[Speex-dev] Server Side AEC

[Tabitha Flash]

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it.
   
  It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers.
   
  Any ideas? 
   
  
Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote:
  > 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 ?

No. The entire path from AEC to loudspeaker and from mic back to AEC
must be free of any non-linearity, codec, drift, ...

> 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.

You're free to do the processing/blocking as you like.

> 3) Occasionaly audio is dropped if it has become delayed but a jitter
> buffer of 120ms is in use.

That'll cause problems.

> 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?

No. You may have good results if you compensate for the delay, though.

> 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?

>From what you write above, no AEC can work at all. Why not just move it
directly to the client instead?

Jean-Marc

> Thanks again, Tabby
> 
> 
> Jean-Marc Valin 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
> 
> 
> 
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