[Speex-dev] Server Side AEC

[Tabitha Flash]

  Hi Jean-Marc,
   
  Regarding you points:
   
  1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ?
  
2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals.
  
3) Occasionaly audio is dropped if it has become delayed but a jitter buffer of 120ms is in use.
   
  People at different distances from the server will have a slightly different round trip time. Do you think if using a large tail or something  we can get near perfect AEC? The same as you get with a hands free phone perhaps?
  
Does it still sound like worth a try? Is Speex AEC as good as it gets or would it be worth contacting some vendors of such software?
   
  Thanks again,
  Tabby 


Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote:  Tabitha Flash a écrit :
> Hi, I am looking for AEC software which can be run on the server
> side. This means there will be a fairly constant 600ms or so gap
> between sending out an audio frame and getting it back with echo.
> Could Speex AEC be configured to handle these conditions? 

Just use a ring buffer to put the delay back to normal.

> If so, how
> good can I expect it to be?

You'll need to try, but to have any chance of working, the following
conditions must be met:
1) No codec must be used in the echo path (maybe G.711 is OK)
2) There must not be any drift in the sampling clocks
3) There must not be any audio samples lost on the echo path

Jean-Marc


       
---------------------------------
 Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070722/09e80c8f/attachment.htm