Merge remote-tracking branch 'asoc/for-5.12' into asoc-linus

sound/soc/amd/raven/acp3x-i2s.c

@@ -264,8 +264,7 @@ static struct snd_soc_dai_driver acp3x_i2s_dai = {
 	.playback = {
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 |
-			SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE |
-			SNDRV_PCM_FMTBIT_S32_LE,
+			SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE,
 		.channels_min = 2,
 		.channels_max = 8,
 		.rate_min = 8000,
@@ -274,8 +273,7 @@ static struct snd_soc_dai_driver acp3x_i2s_dai = {
 	.capture = {
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 |
-			SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE |
-			SNDRV_PCM_FMTBIT_S32_LE,
+			SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE,
 		.channels_min = 2,
 		.channels_max = 2,
 		.rate_min = 8000,

sound/soc/amd/raven/acp3x-pcm-dma.c

@@ -24,8 +24,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = {
 		SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 		SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE |  SNDRV_PCM_FMTBIT_S8 |
-		   SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE |
-		   SNDRV_PCM_FMTBIT_S32_LE,
+		   SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE,
 	.channels_min = 2,
 	.channels_max = 8,
 	.rates = SNDRV_PCM_RATE_8000_96000,
@@ -45,8 +44,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = {
 		SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 		SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 |
-		   SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE |
-		   SNDRV_PCM_FMTBIT_S32_LE,
+		   SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE,
 	.channels_min = 2,
 	.channels_max = 2,
 	.rates = SNDRV_PCM_RATE_8000_48000,

sound/soc/codecs/adau17x1.c

@@ -553,6 +553,7 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai,
 {
 	struct adau *adau = snd_soc_component_get_drvdata(dai->component);
 	unsigned int ctrl0, ctrl1;
+	unsigned int ctrl0_mask;
 	int lrclk_pol;
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -612,8 +613,16 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai,
 	if (lrclk_pol)
 		ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL;
 
-	regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0);
-	regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1);
+	/* Set the mask to update all relevant bits in ADAU17X1_SERIAL_PORT0 */
+	ctrl0_mask = ADAU17X1_SERIAL_PORT0_MASTER |
+		     ADAU17X1_SERIAL_PORT0_LRCLK_POL |
+		     ADAU17X1_SERIAL_PORT0_BCLK_POL |
+		     ADAU17X1_SERIAL_PORT0_PULSE_MODE;
+
+	regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0_mask,
+			   ctrl0);
+	regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1,
+			   ADAU17X1_SERIAL_PORT1_DELAY_MASK, ctrl1);
 
 	adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
 

sound/soc/codecs/rt1011.c

@@ -2239,18 +2239,9 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag)
 	dc_offset |= (value & 0xffff);
 	dev_info(dev, "Gain1 offset=0x%x\n", dc_offset);
 
-	/* check the package info. */
-	regmap_read(rt1011->regmap, RT1011_EFUSE_MATCH_DONE, &value);
-	if (value & 0x4)
-		rt1011->pack_id = 1;
-
 	if (cali_flag) {
 
-		if (rt1011->pack_id)
-			regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x292c);
-		else
-			regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925);
-
+		regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925);
 		/* Class D on */
 		regmap_write(rt1011->regmap, RT1011_CLASS_D_POS, 0x010e);
 		regmap_write(rt1011->regmap,
@@ -2376,10 +2367,7 @@ static void rt1011_calibration_work(struct work_struct *work)
 		rt1011_r0_load(rt1011);
 	}
 
-	if (rt1011->pack_id)
-		snd_soc_component_write(component, RT1011_ADC_SET_1, 0x292c);
-	else
-		snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925);
+	snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925);
 }
 
 static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev)

sound/soc/codecs/rt1011.h

@@ -692,7 +692,6 @@ struct rt1011_priv {
 	unsigned int r0_reg, cali_done;
 	unsigned int r0_calib, temperature_calib;
 	int recv_spk_mode;
-	unsigned int pack_id; /* 0: WLCSP; 1: QFN */
 };
 
 #endif		/* end of _RT1011_H_ */

sound/soc/codecs/rt286.c

@@ -171,6 +171,9 @@ static bool rt286_readable_register(struct device *dev, unsigned int reg)
 	case RT286_PROC_COEF:
 	case RT286_SET_AMP_GAIN_ADC_IN1:
 	case RT286_SET_AMP_GAIN_ADC_IN2:
+	case RT286_SET_GPIO_MASK:
+	case RT286_SET_GPIO_DIRECTION:
+	case RT286_SET_GPIO_DATA:
 	case RT286_SET_POWER(RT286_DAC_OUT1):
 	case RT286_SET_POWER(RT286_DAC_OUT2):
 	case RT286_SET_POWER(RT286_ADC_IN1):
@@ -1204,7 +1207,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c,
 	mdelay(10);
 
 	if (!rt286->pdata.gpio2_en)
-		regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000);
+		regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x40);
 	else
 		regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0);
 

sound/soc/codecs/tas2552.c

@@ -730,8 +730,10 @@ static int tas2552_probe(struct i2c_client *client,
 	ret = devm_snd_soc_register_component(&client->dev,
 				      &soc_component_dev_tas2552,
 				      tas2552_dai, ARRAY_SIZE(tas2552_dai));
-	if (ret < 0)
+	if (ret < 0) {
 		dev_err(&client->dev, "Failed to register component: %d\n", ret);
+		pm_runtime_get_noresume(&client->dev);
+	}
 
 	return ret;
 }

sound/soc/codecs/tlv320aic32x4.c

@@ -577,12 +577,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
 		.window_start = 0,
 		.window_len = 128,
 		.range_min = 0,
-		.range_max = AIC32X4_RMICPGAVOL,
+		.range_max = AIC32X4_REFPOWERUP,
 	},
 };
 
 const struct regmap_config aic32x4_regmap_config = {
-	.max_register = AIC32X4_RMICPGAVOL,
+	.max_register = AIC32X4_REFPOWERUP,
 	.ranges = aic32x4_regmap_pages,
 	.num_ranges = ARRAY_SIZE(aic32x4_regmap_pages),
 };
@@ -1243,6 +1243,10 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap)
 	if (ret)
 		goto err_disable_regulators;
 
+	ret = aic32x4_register_clocks(dev, aic32x4->mclk_name);
+	if (ret)
+		goto err_disable_regulators;
+
 	ret = devm_snd_soc_register_component(dev,
 			&soc_component_dev_aic32x4, &aic32x4_dai, 1);
 	if (ret) {
@@ -1250,10 +1254,6 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap)
 		goto err_disable_regulators;
 	}
 
-	ret = aic32x4_register_clocks(dev, aic32x4->mclk_name);
-	if (ret)
-		goto err_disable_regulators;
-
 	return 0;
 
 err_disable_regulators:

sound/soc/intel/boards/kbl_da7219_max98927.c

@@ -282,11 +282,33 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-	struct snd_soc_dpcm *dpcm = container_of(
-			params, struct snd_soc_dpcm, hw_params);
-	struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
-	struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+	struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
 
+	/*
+	 * The following loop will be called only for playback stream
+	 * In this platform, there is only one playback device on every SSP
+	 */
+	for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+		rtd_dpcm = dpcm;
+		break;
+	}
+
+	/*
+	 * This following loop will be called only for capture stream
+	 * In this platform, there is only one capture device on every SSP
+	 */
+	for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+		rtd_dpcm = dpcm;
+		break;
+	}
+
+	if (!rtd_dpcm)
+		return -EINVAL;
+
+	/*
+	 * The above 2 loops are mutually exclusive based on the stream direction,
+	 * thus rtd_dpcm variable will never be overwritten
+	 */
 	/*
 	 * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE,
 	 * where as kblda7219m98927 & kblmax98927 supports S16_LE by default.
@@ -309,9 +331,9 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	/*
 	 * The ADSP will convert the FE rate to 48k, stereo, 24 bit
 	 */
-	if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+	if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
 		rate->min = rate->max = 48000;
 		chan->min = chan->max = 2;
 		snd_mask_none(fmt);
@@ -322,7 +344,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
 	 * thus changing the mask here
 	 */
-	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+	if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;

sound/soc/intel/keembay/kmb_platform.c

@@ -105,14 +105,15 @@ static unsigned int kmb_pcm_tx_fn(struct kmb_i2s_info *kmb_i2s,
 	void *buf = runtime->dma_area;
 	int i;
 
+	if (kmb_i2s->iec958_fmt)
+		hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr);
+
 	/* KMB i2s uses two separate L/R FIFO */
 	for (i = 0; i < kmb_i2s->fifo_th; i++) {
 		if (kmb_i2s->config.data_width == 16) {
 			writel(((u16(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0));
 			writel(((u16(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0));
 		} else {
-			if (kmb_i2s->iec958_fmt)
-				hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr);
 			writel(((u32(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0));
 			writel(((u32(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0));
 		}

sound/soc/meson/axg-frddr.c

@@ -11,6 +11,7 @@
 #include <linux/regmap.h>
 #include <linux/module.h>
 #include <linux/of_platform.h>
+#include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/soc-dai.h>
 
@@ -46,11 +47,28 @@ static int g12a_frddr_dai_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int axg_frddr_dai_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params,
+				   struct snd_soc_dai *dai)
+{
+	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+	unsigned int period, depth, val;
+
+	period = params_period_bytes(params);
+
+	/* Trim the FIFO depth if the period is small to improve latency */
+	depth = min(period, fifo->depth);
+	val = (depth / AXG_FIFO_BURST) - 1;
+	regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK,
+			   CTRL1_FRDDR_DEPTH(val));
+
+	return 0;
+}
+
 static int axg_frddr_dai_startup(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
 	struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
-	unsigned int val;
 	int ret;
 
 	/* Enable pclk to access registers and clock the fifo ip */
@@ -61,11 +79,6 @@ static int axg_frddr_dai_startup(struct snd_pcm_substream *substream,
 	/* Apply single buffer mode to the interface */
 	regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0);
 
-	/* Use all fifo depth */
-	val = (fifo->depth / AXG_FIFO_BURST) - 1;
-	regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK,
-			   CTRL1_FRDDR_DEPTH(val));
-
 	return 0;
 }
 
@@ -84,6 +97,7 @@ static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd,
 }
 
 static const struct snd_soc_dai_ops axg_frddr_ops = {
+	.hw_params	= axg_frddr_dai_hw_params,
 	.startup	= axg_frddr_dai_startup,
 	.shutdown	= axg_frddr_dai_shutdown,
 };
@@ -157,6 +171,7 @@ static const struct axg_fifo_match_data axg_frddr_match_data = {
 
 static const struct snd_soc_dai_ops g12a_frddr_ops = {
 	.prepare	= g12a_frddr_dai_prepare,
+	.hw_params	= axg_frddr_dai_hw_params,
 	.startup	= axg_frddr_dai_startup,
 	.shutdown	= axg_frddr_dai_shutdown,
 };