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

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)