Merge tag 'asoc-fix-v5.8-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.8

An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them.  For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.

22 files changed, 241 insertions, 84 deletions

diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c
index f25ce50f1a90..ebf4388b6262 100644
--- a/sound/soc/amd/raven/pci-acp3x.c
+++ b/sound/soc/amd/raven/pci-acp3x.c
@@ -232,9 +232,7 @@ static int snd_acp3x_probe(struct pci_dev *pci,
 	}
 	pm_runtime_set_autosuspend_delay(&pci->dev, 2000);
 	pm_runtime_use_autosuspend(&pci->dev);
-	pm_runtime_set_active(&pci->dev);
 	pm_runtime_put_noidle(&pci->dev);
-	pm_runtime_enable(&pci->dev);
 	pm_runtime_allow(&pci->dev);
 	return 0;
 
@@ -303,7 +301,7 @@ static void snd_acp3x_remove(struct pci_dev *pci)
 	ret = acp3x_deinit(adata->acp3x_base);
 	if (ret)
 		dev_err(&pci->dev, "ACP de-init failed\n");
-	pm_runtime_disable(&pci->dev);
+	pm_runtime_forbid(&pci->dev);
 	pm_runtime_get_noresume(&pci->dev);
 	pci_disable_msi(pci);
 	pci_release_regions(pci);
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 96718e3a1ad0..d87402a86c88 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -779,13 +779,6 @@ static int max98373_probe(struct snd_soc_component *component)
 	regmap_write(max98373->regmap,
 		MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2,
 		0x1);
-	/* Set inital volume (0dB) */
-	regmap_write(max98373->regmap,
-		MAX98373_R203D_AMP_DIG_VOL_CTRL,
-		0x00);
-	regmap_write(max98373->regmap,
-		MAX98373_R203E_AMP_PATH_GAIN,
-		0x00);
 	/* Enable DC blocker */
 	regmap_write(max98373->regmap,
 		MAX98373_R203F_AMP_DSP_CFG,
@@ -869,7 +862,6 @@ static const struct snd_soc_component_driver soc_codec_dev_max98373 = {
 	.num_dapm_widgets	= ARRAY_SIZE(max98373_dapm_widgets),
 	.dapm_routes		= max98373_audio_map,
 	.num_dapm_routes	= ARRAY_SIZE(max98373_audio_map),
-	.idle_bias_on		= 1,
 	.use_pmdown_time	= 1,
 	.endianness		= 1,
 	.non_legacy_dai_naming	= 1,
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 9593a9a27bf8..e8d14eefc41b 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -272,13 +272,13 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 		regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf);
 		*mic = buf & 0x80000000;
 	}
-	if (!*mic) {
+
+	if (!*hp) {
 		snd_soc_dapm_disable_pin(dapm, "HV");
 		snd_soc_dapm_disable_pin(dapm, "VREF");
-	}
-	if (!*hp)
 		snd_soc_dapm_disable_pin(dapm, "LDO1");
-	snd_soc_dapm_sync(dapm);
+		snd_soc_dapm_sync(dapm);
+	}
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 70fee6849ab0..dfbc0ca38ff7 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -31,18 +31,19 @@
 #include "rt5670.h"
 #include "rt5670-dsp.h"
 
-#define RT5670_DEV_GPIO     BIT(0)
-#define RT5670_IN2_DIFF     BIT(1)
-#define RT5670_DMIC_EN      BIT(2)
-#define RT5670_DMIC1_IN2P   BIT(3)
-#define RT5670_DMIC1_GPIO6  BIT(4)
-#define RT5670_DMIC1_GPIO7  BIT(5)
-#define RT5670_DMIC2_INR    BIT(6)
-#define RT5670_DMIC2_GPIO8  BIT(7)
-#define RT5670_DMIC3_GPIO5  BIT(8)
-#define RT5670_JD_MODE1     BIT(9)
-#define RT5670_JD_MODE2     BIT(10)
-#define RT5670_JD_MODE3     BIT(11)
+#define RT5670_DEV_GPIO			BIT(0)
+#define RT5670_IN2_DIFF			BIT(1)
+#define RT5670_DMIC_EN			BIT(2)
+#define RT5670_DMIC1_IN2P		BIT(3)
+#define RT5670_DMIC1_GPIO6		BIT(4)
+#define RT5670_DMIC1_GPIO7		BIT(5)
+#define RT5670_DMIC2_INR		BIT(6)
+#define RT5670_DMIC2_GPIO8		BIT(7)
+#define RT5670_DMIC3_GPIO5		BIT(8)
+#define RT5670_JD_MODE1			BIT(9)
+#define RT5670_JD_MODE2			BIT(10)
+#define RT5670_JD_MODE3			BIT(11)
+#define RT5670_GPIO1_IS_EXT_SPK_EN	BIT(12)
 
 static unsigned long rt5670_quirk;
 static unsigned int quirk_override;
@@ -602,9 +603,9 @@ int rt5670_set_jack_detect(struct snd_soc_component *component,
 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_MINMAX(dac_vol_tlv, -6562, 0);
 static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
 static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
 
 /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
@@ -1447,6 +1448,33 @@ static int rt5670_hp_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int rt5670_spk_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+	struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+	if (!rt5670->pdata.gpio1_is_ext_spk_en)
+		return 0;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+				   RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_HI);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+				   RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_LO);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static int rt5670_bst1_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -1860,7 +1888,9 @@ static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = {
-	SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			   rt5670_spk_event, SND_SOC_DAPM_PRE_PMD |
+			   SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_OUTPUT("SPOLP"),
 	SND_SOC_DAPM_OUTPUT("SPOLN"),
 	SND_SOC_DAPM_OUTPUT("SPORP"),
@@ -2857,14 +2887,14 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
 	},
 	{
 		.callback = rt5670_quirk_cb,
-		.ident = "Lenovo Thinkpad Tablet 10",
+		.ident = "Lenovo Miix 2 10",
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
 			DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"),
 		},
 		.driver_data = (unsigned long *)(RT5670_DMIC_EN |
 						 RT5670_DMIC1_IN2P |
-						 RT5670_DEV_GPIO |
+						 RT5670_GPIO1_IS_EXT_SPK_EN |
 						 RT5670_JD_MODE2),
 	},
 	{
@@ -2924,6 +2954,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 		rt5670->pdata.dev_gpio = true;
 		dev_info(&i2c->dev, "quirk dev_gpio\n");
 	}
+	if (rt5670_quirk & RT5670_GPIO1_IS_EXT_SPK_EN) {
+		rt5670->pdata.gpio1_is_ext_spk_en = true;
+		dev_info(&i2c->dev, "quirk GPIO1 is external speaker enable\n");
+	}
 	if (rt5670_quirk & RT5670_IN2_DIFF) {
 		rt5670->pdata.in2_diff = true;
 		dev_info(&i2c->dev, "quirk IN2_DIFF\n");
@@ -3023,6 +3057,13 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 				   RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
 	}
 
+	if (rt5670->pdata.gpio1_is_ext_spk_en) {
+		regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+				   RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_GPIO1);
+		regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+				   RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
+	}
+
 	if (rt5670->pdata.jd_mode) {
 		regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK,
 				   RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index a8c3e44770b8..de0203369b7c 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -757,7 +757,7 @@
 #define RT5670_PWR_VREF2_BIT			4
 #define RT5670_PWR_FV2				(0x1 << 3)
 #define RT5670_PWR_FV2_BIT			3
-#define RT5670_LDO_SEL_MASK			(0x3)
+#define RT5670_LDO_SEL_MASK			(0x7)
 #define RT5670_LDO_SEL_SFT			0
 
 /* Power Management for Analog 2 (0x64) */
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 7d6670abdb08..d503b5bef4ba 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -967,13 +967,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 		rt5682_enable_push_button_irq(component, false);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
 			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
-		if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+		if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
 			snd_soc_component_update_bits(component,
-				RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
-		else
+				RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
+		if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
 			snd_soc_component_update_bits(component,
-				RT5682_PWR_ANLG_1,
-				RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
+				RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
 			RT5682_PWR_CBJ, 0);
 
@@ -992,16 +991,17 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
 
 	rt5682->hs_jack = hs_jack;
 
-	if (!rt5682->is_sdw) {
-		if (!hs_jack) {
-			regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
-				RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
-			regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
-				RT5682_POW_JDH | RT5682_POW_JDL, 0);
-			cancel_delayed_work_sync(&rt5682->jack_detect_work);
-			return 0;
-		}
+	if (!hs_jack) {
+		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+			RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+			RT5682_POW_JDH | RT5682_POW_JDL, 0);
+		cancel_delayed_work_sync(&rt5682->jack_detect_work);
 
+		return 0;
+	}
+
+	if (!rt5682->is_sdw) {
 		switch (rt5682->pdata.jd_src) {
 		case RT5682_JD1:
 			snd_soc_component_update_bits(component,
@@ -1082,7 +1082,8 @@ void rt5682_jack_detect_handler(struct work_struct *work)
 			/* jack was out, report jack type */
 			rt5682->jack_type =
 				rt5682_headset_detect(rt5682->component, 1);
-		} else {
+		} else if ((rt5682->jack_type & SND_JACK_HEADSET) ==
+			SND_JACK_HEADSET) {
 			/* jack is already in, report button event */
 			rt5682->jack_type = SND_JACK_HEADSET;
 			btn_type = rt5682_button_detect(rt5682->component);
@@ -1608,8 +1609,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
 		0, set_filter_clk, SND_SOC_DAPM_PRE_PMU),
 	SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
 		rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
-	SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
-		NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0),
 
 	/* ASRC */
@@ -2492,6 +2492,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw)
 	snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
 	snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
 				RT5682_PWR_MB, RT5682_PWR_MB);
+
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2");
+	snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+			RT5682_PWR_VREF2 | RT5682_PWR_FV2,
+			RT5682_PWR_VREF2);
+	usleep_range(55000, 60000);
+	snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+			RT5682_PWR_FV2, RT5682_PWR_FV2);
+
 	snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1");
 	snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F");
 	snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B");
@@ -2517,9 +2526,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw)
 	snd_soc_dapm_mutex_lock(dapm);
 
 	snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2");
 	if (!rt5682->jack_type)
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+				RT5682_PWR_VREF2 | RT5682_PWR_FV2 |
 				RT5682_PWR_MB, 0);
+
 	snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1");
 	snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F");
 	snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B");
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 06ba36595ddd..7cfc89602fc3 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0),
 
 /* Boost mixer */
 static const struct snd_kcontrol_new wm8974_boost_mixer[] = {
-SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0),
+SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1),
 };
 
 /* Input PGA */
@@ -474,6 +474,10 @@ static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		iface |= 0x0008;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
+		if ((fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_IB_IF ||
+		    (fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_NB_IF) {
+			return -EINVAL;
+		}
 		iface |= 0x00018;
 		break;
 	default:
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 9ad35d9940fe..97b4f5480a31 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -317,8 +317,8 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 	if (ret < 0)
 		goto out_put_node;
 
-	dai_link->dpcm_playback		= 1;
-	dai_link->dpcm_capture		= 1;
+	snd_soc_dai_link_set_capabilities(dai_link);
+
 	dai_link->ops			= &graph_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 55e9f8800b3e..04d4d28ed511 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -231,8 +231,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 	if (ret < 0)
 		goto out_put_node;
 
-	dai_link->dpcm_playback		= 1;
-	dai_link->dpcm_capture		= 1;
+	snd_soc_dai_link_set_capabilities(dai_link);
+
 	dai_link->ops			= &simple_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 5f96d7ac0a22..bed4d5f73d9c 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -354,6 +354,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = {
 	{
 		.name = "Codec DSP",
 		.stream_name = "Wake on Voice",
+		.capture_only = 1,
 		.ops = &bdw_rt5677_dsp_ops,
 		SND_SOC_DAILINK_REG(dsp),
 	},
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 9e5fc9430628..ecbc58e8a37f 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -543,8 +543,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 
 	if (cnt) {
 		ret = device_add_properties(codec_dev, props);
-		if (ret)
+		if (ret) {
+			put_device(codec_dev);
 			return ret;
+		}
 	}
 
 	devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios);
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 7a43c70a1378..22e432768edb 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -253,21 +253,20 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
 	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
 
 	/*
-	 * Default mode for SSP configuration is TDM 4 slot
+	 * Default mode for SSP configuration is TDM 4 slot. One board/design,
+	 * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The
+	 * second piggy-backed, output-only codec is inside the keyboard-dock
+	 * (which has extra speakers). Unlike the main rt5672 codec, we cannot
+	 * configure this codec, it is hard coded to use 2 channel 24 bit I2S.
+	 * Since we only support 2 channels anyways, there is no need for TDM
+	 * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere.
 	 */
-	ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
-				  SND_SOC_DAIFMT_DSP_B |
-				  SND_SOC_DAIFMT_IB_NF |
+	ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
+				  SND_SOC_DAIFMT_I2S     |
+				  SND_SOC_DAIFMT_NB_NF   |
 				  SND_SOC_DAIFMT_CBS_CFS);
 	if (ret < 0) {
-		dev_err(rtd->dev, "can't set format to TDM %d\n", ret);
-		return ret;
-	}
-
-	/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
-	ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24);
-	if (ret < 0) {
-		dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
+		dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
 		return ret;
 	}
 
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index f51b28d1b94d..92f51d0e9fe2 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -72,7 +72,7 @@ config SND_SOC_QDSP6_ASM_DAI
 
 config SND_SOC_QDSP6
 	tristate "SoC ALSA audio driver for QDSP6"
-	depends on QCOM_APR && HAS_DMA
+	depends on QCOM_APR
 	select SND_SOC_QDSP6_COMMON
 	select SND_SOC_QDSP6_CORE
 	select SND_SOC_QDSP6_AFE
diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index f45e5aaa4b30..9539b0d024fe 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -219,19 +219,32 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int rockchip_sound_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+	return snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
+			8000, 96000);
+}
+
 static const struct snd_soc_ops rockchip_sound_max98357a_ops = {
+	.startup = rockchip_sound_startup,
 	.hw_params = rockchip_sound_max98357a_hw_params,
 };
 
 static const struct snd_soc_ops rockchip_sound_rt5514_ops = {
+	.startup = rockchip_sound_startup,
 	.hw_params = rockchip_sound_rt5514_hw_params,
 };
 
 static const struct snd_soc_ops rockchip_sound_da7219_ops = {
+	.startup = rockchip_sound_startup,
 	.hw_params = rockchip_sound_da7219_hw_params,
 };
 
 static const struct snd_soc_ops rockchip_sound_dmic_ops = {
+	.startup = rockchip_sound_startup,
 	.hw_params = rockchip_sound_dmic_hw_params,
 };
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0f30f5aabaa8..2b8abf88ec60 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2573,6 +2573,33 @@ int snd_soc_register_component(struct device *dev,
 EXPORT_SYMBOL_GPL(snd_soc_register_component);
 
 /**
+ * snd_soc_unregister_component_by_driver - Unregister component using a given driver
+ * from the ASoC core
+ *
+ * @dev: The device to unregister
+ * @component_driver: The component driver to unregister
+ */
+void snd_soc_unregister_component_by_driver(struct device *dev,
+					    const struct snd_soc_component_driver *component_driver)
+{
+	struct snd_soc_component *component;
+
+	if (!component_driver)
+		return;
+
+	mutex_lock(&client_mutex);
+	component = snd_soc_lookup_component_nolocked(dev, component_driver->name);
+	if (!component)
+		goto out;
+
+	snd_soc_del_component_unlocked(component);
+
+out:
+	mutex_unlock(&client_mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_component_by_driver);
+
+/**
  * snd_soc_unregister_component - Unregister all related component
  * from the ASoC core
  *
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index b05e18b63a1c..457159975b01 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -391,6 +391,44 @@ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir)
 	return stream->channels_min;
 }
 
+/*
+ * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs
+ */
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
+{
+	struct snd_soc_dai_link_component *cpu;
+	struct snd_soc_dai_link_component *codec;
+	struct snd_soc_dai *dai;
+	bool supported[SNDRV_PCM_STREAM_LAST + 1];
+	int direction;
+	int i;
+
+	for_each_pcm_streams(direction) {
+		supported[direction] = true;
+
+		for_each_link_cpus(dai_link, i, cpu) {
+			dai = snd_soc_find_dai(cpu);
+			if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+				supported[direction] = false;
+				break;
+			}
+		}
+		if (!supported[direction])
+			continue;
+		for_each_link_codecs(dai_link, i, codec) {
+			dai = snd_soc_find_dai(codec);
+			if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+				supported[direction] = false;
+				break;
+			}
+		}
+	}
+
+	dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK];
+	dai_link->dpcm_capture  = supported[SNDRV_PCM_STREAM_CAPTURE];
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities);
+
 void snd_soc_dai_action(struct snd_soc_dai *dai,
 			int stream, int action)
 {
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index 11e5d7962370..4534a1c03e8e 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -48,7 +48,9 @@ EXPORT_SYMBOL_GPL(devm_snd_soc_register_dai);
 
 static void devm_component_release(struct device *dev, void *res)
 {
-	snd_soc_unregister_component(*(struct device **)res);
+	const struct snd_soc_component_driver **cmpnt_drv = res;
+
+	snd_soc_unregister_component_by_driver(dev, *cmpnt_drv);
 }
 
 /**
@@ -65,7 +67,7 @@ int devm_snd_soc_register_component(struct device *dev,
 			 const struct snd_soc_component_driver *cmpnt_drv,
 			 struct snd_soc_dai_driver *dai_drv, int num_dai)
 {
-	struct device **ptr;
+	const struct snd_soc_component_driver **ptr;
 	int ret;
 
 	ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL);
@@ -74,7 +76,7 @@ int devm_snd_soc_register_component(struct device *dev,
 
 	ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai);
 	if (ret == 0) {
-		*ptr = dev;
+		*ptr = cmpnt_drv;
 		devres_add(dev, ptr);
 	} else {
 		devres_free(ptr);
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 80a4e71f2d95..61844403f181 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -478,7 +478,7 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
 
 	pcm = soc_component_to_pcm(component);
 
-	snd_soc_unregister_component(dev);
+	snd_soc_unregister_component_by_driver(dev, component->driver);
 	dmaengine_pcm_release_chan(pcm);
 	kfree(pcm);
 }
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 43e5745b06aa..6eaa00c21011 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1261,17 +1261,29 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list);
 
 		ret = soc_tplg_add_route(tplg, routes[i]);
-		if (ret < 0)
+		if (ret < 0) {
+			/*
+			 * this route was added to the list, it will
+			 * be freed in remove_route() so increment the
+			 * counter to skip it in the error handling
+			 * below.
+			 */
+			i++;
 			break;
+		}
 
 		/* add route, but keep going if some fail */
 		snd_soc_dapm_add_routes(dapm, routes[i], 1);
 	}
 
-	/* free memory allocated for all dapm routes in case of error */
-	if (ret < 0)
-		for (i = 0; i < count ; i++)
-			kfree(routes[i]);
+	/*
+	 * free memory allocated for all dapm routes not added to the
+	 * list in case of error
+	 */
+	if (ret < 0) {
+		while (i < count)
+			kfree(routes[i++]);
+	}
 
 	/*
 	 * free pointer to array of dapm routes as this is no longer needed.
@@ -1359,7 +1371,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
 		if (err < 0) {
 			dev_err(tplg->dev, "ASoC: failed to init %s\n",
 				mc->hdr.name);
-			soc_tplg_free_tlv(tplg, &kc[i]);
 			goto err_sm;
 		}
 	}
@@ -1367,6 +1378,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
 
 err_sm:
 	for (; i >= 0; i--) {
+		soc_tplg_free_tlv(tplg, &kc[i]);
 		sm = (struct soc_mixer_control *)kc[i].private_value;
 		kfree(sm);
 		kfree(kc[i].name);
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 339c4930b0c0..adc7c37145d6 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -345,15 +345,15 @@ int snd_sof_device_remove(struct device *dev)
 	struct snd_sof_pdata *pdata = sdev->pdata;
 	int ret;
 
-	ret = snd_sof_dsp_power_down_notify(sdev);
-	if (ret < 0)
-		dev_warn(dev, "error: %d failed to prepare DSP for device removal",
-			 ret);
-
 	if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
 		cancel_work_sync(&sdev->probe_work);
 
 	if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) {
+		ret = snd_sof_dsp_power_down_notify(sdev);
+		if (ret < 0)
+			dev_warn(dev, "error: %d failed to prepare DSP for device removal",
+				 ret);
+
 		snd_sof_fw_unload(sdev);
 		snd_sof_ipc_free(sdev);
 		snd_sof_free_debug(sdev);
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 63f9c20a1bac..a4fa8451d8cb 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -375,6 +375,14 @@ static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev,
 static struct snd_soc_dai_driver imx8_dai[] = {
 {
 	.name = "esai-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 };
 
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index fa86a9e2990f..287114a37688 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -240,6 +240,14 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev,
 static struct snd_soc_dai_driver imx8m_dai[] = {
 {
 	.name = "sai-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
 },
 };