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Since you're getting 96khz, and have only told dmix to do 96, it's pretty obvious that dmix is being used
no, front-speakers output uses direct playback. actually no software mixing is possible, one output a time.
as of my three lines .asoundrc, they are not working. beside I've found no way to find out if samplerate_best is applied or not.
@mikaID: could be interesting to know how to find out if an audio player does or does not sampling direct playback..
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front-speakers output uses direct playback
What? Are you referring to phonon's mystical involvement in your setup?
Didn't you say it was working before? You saw rate 48 change to 96 in /proc/asound/card0/pcm0p/sub0/hw_params , yes?
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dear ape bro,
yes, I mean, I've wiped .asoundrc out of my hd too and hw_params change according to input source.
this when I use amarok (phonon apps in general), because i've set frontal-speakers as a device. no audio mixing tough in this scenario.
when i use alsa apps (chromium, vlc), dmix is used, it defaults to 32/44100 (so _my_ asoundrc wasn't effective in any way) and have the possibility of different audio sources played synchronously.
well, that's what it see out from it.
I've read in many forums that in order to have bit perfect audio, set alsa to front-speakers for direct playback.
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I've read in many forums that in order to have bit perfect audio, set alsa to front-speakers for direct playback.
LOL, sounds like magic. Are you supposed to say it 3 times? Got a link that de-mysticizes it?
Maybe they are referring to /usr/share/alsa/pcm/front.conf, which is just an abstraction for the front 2 speakers. This is not a magic solution.
It sounds like you're finding that saying "phonon" 3 times isn't working either. Whenever I see phonon mentioned in the Arch or Gentoo forums, I never see any indication that users know what it's doing.
You know what would work? Specifying what you want the audio to do, in ~/.asoundrc, and avoiding whatever mangling phonon is trying to do.
You will notice that in my example, I specify the rate, with:
rate 48000
You could change that to 96000, for starters.
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Two simple questions:
Bypass any dmix setup, as dmix does resampling by default.
- Are you sure that if i play a single stream at 44khz and alsa is configured to use 44khz as default (and my soundcard supports 44khz of course), dmix will resample? Wasn't the resample itself the scope for the "plug" alsa plugin instead?
Use jackd instead, and if you want to output multiple sources at once, put pulseaudio infront of it. But the audioplayer should be connected to jack directly.
- Are you suggesting that pulseaudio mixes better than jack or what? Why one should ever use pulseaudio in a jackd environment?
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Ignore that bad over-simplification! It probably comes from this:
$ grep dmix.rate /usr/share/alsa/alsa.conf
defaults.pcm.dmix.rate 48000
That's a default, easily changed.
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