Merge tag 'sound-fix-6.14-rc1' of git://git.kernel.org/pub/scm/linux/…

…kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Here is a collection of fixes that have been gathered since the
  previous pull request.

  All about device-specific fixes and quirks, and most of them are
  pretty small and trivial"

* tag 'sound-fix-6.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (25 commits)
  ALSA: hda/realtek: Workaround for resume on Dell Venue 11 Pro 7130
  ALSA: hda: Fix headset detection failure due to unstable sort
  ALSA: pcm: use new array-copying-wrapper
  ASoC: codec: es8316: "DAC Soft Ramp Rate" is just a 2 bit control
  ASoC: amd: acp: Fix possible deadlock
  firmware: cs_dsp: FW_CS_DSP_KUNIT_TEST should not select REGMAP
  ALSA: usb-audio: Add delay quirk for iBasso DC07 Pro
  ALSA: hda/realtek: Fix quirk matching for Legion Pro 7
  ASoC: renesas: SND_SIU_MIGOR should depend on DMADEVICES
  ASoC: Intel: bytcr_rt5640: Add DMI quirk for Vexia Edu Atla 10 tablet 5V
  ASoC: da7213: Initialize the mutex
  ASoC: use to_platform_device() instead of container_of()
  ASoC: acp: Support microphone from Lenovo Go S
  ASoC: SOF: imx8m: Add entry for new 8M Plus revision
  ASoC: SOF: imx8: Add entries for new 8QM and 8QXP revisions
  ASoC: SOF: imx: Add mach entry to select cs42888 topology
  dt-bindings: arm: imx: Add board revisions for i.MX8MP, i.MX8QM and i.MX8QXP
  ASoC: fsl_asrc_m2m: select CONFIG_DMA_SHARED_BUFFER
  ASoC: audio-graph-card2: use correct endpoint when getting link parameters
  ASoC: SOF: imx8m: add SAI2,5,6,7
  ...

Documentation/devicetree/bindings/arm/fsl.yaml

@@ -1091,6 +1091,7 @@ properties:
               - dmo,imx8mp-data-modul-edm-sbc # i.MX8MP eDM SBC
               - emcraft,imx8mp-navqp      # i.MX8MP Emcraft Systems NavQ+ Kit
               - fsl,imx8mp-evk            # i.MX8MP EVK Board
+              - fsl,imx8mp-evk-revb4      # i.MX8MP EVK Rev B4 Board
               - gateworks,imx8mp-gw71xx-2x # i.MX8MP Gateworks Board
               - gateworks,imx8mp-gw72xx-2x # i.MX8MP Gateworks Board
               - gateworks,imx8mp-gw73xx-2x # i.MX8MP Gateworks Board
@@ -1271,6 +1272,7 @@ properties:
         items:
           - enum:
               - fsl,imx8qm-mek           # i.MX8QM MEK Board
+              - fsl,imx8qm-mek-revd      # i.MX8QM MEK Rev D Board
               - toradex,apalis-imx8      # Apalis iMX8 Modules
               - toradex,apalis-imx8-v1.1 # Apalis iMX8 V1.1 Modules
           - const: fsl,imx8qm
@@ -1299,6 +1301,7 @@ properties:
           - enum:
               - einfochips,imx8qxp-ai_ml  # i.MX8QXP AI_ML Board
               - fsl,imx8qxp-mek           # i.MX8QXP MEK Board
+              - fsl,imx8qxp-mek-wcpu      # i.MX8QXP MEK WCPU Board
           - const: fsl,imx8qxp
 
       - description: i.MX8DXL based Boards

Documentation/devicetree/bindings/sound/ti,pcm1681.yaml

@@ -4,7 +4,7 @@
 $id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml#
 $schema: http://devicetree.org/meta-schemas/core.yaml#
 
-title: Texas Instruments PCM1681 8-channel PWM Processor
+title: Texas Instruments PCM1681 8-channel Digital-to-Analog Converter
 
 maintainers:
   - Shenghao Ding <[email protected]>

drivers/firmware/cirrus/Kconfig

@@ -6,15 +6,13 @@ config FW_CS_DSP
 
 config FW_CS_DSP_KUNIT_TEST_UTILS
 	tristate
-	depends on KUNIT
-	select REGMAP
+	depends on KUNIT && REGMAP
 	select FW_CS_DSP
 
 config FW_CS_DSP_KUNIT_TEST
 	tristate "KUnit tests for Cirrus Logic cs_dsp" if !KUNIT_ALL_TESTS
-	depends on KUNIT
+	depends on KUNIT && REGMAP
 	default KUNIT_ALL_TESTS
-	select REGMAP
 	select FW_CS_DSP
 	select FW_CS_DSP_KUNIT_TEST_UTILS
 	help

sound/core/pcm_native.c

@@ -3245,7 +3245,7 @@ static int snd_pcm_xfern_frames_ioctl(struct snd_pcm_substream *substream,
 	if (copy_from_user(&xfern, _xfern, sizeof(xfern)))
 		return -EFAULT;
 
-	bufs = memdup_user(xfern.bufs, sizeof(void *) * runtime->channels);
+	bufs = memdup_array_user(xfern.bufs, runtime->channels, sizeof(void *));
 	if (IS_ERR(bufs))
 		return PTR_ERR(bufs);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

sound/pci/hda/hda_auto_parser.c

@@ -80,7 +80,11 @@ static int compare_input_type(const void *ap, const void *bp)
 
 	/* In case one has boost and the other one has not,
 	   pick the one with boost first. */
-	return (int)(b->has_boost_on_pin - a->has_boost_on_pin);
+	if (a->has_boost_on_pin != b->has_boost_on_pin)
+		return (int)(b->has_boost_on_pin - a->has_boost_on_pin);
+
+	/* Keep the original order */
+	return a->order - b->order;
 }
 
 /* Reorder the surround channels
@@ -400,6 +404,8 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
 
 	/* sort inputs in the order of AUTO_PIN_* type */
+	for (i = 0; i < cfg->num_inputs; i++)
+		cfg->inputs[i].order = i;
 	sort(cfg->inputs, cfg->num_inputs, sizeof(cfg->inputs[0]),
 	     compare_input_type, NULL);
 

sound/pci/hda/hda_auto_parser.h

@@ -37,6 +37,7 @@ struct auto_pin_cfg_item {
 	unsigned int is_headset_mic:1;
 	unsigned int is_headphone_mic:1; /* Mic-only in headphone jack */
 	unsigned int has_boost_on_pin:1;
+	int order;
 };
 
 struct auto_pin_cfg;

sound/pci/hda/patch_realtek.c

@@ -7497,6 +7497,16 @@ static void alc287_fixup_lenovo_thinkpad_with_alc1318(struct hda_codec *codec,
 	spec->gen.pcm_playback_hook = alc287_alc1318_playback_pcm_hook;
 }
 
+/*
+ * Clear COEF 0x0d (PCBEEP passthrough) bit 0x40 where BIOS sets it wrongly
+ * at PM resume
+ */
+static void alc283_fixup_dell_hp_resume(struct hda_codec *codec,
+					const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_INIT)
+		alc_write_coef_idx(codec, 0xd, 0x2800);
+}
 
 enum {
 	ALC269_FIXUP_GPIO2,
@@ -7799,6 +7809,7 @@ enum {
 	ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE,
 	ALC233_FIXUP_MEDION_MTL_SPK,
 	ALC294_FIXUP_BASS_SPEAKER_15,
+	ALC283_FIXUP_DELL_HP_RESUME,
 };
 
 /* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -10143,6 +10154,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc294_fixup_bass_speaker_15,
 	},
+	[ALC283_FIXUP_DELL_HP_RESUME] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc283_fixup_dell_hp_resume,
+	},
 };
 
 static const struct hda_quirk alc269_fixup_tbl[] = {
@@ -10203,6 +10218,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0604, "Dell Venue 11 Pro 7130", ALC283_FIXUP_DELL_HP_RESUME),
 	SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK),
 	SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK),
 	SND_PCI_QUIRK(0x1028, 0x062c, "Dell Latitude E5550", ALC292_FIXUP_DELL_E7X),
@@ -10918,7 +10934,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN),
 	HDA_CODEC_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x17aa, 0x386e, "Yoga Pro 7 14ARP8", ALC285_FIXUP_SPEAKER2_TO_DAC1),
-	HDA_CODEC_QUIRK(0x17aa, 0x386f, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C),
+	HDA_CODEC_QUIRK(0x17aa, 0x38a8, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), /* this must match before PCI SSID 17aa:386f below */
 	SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2),

sound/soc/amd/acp/acp-i2s.c

@@ -181,6 +181,7 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas
 			break;
 		default:
 			dev_err(dev, "Unknown chip revision %d\n", chip->acp_rev);
+			spin_unlock_irq(&adata->acp_lock);
 			return -EINVAL;
 		}
 	}

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

@@ -304,6 +304,34 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "83AS"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "83L3"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "83N6"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "83Q2"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "83Q3"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {

sound/soc/codecs/da7213.c

@@ -2203,6 +2203,8 @@ static int da7213_i2c_probe(struct i2c_client *i2c)
 		return ret;
 	}
 
+	mutex_init(&da7213->ctrl_lock);
+
 	pm_runtime_set_autosuspend_delay(&i2c->dev, 100);
 	pm_runtime_use_autosuspend(&i2c->dev);
 	pm_runtime_set_active(&i2c->dev);

sound/soc/codecs/es8316.c

@@ -101,7 +101,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
 	SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
 			 ES8316_DAC_VOLR, 0, 0xc0, 1, dac_vol_tlv),
 	SOC_SINGLE("DAC Soft Ramp Switch", ES8316_DAC_SET1, 4, 1, 1),
-	SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0),
+	SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 3, 0),
 	SOC_SINGLE("DAC Notch Filter Switch", ES8316_DAC_SET2, 6, 1, 0),
 	SOC_SINGLE("DAC Double Fs Switch", ES8316_DAC_SET2, 7, 1, 0),
 	SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0),

sound/soc/codecs/es8326.c

@@ -911,7 +911,7 @@ static void es8326_jack_detect_handler(struct work_struct *work)
 			regmap_write(es8326->regmap, ES8326_INT_SOURCE,
 					(ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
 			regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x1f);
-			regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x08);
+			regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x0d);
 			queue_delayed_work(system_wq, &es8326->jack_detect_work,
 					msecs_to_jiffies(400));
 			es8326->hp = 1;
@@ -1023,7 +1023,7 @@ static void es8326_init(struct snd_soc_component *component)
 	struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component);
 
 	regmap_write(es8326->regmap, ES8326_RESET, 0x1f);
-	regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E);
+	regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x3E);
 	regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0);
 	usleep_range(10000, 15000);
 	regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9);

sound/soc/codecs/rt5514.c

@@ -1091,8 +1091,7 @@ static int rt5514_set_bias_level(struct snd_soc_component *component,
 static int rt5514_probe(struct snd_soc_component *component)
 {
 	struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component);
-	struct platform_device *pdev = container_of(component->dev,
-						   struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(component->dev);
 
 	rt5514->mclk = devm_clk_get_optional(component->dev, "mclk");
 	if (IS_ERR(rt5514->mclk))

sound/soc/fsl/Kconfig

@@ -6,6 +6,7 @@ comment "Common SoC Audio options for Freescale CPUs:"
 config SND_SOC_FSL_ASRC
 	tristate "Asynchronous Sample Rate Converter (ASRC) module support"
 	depends on HAS_DMA
+	select DMA_SHARED_BUFFER
 	select REGMAP_MMIO
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select SND_COMPRESS_ACCEL

sound/soc/fsl/fsl_asrc_m2m.c

@@ -183,7 +183,7 @@ static int asrc_dmaconfig(struct fsl_asrc_pair *pair,
 }
 
 /* main function of converter */
-static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_task_runtime *task)
+static int asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_task_runtime *task)
 {
 	struct fsl_asrc *asrc = pair->asrc;
 	struct device *dev = &asrc->pdev->dev;
@@ -193,7 +193,7 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 	unsigned int out_dma_len;
 	unsigned int width;
 	u32 fifo_addr;
-	int ret;
+	int ret = 0;
 
 	/* set ratio mod */
 	if (asrc->m2m_set_ratio_mod) {
@@ -215,6 +215,7 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 	    in_buf_len > ASRC_M2M_BUFFER_SIZE ||
 	    in_buf_len % (width * pair->channels / 8)) {
 		dev_err(dev, "out buffer size is error: [%d]\n", in_buf_len);
+		ret = -EINVAL;
 		goto end;
 	}
 
@@ -245,6 +246,7 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 		}
 	} else if (out_dma_len > ASRC_M2M_BUFFER_SIZE) {
 		dev_err(dev, "cap buffer size error\n");
+		ret = -EINVAL;
 		goto end;
 	}
 
@@ -263,12 +265,14 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 
 	if (!wait_for_completion_interruptible_timeout(&pair->complete[IN], 10 * HZ)) {
 		dev_err(dev, "out DMA task timeout\n");
+		ret = -ETIMEDOUT;
 		goto end;
 	}
 
 	if (out_dma_len > 0) {
 		if (!wait_for_completion_interruptible_timeout(&pair->complete[OUT], 10 * HZ)) {
 			dev_err(dev, "cap DMA task timeout\n");
+			ret = -ETIMEDOUT;
 			goto end;
 		}
 	}
@@ -278,7 +282,7 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 	/* update payload length for capture */
 	task->output_size = out_dma_len;
 end:
-	return;
+	return ret;
 }
 
 static int fsl_asrc_m2m_comp_open(struct snd_compr_stream *stream)
@@ -525,9 +529,7 @@ static int fsl_asrc_m2m_comp_task_start(struct snd_compr_stream *stream,
 	struct snd_compr_runtime *runtime = stream->runtime;
 	struct fsl_asrc_pair *pair = runtime->private_data;
 
-	asrc_m2m_device_run(pair, task);
-
-	return 0;
+	return asrc_m2m_device_run(pair, task);
 }
 
 static int fsl_asrc_m2m_comp_task_stop(struct snd_compr_stream *stream,
@@ -633,7 +635,7 @@ int fsl_asrc_m2m_suspend(struct fsl_asrc *asrc)
 
 	for (i = 0; i < PAIR_CTX_NUM; i++) {
 		pair = asrc->pair[i];
-		if (!pair)
+		if (!pair || !pair->dma_buffer[IN].area || !pair->dma_buffer[OUT].area)
 			continue;
 		if (!completion_done(&pair->complete[IN])) {
 			if (pair->dma_chan[IN])