Merge branch 'for-linus' into for-next

Back-merge for applying the more HD-audio quirks on top of the latest
code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

MAINTAINERS

@@ -13931,6 +13931,7 @@ S:	Supported
 F:	Documentation/devicetree/bindings/sound/
 F:	Documentation/sound/soc/
 F:	sound/soc/
+F:	include/dt-bindings/sound/
 F:	include/sound/soc*
 
 SOUNDWIRE SUBSYSTEM

include/sound/pcm_params.h

@@ -254,11 +254,13 @@ static inline int snd_interval_empty(const struct snd_interval *i)
 static inline int snd_interval_single(const struct snd_interval *i)
 {
 	return (i->min == i->max || 
-		(i->min + 1 == i->max && i->openmax));
+		(i->min + 1 == i->max && (i->openmin || i->openmax)));
 }
 
 static inline int snd_interval_value(const struct snd_interval *i)
 {
+	if (i->openmin && !i->openmax)
+		return i->max;
 	return i->min;
 }
 

include/sound/soc.h

@@ -1192,7 +1192,7 @@ struct snd_soc_pcm_runtime {
 	     ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \
 	     (i)++)
 #define for_each_rtd_codec_dai_rollback(rtd, i, dai)		\
-	for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);)
+	for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);)
 
 
 /* mixer control */

sound/core/pcm_native.c

@@ -36,6 +36,7 @@
 #include <sound/timer.h>
 #include <sound/minors.h>
 #include <linux/uio.h>
+#include <linux/delay.h>
 
 #include "pcm_local.h"
 
@@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem);
  * and this may lead to a deadlock when the code path takes read sem
  * twice (e.g. one in snd_pcm_action_nonatomic() and another in
  * snd_pcm_stream_lock()).  As a (suboptimal) workaround, let writer to
- * spin until it gets the lock.
+ * sleep until all the readers are completed without blocking by writer.
  */
-static inline void down_write_nonblock(struct rw_semaphore *lock)
+static inline void down_write_nonfifo(struct rw_semaphore *lock)
 {
 	while (!down_write_trylock(lock))
-		cond_resched();
+		msleep(1);
 }
 
 #define PCM_LOCK_DEFAULT	0
@@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 		res = -ENOMEM;
 		goto _nolock;
 	}
-	down_write_nonblock(&snd_pcm_link_rwsem);
+	down_write_nonfifo(&snd_pcm_link_rwsem);
 	write_lock_irq(&snd_pcm_link_rwlock);
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
 	    substream->runtime->status->state != substream1->runtime->status->state ||
@@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
 	struct snd_pcm_substream *s;
 	int res = 0;
 
-	down_write_nonblock(&snd_pcm_link_rwsem);
+	down_write_nonfifo(&snd_pcm_link_rwsem);
 	write_lock_irq(&snd_pcm_link_rwlock);
 	if (!snd_pcm_stream_linked(substream)) {
 		res = -EALREADY;
@@ -2369,7 +2370,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
 
 static void pcm_release_private(struct snd_pcm_substream *substream)
 {
-	snd_pcm_unlink(substream);
+	if (snd_pcm_stream_linked(substream))
+		snd_pcm_unlink(substream);
 }
 
 void snd_pcm_release_substream(struct snd_pcm_substream *substream)

sound/pci/hda/hda_intel.c

@@ -2498,6 +2498,10 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD Hudson */
 	{ PCI_DEVICE(0x1022, 0x780d),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD Stoney */
+	{ PCI_DEVICE(0x1022, 0x157a),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
+			 AZX_DCAPS_PM_RUNTIME },
 	/* AMD Raven */
 	{ PCI_DEVICE(0x1022, 0x15e3),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |

sound/pci/hda/patch_realtek.c

@@ -388,6 +388,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0285:
 	case 0x10ec0298:
 	case 0x10ec0289:
+	case 0x10ec0300:
 		alc_update_coef_idx(codec, 0x10, 1<<9, 0);
 		break;
 	case 0x10ec0275:
@@ -2830,6 +2831,7 @@ enum {
 	ALC269_TYPE_ALC215,
 	ALC269_TYPE_ALC225,
 	ALC269_TYPE_ALC294,
+	ALC269_TYPE_ALC300,
 	ALC269_TYPE_ALC700,
 };
 
@@ -2864,6 +2866,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC215:
 	case ALC269_TYPE_ALC225:
 	case ALC269_TYPE_ALC294:
+	case ALC269_TYPE_ALC300:
 	case ALC269_TYPE_ALC700:
 		ssids = alc269_ssids;
 		break;
@@ -4985,9 +4988,18 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec,
 		{ 0x19, 0x21a11010 }, /* dock mic */
 		{ }
 	};
+	/* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise
+	 * the speaker output becomes too low by some reason on Thinkpads with
+	 * ALC298 codec
+	 */
+	static hda_nid_t preferred_pairs[] = {
+		0x14, 0x03, 0x17, 0x02, 0x21, 0x02,
+		0
+	};
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->gen.preferred_dacs = preferred_pairs;
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 		snd_hda_apply_pincfgs(codec, pincfgs);
 	} else if (action == HDA_FIXUP_ACT_INIT) {
@@ -5358,6 +5370,16 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec,
 	spec->gen.preferred_dacs = preferred_pairs;
 }
 
+/* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */
+static void alc285_fixup_invalidate_dacs(struct hda_codec *codec,
+			      const struct hda_fixup *fix, int action)
+{
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+
+	snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -5492,6 +5514,9 @@ enum {
 	ALC255_FIXUP_DELL_HEADSET_MIC,
 	ALC295_FIXUP_HP_X360,
 	ALC221_FIXUP_HP_HEADSET_MIC,
+	ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
+	ALC295_FIXUP_HP_AUTO_MUTE,
+	ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5656,6 +5681,8 @@ static const struct hda_fixup alc269_fixups[] = {
 	[ALC269_FIXUP_HP_MUTE_LED_MIC3] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_hp_mute_led_mic3,
+		.chained = true,
+		.chain_id = ALC295_FIXUP_HP_AUTO_MUTE
 	},
 	[ALC269_FIXUP_HP_GPIO_LED] = {
 		.type = HDA_FIXUP_FUNC,
@@ -6359,6 +6386,23 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MIC
 	},
+	[ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_invalidate_dacs,
+	},
+	[ALC295_FIXUP_HP_AUTO_MUTE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_auto_mute_via_amp,
+	},
+	[ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -6373,7 +6417,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
+	SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
+	SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	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),
@@ -7032,6 +7080,15 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x12, 0x90a60130},
 		{0x19, 0x03a11020},
 		{0x21, 0x0321101f}),
+	SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
+		{0x12, 0x90a60130},
+		{0x14, 0x90170110},
+		{0x19, 0x04a11040},
+		{0x21, 0x04211020}),
+	SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
+		{0x12, 0x90a60130},
+		{0x17, 0x90170110},
+		{0x21, 0x02211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
 		{0x12, 0x90a60120},
 		{0x14, 0x90170110},
@@ -7167,6 +7224,37 @@ static void alc269_fill_coef(struct hda_codec *codec)
 	alc_update_coef_idx(codec, 0x4, 0, 1<<11);
 }
 
+static void alc294_hp_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	int i, val;
+
+	if (!hp_pin)
+		return;
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	msleep(100);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+	alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */
+	alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */
+
+	/* Wait for depop procedure finish  */
+	val = alc_read_coefex_idx(codec, 0x58, 0x01);
+	for (i = 0; i < 20 && val & 0x0080; i++) {
+		msleep(50);
+		val = alc_read_coefex_idx(codec, 0x58, 0x01);
+	}
+	/* Set HP depop to auto mode */
+	alc_update_coef_idx(codec, 0x6f, 0x000f, 0x000b);
+	msleep(50);
+}
+
 /*
  */
 static int patch_alc269(struct hda_codec *codec)
@@ -7292,6 +7380,11 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->codec_variant = ALC269_TYPE_ALC294;
 		spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */
 		alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */
+		alc294_hp_init(codec);
+		break;
+	case 0x10ec0300:
+		spec->codec_variant = ALC269_TYPE_ALC300;
+		spec->gen.mixer_nid = 0; /* no loopback on ALC300 */
 		break;
 	case 0x10ec0700:
 	case 0x10ec0701:
@@ -7299,6 +7392,7 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->codec_variant = ALC269_TYPE_ALC700;
 		spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */
 		alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */
+		alc294_hp_init(codec);
 		break;
 
 	}
@@ -8403,6 +8497,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0295, "ALC295", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269),
 	HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
 	HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
 	HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),

sound/soc/codecs/hdac_hdmi.c

@@ -2187,11 +2187,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 	 */
 	snd_hdac_codec_read(hdev, hdev->afg, 0,	AC_VERB_SET_POWER_STATE,
 							AC_PWRST_D3);
-	err = snd_hdac_display_power(bus, false);
-	if (err < 0) {
-		dev_err(dev, "Cannot turn on display power on i915\n");
-		return err;
-	}
 
 	hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev));
 	if (!hlink) {
@@ -2201,7 +2196,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 
 	snd_hdac_ext_bus_link_put(bus, hlink);
 
-	return 0;
+	err = snd_hdac_display_power(bus, false);
+	if (err < 0)
+		dev_err(dev, "Cannot turn off display power on i915\n");
+
+	return err;
 }
 
 static int hdac_hdmi_runtime_resume(struct device *dev)

sound/soc/codecs/pcm186x.h

@@ -139,7 +139,7 @@ enum pcm186x_type {
 #define PCM186X_MAX_REGISTER		PCM186X_CURR_TRIM_CTRL
 
 /* PCM186X_PAGE */
-#define PCM186X_RESET			0xff
+#define PCM186X_RESET			0xfe
 
 /* PCM186X_ADCX_INPUT_SEL_X */
 #define PCM186X_ADC_INPUT_SEL_POL	BIT(7)

sound/soc/codecs/pcm3060.c

@@ -198,20 +198,16 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = {
 };
 
 static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = {
-	SND_SOC_DAPM_OUTPUT("OUTL+"),
-	SND_SOC_DAPM_OUTPUT("OUTR+"),
-	SND_SOC_DAPM_OUTPUT("OUTL-"),
-	SND_SOC_DAPM_OUTPUT("OUTR-"),
+	SND_SOC_DAPM_OUTPUT("OUTL"),
+	SND_SOC_DAPM_OUTPUT("OUTR"),
 
 	SND_SOC_DAPM_INPUT("INL"),
 	SND_SOC_DAPM_INPUT("INR"),
 };
 
 static const struct snd_soc_dapm_route pcm3060_dapm_map[] = {
-	{ "OUTL+", NULL, "Playback" },
-	{ "OUTR+", NULL, "Playback" },
-	{ "OUTL-", NULL, "Playback" },
-	{ "OUTR-", NULL, "Playback" },
+	{ "OUTL", NULL, "Playback" },
+	{ "OUTR", NULL, "Playback" },
 
 	{ "Capture", NULL, "INL" },
 	{ "Capture", NULL, "INR" },

sound/soc/codecs/wm_adsp.c

@@ -765,38 +765,41 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem,
 
 static void wm_adsp2_show_fw_status(struct wm_adsp *dsp)
 {
-	u16 scratch[4];
+	unsigned int scratch[4];
+	unsigned int addr = dsp->base + ADSP2_SCRATCH0;
+	unsigned int i;
 	int ret;
 
-	ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0,
-				scratch, sizeof(scratch));
-	if (ret) {
-		adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret);
-		return;
+	for (i = 0; i < ARRAY_SIZE(scratch); ++i) {
+		ret = regmap_read(dsp->regmap, addr + i, &scratch[i]);
+		if (ret) {
+			adsp_err(dsp, "Failed to read SCRATCH%u: %d\n", i, ret);
+			return;
+		}
 	}
 
 	adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n",
-		 be16_to_cpu(scratch[0]),
-		 be16_to_cpu(scratch[1]),
-		 be16_to_cpu(scratch[2]),
-		 be16_to_cpu(scratch[3]));
+		 scratch[0], scratch[1], scratch[2], scratch[3]);
 }
 
 static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp)
 {
-	u32 scratch[2];
+	unsigned int scratch[2];
 	int ret;
 
-	ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1,
-			      scratch, sizeof(scratch));
-
+	ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1,
+			  &scratch[0]);
 	if (ret) {
-		adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret);
+		adsp_err(dsp, "Failed to read SCRATCH0_1: %d\n", ret);
 		return;
 	}
 
-	scratch[0] = be32_to_cpu(scratch[0]);
-	scratch[1] = be32_to_cpu(scratch[1]);
+	ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH2_3,
+			  &scratch[1]);
+	if (ret) {
+		adsp_err(dsp, "Failed to read SCRATCH2_3: %d\n", ret);
+		return;
+	}
 
 	adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n",
 		 scratch[0] & 0xFFFF,

sound/soc/intel/Kconfig

@@ -101,22 +101,42 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	  codec, then enable this option by saying Y or m. This is a
 	  recommended option
 
-config SND_SOC_INTEL_SKYLAKE_SSP_CLK
-	tristate
-
 config SND_SOC_INTEL_SKYLAKE
 	tristate "SKL/BXT/KBL/GLK/CNL... Platforms"
 	depends on PCI && ACPI
+	select SND_SOC_INTEL_SKYLAKE_COMMON
+	help
+	  If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
+	  GeminiLake or CannonLake platform with the DSP enabled in the BIOS
+	  then enable this option by saying Y or m.
+
+if  SND_SOC_INTEL_SKYLAKE
+
+config SND_SOC_INTEL_SKYLAKE_SSP_CLK
+	tristate
+
+config SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+	bool "HDAudio codec support"
+	help
+	  If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
+	  GeminiLake or CannonLake platform with an HDaudio codec
+	  then enable this option by saying Y
+
+config SND_SOC_INTEL_SKYLAKE_COMMON
+	tristate
 	select SND_HDA_EXT_CORE
 	select SND_HDA_DSP_LOADER
 	select SND_SOC_TOPOLOGY
 	select SND_SOC_INTEL_SST
+	select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
 	select SND_SOC_ACPI_INTEL_MATCH
 	help
 	  If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
 	  GeminiLake or CannonLake platform with the DSP enabled in the BIOS
 	  then enable this option by saying Y or m.
 
+endif ## SND_SOC_INTEL_SKYLAKE
+
 config SND_SOC_ACPI_INTEL_MATCH
 	tristate
 	select SND_SOC_ACPI if ACPI

sound/soc/intel/boards/Kconfig

@@ -293,16 +293,6 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH
 	  Say Y if you have such a device.
 	  If unsure select "N".
 
-config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
-	tristate "SKL/KBL/BXT/APL with HDA Codecs"
-	select SND_SOC_HDAC_HDMI
-	select SND_SOC_HDAC_HDA
-	help
-	  This adds support for ASoC machine driver for Intel platforms
-	  SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
-          Say Y or m if you have such a device. This is a recommended option.
-	  If unsure select "N".
-
 config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
 	tristate "GLK with RT5682 and MAX98357A in I2S Mode"
 	depends on MFD_INTEL_LPSS && I2C && ACPI
@@ -319,4 +309,18 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
 
 endif ## SND_SOC_INTEL_SKYLAKE
 
+if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+
+config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
+	tristate "SKL/KBL/BXT/APL with HDA Codecs"
+	select SND_SOC_HDAC_HDMI
+	# SND_SOC_HDAC_HDA is already selected
+	help
+	  This adds support for ASoC machine driver for Intel platforms
+	  SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
+          Say Y or m if you have such a device. This is a recommended option.
+	  If unsure select "N".
+
+endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+
 endif ## SND_SOC_INTEL_MACH

sound/soc/intel/boards/cht_bsw_max98090_ti.c

@@ -19,6 +19,7 @@
  * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  */
 
+#include <linux/dmi.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
@@ -35,6 +36,8 @@
 #define CHT_PLAT_CLK_3_HZ	19200000
 #define CHT_CODEC_DAI	"HiFi"
 
+#define QUIRK_PMC_PLT_CLK_0				0x01
+
 struct cht_mc_private {
 	struct clk *mclk;
 	struct snd_soc_jack jack;
@@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = {
 	.num_controls = ARRAY_SIZE(cht_mc_controls),
 };
 
+static const struct dmi_system_id cht_max98090_quirk_table[] = {
+	{
+		/* Swanky model Chromebook (Toshiba Chromebook 2) */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{}
+};
+
 static int snd_cht_mc_probe(struct platform_device *pdev)
 {
+	const struct dmi_system_id *dmi_id;
 	struct device *dev = &pdev->dev;
 	int ret_val = 0;
 	struct cht_mc_private *drv;
+	const char *mclk_name;
+	int quirks = 0;
+
+	dmi_id = dmi_first_match(cht_max98090_quirk_table);
+	if (dmi_id)
+		quirks = (unsigned long)dmi_id->driver_data;
 
 	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
 	if (!drv)
@@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
 	snd_soc_card_cht.dev = &pdev->dev;
 	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
 
-	drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
+	if (quirks & QUIRK_PMC_PLT_CLK_0)
+		mclk_name = "pmc_plt_clk_0";
+	else
+		mclk_name = "pmc_plt_clk_3";
+
+	drv->mclk = devm_clk_get(&pdev->dev, mclk_name);
 	if (IS_ERR(drv->mclk)) {
 		dev_err(&pdev->dev,
-			"Failed to get MCLK from pmc_plt_clk_3: %ld\n",
-			PTR_ERR(drv->mclk));
+			"Failed to get MCLK from %s: %ld\n",
+			mclk_name, PTR_ERR(drv->mclk));
 		return PTR_ERR(drv->mclk);
 	}
 

sound/soc/intel/skylake/skl.c

@@ -37,7 +37,9 @@
 #include "skl.h"
 #include "skl-sst-dsp.h"
 #include "skl-sst-ipc.h"
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 #include "../../../soc/codecs/hdac_hda.h"
+#endif
 
 /*
  * initialize the PCI registers
@@ -658,6 +660,8 @@ static void skl_clock_device_unregister(struct skl *skl)
 		platform_device_unregister(skl->clk_dev);
 }
 
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
+
 #define IDISP_INTEL_VENDOR_ID	0x80860000
 
 /*
@@ -676,6 +680,8 @@ static void load_codec_module(struct hda_codec *codec)
 #endif
 }
 
+#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */
+
 /*
  * Probe the given codec address
  */
@@ -685,9 +691,11 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	unsigned int res = -1;
 	struct skl *skl = bus_to_skl(bus);
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 	struct hdac_hda_priv *hda_codec;
-	struct hdac_device *hdev;
 	int err;
+#endif
+	struct hdac_device *hdev;
 
 	mutex_lock(&bus->cmd_mutex);
 	snd_hdac_bus_send_cmd(bus, cmd);
@@ -697,6 +705,7 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 		return -EIO;
 	dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res);
 
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 	hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec),
 				 GFP_KERNEL);
 	if (!hda_codec)
@@ -715,6 +724,13 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 		load_codec_module(&hda_codec->codec);
 	}
 	return 0;
+#else
+	hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL);
+	if (!hdev)
+		return -ENOMEM;
+
+	return snd_hdac_ext_bus_device_init(bus, addr, hdev);
+#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */
 }
 
 /* Codec initialization */
@@ -815,6 +831,12 @@ static void skl_probe_work(struct work_struct *work)
 		}
 	}
 
+	/*
+	 * we are done probing so decrement link counts
+	 */
+	list_for_each_entry(hlink, &bus->hlink_list, list)
+		snd_hdac_ext_bus_link_put(bus, hlink);
+
 	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
 		err = snd_hdac_display_power(bus, false);
 		if (err < 0) {
@@ -824,12 +846,6 @@ static void skl_probe_work(struct work_struct *work)
 		}
 	}
 
-	/*
-	 * we are done probing so decrement link counts
-	 */
-	list_for_each_entry(hlink, &bus->hlink_list, list)
-		snd_hdac_ext_bus_link_put(bus, hlink);
-
 	/* configure PM */
 	pm_runtime_put_noidle(bus->dev);
 	pm_runtime_allow(bus->dev);
@@ -870,7 +886,7 @@ static int skl_create(struct pci_dev *pci,
 	hbus = skl_to_hbus(skl);
 	bus = skl_to_bus(skl);
 
-#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA)
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 	ext_ops = snd_soc_hdac_hda_get_ops();
 #endif
 	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops);

sound/soc/omap/omap-abe-twl6040.c

@@ -36,6 +36,8 @@
 #include "../codecs/twl6040.h"
 
 struct abe_twl6040 {
+	struct snd_soc_card card;
+	struct snd_soc_dai_link dai_links[2];
 	int	jack_detection;	/* board can detect jack events */
 	int	mclk_freq;	/* MCLK frequency speed for twl6040 */
 };
@@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
 				ARRAY_SIZE(dmic_audio_map));
 }
 
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
-	{
-		.name = "TWL6040",
-		.stream_name = "TWL6040",
-		.codec_dai_name = "twl6040-legacy",
-		.codec_name = "twl6040-codec",
-		.init = omap_abe_twl6040_init,
-		.ops = &omap_abe_ops,
-	},
-	{
-		.name = "DMIC",
-		.stream_name = "DMIC Capture",
-		.codec_dai_name = "dmic-hifi",
-		.codec_name = "dmic-codec",
-		.init = omap_abe_dmic_init,
-		.ops = &omap_abe_dmic_ops,
-	},
-};
-
-/* Audio machine driver */
-static struct snd_soc_card omap_abe_card = {
-	.owner = THIS_MODULE,
-
-	.dapm_widgets = twl6040_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
-	.dapm_routes = audio_map,
-	.num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
 static int omap_abe_probe(struct platform_device *pdev)
 {
 	struct device_node *node = pdev->dev.of_node;
-	struct snd_soc_card *card = &omap_abe_card;
+	struct snd_soc_card *card;
 	struct device_node *dai_node;
 	struct abe_twl6040 *priv;
 	int num_links = 0;
@@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	card->dev = &pdev->dev;
-
 	priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
 	if (priv == NULL)
 		return -ENOMEM;
 
+	card = &priv->card;
+	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
+	card->dapm_widgets = twl6040_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets);
+	card->dapm_routes = audio_map;
+	card->num_dapm_routes = ARRAY_SIZE(audio_map);
+
 	if (snd_soc_of_parse_card_name(card, "ti,model")) {
 		dev_err(&pdev->dev, "Card name is not provided\n");
 		return -ENODEV;
@@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "McPDM node is not provided\n");
 		return -EINVAL;
 	}
-	abe_twl6040_dai_links[0].cpu_of_node = dai_node;
-	abe_twl6040_dai_links[0].platform_of_node = dai_node;
+
+	priv->dai_links[0].name = "DMIC";
+	priv->dai_links[0].stream_name = "TWL6040";
+	priv->dai_links[0].cpu_of_node = dai_node;
+	priv->dai_links[0].platform_of_node = dai_node;
+	priv->dai_links[0].codec_dai_name = "twl6040-legacy";
+	priv->dai_links[0].codec_name = "twl6040-codec";
+	priv->dai_links[0].init = omap_abe_twl6040_init;
+	priv->dai_links[0].ops = &omap_abe_ops;
 
 	dai_node = of_parse_phandle(node, "ti,dmic", 0);
 	if (dai_node) {
 		num_links = 2;
-		abe_twl6040_dai_links[1].cpu_of_node = dai_node;
-		abe_twl6040_dai_links[1].platform_of_node = dai_node;
+		priv->dai_links[1].name = "TWL6040";
+		priv->dai_links[1].stream_name = "DMIC Capture";
+		priv->dai_links[1].cpu_of_node = dai_node;
+		priv->dai_links[1].platform_of_node = dai_node;
+		priv->dai_links[1].codec_dai_name = "dmic-hifi";
+		priv->dai_links[1].codec_name = "dmic-codec";
+		priv->dai_links[1].init = omap_abe_dmic_init;
+		priv->dai_links[1].ops = &omap_abe_dmic_ops;
 	} else {
 		num_links = 1;
 	}
@@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	card->dai_link = abe_twl6040_dai_links;
+	card->dai_link = priv->dai_links;
 	card->num_links = num_links;
 
 	snd_soc_card_set_drvdata(card, priv);

sound/soc/omap/omap-dmic.c

@@ -48,6 +48,8 @@ struct omap_dmic {
 	struct device *dev;
 	void __iomem *io_base;
 	struct clk *fclk;
+	struct pm_qos_request pm_qos_req;
+	int latency;
 	int fclk_freq;
 	int out_freq;
 	int clk_div;
@@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream,
 
 	mutex_lock(&dmic->mutex);
 
+	pm_qos_remove_request(&dmic->pm_qos_req);
+
 	if (!dai->active)
 		dmic->active = 0;
 
@@ -228,6 +232,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
 	/* packet size is threshold * channels */
 	dma_data = snd_soc_dai_get_dma_data(dai, substream);
 	dma_data->maxburst = dmic->threshold * channels;
+	dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC /
+			params_rate(params);
 
 	return 0;
 }
@@ -238,6 +244,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream,
 	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
 	u32 ctrl;
 
+	if (pm_qos_request_active(&dmic->pm_qos_req))
+		pm_qos_update_request(&dmic->pm_qos_req, dmic->latency);
+
 	/* Configure uplink threshold */
 	omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold);
 

sound/soc/omap/omap-mcbsp.c

@@ -308,9 +308,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 			pkt_size = channels;
 		}
 
-		latency = ((((buffer_size - pkt_size) / channels) * 1000)
-				 / (params->rate_num / params->rate_den));
-
+		latency = (buffer_size - pkt_size) / channels;
+		latency = latency * USEC_PER_SEC /
+			  (params->rate_num / params->rate_den);
 		mcbsp->latency[substream->stream] = latency;
 
 		omap_mcbsp_set_threshold(substream, pkt_size);

sound/soc/omap/omap-mcpdm.c

@@ -54,6 +54,8 @@ struct omap_mcpdm {
 	unsigned long phys_base;
 	void __iomem *io_base;
 	int irq;
+	struct pm_qos_request pm_qos_req;
+	int latency[2];
 
 	struct mutex mutex;
 
@@ -277,6 +279,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock(&mcpdm->mutex);
 
@@ -289,6 +294,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
 		}
 	}
 
+	if (mcpdm->latency[stream2])
+		pm_qos_update_request(&mcpdm->pm_qos_req,
+				      mcpdm->latency[stream2]);
+	else if (mcpdm->latency[stream1])
+		pm_qos_remove_request(&mcpdm->pm_qos_req);
+
+	mcpdm->latency[stream1] = 0;
+
 	mutex_unlock(&mcpdm->mutex);
 }
 
@@ -300,7 +313,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 	int stream = substream->stream;
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	u32 threshold;
-	int channels;
+	int channels, latency;
 	int link_mask = 0;
 
 	channels = params_channels(params);
@@ -344,14 +357,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 
 		dma_data->maxburst =
 				(MCPDM_DN_THRES_MAX - threshold) * channels;
+		latency = threshold;
 	} else {
 		/* If playback is not running assume a stereo stream to come */
 		if (!mcpdm->config[!stream].link_mask)
 			mcpdm->config[!stream].link_mask = (0x3 << 3);
 
 		dma_data->maxburst = threshold * channels;
+		latency = (MCPDM_DN_THRES_MAX - threshold);
 	}
 
+	/*
+	 * The DMA must act to a DMA request within latency time (usec) to avoid
+	 * under/overflow
+	 */
+	mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params);
+
+	if (!mcpdm->latency[stream])
+		mcpdm->latency[stream] = 10;
+
 	/* Check if we need to restart McPDM with this stream */
 	if (mcpdm->config[stream].link_mask &&
 	    mcpdm->config[stream].link_mask != link_mask)
@@ -366,6 +390,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req;
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+	int latency = mcpdm->latency[stream2];
+
+	/* Prevent omap hardware from hitting off between FIFO fills */
+	if (!latency || mcpdm->latency[stream1] < latency)
+		latency = mcpdm->latency[stream1];
+
+	if (pm_qos_request_active(pm_qos_req))
+		pm_qos_update_request(pm_qos_req, latency);
+	else if (latency)
+		pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency);
 
 	if (!omap_mcpdm_active(mcpdm)) {
 		omap_mcpdm_start(mcpdm);
@@ -427,6 +465,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
 	free_irq(mcpdm->irq, (void *)mcpdm);
 	pm_runtime_disable(mcpdm->dev);
 
+	if (pm_qos_request_active(&mcpdm->pm_qos_req))
+		pm_qos_remove_request(&mcpdm->pm_qos_req);
+
 	return 0;
 }
 

sound/soc/qcom/common.c

@@ -13,6 +13,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 	struct device_node *cpu = NULL;
 	struct device *dev = card->dev;
 	struct snd_soc_dai_link *link;
+	struct of_phandle_args args;
 	int ret, num_links;
 
 	ret = snd_soc_of_parse_card_name(card, "model");
@@ -47,12 +48,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			goto err;
 		}
 
-		link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
-		if (!link->cpu_of_node) {
+		ret = of_parse_phandle_with_args(cpu, "sound-dai",
+					"#sound-dai-cells", 0, &args);
+		if (ret) {
 			dev_err(card->dev, "error getting cpu phandle\n");
-			ret = -EINVAL;
 			goto err;
 		}
+		link->cpu_of_node = args.np;
+		link->id = args.args[0];
 
 		ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
 		if (ret) {

sound/soc/qcom/qdsp6/q6afe.c

@@ -49,14 +49,14 @@
 #define AFE_PORT_I2S_SD1		0x2
 #define AFE_PORT_I2S_SD2		0x3
 #define AFE_PORT_I2S_SD3		0x4
-#define AFE_PORT_I2S_SD0_MASK		BIT(0x1)
-#define AFE_PORT_I2S_SD1_MASK		BIT(0x2)
-#define AFE_PORT_I2S_SD2_MASK		BIT(0x3)
-#define AFE_PORT_I2S_SD3_MASK		BIT(0x4)
-#define AFE_PORT_I2S_SD0_1_MASK		GENMASK(2, 1)
-#define AFE_PORT_I2S_SD2_3_MASK		GENMASK(4, 3)
-#define AFE_PORT_I2S_SD0_1_2_MASK	GENMASK(3, 1)
-#define AFE_PORT_I2S_SD0_1_2_3_MASK	GENMASK(4, 1)
+#define AFE_PORT_I2S_SD0_MASK		BIT(0x0)
+#define AFE_PORT_I2S_SD1_MASK		BIT(0x1)
+#define AFE_PORT_I2S_SD2_MASK		BIT(0x2)
+#define AFE_PORT_I2S_SD3_MASK		BIT(0x3)
+#define AFE_PORT_I2S_SD0_1_MASK		GENMASK(1, 0)
+#define AFE_PORT_I2S_SD2_3_MASK		GENMASK(3, 2)
+#define AFE_PORT_I2S_SD0_1_2_MASK	GENMASK(2, 0)
+#define AFE_PORT_I2S_SD0_1_2_3_MASK	GENMASK(3, 0)
 #define AFE_PORT_I2S_QUAD01		0x5
 #define AFE_PORT_I2S_QUAD23		0x6
 #define AFE_PORT_I2S_6CHS		0x7

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

@@ -122,7 +122,6 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
 			.rate_max =	48000,				\
 		},							\
 		.name = "MultiMedia"#num,				\
-		.probe = fe_dai_probe,					\
 		.id = MSM_FRONTEND_DAI_MULTIMEDIA##num,			\
 	}
 
@@ -511,38 +510,6 @@ static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
 	}
 }
 
-static const struct snd_soc_dapm_route afe_pcm_routes[] = {
-	{"MM_DL1",  NULL, "MultiMedia1 Playback" },
-	{"MM_DL2",  NULL, "MultiMedia2 Playback" },
-	{"MM_DL3",  NULL, "MultiMedia3 Playback" },
-	{"MM_DL4",  NULL, "MultiMedia4 Playback" },
-	{"MM_DL5",  NULL, "MultiMedia5 Playback" },
-	{"MM_DL6",  NULL, "MultiMedia6 Playback" },
-	{"MM_DL7",  NULL, "MultiMedia7 Playback" },
-	{"MM_DL7",  NULL, "MultiMedia8 Playback" },
-	{"MultiMedia1 Capture", NULL, "MM_UL1"},
-	{"MultiMedia2 Capture", NULL, "MM_UL2"},
-	{"MultiMedia3 Capture", NULL, "MM_UL3"},
-	{"MultiMedia4 Capture", NULL, "MM_UL4"},
-	{"MultiMedia5 Capture", NULL, "MM_UL5"},
-	{"MultiMedia6 Capture", NULL, "MM_UL6"},
-	{"MultiMedia7 Capture", NULL, "MM_UL7"},
-	{"MultiMedia8 Capture", NULL, "MM_UL8"},
-
-};
-
-static int fe_dai_probe(struct snd_soc_dai *dai)
-{
-	struct snd_soc_dapm_context *dapm;
-
-	dapm = snd_soc_component_get_dapm(dai->component);
-	snd_soc_dapm_add_routes(dapm, afe_pcm_routes,
-				ARRAY_SIZE(afe_pcm_routes));
-
-	return 0;
-}
-
-
 static const struct snd_soc_component_driver q6asm_fe_dai_component = {
 	.name		= DRV_NAME,
 	.ops		= &q6asm_dai_ops,

sound/soc/qcom/qdsp6/q6routing.c

@@ -909,6 +909,25 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MM_UL6", NULL, "MultiMedia6 Mixer"},
 	{"MM_UL7", NULL, "MultiMedia7 Mixer"},
 	{"MM_UL8", NULL, "MultiMedia8 Mixer"},
+
+	{"MM_DL1",  NULL, "MultiMedia1 Playback" },
+	{"MM_DL2",  NULL, "MultiMedia2 Playback" },
+	{"MM_DL3",  NULL, "MultiMedia3 Playback" },
+	{"MM_DL4",  NULL, "MultiMedia4 Playback" },
+	{"MM_DL5",  NULL, "MultiMedia5 Playback" },
+	{"MM_DL6",  NULL, "MultiMedia6 Playback" },
+	{"MM_DL7",  NULL, "MultiMedia7 Playback" },
+	{"MM_DL8",  NULL, "MultiMedia8 Playback" },
+
+	{"MultiMedia1 Capture", NULL, "MM_UL1"},
+	{"MultiMedia2 Capture", NULL, "MM_UL2"},
+	{"MultiMedia3 Capture", NULL, "MM_UL3"},
+	{"MultiMedia4 Capture", NULL, "MM_UL4"},
+	{"MultiMedia5 Capture", NULL, "MM_UL5"},
+	{"MultiMedia6 Capture", NULL, "MM_UL6"},
+	{"MultiMedia7 Capture", NULL, "MM_UL7"},
+	{"MultiMedia8 Capture", NULL, "MM_UL8"},
+
 };
 
 static int routing_hw_params(struct snd_pcm_substream *substream,

sound/soc/rockchip/rockchip_pcm.c

@@ -33,6 +33,7 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = {
 
 static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = {
 	.pcm_hardware = &snd_rockchip_hardware,
+	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
 	.prealloc_buffer_size = 32 * 1024,
 };
 

sound/soc/sh/rcar/ssi.c

@@ -306,7 +306,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	if (rsnd_ssi_is_multi_slave(mod, io))
 		return 0;
 
-	if (ssi->rate) {
+	if (ssi->usrcnt > 1) {
 		if (ssi->rate != rate) {
 			dev_err(dev, "SSI parent/child should use same rate\n");
 			return -EINVAL;

sound/soc/soc-acpi.c

@@ -10,11 +10,17 @@ struct snd_soc_acpi_mach *
 snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines)
 {
 	struct snd_soc_acpi_mach *mach;
+	struct snd_soc_acpi_mach *mach_alt;
 
 	for (mach = machines; mach->id[0]; mach++) {
 		if (acpi_dev_present(mach->id, NULL, -1)) {
-			if (mach->machine_quirk)
-				mach = mach->machine_quirk(mach);
+			if (mach->machine_quirk) {
+				mach_alt = mach->machine_quirk(mach);
+				if (!mach_alt)
+					continue; /* not full match, ignore */
+				mach = mach_alt;
+			}
+
 			return mach;
 		}
 	}

sound/soc/soc-core.c

@@ -2131,6 +2131,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	card->instantiated = 1;
+	dapm_mark_endpoints_dirty(card);
 	snd_soc_dapm_sync(&card->dapm);
 	mutex_unlock(&card->mutex);
 	mutex_unlock(&client_mutex);

sound/soc/stm/stm32_sai_sub.c

@@ -390,7 +390,7 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai)
 	char *mclk_name, *p, *s = (char *)pname;
 	int ret, i = 0;
 
-	mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL);
+	mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL);
 	if (!mclk)
 		return -ENOMEM;
 

sound/soc/sunxi/Kconfig

@@ -31,7 +31,7 @@ config SND_SUN8I_CODEC_ANALOG
 config SND_SUN50I_CODEC_ANALOG
 	tristate "Allwinner sun50i Codec Analog Controls Support"
 	depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST
-	select SND_SUNXI_ADDA_PR_REGMAP
+	select SND_SUN8I_ADDA_PR_REGMAP
 	help
 	  Say Y or M if you want to add support for the analog controls for
 	  the codec embedded in Allwinner A64 SoC.

sound/soc/sunxi/sun8i-codec.c

@@ -481,7 +481,11 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = {
 	{ "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch",
 	  "AIF1 Slot 0 Right"},
 
-	/* ADC routes */
+	/* ADC Routes */
+	{ "AIF1 Slot 0 Right ADC", NULL, "ADC" },
+	{ "AIF1 Slot 0 Left ADC", NULL, "ADC" },
+
+	/* ADC Mixer Routes */
 	{ "Left Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch",
 	  "AIF1 Slot 0 Left ADC" },
 	{ "Right Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch",
@@ -605,16 +609,10 @@ static int sun8i_codec_probe(struct platform_device *pdev)
 
 static int sun8i_codec_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	struct sun8i_codec *scodec = snd_soc_card_get_drvdata(card);
-
 	pm_runtime_disable(&pdev->dev);
 	if (!pm_runtime_status_suspended(&pdev->dev))
 		sun8i_codec_runtime_suspend(&pdev->dev);
 
-	clk_disable_unprepare(scodec->clk_module);
-	clk_disable_unprepare(scodec->clk_bus);
-
 	return 0;
 }
 

sound/usb/card.c

@@ -682,9 +682,12 @@ static int usb_audio_probe(struct usb_interface *intf,
 
  __error:
 	if (chip) {
+		/* chip->active is inside the chip->card object,
+		 * decrement before memory is possibly returned.
+		 */
+		atomic_dec(&chip->active);
 		if (!chip->num_interfaces)
 			snd_card_free(chip->card);
-		atomic_dec(&chip->active);
 	}
 	mutex_unlock(&register_mutex);
 	return err;

sound/usb/quirks-table.h

@@ -3382,5 +3382,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		.ifnum = QUIRK_NO_INTERFACE
 	}
 },
+/* Dell WD19 Dock */
+{
+	USB_DEVICE(0x0bda, 0x402e),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Dell",
+		.product_name = "WD19 Dock",
+		.profile_name = "Dell-WD15-Dock",
+		.ifnum = QUIRK_NO_INTERFACE
+	}
+},
 
 #undef USB_DEVICE_VENDOR_SPEC

sound/usb/quirks.c

@@ -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(0x152a, 0x85de): /* SMSL D1 DAC */
 	case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */
 	case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */
 	case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */