On Saturday, December 01, 2012 06:15:04 PM Neil Jerram wrote: > Given that, I think it's worth me writing a bit more about where/how my > work is going, and there's one bugfix below that you should cherry-pick: > please look for "While doing that I noticed a bug". Apart from that > bugfix I don't want to make any assumptions about what time you have to > consider this, so please feel free to leave the rest of this hanging for > now. But if you do have time and thoughts I'm sure those would be > useful, and in any case it's helpful for me to lay out my thoughts.
Hmm i plan some relaxing now ;-) > But after that, the integrated pulseaudio in-call audio routing seems to > work. Of course I'll be more confident about it if I can have a week of > it working every time... Very nice. > While doing that I noticed a bug in my previous "Rework ALSA state / > QAudioState handling" commit: it will call gsmVoiceStart() and > gsmVoiceStop() even for A4 devices. That's fixed by > > https://github.com/neiljerram/qtmoko/commit/a362c431531d6b75fcb1894c60e0215 > 88ea50d44 > > so that's one commit that you _should_ cherry-pick. done > Next what I'd like to do is to make everything louder! yup, that would be nice > There are 3 > things that aren't loud enough at the moment: > > a) the ringtone > > b) the in-call audio that I hear from the other person > > c) the in-call audio that the other person hears from me. > > (b) and (c) depend on good echo cancellation, and I'm hoping > pulseaudio's module-echo-cancel will do that for me. I think I can test > that, without needing lots of phone calls, simply by playing something > (from file) through the earpiece or speaker and simultaneously recording > from the microphone. If that works well, we can then just increase all > the volumes in the state files. > > (BTW I think that the Speex echo cancellation in gta04-gsm-voice-routing > was less effective because of the playback buffer being 4 times the > frame size. This means that when the code sends frame X to ALSA for > playback, frame X doesn't actually start playing until 3 cycles later. > Therefore we can't immediately use frame X to cancel echo in the > microphone sound that we're capturing _now_. I wonder if there was a > time when the code had buffer size = frame size, and the buffer size was > later increased?) Hmm it was long time ago. IIRC with small buffer size it was not working - but i cant recall the details. :( > Finally, if all of the above works, we can look at whether it all > _still_ works with the squeeze version of pulseaudio. > > Hopefully eventually the software audio routing can be good enough for > A3 audio quality to be on a par with A4. Yes that would be nice. And even A4 could benefit this - e.g. for recording phone calls. Regards Radek _______________________________________________ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community