[Gta04-owner] Routing UMTS sound

[Gilles Filippini]

Neil Jerram a écrit , Le 30/11/2012 01:29:
>> On Sunday 01 January 2012 09:14:14 NeilBrown wrote:
>>
>>> I would suggest trying to set up pulseaudio to do the routing.
> [...]
>>>   pactl load-module module-loopback   \
>>>        source=alsa_input.platform-soc-audio.0.analog-stereo  \
>>>        sink=alsa_output.platform-soc-audio.1.analog-mono \
>>>        rate=8000
>>>
>>>   pactl load-module module-loopback \
>>>        source=alsa_input.platform-soc-audio.1.analog-mono \
>>>        sink=alsa_output.platform-soc-audio.0.analog-stereo \
>>>        rate=8000
> [...]
> A mere 10 months after the above emails... :-)
> 
> I'm happy to report that I just had this approach working - i.e. using
> pulseaudio and those 'pactl ...' lines to route the media during a phone
> call.

That's really good news!

> I think the trick is just to change pulseaudio's default resampling
> method.  Normally it's speex-float-3, but that performs very badly on
> the ARMEL architecture - meaning that there's basically no sound
> transfer at all.  Switching to speex-fixed-3 (in /etc/pulse/daemon.conf)
> gives clear sound.
> 
> (FWIW, the 'trivial' resampler also gives fairly clear sound, but makes
> the voice sound tinny or robotic.)

Could probably be kept as float on GTA04/armhf, then.

> I say 'I think' because I made some other changes before discovering
> that - notably upgrading to the testing version of pulseaudio - so it's
> possible that there's something there that is needed as well as the
> speex-fixed-3 change.
> 
> There's still some work to do to pin that down and to integrate it all
> nicely - but I hope this will be the way forward for completely reliable
> audio routing for A3 phones.  (In QtMoko.  SHR use alsaloop, which I
> presume is also pretty reliable.)

Thanks,

_g.