How does struct snd_pcm_hardware relate to the actual hardware

How does struct snd_pcm_hardware relate to the actual hardware

[Adam Rosenberg]

I am using DMA to transfer data to SPORT1 on my Blackfin BF537.  I am
using a CS42448 CODEC that has 8 DACs.  I am using TDM mode and have
set the following:

DMA Buffer Size For Each SPORT Data Line (Rx or Tx) = 131072 bytes
DMA Periods = 8

static struct snd_pcm_hardware snd_pcm_hw =
{
	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_RESUME),
	.formats = SNDRV_PCM_FMTBIT_S32_LE,
	.rates = SNDRV_PCM_RATE_48000,
	.rate_min = 48000,
	.rate_max = 48000,
	.buffer_bytes_max = 32768,
	.channels_min = 2,
	.channels_max = 2,
	.period_bytes_min = 8,
	.period_bytes_max = 4096,
	.periods_min = 8,
	.periods_max = 8,
};

Everytime I receive a DMA interrupt from the SPORT driver I call
snd_pcm_period_elapsed for all of the substreams that are currently
active.  In the PCM Copy callback I calculate the position within the
DMA buffer to write to based on the TDM slot for the channel and the
position passed into the function.

The audio is not being placed into the DMA buffer properly.  The audio
plays faster than it should and has a constant stutter.  I have read
the "writing an ALSA driver" guide multiple times but it does not
explain the relationship between struct snd_pcm_hardware,
snd_pcm_period_elapsed, and the copy callback clearly enough for me to
understand what I am doing wrong.

Please explain how these interact in more detail to help me fix this
problem.  I can also send the full source code to the CS42448 driver I
am working on if more details are needed.

Thank you,
Adam

Re: How does struct snd_pcm_hardware relate to the actual hardware

[Mark Brown]

On Wed, Jun 30, 2010 at 05:32:41PM -0400, Adam Rosenberg wrote:

> The audio is not being placed into the DMA buffer properly.  The audio
> plays faster than it should and has a constant stutter.  I have read
> the "writing an ALSA driver" guide multiple times but it does not
> explain the relationship between struct snd_pcm_hardware,
> snd_pcm_period_elapsed, and the copy callback clearly enough for me to
> understand what I am doing wrong.

You probably want to provide a little more detail on how you're parsing
the data in your DMA driver here...  The simplest thing when debugging
problems like this is often to place obviously tracable test data into
the audio stream (eg, all but one channel zero) and then work through
where the data ends up compared to where it ought to end up.

Note also that ASoC doesn't do anything explicit to support substreams,
which aren't in themselves ASoC specific.

Re: How does struct snd_pcm_hardware relate to the actual hardware

[Adam Rosenberg]

I determined that the secondary data lines of my SPORT were not being
enabled properly.  With that problem solved I am finally seeing test
data sent to the DACs correctly.  I am still having problems with
placing the data correctly using my custom PCM copy operation.

I do not understand the relationship between snd_pcm_period_elapsed
and the PCM copy and trigger functions.  Where can I find a detailed
description of how this works?

I am attempting to troubleshoot this problem but it seems to change
based on values I enter into struct snd_pcm_hardware, which I copy to
substream->runtime->hw when the PCM open operation occurs.  What are
the correct values for struct snd_pcm_hardware?

I have included the values I am currently using and a brief
description of the hardware configuration below:

I am using one Blackfin BF537 SPORT to transfer 16 channels of audio
to 2 CS42448 CODECs using TDM.  There are 8 channels per TDM frame.
The Primary SPORT data lines are connected to one CS42448 and the
Secondary are connected to another.  I use one DMA buffer of 131072
bytes and configure the DMA for 8 periods. I have to interleave the
channel data so that each CODEC receives one TDM frame for every frame
sync clock.  With 8 channels in each TDM frame, there are 256 bits per
frame sync clock.  The frame sync clock and bit clock are generated
from an ICS661 audio clock source configured for 12.288Mhz (256bits *
48Khz).

static struct snd_pcm_hardware ourCS42448PcmHw =
{
	.info = (SNDRV_PCM_INFO_INTERLEAVED |
                   SNDRV_PCM_INFO_BLOCK_TRANSFER |
                   SNDRV_PCM_INFO_RESUME),

	.formats = SNDRV_PCM_FMTBIT_S32_LE,

	.rates = SNDRV_PCM_RATE_48000,
	.rate_min = 48000,
	.rate_max = 48000,

	.buffer_bytes_max = 32768,

	.channels_min = 2,
	.channels_max = 2,

	.period_bytes_min = 4096,
	.period_bytes_max = 4096,

	.periods_min = 8,
	.periods_max = 8,
};

Thanks,
Adam

Adam Rosenberg
Software Engineer

Alcorn McBride Inc.
3300 South Hiawassee
Building 105
Orlando, FL 32835

(407) 296 - 5800 ext. 5490