Merge branch 'for-linus' into for-next

Pull 5.0 branch for further development of USB-audio quirks
Signed-off-by: Takashi Iwai <tiwai@suse.de>

include/sound/soc.h

@@ -985,6 +985,12 @@ struct snd_soc_dai_link {
 	/* Do not create a PCM for this DAI link (Backend link) */
 	unsigned int ignore:1;
 
+	/*
+	 * This driver uses legacy platform naming. Set by the core, machine
+	 * drivers should not modify this value.
+	 */
+	unsigned int legacy_platform:1;
+
 	struct list_head list; /* DAI link list of the soc card */
 	struct snd_soc_dobj dobj; /* For topology */
 };

sound/core/compress_offload.c

@@ -541,7 +541,8 @@ static int snd_compress_check_input(struct snd_compr_params *params)
 {
 	/* first let's check the buffer parameter's */
 	if (params->buffer.fragment_size == 0 ||
-	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size)
+	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size ||
+	    params->buffer.fragments == 0)
 		return -EINVAL;
 
 	/* now codec parameters */

sound/core/pcm_lib.c

@@ -2112,6 +2112,13 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+/* allow waiting for a capture stream that hasn't been started */
+#if IS_ENABLED(CONFIG_SND_PCM_OSS)
+#define wait_capture_start(substream)	((substream)->oss.oss)
+#else
+#define wait_capture_start(substream)	false
+#endif
+
 /* the common loop for read/write data */
 snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
 				     void *data, bool interleaved,
@@ -2182,7 +2189,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
 			err = snd_pcm_start(substream);
 			if (err < 0)
 				goto _end_unlock;
-		} else {
+		} else if (!wait_capture_start(substream)) {
 			/* nothing to do */
 			err = 0;
 			goto _end_unlock;

sound/pci/cs46xx/dsp_spos.c

@@ -903,6 +903,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip)
 	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
 	int i;
 
+	if (!ins)
+		return 0;
+
 	snd_info_free_entry(ins->proc_sym_info_entry);
 	ins->proc_sym_info_entry = NULL;
 

sound/pci/hda/patch_conexant.c

@@ -931,6 +931,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO),
 	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, 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, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE),

sound/pci/hda/patch_realtek.c

@@ -4102,6 +4102,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
 	case 0x10ec0295:
 	case 0x10ec0289:
 	case 0x10ec0299:
+		alc_process_coef_fw(codec, alc225_pre_hsmode);
 		alc_process_coef_fw(codec, coef0225);
 		break;
 	case 0x10ec0867:
@@ -5440,6 +5441,13 @@ static void alc_fixup_headset_jack(struct hda_codec *codec,
 	}
 }
 
+static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -5549,6 +5557,7 @@ enum {
 	ALC293_FIXUP_LENOVO_SPK_NOISE,
 	ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
 	ALC255_FIXUP_DELL_SPK_NOISE,
+	ALC225_FIXUP_DISABLE_MIC_VREF,
 	ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC295_FIXUP_DISABLE_DAC3,
 	ALC280_FIXUP_HP_HEADSET_MIC,
@@ -6268,6 +6277,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
 	},
+	[ALC225_FIXUP_DISABLE_MIC_VREF] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_mic_vref,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+	},
 	[ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -6277,7 +6292,7 @@ static const struct hda_fixup alc269_fixups[] = {
 			{}
 		},
 		.chained = true,
-		.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+		.chain_id = ALC225_FIXUP_DISABLE_MIC_VREF
 	},
 	[ALC280_FIXUP_HP_HEADSET_MIC] = {
 		.type = HDA_FIXUP_FUNC,
@@ -6911,7 +6926,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"},
 	{.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"},
 	{.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"},
-	{.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"},
+	{.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"},
 	{.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"},
 	{.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"},
 	{.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"},

sound/soc/amd/raven/acp3x-pcm-dma.c

@@ -611,14 +611,16 @@ static int acp3x_audio_probe(struct platform_device *pdev)
 	}
 	irqflags = *((unsigned int *)(pdev->dev.platform_data));
 
-	adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data),
-			     GFP_KERNEL);
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!res) {
 		dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n");
 			return -ENODEV;
 	}
 
+	adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL);
+	if (!adata)
+		return -ENOMEM;
+
 	adata->acp3x_base = devm_ioremap(&pdev->dev, res->start,
 					 resource_size(res));
 

sound/soc/codecs/hdac_hdmi.c

@@ -1890,51 +1890,31 @@ static void hdmi_codec_remove(struct snd_soc_component *component)
 	pm_runtime_disable(&hdev->dev);
 }
 
-#ifdef CONFIG_PM
-static int hdmi_codec_prepare(struct device *dev)
-{
-	struct hdac_device *hdev = dev_to_hdac_dev(dev);
-
-	pm_runtime_get_sync(&hdev->dev);
-
-	/*
-	 * Power down afg.
-	 * codec_read is preferred over codec_write to set the power state.
-	 * This way verb is send to set the power state and response
-	 * is received. So setting power state is ensured without using loop
-	 * to read the state.
-	 */
-	snd_hdac_codec_read(hdev, hdev->afg, 0,	AC_VERB_SET_POWER_STATE,
-							AC_PWRST_D3);
-
-	return 0;
-}
-
-static void hdmi_codec_complete(struct device *dev)
+#ifdef CONFIG_PM_SLEEP
+static int hdmi_codec_resume(struct device *dev)
 {
 	struct hdac_device *hdev = dev_to_hdac_dev(dev);
 	struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev);
+	int ret;
 
-	/* Power up afg */
-	snd_hdac_codec_read(hdev, hdev->afg, 0,	AC_VERB_SET_POWER_STATE,
-							AC_PWRST_D0);
-
-	hdac_hdmi_skl_enable_all_pins(hdev);
-	hdac_hdmi_skl_enable_dp12(hdev);
-
+	ret = pm_runtime_force_resume(dev);
+	if (ret < 0)
+		return ret;
 	/*
 	 * As the ELD notify callback request is not entertained while the
 	 * device is in suspend state. Need to manually check detection of
 	 * all pins here. pin capablity change is not support, so use the
 	 * already set pin caps.
+	 *
+	 * NOTE: this is safe to call even if the codec doesn't actually resume.
+	 * The pin check involves only with DRM audio component hooks, so it
+	 * works even if the HD-audio side is still dreaming peacefully.
 	 */
 	hdac_hdmi_present_sense_all_pins(hdev, hdmi, false);
-
-	pm_runtime_put_sync(&hdev->dev);
+	return 0;
 }
 #else
-#define hdmi_codec_prepare NULL
-#define hdmi_codec_complete NULL
+#define hdmi_codec_resume NULL
 #endif
 
 static const struct snd_soc_component_driver hdmi_hda_codec = {
@@ -2135,75 +2115,6 @@ static int hdac_hdmi_dev_remove(struct hdac_device *hdev)
 }
 
 #ifdef CONFIG_PM
-/*
- * Power management sequences
- * ==========================
- *
- * The following explains the PM handling of HDAC HDMI with its parent
- * device SKL and display power usage
- *
- * Probe
- * -----
- * In SKL probe,
- * 1. skl_probe_work() powers up the display (refcount++ -> 1)
- * 2. enumerates the codecs on the link
- * 3. powers down the display  (refcount-- -> 0)
- *
- * In HDAC HDMI probe,
- * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1)
- * 2. probe the codec
- * 3. put the HDAC HDMI device to runtime suspend
- * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
- *
- * Once children are runtime suspended, SKL device also goes to runtime
- * suspend
- *
- * HDMI Playback
- * -------------
- * Open HDMI device,
- * 1. skl_runtime_resume() invoked
- * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
- *
- * Close HDMI device,
- * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
- * 2. skl_runtime_suspend() invoked
- *
- * S0/S3 Cycle with playback in progress
- * -------------------------------------
- * When the device is opened for playback, the device is runtime active
- * already and the display refcount is 1 as explained above.
- *
- * Entering to S3,
- * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just
- *    increments the PM runtime usage count of the codec since the device
- *    is in use already
- * 2. skl_suspend() powers down the display (refcount-- -> 0)
- *
- * Wakeup from S3,
- * 1. skl_resume() powers up the display (refcount++ -> 1)
- * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just
- *    decrements the PM runtime usage count of the codec since the device
- *    is in use already
- *
- * Once playback is stopped, the display refcount is set to 0 as explained
- * above in the HDMI playback sequence. The PM handlings are designed in
- * such way that to balance the refcount of display power when the codec
- * device put to S3 while playback is going on.
- *
- * S0/S3 Cycle without playback in progress
- * ----------------------------------------
- * Entering to S3,
- * 1. hdmi_codec_prepare() invoke the runtime resume of codec
- * 2. skl_runtime_resume() invoked
- * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1)
- * 4. skl_suspend() powers down the display (refcount-- -> 0)
- *
- * Wakeup from S3,
- * 1. skl_resume() powers up the display (refcount++ -> 1)
- * 2. hdmi_codec_complete() invokes the runtime suspend of codec
- * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0)
- * 4. skl_runtime_suspend() invoked
- */
 static int hdac_hdmi_runtime_suspend(struct device *dev)
 {
 	struct hdac_device *hdev = dev_to_hdac_dev(dev);
@@ -2277,8 +2188,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev)
 
 static const struct dev_pm_ops hdac_hdmi_pm = {
 	SET_RUNTIME_PM_OPS(hdac_hdmi_runtime_suspend, hdac_hdmi_runtime_resume, NULL)
-	.prepare = hdmi_codec_prepare,
-	.complete = hdmi_codec_complete,
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, hdmi_codec_resume)
 };
 
 static const struct hda_device_id hdmi_list[] = {

sound/soc/codecs/pcm512x.c

@@ -1400,24 +1400,20 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute)
 		if (ret != 0) {
 			dev_err(component->dev,
 				"Failed to set digital mute: %d\n", ret);
-			mutex_unlock(&pcm512x->mutex);
-			return ret;
+			goto unlock;
 		}
 
 		regmap_read_poll_timeout(pcm512x->regmap,
 					 PCM512x_ANALOG_MUTE_DET,
 					 mute_det, (mute_det & 0x3) == 0,
 					 200, 10000);
-
-		mutex_unlock(&pcm512x->mutex);
 	} else {
 		pcm512x->mute &= ~0x1;
 		ret = pcm512x_update_mute(pcm512x);
 		if (ret != 0) {
 			dev_err(component->dev,
 				"Failed to update digital mute: %d\n", ret);
-			mutex_unlock(&pcm512x->mutex);
-			return ret;
+			goto unlock;
 		}
 
 		regmap_read_poll_timeout(pcm512x->regmap,
@@ -1428,9 +1424,10 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute)
 					 200, 10000);
 	}
 
+unlock:
 	mutex_unlock(&pcm512x->mutex);
 
-	return 0;
+	return ret;
 }
 
 static const struct snd_soc_dai_ops pcm512x_dai_ops = {

sound/soc/codecs/rt274.c

@@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
-	regmap_read(rt274->regmap,
+	ret = regmap_read(rt274->regmap,
 		RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val);
+	if (ret)
+		return ret;
+
 	if (val != RT274_VENDOR_ID) {
 		dev_err(&i2c->dev,
 			"Device with ID register %#x is not rt274\n", val);

sound/soc/codecs/rt5514-spi.c

@@ -280,6 +280,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component)
 
 	rt5514_dsp = devm_kzalloc(component->dev, sizeof(*rt5514_dsp),
 			GFP_KERNEL);
+	if (!rt5514_dsp)
+		return -ENOMEM;
 
 	rt5514_dsp->dev = &rt5514_spi->dev;
 	mutex_init(&rt5514_dsp->dma_lock);

sound/soc/codecs/rt5682.c

@@ -2512,6 +2512,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 	regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000);
 	regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000);
 	regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005);
+	regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4);
 
 	mutex_unlock(&rt5682->calibrate_mutex);
 

sound/soc/codecs/rt5682.h

@@ -849,18 +849,18 @@
 #define RT5682_SCLK_SRC_PLL2			(0x2 << 13)
 #define RT5682_SCLK_SRC_SDW			(0x3 << 13)
 #define RT5682_SCLK_SRC_RCCLK			(0x4 << 13)
-#define RT5682_PLL1_SRC_MASK			(0x3 << 10)
-#define RT5682_PLL1_SRC_SFT			10
-#define RT5682_PLL1_SRC_MCLK			(0x0 << 10)
-#define RT5682_PLL1_SRC_BCLK1			(0x1 << 10)
-#define RT5682_PLL1_SRC_SDW			(0x2 << 10)
-#define RT5682_PLL1_SRC_RC			(0x3 << 10)
-#define RT5682_PLL2_SRC_MASK			(0x3 << 8)
-#define RT5682_PLL2_SRC_SFT			8
-#define RT5682_PLL2_SRC_MCLK			(0x0 << 8)
-#define RT5682_PLL2_SRC_BCLK1			(0x1 << 8)
-#define RT5682_PLL2_SRC_SDW			(0x2 << 8)
-#define RT5682_PLL2_SRC_RC			(0x3 << 8)
+#define RT5682_PLL2_SRC_MASK			(0x3 << 10)
+#define RT5682_PLL2_SRC_SFT			10
+#define RT5682_PLL2_SRC_MCLK			(0x0 << 10)
+#define RT5682_PLL2_SRC_BCLK1			(0x1 << 10)
+#define RT5682_PLL2_SRC_SDW			(0x2 << 10)
+#define RT5682_PLL2_SRC_RC			(0x3 << 10)
+#define RT5682_PLL1_SRC_MASK			(0x3 << 8)
+#define RT5682_PLL1_SRC_SFT			8
+#define RT5682_PLL1_SRC_MCLK			(0x0 << 8)
+#define RT5682_PLL1_SRC_BCLK1			(0x1 << 8)
+#define RT5682_PLL1_SRC_SDW			(0x2 << 8)
+#define RT5682_PLL1_SRC_RC			(0x3 << 8)
 
 
 

sound/soc/codecs/tlv320aic32x4.c

@@ -822,6 +822,10 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
+		/* Initial cold start */
+		if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF)
+			break;
+
 		/* Switch off BCLK_N Divider */
 		snd_soc_component_update_bits(component, AIC32X4_BCLKN,
 				    AIC32X4_BCLKEN, 0);

sound/soc/fsl/imx-audmux.c

@@ -86,49 +86,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
-	ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
+	ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
 		       pdcr, ptcr);
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxFS output from %s, ",
 				audmux_port_string((ptcr >> 27) & 0x7));
 	else
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxFS input, ");
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxClk output from %s",
 				audmux_port_string((ptcr >> 22) & 0x7));
 	else
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxClk input");
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) {
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"Port is symmetric");
 	} else {
 		if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR)
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxFS output from %s, ",
 					audmux_port_string((ptcr >> 17) & 0x7));
 		else
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxFS input, ");
 
 		if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR)
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxClk output from %s",
 					audmux_port_string((ptcr >> 12) & 0x7));
 		else
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxClk input");
 	}
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret,
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 			"\nData received from %s\n",
 			audmux_port_string((pdcr >> 13) & 0x7));
 

sound/soc/intel/Kconfig

@@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI
 config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
 	default ACPI
-	depends on X86 && ACPI
+	depends on X86 && ACPI && PCI
 	select SND_SST_IPC_ACPI
 	select SND_SST_ATOM_HIFI2_PLATFORM
 	select SND_SOC_ACPI_INTEL_MATCH

sound/soc/intel/atom/sst-mfld-platform-pcm.c

@@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	int ret;
+
+	ret =
+		snd_pcm_lib_malloc_pages(substream,
+				params_buffer_bytes(params));
+	if (ret)
+		return ret;
 	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
 	return 0;
 }

sound/soc/intel/boards/broadwell.c

@@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},

sound/soc/intel/boards/glk_rt5682_max98357a.c

@@ -55,39 +55,6 @@ enum {
 	GLK_DPCM_AUDIO_HDMI3_PB,
 };
 
-static int platform_clock_control(struct snd_soc_dapm_widget *w,
-					struct snd_kcontrol *k, int  event)
-{
-	struct snd_soc_dapm_context *dapm = w->dapm;
-	struct snd_soc_card *card = dapm->card;
-	struct snd_soc_dai *codec_dai;
-	int ret = 0;
-
-	codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI);
-	if (!codec_dai) {
-		dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
-		return -EIO;
-	}
-
-	if (SND_SOC_DAPM_EVENT_OFF(event)) {
-		ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
-		if (ret)
-			dev_err(card->dev, "failed to stop sysclk: %d\n", ret);
-	} else if (SND_SOC_DAPM_EVENT_ON(event)) {
-		ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK,
-					GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ);
-		if (ret < 0) {
-			dev_err(card->dev, "can't set codec pll: %d\n", ret);
-			return ret;
-		}
-	}
-
-	if (ret)
-		dev_err(card->dev, "failed to start internal clk: %d\n", ret);
-
-	return ret;
-}
-
 static const struct snd_kcontrol_new geminilake_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
@@ -102,14 +69,10 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = {
 	SND_SOC_DAPM_SPK("HDMI1", NULL),
 	SND_SOC_DAPM_SPK("HDMI2", NULL),
 	SND_SOC_DAPM_SPK("HDMI3", NULL),
-	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
-			platform_clock_control, SND_SOC_DAPM_PRE_PMU |
-			SND_SOC_DAPM_POST_PMD),
 };
 
 static const struct snd_soc_dapm_route geminilake_map[] = {
 	/* HP jack connectors - unknown if we have jack detection */
-	{ "Headphone Jack", NULL, "Platform Clock" },
 	{ "Headphone Jack", NULL, "HPOL" },
 	{ "Headphone Jack", NULL, "HPOR" },
 
@@ -117,7 +80,6 @@ static const struct snd_soc_dapm_route geminilake_map[] = {
 	{ "Spk", NULL, "Speaker" },
 
 	/* other jacks */
-	{ "Headset Mic", NULL, "Platform Clock" },
 	{ "IN1P", NULL, "Headset Mic" },
 
 	/* digital mics */
@@ -177,6 +139,13 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_jack *jack;
 	int ret;
 
+	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK,
+					GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+		return ret;
+	}
+
 	/* Configure sysclk for codec */
 	ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1,
 					RT5682_PLL_FREQ, SND_SOC_CLOCK_IN);

sound/soc/intel/boards/haswell.c

@@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},

sound/soc/intel/skylake/skl.c

@@ -336,9 +336,6 @@ static int skl_suspend(struct device *dev)
 		skl->skl_sst->fw_loaded = false;
 	}
 
-	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
-		snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false);
-
 	return 0;
 }
 
@@ -350,10 +347,6 @@ static int skl_resume(struct device *dev)
 	struct hdac_ext_link *hlink = NULL;
 	int ret;
 
-	/* Turned OFF in HDMI codec driver after codec reconfiguration */
-	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
-		snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true);
-
 	/*
 	 * resume only when we are not in suspend active, otherwise need to
 	 * restore the device
@@ -446,8 +439,10 @@ static int skl_free(struct hdac_bus *bus)
 	snd_hdac_ext_bus_exit(bus);
 
 	cancel_work_sync(&skl->probe_work);
-	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI))
+	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
+		snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false);
 		snd_hdac_i915_exit(bus);
+	}
 
 	return 0;
 }
@@ -814,7 +809,7 @@ static void skl_probe_work(struct work_struct *work)
 	err = skl_platform_register(bus->dev);
 	if (err < 0) {
 		dev_err(bus->dev, "platform register failed: %d\n", err);
-		return;
+		goto out_err;
 	}
 
 	err = skl_machine_device_register(skl);

sound/soc/qcom/qdsp6/q6asm-dai.c

@@ -570,10 +570,10 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
 	prtd->audio_client = q6asm_audio_client_alloc(dev,
 					(q6asm_cb)compress_event_handler,
 					prtd, stream_id, LEGACY_PCM_MODE);
-	if (!prtd->audio_client) {
+	if (IS_ERR(prtd->audio_client)) {
 		dev_err(dev, "Could not allocate memory\n");
-		kfree(prtd);
-		return -ENOMEM;
+		ret = PTR_ERR(prtd->audio_client);
+		goto free_prtd;
 	}
 
 	size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE *
@@ -582,7 +582,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
 				  &prtd->dma_buffer);
 	if (ret) {
 		dev_err(dev, "Cannot allocate buffer(s)\n");
-		return ret;
+		goto free_client;
 	}
 
 	if (pdata->sid < 0)
@@ -595,6 +595,13 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream)
 	runtime->private_data = prtd;
 
 	return 0;
+
+free_client:
+	q6asm_audio_client_free(prtd->audio_client);
+free_prtd:
+	kfree(prtd);
+
+	return ret;
 }
 
 static int q6asm_dai_compr_free(struct snd_compr_stream *stream)
@@ -874,7 +881,7 @@ static int of_q6asm_parse_dai_data(struct device *dev,
 
 	for_each_child_of_node(dev->of_node, node) {
 		ret = of_property_read_u32(node, "reg", &id);
-		if (ret || id > MAX_SESSIONS || id < 0) {
+		if (ret || id >= MAX_SESSIONS || id < 0) {
 			dev_err(dev, "valid dai id not found:%d\n", ret);
 			continue;
 		}

sound/soc/qcom/sdm845.c

@@ -158,17 +158,24 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static void sdm845_jack_free(struct snd_jack *jack)
+{
+	struct snd_soc_component *component = jack->private_data;
+
+	snd_soc_component_set_jack(component, NULL, NULL);
+}
+
 static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_component *component;
-	struct snd_soc_dai_link *dai_link = rtd->dai_link;
 	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
-	int i, rval;
+	struct snd_jack *jack;
+	int rval;
 
 	if (!pdata->jack_setup) {
-		struct snd_jack *jack;
-
 		rval = snd_soc_card_jack_new(card, "Headset Jack",
 				SND_JACK_HEADSET |
 				SND_JACK_HEADPHONE |
@@ -190,16 +197,22 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
 		pdata->jack_setup = true;
 	}
 
-	for (i = 0 ; i < dai_link->num_codecs; i++) {
-		struct snd_soc_dai *dai = rtd->codec_dais[i];
+	switch (cpu_dai->id) {
+	case PRIMARY_MI2S_RX:
+		jack  = pdata->jack.jack;
+		component = codec_dai->component;
 
-		component = dai->component;
-		rval = snd_soc_component_set_jack(
-				component, &pdata->jack, NULL);
+		jack->private_data = component;
+		jack->private_free = sdm845_jack_free;
+		rval = snd_soc_component_set_jack(component,
+						  &pdata->jack, NULL);
 		if (rval != 0 && rval != -ENOTSUPP) {
 			dev_warn(card->dev, "Failed to set jack: %d\n", rval);
 			return rval;
 		}
+		break;
+	default:
+		break;
 	}
 
 	return 0;

sound/soc/sh/dma-sh7760.c

@@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id];
 
-	pr_debug("PCM data: addr 0x%08ulx len %d\n",
+	pr_debug("PCM data: addr 0x%08lx len %d\n",
 		 (u32)runtime->dma_addr, runtime->dma_bytes);
  
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {

sound/soc/soc-core.c

@@ -742,7 +742,7 @@ static struct snd_soc_component *soc_find_component(
 		if (of_node) {
 			if (component->dev->of_node == of_node)
 				return component;
-		} else if (strcmp(component->name, name) == 0) {
+		} else if (name && strcmp(component->name, name) == 0) {
 			return component;
 		}
 	}
@@ -1034,17 +1034,18 @@ static int snd_soc_init_platform(struct snd_soc_card *card,
 	 * this function should be removed in the future
 	 */
 	/* convert Legacy platform link */
-	if (!platform) {
+	if (!platform || dai_link->legacy_platform) {
 		platform = devm_kzalloc(card->dev,
 				sizeof(struct snd_soc_dai_link_component),
 				GFP_KERNEL);
 		if (!platform)
 			return -ENOMEM;
 
-		dai_link->platform	= platform;
-		platform->name		= dai_link->platform_name;
-		platform->of_node	= dai_link->platform_of_node;
-		platform->dai_name	= NULL;
+		dai_link->platform	  = platform;
+		dai_link->legacy_platform = 1;
+		platform->name		  = dai_link->platform_name;
+		platform->of_node	  = dai_link->platform_of_node;
+		platform->dai_name	  = NULL;
 	}
 
 	/* if there's no platform we match on the empty platform */
@@ -1129,6 +1130,15 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 			link->name);
 		return -EINVAL;
 	}
+
+	/*
+	 * Defer card registartion if platform dai component is not added to
+	 * component list.
+	 */
+	if ((link->platform->of_node || link->platform->name) &&
+	    !soc_find_component(link->platform->of_node, link->platform->name))
+		return -EPROBE_DEFER;
+
 	/*
 	 * CPU device may be specified by either name or OF node, but
 	 * can be left unspecified, and will be matched based on DAI
@@ -1140,6 +1150,15 @@ static int soc_init_dai_link(struct snd_soc_card *card,
 			link->name);
 		return -EINVAL;
 	}
+
+	/*
+	 * Defer card registartion if cpu dai component is not added to
+	 * component list.
+	 */
+	if ((link->cpu_of_node || link->cpu_name) &&
+	    !soc_find_component(link->cpu_of_node, link->cpu_name))
+		return -EPROBE_DEFER;
+
 	/*
 	 * At least one of CPU DAI name or CPU device name/node must be
 	 * specified
@@ -2739,15 +2758,18 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	if (!card->name || !card->dev)
 		return -EINVAL;
 
+	mutex_lock(&client_mutex);
 	for_each_card_prelinks(card, i, link) {
 
 		ret = soc_init_dai_link(card, link);
 		if (ret) {
 			dev_err(card->dev, "ASoC: failed to init link %s\n",
 				link->name);
+			mutex_unlock(&client_mutex);
 			return ret;
 		}
 	}
+	mutex_unlock(&client_mutex);
 
 	dev_set_drvdata(card->dev, card);
 

sound/soc/soc-dapm.c

@@ -2019,19 +2019,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 		out = is_connected_output_ep(w, NULL, NULL);
 	}
 
-	ret = snprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
+	ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
 		       w->name, w->power ? "On" : "Off",
 		       w->force ? " (forced)" : "", in, out);
 
 	if (w->reg >= 0)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				" - R%d(0x%x) mask 0x%x",
 				w->reg, w->reg, w->mask << w->shift);
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
 	if (w->sname)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
 				w->sname,
 				w->active ? "active" : "inactive");
 
@@ -2044,7 +2044,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 			if (!p->connect)
 				continue;
 
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					" %s  \"%s\" \"%s\"\n",
 					(rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out",
 					p->name ? p->name : "static",

sound/soc/ti/davinci-mcasp.c

@@ -108,7 +108,7 @@ struct davinci_mcasp {
 	/* Used for comstraint setting on the second stream */
 	u32	channels;
 
-#ifdef CONFIG_PM_SLEEP
+#ifdef CONFIG_PM
 	struct davinci_mcasp_context context;
 #endif
 
@@ -1486,74 +1486,6 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-#ifdef CONFIG_PM_SLEEP
-static int davinci_mcasp_suspend(struct snd_soc_dai *dai)
-{
-	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
-	struct davinci_mcasp_context *context = &mcasp->context;
-	u32 reg;
-	int i;
-
-	context->pm_state = pm_runtime_active(mcasp->dev);
-	if (!context->pm_state)
-		pm_runtime_get_sync(mcasp->dev);
-
-	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
-		context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]);
-
-	if (mcasp->txnumevt) {
-		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
-		context->afifo_regs[0] = mcasp_get_reg(mcasp, reg);
-	}
-	if (mcasp->rxnumevt) {
-		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
-		context->afifo_regs[1] = mcasp_get_reg(mcasp, reg);
-	}
-
-	for (i = 0; i < mcasp->num_serializer; i++)
-		context->xrsr_regs[i] = mcasp_get_reg(mcasp,
-						DAVINCI_MCASP_XRSRCTL_REG(i));
-
-	pm_runtime_put_sync(mcasp->dev);
-
-	return 0;
-}
-
-static int davinci_mcasp_resume(struct snd_soc_dai *dai)
-{
-	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
-	struct davinci_mcasp_context *context = &mcasp->context;
-	u32 reg;
-	int i;
-
-	pm_runtime_get_sync(mcasp->dev);
-
-	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
-		mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]);
-
-	if (mcasp->txnumevt) {
-		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
-		mcasp_set_reg(mcasp, reg, context->afifo_regs[0]);
-	}
-	if (mcasp->rxnumevt) {
-		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
-		mcasp_set_reg(mcasp, reg, context->afifo_regs[1]);
-	}
-
-	for (i = 0; i < mcasp->num_serializer; i++)
-		mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
-			      context->xrsr_regs[i]);
-
-	if (!context->pm_state)
-		pm_runtime_put_sync(mcasp->dev);
-
-	return 0;
-}
-#else
-#define davinci_mcasp_suspend NULL
-#define davinci_mcasp_resume NULL
-#endif
-
 #define DAVINCI_MCASP_RATES	SNDRV_PCM_RATE_8000_192000
 
 #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
@@ -1571,8 +1503,6 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 	{
 		.name		= "davinci-mcasp.0",
 		.probe		= davinci_mcasp_dai_probe,
-		.suspend	= davinci_mcasp_suspend,
-		.resume		= davinci_mcasp_resume,
 		.playback	= {
 			.channels_min	= 1,
 			.channels_max	= 32 * 16,
@@ -1976,7 +1906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	}
 
 	mcasp->num_serializer = pdata->num_serializer;
-#ifdef CONFIG_PM_SLEEP
+#ifdef CONFIG_PM
 	mcasp->context.xrsr_regs = devm_kcalloc(&pdev->dev,
 					mcasp->num_serializer, sizeof(u32),
 					GFP_KERNEL);
@@ -2196,11 +2126,73 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int davinci_mcasp_runtime_suspend(struct device *dev)
+{
+	struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
+	struct davinci_mcasp_context *context = &mcasp->context;
+	u32 reg;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+		context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]);
+
+	if (mcasp->txnumevt) {
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		context->afifo_regs[0] = mcasp_get_reg(mcasp, reg);
+	}
+	if (mcasp->rxnumevt) {
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		context->afifo_regs[1] = mcasp_get_reg(mcasp, reg);
+	}
+
+	for (i = 0; i < mcasp->num_serializer; i++)
+		context->xrsr_regs[i] = mcasp_get_reg(mcasp,
+						DAVINCI_MCASP_XRSRCTL_REG(i));
+
+	return 0;
+}
+
+static int davinci_mcasp_runtime_resume(struct device *dev)
+{
+	struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
+	struct davinci_mcasp_context *context = &mcasp->context;
+	u32 reg;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+		mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]);
+
+	if (mcasp->txnumevt) {
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		mcasp_set_reg(mcasp, reg, context->afifo_regs[0]);
+	}
+	if (mcasp->rxnumevt) {
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		mcasp_set_reg(mcasp, reg, context->afifo_regs[1]);
+	}
+
+	for (i = 0; i < mcasp->num_serializer; i++)
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+			      context->xrsr_regs[i]);
+
+	return 0;
+}
+
+#endif
+
+static const struct dev_pm_ops davinci_mcasp_pm_ops = {
+	SET_RUNTIME_PM_OPS(davinci_mcasp_runtime_suspend,
+			   davinci_mcasp_runtime_resume,
+			   NULL)
+};
+
 static struct platform_driver davinci_mcasp_driver = {
 	.probe		= davinci_mcasp_probe,
 	.remove		= davinci_mcasp_remove,
 	.driver		= {
 		.name	= "davinci-mcasp",
+		.pm     = &davinci_mcasp_pm_ops,
 		.of_match_table = mcasp_dt_ids,
 	},
 };

sound/soc/xilinx/Kconfig

 config SND_SOC_XILINX_I2S
-	tristate "Audio support for the the Xilinx I2S"
+	tristate "Audio support for the Xilinx I2S"
 	help
 	  Select this option to enable Xilinx I2S Audio. This enables
 	  I2S playback and capture using xilinx soft IP. In transmitter

sound/soc/xilinx/xlnx_i2s.c

 // SPDX-License-Identifier: GPL-2.0
-/*
- * Xilinx ASoC I2S audio support
- *
- * Copyright (C) 2018 Xilinx, Inc.
- *
- * Author: Praveen Vuppala <praveenv@xilinx.com>
- * Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com>
- */
+//
+// Xilinx ASoC I2S audio support
+//
+// Copyright (C) 2018 Xilinx, Inc.
+//
+// Author: Praveen Vuppala <praveenv@xilinx.com>
+// Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com>
 
 #include <linux/io.h>
 #include <linux/module.h>

sound/usb/quirks.c

@@ -768,7 +768,7 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
 		 * REG1: PLL binary search enable, soft mute enable.
 		 */
 		CM6206_REG1_PLLBIN_EN |
-		CM6206_REG1_SOFT_MUTE_EN |
+		CM6206_REG1_SOFT_MUTE_EN,
 		/*
 		 * REG2: enable output drivers,
 		 * select front channels to the headphone output,
@@ -1492,6 +1492,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
 		break;
 
+	case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */
 	case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */
 	case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */
 	case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */