Merge tag 'sound-fix-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A few small remaining wrap-up for this merge window.

  Most of patches are device-specific (HD-audio and USB-audio quirks,
  FireWire, pcm316a, fsl, rsnd, Atmel, and TI fixes), while there is a
  simple fix (actually two commits) for ASoC core"

* tag 'sound-fix-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Add DSD support for EVGA NU Audio
  ALSA: hda - Add laptop imic fixup for ASUS M9V laptop
  ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies
  ASoC: pcm3168a: The codec does not support S32_LE
  ASoC: core: use list_del_init and move it back to soc_cleanup_component
  ALSA: hda/realtek - PCI quirk for Medion E4254
  ALSA: hda - Apply AMD controller workaround for Raven platform
  ASoC: rsnd: do error check after rsnd_channel_normalization()
  ASoC: atmel_ssc_dai: Remove wrong spinlock usage
  ASoC: core: delete component->card_list in soc_remove_component only
  ASoC: fsl_sai: Fix noise when using EDMA
  ALSA: usb-audio: Add Hiby device family to quirks for native DSD support
  ALSA: hda/realtek - Fix alienware headset mic
  ALSA: dice: fix wrong packet parameter for Alesis iO26

13 files changed, 67 insertions, 24 deletions

diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c
index 218292bdace6..f5b325263b67 100644
--- a/sound/firewire/dice/dice-alesis.c
+++ b/sound/firewire/dice/dice-alesis.c
@@ -15,7 +15,7 @@ alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
 
 static const unsigned int
 alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
-	{10, 10, 8},	/* Tx0 = Analog + S/PDIF. */
+	{10, 10, 4},	/* Tx0 = Analog + S/PDIF. */
 	{16, 8, 0},	/* Tx1 = ADAT1 + ADAT2. */
 };
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 91e71be42fa4..240f4ca76391 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2485,8 +2485,7 @@ static const struct pci_device_id azx_ids[] = {
 			 AZX_DCAPS_PM_RUNTIME },
 	/* AMD Raven */
 	{ PCI_DEVICE(0x1022, 0x15e3),
-	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
-			 AZX_DCAPS_PM_RUNTIME },
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* ATI HDMI */
 	{ PCI_DEVICE(0x1002, 0x0002),
 	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e283966bdbb1..bc9dd8e6fd86 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -357,6 +357,7 @@ static const struct hda_fixup ad1986a_fixups[] = {
 
 static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+	SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9V", AD1986A_FIXUP_LAPTOP_IMIC),
 	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD),
 	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index da1695418731..b000b36ac3c6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5817,6 +5817,7 @@ enum {
 	ALC292_FIXUP_DELL_E7X,
 	ALC292_FIXUP_DISABLE_AAMIX,
 	ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK,
+	ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE,
 	ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE,
 	ALC275_FIXUP_DELL_XPS,
@@ -5871,6 +5872,7 @@ enum {
 	ALC256_FIXUP_ASUS_MIC_NO_PRESENCE,
 	ALC299_FIXUP_PREDATOR_SPK,
 	ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC,
+	ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -6506,6 +6508,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC292_FIXUP_DISABLE_AAMIX
 	},
+	[ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x01a1913c }, /* headset mic w/o jack detect */
+			{ }
+		},
+		.chained_before = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE,
+	},
 	[ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -6927,6 +6938,16 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
 	},
+	[ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x04a11040 },
+			{ 0x21, 0x04211020 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7190,6 +7211,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
+	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
 
 #if 0
 	/* Below is a quirk table taken from the old code.
@@ -7358,6 +7380,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"},
 	{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
 	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
+	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
@@ -7770,6 +7793,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x17, 0x90170110},
 		{0x1a, 0x03011020},
 		{0x21, 0x03211030}),
+	SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE,
+		{0x12, 0xb7a60140},
+		{0x17, 0x90170110},
+		{0x1a, 0x03a11030},
+		{0x21, 0x03211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
 		ALC225_STANDARD_PINS,
 		{0x12, 0xb7a60130},
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 48e9eef34c0f..ca603397651c 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
 static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = {
 	{
 	.name		= "ssc0",
-	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
 	.dir_mask	= SSC_DIR_MASK_UNUSED,
 	.initialized	= 0,
 	},
 	{
 	.name		= "ssc1",
-	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
 	.dir_mask	= SSC_DIR_MASK_UNUSED,
 	.initialized	= 0,
 	},
 	{
 	.name		= "ssc2",
-	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
 	.dir_mask	= SSC_DIR_MASK_UNUSED,
 	.initialized	= 0,
 	},
@@ -317,13 +314,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
 
 	snd_soc_dai_set_dma_data(dai, substream, dma_params);
 
-	spin_lock_irq(&ssc_p->lock);
-	if (ssc_p->dir_mask & dir_mask) {
-		spin_unlock_irq(&ssc_p->lock);
+	if (ssc_p->dir_mask & dir_mask)
 		return -EBUSY;
-	}
+
 	ssc_p->dir_mask |= dir_mask;
-	spin_unlock_irq(&ssc_p->lock);
 
 	return 0;
 }
@@ -355,7 +349,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
 
 	dir_mask = 1 << dir;
 
-	spin_lock_irq(&ssc_p->lock);
 	ssc_p->dir_mask &= ~dir_mask;
 	if (!ssc_p->dir_mask) {
 		if (ssc_p->initialized) {
@@ -369,7 +362,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
 		ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
 		ssc_p->forced_divider = 0;
 	}
-	spin_unlock_irq(&ssc_p->lock);
 
 	/* Shutdown the SSC clock. */
 	pr_debug("atmel_ssc_dai: Stopping clock\n");
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
index ae764cb541c7..3470b966e449 100644
--- a/sound/soc/atmel/atmel_ssc_dai.h
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -93,7 +93,6 @@ struct atmel_ssc_state {
 struct atmel_ssc_info {
 	char *name;
 	struct ssc_device *ssc;
-	spinlock_t lock;	/* lock for dir_mask */
 	unsigned short dir_mask;	/* 0=unused, 1=playback, 2=capture */
 	unsigned short initialized;	/* true if SSC has been initialized */
 	unsigned short daifmt;
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 50ed86d45c26..88b75695fbf7 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -21,8 +21,7 @@
 
 #define PCM3168A_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | \
-			 SNDRV_PCM_FMTBIT_S24_LE | \
-			 SNDRV_PCM_FMTBIT_S32_LE)
+			 SNDRV_PCM_FMTBIT_S24_LE)
 
 #define PCM3168A_FMT_I2S		0x0
 #define PCM3168A_FMT_LEFT_J		0x1
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ef0b74693093..b517e4bc1b87 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -628,6 +628,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
 			   FSL_SAI_CR3_TRCE_MASK,
 			   FSL_SAI_CR3_TRCE);
 
+	/*
+	 * EDMA controller needs period size to be a multiple of
+	 * tx/rx maxburst
+	 */
+	if (sai->soc_data->use_edma)
+		snd_pcm_hw_constraint_step(substream->runtime, 0,
+					   SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+					   tx ? sai->dma_params_tx.maxburst :
+					   sai->dma_params_rx.maxburst);
+
 	ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
 			SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints);
 
@@ -1026,30 +1036,35 @@ static int fsl_sai_remove(struct platform_device *pdev)
 
 static const struct fsl_sai_soc_data fsl_sai_vf610_data = {
 	.use_imx_pcm = false,
+	.use_edma = false,
 	.fifo_depth = 32,
 	.reg_offset = 0,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = {
 	.use_imx_pcm = true,
+	.use_edma = false,
 	.fifo_depth = 32,
 	.reg_offset = 0,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = {
 	.use_imx_pcm = true,
+	.use_edma = false,
 	.fifo_depth = 16,
 	.reg_offset = 8,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = {
 	.use_imx_pcm = true,
+	.use_edma = false,
 	.fifo_depth = 128,
 	.reg_offset = 8,
 };
 
 static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = {
 	.use_imx_pcm = true,
+	.use_edma = true,
 	.fifo_depth = 64,
 	.reg_offset = 0,
 };
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index b12cb578f6d0..76b15deea80c 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -157,6 +157,7 @@
 
 struct fsl_sai_soc_data {
 	bool use_imx_pcm;
+	bool use_edma;
 	unsigned int fifo_depth;
 	unsigned int reg_offset;
 };
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index f6a7466622ea..fc5d089868df 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -286,6 +286,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	if (rsnd_ssi_is_multi_slave(mod, io))
 		return 0;
 
+	if (rsnd_runtime_is_tdm_split(io))
+		chan = rsnd_io_converted_chan(io);
+
+	chan = rsnd_channel_normalization(chan);
+
 	if (ssi->usrcnt > 0) {
 		if (ssi->rate != rate) {
 			dev_err(dev, "SSI parent/child should use same rate\n");
@@ -300,11 +305,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 		return 0;
 	}
 
-	if (rsnd_runtime_is_tdm_split(io))
-		chan = rsnd_io_converted_chan(io);
-
-	chan = rsnd_channel_normalization(chan);
-
 	main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx);
 	if (!main_rate) {
 		dev_err(dev, "unsupported clock rate\n");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 35f48e9c5ead..88978a3036c4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -978,7 +978,7 @@ static void soc_cleanup_component(struct snd_soc_component *component)
 	/* For framework level robustness */
 	snd_soc_component_set_jack(component, NULL, NULL);
 
-	list_del(&component->card_list);
+	list_del_init(&component->card_list);
 	snd_soc_dapm_free(snd_soc_component_get_dapm(component));
 	soc_cleanup_component_debugfs(component);
 	component->card = NULL;
diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
index 87a9b9dd4e98..29f61053ab62 100644
--- a/sound/soc/ti/Kconfig
+++ b/sound/soc/ti/Kconfig
@@ -200,11 +200,18 @@ config SND_SOC_DM365_AIC3X_CODEC
 
 config SND_SOC_DM365_VOICE_CODEC
 	bool "Voice Codec - CQ93VC"
-	select MFD_DAVINCI_VOICECODEC
-	select SND_SOC_CQ0093VC
 	help
 	  Say Y if you want to add support for SoC On-chip voice codec
 endchoice
 
+config SND_SOC_DM365_VOICE_CODEC_MODULE
+	def_tristate y
+	depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC
+	select MFD_DAVINCI_VOICECODEC
+	select SND_SOC_CQ0093VC
+	help
+	  The is an internal symbol needed to ensure that the codec
+	  and MFD driver can be built as loadable modules if necessary.
+
 endmenu
 
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 25faf2d3c639..fbfde996fee7 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1658,6 +1658,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 	case 0x25ce:  /* Mytek devices */
 	case 0x278b:  /* Rotel? */
 	case 0x2ab6:  /* T+A devices */
+	case 0x3842:  /* EVGA */
+	case 0xc502:  /* HiBy devices */
 		if (fp->dsd_raw)
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
 		break;