Merge tag 'v5.13-rc3' into asoc-5.13

Linux 5.13-rc3

32 files changed, 212 insertions, 115 deletions

diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c
index 396cb7d619e2..de6f70d7ef36 100644
--- a/sound/soc/amd/raven/acp3x-i2s.c
+++ b/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,
diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index 2447a1e6e913..8148b0d22e88 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/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,
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
index fea296b41a43..ba03bb62ba96 100644
--- a/sound/soc/bcm/cygnus-ssp.c
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -1348,8 +1348,10 @@ static int cygnus_ssp_probe(struct platform_device *pdev)
 					&cygnus_ssp_dai[active_port_count]);
 
 		/* negative is err, 0 is active and good, 1 is disabled */
-		if (err < 0)
+		if (err < 0) {
+			of_node_put(child_node);
 			return err;
+		}
 		else if (!err) {
 			dev_dbg(dev, "Activating DAI: %s\n",
 				cygnus_ssp_dai[active_port_count].name);
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 546ee8178038..8aae7ab74091 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/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;
 
diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c
index 80bc7c10ed75..80cd3ea0c157 100644
--- a/sound/soc/codecs/cs43130.c
+++ b/sound/soc/codecs/cs43130.c
@@ -1735,6 +1735,14 @@ static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL);
 static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL);
 static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL);
 
+static struct attribute *hpload_attrs[] = {
+	&dev_attr_hpload_dc_l.attr,
+	&dev_attr_hpload_dc_r.attr,
+	&dev_attr_hpload_ac_l.attr,
+	&dev_attr_hpload_ac_r.attr,
+};
+ATTRIBUTE_GROUPS(hpload);
+
 static struct reg_sequence hp_en_cal_seq[] = {
 	{CS43130_INT_MASK_4, CS43130_INT_MASK_ALL},
 	{CS43130_HP_MEAS_LOAD_1, 0},
@@ -2302,25 +2310,15 @@ static int cs43130_probe(struct snd_soc_component *component)
 
 	cs43130->hpload_done = false;
 	if (cs43130->dc_meas) {
-		ret = device_create_file(component->dev, &dev_attr_hpload_dc_l);
-		if (ret < 0)
-			return ret;
-
-		ret = device_create_file(component->dev, &dev_attr_hpload_dc_r);
-		if (ret < 0)
-			return ret;
-
-		ret = device_create_file(component->dev, &dev_attr_hpload_ac_l);
-		if (ret < 0)
-			return ret;
-
-		ret = device_create_file(component->dev, &dev_attr_hpload_ac_r);
-		if (ret < 0)
+		ret = sysfs_create_groups(&component->dev->kobj, hpload_groups);
+		if (ret)
 			return ret;
 
 		cs43130->wq = create_singlethread_workqueue("cs43130_hp");
-		if (!cs43130->wq)
+		if (!cs43130->wq) {
+			sysfs_remove_groups(&component->dev->kobj, hpload_groups);
 			return -ENOMEM;
+		}
 		INIT_WORK(&cs43130->work, cs43130_imp_meas);
 	}
 
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 161be8b7d131..61dfa86d444d 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -285,7 +285,6 @@ static void v253_wakeup(struct tty_struct *tty)
 }
 
 struct tty_ldisc_ops v253_ops = {
-	.magic = TTY_LDISC_MAGIC,
 	.name = "cx20442",
 	.owner = THIS_MODULE,
 	.open = v253_open,
diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c
index e074c7908c23..171ab7f519c0 100644
--- a/sound/soc/codecs/lpass-rx-macro.c
+++ b/sound/soc/codecs/lpass-rx-macro.c
@@ -3549,7 +3549,7 @@ static int rx_macro_probe(struct platform_device *pdev)
 
 	/* set MCLK and NPL rates */
 	clk_set_rate(rx->clks[2].clk, MCLK_FREQ);
-	clk_set_rate(rx->clks[3].clk, MCLK_FREQ);
+	clk_set_rate(rx->clks[3].clk, 2 * MCLK_FREQ);
 
 	ret = clk_bulk_prepare_enable(RX_NUM_CLKS_MAX, rx->clks);
 	if (ret)
diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
index 3d3a6e31551b..27a0d5defd27 100644
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@@ -1811,7 +1811,7 @@ static int tx_macro_probe(struct platform_device *pdev)
 
 	/* set MCLK and NPL rates */
 	clk_set_rate(tx->clks[2].clk, MCLK_FREQ);
-	clk_set_rate(tx->clks[3].clk, MCLK_FREQ);
+	clk_set_rate(tx->clks[3].clk, 2 * MCLK_FREQ);
 
 	ret = clk_bulk_prepare_enable(TX_NUM_CLKS_MAX, tx->clks);
 	if (ret)
diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c
index 85f6865019d4..ddb6436835d7 100644
--- a/sound/soc/codecs/max98373-i2c.c
+++ b/sound/soc/codecs/max98373-i2c.c
@@ -446,6 +446,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg)
 	case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK:
 	case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK:
 	case MAX98373_R20B6_BDE_CUR_STATE_READBACK:
+	case MAX98373_R20FF_GLOBAL_SHDN:
 	case MAX98373_R21FF_REV_ID:
 		return true;
 	default:
diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c
index d1d26bdca1ca..dc520effc61c 100644
--- a/sound/soc/codecs/max98373-sdw.c
+++ b/sound/soc/codecs/max98373-sdw.c
@@ -220,6 +220,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg)
 	case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK:
 	case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK:
 	case MAX98373_R20B6_BDE_CUR_STATE_READBACK:
+	case MAX98373_R20FF_GLOBAL_SHDN:
 	case MAX98373_R21FF_REV_ID:
 	/* SoundWire Control Port Registers */
 	case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR:
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index b1e43c26dd6b..e14fe98349a5 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -28,11 +28,13 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w,
 		regmap_update_bits(max98373->regmap,
 			MAX98373_R20FF_GLOBAL_SHDN,
 			MAX98373_GLOBAL_EN_MASK, 1);
+		usleep_range(30000, 31000);
 		break;
 	case SND_SOC_DAPM_POST_PMD:
 		regmap_update_bits(max98373->regmap,
 			MAX98373_R20FF_GLOBAL_SHDN,
 			MAX98373_GLOBAL_EN_MASK, 0);
+		usleep_range(30000, 31000);
 		max98373->tdm_mode = false;
 		break;
 	default:
diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index 238cb66f4c6a..faff2b558687 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -2236,18 +2236,9 @@ 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,
@@ -2373,10 +2364,7 @@ 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)
diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h
index f3a9a96640f1..68fadc15fa8c 100644
--- a/sound/soc/codecs/rt1011.h
+++ b/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_ */
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 9408ee63cb26..438fa18bcb55 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3388,30 +3388,44 @@ static int rt5645_probe(struct snd_soc_component *component)
 {
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
 	struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+	int ret = 0;
 
 	rt5645->component = component;
 
 	switch (rt5645->codec_type) {
 	case CODEC_TYPE_RT5645:
-		snd_soc_dapm_new_controls(dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			rt5645_specific_dapm_widgets,
 			ARRAY_SIZE(rt5645_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(dapm,
+		if (ret < 0)
+			goto exit;
+
+		ret = snd_soc_dapm_add_routes(dapm,
 			rt5645_specific_dapm_routes,
 			ARRAY_SIZE(rt5645_specific_dapm_routes));
+		if (ret < 0)
+			goto exit;
+
 		if (rt5645->v_id < 3) {
-			snd_soc_dapm_add_routes(dapm,
+			ret = snd_soc_dapm_add_routes(dapm,
 				rt5645_old_dapm_routes,
 				ARRAY_SIZE(rt5645_old_dapm_routes));
+			if (ret < 0)
+				goto exit;
 		}
 		break;
 	case CODEC_TYPE_RT5650:
-		snd_soc_dapm_new_controls(dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			rt5650_specific_dapm_widgets,
 			ARRAY_SIZE(rt5650_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(dapm,
+		if (ret < 0)
+			goto exit;
+
+		ret = snd_soc_dapm_add_routes(dapm,
 			rt5650_specific_dapm_routes,
 			ARRAY_SIZE(rt5650_specific_dapm_routes));
+		if (ret < 0)
+			goto exit;
 		break;
 	}
 
@@ -3419,9 +3433,17 @@ static int rt5645_probe(struct snd_soc_component *component)
 
 	/* for JD function */
 	if (rt5645->pdata.jd_mode) {
-		snd_soc_dapm_force_enable_pin(dapm, "JD Power");
-		snd_soc_dapm_force_enable_pin(dapm, "LDO2");
-		snd_soc_dapm_sync(dapm);
+		ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power");
+		if (ret < 0)
+			goto exit;
+
+		ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2");
+		if (ret < 0)
+			goto exit;
+
+		ret = snd_soc_dapm_sync(dapm);
+		if (ret < 0)
+			goto exit;
 	}
 
 	if (rt5645->pdata.long_name)
@@ -3432,9 +3454,14 @@ static int rt5645_probe(struct snd_soc_component *component)
 		GFP_KERNEL);
 
 	if (!rt5645->eq_param)
-		return -ENOMEM;
-
-	return 0;
+		ret = -ENOMEM;
+exit:
+	/*
+	 * If there was an error above, everything will be cleaned up by the
+	 * caller if we return an error here.  This will be done with a later
+	 * call to rt5645_remove().
+	 */
+	return ret;
 }
 
 static void rt5645_remove(struct snd_soc_component *component)
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index bd00c35116cd..700baa6314aa 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/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;
 }
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index f04f88c8d425..b689f26fc4be 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/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:
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 847ca16b9841..9e621a254392 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -697,7 +697,13 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in,
 	best_freq_out = -EINVAL;
 	*sysclk_idx = *dac_idx = *bclk_idx = -1;
 
-	for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
+	/*
+	 * From Datasheet, the PLL performs best when f2 is between
+	 * 90MHz and 100MHz, the desired sysclk output is 11.2896MHz
+	 * or 12.288MHz, then sysclkdiv = 2 is the best choice.
+	 * So search sysclk_divs from 2 to 1 other than from 1 to 2.
+	 */
+	for (i = ARRAY_SIZE(sysclk_divs) - 1; i >= 0; --i) {
 		if (sysclk_divs[i] == -1)
 			continue;
 		for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c0d4f3c5dbb1..f356ae5925af 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -519,11 +519,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream,
 				   ESAI_SAICR_SYNC, esai_priv->synchronous ?
 				   ESAI_SAICR_SYNC : 0);
 
-		/* Set a default slot number -- 2 */
+		/* Set slots count */
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
-				   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+				   ESAI_xCCR_xDC_MASK,
+				   ESAI_xCCR_xDC(esai_priv->slots));
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
-				   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+				   ESAI_xCCR_xDC_MASK,
+				   ESAI_xCCR_xDC(esai_priv->slots));
 	}
 
 	return 0;
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 9e9b05883557..4124aa2fc247 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -487,15 +487,15 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
 		.stream_name = "Headset Playback",
 		.channels_min = SST_STEREO,
 		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
 		.stream_name = "Headset Capture",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 },
 {
@@ -505,8 +505,8 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
 		.stream_name = "Deepbuffer Playback",
 		.channels_min = SST_STEREO,
 		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 },
 {
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index 9dfe5bd9180d..4b7b4a044f81 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -284,12 +284,34 @@ 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.
 	 * Skipping the port wise FE and BE configuration for kblda7219m98373 &
@@ -311,9 +333,9 @@ 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);
@@ -324,7 +346,7 @@ 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;
diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c
index ef73392f8cff..a6fb74ba1c42 100644
--- a/sound/soc/intel/keembay/kmb_platform.c
+++ b/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));
 		}
diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c
index c3ae8ac30745..37f4bb3469b5 100644
--- a/sound/soc/meson/axg-frddr.c
+++ b/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,
 };
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index f57e363b5974..3e4dd4a86363 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -405,7 +405,13 @@ int snd_sof_device_shutdown(struct device *dev)
 {
 	struct snd_sof_dev *sdev = dev_get_drvdata(dev);
 
-	return snd_sof_shutdown(sdev);
+	if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
+		cancel_work_sync(&sdev->probe_work);
+
+	if (sdev->fw_state == SOF_FW_BOOT_COMPLETE)
+		return snd_sof_shutdown(sdev);
+
+	return 0;
 }
 EXPORT_SYMBOL(snd_sof_device_shutdown);
 
diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c
index fc29b91b8932..c7ed2b3d6abc 100644
--- a/sound/soc/sof/intel/apl.c
+++ b/sound/soc/sof/intel/apl.c
@@ -27,9 +27,10 @@ static const struct snd_sof_debugfs_map apl_dsp_debugfs[] = {
 
 /* apollolake ops */
 const struct snd_sof_dsp_ops sof_apl_ops = {
-	/* probe and remove */
+	/* probe/remove/shutdown */
 	.probe		= hda_dsp_probe,
 	.remove		= hda_dsp_remove,
+	.shutdown	= hda_dsp_shutdown,
 
 	/* Register IO */
 	.write		= sof_io_write,
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index e38db519f38d..821f25fbcf08 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -232,9 +232,10 @@ void cnl_ipc_dump(struct snd_sof_dev *sdev)
 
 /* cannonlake ops */
 const struct snd_sof_dsp_ops sof_cnl_ops = {
-	/* probe and remove */
+	/* probe/remove/shutdown */
 	.probe		= hda_dsp_probe,
 	.remove		= hda_dsp_remove,
+	.shutdown	= hda_dsp_shutdown,
 
 	/* Register IO */
 	.write		= sof_io_write,
@@ -349,22 +350,6 @@ const struct sof_intel_dsp_desc cnl_chip_info = {
 };
 EXPORT_SYMBOL_NS(cnl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
 
-const struct sof_intel_dsp_desc ehl_chip_info = {
-	/* Elkhartlake */
-	.cores_num = 4,
-	.init_core_mask = 1,
-	.host_managed_cores_mask = BIT(0),
-	.ipc_req = CNL_DSP_REG_HIPCIDR,
-	.ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY,
-	.ipc_ack = CNL_DSP_REG_HIPCIDA,
-	.ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE,
-	.ipc_ctl = CNL_DSP_REG_HIPCCTL,
-	.rom_init_timeout	= 300,
-	.ssp_count = ICL_SSP_COUNT,
-	.ssp_base_offset = CNL_SSP_BASE_OFFSET,
-};
-EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
-
 const struct sof_intel_dsp_desc jsl_chip_info = {
 	/* Jasperlake */
 	.cores_num = 2,
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index df4cb29b9a1b..623cf291e207 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -226,10 +226,17 @@ bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev,
 
 	val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS);
 
-	is_enable = (val & HDA_DSP_ADSPCS_CPA_MASK(core_mask)) &&
-		    (val & HDA_DSP_ADSPCS_SPA_MASK(core_mask)) &&
-		    !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) &&
-		    !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask));
+#define MASK_IS_EQUAL(v, m, field) ({	\
+	u32 _m = field(m);		\
+	((v) & _m) == _m;		\
+})
+
+	is_enable = MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_CPA_MASK) &&
+		MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_SPA_MASK) &&
+		!(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) &&
+		!(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask));
+
+#undef MASK_IS_EQUAL
 
 	dev_dbg(sdev->dev, "DSP core(s) enabled? %d : core_mask %x\n",
 		is_enable, core_mask);
@@ -884,6 +891,12 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state)
 	return snd_sof_dsp_set_power_state(sdev, &target_dsp_state);
 }
 
+int hda_dsp_shutdown(struct snd_sof_dev *sdev)
+{
+	sdev->system_suspend_target = SOF_SUSPEND_S3;
+	return snd_sof_suspend(sdev->dev);
+}
+
 int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev)
 {
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 5ef29dcc052d..4d44f8910393 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -517,6 +517,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev);
 int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev);
 int hda_dsp_runtime_resume(struct snd_sof_dev *sdev);
 int hda_dsp_runtime_idle(struct snd_sof_dev *sdev);
+int hda_dsp_shutdown(struct snd_sof_dev *sdev);
 int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev);
 void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags);
 void hda_ipc_dump(struct snd_sof_dev *sdev);
diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c
index e9d5a0a58504..88a74be8a0c1 100644
--- a/sound/soc/sof/intel/icl.c
+++ b/sound/soc/sof/intel/icl.c
@@ -26,9 +26,10 @@ static const struct snd_sof_debugfs_map icl_dsp_debugfs[] = {
 
 /* Icelake ops */
 const struct snd_sof_dsp_ops sof_icl_ops = {
-	/* probe and remove */
+	/* probe/remove/shutdown */
 	.probe		= hda_dsp_probe,
 	.remove		= hda_dsp_remove,
+	.shutdown	= hda_dsp_shutdown,
 
 	/* Register IO */
 	.write		= sof_io_write,
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index 21f69204d5ad..88c3bf404dd7 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -66,7 +66,7 @@ static const struct sof_dev_desc ehl_desc = {
 	.default_tplg_path = "intel/sof-tplg",
 	.default_fw_filename = "sof-ehl.ri",
 	.nocodec_tplg_filename = "sof-ehl-nocodec.tplg",
-	.ops = &sof_cnl_ops,
+	.ops = &sof_tgl_ops,
 };
 
 static const struct sof_dev_desc adls_desc = {
diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c
index 940f169047a0..2ed788304414 100644
--- a/sound/soc/sof/intel/tgl.c
+++ b/sound/soc/sof/intel/tgl.c
@@ -25,7 +25,7 @@ const struct snd_sof_dsp_ops sof_tgl_ops = {
 	/* probe/remove/shutdown */
 	.probe		= hda_dsp_probe,
 	.remove		= hda_dsp_remove,
-	.shutdown	= hda_dsp_remove,
+	.shutdown	= hda_dsp_shutdown,
 
 	/* Register IO */
 	.write		= sof_io_write,
@@ -156,6 +156,22 @@ const struct sof_intel_dsp_desc tglh_chip_info = {
 };
 EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
 
+const struct sof_intel_dsp_desc ehl_chip_info = {
+	/* Elkhartlake */
+	.cores_num = 4,
+	.init_core_mask = 1,
+	.host_managed_cores_mask = BIT(0),
+	.ipc_req = CNL_DSP_REG_HIPCIDR,
+	.ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY,
+	.ipc_ack = CNL_DSP_REG_HIPCIDA,
+	.ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE,
+	.ipc_ctl = CNL_DSP_REG_HIPCCTL,
+	.rom_init_timeout	= 300,
+	.ssp_count = ICL_SSP_COUNT,
+	.ssp_base_offset = CNL_SSP_BASE_OFFSET,
+};
+EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON);
+
 const struct sof_intel_dsp_desc adls_chip_info = {
 	/* Alderlake-S */
 	.cores_num = 2,
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index 00b73b7444b7..6f3d9148a185 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -1364,6 +1364,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev)
 		return ERR_PTR(-ENOMEM);
 
 	card->dev		= dev;
+	card->owner		= THIS_MODULE;
 	card->name		= "sun4i-codec";
 	card->dapm_widgets	= sun4i_codec_card_dapm_widgets;
 	card->num_dapm_widgets	= ARRAY_SIZE(sun4i_codec_card_dapm_widgets);
@@ -1396,6 +1397,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev)
 		return ERR_PTR(-ENOMEM);
 
 	card->dev		= dev;
+	card->owner		= THIS_MODULE;
 	card->name		= "A31 Audio Codec";
 	card->dapm_widgets	= sun6i_codec_card_dapm_widgets;
 	card->num_dapm_widgets	= ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1449,6 +1451,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev)
 		return ERR_PTR(-ENOMEM);
 
 	card->dev		= dev;
+	card->owner		= THIS_MODULE;
 	card->name		= "A23 Audio Codec";
 	card->dapm_widgets	= sun6i_codec_card_dapm_widgets;
 	card->num_dapm_widgets	= ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1487,6 +1490,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev)
 		return ERR_PTR(-ENOMEM);
 
 	card->dev		= dev;
+	card->owner		= THIS_MODULE;
 	card->name		= "H3 Audio Codec";
 	card->dapm_widgets	= sun6i_codec_card_dapm_widgets;
 	card->num_dapm_widgets	= ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1525,6 +1529,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev)
 		return ERR_PTR(-ENOMEM);
 
 	card->dev		= dev;
+	card->owner		= THIS_MODULE;
 	card->name		= "V3s Audio Codec";
 	card->dapm_widgets	= sun6i_codec_card_dapm_widgets;
 	card->num_dapm_widgets	= ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index 31462587f922..aba0017e870b 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -395,7 +395,6 @@ static void cx81801_wakeup(struct tty_struct *tty)
 }
 
 static struct tty_ldisc_ops cx81801_ops = {
-	.magic = TTY_LDISC_MAGIC,
 	.name = "cx81801",
 	.owner = THIS_MODULE,
 	.open = cx81801_open,