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AFAIK, that's a common point for all AECs. But some of them solve the problem by resampling on of the end to keep it in sync with the other. On Tue, Jul 7, 2009 at 5:14 PM, ggb<ggb at tid.es> wrote: > Thank you John. > > On 07/06/2009 11:03 PM, John Ridges wrote: > > ly synchronized, and therefore the clock drift adds a non-linear > factor to the audio path. The AEC can only cancel linear changes to the > audio path, and so the AEC never converg...

The problem with different sound cards is that their clocks are not usually synchronized, and therefore the clock drift adds a non-linear factor to the audio path. The AEC can only cancel linear changes to the audio path, and so the AEC never converges.One solution is to measure the clock drift and resample either the input or output signal so that they *are* synchronized, and then the AEC

...ot at liberty to get any more detailed than that. Sorry. John Ridges Alexander Chemeris wrote: > AFAIK, that's a common point for all AECs. But some of them > solve the problem by resampling on of the end to keep it in sync > with the other. > > On Tue, Jul 7, 2009 at 5:14 PM, ggb<ggb at tid.es> wrote: >??? >> Thank you John. >> >> On 07/06/2009 11:03 PM, John Ridges wrote: >> >> ly synchronized, and therefore the clock drift adds a non-linear >> factor to the audio path. The AEC can only cancel linear changes to the >> audi...

Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G

Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...nes?" I told her, "I hope so, but we're lucky to have something as good as what we have now!" :) Anyway, if you search the list archives you should be able to find some comments about how the echo functions are supposed to be used... Good luck, Tom GUSTAVO GARCIA BERNARDO (ggb@tid.es) wrote: > > Hi, > I think speex library support echo suppresion/cancellation (may be experimental). How can i test it? Are there any sample or documentation about it? > > Thx. Best regards. > > ..g --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg proj...

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Hi, Somebody has tested succesfully the speex echo cancellator? Sometimes ago I post the same question, and it was very untested. Best regards, and thank you very much for your work. G.

Hello, It's possible to provide threewaycalling service in asterisk (nor in terminals) for SIP users? I would like to be able to join to calls in a threewaycall sending some dtfm. Thank you in advance for the information G. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm trying to establish jingle call in this network scenario: Asterisk -> NAT -> Internet -> HTTP_PROXY -> GTalk client The call is received and answered in gtalk but there is no audio in the call. I suppose it could be related to the support for relay candidates in asterisk jingle implementation. Anybody else has faced this problem? Notes: Asterisk 1.6.2. It

Hi, When i include a ',' in the application data, for example: exten => 1,1,Plaback(echo,file); [My file name is "echo,file"] the Playback application receives echo|file as data parameter. How can avoid asterisk change the ',' character? (The change code is in asterisk/pbx/pbx_config.c Ln 1150). I'm developping an application for asterisk and i need to pass

Hello, I would like to test the echo cancel algorithm available in unstable version 1.1.4. This echo canceller can be used with other codecs like G711? Somebody could send me some documentation or sample, or explain the next functions: SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length); void speex_echo_state_destroy(SpeexEchoState *st); void