linphone-users
[Top][All Lists]
Advanced

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

Re: [Linphone-users] Noise and artefacts in sound


From: Matthias Peter Walther
Subject: Re: [Linphone-users] Noise and artefacts in sound
Date: Sat, 27 Feb 2016 18:45:04 +0100
User-agent: Mozilla/5.0 (X11; Linux x86_64; rv:38.0) Gecko/20100101 Thunderbird/38.5.1

Hi,

thanks for your reply. The option was set to speex-float-1 by default
and even resample-method = speex-float-3 didn't improve much.
(pulseaudio -k && restart of linphone.)

I don't think that the CPU is the problem here. It's an i7-2600K. Not up
to date anymore, but still a powerful processor. Skyping, TeamSpeak and
Mumble is no problem at all. (But still I want real SIP.)

Any other ideas?

Another info might helpful for a diagnosis: I recorded a call made via
linphone. When I play the waf-file the with mplayer (or any other
software) the sounds is much clearer and almost perfect.

So I think this is definitly a playback issue not a network or codec issue.

Or maybe it's a jitter problem? I don't know how linphone creates the
recording. Maybe some packages haven't arrived to the time where they're
needed for live stream, but are used to store the stream to the harddrive?

Regards,
Matthias


On 27.02.2016 17:19, J G Miller wrote:
> At 21:20h, on Friday, February 26, 2016,
> in message <address@hidden>,
> on the subject of "[Linphone-users] Noise and artefacts in sound",
> Matthias Peter Walther explained -
>
>  > I think it might be a problem between linphone and pulseaudio as Skype has
>  > had similiar problems some years ago.
>
> I think it may be a similar problem which I was suffering some time ago,
> which despite trying all the different codecs in linphone and bitrate settings
> resulted in audio distortion.
>
> At the time I just could not understand why the audio was bad even when the 
> best
> quality or least quality codec was selected.
>
> But, if the problem is correctly identified, it can be very easily fixed.
>
> Pulseaudio can use different resampling methods and this is set in the file
>
>                /etc/pulse/daemon.conf
>
> If the resample method is set to use a method which puts too much demand on
> your PC, then the audio is distorted.  The best compromise setting for my
> system after various tests was found to be
>
>               resample-method = speex-float-1
>
> You may well be able to use a better quality method but not the absolute (?) 
> best
>
>                    src-sinc-best-quality
>
> For further information see the table entry for "resample-method" with the
> "Tip" highlighted in green at
>
>       <https://wiki.archlinux.ORG/index.php/PulseAudio/Configuration>
>
> and also the section "Re-sampling using up a lot of CPU time" at
>
>       <https://wiki.gentoo.org/wiki/PulseAudio>
>
> Note that lowering the resample-method specifiction from the absolute best 
> will result
> in a lowering of sound quality but if too high a resample-method is specified 
> then the
> sound quality will suffer from artefacts and so be distorted, so it all 
> depends on
> what your resources your PC can provide.
>
> In fact to achieve the best sound quality means using the same sample-rate 
> and sample-format
> directly supported by the hardware and avoiding any resampling conversion at 
> all.
>
> The difficulty is getting pulseaudio not to resample from input to output has 
> been
> the most important criticism of pulseaudio by the sound quality experts and 
> purists.
>
> Hope this addresses your problem, which could be something else entirely ...




reply via email to

[Prev in Thread] Current Thread [Next in Thread]