similar to: AEC in Sipek

On 04/14/2011 07:26 PM, LiMaoquan2000 wrote: > Hi All, > Many Thanks to Underwood for her excellent review of our big trouble > which prevent LMS-based AEC algorithms to be used in most computer. > Maybe it can be summaried as follows: > 1. Different sample rate of sampling and rendering does exists in most > low-cost soundcards (In my experiments over more than 20 soundcards,

I read the question: http://stackoverflow.com/questions/3403152/acoustic-echo-cancellation-in-java/3403198 But my problem is that my project is using Encoding and decoding mechanism using JSpeex and I am not sure whether I can integrate the above java code with my project. So can someone tell me how can I implement Acoustic Echo Cancellation (AEC) using JSpeex(Java Port of Speex).

Well, I am not the person to be asked about how Speex AEC works :) And I was referring speaker input as having direct access into what has been played thru speaker. (ASIO Soundcard provides interface for that) So if I am remebering correctly, with data from Mic and Speaker in sync, speex AEC should do the job. Tim _____ From: Guilherme Balena Versiani [mailto:guibv at

I've looked at some partial examples and read in the pdf Manual that the Acoustic Echo Canceller can be used outside of the speex codec as a preprocessor. I currently have a java app that uses the ffmpeg api for live streaming. I'm exploring integrating echo cancellation & I'd like to use the speexdsp AEC by mapping a javacpp library to communicate between java & speex. Audio

Hi all, I have a query. I developed NLMS acoustic echo cancellation algorithm for real time voip application and using general purpose processor. Used more filter length (4096/2048) for NLMS and faced CPU consumption.With less filter length unable to estimate the exact echo signal and the quality of the speech also getting degraded. Could any one please suggest me the filter length

Ok, let me be more clear on this. AFAIK, Windows OS doesn't expose speaker input as other OS ( Linux, Mac OS...). That puts you in bad spot in using Speex AEC with windows. Only way to work is to use Soundcard with ASIO functionality which does give you speaker input. But then you would have to impose that requirement on all your users. I heard Speex AEC works great from developer who

Jean-Marc Valin <Jean-Marc.Valin <at> USherbrooke.ca> writes: > > Le mercredi 07 juin 2006 ? 10:03 +0000, shridhar desai a ?crit : > > > > hi all, > > i am using the Acoustic Echo Cancellation from "Speex 1.1.12 version" > > in my VOIP application. Is it that the frame length to be chosen > > should always be 20ms or can i have

On Mar 15, 2010, at 8:46 AM, Jean-Marc Valin wrote: > If more than one speaker receives the *same* signal, it doesn't matter the > number of speakers. It only gets tricky when the speakers are playing slightly > different signals (e.g. from a stereo song). > Does "tricky" mean that the Speex AEC won't handle such situations well? Or just that you had to be

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

One thing I can think of is if you are using two or more speakers. If the speakers are not at the exact same distance from the mic, you will get more than one echo. AEC can not handle that. Try disconnecting all but one speaker and see if it makes any difference. cheers Greger 2010/3/15 Anton A. Shpakovsky <saa at tomsksoft.com> > Hello. > > I have the following situation. AEC

Ok. Thanks J-M for clearing that up. What if you mix up the polarity on one speaker (180 degree phase change), would that matter? cheers Greger 2010/3/15 Jean-Marc Valin <Jean-Marc.Valin at usherbrooke.ca> > If more than one speaker receives the *same* signal, it doesn't matter the > number of speakers. It only gets tricky when the speakers are playing > slightly >

On 2010-03-16 06:35, Greger Burman wrote: > Ok. Thanks J-M for clearing that up. > What if you mix up the polarity on one speaker (180 degree phase > change), would that matter? Not at all. It's still all linear. You can even apply a different equalizer to each speaker and it'll still be linear. Jean-Marc > cheers > Greger > > 2010/3/15 Jean-Marc Valin

Well, AFAIK AEC engine needs time to adapt. So probably that's why at the beginning of the conversation you hear the consequences of this adaptation. Btw, what sound system do you have? I mean laptop or arbitrary acoustic system? I'd like to know, cause I couldn't get enough echo cancellation with speex in my environment with standalone speakers AT ALL. -----Original Message-----

Hi All, Many Thanks to Underwood for her excellent review of our big trouble which prevent LMS-based AEC algorithms to be used in most computer. Maybe it can be summaried as follows: 1. Different sample rate of sampling and rendering does exists in most low-cost soundcards (In my experiments over more than 20 soundcards, the differences range from 0.5Hz to more than 50Hz when sample rate is set

Hi, Two related news here. First, I've merged in support for stereo (or more channels) acoustic echo cancellation in mainline. The standard AEC remains the same, but I added a speex_echo_state_init_mc() call that takes the number of speakers and microphones. Now, one of the consequence of this new work is that by merging this work into the main tree, I had to do things in git that svn just

On 04/04/2011 06:58 PM, Simon Morlat wrote: > Hi, > > I implemented a small patch that allows the internal convergence state > of the echo canceller to be saved in a file for later use, especially > after a process restart or machine reboot. This enables immediate echo > cancellation the second time the AEC is run. > > Of course this works only if the acoustic environment

Hello. I have the following situation. AEC is used in network chat software over DirectSound API. Echo and reference signals are almost aligned (delay is no more than 30ms). When echo is emulated in notebook (built-in speakers + mic) everything goes fine and echo is cancelled. But when configuration includes stand-alone speakers and mic no echo is removed. Audio is in 22050 hz at 16 bit

Hi, I've got a question about the acoustic echo canceler included in the speexdsp library. Is there any limit on the maximum sampling rate it supports? Looking at the docs and the source code, I can't seem to find any. Is it capable of operating at 48000 Hz out of the box, or are there some modifications required? Thank you in advance, Gregory.

Yes. I guessed that too, however I am not sure why it keeps repeating every time the user stops / pauses and starts speaking again in a single session. I am using a laptop with standalone speakers. For echo cancellation to work one has to make sure that the ref and echo buffers are synchronized. I guess this is the most common problem. -Elston -----Original Message----- From: Anton A.

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