similar to: Server Side AEC

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

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@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using

Hi, I am very glad to enter this mail list. i am a new guy,i am interested in speech coding.so is there anybody working on this ? please contact me,thank you.i am studing on fixed codebook search. and i have read almost all the articals(including patent on this) about codebook search. but i found that all the suboptimize methods have their own flaw. could anybody have some comments on

> 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

Tabitha Flash a ?crit : > 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

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

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: comunip.gif Type: image/gif Size: 1663 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.gif

Hi,all I have tested AEC on files, it works well.I have some files,one is echo file, others are echo-added files(an origin file adding echo at different delay,such as 20ms,40ms...120ms,140ms).AEC do wonderfully on those files except echo added at 140ms-delay. But ,when i use AEC in my voip project, it does feebly. Who can give me some hints why caused this.How long can sound be picked up by

Hi, I couldn't find a discussion that specifically addresses this, so here it is. I'm using Speex AEC in my mobile VoIP application to cancel speaker echo. The used version is 1.2rc1 from the website, and I'm compiling with fixed-point. On most occasions, the AEC works very well and cancels most of the echo (combined with the preprocessor). On some devices, where the microphone signal

speex-devDear Jean-Marc Valin I got some problems with evaluating the AEC module of speex. I wrote a test main function and compiled it with the speex lib in VC6.0, it initialized the AEC state and called the AEC main function in the same way as what was done in testecho.c. The near-end input wave file was a simple delaying and adding version of the far-end input wave, eg. y(n) =

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

Analog Devices sponsored a fixed point version for AEC which was posted early this year, but it seems from the source code that the fft routines are still in floating point. Is the port still not complete or am I missing something? Has anyone out there ported speex AEC on RISC architecture? Please let me know. Thanks in advance, -Deepa -------------- next part -------------- An HTML

> Yes, you are not forced to reset the AEC between calls at all, if you > can assume the actual echo path (physical environment) will not change > much, you can keep the instance and continue using it for the next call. > > Also what we did was keep the AEC running between calls - but we had a > rather high-duty system with sounds and signals played between calls, so > the AEC

Hi Jean-Marc, thanks for the response. First, I will clarify again that floating-point solves this - so isn't that a bug in fixed-point? Also, I understand that algorithmically the AEC won't cancel echo properly on a non-linear signal, but why completely distort the output? If the echo just won't get cancelled it would be acceptable, but in the current state it disables the ability to

On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: > The fact that the AEC takes a few seconds to converge is normal. The Do you think there might be a way to reduce this? > fact that it needs to completely re-converge in the middle of a call > probably indicates that something went "wrong" in the audio > capture/playback. For example, that could be an

Perhaps you could add a warm-start to the AEC, such that the parameters start near the correct values on all but the first use? Stuart On 05/25/2011 08:22 AM, Andras Kadinger wrote: > 2011.05.25. 4:52 keltez?ssel, Arun Raghavan ?rta: >> On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: >>> The fact that the AEC takes a few seconds to converge is normal. The >> Do

Hi: By 'no success', I mean result gets worse in terms that, there comes in a shrill noise with high gain and also *no* attenuation of the echo. I am trying to validate the DC offset part. But can you please feed some light on how does the DC offset matter for AEC. As I have already mentioned, I tried adding silence (from 2 ms to 10 ms) in the start of the reference signal. This is for

> Today's Topics: > > 1. AEC with WB mode (Jean-Christophe.Berge@etu.enseeiht.fr) > 2. Multiple frame encode and decode (Reza Fatahillah) > 3. cant link speex_echo.h (jesus) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 20 Sep 2006 08:46:03 +0200 (CEST) > From:

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

Hello all, I'm just new to this list. I'm user of a voIP application which uses Speex Acoustic Echo Canceller (so only user of speex). The only parameter it has to tune is the length of the ec tail. I'm testing the echo canceller on an arm9 platform. Tests made only allow me to raise the length of the tail until 250 msec (upper the cpu power consumption is 100% and there are audio