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[Iaxclient-devel] [Fwd: Re: Re; iaxclient on Linux ...]
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From: atuc <at...@gm...> - 2009-01-17 14:07:36
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-------- Original-Nachricht --------
Betreff: Re: [Iaxclient-devel] Re; iaxclient on Linux ...
Datum: Sat, 17 Jan 2009 14:49:47 +0100
Von: Alexander Tuchacek <tuc...@kr...>
An: Gareth Bult <ga...@en...>
Referenzen: <11933176.342711232103683798.JavaMail.root@zimbra>
Gareth Bult schrieb:
> Hi,
>
> I've been developing with iaxclient on Linux and had a fair amount of success in general, but I'm now running up against the buffers. Linux (Ubuntu in particular) seem to be standardising on "PulseAudio", which generally works fine - when you start an Alsa or OSS application, it activates a plugin which connects to PluseAudio allowing different applications to output sound at the same time. Great!
>
> First problem I notice is that the ALSA option in portaudio doesn't work, apparently because it's trying to use a sample rate of 8000 when the minimum for alsa appears to be 44100. Anyway, this is bypassable by using OSS and /dev/dsp ...
>
> aoss <application> "looks" like it's going to work .. when I run the app and the app opens /dev/dsp, off it goes and the sound comes out nicely. However, when my iaxclient based app is outputting sound and I try to get another app to output sound - nothing happens! Conversely, if I'm playing a CD and I get an incoming call .. silence from my app! (the underlying code reports it was unable to open /dev/dsp)
>
> Checking other docs, apparently there is a specific issue with PortAudio which means it's about the only library out there which doesn't play ball with PulseAudio .. :-(
>
> It doesn't look like there's much going on with PortAudio any more, I've seen no end of recommendations that developers use *anything* else, and at the moment they don't even have a mailing list (!) , so are there any plans to move to OSS/ALSA/Pulse directly and remove PortAudio .. ? Alternatively are there any moves afoot to make PortAudio play nice with Pluse ??
>
> .. or (!) can anyone suggest how I might get IAX2 working in a Linux app such that I can get sound from both the app and the rest of the Linux application base to run at the same time ??
>
i use a the alsa dmix plugin for default, it forces samplerate convert
to 44100, pulse audio shoes it as "My Alsa DMIX Soundcard"
my .asoundrc:
best alex
> pcm.dmixer {
> type dmix
> ipc_key 1024
> ipc_perm 0666 # Andere Benutzer können ebenfalls dmix gleichzeitig nutzen
> slave.pcm "snd_card"
> slave {
> pcm "hw:0,0"
> # buffer_size kann bei Problemen der jeweiligen Karte angepasst werden.
> period_time 0
> period_size 128
> buffer_size 8192
> # bei Störungen kann die Konvertierung auf die Rate 44100 eingeschaltet werden.
> rate 44100
> # einige Soundkarten benötigen das exakte Datenformat (zB ice1712)
> # format S32_LE
> }
> bindings {
> 0 0
> 1 1
> }
> }
>
> # Das dsnoop-Plugin, welches es erlaubt, mehrere Programme gleichzeitig aufnehmen zu lassen.
> pcm.dsnooper {
> type dsnoop
> ipc_key 2048
> ipc_perm 0666
> slave.pcm "snd_card"
> slave
> {
> pcm "hw:0,0"
> period_time 0
> period_size 128
> buffer_size 8192
> # bei Störungen kann die Konvertierung auf die Rate 44100 eingeschaltet werden.
> rate 44100
> # einige Soundkarten benötigen das exakte Datenformat (zB ice1712)
> # format S32_LE
> format S16_LE
> }
> bindings {
> 0 0
> 1 1
> }
> }
>
> # Dies definiert unser Fullduplex-Plugin als Standard für alle ALSA-Programme.
> pcm.duplex {
> type asym
> playback.pcm "dmixer"
> capture.pcm "dsnooper"
> hint {
> show on
> description "My Alsa DMIX Soundcard "
> }
> }
>
> pcm.!default {
> type plug
> slave.pcm "duplex"
> }
>
>
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