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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- 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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