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author | Linus Torvalds <torvalds@linux-foundation.org> | 2018-10-25 18:00:15 +0200 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2018-10-25 18:00:15 +0200 |
commit | 3acbd2de6bc3af215c6ed7732dfc097d1e238503 (patch) | |
tree | 5152e90a4d2d586dd6ad1cf0b8f28c4de2e46e66 /sound/soc/intel/skylake/skl-pcm.c | |
parent | Merge tag 'scsi-misc' of git://git.kernel.org/pub/scm/linux/kernel/git/jejb/scsi (diff) | |
parent | Merge tag 'asoc-v5.0-2' of https://git.kernel.org/pub/scm/linux/kernel/git/br... (diff) | |
download | linux-3acbd2de6bc3af215c6ed7732dfc097d1e238503.tar.xz linux-3acbd2de6bc3af215c6ed7732dfc097d1e238503.zip |
Merge tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There have been little changes in ALSA core stuff, but ASoC core still
kept rolling for the continued restructuring. The rest are lots of
small driver-specific changes and some minor API updates. Here are
highlights:
General:
- Appropriate fall-through annotations everywhere
- Some code cleanup in memalloc code, handling non-cacahed pages more
commonly in the helper
- Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently
Drivers:
- More HD-audio CA0132 codec improvement for supporting other Creative
boards
- Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
give move support of existing HD-audio devices with DSP
- A few device-specific HD-audio quirks as usual
- New quirk for RME CC devices and correction for B&W PX for USB-audio
- FireWire: code refactoring including devres usages
ASoC Core:
- Continued componentization works; it's almost done!
- A bunch of new for_each_foo macros
- Cleanups and fixes in DAPM code
ASoC Drivers:
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060"
* tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
ASoC: stm32: sai: fix master clock naming
ASoC: stm32: add clock dependency for sai
ALSA: hda/ca0132 - Actually fix microphone issue
ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
ASoC: wm2000: Remove wm2000_read helper function
ASoC: cs42l51: fix mclk support
ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate
ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
ASoC: cs42l51: add mclk support
ASoC: stm32: sai: set sai as mclk clock provider
ASoC: dt-bindings: add mclk support to cs42l51
ASoC: dt-bindings: add mclk provider support to stm32 sai
ASoC: soc-core: fix trivial checkpatch issues
ASoC: dapm: Add support for hw_free on CODEC to CODEC links
ASoC: Intel: kbl_da7219_max98927: minor white space clean up
ALSA: i2c/cs8427: Fix int to char conversion
ALSA: doc: Brush up the old writing-an-alsa-driver
ASoC: rsnd: tidyup SSICR::SWSP for TDM
ASoC: rsnd: enable TDM settings for SSI parent
ASoC: pcm3168a: add hw constraint for capture channel
...
Diffstat (limited to 'sound/soc/intel/skylake/skl-pcm.c')
-rw-r--r-- | sound/soc/intel/skylake/skl-pcm.c | 71 |
1 files changed, 59 insertions, 12 deletions
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..557f80c0bfe5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -32,6 +32,7 @@ #define HDA_MONO 1 #define HDA_STEREO 2 #define HDA_QUAD 4 +#define HDA_MAX 8 static const struct snd_pcm_hardware azx_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | @@ -494,6 +495,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -569,7 +571,10 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, stream_tag = hdac_stream(link_dev)->stream_tag; /* set the stream tag in the codec dai dma params */ - snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + else + snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0); p_params.s_fmt = snd_pcm_format_width(params_format(params)); p_params.ch = params_channels(params); @@ -995,21 +1000,63 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }, { - .name = "HD-Codec Pin", + .name = "Analog CPU DAI", .ops = &skl_link_dai_ops, .playback = { - .stream_name = "HD-Codec Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .stream_name = "Analog CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, }, .capture = { - .stream_name = "HD-Codec Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .stream_name = "Analog CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, +}, +{ + .name = "Alt Analog CPU DAI", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "Alt Analog CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Alt Analog CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, +}, +{ + .name = "Digital CPU DAI", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "Digital CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Digital CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, }, }, }; |