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author | Linus Torvalds <torvalds@linux-foundation.org> | 2010-04-07 19:42:25 +0400 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2010-04-07 19:42:25 +0400 |
commit | 84db18bbeb5c9c1a9c86e38a89d76ee526fd2c6f (patch) | |
tree | 49d3959eb24cd7c0754ed50e05fb96b0fb8d04aa /include | |
parent | 6948ec70355ae6cf6082519e3d76b280373dade1 (diff) | |
parent | 55b371d4ac5ed6f3338a398fbf9f2eb9ace78799 (diff) | |
download | linux-84db18bbeb5c9c1a9c86e38a89d76ee526fd2c6f.tar.xz |
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.h | 2 | ||||
-rw-r--r-- | include/sound/soc-dai.h | 18 | ||||
-rw-r--r-- | include/sound/soc.h | 1 |
3 files changed, 19 insertions, 2 deletions
diff --git a/include/sound/ak4113.h b/include/sound/ak4113.h index 8988edae1609..2609048c1d44 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 061f16d4c878..0a0b019d41ad 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 5d234a8c2506..a57fbfcd4c8f 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 */ |