Hi Saurabh!
You can use the following script for routing playback between speaker
and headphones:
#!/bin/sh
if [ "${1}" == "speaker" ] ; then
# low headphone playback volume
amixer -qc0 cset iface=MIXER,name='Headphone Playback Volume' 5,5
amixer -qc0 cset iface=MIXER,name='Left DAC_L1 Mixer Line Switch' off
amixer -qc0 cset iface=MIXER,name='Left DAC_L1 Mixer HP Switch' on
amixer -qc0 cset iface=MIXER,name='Right DAC_R1 Mixer Line Switch' off
amixer -qc0 cset iface=MIXER,name='Right DAC_R1 Mixer HP Switch' on
amixer -qc0 cset iface=MIXER,name='HP DAC Playback Volume' 50,50
amixer -qc0 cset iface=MIXER,name='Speaker Function' 1
else if [ "${1}" == "headphone" ] ; then
amixer -qc0 cset iface=MIXER,name='Left DAC_L1 Mixer HP Switch' off
amixer -qc0 cset iface=MIXER,name='Left DAC_L1 Mixer Line Switch' on
amixer -qc0 cset iface=MIXER,name='Right DAC_R1 Mixer HP Switch' off
amixer -qc0 cset iface=MIXER,name='Right DAC_R1 Mixer Line Switch' on
amixer -qc0 cset iface=MIXER,name='HP DAC Playback Volume' 0,0
amixer -qc0 cset iface=MIXER,name='Speaker Function' 0
else
echo 'Please specify option: "speaker" or "headphone"' ;
fi
fi
Thanks, Daniil.
On Sun, May 2, 2010 at 6:29 PM, Chitti Babu <chitti_ma...@yahoo.com> wrote:
>
> Hi Saurabh,
> Were n't you 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-developers-requ...@maemo.org
> <maemo-developers-requ...@maemo.org> wrote:
>
> Message: 8
> Date: Sun, 2 May 2010 08:43:04 +0530
> From: saurabh aggarwal <aggarwal.saur...@gmail.com>
> Subject: Re: Audio Routing API
> To: Martin Grimme <martin.gri...@gmail.com>
> Cc: maemo-developers@maemo.org
> Message-ID:
> <l2le3a9904f1005012013q167658ccre8a0d368ff6b3...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> 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. Like x-maemo.mode which seems to be IHF or HP
> depending on whether pulsesink is in headset or speaker mode.
>
> I have been trying to use these flags, but haven't been able to do so till
> now. If someone can help here, that would be great.
>
> -Saurabh
>
> On Sat, May 1, 2010 at 3:46 AM, Martin Grimme <martin.gri...@gmail.com>wrote:
>
> > 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 <aggarwal.saur...@gmail.com>:
> > > We are developing a VoIP application, and have it running for Maemo5
> > (using
> > > gstreamer/pulseaudio).
> > >
> > > The problem now is that the output being used is the speaker, and not the
> > > headphone/earpiece.
> > >
> > > I have tried a couple of things (which haven't worked) like -
> > > gst-launch pulsesrc device=source.voice ! pulsesink device=sink.voice
> > > (outputs voice from speaker)
> > >
> > > gst-launch pulsesrc ! pulsesink device=sink.hw0 x-maemo.mode="hp" (I get
> > an
> > > error no element "hp"). Have tried the same with "ihf".
> > >
> > > If someone can point me to the right direction, that would be really
> > > appreciated.
> > >
> > > -Saurabh
> > >
> >
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