----- Original Message ----
From: "freeswitch@dalethatcher.com" <freeswitch@dalethatcher.com>
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, January 28, 2008 8:01:00 AM
Subject: Re: [Freeswitch-users] Audio rate changes and clipping on playback
I mean it sounds like parts of the sound are missing. e.g. the
frequency of playback seems to change and I can hear a warbling/slow
down from the Freeswitch instance on the audio samples.
If you (or any others on the list) have a spare couple of minutes then
I'm happy to send you the details of my test server. It would be
reassuring if someone else could hear and verify the issue.
thanks,
- Dale
On Mon, Jan 28, 2008 at 02:05:20PM +0100, Sluschny, Thomas wrote:
> don't know what you mean with clipping, but I've notice clipping
errors when I play sound files with maximum levels with G.711, clipping
means that on max. posivite value a max. negative value is played.
> this is true only on G.711, e. g. Linear16 codes are fine.
>
> regards,
> thomas
>
>
>
> -----Ursprüngliche Nachricht-----
> Von:
freeswitch-users-bounces@lists.freeswitch.org
[mailto:
freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von
freeswitch@dalethatcher.com> Gesendet: Samstag, 26. Januar 2008 16:30
> An:
freeswitch-users@lists.freeswitch.org> Betreff: [Freeswitch-users] Audio rate changes and clipping on
playback
>
> Hello all!
>
> I'm experiencing some problems with clipping and the rate of playback
of
> wav files from Freeswitch. I've not raised this before as I thought
the
> problems were due to hosting over an ADSL line. However I've now
seen
> the same issues running from a dedicated host.
>
> My test setup is:
>
> Client <-> NAT ADSL <-> Freeswitch
>
> I've replicated using both the Gizmo and FWD clients from Windows.
If I
> substitute Asterisk with Freeswitch then the audio comes over fine.
The
> two test samples are one of myself and the fpm-sunshine.wav that is
> bundled with asterisk.
>
> I've tried playing around with the codecs but see the same problems
> regardless. Also I've performed a test with a conference hosted on
the
> Freeswitch instance with myself and friend dialling in using Gizmo
and
> the audio is fine. Which implied to me that the link between Gizmo
to
> Freeswitch is ok.
>
> So I'm guessing that there is something wrong with the samples that
is
> causing the problem? They're both in this format:
>
> $ file fpm-sunshine.wav
> RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono
8000 Hz
>
> My deployment platform is Debian stable and the Freeswitch SVN
revision
> is 7343. I've tried looking in Jira but keep getting a system error
> page (for search terms 'audio', 'wav' and others).
>
> Is anyone else seeing the same issues or has a solution?
>
> thanks,
>
> - Dale
>
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