At Thu, 8 Feb 2007 10:44:10 -0800,
<daniel.car@xxxxxxx> wrote:
>
> Hi,
>
> I'm working on an unannounced SOC audio driver on an ARM system. I have a problem were audio works fine with out dmix, but when I try to use dmix no audio data gets to the driver. The dmix plug-in does work when attaching external usb audio device. Trying to use dmix so that more then one application can generate audio simultaneously.
>
> If I try with version 1.0.11 of ALSA the audio driver doesn't receive any data after the snd_pcm_period_elapsed() function call. If I try with version 1.0.13 of ALSA, aplay returns an error saying:
> # aplay -D hw:0,0 pcm48khz_2ch_stereo.wav open_stream success
> Playing WAVE 'pcm48khz_2ch_stereo.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
> aplay: set_params:882: Broken configuration for this PCM: no configurations available
>
> I'm planning on building a debug version of 1.0.14rc2 and tracing thru the code to see why aplay fails. Is there a better way to debug this? Are there any tricks for building ALSA with debug info?
So, actually it's a problem of driver / alsa-lib hw layer rather than
dmix? You can get more verbose messages by setting LIBASOUND_DEBUG
variable. See alsa-lib/NOTES for details.
> Here is how the driver is set up:
> static snd_pcm_hardware_t snd_oloriver_playback_hw = {
> .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
> SNDRV_PCM_INFO_MMAP_VALID),
> .formats = SNDRV_PCM_FMTBIT_S16_LE,
> .rates = SNDRV_PCM_RATE_48000,
> .rate_min = 48000,
> .rate_max = 48000,
> .channels_min = 2,
> .channels_max = 2,
> .buffer_bytes_max = 65536,
> .period_bytes_min = 32768,
> .period_bytes_max = 32768,
> .periods_min = 2,
> .periods_max = 2,
My rough guess is that the buffer and period size constraints are too
restrictive. You can try to pass the exact period/buffer sizes to
aplay via --period-size=8192 --buffer-size=16384 options.
Takashi
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