t;
> > If your application is SIP based, you can just use telepathy-sofiasip as
> your backend.
> >
> > Regards,
> > Chitti
> >
> > --- On Sun, 5/2/10, maemo-developers-requ...@maemo.org <
> maemo-developers-requ...@maemo.org> wrote:
> >
> >
ou using Telepathy for your VOIP application ? I think farsight
> (StreamedMedia plugin on gstreamer) handles all such intricacies.
>
> If your application is SIP based, you can just use telepathy-sofiasip as your
> backend.
>
> Regards,
> Chitti
>
> --- On Sun, 5/2/10, maemo-d
opers-requ...@maemo.org
wrote:
Message: 8
Date: Sun, 2 May 2010 08:43:04 +0530
From: saurabh aggarwal
Subject: Re: Audio Routing API
To: Martin Grimme
Cc: maemo-developers@maemo.org
Message-ID:
Content-Type: text/plain; charset="iso-8859-1"
Thanks Martin, I have a feeling that I
Thanks Martin, I have a feeling that I should be able to set those flags in
alsa using the gstreamer APIs and pulsesink.
The pulsesink object does accept a propllist, and using pactl if you look at
the pulsesink, there seem to be properties that should be able to switch
between headset and speaker
Hi,
alsamixer -c0
reveals a switch to route audio to the earpiece (I don't remember it's
exact name). I don't think you can route sound to the earpiece any
other way. Somebody please correct me if I'm wrong.
Martin
2010/4/28, saurabh aggarwal :
> We are developing a VoIP application, and have
