Merge tag 'asoc-v3.16-rc5' into asoc-linus

ASoC: Fixes for v3.16

A bigger batch of changes than I would like as I didn't send any for a
few weeks without noticing how many had built up.  They are almost all
driver specific though, larger changes are:

 - Fixes to the newly added Baytrail/MAX98090 which look like some QA
   was missed on the microphone detection.
 - Deletion of some erroniously listed audio formats for Haswell.
 - Fix debugfs creation in the core so that we don't try to generate
   multiple directories with the same name, relatively large textually
   but simple to inspect by eye and test.
 - A couple of bugfixes for the rcar driver one of which which involves
   a bit of code motion to move initailisation of some hardware out of
   common paths into device specific ones.
 - Ensure both channels are powered up for mono outputs on Arizona
   devices, involving some simple data tables listing the outputs and a
   loop over them.
 - A couple of fixes to save and restore information on suspended and
   idle Samsung I2S controllers.

# gpg: Signature made Tue 22 Jul 2014 00:52:53 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <[email protected]>"
# gpg:                 aka "Mark Brown <[email protected]>"
# gpg:                 aka "Mark Brown <[email protected]>"
# gpg:                 aka "Mark Brown <[email protected]>"
# gpg:                 aka "Mark Brown <[email protected]>"
# gpg:                 aka "Mark Brown <[email protected]>"

sound/soc/blackfin/bf5xx-i2s-pcm.c

@@ -290,19 +290,19 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream,
 	unsigned int sample_size = runtime->sample_bits / 8;
 	void *buf = runtime->dma_area;
 	struct bf5xx_i2s_pcm_data *dma_data;
-	unsigned int offset, size;
+	unsigned int offset, samples;
 
 	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
 	if (dma_data->tdm_mode) {
 		offset = pos * 8 * sample_size;
-		size = count * 8 * sample_size;
+		samples = count * 8;
 	} else {
 		offset = frames_to_bytes(runtime, pos);
-		size = frames_to_bytes(runtime, count);
+		samples = count * runtime->channels;
 	}
 
-	snd_pcm_format_set_silence(runtime->format, buf + offset, size);
+	snd_pcm_format_set_silence(runtime->format, buf + offset, samples);
 
 	return 0;
 }

sound/soc/codecs/adau1701.c

@@ -230,8 +230,10 @@ static int adau1701_reg_read(void *context, unsigned int reg,
 
 	*value = 0;
 
-	for (i = 0; i < size; i++)
-		*value |= recv_buf[i] << (i * 8);
+	for (i = 0; i < size; i++) {
+		*value <<= 8;
+		*value |= recv_buf[i];
+	}
 
 	return 0;
 }

sound/soc/codecs/arizona.c

@@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(arizona_init_spk);
 
+static const struct snd_soc_dapm_route arizona_mono_routes[] = {
+	{ "OUT1R", NULL, "OUT1L" },
+	{ "OUT2R", NULL, "OUT2L" },
+	{ "OUT3R", NULL, "OUT3L" },
+	{ "OUT4R", NULL, "OUT4L" },
+	{ "OUT5R", NULL, "OUT5L" },
+	{ "OUT6R", NULL, "OUT6L" },
+};
+
+int arizona_init_mono(struct snd_soc_codec *codec)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	int i;
+
+	for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) {
+		if (arizona->pdata.out_mono[i])
+			snd_soc_dapm_add_routes(&codec->dapm,
+						&arizona_mono_routes[i], 1);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_mono);
+
 int arizona_init_gpio(struct snd_soc_codec *codec)
 {
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);

sound/soc/codecs/arizona.h

@@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source,
 
 extern int arizona_init_spk(struct snd_soc_codec *codec);
 extern int arizona_init_gpio(struct snd_soc_codec *codec);
+extern int arizona_init_mono(struct snd_soc_codec *codec);
 
 extern int arizona_init_dai(struct arizona_priv *priv, int dai);
 

sound/soc/codecs/cs42l56.c

@@ -445,9 +445,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
 	SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1),
 
 	SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME,
-			      CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv),
+			      CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv),
 	SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME,
-			      CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv),
+			      CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv),
 
 	SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL,
 			0, 0x00, 1, tone_tlv),

sound/soc/codecs/max98090.c

@@ -2284,7 +2284,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	/* Register for interrupts */
 	dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
 
-	ret = request_threaded_irq(max98090->irq, NULL,
+	ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL,
 		max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
 		"max98090_interrupt", codec);
 	if (ret < 0) {

sound/soc/codecs/sgtl5000.c

@@ -1277,21 +1277,24 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 			return ret;
 	}
 
-	ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
 				 sgtl5000->supplies);
 	if (ret)
 		goto err_ldo_remove;
 
 	ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
 					sgtl5000->supplies);
 	if (ret)
-		goto err_ldo_remove;
+		goto err_regulator_free;
 
 	/* wait for all power rails bring up */
 	udelay(10);
 
 	return 0;
 
+err_regulator_free:
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
 err_ldo_remove:
 	if (!external_vddd)
 		ldo_regulator_remove(codec);
@@ -1361,6 +1364,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 err:
 	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
 						sgtl5000->supplies);
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
 	ldo_regulator_remove(codec);
 
 	return ret;
@@ -1374,6 +1379,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec)
 
 	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
 						sgtl5000->supplies);
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
 	ldo_regulator_remove(codec);
 
 	return 0;

sound/soc/codecs/tlv320aic3x.c

@@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_FORMAT_S20_3LE:
 		data |= (0x01 << 4);
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case SNDRV_PCM_FORMAT_S24_3LE:
 		data |= (0x02 << 4);
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:

sound/soc/codecs/wm5110.c

@@ -1596,6 +1596,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 
 	arizona_init_spk(codec);
 	arizona_init_gpio(codec);
+	arizona_init_mono(codec);
 
 	ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8);
 	if (ret != 0)

sound/soc/codecs/wm_adsp.c

@@ -1758,3 +1758,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs)
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm_adsp2_init);
+
+MODULE_LICENSE("GPL v2");

sound/soc/davinci/Kconfig

@@ -6,6 +6,7 @@ config SND_DAVINCI_SOC_I2S
 	tristate
 
 config SND_DAVINCI_SOC_MCASP
+	depends on SND_DAVINCI_SOC || SND_OMAP_SOC
 	tristate
 
 config SND_DAVINCI_SOC_VCIF

sound/soc/davinci/davinci-mcasp.c

@@ -720,6 +720,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 
 	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;
@@ -1223,14 +1227,22 @@ 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:
 	case MCASP_VERSION_3:
 		ret = davinci_soc_platform_register(&pdev->dev);
 		break;
+#endif
+#if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \
+	(IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+	 IS_MODULE(CONFIG_SND_OMAP_SOC))
 	case MCASP_VERSION_4:
 		ret = omap_pcm_platform_register(&pdev->dev);
 		break;
+#endif
 	default:
 		dev_err(&pdev->dev, "Invalid McASP version: %d\n",
 			mcasp->version);

sound/soc/fsl/fsl_sai.c

@@ -106,7 +106,7 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid)
 	xcsr &= ~FSL_SAI_CSR_xF_MASK;
 
 	if (flags)
-		regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr);
+		regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr);
 
 out:
 	if (irq_none)
@@ -371,10 +371,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 
 		/* Check if the opposite FRDE is also disabled */
 		if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) {
+			/* Disable both directions and reset their FIFOs */
 			regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
-					   FSL_SAI_CSR_TERE, 0);
+					   FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR,
+					   FSL_SAI_CSR_FR);
 			regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
-					   FSL_SAI_CSR_TERE, 0);
+					   FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR,
+					   FSL_SAI_CSR_FR);
 		}
 		break;
 	default:

sound/soc/generic/simple-card.c

@@ -116,6 +116,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 {
 	struct device_node *node;
 	struct clk *clk;
+	u32 val;
 	int ret;
 
 	/*
@@ -151,10 +152,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 		}
 
 		dai->sysclk = clk_get_rate(clk);
-	} else if (of_property_read_bool(np, "system-clock-frequency")) {
-		of_property_read_u32(np,
-				     "system-clock-frequency",
-				     &dai->sysclk);
+	} else if (!of_property_read_u32(np, "system-clock-frequency", &val)) {
+		dai->sysclk = val;
 	} else {
 		clk = of_clk_get(node, 0);
 		if (!IS_ERR(clk))
@@ -303,6 +302,7 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 {
 	struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link;
 	struct simple_dai_props *dai_props = priv->dai_props;
+	u32 val;
 	int ret;
 
 	/* parsing the card name from DT */
@@ -325,8 +325,9 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 	}
 
 	/* Factor to mclk, used in hw_params() */
-	of_property_read_u32(node, "simple-audio-card,mclk-fs",
-			     &priv->mclk_fs);
+	ret = of_property_read_u32(node, "simple-audio-card,mclk-fs", &val);
+	if (ret == 0)
+		priv->mclk_fs = val;
 
 	dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ?
 		priv->snd_card.name : "");

sound/soc/intel/byt-max98090.c

@@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = {
 
 static const struct snd_soc_dapm_route byt_max98090_audio_map[] = {
 	{"IN34", NULL, "Headset Mic"},
-	{"IN34", NULL, "MICBIAS"},
-	{"MICBIAS", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "MICBIAS"},
 	{"DMICL", NULL, "Int Mic"},
 	{"Headphone", NULL, "HPL"},
 	{"Headphone", NULL, "HPR"},
@@ -84,7 +83,8 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = {
 	{
 		.name		= "mic-gpio",
 		.idx		= 1,
-		.report		= SND_JACK_MICROPHONE | SND_JACK_LINEIN,
+		.invert		= 1,
+		.report		= SND_JACK_MICROPHONE,
 		.debounce_time	= 200,
 	},
 };
@@ -108,7 +108,8 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
 	}
 
 	/* Enable jack detection */
-	ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack);
+	ret = snd_soc_jack_new(codec, "Headset",
+			       SND_JACK_LINEOUT | SND_JACK_HEADSET, jack);
 	if (ret)
 		return ret;
 
@@ -117,13 +118,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
 	if (ret)
 		return ret;
 
-	ret = snd_soc_jack_add_gpiods(card->dev->parent, jack,
-				      ARRAY_SIZE(hs_jack_gpios),
-				      hs_jack_gpios);
-	if (ret)
-		return ret;
-
-	return max98090_mic_detect(codec, jack);
+	return snd_soc_jack_add_gpiods(card->dev->parent, jack,
+				       ARRAY_SIZE(hs_jack_gpios),
+				       hs_jack_gpios);
 }
 
 static struct snd_soc_dai_link byt_max98090_dais[] = {

sound/soc/intel/sst-baytrail-pcm.c

@@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = {
 				  SNDRV_PCM_INFO_PAUSE |
 				  SNDRV_PCM_INFO_RESUME,
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-				  SNDRV_PCM_FORMAT_S24_LE,
+				  SNDRV_PCM_FMTBIT_S24_LE,
 	.period_bytes_min	= 384,
 	.period_bytes_max	= 48000,
 	.periods_min		= 2,

sound/soc/intel/sst-haswell-dsp.c

@@ -359,6 +359,17 @@ static u32 hsw_block_get_bit(struct sst_mem_block *block)
 	return bit;
 }
 
+/*dummy read a SRAM block.*/
+static void sst_mem_block_dummy_read(struct sst_mem_block *block)
+{
+	u32 size;
+	u8 tmp_buf[4];
+	struct sst_dsp *sst = block->dsp;
+
+	size = block->size > 4 ? 4 : block->size;
+	memcpy_fromio(tmp_buf, sst->addr.lpe + block->offset, size);
+}
+
 /* enable 32kB memory block - locks held by caller */
 static int hsw_block_enable(struct sst_mem_block *block)
 {
@@ -378,6 +389,8 @@ static int hsw_block_enable(struct sst_mem_block *block)
 	/* wait 18 DSP clock ticks */
 	udelay(10);
 
+	/*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/
+	sst_mem_block_dummy_read(block);
 	return 0;
 }