +} + +static int tx_wait_done(struct sst_generic_ipc *ipc, + struct ipc_message *msg, void *rx_data) +{ + unsigned long flags; + int ret; + + /* wait for DSP completion (in all cases atm inc pending) */ + ret = wait_event_timeout(msg->waitq, msg->complete, + msecs_to_jiffies(IPC_TIMEOUT_MSECS)); + + spin_lock_irqsave(&ipc->dsp->spinlock, flags); + if (ret == 0) { + if (ipc->ops.shim_dbg != NULL) + ipc->ops.shim_dbg(ipc, "message timeout"); + + list_del(&msg->list); + ret = -ETIMEDOUT; + } else { + + /* copy the data returned from DSP */ + if (msg->rx_size) + memcpy(rx_data, msg->rx_data, msg->rx_size); + ret = msg->errno; + } + + list_add_tail(&msg->list, &ipc->empty_list); + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); + return ret; +} + +static int ipc_tx_message(struct sst_generic_ipc *ipc, u64 header, + void *tx_data, size_t tx_bytes, void *rx_data, + size_t rx_bytes, int wait) +{ + struct ipc_message *msg; + unsigned long flags; + + spin_lock_irqsave(&ipc->dsp->spinlock, flags); + + msg = msg_get_empty(ipc); + if (msg == NULL) { + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); + return -EBUSY; + } + + msg->header = header; + msg->tx_size = tx_bytes; + msg->rx_size = rx_bytes; + msg->wait = wait; + msg->errno = 0; + msg->pending = false; + msg->complete = false; + + if ((tx_bytes) && (ipc->ops.tx_data_copy != NULL)) + ipc->ops.tx_data_copy(msg, tx_data, tx_bytes); + + list_add_tail(&msg->list, &ipc->tx_list); + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); + + queue_kthread_work(&ipc->kworker, &ipc->kwork); + + if (wait) + return tx_wait_done(ipc, msg, rx_data); + else + return 0; +} + +static int msg_empty_list_init(struct sst_generic_ipc *ipc) +{ + int i; + + ipc->msg = kzalloc(sizeof(struct ipc_message) * + IPC_EMPTY_LIST_SIZE, GFP_KERNEL); + if (ipc->msg == NULL) + return -ENOMEM; + + for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + init_waitqueue_head(&ipc->msg[i].waitq); + list_add(&ipc->msg[i].list, &ipc->empty_list); + } + + return 0; +} + +static void ipc_tx_msgs(struct kthread_work *work) +{ + struct sst_generic_ipc *ipc = + container_of(work, struct sst_generic_ipc, kwork); + struct ipc_message *msg; + unsigned long flags; + u64 ipcx; + + spin_lock_irqsave(&ipc->dsp->spinlock, flags); + + if (list_empty(&ipc->tx_list) || ipc->pending) { + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); + return; + } + + /* if the DSP is busy, we will TX messages after IRQ. + * also postpone if we are in the middle of procesing completion irq*/ + ipcx = sst_dsp_shim_read_unlocked(ipc->dsp, SST_IPCX); + if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); + return; + } + + msg = list_first_entry(&ipc->tx_list, struct ipc_message, list); + list_move(&msg->list, &ipc->rx_list); + + if (ipc->ops.tx_msg != NULL) + ipc->ops.tx_msg(ipc, msg); + + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); +} + +int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc, u64 header, + void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes) +{ + return ipc_tx_message(ipc, header, tx_data, tx_bytes, + rx_data, rx_bytes, 1); +} +EXPORT_SYMBOL_GPL(sst_ipc_tx_message_wait); + +int sst_ipc_tx_message_nowait(struct sst_generic_ipc *ipc, u64 header, + void *tx_data, size_t tx_bytes) +{ + return ipc_tx_message(ipc, header, tx_data, tx_bytes, + NULL, 0, 0); +} +EXPORT_SYMBOL_GPL(sst_ipc_tx_message_nowait); + +struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, + u64 header) +{ + struct ipc_message *msg; + u64 mask; + + if (ipc->ops.reply_msg_match != NULL) + header = ipc->ops.reply_msg_match(header, &mask); + + if (list_empty(&ipc->rx_list)) { + dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n", + header); + return NULL; + } + + list_for_each_entry(msg, &ipc->rx_list, list) { + if ((msg->header & mask) == header) + return msg; + } + + return NULL; +} +EXPORT_SYMBOL_GPL(sst_ipc_reply_find_msg); + +/* locks held by caller */ +void sst_ipc_tx_msg_reply_complete(struct sst_generic_ipc *ipc, + struct ipc_message *msg) +{ + msg->complete = true; + + if (!msg->wait) + list_add_tail(&msg->list, &ipc->empty_list); + else + wake_up(&msg->waitq); +} +EXPORT_SYMBOL_GPL(sst_ipc_tx_msg_reply_complete); + +void sst_ipc_drop_all(struct sst_generic_ipc *ipc) +{ + struct ipc_message *msg, *tmp; + unsigned long flags; + int tx_drop_cnt = 0, rx_drop_cnt = 0; + + /* drop all TX and Rx messages before we stall + reset DSP */ + spin_lock_irqsave(&ipc->dsp->spinlock, flags); + + list_for_each_entry_safe(msg, tmp, &ipc->tx_list, list) { + list_move(&msg->list, &ipc->empty_list); + tx_drop_cnt++; + } + + list_for_each_entry_safe(msg, tmp, &ipc->rx_list, list) { + list_move(&msg->list, &ipc->empty_list); + rx_drop_cnt++; + } + + spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); + + if (tx_drop_cnt || rx_drop_cnt) + dev_err(ipc->dev, "dropped IPC msg RX=%d, TX=%d\n", + tx_drop_cnt, rx_drop_cnt); +} +EXPORT_SYMBOL_GPL(sst_ipc_drop_all); + +int sst_ipc_init(struct sst_generic_ipc *ipc) +{ + int ret; + + INIT_LIST_HEAD(&ipc->tx_list); + INIT_LIST_HEAD(&ipc->rx_list); + INIT_LIST_HEAD(&ipc->empty_list); + init_waitqueue_head(&ipc->wait_txq); + + ret = msg_empty_list_init(ipc); + if (ret < 0) + return -ENOMEM; + + /* start the IPC message thread */ + init_kthread_worker(&ipc->kworker); + ipc->tx_thread = kthread_run(kthread_worker_fn, + &ipc->kworker, "%s", + dev_name(ipc->dev)); + if (IS_ERR(ipc->tx_thread)) { + dev_err(ipc->dev, "error: failed to create message TX task\n"); + ret = PTR_ERR(ipc->tx_thread); + kfree(ipc->msg); + return ret; + } + + init_kthread_work(&ipc->kwork, ipc_tx_msgs); + return 0; +} +EXPORT_SYMBOL_GPL(sst_ipc_init); + +void sst_ipc_fini(struct sst_generic_ipc *ipc) +{ + if (ipc->tx_thread) + kthread_stop(ipc->tx_thread); + + if (ipc->msg) + kfree(ipc->msg); +} +EXPORT_SYMBOL_GPL(sst_ipc_fini); + +/* Module information */ +MODULE_AUTHOR("Jin Yao"); +MODULE_DESCRIPTION("Intel SST IPC generic"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h new file mode 100644 index 00000000000000..125ea451a37369 --- /dev/null +++ b/sound/soc/intel/common/sst-ipc.h @@ -0,0 +1,91 @@ +/* + * Intel SST generic IPC Support + * + * Copyright (C) 2015, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SST_GENERIC_IPC_H +#define __SST_GENERIC_IPC_H + +#include +#include +#include +#include +#include +#include +#include + +#define IPC_MAX_MAILBOX_BYTES 256 + +struct ipc_message { + struct list_head list; + u64 header; + + /* direction wrt host CPU */ + char tx_data[IPC_MAX_MAILBOX_BYTES]; + size_t tx_size; + char rx_data[IPC_MAX_MAILBOX_BYTES]; + size_t rx_size; + + wait_queue_head_t waitq; + bool pending; + bool complete; + bool wait; + int errno; +}; + +struct sst_generic_ipc; + +struct sst_plat_ipc_ops { + void (*tx_msg)(struct sst_generic_ipc *, struct ipc_message *); + void (*shim_dbg)(struct sst_generic_ipc *, const char *); + void (*tx_data_copy)(struct ipc_message *, char *, size_t); + u64 (*reply_msg_match)(u64 header, u64 *mask); +}; + +/* SST generic IPC data */ +struct sst_generic_ipc { + struct device *dev; + struct sst_dsp *dsp; + + /* IPC messaging */ + struct list_head tx_list; + struct list_head rx_list; + struct list_head empty_list; + wait_queue_head_t wait_txq; + struct task_struct *tx_thread; + struct kthread_worker kworker; + struct kthread_work kwork; + bool pending; + struct ipc_message *msg; + + struct sst_plat_ipc_ops ops; +}; + +int sst_ipc_tx_message_wait(struct sst_generic_ipc *ipc, u64 header, + void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes); + +int sst_ipc_tx_message_nowait(struct sst_generic_ipc *ipc, u64 header, + void *tx_data, size_t tx_bytes); + +struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, + u64 header); + +void sst_ipc_tx_msg_reply_complete(struct sst_generic_ipc *ipc, + struct ipc_message *msg); + +void sst_ipc_drop_all(struct sst_generic_ipc *ipc); +int sst_ipc_init(struct sst_generic_ipc *ipc); +void sst_ipc_fini(struct sst_generic_ipc *ipc); + +#endif From 48cec59b6f383c63b2b828b93656ee2030abecc0 Mon Sep 17 00:00:00 2001 From: Jin Yao Date: Tue, 7 Apr 2015 09:33:31 +0800 Subject: [PATCH 387/411] ASoC: Intel: Use the generic IPC/mailbox APIs in Baytrail Use the generic IPC/mailbox APIs to replace the original processing code for Baytrail platform. Signed-off-by: Jin Yao Acked-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 360 +++++--------------- 1 file changed, 77 insertions(+), 283 deletions(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index aabb9b0f48b8c4..1efb33b36303ea 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -31,6 +31,7 @@ #include "sst-baytrail-ipc.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "../common/sst-ipc.h" /* IPC message timeout */ #define IPC_TIMEOUT_MSECS 300 @@ -142,23 +143,6 @@ struct sst_byt_fw_init { u8 debug_info; } __packed; -/* driver internal IPC message structure */ -struct ipc_message { - struct list_head list; - u64 header; - - /* direction wrt host CPU */ - char tx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE]; - size_t tx_size; - char rx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE]; - size_t rx_size; - - wait_queue_head_t waitq; - bool complete; - bool wait; - int errno; -}; - struct sst_byt_stream; struct sst_byt; @@ -195,14 +179,7 @@ struct sst_byt { struct sst_fw *fw; /* IPC messaging */ - struct list_head tx_list; - struct list_head rx_list; - struct list_head empty_list; - wait_queue_head_t wait_txq; - struct task_struct *tx_thread; - struct kthread_worker kworker; - struct kthread_work kwork; - struct ipc_message *msg; + struct sst_generic_ipc ipc; }; static inline u64 sst_byt_header(int msg_id, int data, bool large, int str_id) @@ -246,209 +223,6 @@ static struct sst_byt_stream *sst_byt_get_stream(struct sst_byt *byt, return NULL; } -static void sst_byt_ipc_shim_dbg(struct sst_byt *byt, const char *text) -{ - struct sst_dsp *sst = byt->dsp; - u64 isr, ipcd, imrx, ipcx; - - ipcx = sst_dsp_shim_read64_unlocked(sst, SST_IPCX); - isr = sst_dsp_shim_read64_unlocked(sst, SST_ISRX); - ipcd = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); - imrx = sst_dsp_shim_read64_unlocked(sst, SST_IMRX); - - dev_err(byt->dev, - "ipc: --%s-- ipcx 0x%llx isr 0x%llx ipcd 0x%llx imrx 0x%llx\n", - text, ipcx, isr, ipcd, imrx); -} - -/* locks held by caller */ -static struct ipc_message *sst_byt_msg_get_empty(struct sst_byt *byt) -{ - struct ipc_message *msg = NULL; - - if (!list_empty(&byt->empty_list)) { - msg = list_first_entry(&byt->empty_list, - struct ipc_message, list); - list_del(&msg->list); - } - - return msg; -} - -static void sst_byt_ipc_tx_msgs(struct kthread_work *work) -{ - struct sst_byt *byt = - container_of(work, struct sst_byt, kwork); - struct ipc_message *msg; - u64 ipcx; - unsigned long flags; - - spin_lock_irqsave(&byt->dsp->spinlock, flags); - if (list_empty(&byt->tx_list)) { - spin_unlock_irqrestore(&byt->dsp->spinlock, flags); - return; - } - - /* if the DSP is busy we will TX messages after IRQ */ - ipcx = sst_dsp_shim_read64_unlocked(byt->dsp, SST_IPCX); - if (ipcx & SST_BYT_IPCX_BUSY) { - spin_unlock_irqrestore(&byt->dsp->spinlock, flags); - return; - } - - msg = list_first_entry(&byt->tx_list, struct ipc_message, list); - - list_move(&msg->list, &byt->rx_list); - - /* send the message */ - if (msg->header & IPC_HEADER_LARGE(true)) - sst_dsp_outbox_write(byt->dsp, msg->tx_data, msg->tx_size); - sst_dsp_shim_write64_unlocked(byt->dsp, SST_IPCX, msg->header); - - spin_unlock_irqrestore(&byt->dsp->spinlock, flags); -} - -static inline void sst_byt_tx_msg_reply_complete(struct sst_byt *byt, - struct ipc_message *msg) -{ - msg->complete = true; - - if (!msg->wait) - list_add_tail(&msg->list, &byt->empty_list); - else - wake_up(&msg->waitq); -} - -static void sst_byt_drop_all(struct sst_byt *byt) -{ - struct ipc_message *msg, *tmp; - unsigned long flags; - - /* drop all TX and Rx messages before we stall + reset DSP */ - spin_lock_irqsave(&byt->dsp->spinlock, flags); - list_for_each_entry_safe(msg, tmp, &byt->tx_list, list) { - list_move(&msg->list, &byt->empty_list); - } - - list_for_each_entry_safe(msg, tmp, &byt->rx_list, list) { - list_move(&msg->list, &byt->empty_list); - } - - spin_unlock_irqrestore(&byt->dsp->spinlock, flags); -} - -static int sst_byt_tx_wait_done(struct sst_byt *byt, struct ipc_message *msg, - void *rx_data) -{ - unsigned long flags; - int ret; - - /* wait for DSP completion */ - ret = wait_event_timeout(msg->waitq, msg->complete, - msecs_to_jiffies(IPC_TIMEOUT_MSECS)); - - spin_lock_irqsave(&byt->dsp->spinlock, flags); - if (ret == 0) { - list_del(&msg->list); - sst_byt_ipc_shim_dbg(byt, "message timeout"); - - ret = -ETIMEDOUT; - } else { - - /* copy the data returned from DSP */ - if (msg->rx_size) - memcpy(rx_data, msg->rx_data, msg->rx_size); - ret = msg->errno; - } - - list_add_tail(&msg->list, &byt->empty_list); - spin_unlock_irqrestore(&byt->dsp->spinlock, flags); - return ret; -} - -static int sst_byt_ipc_tx_message(struct sst_byt *byt, u64 header, - void *tx_data, size_t tx_bytes, - void *rx_data, size_t rx_bytes, int wait) -{ - unsigned long flags; - struct ipc_message *msg; - - spin_lock_irqsave(&byt->dsp->spinlock, flags); - - msg = sst_byt_msg_get_empty(byt); - if (msg == NULL) { - spin_unlock_irqrestore(&byt->dsp->spinlock, flags); - return -EBUSY; - } - - msg->header = header; - msg->tx_size = tx_bytes; - msg->rx_size = rx_bytes; - msg->wait = wait; - msg->errno = 0; - msg->complete = false; - - if (tx_bytes) { - /* msg content = lower 32-bit of the header + data */ - *(u32 *)msg->tx_data = (u32)(header & (u32)-1); - memcpy(msg->tx_data + sizeof(u32), tx_data, tx_bytes); - msg->tx_size += sizeof(u32); - } - - list_add_tail(&msg->list, &byt->tx_list); - spin_unlock_irqrestore(&byt->dsp->spinlock, flags); - - queue_kthread_work(&byt->kworker, &byt->kwork); - - if (wait) - return sst_byt_tx_wait_done(byt, msg, rx_data); - else - return 0; -} - -static inline int sst_byt_ipc_tx_msg_wait(struct sst_byt *byt, u64 header, - void *tx_data, size_t tx_bytes, - void *rx_data, size_t rx_bytes) -{ - return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes, - rx_data, rx_bytes, 1); -} - -static inline int sst_byt_ipc_tx_msg_nowait(struct sst_byt *byt, u64 header, - void *tx_data, size_t tx_bytes) -{ - return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes, - NULL, 0, 0); -} - -static struct ipc_message *sst_byt_reply_find_msg(struct sst_byt *byt, - u64 header) -{ - struct ipc_message *msg = NULL, *_msg; - u64 mask; - - /* match reply to message sent based on msg and stream IDs */ - mask = IPC_HEADER_MSG_ID_MASK | - IPC_HEADER_STR_ID_MASK << IPC_HEADER_STR_ID_SHIFT; - header &= mask; - - if (list_empty(&byt->rx_list)) { - dev_err(byt->dev, - "ipc: rx list is empty but received 0x%llx\n", header); - goto out; - } - - list_for_each_entry(_msg, &byt->rx_list, list) { - if ((_msg->header & mask) == header) { - msg = _msg; - break; - } - } - -out: - return msg; -} - static void sst_byt_stream_update(struct sst_byt *byt, struct ipc_message *msg) { struct sst_byt_stream *stream; @@ -477,7 +251,7 @@ static int sst_byt_process_reply(struct sst_byt *byt, u64 header) { struct ipc_message *msg; - msg = sst_byt_reply_find_msg(byt, header); + msg = sst_ipc_reply_find_msg(&byt->ipc, header); if (msg == NULL) return 1; @@ -491,7 +265,7 @@ static int sst_byt_process_reply(struct sst_byt *byt, u64 header) list_del(&msg->list); /* wake up */ - sst_byt_tx_msg_reply_complete(byt, msg); + sst_ipc_tx_msg_reply_complete(&byt->ipc, msg); return 1; } @@ -538,6 +312,7 @@ static irqreturn_t sst_byt_irq_thread(int irq, void *context) { struct sst_dsp *sst = (struct sst_dsp *) context; struct sst_byt *byt = sst_dsp_get_thread_context(sst); + struct sst_generic_ipc *ipc = &byt->ipc; u64 header; unsigned long flags; @@ -569,7 +344,7 @@ static irqreturn_t sst_byt_irq_thread(int irq, void *context) spin_unlock_irqrestore(&sst->spinlock, flags); /* continue to send any remaining messages... */ - queue_kthread_work(&byt->kworker, &byt->kwork); + queue_kthread_work(&ipc->kworker, &ipc->kwork); return IRQ_HANDLED; } @@ -656,7 +431,8 @@ int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream) header = sst_byt_header(IPC_IA_ALLOC_STREAM, sizeof(*str_req) + sizeof(u32), true, stream->str_id); - ret = sst_byt_ipc_tx_msg_wait(byt, header, str_req, sizeof(*str_req), + ret = sst_ipc_tx_message_wait(&byt->ipc, header, str_req, + sizeof(*str_req), reply, sizeof(*reply)); if (ret < 0) { dev_err(byt->dev, "ipc: error stream commit failed\n"); @@ -679,7 +455,7 @@ int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream) goto out; header = sst_byt_header(IPC_IA_FREE_STREAM, 0, false, stream->str_id); - ret = sst_byt_ipc_tx_msg_wait(byt, header, NULL, 0, NULL, 0); + ret = sst_ipc_tx_message_wait(&byt->ipc, header, NULL, 0, NULL, 0); if (ret < 0) { dev_err(byt->dev, "ipc: free stream %d failed\n", stream->str_id); @@ -703,9 +479,11 @@ static int sst_byt_stream_operations(struct sst_byt *byt, int type, header = sst_byt_header(type, 0, false, stream_id); if (wait) - return sst_byt_ipc_tx_msg_wait(byt, header, NULL, 0, NULL, 0); + return sst_ipc_tx_message_wait(&byt->ipc, header, NULL, + 0, NULL, 0); else - return sst_byt_ipc_tx_msg_nowait(byt, header, NULL, 0); + return sst_ipc_tx_message_nowait(&byt->ipc, header, + NULL, 0); } /* stream ALSA trigger operations */ @@ -725,7 +503,7 @@ int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream, tx_msg = &start_stream; size = sizeof(start_stream); - ret = sst_byt_ipc_tx_msg_nowait(byt, header, tx_msg, size); + ret = sst_ipc_tx_message_nowait(&byt->ipc, header, tx_msg, size); if (ret < 0) dev_err(byt->dev, "ipc: error failed to start stream %d\n", stream->str_id); @@ -790,23 +568,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt, return do_div(fw_tstamp.ring_buffer_counter, buffer_size); } -static int msg_empty_list_init(struct sst_byt *byt) -{ - struct ipc_message *msg; - int i; - - byt->msg = kzalloc(sizeof(*msg) * IPC_EMPTY_LIST_SIZE, GFP_KERNEL); - if (byt->msg == NULL) - return -ENOMEM; - - for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { - init_waitqueue_head(&byt->msg[i].waitq); - list_add(&byt->msg[i].list, &byt->empty_list); - } - - return 0; -} - struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt) { return byt->dsp; @@ -823,7 +584,7 @@ int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) dev_dbg(byt->dev, "dsp reset\n"); sst_dsp_reset(byt->dsp); - sst_byt_drop_all(byt); + sst_ipc_drop_all(&byt->ipc); dev_dbg(byt->dev, "dsp in reset\n"); dev_dbg(byt->dev, "free all blocks and unload fw\n"); @@ -876,9 +637,52 @@ int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata) } EXPORT_SYMBOL_GPL(sst_byt_dsp_wait_for_ready); +static void byt_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg) +{ + if (msg->header & IPC_HEADER_LARGE(true)) + sst_dsp_outbox_write(ipc->dsp, msg->tx_data, msg->tx_size); + + sst_dsp_shim_write64_unlocked(ipc->dsp, SST_IPCX, msg->header); +} + +static void byt_shim_dbg(struct sst_generic_ipc *ipc, const char *text) +{ + struct sst_dsp *sst = ipc->dsp; + u64 isr, ipcd, imrx, ipcx; + + ipcx = sst_dsp_shim_read64_unlocked(sst, SST_IPCX); + isr = sst_dsp_shim_read64_unlocked(sst, SST_ISRX); + ipcd = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); + imrx = sst_dsp_shim_read64_unlocked(sst, SST_IMRX); + + dev_err(ipc->dev, + "ipc: --%s-- ipcx 0x%llx isr 0x%llx ipcd 0x%llx imrx 0x%llx\n", + text, ipcx, isr, ipcd, imrx); +} + +static void byt_tx_data_copy(struct ipc_message *msg, char *tx_data, + size_t tx_size) +{ + /* msg content = lower 32-bit of the header + data */ + *(u32 *)msg->tx_data = (u32)(msg->header & (u32)-1); + memcpy(msg->tx_data + sizeof(u32), tx_data, tx_size); + msg->tx_size += sizeof(u32); +} + +static u64 byt_reply_msg_match(u64 header, u64 *mask) +{ + /* match reply to message sent based on msg and stream IDs */ + *mask = IPC_HEADER_MSG_ID_MASK | + IPC_HEADER_STR_ID_MASK << IPC_HEADER_STR_ID_SHIFT; + header &= *mask; + + return header; +} + int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt; + struct sst_generic_ipc *ipc; struct sst_fw *byt_sst_fw; struct sst_byt_fw_init init; int err; @@ -889,39 +693,30 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) if (byt == NULL) return -ENOMEM; - byt->dev = dev; - INIT_LIST_HEAD(&byt->stream_list); - INIT_LIST_HEAD(&byt->tx_list); - INIT_LIST_HEAD(&byt->rx_list); - INIT_LIST_HEAD(&byt->empty_list); - init_waitqueue_head(&byt->boot_wait); - init_waitqueue_head(&byt->wait_txq); + ipc = &byt->ipc; + ipc->dev = dev; + ipc->ops.tx_msg = byt_tx_msg; + ipc->ops.shim_dbg = byt_shim_dbg; + ipc->ops.tx_data_copy = byt_tx_data_copy; + ipc->ops.reply_msg_match = byt_reply_msg_match; - err = msg_empty_list_init(byt); - if (err < 0) - return -ENOMEM; - - /* start the IPC message thread */ - init_kthread_worker(&byt->kworker); - byt->tx_thread = kthread_run(kthread_worker_fn, - &byt->kworker, "%s", - dev_name(byt->dev)); - if (IS_ERR(byt->tx_thread)) { - err = PTR_ERR(byt->tx_thread); - dev_err(byt->dev, "error failed to create message TX task\n"); - goto err_free_msg; - } - init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs); + err = sst_ipc_init(ipc); + if (err != 0) + goto ipc_init_err; + INIT_LIST_HEAD(&byt->stream_list); + init_waitqueue_head(&byt->boot_wait); byt_dev.thread_context = byt; /* init SST shim */ byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); if (byt->dsp == NULL) { err = -ENODEV; - goto dsp_err; + goto dsp_new_err; } + ipc->dsp = byt->dsp; + /* keep the DSP in reset state for base FW loading */ sst_dsp_reset(byt->dsp); @@ -961,10 +756,10 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) sst_fw_free(byt_sst_fw); fw_err: sst_dsp_free(byt->dsp); -dsp_err: - kthread_stop(byt->tx_thread); -err_free_msg: - kfree(byt->msg); +dsp_new_err: + sst_ipc_fini(ipc); +ipc_init_err: + kfree(byt); return err; } @@ -977,7 +772,6 @@ void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata) sst_dsp_reset(byt->dsp); sst_fw_free_all(byt->dsp); sst_dsp_free(byt->dsp); - kthread_stop(byt->tx_thread); - kfree(byt->msg); + sst_ipc_fini(&byt->ipc); } EXPORT_SYMBOL_GPL(sst_byt_dsp_free); From 0e7921e9583b72be93d8fa82536a7594974b7eea Mon Sep 17 00:00:00 2001 From: Jin Yao Date: Tue, 7 Apr 2015 09:33:32 +0800 Subject: [PATCH 388/411] ASoC: Intel: Use the generic IPC/mailbox APIs in Broadwell Use the generic IPC/mailbox APIs to replace the original processing code for Broadwell platform. Signed-off-by: Jin Yao Acked-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-ipc.c | 382 +++++----------------- 1 file changed, 87 insertions(+), 295 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 28667d8e200506..d75f09eb30b772 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -36,6 +36,7 @@ #include "sst-haswell-ipc.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "../common/sst-ipc.h" /* Global Message - Generic */ #define IPC_GLB_TYPE_SHIFT 24 @@ -210,23 +211,6 @@ struct sst_hsw_ipc_fw_ready { u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; } __attribute__((packed)); -struct ipc_message { - struct list_head list; - u32 header; - - /* direction wrt host CPU */ - char tx_data[IPC_MAX_MAILBOX_BYTES]; - size_t tx_size; - char rx_data[IPC_MAX_MAILBOX_BYTES]; - size_t rx_size; - - wait_queue_head_t waitq; - bool pending; - bool complete; - bool wait; - int errno; -}; - struct sst_hsw_stream; struct sst_hsw; @@ -325,15 +309,7 @@ struct sst_hsw { bool shutdown; /* IPC messaging */ - struct list_head tx_list; - struct list_head rx_list; - struct list_head empty_list; - wait_queue_head_t wait_txq; - struct task_struct *tx_thread; - struct kthread_worker kworker; - struct kthread_work kwork; - bool pending; - struct ipc_message *msg; + struct sst_generic_ipc ipc; /* FW log stream */ struct sst_hsw_log_stream log_stream; @@ -456,159 +432,6 @@ static struct sst_hsw_stream *get_stream_by_id(struct sst_hsw *hsw, return NULL; } -static void ipc_shim_dbg(struct sst_hsw *hsw, const char *text) -{ - struct sst_dsp *sst = hsw->dsp; - u32 isr, ipcd, imrx, ipcx; - - ipcx = sst_dsp_shim_read_unlocked(sst, SST_IPCX); - isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX); - ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD); - imrx = sst_dsp_shim_read_unlocked(sst, SST_IMRX); - - dev_err(hsw->dev, "ipc: --%s-- ipcx 0x%8.8x isr 0x%8.8x ipcd 0x%8.8x imrx 0x%8.8x\n", - text, ipcx, isr, ipcd, imrx); -} - -/* locks held by caller */ -static struct ipc_message *msg_get_empty(struct sst_hsw *hsw) -{ - struct ipc_message *msg = NULL; - - if (!list_empty(&hsw->empty_list)) { - msg = list_first_entry(&hsw->empty_list, struct ipc_message, - list); - list_del(&msg->list); - } - - return msg; -} - -static void ipc_tx_msgs(struct kthread_work *work) -{ - struct sst_hsw *hsw = - container_of(work, struct sst_hsw, kwork); - struct ipc_message *msg; - unsigned long flags; - u32 ipcx; - - spin_lock_irqsave(&hsw->dsp->spinlock, flags); - - if (list_empty(&hsw->tx_list) || hsw->pending) { - spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); - return; - } - - /* if the DSP is busy, we will TX messages after IRQ. - * also postpone if we are in the middle of procesing completion irq*/ - ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX); - if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { - spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); - return; - } - - msg = list_first_entry(&hsw->tx_list, struct ipc_message, list); - - list_move(&msg->list, &hsw->rx_list); - - /* send the message */ - sst_dsp_outbox_write(hsw->dsp, msg->tx_data, msg->tx_size); - sst_dsp_ipc_msg_tx(hsw->dsp, msg->header | SST_IPCX_BUSY); - - spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); -} - -/* locks held by caller */ -static void tx_msg_reply_complete(struct sst_hsw *hsw, struct ipc_message *msg) -{ - msg->complete = true; - trace_ipc_reply("completed", msg->header); - - if (!msg->wait) - list_add_tail(&msg->list, &hsw->empty_list); - else - wake_up(&msg->waitq); -} - -static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg, - void *rx_data) -{ - unsigned long flags; - int ret; - - /* wait for DSP completion (in all cases atm inc pending) */ - ret = wait_event_timeout(msg->waitq, msg->complete, - msecs_to_jiffies(IPC_TIMEOUT_MSECS)); - - spin_lock_irqsave(&hsw->dsp->spinlock, flags); - if (ret == 0) { - ipc_shim_dbg(hsw, "message timeout"); - - trace_ipc_error("error message timeout for", msg->header); - list_del(&msg->list); - ret = -ETIMEDOUT; - } else { - - /* copy the data returned from DSP */ - if (msg->rx_size) - memcpy(rx_data, msg->rx_data, msg->rx_size); - ret = msg->errno; - } - - list_add_tail(&msg->list, &hsw->empty_list); - spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); - return ret; -} - -static int ipc_tx_message(struct sst_hsw *hsw, u32 header, void *tx_data, - size_t tx_bytes, void *rx_data, size_t rx_bytes, int wait) -{ - struct ipc_message *msg; - unsigned long flags; - - spin_lock_irqsave(&hsw->dsp->spinlock, flags); - - msg = msg_get_empty(hsw); - if (msg == NULL) { - spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); - return -EBUSY; - } - - if (tx_bytes) - memcpy(msg->tx_data, tx_data, tx_bytes); - - msg->header = header; - msg->tx_size = tx_bytes; - msg->rx_size = rx_bytes; - msg->wait = wait; - msg->errno = 0; - msg->pending = false; - msg->complete = false; - - list_add_tail(&msg->list, &hsw->tx_list); - spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); - - queue_kthread_work(&hsw->kworker, &hsw->kwork); - - if (wait) - return tx_wait_done(hsw, msg, rx_data); - else - return 0; -} - -static inline int ipc_tx_message_wait(struct sst_hsw *hsw, u32 header, - void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes) -{ - return ipc_tx_message(hsw, header, tx_data, tx_bytes, rx_data, - rx_bytes, 1); -} - -static inline int ipc_tx_message_nowait(struct sst_hsw *hsw, u32 header, - void *tx_data, size_t tx_bytes) -{ - return ipc_tx_message(hsw, header, tx_data, tx_bytes, NULL, 0, 0); -} - static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) { struct sst_hsw_ipc_fw_ready fw_ready; @@ -696,27 +519,6 @@ static void hsw_notification_work(struct work_struct *work) sst_dsp_shim_update_bits(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); } -static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) -{ - struct ipc_message *msg; - - /* clear reply bits & status bits */ - header &= ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK); - - if (list_empty(&hsw->rx_list)) { - dev_err(hsw->dev, "error: rx list empty but received 0x%x\n", - header); - return NULL; - } - - list_for_each_entry(msg, &hsw->rx_list, list) { - if (msg->header == header) - return msg; - } - - return NULL; -} - static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg) { struct sst_hsw_stream *stream; @@ -755,7 +557,7 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) trace_ipc_reply("processing -->", header); - msg = reply_find_msg(hsw, header); + msg = sst_ipc_reply_find_msg(&hsw->ipc, header); if (msg == NULL) { trace_ipc_error("error: can't find message header", header); return -EIO; @@ -766,14 +568,14 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) case IPC_GLB_REPLY_PENDING: trace_ipc_pending_reply("received", header); msg->pending = true; - hsw->pending = true; + hsw->ipc.pending = true; return 1; case IPC_GLB_REPLY_SUCCESS: if (msg->pending) { trace_ipc_pending_reply("completed", header); sst_dsp_inbox_read(hsw->dsp, msg->rx_data, msg->rx_size); - hsw->pending = false; + hsw->ipc.pending = false; } else { /* copy data from the DSP */ sst_dsp_outbox_read(hsw->dsp, msg->rx_data, @@ -829,7 +631,7 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) /* wake up and return the error if we have waiters on this message ? */ list_del(&msg->list); - tx_msg_reply_complete(hsw, msg); + sst_ipc_tx_msg_reply_complete(&hsw->ipc, msg); return 1; } @@ -970,6 +772,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) { struct sst_dsp *sst = (struct sst_dsp *) context; struct sst_hsw *hsw = sst_dsp_get_thread_context(sst); + struct sst_generic_ipc *ipc = &hsw->ipc; u32 ipcx, ipcd; int handled; unsigned long flags; @@ -1016,7 +819,7 @@ static irqreturn_t hsw_irq_thread(int irq, void *context) spin_unlock_irqrestore(&sst->spinlock, flags); /* continue to send any remaining messages... */ - queue_kthread_work(&hsw->kworker, &hsw->kwork); + queue_kthread_work(&ipc->kworker, &ipc->kwork); return IRQ_HANDLED; } @@ -1026,7 +829,8 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, { int ret; - ret = ipc_tx_message_wait(hsw, IPC_GLB_TYPE(IPC_GLB_GET_FW_VERSION), + ret = sst_ipc_tx_message_wait(&hsw->ipc, + IPC_GLB_TYPE(IPC_GLB_GET_FW_VERSION), NULL, 0, version, sizeof(*version)); if (ret < 0) dev_err(hsw->dev, "error: get version failed\n"); @@ -1090,7 +894,8 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, req->channel = channel; } - ret = ipc_tx_message_wait(hsw, header, req, sizeof(*req), NULL, 0); + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, req, + sizeof(*req), NULL, 0); if (ret < 0) { dev_err(hsw->dev, "error: set stream volume failed\n"); return ret; @@ -1155,7 +960,8 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, req.curve_type = hsw->curve_type; req.target_volume = volume; - ret = ipc_tx_message_wait(hsw, header, &req, sizeof(req), NULL, 0); + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &req, + sizeof(req), NULL, 0); if (ret < 0) { dev_err(hsw->dev, "error: set mixer volume failed\n"); return ret; @@ -1213,7 +1019,7 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) stream->free_req.stream_id = stream->reply.stream_hw_id; header = IPC_GLB_TYPE(IPC_GLB_FREE_STREAM); - ret = ipc_tx_message_wait(hsw, header, &stream->free_req, + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &stream->free_req, sizeof(stream->free_req), NULL, 0); if (ret < 0) { dev_err(hsw->dev, "error: free stream %d failed\n", @@ -1405,8 +1211,8 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM); - ret = ipc_tx_message_wait(hsw, header, str_req, sizeof(*str_req), - reply, sizeof(*reply)); + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, str_req, + sizeof(*str_req), reply, sizeof(*reply)); if (ret < 0) { dev_err(hsw->dev, "error: stream commit failed\n"); return ret; @@ -1455,7 +1261,8 @@ int sst_hsw_mixer_get_info(struct sst_hsw *hsw) trace_ipc_request("get global mixer info", 0); - ret = ipc_tx_message_wait(hsw, header, NULL, 0, reply, sizeof(*reply)); + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, NULL, 0, + reply, sizeof(*reply)); if (ret < 0) { dev_err(hsw->dev, "error: get stream info failed\n"); return ret; @@ -1476,9 +1283,10 @@ static int sst_hsw_stream_operations(struct sst_hsw *hsw, int type, header |= (stream_id << IPC_STR_ID_SHIFT); if (wait) - return ipc_tx_message_wait(hsw, header, NULL, 0, NULL, 0); + return sst_ipc_tx_message_wait(&hsw->ipc, header, + NULL, 0, NULL, 0); else - return ipc_tx_message_nowait(hsw, header, NULL, 0); + return sst_ipc_tx_message_nowait(&hsw->ipc, header, NULL, 0); } /* Stream ALSA trigger operations */ @@ -1605,8 +1413,8 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, header = IPC_GLB_TYPE(IPC_GLB_SET_DEVICE_FORMATS); - ret = ipc_tx_message_wait(hsw, header, &config, sizeof(config), - NULL, 0); + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &config, + sizeof(config), NULL, 0); if (ret < 0) dev_err(hsw->dev, "error: set device formats failed\n"); @@ -1626,8 +1434,8 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, trace_ipc_request("PM enter Dx state", state); - ret = ipc_tx_message_wait(hsw, header, &state_, sizeof(state_), - dx, sizeof(*dx)); + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &state_, + sizeof(state_), dx, sizeof(*dx)); if (ret < 0) { dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state); return ret; @@ -1770,32 +1578,6 @@ static int sst_hsw_dx_state_restore(struct sst_hsw *hsw) return 0; } -static void sst_hsw_drop_all(struct sst_hsw *hsw) -{ - struct ipc_message *msg, *tmp; - unsigned long flags; - int tx_drop_cnt = 0, rx_drop_cnt = 0; - - /* drop all TX and Rx messages before we stall + reset DSP */ - spin_lock_irqsave(&hsw->dsp->spinlock, flags); - - list_for_each_entry_safe(msg, tmp, &hsw->tx_list, list) { - list_move(&msg->list, &hsw->empty_list); - tx_drop_cnt++; - } - - list_for_each_entry_safe(msg, tmp, &hsw->rx_list, list) { - list_move(&msg->list, &hsw->empty_list); - rx_drop_cnt++; - } - - spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); - - if (tx_drop_cnt || rx_drop_cnt) - dev_err(hsw->dev, "dropped IPC msg RX=%d, TX=%d\n", - tx_drop_cnt, rx_drop_cnt); -} - int sst_hsw_dsp_load(struct sst_hsw *hsw) { struct sst_dsp *dsp = hsw->dsp; @@ -1875,7 +1657,7 @@ int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw) if (ret < 0) return ret; - sst_hsw_drop_all(hsw); + sst_ipc_drop_all(&hsw->ipc); return 0; } @@ -1933,23 +1715,6 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) } #endif -static int msg_empty_list_init(struct sst_hsw *hsw) -{ - int i; - - hsw->msg = kzalloc(sizeof(struct ipc_message) * - IPC_EMPTY_LIST_SIZE, GFP_KERNEL); - if (hsw->msg == NULL) - return -ENOMEM; - - for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { - init_waitqueue_head(&hsw->msg[i].waitq); - list_add(&hsw->msg[i].list, &hsw->empty_list); - } - - return 0; -} - struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw) { return hsw->dsp; @@ -2184,7 +1949,7 @@ int sst_hsw_module_enable(struct sst_hsw *hsw, config.scratch_mem.size, config.scratch_mem.offset, config.map.module_entries[0].entry_point); - ret = ipc_tx_message_wait(hsw, header, + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, &config, sizeof(config), NULL, 0); if (ret < 0) dev_err(dev, "ipc: module enable failed - %d\n", ret); @@ -2223,7 +1988,7 @@ int sst_hsw_module_disable(struct sst_hsw *hsw, IPC_MODULE_OPERATION(IPC_MODULE_DISABLE) | IPC_MODULE_ID(module_id); - ret = ipc_tx_message_wait(hsw, header, NULL, 0, NULL, 0); + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, NULL, 0, NULL, 0); if (ret < 0) dev_err(dev, "module disable failed - %d\n", ret); else @@ -2277,7 +2042,7 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw, parameter->parameter_id = parameter_id; parameter->data_size = param_size; - ret = ipc_tx_message_wait(hsw, header, + ret = sst_ipc_tx_message_wait(&hsw->ipc, header, parameter, transfer_parameter_size , NULL, 0); if (ret < 0) dev_err(dev, "ipc: module set parameter failed - %d\n", ret); @@ -2296,10 +2061,48 @@ static struct sst_dsp_device hsw_dev = { .ops = &haswell_ops, }; +static void hsw_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg) +{ + /* send the message */ + sst_dsp_outbox_write(ipc->dsp, msg->tx_data, msg->tx_size); + sst_dsp_ipc_msg_tx(ipc->dsp, msg->header); +} + +static void hsw_shim_dbg(struct sst_generic_ipc *ipc, const char *text) +{ + struct sst_dsp *sst = ipc->dsp; + u32 isr, ipcd, imrx, ipcx; + + ipcx = sst_dsp_shim_read_unlocked(sst, SST_IPCX); + isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX); + ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD); + imrx = sst_dsp_shim_read_unlocked(sst, SST_IMRX); + + dev_err(ipc->dev, + "ipc: --%s-- ipcx 0x%8.8x isr 0x%8.8x ipcd 0x%8.8x imrx 0x%8.8x\n", + text, ipcx, isr, ipcd, imrx); +} + +static void hsw_tx_data_copy(struct ipc_message *msg, char *tx_data, + size_t tx_size) +{ + memcpy(msg->tx_data, tx_data, tx_size); +} + +static u64 hsw_reply_msg_match(u64 header, u64 *mask) +{ + /* clear reply bits & status bits */ + header &= ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK); + *mask = (u64)-1; + + return header; +} + int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_hsw_ipc_fw_version version; struct sst_hsw *hsw; + struct sst_generic_ipc *ipc; int ret; dev_dbg(dev, "initialising Audio DSP IPC\n"); @@ -2308,39 +2111,30 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (hsw == NULL) return -ENOMEM; - hsw->dev = dev; - INIT_LIST_HEAD(&hsw->stream_list); - INIT_LIST_HEAD(&hsw->tx_list); - INIT_LIST_HEAD(&hsw->rx_list); - INIT_LIST_HEAD(&hsw->empty_list); - init_waitqueue_head(&hsw->boot_wait); - init_waitqueue_head(&hsw->wait_txq); + ipc = &hsw->ipc; + ipc->dev = dev; + ipc->ops.tx_msg = hsw_tx_msg; + ipc->ops.shim_dbg = hsw_shim_dbg; + ipc->ops.tx_data_copy = hsw_tx_data_copy; + ipc->ops.reply_msg_match = hsw_reply_msg_match; - ret = msg_empty_list_init(hsw); - if (ret < 0) - return -ENOMEM; - - /* start the IPC message thread */ - init_kthread_worker(&hsw->kworker); - hsw->tx_thread = kthread_run(kthread_worker_fn, - &hsw->kworker, "%s", - dev_name(hsw->dev)); - if (IS_ERR(hsw->tx_thread)) { - ret = PTR_ERR(hsw->tx_thread); - dev_err(hsw->dev, "error: failed to create message TX task\n"); - goto err_free_msg; - } - init_kthread_work(&hsw->kwork, ipc_tx_msgs); + ret = sst_ipc_init(ipc); + if (ret != 0) + goto ipc_init_err; + INIT_LIST_HEAD(&hsw->stream_list); + init_waitqueue_head(&hsw->boot_wait); hsw_dev.thread_context = hsw; /* init SST shim */ hsw->dsp = sst_dsp_new(dev, &hsw_dev, pdata); if (hsw->dsp == NULL) { ret = -ENODEV; - goto dsp_err; + goto dsp_new_err; } + ipc->dsp = hsw->dsp; + /* allocate DMA buffer for context storage */ hsw->dx_context = dma_alloc_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, &hsw->dx_context_paddr, GFP_KERNEL); @@ -2404,11 +2198,10 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) hsw->dx_context, hsw->dx_context_paddr); dma_err: sst_dsp_free(hsw->dsp); -dsp_err: - kthread_stop(hsw->tx_thread); -err_free_msg: - kfree(hsw->msg); - +dsp_new_err: + sst_ipc_fini(ipc); +ipc_init_err: + kfree(hsw); return ret; } EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); @@ -2422,7 +2215,6 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); sst_dsp_free(hsw->dsp); - kthread_stop(hsw->tx_thread); - kfree(hsw->msg); + sst_ipc_fini(&hsw->ipc); } EXPORT_SYMBOL_GPL(sst_hsw_dsp_free); From 3e21a19d1d6775591415efd5617375ba42c41bbd Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 11 Apr 2015 13:24:47 +0900 Subject: [PATCH 389/411] ALSA: seq: fill client ID in return value of pool operation The returned value of 'get/seq client pool' operation has zeroed value for its client ID, against requested client ID. This commit fix the bug by filling it with index value of referred client object. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 48287651ac7795..edbdab85fc02f6 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1879,6 +1879,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, if (cptr == NULL) return -ENOENT; memset(&info, 0, sizeof(info)); + info.client = cptr->number; info.output_pool = cptr->pool->size; info.output_room = cptr->pool->room; info.output_free = info.output_pool; From c0278669fb61596cc1a10ab8686d27c37269c37b Mon Sep 17 00:00:00 2001 From: Yves-Alexis Perez Date: Sat, 11 Apr 2015 09:31:35 +0200 Subject: [PATCH 390/411] ALSA: hda - Add dock support for ThinkPad X250 (17aa:2226) This model uses the same dock port as the previous generation. Signed-off-by: Yves-Alexis Perez Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f9d12c0a7e5a34..3ad85c76240041 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5047,6 +5047,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), From fa863b2d5e62e2ea7d86ccfa9a888bd28dd79ebe Mon Sep 17 00:00:00 2001 From: Michael Gernoth Date: Sat, 11 Apr 2015 14:34:44 +0200 Subject: [PATCH 391/411] ALSA: emu10k1: handle dock disconnects When the dock on an E-mu 1010 card is disconnected, all outputs get muted by the hardware. Add logic to detect a disconnect and unmute. Signed-off-by: Michael Gernoth Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 4887299011efcc..54079f5d567395 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -707,6 +707,7 @@ static int emu1010_firmware_thread(void *data) { struct snd_emu10k1 *emu = data; u32 tmp, tmp2, reg; + u32 last_reg = 0; int err; for (;;) { @@ -782,7 +783,15 @@ static int emu1010_firmware_thread(void *data) msleep(10); /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); + } else if (!reg && last_reg) { + /* Audio Dock removed */ + dev_info(emu->card->dev, + "emu1010: Audio Dock detached\n"); + /* Unmute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } + + last_reg = reg; } dev_info(emu->card->dev, "emu1010: firmware thread stopping\n"); return 0; From c78497e010ae62c8abfb33a45d0e0b361218e9bb Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 11 Apr 2015 17:41:02 +0900 Subject: [PATCH 392/411] ALSA: ctl: confirm to return all identical information in 'activate' event When event originator doesn't set numerical ID in identical information, the event data includes no numerical ID, thus userspace applications cannot identify the control just by unique ID in event data. This commit fix this bug so as the event data includes all of identical information. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 00fcaa0ca647e3..90a9e5d9819af8 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -578,6 +578,7 @@ static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file, * * Finds the control instance with the given id, and activate or * inactivate the control together with notification, if changed. + * The given ID data is filled with full information. * * Return: 0 if unchanged, 1 if changed, or a negative error code on failure. */ @@ -607,6 +608,7 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, goto unlock; vd->access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; } + snd_ctl_build_ioff(id, kctl, index_offset); ret = 1; unlock: up_write(&card->controls_rwsem); From c378c3b03c8d6eef2d2600d0279e2c718d6a0a44 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 11 Apr 2015 17:41:03 +0900 Subject: [PATCH 393/411] ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls In operations of SNDRV_CTL_IOCTL_ELEM_INFO, identical information in returned value is cleared. This is not better to userspace application. This commit confirms to return full identical information to the operations. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 90a9e5d9819af8..a750846514dc05 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1040,8 +1040,12 @@ static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct user_element *ue = kcontrol->private_data; + unsigned int offset; + offset = snd_ctl_get_ioff(kcontrol, &uinfo->id); *uinfo = ue->info; + snd_ctl_build_ioff(&uinfo->id, kcontrol, offset); + return 0; } @@ -1051,10 +1055,13 @@ static int snd_ctl_elem_user_enum_info(struct snd_kcontrol *kcontrol, struct user_element *ue = kcontrol->private_data; const char *names; unsigned int item; + unsigned int offset; item = uinfo->value.enumerated.item; + offset = snd_ctl_get_ioff(kcontrol, &uinfo->id); *uinfo = ue->info; + snd_ctl_build_ioff(&uinfo->id, kcontrol, offset); item = min(item, uinfo->value.enumerated.items - 1); uinfo->value.enumerated.item = item; From cab2ed7474bffafd2a68a885e03b85526194abcd Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 11 Apr 2015 17:41:04 +0900 Subject: [PATCH 394/411] ALSA: ctl: fill identical information to return value when adding userspace elements currently some members related identical information are not fiiled in returned parameter of SNDRV_CTL_IOCTL_ELEM_ADD. This is not better for userspace application. This commit copies information to returned value. When failing to copy into userspace, the added elements are going to be removed. Then, no applications can lock these elements between adding and removing because these are already locked. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index a750846514dc05..ccb1ca26a71e96 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1214,6 +1214,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, unsigned int access; long private_size; struct user_element *ue; + unsigned int offset; int err; if (!*info->id.name) @@ -1316,6 +1317,15 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, err = snd_ctl_add(card, kctl); if (err < 0) return err; + offset = snd_ctl_get_ioff(kctl, &info->id); + snd_ctl_build_ioff(&info->id, kctl, offset); + /* + * Here we cannot fill any field for the number of elements added by + * this operation because there're no specific fields. The usage of + * 'owner' field for this purpose may cause any bugs to userspace + * applications because the field originally means PID of a process + * which locks the element. + */ down_write(&card->controls_rwsem); card->user_ctl_count++; @@ -1328,9 +1338,19 @@ static int snd_ctl_elem_add_user(struct snd_ctl_file *file, struct snd_ctl_elem_info __user *_info, int replace) { struct snd_ctl_elem_info info; + int err; + if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; - return snd_ctl_elem_add(file, &info, replace); + err = snd_ctl_elem_add(file, &info, replace); + if (err < 0) + return err; + if (copy_to_user(_info, &info, sizeof(info))) { + snd_ctl_remove_user_ctl(file, &info.id); + return -EFAULT; + } + + return 0; } static int snd_ctl_elem_remove(struct snd_ctl_file *file, From 99dcab46b5a5b470074bec6d8386e2c7807684cf Mon Sep 17 00:00:00 2001 From: Michael Gernoth Date: Sat, 11 Apr 2015 18:00:19 +0200 Subject: [PATCH 395/411] ALSA: emu10k1: add toggles for E-mu 1010 optical ports The optical ports on the E-mu 1010 (and dock) can be configured for ADAT- or S/PDIF-mode, which is currently hardcoded to ADAT. Add two mixer elements to expose this setting. Tested on an E-mu 1010 PCIe with connected Micro Dock. Signed-off-by: Michael Gernoth Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emumixer.c | 118 +++++++++++++++++++++++++++++++++++ 1 file changed, 118 insertions(+) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 1de33025669a11..55e57166256e99 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -806,6 +806,108 @@ static struct snd_kcontrol_new snd_emu1010_internal_clock = .put = snd_emu1010_internal_clock_put }; +static int snd_emu1010_optical_out_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { + "SPDIF", "ADAT" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_emu1010_optical_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->emu1010.optical_out; + return 0; +} + +static int snd_emu1010_optical_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + u32 tmp; + int change = 0; + + val = ucontrol->value.enumerated.item[0]; + /* Limit: uinfo->value.enumerated.items = 2; */ + if (val >= 2) + return -EINVAL; + change = (emu->emu1010.optical_out != val); + if (change) { + emu->emu1010.optical_out = val; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + } + return change; +} + +static struct snd_kcontrol_new snd_emu1010_optical_out = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Optical Output Mode", + .count = 1, + .info = snd_emu1010_optical_out_info, + .get = snd_emu1010_optical_out_get, + .put = snd_emu1010_optical_out_put +}; + +static int snd_emu1010_optical_in_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { + "SPDIF", "ADAT" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_emu1010_optical_in_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->emu1010.optical_in; + return 0; +} + +static int snd_emu1010_optical_in_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + u32 tmp; + int change = 0; + + val = ucontrol->value.enumerated.item[0]; + /* Limit: uinfo->value.enumerated.items = 2; */ + if (val >= 2) + return -EINVAL; + change = (emu->emu1010.optical_in != val); + if (change) { + emu->emu1010.optical_in = val; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + } + return change; +} + +static struct snd_kcontrol_new snd_emu1010_optical_in = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Optical Input Mode", + .count = 1, + .info = snd_emu1010_optical_in_info, + .get = snd_emu1010_optical_in_get, + .put = snd_emu1010_optical_in_put +}; + static int snd_audigy_i2c_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2051,6 +2153,14 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_out, emu)); + if (err < 0) + return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_in, emu)); + if (err < 0) + return err; } else if (emu->card_capabilities->emu_model) { /* all other e-mu cards for now */ @@ -2086,6 +2196,14 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_out, emu)); + if (err < 0) + return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_in, emu)); + if (err < 0) + return err; } if ( emu->card_capabilities->i2c_adc) { From 8616774968f3baf0c49fc6bcca9550cac49034ee Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sat, 11 Apr 2015 00:18:37 +0200 Subject: [PATCH 396/411] ASoC: rnsd: fix build regression without CONFIG_OF The r-car sound driver only works when CONFIG_OF is set, and after a recent change has a compile-time dependency as well: sound/built-in.o: In function `rsnd_dma_request_channel': :(.text+0x9fb84): undefined reference to `of_dma_request_slave_channel' This could be fixed either by adding a static inline wrapper for the function, or by adding a Kconfig dependency. This implements the second approach, which seems appropriate because the driver in fact has a hard dependency. Signed-off-by: Arnd Bergmann Fixes: 72adc61f4637aa3 ("ASoC: rsnd: 1st DMAC dma-names cares subnode") Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 2b30304155734f..07114b0b0dc12b 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" + depends on DMA_OF select SND_SIMPLE_CARD select REGMAP_MMIO help From eef0342cf32689f77d78ee3302999e5caaa6a8f3 Mon Sep 17 00:00:00 2001 From: Adam Honse Date: Sun, 12 Apr 2015 01:03:07 -0500 Subject: [PATCH 397/411] ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate Adds Microsoft LifeCam Cinema USB ID to the snd_usb_get_sample_rate_quirk list as the Lifecam Cinema does not appear to support getting the sample rate. Fixes the issue where the LifeCam Cinema would wait for USB timeout and log the message "cannot get freq at ep 0x82" when accessed. Addresses bug report https://bugzilla.kernel.org/show_bug.cgi?id=95961. Signed-off-by: Adam Honse Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9a28365126f9ab..32631a86078bf4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1115,6 +1115,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) { /* devices which do not support reading the sample rate. */ switch (chip->usb_id) { + case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema */ case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ return true; From c30cf8cbe55413cd643a0bdd3442d75950caa918 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 12 Apr 2015 09:16:11 +0200 Subject: [PATCH 398/411] ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_* The commit [39d118677baa: ALSA: ctl: evaluate macro instead of numerical value] replaced the numbers with constants, but one place was replaced wrongly with a different type. Fixed now. Fixes: 39d118677baa ('ALSA: ctl: evaluate macro instead of numerical value') Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index ccb1ca26a71e96..be5b97cd8dc3c3 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1432,7 +1432,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, return 0; } } else { - if (op_flag != SNDRV_CTL_ELEM_ACCESS_TLV_READ) { + if (op_flag != SNDRV_CTL_TLV_OP_READ) { err = -ENXIO; goto __kctl_end; } From 7ca92b8f5ad80e0d97f093dad81a4e4143c8edeb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Apr 2015 11:16:18 +0200 Subject: [PATCH 399/411] ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary The SND_AT91_SOC_SAM9G20_WM8731 and SND_AT91_SOC_SAM9X5_WM8731 machine driver symbols select SND_SOC_WM8731 which depends on SND_SOC_I2C_AND_SPI. So the machine driver symbols need to depend on SND_SOC_I2C_AND_SPI as well, otherwise we might end up with a invalid configuration, which will sooner or later upset the randconfig auto-builders. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1579e994acf881..2f185e51862b17 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,7 +25,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -45,7 +45,7 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 From ada602b30e070e786caa6e14a2e17100c6bd6998 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Apr 2015 11:16:19 +0200 Subject: [PATCH 400/411] ASoC: atmel: Improve machine driver compile test coverage The Atmel ASoC machine drivers don't have any compile time arch dependencies anymore. Make it possible to select them when COMPILE_TEST is enabled to get better compile test coverage. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 2f185e51862b17..e7d08806f3e92d 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,7 +25,8 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI + depends on ARCH_AT91 || COMPILE_TEST + depends on ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -35,7 +36,8 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_ATMEL_SOC_WM8904 tristate "Atmel ASoC driver for boards using WM8904 codec" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && I2C + depends on ARCH_AT91 || COMPILE_TEST + depends on ATMEL_SSC && SND_ATMEL_SOC && I2C select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8904 @@ -45,7 +47,8 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI + depends on ARCH_AT91 || COMPILE_TEST + depends on ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 From 02f51640fedb61ab17fcddbd6fdb40239a4f46d9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Apr 2015 12:52:01 +0200 Subject: [PATCH 401/411] ASoC: wm1133-ev1: Use card DAPM context to access widgets The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/wm1133-ev1.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 0653aa83c9273c..b454972dce3521 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -202,7 +202,6 @@ static struct snd_soc_jack_pin mic_jack_pins[] = { static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; /* Headphone jack detection */ snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, @@ -216,7 +215,7 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias"); return 0; } From 5cf57f0f6b25d046e6ea219d99681077edca5d7c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Apr 2015 13:01:02 +0200 Subject: [PATCH 402/411] ASoC: mop500_ab8500: Use card DAPM context to access widgets The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/ux500/mop500_ab8500.c | 36 ++++++++++++++++----------------- 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index aa65370db82aa5..b81a7a4c938b80 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -362,7 +362,7 @@ struct snd_soc_ops mop500_ab8500_ops[] = { int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; struct device *dev = rtd->card->dev; struct mop500_ab8500_drvdata *drvdata; int ret; @@ -407,23 +407,23 @@ int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + ret = snd_soc_dapm_disable_pin(dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 6"); return ret; } From b213b44a96ed1f868f68b094dbcf8fc9622984ef Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Apr 2015 13:11:28 +0200 Subject: [PATCH 403/411] ASoC: davinci-evm: Use card DAPM context to access widgets The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b6bb5947a8a843..1f314a836f2a9e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -117,7 +117,6 @@ static const struct snd_soc_dapm_route audio_map[] = { static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct snd_soc_codec *codec = rtd->codec; struct device_node *np = card->dev->of_node; int ret; @@ -136,9 +135,9 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) } /* not connected */ - snd_soc_dapm_nc_pin(&codec->dapm, "MONO_LOUT"); - snd_soc_dapm_nc_pin(&codec->dapm, "HPLCOM"); - snd_soc_dapm_nc_pin(&codec->dapm, "HPRCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&card->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "HPRCOM"); return 0; } From d4bdaced1a81ca2953557f8ecae842b42879fda4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Apr 2015 10:47:57 +0200 Subject: [PATCH 404/411] ASoC: n810: Consistently pass the card DAPM context to n810_ext_control() Some callers of n810_ext_control() pass the card DAPM context and some pass the CODEC DAPM context. Given that some of the widgets that are accessed in the function are in the card's context, always passing it is the obvious choice. Signed-off-by: Lars-Peter Clausen Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 5d7f9cebe04142..617eae37581c2f 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -98,12 +98,11 @@ static int n810_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); - n810_ext_control(&codec->dapm); + n810_ext_control(&rtd->card->dapm); return clk_prepare_enable(sys_clkout2); } From 59c41d15e63063545b1f91f7ad5411c25f6d046d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Apr 2015 10:47:58 +0200 Subject: [PATCH 405/411] ASoC: n810: Automatically disconnect non-connected pins All CODEC input and output widgets are either in the DAPM routing table or manually marked as non-connected. This means the card is fully routed and we can let the core take care of disconnecting non-connected pins. Signed-off-by: Lars-Peter Clausen Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 617eae37581c2f..dcb5336b569815 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -254,24 +254,6 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { n810_get_input, n810_set_input), }; -static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Not connected */ - snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); - snd_soc_dapm_nc_pin(dapm, "HPLCOM"); - snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - snd_soc_dapm_nc_pin(dapm, "MIC3L"); - snd_soc_dapm_nc_pin(dapm, "MIC3R"); - snd_soc_dapm_nc_pin(dapm, "LINE1R"); - snd_soc_dapm_nc_pin(dapm, "LINE2L"); - snd_soc_dapm_nc_pin(dapm, "LINE2R"); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link n810_dai = { .name = "TLV320AIC33", @@ -282,7 +264,6 @@ static struct snd_soc_dai_link n810_dai = { .codec_dai_name = "tlv320aic3x-hifi", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = n810_aic33_init, .ops = &n810_ops, }; @@ -299,6 +280,7 @@ static struct snd_soc_card snd_soc_n810 = { .num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), + .fully_routed = true, }; static struct platform_device *n810_snd_device; From a5e5e12bd4ed5cd1123ace4300b5c07230fbf21e Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Mon, 13 Apr 2015 02:16:21 +0800 Subject: [PATCH 406/411] ASoC: Intel: fix array_size.cocci warnings sound/soc/intel/haswell/sst-haswell-ipc.c:646:28-29: WARNING: Use ARRAY_SIZE Use ARRAY_SIZE instead of dividing sizeof array with sizeof an element Semantic patch information: This makes an effort to find cases where ARRAY_SIZE can be used such as where there is a division of sizeof the array by the sizeof its first element or by any indexed element or the element type. It replaces the division of the two sizeofs by ARRAY_SIZE. Generated by: scripts/coccinelle/misc/array_size.cocci CC: Jie Yang Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index d75f09eb30b772..344a1e9bbce579 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -466,7 +466,7 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) /* log the FW version info got from the mailbox here. */ memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); pinfo = &fw_info[0]; - for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + for (i = 0; i < ARRAY_SIZE(tmp); i++) tmp[i] = strsep(&pinfo, " "); dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " "version: %s.%s, build %s, source commit id: %s\n", From e1c78df1da112f2644058af2425dd5ca3eb1a96a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 12 Apr 2015 10:12:25 +0900 Subject: [PATCH 407/411] ALSA: ctl: fix to handle several elements added by one operation for userspace element An element instance can have several elements with the same feature. Some userspace applications can add such an element instance by add operation with the number of elements. Then, the element instance gets a memory object to keep states of these elements. But the element instance has just one memory object for the elements. This causes the same result to each read/write operations to the different elements. This commit fixes this bug by allocating enough memory objects to the element instance for each of elements. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index be5b97cd8dc3c3..196a6fe100ca8f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1029,7 +1029,7 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, struct user_element { struct snd_ctl_elem_info info; struct snd_card *card; - void *elem_data; /* element data */ + char *elem_data; /* element data */ unsigned long elem_data_size; /* size of element data in bytes */ void *tlv_data; /* TLV data */ unsigned long tlv_data_size; /* TLV data size */ @@ -1078,9 +1078,12 @@ static int snd_ctl_elem_user_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct user_element *ue = kcontrol->private_data; + unsigned int size = ue->elem_data_size; + char *src = ue->elem_data + + snd_ctl_get_ioff(kcontrol, &ucontrol->id) * size; mutex_lock(&ue->card->user_ctl_lock); - memcpy(&ucontrol->value, ue->elem_data, ue->elem_data_size); + memcpy(&ucontrol->value, src, size); mutex_unlock(&ue->card->user_ctl_lock); return 0; } @@ -1090,11 +1093,14 @@ static int snd_ctl_elem_user_put(struct snd_kcontrol *kcontrol, { int change; struct user_element *ue = kcontrol->private_data; + unsigned int size = ue->elem_data_size; + char *dst = ue->elem_data + + snd_ctl_get_ioff(kcontrol, &ucontrol->id) * size; mutex_lock(&ue->card->user_ctl_lock); - change = memcmp(&ucontrol->value, ue->elem_data, ue->elem_data_size) != 0; + change = memcmp(&ucontrol->value, dst, size) != 0; if (change) - memcpy(ue->elem_data, &ucontrol->value, ue->elem_data_size); + memcpy(dst, &ucontrol->value, size); mutex_unlock(&ue->card->user_ctl_lock); return change; } @@ -1278,7 +1284,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (err < 0) return err; memcpy(&kctl->id, &info->id, sizeof(kctl->id)); - kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, + kctl->private_data = kzalloc(sizeof(struct user_element) + private_size * count, GFP_KERNEL); if (kctl->private_data == NULL) { kfree(kctl); From eacf6e0a238923dfce0626450adcb6d486072f28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2015 10:43:54 +0200 Subject: [PATCH 408/411] ALSA: hda - Expose codec type sysfs The type field of HD-audio codec object should be exposed to user-space so that it can identify which driver type to bind (legacy / asoc). Signed-off-by: Takashi Iwai --- sound/hda/hdac_sysfs.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 18aea43d230d9f..0a6ce3b84cc474 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -36,6 +36,7 @@ static ssize_t type##_show(struct device *dev, \ } \ static DEVICE_ATTR_RO(type) +CODEC_ATTR(type); CODEC_ATTR(vendor_id); CODEC_ATTR(subsystem_id); CODEC_ATTR(revision_id); @@ -45,6 +46,7 @@ CODEC_ATTR_STR(vendor_name); CODEC_ATTR_STR(chip_name); static struct attribute *hdac_dev_attrs[] = { + &dev_attr_type.attr, &dev_attr_vendor_id.attr, &dev_attr_subsystem_id.attr, &dev_attr_revision_id.attr, From c3aeda62878f09da91329693a60a1f08ec97e0b8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2015 11:01:14 +0200 Subject: [PATCH 409/411] ALSA: hda - Fix another race in runtime PM refcounting Although some races in runtime PM refcount was fixed by the commit [664c715573c2: ALSA: hda - Work around races of power up/down with runtime PM], there is still a race in the following case: CPU0: CPU1 : runtime suspend: codec->in_pm = 1 snd_hdac_power_up_pm(): pm_runtime_get_sync() skipped suspend finished: codec->in_pm = 0 snd_hdac_power_down_pm(): pm_runtime_put_*() is called! For avoiding this situation, increment in_pm flag atomically when it's non-zero, and decrement accordingly, to ensure that in_pm is set consistently for the whole concurrent operations. Also, since atomic_inc_not_zero() and atomic_dec_if_positive() are lengthy inline functions, move snd_hdac_power_up_pm() and _down_pm() to sound/hda/hdac_device.c as no inline functions. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 32 ++++---------------------------- sound/hda/hdac_device.c | 30 ++++++++++++++++++++++++++++++ 2 files changed, 34 insertions(+), 28 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 30446f17c6a65c..2a8aa9dfb83d73 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -139,39 +139,15 @@ static inline int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, #ifdef CONFIG_PM void snd_hdac_power_up(struct hdac_device *codec); void snd_hdac_power_down(struct hdac_device *codec); +void snd_hdac_power_up_pm(struct hdac_device *codec); +void snd_hdac_power_down_pm(struct hdac_device *codec); #else static inline void snd_hdac_power_up(struct hdac_device *codec) {} static inline void snd_hdac_power_down(struct hdac_device *codec) {} +static inline void snd_hdac_power_up_pm(struct hdac_device *codec) {} +static inline void snd_hdac_power_down_pm(struct hdac_device *codec) {} #endif -/** - * snd_hdac_power_up_pm - power up the codec - * @codec: the codec object - * - * This function can be called in a recursive code path like init code - * which may be called by PM suspend/resume again. OTOH, if a power-up - * call must wake up the sleeper (e.g. in a kctl callback), use - * snd_hdac_power_up() instead. - */ -static inline void snd_hdac_power_up_pm(struct hdac_device *codec) -{ - if (!atomic_read(&codec->in_pm)) - snd_hdac_power_up(codec); -} - -/** - * snd_hdac_power_down_pm - power down the codec - * @codec: the codec object - * - * Like snd_hdac_power_up_pm(), this function is used in a recursive - * code path like init code which may be called by PM suspend/resume again. - */ -static inline void snd_hdac_power_down_pm(struct hdac_device *codec) -{ - if (!atomic_read(&codec->in_pm)) - snd_hdac_power_down(codec); -} - /* * HD-audio codec base driver */ diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 92604bbcee5ff1..f75bf562268783 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -519,6 +519,36 @@ void snd_hdac_power_down(struct hdac_device *codec) pm_runtime_put_autosuspend(dev); } EXPORT_SYMBOL_GPL(snd_hdac_power_down); + +/** + * snd_hdac_power_up_pm - power up the codec + * @codec: the codec object + * + * This function can be called in a recursive code path like init code + * which may be called by PM suspend/resume again. OTOH, if a power-up + * call must wake up the sleeper (e.g. in a kctl callback), use + * snd_hdac_power_up() instead. + */ +void snd_hdac_power_up_pm(struct hdac_device *codec) +{ + if (!atomic_inc_not_zero(&codec->in_pm)) + snd_hdac_power_up(codec); +} +EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); + +/** + * snd_hdac_power_down_pm - power down the codec + * @codec: the codec object + * + * Like snd_hdac_power_up_pm(), this function is used in a recursive + * code path like init code which may be called by PM suspend/resume again. + */ +void snd_hdac_power_down_pm(struct hdac_device *codec) +{ + if (atomic_dec_if_positive(&codec->in_pm) < 0) + snd_hdac_power_down(codec); +} +EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm); #endif /* codec vendor labels */ From f2aa111041ce36b94e651d882458dea502e76721 Mon Sep 17 00:00:00 2001 From: Jo-Philipp Wich Date: Mon, 13 Apr 2015 12:47:26 +0200 Subject: [PATCH 410/411] ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450 The Lenovo Thinkpad T450 requires the ALC292_FIXUP_TPT440_DOCK as well in order to get working sound output on the docking stations headphone jack. Patch tested on a Thinkpad T450 (20BVCTO1WW) using kernel 4.0-rc7 in conjunction with a ThinkPad Ultradock. Signed-off-by: Jo-Philipp Wich Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b26a7189fb3a70..b18b9c67b262bb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5141,6 +5141,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), From d6eb9e3ec78c98324097bab8eea266c3bb0d0ac7 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 14 Apr 2015 11:25:36 +0800 Subject: [PATCH 411/411] ALSA: hda - set GET bit when adding a vendor verb to the codec regmap Some HD-A codecs may add their own vendor 'set' verb to the regmap, thru func snd_hdac_add_vendor_verb(). This patch sets the GET bit (bit 11) when adding the verb so that its peer vendor 'get' verb is actually added. This can avoid I/O error when writing the 'set' verb thru remap, since HD-A regmap internally looks up a writable vendor verb with GET bit set at first. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/hda/hdac_regmap.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 51f1b5c8a91ccc..7371e0c3926f32 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -368,7 +368,7 @@ int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec, if (!p) return -ENOMEM; - *p = verb; + *p = verb | 0x800; /* set GET bit */ return 0; } EXPORT_SYMBOL_GPL(snd_hdac_regmap_add_vendor_verb);