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authorLinus Torvalds <torvalds@linux-foundation.org>2011-10-28 14:25:01 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2011-10-28 14:25:01 -0700
commit68d99b2c8efcb6ed3807a55569300c53b5f88be5 (patch)
treef189c8f2132d3668a2f0e503f5c3f8695b26a1c8 /sound/soc/davinci/davinci-mcasp.c
parent0e59e7e7feb5a12938fbf9135147eeda3238c6c4 (diff)
parent8128c9f21509f9a8b6da94ac432d845dda458406 (diff)
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits) ALSA: hda - Fix ADC input-amp handling for Cx20549 codec ALSA: hda - Keep EAPD turned on for old Conexant chips ALSA: hda/realtek - Fix missing volume controls with ALC260 ASoC: wm8940: Properly set codec->dapm.bias_level ALSA: hda - Fix pin-config for ASUS W90V ALSA: hda - Fix surround/CLFE headphone and speaker pins order ALSA: hda - Fix typo ALSA: Update the sound git tree URL ALSA: HDA: Add new revision for ALC662 ASoC: max98095: Convert codec->hw_write to snd_soc_write ASoC: keep pointer to resource so it can be freed ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2 ASoC: da7210: Add support for line out and DAC ASoC: da7210: Add support for DAPM ALSA: hda/realtek - Fix DAC assignments of multiple speakers ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value ASoC: Set sgtl5000->ldo in ldo_regulator_register ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture ...
Diffstat (limited to 'sound/soc/davinci/davinci-mcasp.c')
-rw-r--r--sound/soc/davinci/davinci-mcasp.c20
1 files changed, 13 insertions, 7 deletions
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 8566238db2a..7173df254a9 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -732,16 +732,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
davinci_hw_param(dev, substream->stream);
switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_U8:
case SNDRV_PCM_FORMAT_S8:
dma_params->data_type = 1;
word_length = DAVINCI_AUDIO_WORD_8;
break;
+ case SNDRV_PCM_FORMAT_U16_LE:
case SNDRV_PCM_FORMAT_S16_LE:
dma_params->data_type = 2;
word_length = DAVINCI_AUDIO_WORD_16;
break;
+ case SNDRV_PCM_FORMAT_U32_LE:
case SNDRV_PCM_FORMAT_S32_LE:
dma_params->data_type = 4;
word_length = DAVINCI_AUDIO_WORD_32;
@@ -818,6 +821,13 @@ static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
};
+#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | \
+ SNDRV_PCM_FMTBIT_U32_LE)
+
static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
{
.name = "davinci-mcasp.0",
@@ -825,17 +835,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.channels_min = 2,
.channels_max = 2,
.rates = DAVINCI_MCASP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
+ .formats = DAVINCI_MCASP_PCM_FMTS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
.rates = DAVINCI_MCASP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S8 |
- SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
+ .formats = DAVINCI_MCASP_PCM_FMTS,
},
.ops = &davinci_mcasp_dai_ops,
@@ -846,7 +852,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.channels_min = 1,
.channels_max = 384,
.rates = DAVINCI_MCASP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = DAVINCI_MCASP_PCM_FMTS,
},
.ops = &davinci_mcasp_dai_ops,
},