Merge remote-tracking branch 'torvalds/master' into perf/core

Minor conflict in tools/perf/arch/arm/util/auxtrace.c as one fix there
was cherry-picked for the last perf/urgent pull req to Linus, so was
already there.

Signed-off-by: Arnaldo Carvalho de Melo <acme@redhat.com>

24 files changed, 267 insertions, 97 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/amd/renoir/Makefile b/sound/soc/amd/renoir/Makefile
index e4371932a55a..4a82690aec16 100644
--- a/sound/soc/amd/renoir/Makefile
+++ b/sound/soc/amd/renoir/Makefile
@@ -2,6 +2,7 @@
 # Renoir platform Support
 snd-rn-pci-acp3x-objs	:= rn-pci-acp3x.o
 snd-acp3x-pdm-dma-objs	:= acp3x-pdm-dma.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR)	 += snd-rn-pci-acp3x.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR)	 += snd-acp3x-pdm-dma.o
-obj-$(CONFIG_SND_SOC_AMD_RENOIR_MACH)	+= acp3x-rn.o
+snd-acp3x-rn-objs	:= acp3x-rn.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR)	+= snd-rn-pci-acp3x.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR)	+= snd-acp3x-pdm-dma.o
+obj-$(CONFIG_SND_SOC_AMD_RENOIR_MACH)	+= snd-acp3x-rn.o
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 3e9d2c6c51f9..d503b5bef4ba 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -932,7 +932,9 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 			RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
 			RT5682_PWR_CBJ, RT5682_PWR_CBJ);
-
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1,
+			RT5682_OSW_L_MASK | RT5682_OSW_R_MASK, 0);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
 			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH);
 
@@ -956,17 +958,21 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 			rt5682->jack_type = SND_JACK_HEADPHONE;
 			break;
 		}
+
+		snd_soc_component_update_bits(component,
+			RT5682_HP_CHARGE_PUMP_1,
+			RT5682_OSW_L_MASK | RT5682_OSW_R_MASK,
+			RT5682_OSW_L_EN | RT5682_OSW_R_EN);
 	} else {
 		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);
 
@@ -985,16 +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,
@@ -1075,7 +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);
@@ -1601,8 +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 */
@@ -2485,6 +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");
@@ -2510,9 +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/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c
index 0c813a45bba7..69aeb0e71844 100644
--- a/sound/soc/fsl/fsl_mqs.c
+++ b/sound/soc/fsl/fsl_mqs.c
@@ -265,12 +265,20 @@ static int fsl_mqs_remove(struct platform_device *pdev)
 static int fsl_mqs_runtime_resume(struct device *dev)
 {
 	struct fsl_mqs *mqs_priv = dev_get_drvdata(dev);
+	int ret;
 
-	if (mqs_priv->ipg)
-		clk_prepare_enable(mqs_priv->ipg);
+	ret = clk_prepare_enable(mqs_priv->ipg);
+	if (ret) {
+		dev_err(dev, "failed to enable ipg clock\n");
+		return ret;
+	}
 
-	if (mqs_priv->mclk)
-		clk_prepare_enable(mqs_priv->mclk);
+	ret = clk_prepare_enable(mqs_priv->mclk);
+	if (ret) {
+		dev_err(dev, "failed to enable mclk clock\n");
+		clk_disable_unprepare(mqs_priv->ipg);
+		return ret;
+	}
 
 	if (mqs_priv->use_gpr)
 		regmap_write(mqs_priv->regmap, IOMUXC_GPR2,
@@ -292,11 +300,8 @@ static int fsl_mqs_runtime_suspend(struct device *dev)
 		regmap_read(mqs_priv->regmap, REG_MQS_CTRL,
 			    &mqs_priv->reg_mqs_ctrl);
 
-	if (mqs_priv->mclk)
-		clk_disable_unprepare(mqs_priv->mclk);
-
-	if (mqs_priv->ipg)
-		clk_disable_unprepare(mqs_priv->ipg);
+	clk_disable_unprepare(mqs_priv->mclk);
+	clk_disable_unprepare(mqs_priv->ipg);
 
 	return 0;
 }
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,
+	},
 },
 };