<div>The client is the adobe flash player. No install and on 98% of all desktops but we can't change it.</div> <div> </div> <div>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.</div> <div> </div> <div>Any ideas? </div> <div> </div> <div><BR><B><I>Jean-Marc Valin <jean-marc.valin@usherbrooke.ca></I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">> 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent<BR>> to the client and decoded when it is recevied so the AEC is always<BR>> performed on raw PCM16 8KHZ ?<BR><BR>No. The entire path from AEC to loudspeaker and from mic back to AEC<BR>must be free of any non-linearity, codec, drift, ...<BR><BR>> 2) The audio is moved in 32ms (512 byte) chunks and the reading
and<BR>> writing to the AEC code will be done by separate threads at regular<BR>> 32 ms intervals.<BR><BR>You're free to do the processing/blocking as you like.<BR><BR>> 3) Occasionaly audio is dropped if it has become delayed but a jitter<BR>> buffer of 120ms is in use.<BR><BR>That'll cause problems.<BR><BR>> People at different distances from the server will have a slightly<BR>> different round trip time. Do you think if using a large tail or<BR>> something we can get near perfect AEC? The same as you get with a<BR>> hands free phone perhaps?<BR><BR>No. You may have good results if you compensate for the delay, though.<BR><BR>> Does it still sound like worth a try? Is Speex AEC as good as it gets<BR>> or would it be worth contacting some vendors of such software?<BR><BR>From what you write above, no AEC can work at all. Why not just move it<BR>directly to the client instead?<BR><BR>Jean-Marc<BR><BR>> Thanks again, Tabby<BR>> <BR>>
<BR>> Jean-Marc Valin <JEAN-MARC.VALIN@USHERBROOKE.CA>wrote: Tabitha<BR>> Flash a écrit :<BR>>> Hi, I am looking for AEC software which can be run on the server <BR>>> side. This means there will be a fairly constant 600ms or so gap <BR>>> between sending out an audio frame and getting it back with echo. <BR>>> Could Speex AEC be configured to handle these conditions?<BR>> <BR>> Just use a ring buffer to put the delay back to normal.<BR>> <BR>>> If so, how good can I expect it to be?<BR>> <BR>> You'll need to try, but to have any chance of working, the following <BR>> conditions must be met: 1) No codec must be used in the echo path<BR>> (maybe G.711 is OK) 2) There must not be any drift in the sampling<BR>> clocks 3) There must not be any audio samples lost on the echo path<BR>> <BR>> Jean-Marc<BR>> <BR>> <BR>> <BR>> --------------------------------- Yahoo! Mail is the world's<BR>> favourite
email. Don't settle for less, sign up for your freeaccount<BR>> today.<BR></BLOCKQUOTE><BR><p> 
<hr size=1>
Yahoo! Answers - Get better answers from someone who knows. <a
href="http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU">Try
it now</a>.