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Re: [Linphone-users] Noise and artefacts in sound
From: |
Matthias Peter Walther |
Subject: |
Re: [Linphone-users] Noise and artefacts in sound |
Date: |
Mon, 29 Feb 2016 21:43:22 +0100 |
User-agent: |
Mozilla/5.0 (X11; Linux x86_64; rv:38.0) Gecko/20100101 Thunderbird/38.5.1 |
Hi,
I googled to search, what I have to put in the alsa configuration to
route the sound directly to the output device and not through pulse
again. But I didn't find a solution. Do you know what option needs to be
set there?
Regards,
Matthias
Am 28.02.2016 um 00:40 schrieb J G Miller:
> At 23:12h, on Saturday, February 27, 2016,
> in message <address@hidden>,
> on the subject of "Re: Noise and artefacts in sound", you wrote -
>
> > Yeah, you're a little smarter than me I guess.
>
> Probably not -- just have suffered more from these types of bugs.
>
> > What I still don't get: Playing the ringtone from the audio options just
> > works fine.
>
> Which would indicate that it is an incoming stream related problem.
>
> Have you checked that when you call, whether the person on the remote side
> hears your voice with or without distortion?
>
> > I think this is a jitter problem or a decoding problem which would
> > explain why the sound on the recording is fine.
>
> Well if a recording has no distortion but the live call which you recorded
> did,
> then I would have thought the problem would not be jitter but the linphone
> decoding.
>
> And you have checked that the distortion still occurs if you put the output
> direct to
> ALSA and not pulseaudio (taking into account any default ALSA pcm being sent
> to pulse
> audio as already explained) as previously requested?
>
> > Where can I set the jitter?
>
> As far as I can tell, the only parameter you can change is in rtp.h
>
>
> <https://github.com/BelledonneCommunications/ortp/blob/master/include/ortp/rtp.h>
>
> #define RTP_DEFAULT_JITTER_TIME 80 /*miliseconds*/
>
> and the jitter buffer size
>
> float jitter_buffer_size_ms;/* mean jitter buffer size in milliseconds
>
> is calculated by some magic in rtp.c
>
>
> session->rtp.jitter_stats.jitter_buffer_size_ms=jitter_control_compute_mean_size(&session->rtp.jittctl);
>
> If you want to read a discussion about the way that Linphone handles jitter
> with ortp,
> take a look at
>
>
> <http://nongnu.13855.n7.nabble.COM/Bug-in-adaptive-jitter-buffer-td174758.html>
>
> Nonetheless, and I may be completely wrong and in all probability I am, I
> doubt that your
> audio distortion problem is because of the jitter settings in linphone.