[Freeswitch-users] Audio rate changes and clipping on playback

Anthony Minessale anthmct at yahoo.com
Mon Jan 28 07:14:33 PST 2008


yes please install our ssh key and send me the details.

the key is in the svn tree in

ssh_keys/shinzon.pub

you can find us on irc all day or im or email
me.

 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
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pstn:213-799-1400


----- Original Message ----
From: "freeswitch at dalethatcher.com" <freeswitch at dalethatcher.com>
To: freeswitch-users at 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 at lists.freeswitch.org
 [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
 freeswitch at dalethatcher.com
> Gesendet: Samstag, 26. Januar 2008 16:30
> An: freeswitch-users at 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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