similar to: Using AEC on a mobile device where earpiece is routed differently

Hi, Speex devs, I apologize in advance if this is not the proper venue for this question, but I had seen in the archives that other threads of this nature had been discussed. In brief, I'm trying to get AEC working on a simple Android NDK app. It's a basic "play from a file, record from the mic to file" loopback test. I'm using OpenSL ES. I establish a player and a

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

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

Hi Guys, I am trying to incorporate the speex AEC into my RTP stack. Trying to diagnose why the echo is not being cancelled I enabled DUMP_ECHO_CANCEL_DATA as per the docs. The problem is that the dump files the AEC generates are of different size. As I understand this and looking at echo_diagnostic.m, the files are supposed to be of same size. Can someone give me a hint as to why this could

Hi, Rhett, echo_diagnostic.m can produce an echo canceled file along with the possible warning messages. Can it cancel the echo properly? It may announce the "Drift estimate is ..." message if the recorded sound contains not only echo but also other sound like a near-end speech. In other cases, it may truly indicate a mismatch of the sampling frequency. Regards Kaiyu On Thu, Feb 6,

Sorry for not being accurate in the original question. My testing environment is set to 20ms frames (PCM 16 8 1 to be precise). I have verified that all the frames that I feed to the AEC have the fixed size of 320 bytes. So this is why I am presuming that the the dumped files should be of equal size. The output signal is good and not distorted. It has echo, though. In a try to diagnose that I

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

Hello everyone, I'm a new comer to this mailing list. I found that the echo_diagnostic.m included in speex-1.2rc1 seems to have grammar mistakes which cause parse error in octave 3.6.2. I wonder if anyone have ever executed this code under octave/matlab. I also made some minor modifications to the file to correct those errors and publish it here. If it can helps the project to be more

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

Hi, The Speex AEC is simply not designed to deal with non-linear echo, as is the case when clipping of AGCs are involved. Make sure all your path is linear or forget about the Speex AEC. Jean-Marc On 11-01-03 11:31 AM, Omer Gilad wrote: > Hi, > I couldn't find a discussion that specifically addresses this, so here > it is. > > I'm using Speex AEC in my mobile VoIP

Hi Dmitry, I met the same problem a few months ago and at that time I used 1.2beta2 code/library downloaded from Speex homepage. That bug has been fixed in the most updated source code in svn. Regards, Eddy ----- Original Message ---- From: Dmitry Nikitin <dmitry.nikitin@advatel.com.au> To: speex-dev@xiph.org Sent: Wednesday, December 5, 2007 12:31:57 PM Subject: [Speex-dev] AEC

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

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Definitely! It tells me that my delay is fine but drifting is -15. I use the same onboard sound to record the near end and far end voice. So I can assume it should be fine? But I don't see the any different between input and output the echo_canceller function for this recording. Maybe there is some nonlinearity issue there? The link of this file(Capture1.zip) is in

All I can say for now is that: 1) My diagnostic tool does funny things that need to be fixed 2) It was probably also getting confused by the clicks at the end of the files 3) There seems to be odd things with your recordings, though I can't say what that would be. Are you changing something during the recording by any chance? Note that if a person close to the mic of speaker moves, that

> Thanks for the great work. It indeed helps to diagnose of echo > canceller and narrow down the > problem area. Good, so it worked for you? > I am not an expert in the signaling processing but I > wonder if it is possible > to add the linearity check in the echo_diagnostic function? Theoretically, yes. In practice, I'm not sure how accurate it would be. Do you have any

Hi, I've sorted the performance issues i had in a previous post. Now the AEC runs but does not cancel any echo. I have a scenario where i have a single thread processing thread messages WOM_DONE and WIM_DATA which tell me when Windows Mobile OS has played the buffer or received the buffer respectively. Anyone any ideas how to sync the mic and the speaker. Any tips would be useful. Thanks

Hi everyone, I think this should be interesting to all of those with echo cancellation problems. I finally check in some code to make these problems easier to debug. You'll need to have svn version for this. First, you need to manually define DUMP_ECHO_CANCEL_DATA in the compilation (sorry configure switch for now). With that, the AEC will automatically save the near-end, far-end and output

Thank you for you quick reply, Jean-Marc. I have just used a human voice to replace the sine wave, but the result is like a broken voice after tens of seconds. Do you have some sample voice in .sw that I can test with? I want to have one set or a few sets of sample voice that work for my understanding and debugging. This is the output from echo_diagnostic.m in my human voice test: Far end to

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