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?
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,
};
No config file is used for the working version. For the dmix version the etc/asound.conf file is:
pcm.chemamix {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
rate 48000
}
}
pcm.!default {
type plug
slave.pcm chemamix
}
Thanks,
Daniel
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