Linus,
please pull sound fixes for v5.0-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
tags/sound-5.0-rc5
The topmost commit is 693abe11aa6b27aed6eb8222162f8fb986325cef
----------------------------------------------------------------
sound fixes for 5.0-rc5
Only three fixes: a 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.
----------------------------------------------------------------
Kailang Yang (1):
ALSA: hda/realtek - Fixed hp_pin no value
Olek Poplavsky (1):
ALSA: usb-audio: Add Opus #3 to quirks for native DSD support
Takashi Iwai (1):
ALSA: pcm: Fix tight loop of OSS capture stream
---
sound/core/pcm_lib.c | 9 ++++-
sound/pci/hda/patch_realtek.c | 78 +++++++++++++++++++++++++------------------
sound/usb/quirks.c | 1 +
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 */
