similar to: NLMS filter length for AEC

Hello Jean-Marc Valin and all the Speex-dev team, Today I have a little question about patents and the AEC. I' am not a lawyer, but after a deep research in esp@cenet.com, I found that in August 2003, an AEC called "Double talk insensitive NLMS algorithm" (DNLMS) have been patented by Gary Q Jin, Gordon J Reesor, and others (US 6.608.897 B1). It uses a continuous adaptation step

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) =

Hello, I tried to use speex echo canceller for a live performance setup to cancell feedback. My program buffers input until collecting a speex frame size of samples, then calls speex_echo_cancellation passing this buffer as input signal and previous speex output as echo signal, then immediately plays current speex output. First thing I noticed was huge high pass filtering. Then I found out there

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've tried some further debugging to see what mdf is actually doing. Instead of sending: tmp_out = (float)ref[i] - st->y[i+st->frame_size] to the output, I just sent st->y[i+st->frame_size] to see what was being subtracted from the microphone input. When I open this in Audacity, I see a very small signal at about -40dBm. The actual echo in my sample has a power closer to -20dBm.

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

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

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

> 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:

The AEC is not designed to cancel sinusoids. Try with speech. Jean-Marc Johnson Peng a ?crit : > 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. >

Thanks Jean, I missed that detail. I have a few more compile errors if you could help me with it => In function speex_echo_state_init in mdf.c I am getting error: "undefined reference to exp". I don't see a fixed point implementation of "exp". Can you point me to the correct files. I am using the following files from speex 1.2 beta1 => mdf.c,

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

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, I've just caught a strange crash in speex_echo_state_init routine. It happened only on one WinXP machine, while on others using XP, Vista and 7 everything is fine. Crash occures in mdf.c line 434: st->spec_average = DIV32_16(SHL32(EXTEND32(st->frame_size), 15), st->sampling_rate); Got any ideas or should I provide more information of the OS? Thanks a lot!

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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

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). Jean-Marc Quoting Greger Burman <greger at mobile-robotics.com>: > One thing I can think of is if you are using two or more speakers. If the > speakers are not at the exact same

On 2010-03-16 14:22, Josh Gargus 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

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

Le vendredi 11 novembre 2005 ? 01:21 -0800, Duane Storey a ?crit : > This is a very real problem though.. I've encountered many sound cards that > use different clocks for input and output (even on the same card!) Also, if > you open up a sound device on windows at 8kHz, the microphone is often > around 8100Hz, while the output is 8000Hz.. I'm not sure if there's a bug >