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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 /include
parent6948ec70355ae6cf6082519e3d76b280373dade1 (diff)
parent55b371d4ac5ed6f3338a398fbf9f2eb9ace78799 (diff)
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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 'include')
-rw-r--r--include/sound/ak4113.h2
-rw-r--r--include/sound/soc-dai.h18
-rw-r--r--include/sound/soc.h1
3 files changed, 19 insertions, 2 deletions
diff --git a/include/sound/ak4113.h b/include/sound/ak4113.h
index 8988edae160..2609048c1d4 100644
--- a/include/sound/ak4113.h
+++ b/include/sound/ak4113.h
@@ -307,7 +307,7 @@ struct ak4113 {
int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
ak4113_write_t *write,
- const unsigned char pgm[AK4113_WRITABLE_REGS],
+ const unsigned char *pgm,
void *private_data, struct ak4113 **r_ak4113);
void snd_ak4113_reg_write(struct ak4113 *ak4113, unsigned char reg,
unsigned char mask, unsigned char val);
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 061f16d4c87..0a0b019d41a 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -219,7 +219,6 @@ struct snd_soc_dai {
struct snd_soc_codec *codec;
unsigned int active;
unsigned char pop_wait:1;
- void *dma_data;
/* DAI private data */
void *private_data;
@@ -230,4 +229,21 @@ struct snd_soc_dai {
struct list_head list;
};
+static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
+ const struct snd_pcm_substream *ss)
+{
+ return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ dai->playback.dma_data : dai->capture.dma_data;
+}
+
+static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
+ const struct snd_pcm_substream *ss,
+ void *data)
+{
+ if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->playback.dma_data = data;
+ else
+ dai->capture.dma_data = data;
+}
+
#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5d234a8c250..a57fbfcd4c8 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -375,6 +375,7 @@ struct snd_soc_pcm_stream {
unsigned int channels_min; /* min channels */
unsigned int channels_max; /* max channels */
unsigned int active:1; /* stream is in use */
+ void *dma_data; /* used by platform code */
};
/* SoC audio ops */