similar to: Speex 1.1.9 is out -- Try the new echo canceller

Hello Jean-Marc, I solved the variable delay problem, but I still have trouble with speex_echo_cancel(). When i try testecho.c with clean speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

I ran some further tests on mdf and here are the results: 1. reduced tail length to 100ms, aligned mic and speaker signals to within 10ms - almost no echo attenuation 2. aligned mic and speaker signals to within 5 samples - still almost no echo attenuation 3. ran testecho using the same file for mic and speaker - very good echo cancellation (of course this is expected, but I needed to do a sanity

I'm pretty much sure of it. When I test inverting the inputs, my output is pretty much the same as my speaker signal. Whereas the way that I normally test the output is my mic signal with very little attenuation. If you are interested I can send my test files; they are about 94KB each. -Jason --- Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > Are you sure you're

Hi Julia, Version 1.2-beta1 has a bug in the echo canceller. Try either svn or 1.1.12 -- or apply this patch to 1.2beta1: https://trac.xiph.org/changeset/11882 Jean-Marc julia rg a ?crit : > Hello Jean-Marc, > > > > I solved the variable delay problem, but I still have trouble with > speex_echo_cancel(). When i try testecho.c with clean speech for far-end input > and

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being

(from thread Re: [Speex-dev] Speex echo canceller on TI C55 DSP, but this is a more general topic) >> With these changes, the link was successful, using testecho.c with some >> modifications for the C55 environment. The code and data memory >> requirements were a lot more than I had hoped (>20kbytes of dynamic data >> memory for block size=128, tail length = 1024),

Had a try. The reason why a simple delay is not that good is mainly due to the initialization of the filter parameter that still takes a few seconds (if they are perfectly in sync, you sort of get lucky). Otherwise, you real recording seems to have something odd in it. Are you sampling from a different card then the one that's playing the sound? or maybe the mic (or something else) in the room

Thanks for alerting me to the new changes. I just tried the latest code from SVN, but unfortunately I still have just about the same results. The estimated echo that gets subtracted from the actual echo is such a small signal that it doesn't really result in any noticeable echo attenuation. I currently have my filter size set to 2 seconds even though the echo in my microphone file is only

> I built and ran the same test on the TI C64 simulator, and the echo was > canceled nicely (about 10:1 reduction in the peak amplitude during the > second of two brief speech bursts). So, my problem must again be related to > the 16-bit processing on the C5X DSPs. Good. At least we've narrowed it down a bit. > Also, the line where it is hanging is: >

>Just tried your files and I'm not running into any infinite loops and >the cancellation works fine. Unless the C6x has the same problem, I >suspect a 16-bit problem. I'll check and see if I find something. About >the r=0 problem, I can't find where it ends up in a denominator, so I >suspect is not (directly) the problem. I built and ran the same test on the TI C64

Hi, I am currently encoding 32 Khz/Qual-10/UWB encode mode, with MANUAL_ALLOC enabled ( similar to the c5x/c6x configuration). In file sb_celp.c, I noticed the scratch memory grow during recursive (UWB -> WB) calls to 'sb_encode'. 1. 'stack' was not tracked (with 'tmp_stack' as done at other palaces) after/before - auto-correlation/Levinson-Durbin scope (~line

Hi Jim, I've just been made aware of these problems (look for the thread "speex echo cancellation limitations"). It's on my short-term TODO list. > If fftwrap.c, I ifdefed out the spx_fft_float and spx_ifft_float routines, > because there were not used and required smallft.c (which is not so small at > all) to be added to the build. Right, need to cleanup that

? Do you have any references for me to investigate, I am trying to understand how noise is reduced by introducing higher sampling rates. (I tried to search, but maybe it is so obvious that nobody even explains it) This is not very obvious. It requires you to understand basic signal processing theory. I will give some pointers below. Any physical signal (e.g. audio coming out of speaker, current

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

Just tried your files and I'm not running into any infinite loops and the cancellation works fine. Unless the C6x has the same problem, I suspect a 16-bit problem. I'll check and see if I find something. About the r=0 problem, I can't find where it ends up in a denominator, so I suspect is not (directly) the problem. Jean-Marc Le lundi 08 mai 2006 ? 20:05 -0400, Jim Crichton a ?crit

Jean-Marc, Well I finally tracked down the problem. Then I checked my mail and found that you had fixed it several hours earlier. :( Build 11387 produces the same result as my modified build 11343. Because of compiler limitations in the TI tools, I did have to make modifications to pseudofloat.h (separating return of float values) and nb_celp.c (adding braces around a variable declaration

Hello? My name is Hyeonwoo Oh. I?m developing video conference software using speex library. Can you give me some sample codes for acoustic echo canceller and noise reduction? I tried noise reduction API, but failed. Normal encode/decode API is working well. Thank you very much for your good job. I?m very pleased with speex. Best regards, Hyeonwoo Oh

I have successfully DECODED speex on TI C5509: #define TESTENC_BYTES_PER_FRAME 20 /* 8kbps */ #define TESTENC_QUALITY 4 /* 8kbps */ I am trying to generate the files I need with speexenc.exe: speexenc -n --quality 4 -V male.wav male.spx But I can't decode the files on C5x. Yes, I have seen that speexenc.exe adds Ogg header and

Hello. I'm from Federal University of Rio Grande do Sul - Brazil and I'm trying to adapt speex_echo.h (speex-1.2beta3-win32) to our video-conference software code. We are working on amplitude values, but I realised that in order to AEC works, one has to convert amplitude to frequency and filter the echo frequencies. So, my question is: Do I have to manually convert amplitude samples to