Merge tag 'sound-5.19-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Hopefully the last one for 5.19. This became bigger than wished, but
  all changes are pretty device-specific small fixes, which look less
  worrisome.

  The majority of changes are about various ASoC fixes, while the usual
  HD-audio quirks are included as well"

* tag 'sound-5.19-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (28 commits)
  ALSA: hda/realtek - Enable the headset-mic on a Xiaomi's laptop
  ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc221
  ALSA: hda/realtek: fix mute/micmute LEDs for HP machines
  ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc671
  ALSA: hda - Add fixup for Dell Latitidue E5430
  ALSA: hda/conexant: Apply quirk for another HP ProDesk 600 G3 model
  ALSA: hda/realtek: Fix headset mic for Acer SF313-51
  ASoC: Intel: Skylake: Correct the handling of fmt_config flexible array
  ASoC: Intel: Skylake: Correct the ssp rate discovery in skl_get_ssp_clks()
  ASoC: rt5640: Fix the wrong state of JD1 and JD2
  ASoC: Intel: sof_rt5682: fix out-of-bounds array access
  ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph()
  ASoC: tas2764: Fix amp gain register offset & default
  ASoC: tas2764: Correct playback volume range
  ASoC: tas2764: Fix and extend FSYNC polarity handling
  ASoC: tas2764: Add post reset delays
  ASoC: dt-bindings: Fix description for msm8916
  ASoC: doc: Capitalize RESET line name
  ASoC: arizona: Update arizona_aif_cfg_changed to use RX_BCLK_RATE
  ASoC: cs47l92: Fix event generation for OUT1 demux
  ...

Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml

@@ -25,12 +25,12 @@ properties:
       - qcom,sc7280-lpass-cpu
 
   reg:
-    minItems: 2
+    minItems: 1
     maxItems: 6
     description: LPAIF core registers
 
   reg-names:
-    minItems: 2
+    minItems: 1
     maxItems: 6
 
   clocks:
@@ -42,12 +42,12 @@ properties:
     maxItems: 10
 
   interrupts:
-    minItems: 2
+    minItems: 1
     maxItems: 4
     description: LPAIF DMA buffer interrupt
 
   interrupt-names:
-    minItems: 2
+    minItems: 1
     maxItems: 4
 
   qcom,adsp:

Documentation/sound/soc/dai.rst

@@ -10,7 +10,7 @@ AC97
 ====
 
 AC97 is a five wire interface commonly found on many PC sound cards. It is
-now also popular in many portable devices. This DAI has a reset line and time
+now also popular in many portable devices. This DAI has a RESET line and time
 multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
 The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
 frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97

sound/pci/hda/patch_conexant.c

@@ -944,6 +944,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x103c, 0x82b4, "HP ProDesk 600 G3", CXT_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),

sound/pci/hda/patch_realtek.c

@@ -6901,6 +6901,7 @@ enum {
 	ALC298_FIXUP_LENOVO_SPK_VOLUME,
 	ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER,
 	ALC269_FIXUP_ATIV_BOOK_8,
+	ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE,
 	ALC221_FIXUP_HP_MIC_NO_PRESENCE,
 	ALC256_FIXUP_ASUS_HEADSET_MODE,
 	ALC256_FIXUP_ASUS_MIC,
@@ -7837,6 +7838,16 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_NO_SHUTUP
 	},
+	[ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ 0x1a, 0x01813030 }, /* use as headphone mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE
+	},
 	[ALC221_FIXUP_HP_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -8886,6 +8897,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1025, 0x129d, "Acer SWIFT SF313-51", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
@@ -8895,6 +8907,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+	SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X),
 	SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
 	SND_PCI_QUIRK(0x1028, 0x05be, "Dell Latitude E6540", ALC292_FIXUP_DELL_E7X),
@@ -9010,6 +9023,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+	SND_PCI_QUIRK(0x103c, 0x2b5e, "HP 288 Pro G2 MT", ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC),
@@ -9096,6 +9110,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -9355,6 +9373,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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),
+	SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC),
 	SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
@@ -11217,6 +11236,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
 	SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB),
 	SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2),
+	SND_PCI_QUIRK(0x103c, 0x877e, "HP 288 Pro G6", ALC671_FIXUP_HP_HEADSET_MIC2),
 	SND_PCI_QUIRK(0x103c, 0x885f, "HP 288 Pro G8", ALC671_FIXUP_HP_HEADSET_MIC2),
 	SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE),
 	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),

sound/soc/codecs/arizona.c

@@ -1760,8 +1760,8 @@ static bool arizona_aif_cfg_changed(struct snd_soc_component *component,
 	if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK))
 		return true;
 
-	val = snd_soc_component_read(component, base + ARIZONA_AIF_TX_BCLK_RATE);
-	if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK))
+	val = snd_soc_component_read(component, base + ARIZONA_AIF_RX_BCLK_RATE);
+	if (lrclk != (val & ARIZONA_AIF1RX_BCPF_MASK))
 		return true;
 
 	val = snd_soc_component_read(component, base + ARIZONA_AIF_FRAME_CTRL_1);

sound/soc/codecs/cs47l92.c

@@ -119,7 +119,13 @@ static int cs47l92_put_demux(struct snd_kcontrol *kcontrol,
 end:
 	snd_soc_dapm_mutex_unlock(dapm);
 
-	return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+	ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
+	if (ret < 0) {
+		dev_err(madera->dev, "Failed to update demux power state: %d\n", ret);
+		return ret;
+	}
+
+	return change;
 }
 
 static SOC_ENUM_SINGLE_DECL(cs47l92_outdemux_enum,

sound/soc/codecs/max98396.c

@@ -342,12 +342,15 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	struct max98396_priv *max98396 = snd_soc_component_get_drvdata(component);
-	unsigned int format = 0;
+	unsigned int format_mask, format = 0;
 	unsigned int bclk_pol = 0;
 	int ret, status;
 	int reg;
 	bool update = false;
 
+	format_mask = MAX98396_PCM_MODE_CFG_FORMAT_MASK |
+		      MAX98396_PCM_MODE_CFG_LRCLKEDGE;
+
 	dev_dbg(component->dev, "%s: fmt 0x%08X\n", __func__, fmt);
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -395,7 +398,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		ret = regmap_read(max98396->regmap, MAX98396_R2041_PCM_MODE_CFG, &reg);
 		if (ret < 0)
 			return -EINVAL;
-		if (format != (reg & MAX98396_PCM_BCLKEDGE_BSEL_MASK)) {
+		if (format != (reg & format_mask)) {
 			update = true;
 		} else {
 			ret = regmap_read(max98396->regmap,
@@ -412,8 +415,7 @@ static int max98396_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 
 	regmap_update_bits(max98396->regmap,
 			   MAX98396_R2041_PCM_MODE_CFG,
-			   MAX98396_PCM_BCLKEDGE_BSEL_MASK,
-			   format);
+			   format_mask, format);
 
 	regmap_update_bits(max98396->regmap,
 			   MAX98396_R2042_PCM_CLK_SETUP,

sound/soc/codecs/rt5640.c

@@ -1984,7 +1984,12 @@ static int rt5640_set_bias_level(struct snd_soc_component *component,
 		snd_soc_component_write(component, RT5640_PWR_DIG2, 0x0000);
 		snd_soc_component_write(component, RT5640_PWR_VOL, 0x0000);
 		snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000);
-		snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000);
+		if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
+			snd_soc_component_write(component, RT5640_PWR_ANLG1,
+				0x0018);
+		else
+			snd_soc_component_write(component, RT5640_PWR_ANLG1,
+				0x0000);
 		snd_soc_component_write(component, RT5640_PWR_ANLG2, 0x0000);
 		break;
 
@@ -2393,9 +2398,15 @@ static void rt5640_jack_work(struct work_struct *work)
 static irqreturn_t rt5640_irq(int irq, void *data)
 {
 	struct rt5640_priv *rt5640 = data;
+	int delay = 0;
+
+	if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
+		cancel_delayed_work_sync(&rt5640->jack_work);
+		delay = 100;
+	}
 
 	if (rt5640->jack)
-		queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
+		queue_delayed_work(system_long_wq, &rt5640->jack_work, delay);
 
 	return IRQ_HANDLED;
 }
@@ -2580,6 +2591,12 @@ static void rt5640_enable_hda_jack_detect(
 
 	snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0);
 
+	snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+		RT5640_PWR_VREF2, RT5640_PWR_VREF2);
+	usleep_range(10000, 15000);
+	snd_soc_component_update_bits(component, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV2, RT5640_PWR_FV2);
+
 	rt5640->jack = jack;
 
 	ret = request_irq(rt5640->irq, rt5640_irq,
@@ -2696,16 +2713,13 @@ static int rt5640_probe(struct snd_soc_component *component)
 
 	if (device_property_read_u32(component->dev,
 				     "realtek,jack-detect-source", &val) == 0) {
-		if (val <= RT5640_JD_SRC_GPIO4) {
+		if (val <= RT5640_JD_SRC_GPIO4)
 			rt5640->jd_src = val << RT5640_JD_SFT;
-		} else if (val == RT5640_JD_SRC_HDA_HEADER) {
+		else if (val == RT5640_JD_SRC_HDA_HEADER)
 			rt5640->jd_src = RT5640_JD_SRC_HDA_HEADER;
-			snd_soc_component_update_bits(component, RT5640_DUMMY1,
-				0x0300, 0x0);
-		} else {
+		else
 			dev_warn(component->dev, "Warning: Invalid jack-detect-source value: %d, leaving jack-detect disabled\n",
 				 val);
-		}
 	}
 
 	if (!device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted"))

sound/soc/codecs/sgtl5000.c

@@ -1795,6 +1795,9 @@ static int sgtl5000_i2c_remove(struct i2c_client *client)
 {
 	struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client);
 
+	regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT);
+	regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT);
+
 	clk_disable_unprepare(sgtl5000->mclk);
 	regulator_bulk_disable(sgtl5000->num_supplies, sgtl5000->supplies);
 	regulator_bulk_free(sgtl5000->num_supplies, sgtl5000->supplies);
@@ -1802,6 +1805,11 @@ static int sgtl5000_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+static void sgtl5000_i2c_shutdown(struct i2c_client *client)
+{
+	sgtl5000_i2c_remove(client);
+}
+
 static const struct i2c_device_id sgtl5000_id[] = {
 	{"sgtl5000", 0},
 	{},
@@ -1822,6 +1830,7 @@ static struct i2c_driver sgtl5000_i2c_driver = {
 	},
 	.probe_new = sgtl5000_i2c_probe,
 	.remove = sgtl5000_i2c_remove,
+	.shutdown = sgtl5000_i2c_shutdown,
 	.id_table = sgtl5000_id,
 };
 

sound/soc/codecs/sgtl5000.h

@@ -80,6 +80,7 @@
 /*
  * SGTL5000_CHIP_DIG_POWER
  */
+#define SGTL5000_DIG_POWER_DEFAULT		0x0000
 #define SGTL5000_ADC_EN				0x0040
 #define SGTL5000_DAC_EN				0x0020
 #define SGTL5000_DAP_POWERUP			0x0010

sound/soc/codecs/tas2764.c

@@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764)
 		gpiod_set_value_cansleep(tas2764->reset_gpio, 0);
 		msleep(20);
 		gpiod_set_value_cansleep(tas2764->reset_gpio, 1);
+		usleep_range(1000, 2000);
 	}
 
 	snd_soc_component_write(tas2764->component, TAS2764_SW_RST,
 				TAS2764_RST);
+	usleep_range(1000, 2000);
 }
 
 static int tas2764_set_bias_level(struct snd_soc_component *component,
@@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component)
 	struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
 	int ret;
 
-	if (tas2764->sdz_gpio)
+	if (tas2764->sdz_gpio) {
 		gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+		usleep_range(1000, 2000);
+	}
 
 	ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL,
 					    TAS2764_PWR_CTRL_MASK,
@@ -131,7 +135,8 @@ static const char * const tas2764_ASI1_src[] = {
 };
 
 static SOC_ENUM_SINGLE_DECL(
-	tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, 4, tas2764_ASI1_src);
+	tas2764_ASI1_src_enum, TAS2764_TDM_CFG2, TAS2764_TDM_CFG2_SCFG_SHIFT,
+	tas2764_ASI1_src);
 
 static const struct snd_kcontrol_new tas2764_asi1_mux =
 	SOC_DAPM_ENUM("ASI1 Source", tas2764_ASI1_src_enum);
@@ -329,20 +334,22 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
-	u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0;
-	int iface;
+	u8 tdm_rx_start_slot = 0, asi_cfg_0 = 0, asi_cfg_1 = 0;
 	int ret;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_IF:
+		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+		fallthrough;
 	case SND_SOC_DAIFMT_NB_NF:
 		asi_cfg_1 = TAS2764_TDM_CFG1_RX_RISING;
 		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+		fallthrough;
 	case SND_SOC_DAIFMT_IB_NF:
 		asi_cfg_1 = TAS2764_TDM_CFG1_RX_FALLING;
 		break;
-	default:
-		dev_err(tas2764->dev, "ASI format Inverse is not found\n");
-		return -EINVAL;
 	}
 
 	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
@@ -353,13 +360,13 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
+		asi_cfg_0 ^= TAS2764_TDM_CFG0_FRAME_START;
+		fallthrough;
 	case SND_SOC_DAIFMT_DSP_A:
-		iface = TAS2764_TDM_CFG2_SCFG_I2S;
 		tdm_rx_start_slot = 1;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 	case SND_SOC_DAIFMT_LEFT_J:
-		iface = TAS2764_TDM_CFG2_SCFG_LEFT_J;
 		tdm_rx_start_slot = 0;
 		break;
 	default:
@@ -368,14 +375,15 @@ static int tas2764_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
-	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
-					    TAS2764_TDM_CFG1_MASK,
-					    (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
+	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG0,
+					    TAS2764_TDM_CFG0_FRAME_START,
+					    asi_cfg_0);
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG2,
-					    TAS2764_TDM_CFG2_SCFG_MASK, iface);
+	ret = snd_soc_component_update_bits(component, TAS2764_TDM_CFG1,
+					    TAS2764_TDM_CFG1_MASK,
+					    (tdm_rx_start_slot << TAS2764_TDM_CFG1_51_SHIFT));
 	if (ret < 0)
 		return ret;
 
@@ -501,8 +509,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
 
 	tas2764->component = component;
 
-	if (tas2764->sdz_gpio)
+	if (tas2764->sdz_gpio) {
 		gpiod_set_value_cansleep(tas2764->sdz_gpio, 1);
+		usleep_range(1000, 2000);
+	}
 
 	tas2764_reset(tas2764);
 
@@ -526,12 +536,12 @@ static int tas2764_codec_probe(struct snd_soc_component *component)
 }
 
 static DECLARE_TLV_DB_SCALE(tas2764_digital_tlv, 1100, 50, 0);
-static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10000, 50, 0);
+static DECLARE_TLV_DB_SCALE(tas2764_playback_volume, -10050, 50, 1);
 
 static const struct snd_kcontrol_new tas2764_snd_controls[] = {
 	SOC_SINGLE_TLV("Speaker Volume", TAS2764_DVC, 0,
 		       TAS2764_DVC_MAX, 1, tas2764_playback_volume),
-	SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 0, 0x14, 0,
+	SOC_SINGLE_TLV("Amp Gain Volume", TAS2764_CHNL_0, 1, 0x14, 0,
 		       tas2764_digital_tlv),
 };
 
@@ -556,7 +566,7 @@ static const struct reg_default tas2764_reg_defaults[] = {
 	{ TAS2764_SW_RST, 0x00 },
 	{ TAS2764_PWR_CTRL, 0x1a },
 	{ TAS2764_DVC, 0x00 },
-	{ TAS2764_CHNL_0, 0x00 },
+	{ TAS2764_CHNL_0, 0x28 },
 	{ TAS2764_TDM_CFG0, 0x09 },
 	{ TAS2764_TDM_CFG1, 0x02 },
 	{ TAS2764_TDM_CFG2, 0x0a },

sound/soc/codecs/tas2764.h

@@ -47,6 +47,7 @@
 #define TAS2764_TDM_CFG0_MASK		GENMASK(3, 1)
 #define TAS2764_TDM_CFG0_44_1_48KHZ	BIT(3)
 #define TAS2764_TDM_CFG0_88_2_96KHZ	(BIT(3) | BIT(1))
+#define TAS2764_TDM_CFG0_FRAME_START	BIT(0)
 
 /* TDM Configuration Reg1 */
 #define TAS2764_TDM_CFG1		TAS2764_REG(0X0, 0x09)
@@ -66,10 +67,7 @@
 #define TAS2764_TDM_CFG2_RXS_16BITS	0x0
 #define TAS2764_TDM_CFG2_RXS_24BITS	BIT(0)
 #define TAS2764_TDM_CFG2_RXS_32BITS	BIT(1)
-#define TAS2764_TDM_CFG2_SCFG_MASK	GENMASK(5, 4)
-#define TAS2764_TDM_CFG2_SCFG_I2S	0x0
-#define TAS2764_TDM_CFG2_SCFG_LEFT_J	BIT(4)
-#define TAS2764_TDM_CFG2_SCFG_RIGHT_J	BIT(5)
+#define TAS2764_TDM_CFG2_SCFG_SHIFT	4
 
 /* TDM Configuration Reg3 */
 #define TAS2764_TDM_CFG3		TAS2764_REG(0X0, 0x0c)

sound/soc/codecs/tlv320adcx140.c

@@ -33,7 +33,6 @@ struct adcx140_priv {
 	bool micbias_vg;
 
 	unsigned int dai_fmt;
-	unsigned int tdm_delay;
 	unsigned int slot_width;
 };
 
@@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_component *component = codec_dai->component;
 	struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
-	unsigned int lsb;
 
-	/* TDM based on DSP mode requires slots to be adjacent */
-	lsb = __ffs(tx_mask);
-	if ((lsb + 1) != __fls(tx_mask)) {
-		dev_err(component->dev, "Invalid mask, slots must be adjacent\n");
+	/*
+	 * The chip itself supports arbitrary masks, but the driver currently
+	 * only supports adjacent slots beginning at the first slot.
+	 */
+	if (tx_mask != GENMASK(__fls(tx_mask), 0)) {
+		dev_err(component->dev, "Only lower adjacent slots are supported\n");
 		return -EINVAL;
 	}
 
@@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	adcx140->tdm_delay = lsb;
 	adcx140->slot_width = slot_width;
 
 	return 0;

sound/soc/codecs/wcd9335.c

@@ -342,7 +342,7 @@ struct wcd9335_codec {
 	struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY];
 
 	unsigned int rx_port_value[WCD9335_RX_MAX];
-	unsigned int tx_port_value;
+	unsigned int tx_port_value[WCD9335_TX_MAX];
 	int hph_l_gain;
 	int hph_r_gain;
 	u32 rx_bias_count;
@@ -1334,8 +1334,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc,
 
 	struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc);
 	struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev);
+	struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc);
+	struct soc_mixer_control *mixer =
+			(struct soc_mixer_control *)kc->private_value;
+	int dai_id = widget->shift;
+	int port_id = mixer->shift;
 
-	ucontrol->value.integer.value[0] = wcd->tx_port_value;
+	ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id;
 
 	return 0;
 }
@@ -1358,12 +1363,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc,
 	case AIF2_CAP:
 	case AIF3_CAP:
 		/* only add to the list if value not set */
-		if (enable && !(wcd->tx_port_value & BIT(port_id))) {
-			wcd->tx_port_value |= BIT(port_id);
+		if (enable && wcd->tx_port_value[port_id] != dai_id) {
+			wcd->tx_port_value[port_id] = dai_id;
 			list_add_tail(&wcd->tx_chs[port_id].list,
 					&wcd->dai[dai_id].slim_ch_list);
-		} else if (!enable && (wcd->tx_port_value & BIT(port_id))) {
-			wcd->tx_port_value &= ~BIT(port_id);
+		} else if (!enable && wcd->tx_port_value[port_id] == dai_id) {
+			wcd->tx_port_value[port_id] = -1;
 			list_del_init(&wcd->tx_chs[port_id].list);
 		}
 		break;

sound/soc/codecs/wm5102.c

@@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+	uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+	int ret = 0;
 
 	mutex_lock(&arizona->dac_comp_lock);
-	arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data);
+	if (arizona->dac_comp_coeff != dac_comp_coeff) {
+		arizona->dac_comp_coeff = dac_comp_coeff;
+		ret = 1;
+	}
 	mutex_unlock(&arizona->dac_comp_lock);
 
-	return 0;
+	return ret;
 }
 
 static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol,
@@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
 	struct arizona *arizona = dev_get_drvdata(component->dev->parent);
+	struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value;
+	int ret = 0;
+
+	if (ucontrol->value.integer.value[0] > mc->max)
+		return -EINVAL;
 
 	mutex_lock(&arizona->dac_comp_lock);
-	arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+	if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) {
+		arizona->dac_comp_enabled = ucontrol->value.integer.value[0];
+		ret = 1;
+	}
 	mutex_unlock(&arizona->dac_comp_lock);
 
-	return 0;
+	return ret;
 }
 
 static const char * const wm5102_osr_text[] = {

sound/soc/codecs/wm8998.c

@@ -108,6 +108,7 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int mode_reg, mode_index;
 	unsigned int mux, inmode, src_val, mode_val;
+	int change, ret;
 
 	mux = ucontrol->value.enumerated.item[0];
 	if (mux > 1)
@@ -137,14 +138,20 @@ static int wm8998_inmux_put(struct snd_kcontrol *kcontrol,
 	snd_soc_component_update_bits(component, mode_reg,
 				      ARIZONA_IN1_MODE_MASK, mode_val);
 
-	snd_soc_component_update_bits(component, e->reg,
-				      ARIZONA_IN1L_SRC_MASK |
-				      ARIZONA_IN1L_SRC_SE_MASK,
-				      src_val);
+	change = snd_soc_component_update_bits(component, e->reg,
+					       ARIZONA_IN1L_SRC_MASK |
+					       ARIZONA_IN1L_SRC_SE_MASK,
+					       src_val);
 
-	return snd_soc_dapm_mux_update_power(dapm, kcontrol,
-					     ucontrol->value.enumerated.item[0],
-					     e, NULL);
+	ret = snd_soc_dapm_mux_update_power(dapm, kcontrol,
+					    ucontrol->value.enumerated.item[0],
+					    e, NULL);
+	if (ret < 0) {
+		dev_err(arizona->dev, "Failed to update demux power state: %d\n", ret);
+		return ret;
+	}
+
+	return change;
 }
 
 static const char * const wm8998_inmux_texts[] = {

sound/soc/generic/audio-graph-card2.c

@@ -90,12 +90,12 @@ links indicates connection part of CPU side (= A).
 			ports@0 {
 (X) (A)			mcpu:	port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; };
 (y)				port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; };
-(y)				port@1 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
+(y)				port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
 			};
 			ports@1 {
 (X)				port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; };
-(y)				port@0 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
-(y)				port@1 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
+(y)				port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
+(y)				port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
 			};
 		};
 	};

sound/soc/intel/boards/sof_rt5682.c

@@ -69,11 +69,10 @@ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
 
 static int is_legacy_cpu;
 
-static struct snd_soc_jack sof_hdmi[3];
-
 struct sof_hdmi_pcm {
 	struct list_head head;
 	struct snd_soc_dai *codec_dai;
+	struct snd_soc_jack hdmi_jack;
 	int device;
 };
 
@@ -434,7 +433,6 @@ static int sof_card_late_probe(struct snd_soc_card *card)
 	char jack_name[NAME_SIZE];
 	struct sof_hdmi_pcm *pcm;
 	int err;
-	int i = 0;
 
 	/* HDMI is not supported by SOF on Baytrail/CherryTrail */
 	if (is_legacy_cpu || !ctx->idisp_codec)
@@ -455,17 +453,15 @@ static int sof_card_late_probe(struct snd_soc_card *card)
 		snprintf(jack_name, sizeof(jack_name),
 			 "HDMI/DP, pcm=%d Jack", pcm->device);
 		err = snd_soc_card_jack_new(card, jack_name,
-					    SND_JACK_AVOUT, &sof_hdmi[i]);
+					    SND_JACK_AVOUT, &pcm->hdmi_jack);
 
 		if (err)
 			return err;
 
 		err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device,
-					  &sof_hdmi[i]);
+					  &pcm->hdmi_jack);
 		if (err < 0)
 			return err;
-
-		i++;
 	}
 
 	if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) {

sound/soc/intel/skylake/skl-nhlt.c

@@ -99,7 +99,6 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 	struct nhlt_fmt_cfg *fmt_cfg;
 	struct wav_fmt_ext *wav_fmt;
 	unsigned long rate;
-	bool present = false;
 	int rate_index = 0;
 	u16 channels, bps;
 	u8 clk_src;
@@ -112,9 +111,12 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 	if (fmt->fmt_count == 0)
 		return;
 
+	fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
 	for (i = 0; i < fmt->fmt_count; i++) {
-		fmt_cfg = &fmt->fmt_config[i];
-		wav_fmt = &fmt_cfg->fmt_ext;
+		struct nhlt_fmt_cfg *saved_fmt_cfg = fmt_cfg;
+		bool present = false;
+
+		wav_fmt = &saved_fmt_cfg->fmt_ext;
 
 		channels = wav_fmt->fmt.channels;
 		bps = wav_fmt->fmt.bits_per_sample;
@@ -132,12 +134,18 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 		 * derive the rate.
 		 */
 		for (j = i; j < fmt->fmt_count; j++) {
-			fmt_cfg = &fmt->fmt_config[j];
-			wav_fmt = &fmt_cfg->fmt_ext;
+			struct nhlt_fmt_cfg *tmp_fmt_cfg = fmt_cfg;
+
+			wav_fmt = &tmp_fmt_cfg->fmt_ext;
 			if ((fs == wav_fmt->fmt.samples_per_sec) &&
-			   (bps == wav_fmt->fmt.bits_per_sample))
+			   (bps == wav_fmt->fmt.bits_per_sample)) {
 				channels = max_t(u16, channels,
 						wav_fmt->fmt.channels);
+				saved_fmt_cfg = tmp_fmt_cfg;
+			}
+			/* Move to the next nhlt_fmt_cfg */
+			tmp_fmt_cfg = (struct nhlt_fmt_cfg *)(tmp_fmt_cfg->config.caps +
+							      tmp_fmt_cfg->config.size);
 		}
 
 		rate = channels * bps * fs;
@@ -153,8 +161,11 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 
 		/* Fill rate and parent for sclk/sclkfs */
 		if (!present) {
+			struct nhlt_fmt_cfg *first_fmt_cfg;
+
+			first_fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
 			i2s_config_ext = (struct skl_i2s_config_blob_ext *)
-						fmt->fmt_config[0].config.caps;
+						first_fmt_cfg->config.caps;
 
 			/* MCLK Divider Source Select */
 			if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -168,6 +179,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 
 			parent = skl_get_parent_clk(clk_src);
 
+			/* Move to the next nhlt_fmt_cfg */
+			fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps +
+							  fmt_cfg->config.size);
 			/*
 			 * Do not copy the config data if there is no parent
 			 * clock available for this clock source select
@@ -176,9 +190,9 @@ static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks,
 				continue;
 
 			sclk[id].rate_cfg[rate_index].rate = rate;
-			sclk[id].rate_cfg[rate_index].config = fmt_cfg;
+			sclk[id].rate_cfg[rate_index].config = saved_fmt_cfg;
 			sclkfs[id].rate_cfg[rate_index].rate = rate;
-			sclkfs[id].rate_cfg[rate_index].config = fmt_cfg;
+			sclkfs[id].rate_cfg[rate_index].config = saved_fmt_cfg;
 			sclk[id].parent_name = parent->name;
 			sclkfs[id].parent_name = parent->name;
 
@@ -192,13 +206,13 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
 {
 	struct skl_i2s_config_blob_ext *i2s_config_ext;
 	struct skl_i2s_config_blob_legacy *i2s_config;
-	struct nhlt_specific_cfg *fmt_cfg;
+	struct nhlt_fmt_cfg *fmt_cfg;
 	struct skl_clk_parent_src *parent;
 	u32 clkdiv, div_ratio;
 	u8 clk_src;
 
-	fmt_cfg = &fmt->fmt_config[0].config;
-	i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->caps;
+	fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config;
+	i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->config.caps;
 
 	/* MCLK Divider Source Select and divider */
 	if (is_legacy_blob(i2s_config_ext->hdr.sig)) {
@@ -227,7 +241,7 @@ static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk,
 		return;
 
 	mclk[id].rate_cfg[0].rate = parent->rate/div_ratio;
-	mclk[id].rate_cfg[0].config = &fmt->fmt_config[0];
+	mclk[id].rate_cfg[0].config = fmt_cfg;
 	mclk[id].parent_name = parent->name;
 }
 

sound/soc/qcom/qdsp6/q6apm.c

@@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui
 	id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL);
 	if (id < 0) {
 		dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id);
+		kfree(graph->graph);
 		kfree(graph);
 		mutex_unlock(&apm->lock);
 		return ERR_PTR(id);

sound/soc/ti/omap-mcbsp-priv.h

@@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg,
 
 /* Sidetone specific API */
 int omap_mcbsp_st_init(struct platform_device *pdev);
-void omap_mcbsp_st_cleanup(struct platform_device *pdev);
-
 int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp);
 int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp);
 

sound/soc/ti/omap-mcbsp-st.c

@@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
 	if (!st_data)
 		return -ENOMEM;
 
-	st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick");
+	st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick");
 	if (IS_ERR(st_data->mcbsp_iclk)) {
 		dev_warn(mcbsp->dev,
 			 "Failed to get ick, sidetone might be broken\n");
@@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
 	if (!st_data->io_base_st)
 		return -ENOMEM;
 
-	ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group);
+	ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group);
 	if (ret)
 		return ret;
 
@@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev)
 	return 0;
 }
 
-void omap_mcbsp_st_cleanup(struct platform_device *pdev)
-{
-	struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
-
-	if (mcbsp->st_data) {
-		sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group);
-		clk_put(mcbsp->st_data->mcbsp_iclk);
-	}
-}
-
 static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
 				    struct snd_ctl_elem_info *uinfo)
 {

sound/soc/ti/omap-mcbsp.c

@@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 		mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10;
 		mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10;
 
-		ret = sysfs_create_group(&mcbsp->dev->kobj,
-					 &additional_attr_group);
+		ret = devm_device_add_group(mcbsp->dev, &additional_attr_group);
 		if (ret) {
 			dev_err(mcbsp->dev,
 				"Unable to create additional controls\n");
@@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev)
 		}
 	}
 
-	ret = omap_mcbsp_st_init(pdev);
-	if (ret)
-		goto err_st;
-
-	return 0;
-
-err_st:
-	if (mcbsp->pdata->buffer_size)
-		sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-	return ret;
+	return omap_mcbsp_st_init(pdev);
 }
 
 /*
@@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
 	if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req))
 		cpu_latency_qos_remove_request(&mcbsp->pm_qos_req);
 
-	if (mcbsp->pdata->buffer_size)
-		sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-
-	omap_mcbsp_st_cleanup(pdev);
-
 	return 0;
 }