[alsa-devel] Applied "ASoC: Intel: Atom: add 24-bit support for media playback and capture" to the asoc tree

Mark Brown broonie at kernel.org
Sat Dec 19 12:51:29 CET 2015


The patch

   ASoC: Intel: Atom: add 24-bit support for media playback and capture

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 098c2cd2814098b6cf98ab8c068d69eefbc46716 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>
Date: Thu, 17 Dec 2015 20:35:46 -0600
Subject: [PATCH] ASoC: Intel: Atom: add 24-bit support for media playback and
 capture

DSP firmware supports 24-bit data, expose functionality to
userspace/apps.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>
Signed-off-by: Mark Brown <broonie at kernel.org>
---
 sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 60b73b7eed0f..c1f618ed183b 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -501,14 +501,14 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
 		.channels_min = SST_STEREO,
 		.channels_max = SST_STEREO,
 		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
 	},
 	.capture = {
 		.stream_name = "Headset Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
 	},
 },
 {
@@ -519,7 +519,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
 		.channels_min = SST_STEREO,
 		.channels_max = SST_STEREO,
 		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
 	},
 },
 {
-- 
2.6.2



More information about the Alsa-devel mailing list