On Wednesday 29 April 2009, Nicola Mfb wrote: > 2009/4/19 Nicola Mfb <[email protected]>: > > 2009/4/19 Al Johnson <[email protected]>: > > [...] > > As AMI emits all needed events I'll add fso support for the GUI to > > handle the switching automatically, while for a true voip fso > > [...] > > I added fso support to switch between stereoout when ringing and > voip-handset when the call is established but asterisk does not reacts > well on this and stop to capture audio. > It works well if I set the voip scenario before launching it and never > switches to stereoout. > Before digging again in the asterisk alsa code I'd like to know if the > scenario switching is transparent to alsa applications, or may brings > underrun/overrun or other problems that needs to be managed in a > stronger way.
Scenario switching ought to be transparent to apps, but that might not be true if there's a change in the 'DAI mode' setting. There's more on this in the wiki: http://wiki.openmoko.org/wiki/Neo_1973_audio_subsystem I don't have the state files too hand to see if this is being changed, but it's the only setting I can think of that might upset an app. Can you reload chan_alsa after the state change? I don't remember how granular the asterisk reload options are, but it might be a quick'n'dirty workaround. _______________________________________________ Openmoko community mailing list [email protected] http://lists.openmoko.org/mailman/listinfo/community
