Merge tag 'sound-5.16-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Another collection of small fixes. It's still not quite calm yet, but
  nothing looks scary.

  ALSA core got a few fixes for covering the issues detected by fuzzer
  and the 32bit compat problem of control API, while the rest are all
  device-specific small fixes, including the continued fixes for Tegra"

* tag 'sound-5.16-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits)
  ALSA: hda/realtek - Add headset Mic support for Lenovo ALC897 platform
  ALSA: usb-audio: Reorder snd_djm_devices[] entries
  ALSA: hda/realtek: Fix quirk for TongFang PHxTxX1
  ALSA: ctl: Fix copy of updated id with element read/write
  ALSA: pcm: oss: Handle missing errors in snd_pcm_oss_change_params*()
  ALSA: pcm: oss: Limit the period size to 16MB
  ALSA: pcm: oss: Fix negative period/buffer sizes
  ASoC: codecs: wsa881x: fix return values from kcontrol put
  ASoC: codecs: wcd934x: return correct value from mixer put
  ASoC: codecs: wcd934x: handle channel mappping list correctly
  ASoC: qdsp6: q6routing: Fix return value from msm_routing_put_audio_mixer
  ASoC: SOF: Intel: Retry codec probing if it fails
  ASoC: amd: fix uninitialized variable in snd_acp6x_probe()
  ASoC: rockchip: i2s_tdm: Dup static DAI template
  ASoC: rt5682s: Fix crash due to out of scope stack vars
  ASoC: rt5682: Fix crash due to out of scope stack vars
  ASoC: tegra: Use normal system sleep for ADX
  ASoC: tegra: Use normal system sleep for AMX
  ASoC: tegra: Use normal system sleep for Mixer
  ASoC: tegra: Use normal system sleep for MVC
  ...

Documentation/devicetree/bindings/sound/wlf,wm8962.yaml

@@ -19,6 +19,9 @@ properties:
   clocks:
     maxItems: 1
 
+  interrupts:
+    maxItems: 1
+
   "#sound-dai-cells":
     const: 0
 

sound/core/control_compat.c

@@ -264,6 +264,7 @@ static int copy_ctl_value_to_user(void __user *userdata,
 				  struct snd_ctl_elem_value *data,
 				  int type, int count)
 {
+	struct snd_ctl_elem_value32 __user *data32 = userdata;
 	int i, size;
 
 	if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN ||
@@ -280,6 +281,8 @@ static int copy_ctl_value_to_user(void __user *userdata,
 		if (copy_to_user(valuep, data->value.bytes.data, size))
 			return -EFAULT;
 	}
+	if (copy_to_user(&data32->id, &data->id, sizeof(data32->id)))
+		return -EFAULT;
 	return 0;
 }
 

sound/core/oss/pcm_oss.c

@@ -147,7 +147,7 @@ snd_pcm_hw_param_value_min(const struct snd_pcm_hw_params *params,
  *
  * Return the maximum value for field PAR.
  */
-static unsigned int
+static int
 snd_pcm_hw_param_value_max(const struct snd_pcm_hw_params *params,
 			   snd_pcm_hw_param_t var, int *dir)
 {
@@ -682,18 +682,24 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *oss_params,
 				   struct snd_pcm_hw_params *slave_params)
 {
-	size_t s;
-	size_t oss_buffer_size, oss_period_size, oss_periods;
-	size_t min_period_size, max_period_size;
+	ssize_t s;
+	ssize_t oss_buffer_size;
+	ssize_t oss_period_size, oss_periods;
+	ssize_t min_period_size, max_period_size;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	size_t oss_frame_size;
 
 	oss_frame_size = snd_pcm_format_physical_width(params_format(oss_params)) *
 			 params_channels(oss_params) / 8;
 
+	oss_buffer_size = snd_pcm_hw_param_value_max(slave_params,
+						     SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+						     NULL);
+	if (oss_buffer_size <= 0)
+		return -EINVAL;
 	oss_buffer_size = snd_pcm_plug_client_size(substream,
-						   snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size;
-	if (!oss_buffer_size)
+						   oss_buffer_size * oss_frame_size);
+	if (oss_buffer_size <= 0)
 		return -EINVAL;
 	oss_buffer_size = rounddown_pow_of_two(oss_buffer_size);
 	if (atomic_read(&substream->mmap_count)) {
@@ -730,7 +736,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
 
 	min_period_size = snd_pcm_plug_client_size(substream,
 						   snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
-	if (min_period_size) {
+	if (min_period_size > 0) {
 		min_period_size *= oss_frame_size;
 		min_period_size = roundup_pow_of_two(min_period_size);
 		if (oss_period_size < min_period_size)
@@ -739,7 +745,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
 
 	max_period_size = snd_pcm_plug_client_size(substream,
 						   snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
-	if (max_period_size) {
+	if (max_period_size > 0) {
 		max_period_size *= oss_frame_size;
 		max_period_size = rounddown_pow_of_two(max_period_size);
 		if (oss_period_size > max_period_size)
@@ -752,7 +758,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
 		oss_periods = substream->oss.setup.periods;
 
 	s = snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIODS, NULL);
-	if (runtime->oss.maxfrags && s > runtime->oss.maxfrags)
+	if (s > 0 && runtime->oss.maxfrags && s > runtime->oss.maxfrags)
 		s = runtime->oss.maxfrags;
 	if (oss_periods > s)
 		oss_periods = s;
@@ -878,8 +884,15 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
 		err = -EINVAL;
 		goto failure;
 	}
-	choose_rate(substream, sparams, runtime->oss.rate);
-	snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_CHANNELS, runtime->oss.channels, NULL);
+
+	err = choose_rate(substream, sparams, runtime->oss.rate);
+	if (err < 0)
+		goto failure;
+	err = snd_pcm_hw_param_near(substream, sparams,
+				    SNDRV_PCM_HW_PARAM_CHANNELS,
+				    runtime->oss.channels, NULL);
+	if (err < 0)
+		goto failure;
 
 	format = snd_pcm_oss_format_from(runtime->oss.format);
 
@@ -1956,7 +1969,7 @@ static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsign
 	if (runtime->oss.subdivision || runtime->oss.fragshift)
 		return -EINVAL;
 	fragshift = val & 0xffff;
-	if (fragshift >= 31)
+	if (fragshift >= 25) /* should be large enough */
 		return -EINVAL;
 	runtime->oss.fragshift = fragshift;
 	runtime->oss.maxfrags = (val >> 16) & 0xffff;

sound/pci/hda/patch_realtek.c

@@ -6503,22 +6503,26 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec,
 /* for alc285_fixup_ideapad_s740_coef() */
 #include "ideapad_s740_helper.c"
 
-static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *codec,
-							    const struct hda_fixup *fix,
-							    int action)
+static const struct coef_fw alc256_fixup_set_coef_defaults_coefs[] = {
+	WRITE_COEF(0x10, 0x0020), WRITE_COEF(0x24, 0x0000),
+	WRITE_COEF(0x26, 0x0000), WRITE_COEF(0x29, 0x3000),
+	WRITE_COEF(0x37, 0xfe05), WRITE_COEF(0x45, 0x5089),
+	{}
+};
+
+static void alc256_fixup_set_coef_defaults(struct hda_codec *codec,
+					   const struct hda_fixup *fix,
+					   int action)
 {
 	/*
-	* A certain other OS sets these coeffs to different values. On at least one TongFang
-	* barebone these settings might survive even a cold reboot. So to restore a clean slate the
-	* values are explicitly reset to default here. Without this, the external microphone is
-	* always in a plugged-in state, while the internal microphone is always in an unplugged
-	* state, breaking the ability to use the internal microphone.
-	*/
-	alc_write_coef_idx(codec, 0x24, 0x0000);
-	alc_write_coef_idx(codec, 0x26, 0x0000);
-	alc_write_coef_idx(codec, 0x29, 0x3000);
-	alc_write_coef_idx(codec, 0x37, 0xfe05);
-	alc_write_coef_idx(codec, 0x45, 0x5089);
+	 * A certain other OS sets these coeffs to different values. On at least
+	 * one TongFang barebone these settings might survive even a cold
+	 * reboot. So to restore a clean slate the values are explicitly reset
+	 * to default here. Without this, the external microphone is always in a
+	 * plugged-in state, while the internal microphone is always in an
+	 * unplugged state, breaking the ability to use the internal microphone.
+	 */
+	alc_process_coef_fw(codec, alc256_fixup_set_coef_defaults_coefs);
 }
 
 static const struct coef_fw alc233_fixup_no_audio_jack_coefs[] = {
@@ -6759,7 +6763,7 @@ enum {
 	ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE,
 	ALC287_FIXUP_YOGA7_14ITL_SPEAKERS,
 	ALC287_FIXUP_13S_GEN2_SPEAKERS,
-	ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS,
+	ALC256_FIXUP_SET_COEF_DEFAULTS,
 	ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE,
 	ALC233_FIXUP_NO_AUDIO_JACK,
 };
@@ -8465,9 +8469,9 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE,
 	},
-	[ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS] = {
+	[ALC256_FIXUP_SET_COEF_DEFAULTS] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = alc256_fixup_tongfang_reset_persistent_settings,
+		.v.func = alc256_fixup_set_coef_defaults,
 	},
 	[ALC245_FIXUP_HP_GPIO_LED] = {
 		.type = HDA_FIXUP_FUNC,
@@ -8929,7 +8933,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
 	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
 	SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
-	SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS),
+	SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS),
 	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
@@ -10231,6 +10235,27 @@ static void alc671_fixup_hp_headset_mic2(struct hda_codec *codec,
 	}
 }
 
+static void alc897_hp_automute_hook(struct hda_codec *codec,
+					 struct hda_jack_callback *jack)
+{
+	struct alc_spec *spec = codec->spec;
+	int vref;
+
+	snd_hda_gen_hp_automute(codec, jack);
+	vref = spec->gen.hp_jack_present ? (PIN_HP | AC_PINCTL_VREF_100) : PIN_HP;
+	snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    vref);
+}
+
+static void alc897_fixup_lenovo_headset_mic(struct hda_codec *codec,
+				     const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->gen.hp_automute_hook = alc897_hp_automute_hook;
+	}
+}
+
 static const struct coef_fw alc668_coefs[] = {
 	WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03,    0x0),
 	WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06,    0x0), WRITE_COEF(0x07, 0x0f80),
@@ -10311,6 +10336,8 @@ enum {
 	ALC668_FIXUP_ASUS_NO_HEADSET_MIC,
 	ALC668_FIXUP_HEADSET_MIC,
 	ALC668_FIXUP_MIC_DET_COEF,
+	ALC897_FIXUP_LENOVO_HEADSET_MIC,
+	ALC897_FIXUP_HEADSET_MIC_PIN,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -10717,6 +10744,19 @@ static const struct hda_fixup alc662_fixups[] = {
 			{}
 		},
 	},
+	[ALC897_FIXUP_LENOVO_HEADSET_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc897_fixup_lenovo_headset_mic,
+	},
+	[ALC897_FIXUP_HEADSET_MIC_PIN] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x03a11050 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC897_FIXUP_LENOVO_HEADSET_MIC
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -10761,6 +10801,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE),
 	SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS),
+	SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN),
+	SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO),

sound/soc/amd/yc/pci-acp6x.c

@@ -146,10 +146,11 @@ static int snd_acp6x_probe(struct pci_dev *pci,
 {
 	struct acp6x_dev_data *adata;
 	struct platform_device_info pdevinfo[ACP6x_DEVS];
-	int ret, index;
+	int index = 0;
 	int val = 0x00;
 	u32 addr;
 	unsigned int irqflags;
+	int ret;
 
 	irqflags = IRQF_SHARED;
 	/* Yellow Carp device check */

sound/soc/codecs/rt5682.c

@@ -2858,6 +2858,8 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682)
 
 	for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) {
 		struct clk_init_data init = { };
+		struct clk_parent_data parent_data;
+		const struct clk_hw *parent;
 
 		dai_clk_hw = &rt5682->dai_clks_hw[i];
 
@@ -2865,17 +2867,17 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682)
 		case RT5682_DAI_WCLK_IDX:
 			/* Make MCLK the parent of WCLK */
 			if (rt5682->mclk) {
-				init.parent_data = &(struct clk_parent_data){
+				parent_data = (struct clk_parent_data){
 					.fw_name = "mclk",
 				};
+				init.parent_data = &parent_data;
 				init.num_parents = 1;
 			}
 			break;
 		case RT5682_DAI_BCLK_IDX:
 			/* Make WCLK the parent of BCLK */
-			init.parent_hws = &(const struct clk_hw *){
-				&rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX]
-			};
+			parent = &rt5682->dai_clks_hw[RT5682_DAI_WCLK_IDX];
+			init.parent_hws = &parent;
 			init.num_parents = 1;
 			break;
 		default:

sound/soc/codecs/rt5682s.c

@@ -2693,6 +2693,8 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component)
 
 	for (i = 0; i < RT5682S_DAI_NUM_CLKS; ++i) {
 		struct clk_init_data init = { };
+		struct clk_parent_data parent_data;
+		const struct clk_hw *parent;
 
 		dai_clk_hw = &rt5682s->dai_clks_hw[i];
 
@@ -2700,17 +2702,17 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component)
 		case RT5682S_DAI_WCLK_IDX:
 			/* Make MCLK the parent of WCLK */
 			if (rt5682s->mclk) {
-				init.parent_data = &(struct clk_parent_data){
+				parent_data = (struct clk_parent_data){
 					.fw_name = "mclk",
 				};
+				init.parent_data = &parent_data;
 				init.num_parents = 1;
 			}
 			break;
 		case RT5682S_DAI_BCLK_IDX:
 			/* Make WCLK the parent of BCLK */
-			init.parent_hws = &(const struct clk_hw *){
-				&rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX]
-			};
+			parent = &rt5682s->dai_clks_hw[RT5682S_DAI_WCLK_IDX];
+			init.parent_hws = &parent;
 			init.num_parents = 1;
 			break;
 		default:

sound/soc/codecs/wcd934x.c

@@ -3256,6 +3256,9 @@ static int wcd934x_compander_set(struct snd_kcontrol *kc,
 	int value = ucontrol->value.integer.value[0];
 	int sel;
 
+	if (wcd->comp_enabled[comp] == value)
+		return 0;
+
 	wcd->comp_enabled[comp] = value;
 	sel = value ? WCD934X_HPH_GAIN_SRC_SEL_COMPANDER :
 		WCD934X_HPH_GAIN_SRC_SEL_REGISTER;
@@ -3279,10 +3282,10 @@ static int wcd934x_compander_set(struct snd_kcontrol *kc,
 	case COMPANDER_8:
 		break;
 	default:
-		break;
+		return 0;
 	}
 
-	return 0;
+	return 1;
 }
 
 static int wcd934x_rx_hph_mode_get(struct snd_kcontrol *kc,
@@ -3326,6 +3329,31 @@ static int slim_rx_mux_get(struct snd_kcontrol *kc,
 	return 0;
 }
 
+static int slim_rx_mux_to_dai_id(int mux)
+{
+	int aif_id;
+
+	switch (mux) {
+	case 1:
+		aif_id = AIF1_PB;
+		break;
+	case 2:
+		aif_id = AIF2_PB;
+		break;
+	case 3:
+		aif_id = AIF3_PB;
+		break;
+	case 4:
+		aif_id = AIF4_PB;
+		break;
+	default:
+		aif_id = -1;
+		break;
+	}
+
+	return aif_id;
+}
+
 static int slim_rx_mux_put(struct snd_kcontrol *kc,
 			   struct snd_ctl_elem_value *ucontrol)
 {
@@ -3333,43 +3361,59 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc,
 	struct wcd934x_codec *wcd = dev_get_drvdata(w->dapm->dev);
 	struct soc_enum *e = (struct soc_enum *)kc->private_value;
 	struct snd_soc_dapm_update *update = NULL;
+	struct wcd934x_slim_ch *ch, *c;
 	u32 port_id = w->shift;
+	bool found = false;
+	int mux_idx;
+	int prev_mux_idx = wcd->rx_port_value[port_id];
+	int aif_id;
 
-	if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0])
-		return 0;
+	mux_idx = ucontrol->value.enumerated.item[0];
 
-	wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0];
+	if (mux_idx == prev_mux_idx)
+		return 0;
 
-	switch (wcd->rx_port_value[port_id]) {
+	switch(mux_idx) {
 	case 0:
-		list_del_init(&wcd->rx_chs[port_id].list);
-		break;
-	case 1:
-		list_add_tail(&wcd->rx_chs[port_id].list,
-			      &wcd->dai[AIF1_PB].slim_ch_list);
-		break;
-	case 2:
-		list_add_tail(&wcd->rx_chs[port_id].list,
-			      &wcd->dai[AIF2_PB].slim_ch_list);
-		break;
-	case 3:
-		list_add_tail(&wcd->rx_chs[port_id].list,
-			      &wcd->dai[AIF3_PB].slim_ch_list);
+		aif_id = slim_rx_mux_to_dai_id(prev_mux_idx);
+		if (aif_id < 0)
+			return 0;
+
+		list_for_each_entry_safe(ch, c, &wcd->dai[aif_id].slim_ch_list, list) {
+			if (ch->port == port_id + WCD934X_RX_START) {
+				found = true;
+				list_del_init(&ch->list);
+				break;
+			}
+		}
+		if (!found)
+			return 0;
+
 		break;
-	case 4:
-		list_add_tail(&wcd->rx_chs[port_id].list,
-			      &wcd->dai[AIF4_PB].slim_ch_list);
+	case 1 ... 4:
+		aif_id = slim_rx_mux_to_dai_id(mux_idx);
+		if (aif_id < 0)
+			return 0;
+
+		if (list_empty(&wcd->rx_chs[port_id].list)) {
+			list_add_tail(&wcd->rx_chs[port_id].list,
+				      &wcd->dai[aif_id].slim_ch_list);
+		} else {
+			dev_err(wcd->dev ,"SLIM_RX%d PORT is busy\n", port_id);
+			return 0;
+		}
 		break;
+
 	default:
-		dev_err(wcd->dev, "Unknown AIF %d\n",
-			wcd->rx_port_value[port_id]);
+		dev_err(wcd->dev, "Unknown AIF %d\n", mux_idx);
 		goto err;
 	}
 
+	wcd->rx_port_value[port_id] = mux_idx;
 	snd_soc_dapm_mux_update_power(w->dapm, kc, wcd->rx_port_value[port_id],
 				      e, update);
 
-	return 0;
+	return 1;
 err:
 	return -EINVAL;
 }
@@ -3815,6 +3859,7 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc,
 	struct soc_mixer_control *mixer =
 			(struct soc_mixer_control *)kc->private_value;
 	int enable = ucontrol->value.integer.value[0];
+	struct wcd934x_slim_ch *ch, *c;
 	int dai_id = widget->shift;
 	int port_id = mixer->shift;
 
@@ -3822,17 +3867,32 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc,
 	if (enable == wcd->tx_port_value[port_id])
 		return 0;
 
-	wcd->tx_port_value[port_id] = enable;
-
-	if (enable)
-		list_add_tail(&wcd->tx_chs[port_id].list,
-			      &wcd->dai[dai_id].slim_ch_list);
-	else
-		list_del_init(&wcd->tx_chs[port_id].list);
+	if (enable) {
+		if (list_empty(&wcd->tx_chs[port_id].list)) {
+			list_add_tail(&wcd->tx_chs[port_id].list,
+				      &wcd->dai[dai_id].slim_ch_list);
+		} else {
+			dev_err(wcd->dev ,"SLIM_TX%d PORT is busy\n", port_id);
+			return 0;
+		}
+	 } else {
+		bool found = false;
+
+		list_for_each_entry_safe(ch, c, &wcd->dai[dai_id].slim_ch_list, list) {
+			if (ch->port == port_id) {
+				found = true;
+				list_del_init(&wcd->tx_chs[port_id].list);
+				break;
+			}
+		}
+		if (!found)
+			return 0;
+	 }
 
+	wcd->tx_port_value[port_id] = enable;
 	snd_soc_dapm_mixer_update_power(widget->dapm, kc, enable, update);
 
-	return 0;
+	return 1;
 }
 
 static const struct snd_kcontrol_new aif1_slim_cap_mixer[] = {

sound/soc/codecs/wsa881x.c

@@ -772,7 +772,8 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc,
 
 		usleep_range(1000, 1010);
 	}
-	return 0;
+
+	return 1;
 }
 
 static int wsa881x_get_port(struct snd_kcontrol *kcontrol,
@@ -816,15 +817,22 @@ static int wsa881x_set_port(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	int portidx = mixer->reg;
 
-	if (ucontrol->value.integer.value[0])
+	if (ucontrol->value.integer.value[0]) {
+		if (data->port_enable[portidx])
+			return 0;
+
 		data->port_enable[portidx] = true;
-	else
+	} else {
+		if (!data->port_enable[portidx])
+			return 0;
+
 		data->port_enable[portidx] = false;
+	}
 
 	if (portidx == WSA881X_PORT_BOOST) /* Boost Switch */
 		wsa881x_boost_ctrl(comp, data->port_enable[portidx]);
 
-	return 0;
+	return 1;
 }
 
 static const char * const smart_boost_lvl_text[] = {

sound/soc/qcom/qdsp6/q6routing.c

@@ -498,14 +498,16 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
 	struct session_data *session = &data->sessions[session_id];
 
 	if (ucontrol->value.integer.value[0]) {
+		if (session->port_id == be_id)
+			return 0;
+
 		session->port_id = be_id;
 		snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update);
 	} else {
-		if (session->port_id == be_id) {
-			session->port_id = -1;
+		if (session->port_id == -1 || session->port_id != be_id)
 			return 0;
-		}
 
+		session->port_id = -1;
 		snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update);
 	}
 

sound/soc/rockchip/rockchip_i2s_tdm.c

@@ -95,6 +95,7 @@ struct rk_i2s_tdm_dev {
 	spinlock_t lock; /* xfer lock */
 	bool has_playback;
 	bool has_capture;
+	struct snd_soc_dai_driver *dai;
 };
 
 static int to_ch_num(unsigned int val)
@@ -1310,19 +1311,14 @@ static const struct of_device_id rockchip_i2s_tdm_match[] = {
 	{},
 };
 
-static struct snd_soc_dai_driver i2s_tdm_dai = {
+static const struct snd_soc_dai_driver i2s_tdm_dai = {
 	.probe = rockchip_i2s_tdm_dai_probe,
-	.playback = {
-		.stream_name  = "Playback",
-	},
-	.capture = {
-		.stream_name  = "Capture",
-	},
 	.ops = &rockchip_i2s_tdm_dai_ops,
 };
 
-static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm)
+static int rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm)
 {
+	struct snd_soc_dai_driver *dai;
 	struct property *dma_names;
 	const char *dma_name;
 	u64 formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |
@@ -1337,19 +1333,33 @@ static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm)
 			i2s_tdm->has_capture = true;
 	}
 
+	dai = devm_kmemdup(i2s_tdm->dev, &i2s_tdm_dai,
+			   sizeof(*dai), GFP_KERNEL);
+	if (!dai)
+		return -ENOMEM;
+
 	if (i2s_tdm->has_playback) {
-		i2s_tdm_dai.playback.channels_min = 2;
-		i2s_tdm_dai.playback.channels_max = 8;
-		i2s_tdm_dai.playback.rates = SNDRV_PCM_RATE_8000_192000;
-		i2s_tdm_dai.playback.formats = formats;
+		dai->playback.stream_name  = "Playback";
+		dai->playback.channels_min = 2;
+		dai->playback.channels_max = 8;
+		dai->playback.rates = SNDRV_PCM_RATE_8000_192000;
+		dai->playback.formats = formats;
 	}
 
 	if (i2s_tdm->has_capture) {
-		i2s_tdm_dai.capture.channels_min = 2;
-		i2s_tdm_dai.capture.channels_max = 8;
-		i2s_tdm_dai.capture.rates = SNDRV_PCM_RATE_8000_192000;
-		i2s_tdm_dai.capture.formats = formats;
+		dai->capture.stream_name  = "Capture";
+		dai->capture.channels_min = 2;
+		dai->capture.channels_max = 8;
+		dai->capture.rates = SNDRV_PCM_RATE_8000_192000;
+		dai->capture.formats = formats;
 	}
+
+	if (i2s_tdm->clk_trcm != TRCM_TXRX)
+		dai->symmetric_rate = 1;
+
+	i2s_tdm->dai = dai;
+
+	return 0;
 }
 
 static int rockchip_i2s_tdm_path_check(struct rk_i2s_tdm_dev *i2s_tdm,
@@ -1541,8 +1551,6 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev)
 	spin_lock_init(&i2s_tdm->lock);
 	i2s_tdm->soc_data = (struct rk_i2s_soc_data *)of_id->data;
 
-	rockchip_i2s_tdm_init_dai(i2s_tdm);
-
 	i2s_tdm->frame_width = 64;
 
 	i2s_tdm->clk_trcm = TRCM_TXRX;
@@ -1555,8 +1563,10 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev)
 		}
 		i2s_tdm->clk_trcm = TRCM_RX;
 	}
-	if (i2s_tdm->clk_trcm != TRCM_TXRX)
-		i2s_tdm_dai.symmetric_rate = 1;
+
+	ret = rockchip_i2s_tdm_init_dai(i2s_tdm);
+	if (ret)
+		return ret;
 
 	i2s_tdm->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf");
 	if (IS_ERR(i2s_tdm->grf))
@@ -1678,7 +1688,7 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev)
 
 	ret = devm_snd_soc_register_component(&pdev->dev,
 					      &rockchip_i2s_tdm_component,
-					      &i2s_tdm_dai, 1);
+					      i2s_tdm->dai, 1);
 
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register DAI\n");

sound/soc/sof/intel/hda-codec.c

@@ -22,6 +22,7 @@
 
 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC)
 #define IDISP_VID_INTEL	0x80860000
+#define CODEC_PROBE_RETRIES 3
 
 /* load the legacy HDA codec driver */
 static int request_codec_module(struct hda_codec *codec)
@@ -121,12 +122,15 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address,
 	u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) |
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	u32 resp = -1;
-	int ret;
+	int ret, retry = 0;
+
+	do {
+		mutex_lock(&hbus->core.cmd_mutex);
+		snd_hdac_bus_send_cmd(&hbus->core, hda_cmd);
+		snd_hdac_bus_get_response(&hbus->core, address, &resp);
+		mutex_unlock(&hbus->core.cmd_mutex);
+	} while (resp == -1 && retry++ < CODEC_PROBE_RETRIES);
 
-	mutex_lock(&hbus->core.cmd_mutex);
-	snd_hdac_bus_send_cmd(&hbus->core, hda_cmd);
-	snd_hdac_bus_get_response(&hbus->core, address, &resp);
-	mutex_unlock(&hbus->core.cmd_mutex);
 	if (resp == -1)
 		return -EIO;
 	dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n",

sound/soc/tegra/tegra210_adx.c

@@ -514,8 +514,8 @@ static int tegra210_adx_platform_remove(struct platform_device *pdev)
 static const struct dev_pm_ops tegra210_adx_pm_ops = {
 	SET_RUNTIME_PM_OPS(tegra210_adx_runtime_suspend,
 			   tegra210_adx_runtime_resume, NULL)
-	SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
-				     pm_runtime_force_resume)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
 };
 
 static struct platform_driver tegra210_adx_driver = {

sound/soc/tegra/tegra210_amx.c

@@ -583,8 +583,8 @@ static int tegra210_amx_platform_remove(struct platform_device *pdev)
 static const struct dev_pm_ops tegra210_amx_pm_ops = {
 	SET_RUNTIME_PM_OPS(tegra210_amx_runtime_suspend,
 			   tegra210_amx_runtime_resume, NULL)
-	SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
-				     pm_runtime_force_resume)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
 };
 
 static struct platform_driver tegra210_amx_driver = {

sound/soc/tegra/tegra210_mixer.c

@@ -666,8 +666,8 @@ static int tegra210_mixer_platform_remove(struct platform_device *pdev)
 static const struct dev_pm_ops tegra210_mixer_pm_ops = {
 	SET_RUNTIME_PM_OPS(tegra210_mixer_runtime_suspend,
 			   tegra210_mixer_runtime_resume, NULL)
-	SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
-				     pm_runtime_force_resume)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
 };
 
 static struct platform_driver tegra210_mixer_driver = {

sound/soc/tegra/tegra210_mvc.c

@@ -164,7 +164,7 @@ static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol,
 	if (err < 0)
 		goto end;
 
-	return 1;
+	err = 1;
 
 end:
 	pm_runtime_put(cmpnt->dev);
@@ -236,7 +236,7 @@ static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol,
 			   TEGRA210_MVC_VOLUME_SWITCH_MASK,
 			   TEGRA210_MVC_VOLUME_SWITCH_TRIGGER);
 
-	return 1;
+	err = 1;
 
 end:
 	pm_runtime_put(cmpnt->dev);
@@ -639,8 +639,8 @@ static int tegra210_mvc_platform_remove(struct platform_device *pdev)
 static const struct dev_pm_ops tegra210_mvc_pm_ops = {
 	SET_RUNTIME_PM_OPS(tegra210_mvc_runtime_suspend,
 			   tegra210_mvc_runtime_resume, NULL)
-	SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
-				     pm_runtime_force_resume)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
 };
 
 static struct platform_driver tegra210_mvc_driver = {

sound/soc/tegra/tegra210_sfc.c

@@ -3594,8 +3594,8 @@ static int tegra210_sfc_platform_remove(struct platform_device *pdev)
 static const struct dev_pm_ops tegra210_sfc_pm_ops = {
 	SET_RUNTIME_PM_OPS(tegra210_sfc_runtime_suspend,
 			   tegra210_sfc_runtime_resume, NULL)
-	SET_LATE_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
-				     pm_runtime_force_resume)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
 };
 
 static struct platform_driver tegra210_sfc_driver = {

sound/usb/mixer_quirks.c

@@ -3016,11 +3016,11 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = {
 
 
 static const struct snd_djm_device snd_djm_devices[] = {
-	SND_DJM_DEVICE(250mk2),
-	SND_DJM_DEVICE(750),
-	SND_DJM_DEVICE(750mk2),
-	SND_DJM_DEVICE(850),
-	SND_DJM_DEVICE(900nxs2)
+	[SND_DJM_250MK2_IDX] = SND_DJM_DEVICE(250mk2),
+	[SND_DJM_750_IDX] = SND_DJM_DEVICE(750),
+	[SND_DJM_850_IDX] = SND_DJM_DEVICE(850),
+	[SND_DJM_900NXS2_IDX] = SND_DJM_DEVICE(900nxs2),
+	[SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2),
 };