On Fri, 2009-07-10 at 00:39 +0300, George Kiagiadakis wrote:
> First, I have tried doing calls between my client (on the desktop) and google 
> talk (on the laptop). This scenario mostly works, I get correct audio on both 
> sides, but after a while (~30 seconds or so) the audio stream from my client 
> stops being delivered to gtalk and instead I get a continuous noise (always 
> the same) on gtalk.
> 
> Second, I have tried doing calls between my client and empathy. In this case, 
> although they connect fine, I don't get any audio on either side, instead I 
> get 
> a "tak tak tak" noise with exactly 1 Hz frequency. This noise seems to be 
> coming from FsConference itself. I'll try to explain what makes me believe 
> that: I have a gstreamer pipeline that looks like this:
> 
> audiosrc -> volume -> FsConference -> liveadder -> audioresample -> volume -> 
> audiosink
> 
> (audiosrc and audiosink can be anything. I have tried so far autoaudio{src, 
> sink}, alsa{src,sink}, oss{src, sink} and audiotestsrc.)
> 
> Empathy has a similar one that looks like this:
> 
> gconfaudiosrc -> volume -> level -> FsConference -> liveadder -> 
> audioresample 
> -> volume -> gconfaudiosink
> 
> The only practical difference is the "level" element, which apparently feeds 
> a 
> small visualization widget on the empathy window that shows the level of the 
> audio coming from the microphone.
> 
> Both sides receive this "tak" noise. Now, if I mute one side by setting the 
> volume on the "volume" element of the input to 0, the other side stops 
> getting 
> the noise. However, increasing the volume doesn't make any significant 
> difference. An important detail is that the visualization widget on empathy 
> reacts to the microphone and is not synchronized with the noise that I get on 
> the other side. So, that leads me to the conclusion that the FsConference 
> element somehow gets correct audio from the input and transforms it into this 
> noise. Thus, if it gets no audio, there is no noise.
> 
> The same exactly happens when I am trying to make calls with my client on 
> both 
> sides. However, calls with empathy on both sides work fine. This makes me 
> wonder what can be wrong with my client. Both my client and empathy are very 
> similar applications, both using telepathy, telepathy-farsight, gstreamer and 
> even (almost) the same pipeline.
> 
> Does anybody have any clue? What can be wrong?

This is strange, all that FsRtpConference does to audio is to
encode/decode it. Which codec is used?

-- 
Olivier Crête
olivier.cr...@collabora.co.uk

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