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authorLinus Torvalds <torvalds@linux-foundation.org>2018-10-25 18:00:15 +0200
committerLinus Torvalds <torvalds@linux-foundation.org>2018-10-25 18:00:15 +0200
commit3acbd2de6bc3af215c6ed7732dfc097d1e238503 (patch)
tree5152e90a4d2d586dd6ad1cf0b8f28c4de2e46e66 /sound/soc/intel/skylake/skl-pcm.c
parentMerge tag 'scsi-misc' of git://git.kernel.org/pub/scm/linux/kernel/git/jejb/scsi (diff)
parentMerge tag 'asoc-v5.0-2' of https://git.kernel.org/pub/scm/linux/kernel/git/br... (diff)
downloadlinux-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.c71
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,
},
},
};