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authorDaniel Mack <daniel@caiaq.de>2010-03-22 10:11:15 +0100
committerMark Brown <broonie@opensource.wolfsonmicro.com>2010-04-05 19:14:11 +0100
commit5f712b2b73a9fc87fcc52124cfe8adefaa0c92f5 (patch)
tree0e7ab3cedba6b50cdf603c433b79ceebf23972b0 /include/sound
parentd522ffbfb9fccf6eca283cd2e8b03cf3d21fb616 (diff)
downloadop-kernel-dev-5f712b2b73a9fc87fcc52124cfe8adefaa0c92f5.zip
op-kernel-dev-5f712b2b73a9fc87fcc52124cfe8adefaa0c92f5.tar.gz
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
This fixes a memory corruption when ASoC devices are used in full-duplex mode. Specifically for pxa-ssp code, where this pointer is dynamically allocated for each direction and destroyed upon each stream start. All other platforms are fixed blindly, I couldn't even compile-test them. Sorry for any breakage I may have caused. [Note that this is a backported version for 2.6.34. Upstream commit is fd23b7dee] Signed-off-by: Daniel Mack <daniel@caiaq.de> Reported-by: Sven Neumann <s.neumann@raumfeld.com> Reported-by: Michael Hirsch <m.hirsch@raumfeld.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'include/sound')
-rw-r--r--include/sound/soc-dai.h18
-rw-r--r--include/sound/soc.h1
2 files changed, 18 insertions, 1 deletions
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 061f16d..0a0b019 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 5d234a8..a57fbfc 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 */
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