[alsa-devel] [PATCH] ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
Daniel Mack
daniel at caiaq.de
Thu Mar 18 19:08:59 CET 2010
This fixes a memory corruption when using 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.
Signed-off-by: Daniel Mack <daniel at caiaq.de>
Cc: Liam Girdwood <lrg at slimlogic.co.uk>
Cc: Mark Brown <broonie at opensource.wolfsonmicro.com>
Cc: Sven Neumann <s.neumann at raumfeld.com>
Cc: Michael Hirsch <m.hirsch at raumfeld.com>
Reported-by: Sven Neumann <s.neumann at raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch at raumfeld.com>
---
include/sound/soc-dai.h | 18 +++++++++++++++++-
include/sound/soc.h | 1 +
sound/soc/atmel/atmel-pcm.c | 2 +-
sound/soc/atmel/atmel_ssc_dai.c | 6 +++---
sound/soc/davinci/davinci-i2s.c | 3 ++-
sound/soc/davinci/davinci-mcasp.c | 3 ++-
sound/soc/davinci/davinci-pcm.c | 4 +++-
sound/soc/omap/omap-mcbsp.c | 4 +++-
sound/soc/omap/omap-pcm.c | 4 +++-
sound/soc/pxa/pxa-ssp.c | 23 ++++++++++++-----------
sound/soc/pxa/pxa2xx-pcm.c | 4 +++-
11 files changed, 50 insertions(+), 22 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 6cf76a4..377693a 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -224,7 +224,6 @@ 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;
@@ -235,4 +234,21 @@ 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 dbfec16..b8bac6a 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -376,6 +376,7 @@ struct snd_soc_pcm_stream {
unsigned int channels_min; /* min channels */
unsigned int channels_max; /* max channels */
unsigned int active; /* num of active users of the stream */
+ void *dma_data; /* used by platform code */
};
/* SoC audio ops */
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index fdb2553..f6b3cc0 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -180,7 +180,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = params_buffer_bytes(params);
- prtd->params = rtd->dai->cpu_dai->dma_data;
+ prtd->params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
prtd->dma_buffer = runtime->dma_addr;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index e588e63..0b59806 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -363,12 +363,12 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
ssc_p->dma_params[dir] = dma_params;
/*
- * The cpu_dai->dma_data field is only used to communicate the
- * appropriate DMA parameters to the pcm driver hw_params()
+ * The snd_soc_pcm_stream->dma_data field is only used to communicate
+ * the appropriate DMA parameters to the pcm driver hw_params()
* function. It should not be used for other purposes
* as it is common to all substreams.
*/
- rtd->dai->cpu_dai->dma_data = dma_params;
+ snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params);
channels = params_channels(params);
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 6362ca0..4aad7ec 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -585,7 +585,8 @@ static int davinci_i2s_probe(struct platform_device *pdev)
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
davinci_i2s_dai.private_data = dev;
- davinci_i2s_dai.dma_data = dev->dma_params;
+ davinci_i2s_dai.capture.dma_data = dev->dma_params;
+ davinci_i2s_dai.playback.dma_data = dev->dma_params;
ret = snd_soc_register_dai(&davinci_i2s_dai);
if (ret != 0)
goto err_free_mem;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index ab6518d..c056bfb 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -917,7 +917,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dma_data->channel = res->start;
davinci_mcasp_dai[pdata->op_mode].private_data = dev;
- davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params;
+ davinci_mcasp_dai[pdata->op_mode].capture.dma_data = dev->dma_params;
+ davinci_mcasp_dai[pdata->op_mode].playback.dma_data = dev->dma_params;
davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 80c7fdf..2dc406f 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -649,8 +649,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
struct snd_pcm_hardware *ppcm;
int ret = 0;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data;
+ struct davinci_pcm_dma_params *pa;
struct davinci_pcm_dma_params *params;
+
+ pa = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
if (!pa)
return -ENODEV;
params = &pa[substream->stream];
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 2952fb0..e21a3e5 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -322,7 +322,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
OMAP_DMA_DATA_TYPE_S16;
- cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream,
+ &omap_mcbsp_dai_dma_params[id][substream->stream]);
if (mcbsp_data->configured) {
/* McBSP already configured by another stream */
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 825db38..39538c0 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -100,9 +100,11 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct omap_runtime_data *prtd = runtime->private_data;
- struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
+ struct omap_pcm_dma_data *dma_data;
int err = 0;
+ dma_data = 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_data)
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 9e95e51..6959c51 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -121,10 +121,9 @@ 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;
}
@@ -141,10 +140,8 @@ 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
@@ -569,19 +566,23 @@ 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-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index d38e395..adc7e6f 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)
--
1.6.6.2
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