Merge tag 'sound-5.0-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Only three fixes.

  The fix for Realtek HD-audio looks lengthy, but it's just a code
  shuffling, and the actual changes are fairly small.

  The rest are a PCM core fix for a long-standing bug that was recently
  scratched by syzkaller, and a trivial USB-audio quirk for DSD support"

* tag 'sound-5.0-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/realtek - Fixed hp_pin no value
  ALSA: pcm: Fix tight loop of OSS capture stream
  ALSA: usb-audio: Add Opus #3 to quirks for native DSD support

3 files changed, 54 insertions, 34 deletions

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 40013b26f671..6c99fa8ac5fa 100644
--- a/sound/core/pcm_lib.c
+++ b/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;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b4f472157ebd..4139aced63f8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -117,6 +117,7 @@ struct alc_spec {
 	int codec_variant;	/* flag for other variants */
 	unsigned int has_alc5505_dsp:1;
 	unsigned int no_depop_delay:1;
+	unsigned int done_hp_init:1;
 
 	/* for PLL fix */
 	hda_nid_t pll_nid;
@@ -3372,6 +3373,48 @@ static void alc_default_shutup(struct hda_codec *codec)
 	snd_hda_shutup_pins(codec);
 }
 
+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 void alc294_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (!spec->done_hp_init) {
+		alc294_hp_init(codec);
+		spec->done_hp_init = true;
+	}
+	alc_default_init(codec);
+}
+
 static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg,
 			     unsigned int val)
 {
@@ -7373,37 +7416,6 @@ 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)
@@ -7529,7 +7541,7 @@ 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);
+		spec->init_hook = alc294_init;
 		break;
 	case 0x10ec0300:
 		spec->codec_variant = ALC269_TYPE_ALC300;
@@ -7541,7 +7553,7 @@ 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);
+		spec->init_hook = alc294_init;
 		break;
 
 	}
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ebbadb3a7094..bb8372833fc2 100644
--- a/sound/usb/quirks.c
+++ b/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(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 */