I'm using ALSA 1.0.0rc2 on snd_intel8x0 and .asoundrc:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
pcm.mix {
type dmix
ipc_key 321456 # any unique value
ipc_key_add_uid true
slave {
pcm "hw:0,0"
periods 0
period_time 0
period_size 1024 # must be power of 2
buffer_size 4096 # ditto
}
bindings {
0 0
1 1
}
}
ctl.mix { type hw card 0 }
pcm.dsp0 { type plug slave.pcm "mix" }
ctl.mixer0 { type hw card 0 }
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~My problem:
when using wavplay with OSS ALSA emulation playing 22050 and 11025 Hz .wav files works OK
but with AOSS lib using pcm.dsp0 defined above I can't play 11025 Hz .wav file:
$ aoss wavplay lburn2.wav Pathname: lburn2.wav Device: /dev/dsp Sampling Rate: 11025 Hz Mode: Mono Samples: 5999 Bits: 8
Unable to set audio sampling rate
but 22050 HZ .wav is played correctly.
So my question is how properly compose .asoundrc file so we could mix all sound sources which use ALSA natively
through -Dmix virtual device and OSS apps routed through AOSS lib? Is this possible? Is this explained somewhere?
Next case is a possibility to create a virtual mixer so ctl.mix virtually sets level of particular source before mixing
it with others sources so playind applications could set volumen of only his source. Is that possible?
I thing that solution should be a snd kernel module which does this jobs (mixing and correcly setting volume level independently
for all apps, using hardware mixing if possible and software if not, DMA mapping etc.) which lays below OSS pcm emulation
module and pcm0p device and above hardware so no configuration changes in all apps are needed.
If it will working as a kernel high priority thread then some latency problems will gone too.
Regards
-- Adam Tla/lka mailto:[EMAIL PROTECTED] ^v^ ^v^ ^v^ Computer Center, Technical University of Gdansk, Poland PGP public key: finger [EMAIL PROTECTED]
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