Merge tag 'sound-fix-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Here are a few fixes that have been pending since the previous pull
  request: a regression fix for HD-audio multiple SPDIF / HDMI devices,
  several ALC256 codec fixes, a couple of i915 HDMI audio fixes, and
  various small fixes.

  Nothing exciting, just boring, but things good to have"

* tag 'sound-fix-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - fix headset mic detection problem for one more machine
  ALSA: hda/realtek - Fix Headphone Mic doesn't recording for ALC256
  ALSA: hda - fix "num_steps = 0" error on ALC256
  ALSA: usb-audio: Fix audio output on Roland SC-D70 sound module
  ALSA: hda - add AZX_DCAPS_I915_POWERWELL to Baytrail
  ALSA: hda - only sync BCLK to the display clock for Haswell & Broadwell
  ALSA: hda - Mute headphone pin on suspend on XPS13 9333
  sound/oss: fix deadlock in sequencer_ioctl(SNDCTL_SEQ_OUTOFBAND)
  ALSA: asound.h - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN
  ALSA: hda - potential (but unlikely) uninitialized variable
  ALSA: hda - Fix regression for slave SPDIF setups
  ALSA: intel8x0: Check pci_iomap() success for DEVICE_ALI
  ALSA: hda - simplify azx_has_pm_runtime

11 files changed, 51 insertions, 68 deletions

diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 46145a5277fe..a45be6bdcf5b 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -864,7 +864,7 @@ struct snd_ctl_elem_id {
 	snd_ctl_elem_iface_t iface;	/* interface identifier */
 	unsigned int device;		/* device/client number */
 	unsigned int subdevice;		/* subdevice (substream) number */
-	unsigned char name[44];		/* ASCII name of item */
+	unsigned char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];		/* ASCII name of item */
 	unsigned int index;		/* index of item */
 };
 
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index c0eea1dfe90f..f19da4b47c1d 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -681,13 +681,8 @@ static int seq_timing_event(unsigned char *event_rec)
 			break;
 
 		case TMR_ECHO:
-			if (seq_mode == SEQ_2)
-				seq_copy_to_input(event_rec, 8);
-			else
-			{
-				parm = (parm << 8 | SEQ_ECHO);
-				seq_copy_to_input((unsigned char *) &parm, 4);
-			}
+			parm = (parm << 8 | SEQ_ECHO);
+			seq_copy_to_input((unsigned char *) &parm, 4);
 			break;
 
 		default:;
@@ -1324,7 +1319,6 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a
 	int mode = translate_mode(file);
 	struct synth_info inf;
 	struct seq_event_rec event_rec;
-	unsigned long flags;
 	int __user *p = arg;
 
 	orig_dev = dev = dev >> 4;
@@ -1479,9 +1473,7 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a
 		case SNDCTL_SEQ_OUTOFBAND:
 			if (copy_from_user(&event_rec, arg, sizeof(event_rec)))
 				return -EFAULT;
-			spin_lock_irqsave(&lock,flags);
 			play_event(event_rec.arr);
-			spin_unlock_irqrestore(&lock,flags);
 			return 0;
 
 		case SNDCTL_MIDI_INFO:
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e70a7fb393dd..873ed1bce12b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2529,7 +2529,7 @@ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
 	if (!d)
 		return;
 	for (; *d; d++)
-		snd_hdac_regmap_update(&codec->core, nid,
+		snd_hdac_regmap_update(&codec->core, *d,
 				       AC_VERB_SET_DIGI_CONVERT_1, mask, val);
 }
 
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index be1b7ded8d82..0efdb094d21c 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -404,7 +404,7 @@ struct azx {
 	((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg))
 
 #define azx_has_pm_runtime(chip) \
-	(!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME))
+	((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)
 
 /* PCM setup */
 static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream)
diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c
index 52a85d87c23c..3052a2b095f7 100644
--- a/sound/pci/hda/hda_i915.c
+++ b/sound/pci/hda/hda_i915.c
@@ -55,6 +55,12 @@ void haswell_set_bclk(struct hda_intel *hda)
 	int cdclk_freq;
 	unsigned int bclk_m, bclk_n;
 	struct i915_audio_component *acomp = &hda->audio_component;
+	struct pci_dev *pci = hda->chip.pci;
+
+	/* Only Haswell/Broadwell need set BCLK */
+	if (pci->device != 0x0a0c && pci->device != 0x0c0c
+	   && pci->device != 0x0d0c && pci->device != 0x160c)
+		return;
 
 	if (!acomp->ops)
 		return;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e1c210515581..34040d26c94f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -297,6 +297,9 @@ enum {
 	 AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
 	 AZX_DCAPS_SNOOP_TYPE(SCH))
 
+#define AZX_DCAPS_INTEL_BAYTRAIL \
+	(AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_I915_POWERWELL)
+
 #define AZX_DCAPS_INTEL_BRASWELL \
 	(AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL)
 
@@ -1992,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = {
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM },
 	/* BayTrail */
 	{ PCI_DEVICE(0x8086, 0x0f04),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL },
 	/* Braswell */
 	{ PCI_DEVICE(0x8086, 0x2284),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL },
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ee6230767c64..baaf7ed06875 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -582,8 +582,8 @@ static void print_conn_list(struct snd_info_buffer *buffer,
 
 	/* Get Cache connections info */
 	cache_len = snd_hda_get_conn_list(codec, nid, &list);
-	if (cache_len != conn_len
-			|| memcmp(list, conn, conn_len)) {
+	if (cache_len >= 0 && (cache_len != conn_len ||
+			      memcmp(list, conn, conn_len) != 0)) {
 		snd_iprintf(buffer, "  In-driver Connection: %d\n", cache_len);
 		if (cache_len > 0) {
 			snd_iprintf(buffer, "    ");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b18b9c67b262..06199e4e930f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4176,17 +4176,15 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec,
 	}
 }
 
-static unsigned int alc_power_filter_xps13(struct hda_codec *codec,
-				hda_nid_t nid,
-				unsigned int power_state)
+static void alc_shutup_dell_xps13(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
+	int hp_pin = spec->gen.autocfg.hp_pins[0];
 
-	/* Avoid pop noises when headphones are plugged in */
-	if (spec->gen.hp_jack_present)
-		if (nid == codec->core.afg || nid == 0x02 || nid == 0x15)
-			return AC_PWRST_D0;
-	return snd_hda_gen_path_power_filter(codec, nid, power_state);
+	/* Prevent pop noises when headphones are plugged in */
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+	msleep(20);
 }
 
 static void alc_fixup_dell_xps13(struct hda_codec *codec,
@@ -4197,8 +4195,7 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec,
 		struct hda_input_mux *imux = &spec->gen.input_mux;
 		int i;
 
-		spec->shutup = alc_no_shutup;
-		codec->power_filter = alc_power_filter_xps13;
+		spec->shutup = alc_shutup_dell_xps13;
 
 		/* Make the internal mic the default input source. */
 		for (i = 0; i < imux->num_items; i++) {
@@ -5231,6 +5228,16 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{0x1b, 0x411111f0}, \
 	{0x1e, 0x411111f0}
 
+#define ALC256_STANDARD_PINS \
+	{0x12, 0x90a60140}, \
+	{0x14, 0x90170110}, \
+	{0x19, 0x411111f0}, \
+	{0x1a, 0x411111f0}, \
+	{0x1b, 0x411111f0}, \
+	{0x1d, 0x40700001}, \
+	{0x1e, 0x411111f0}, \
+	{0x21, 0x02211020}
+
 #define ALC282_STANDARD_PINS \
 	{0x14, 0x90170110}, \
 	{0x18, 0x411111f0}, \
@@ -5331,15 +5338,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x1d, 0x40700001},
 		{0x21, 0x02211050}),
 	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
-		{0x12, 0x90a60140},
-		{0x13, 0x40000000},
-		{0x14, 0x90170110},
-		{0x19, 0x411111f0},
-		{0x1a, 0x411111f0},
-		{0x1b, 0x411111f0},
-		{0x1d, 0x40700001},
-		{0x1e, 0x411111f0},
-		{0x21, 0x02211020}),
+		ALC256_STANDARD_PINS,
+		{0x13, 0x40000000}),
+	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+		ALC256_STANDARD_PINS,
+		{0x13, 0x411111f0}),
 	SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
 		{0x12, 0x90a60130},
 		{0x13, 0x40000000},
@@ -5667,6 +5670,8 @@ static int patch_alc269(struct hda_codec *codec)
 		break;
 	case 0x10ec0256:
 		spec->codec_variant = ALC269_TYPE_ALC256;
+		spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */
+		alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
 		break;
 	}
 
@@ -5680,8 +5685,8 @@ static int patch_alc269(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	if (!spec->gen.no_analog && spec->gen.beep_nid)
-		set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+	if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid)
+		set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT);
 
 	codec->patch_ops = alc_patch_ops;
 	codec->patch_ops.stream_pm = snd_hda_gen_stream_pm;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 749069aa6997..b120925223ae 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -3101,13 +3101,13 @@ static int snd_intel8x0_create(struct snd_card *card,
 		chip->bmaddr = pci_iomap(pci, 3, 0);
 	else
 		chip->bmaddr = pci_iomap(pci, 1, 0);
+
+ port_inited:
 	if (!chip->bmaddr) {
 		dev_err(card->dev, "Controller space ioremap problem\n");
 		snd_intel8x0_free(chip);
 		return -EIO;
 	}
-
- port_inited:
 	chip->bdbars_count = bdbars[device_type];
 
 	/* initialize offsets */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 8bcc87cf5667..789d19ec035d 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -79,7 +79,10 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
 		format = 1 << UAC_FORMAT_TYPE_I_PCM;
 	}
 	if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) {
-		if (chip->usb_id == USB_ID(0x0582, 0x0016) /* Edirol SD-90 */ &&
+		if (((chip->usb_id == USB_ID(0x0582, 0x0016)) ||
+		     /* Edirol SD-90 */
+		     (chip->usb_id == USB_ID(0x0582, 0x000c))) &&
+		     /* Roland SC-D70 */
 		    sample_width == 24 && sample_bytes == 2)
 			sample_bytes = 3;
 		else if (sample_width > sample_bytes * 8) {
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 07f984d5f516..2f6d3e9a1bcd 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -816,37 +816,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.data = (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
-				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
-				.data = & (const struct audioformat) {
-					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
-					.channels = 2,
-					.iface = 0,
-					.altsetting = 1,
-					.altset_idx = 1,
-					.attributes = 0,
-					.endpoint = 0x01,
-					.ep_attr = 0x01,
-					.rates = SNDRV_PCM_RATE_CONTINUOUS,
-					.rate_min = 44100,
-					.rate_max = 44100,
-				}
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
 			},
 			{
 				.ifnum = 1,
-				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
-				.data = & (const struct audioformat) {
-					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
-					.channels = 2,
-					.iface = 1,
-					.altsetting = 1,
-					.altset_idx = 1,
-					.attributes = 0,
-					.endpoint = 0x81,
-					.ep_attr = 0x01,
-					.rates = SNDRV_PCM_RATE_CONTINUOUS,
-					.rate_min = 44100,
-					.rate_max = 44100,
-				}
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
 			},
 			{
 				.ifnum = 2,