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| author | Linus Torvalds <torvalds@linux-foundation.org> | 2020-04-24 20:27:43 +0300 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2020-04-24 20:27:43 +0300 |
| commit | b4ecf26ea2ed744715753ae11e6928fbda9b65ad (patch) | |
| tree | 3084d8cb71f073deeb4ee03f52c82ce4298e2ac2 /include | |
| parent | 88412a4e00f6baab2752e99ffdbdb0ee661cac30 (diff) | |
| parent | 8d6762af302d69f76fa788a277a56a9d9cd275d5 (diff) | |
| download | linux-b4ecf26ea2ed744715753ae11e6928fbda9b65ad.tar.xz | |
Merge tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This became a slightly big pull request, as the accumulated ASoC fixes
are included here. Some highlights:
- Revert of ASoC DAI startup changes that caused regression on some
x86 platforms
- Regression fix in HD-audio power management and driver blacklist
- A collection of ASoC DAPM and topology fixes
- Continued USB-audio fixes and quirks
- Lots of small device-specific fixes
- Rockchip S/PDIF DT stuff update for validation issues"
* tag 'sound-5.7-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (51 commits)
ALSA: hda: Always use jackpoll helper for jack update after resume
ALSA: hda/realtek - Add new codec supported for ALC245
ALSA: usb-audio: Fix usb audio refcnt leak when getting spdif
ALSA: usb-audio: Add connector notifier delegation
ALSA: usb-audio: Apply async workaround for Scarlett 2i4 2nd gen
ASoC: wm8960: Fix wrong clock after suspend & resume
ALSA: usx2y: Fix potential NULL dereference
ALSA: usb-audio: Add quirk for Focusrite Scarlett 2i2
ASoC: wm89xx: Add missing dependency
ASoC: dapm: fixup dapm kcontrol widget
ASoC: rsnd: Fix "status check failed" spam for multi-SSI
ASoC: rsnd: Don't treat master SSI in multi SSI setup as parent
ASoC: meson: gx-card: fix codec-to-codec link setup
ASoC: meson: axg-card: fix codec-to-codec link setup
ALSA: usb-audio: Add static mapping table for ALC1220-VB-based mobos
ALSA: hda: Remove ASUS ROG Zenith from the blacklist
ALSA: hda/realtek - Fix unexpected init_amp override
ALSA: usb-audio: Filter out unsupported sample rates on Focusrite devices
ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
ASoC: stm32: sai: fix sai probe
...
Diffstat (limited to 'include')
| -rw-r--r-- | include/sound/soc-dai.h | 1 | ||||
| -rw-r--r-- | include/sound/soc.h | 3 |
2 files changed, 3 insertions, 1 deletions
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index d4825b82c7a3..b33abe93b905 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -351,7 +351,6 @@ struct snd_soc_dai { /* bit field */ unsigned int probed:1; - unsigned int started[SNDRV_PCM_STREAM_LAST + 1]; }; static inline struct snd_soc_pcm_stream * diff --git a/include/sound/soc.h b/include/sound/soc.h index 13458e4fbb13..946f88a6c63d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -790,6 +790,9 @@ struct snd_soc_dai_link { const struct snd_soc_pcm_stream *params; unsigned int num_params; + struct snd_soc_dapm_widget *playback_widget; + struct snd_soc_dapm_widget *capture_widget; + unsigned int dai_fmt; /* format to set on init */ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ |