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commit 83f4997a01e72844045a2bf1f348359a1acbfb19
parent af0c9af1b3f66052c369d08be3f60fa9a9559e48
Author: Linus Torvalds <torvalds@linux-foundation.org>
Date:   Thu, 31 Jan 2019 10:00:00 -0800

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

Diffstat:
Msound/core/pcm_lib.c | 9++++++++-
Msound/pci/hda/patch_realtek.c | 78+++++++++++++++++++++++++++++++++++++++++++++---------------------------------
Msound/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 @@ -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 @@ -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 @@ -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 */