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ALSA: ASoC: fix SNDCTL_DSP_SYNC support in Freescale 8610 sound drivers 

If an OSS application calls SNDCTL_DSP_SYNC, then ALSA will call the driver's
_hw_params and _prepare functions again.  On the Freescale MPC8610 DMA ASoC
driver, this caused the DMA controller to be unneccessarily re-programmed, and
apparently it doesn't like that.  The DMA will then not operate when
instructed.  This patch relocates much of the DMA programming to
fsl_dma_open(), which is called only once.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is contained in:
Timur Tabi 2008-08-01 14:58:44 -05:00 committed by Takashi Iwai
parent 11589418a1
commit bf9c8c9dde
1 changed files with 131 additions and 118 deletions
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249
sound/soc/fsl/fsl_dma.c
View File

@ -327,14 +327,75 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
* fsl_dma_open: open a new substream.
*
* Each substream has its own DMA buffer.
*
* ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link
* descriptors that ping-pong from one period to the next. For example, if
* there are six periods and two link descriptors, this is how they look
* before playback starts:
*
* The last link descriptor
* ____________ points back to the first
* | |
* V |
* ___ ___ |
* | |->| |->|
* |___| |___|
* | |
* | |
* V V
* _________________________________________
* | | | | | | | The DMA buffer is
* | | | | | | | divided into 6 parts
* |______|______|______|______|______|______|
*
* and here's how they look after the first period is finished playing:
*
* ____________
* | |
* V |
* ___ ___ |
* | |->| |->|
* |___| |___|
* | |
* |______________
* | |
* V V
* _________________________________________
* | | | | | | |
* | | | | | | |
* |______|______|______|______|______|______|
*
* The first link descriptor now points to the third period. The DMA
* controller is currently playing the second period. When it finishes, it
* will jump back to the first descriptor and play the third period.
*
* There are four reasons we do this:
*
* 1. The only way to get the DMA controller to automatically restart the
* transfer when it gets to the end of the buffer is to use chaining
* mode. Basic direct mode doesn't offer that feature.
* 2. We need to receive an interrupt at the end of every period. The DMA
* controller can generate an interrupt at the end of every link transfer
* (aka segment). Making each period into a DMA segment will give us the
* interrupts we need.
* 3. By creating only two link descriptors, regardless of the number of
* periods, we do not need to reallocate the link descriptors if the
* number of periods changes.
* 4. All of the audio data is still stored in a single, contiguous DMA
* buffer, which is what ALSA expects. We're just dividing it into
* contiguous parts, and creating a link descriptor for each one.
*/
static int fsl_dma_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private;
struct ccsr_dma_channel __iomem *dma_channel;
dma_addr_t ld_buf_phys;
u64 temp_link; /* Pointer to next link descriptor */
u32 mr;
unsigned int channel;
int ret = 0;
unsigned int i;
/*
* Reject any DMA buffer whose size is not a multiple of the period
@ -395,135 +456,20 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
runtime->private_data = dma_private;
return 0;
}
/* Program the fixed DMA controller parameters */
/**
* fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors.
*
* ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link
* descriptors that ping-pong from one period to the next. For example, if
* there are six periods and two link descriptors, this is how they look
* before playback starts:
*
* The last link descriptor
* ____________ points back to the first
* | |
* V |
* ___ ___ |
* | |->| |->|
* |___| |___|
* | |
* | |
* V V
* _________________________________________
* | | | | | | | The DMA buffer is
* | | | | | | | divided into 6 parts
* |______|______|______|______|______|______|
*
* and here's how they look after the first period is finished playing:
*
* ____________
* | |
* V |
* ___ ___ |
* | |->| |->|
* |___| |___|
* | |
* |______________
* | |
* V V
* _________________________________________
* | | | | | | |
* | | | | | | |
* |______|______|______|______|______|______|
*
* The first link descriptor now points to the third period. The DMA
* controller is currently playing the second period. When it finishes, it
* will jump back to the first descriptor and play the third period.
*
* There are four reasons we do this:
*
* 1. The only way to get the DMA controller to automatically restart the
* transfer when it gets to the end of the buffer is to use chaining
* mode. Basic direct mode doesn't offer that feature.
* 2. We need to receive an interrupt at the end of every period. The DMA
* controller can generate an interrupt at the end of every link transfer
* (aka segment). Making each period into a DMA segment will give us the
* interrupts we need.
* 3. By creating only two link descriptors, regardless of the number of
* periods, we do not need to reallocate the link descriptors if the
* number of periods changes.
* 4. All of the audio data is still stored in a single, contiguous DMA
* buffer, which is what ALSA expects. We're just dividing it into
* contiguous parts, and creating a link descriptor for each one.
*
* Note that due to a quirk of the SSI's STX register, the target address
* for the DMA operations depends on the sample size. So we don't program
* the dest_addr (for playback -- source_addr for capture) fields in the
* link descriptors here. We do that in fsl_dma_prepare()
*/
static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private = runtime->private_data;
struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
dma_channel = dma_private->dma_channel;
dma_addr_t temp_addr; /* Pointer to next period */
u64 temp_link; /* Pointer to next link descriptor */
u32 mr; /* Temporary variable for MR register */
unsigned int i;
/* Get all the parameters we need */
size_t buffer_size = params_buffer_bytes(hw_params);
size_t period_size = params_period_bytes(hw_params);
/* Initialize our DMA tracking variables */
dma_private->period_size = period_size;
dma_private->num_periods = params_periods(hw_params);
dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
dma_private->dma_buf_next = dma_private->dma_buf_phys +
(NUM_DMA_LINKS * period_size);
if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
dma_private->dma_buf_next = dma_private->dma_buf_phys;
/*
* Initialize each link descriptor.
*
* The actual address in STX0 (destination for playback, source for
* capture) is based on the sample size, but we don't know the sample
* size in this function, so we'll have to adjust that later. See
* comments in fsl_dma_prepare().
*
* The DMA controller does not have a cache, so the CPU does not
* need to tell it to flush its cache. However, the DMA
* controller does need to tell the CPU to flush its cache.
* That's what the SNOOP bit does.
*
* Also, even though the DMA controller supports 36-bit addressing, for
* simplicity we currently support only 32-bit addresses for the audio
* buffer itself.
*/
temp_addr = substream->dma_buffer.addr;
temp_link = dma_private->ld_buf_phys +
sizeof(struct fsl_dma_link_descriptor);
for (i = 0; i < NUM_DMA_LINKS; i++) {
struct fsl_dma_link_descriptor *link = &dma_private->link[i];
link->count = cpu_to_be32(period_size);
link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
link->next = cpu_to_be64(temp_link);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
link->source_addr = cpu_to_be32(temp_addr);
else
link->dest_addr = cpu_to_be32(temp_addr);
temp_addr += period_size;
temp_link += sizeof(struct fsl_dma_link_descriptor);
}
/* The last link descriptor points to the first */
@ -549,7 +495,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
* We want External Master Start and External Master Pause enabled,
* because the SSI is controlling the DMA controller. We want the DMA
* controller to be set up in advance, and then we signal only the SSI
* to start transfering.
* to start transferring.
*
* We want End-Of-Segment Interrupts enabled, because this will generate
* an interrupt at the end of each segment (each link descriptor
@ -573,6 +519,73 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
return 0;
}
/**
* fsl_dma_hw_params: continue initializing the DMA links
*
* This function obtains hardware parameters about the opened stream and
* programs the DMA controller accordingly.
*
* Note that due to a quirk of the SSI's STX register, the target address
* for the DMA operations depends on the sample size. So we don't program
* the dest_addr (for playback -- source_addr for capture) fields in the
* link descriptors here. We do that in fsl_dma_prepare()
*/
static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct fsl_dma_private *dma_private = runtime->private_data;
dma_addr_t temp_addr; /* Pointer to next period */
unsigned int i;
/* Get all the parameters we need */
size_t buffer_size = params_buffer_bytes(hw_params);
size_t period_size = params_period_bytes(hw_params);
/* Initialize our DMA tracking variables */
dma_private->period_size = period_size;
dma_private->num_periods = params_periods(hw_params);
dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
dma_private->dma_buf_next = dma_private->dma_buf_phys +
(NUM_DMA_LINKS * period_size);
if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
dma_private->dma_buf_next = dma_private->dma_buf_phys;
/*
* The actual address in STX0 (destination for playback, source for
* capture) is based on the sample size, but we don't know the sample
* size in this function, so we'll have to adjust that later. See
* comments in fsl_dma_prepare().
*
* The DMA controller does not have a cache, so the CPU does not
* need to tell it to flush its cache. However, the DMA
* controller does need to tell the CPU to flush its cache.
* That's what the SNOOP bit does.
*
* Also, even though the DMA controller supports 36-bit addressing, for
* simplicity we currently support only 32-bit addresses for the audio
* buffer itself.
*/
temp_addr = substream->dma_buffer.addr;
for (i = 0; i < NUM_DMA_LINKS; i++) {
struct fsl_dma_link_descriptor *link = &dma_private->link[i];
link->count = cpu_to_be32(period_size);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
link->source_addr = cpu_to_be32(temp_addr);
else
link->dest_addr = cpu_to_be32(temp_addr);
temp_addr += period_size;
}
return 0;
}
/**
* fsl_dma_prepare - prepare the DMA registers for playback.
*