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authorLinus Torvalds <torvalds@linux-foundation.org>2010-04-07 08:42:25 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2010-04-07 08:42:25 -0700
commit84db18bbeb5c9c1a9c86e38a89d76ee526fd2c6f (patch)
tree49d3959eb24cd7c0754ed50e05fb96b0fb8d04aa /sound/soc/pxa
parent6948ec70355ae6cf6082519e3d76b280373dade1 (diff)
parent55b371d4ac5ed6f3338a398fbf9f2eb9ace78799 (diff)
downloadblackbird-op-linux-84db18bbeb5c9c1a9c86e38a89d76ee526fd2c6f.tar.gz
blackbird-op-linux-84db18bbeb5c9c1a9c86e38a89d76ee526fd2c6f.zip
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: mixart: range checking proc file ALSA: hda - Fix a wrong array range check in patch_realtek.c ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream ALSA: hda - Enable amplifiers on Acer Inspire 6530G ASoC: Only do WM8994 bias off transition from standby ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction ASoC: Support second DC servo readback method for wm_hubs ASoC: Avoid wraparound in wm_hubs DC servo correction ALSA: echoaudio - Eliminate use after free ALSA: i2c: cleanup: change parameter to pointer ALSA: hda - Add MSI blacklist for Aopen MZ915-M ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code ALSA: hda - Update document about MSI and interrupts ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981 ALSA: hda - Add missing printk argument in previous patch ASoC: Fix passing platform_data to ac97 bus users and fix a leak ALSA: hda - Fix ADC/MUX assignment of ALC269 codec ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo() ASoC: wm8994: playback => capture
Diffstat (limited to 'sound/soc/pxa')
-rw-r--r--sound/soc/pxa/pxa-ssp.c23
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c17
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c7
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c4
4 files changed, 32 insertions, 19 deletions
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d5fc52d0a3c4..544fd9566f4d 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -122,10 +122,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
ssp_disable(ssp);
}
- if (cpu_dai->dma_data) {
- kfree(cpu_dai->dma_data);
- cpu_dai->dma_data = NULL;
- }
+ kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+ snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+
return ret;
}
@@ -142,10 +141,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
clk_disable(ssp->clk);
}
- if (cpu_dai->dma_data) {
- kfree(cpu_dai->dma_data);
- cpu_dai->dma_data = NULL;
- }
+ kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+ snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
}
#ifdef CONFIG_PM
@@ -570,19 +567,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
u32 sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
+ struct pxa2xx_pcm_dma_params *dma_data;
+
+ dma_data = snd_soc_dai_get_dma_data(dai, substream);
/* generate correct DMA params */
- if (cpu_dai->dma_data)
- kfree(cpu_dai->dma_data);
+ kfree(dma_data);
/* Network mode with one active slot (ttsa == 1) can be used
* to force 16-bit frame width on the wire (for S16_LE), even
* with two channels. Use 16-bit DMA transfers for this case.
*/
- cpu_dai->dma_data = ssp_get_dma_params(ssp,
+ dma_data = ssp_get_dma_params(ssp,
((chn == 2) && (ttsa != 1)) || (width == 32),
substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+ snd_soc_dai_set_dma_data(dai, substream, dma_data);
+
/* we can only change the settings if the port is not in use */
if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
return 0;
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index e9ae7b3a7e00..d314115e3dd7 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -122,11 +122,14 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct pxa2xx_pcm_dma_params *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
+ dma_data = &pxa2xx_ac97_pcm_stereo_out;
else
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in;
+ dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
return 0;
}
@@ -137,11 +140,14 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct pxa2xx_pcm_dma_params *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+ dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
else
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+ dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
return 0;
}
@@ -156,7 +162,8 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
else
- cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
+ snd_soc_dai_set_dma_data(cpu_dai, substream,
+ &pxa2xx_ac97_pcm_mic_mono_in);
return 0;
}
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 6b8f655d1ad8..c1a5275721e4 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -164,6 +164,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct pxa2xx_pcm_dma_params *dma_data;
BUG_ON(IS_ERR(clk_i2s));
clk_enable(clk_i2s);
@@ -171,9 +172,11 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
pxa_i2s_wait();
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
+ dma_data = &pxa2xx_i2s_pcm_stereo_out;
else
- cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in;
+ dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
/* is port used by another stream */
if (!(SACR0 & SACR0_ENB)) {
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index d38e39575f51..adc7e6f15f93 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -25,9 +25,11 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_runtime *runtime = substream->runtime;
struct pxa2xx_runtime_data *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+ struct pxa2xx_pcm_dma_params *dma;
int ret;
+ dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
if (!dma)
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