similar to: Stable 1.2 release

It appears that both the echo canceller and the noise reducer (NR) introduce rather severe clicking artifacts when presented with audio that has slightly overloaded the A/D converters. I am talking about speech that sounds just slightly distorted, due to clipping, when simply played back. If I pass that speech through the echo canceller or the noise reducer, it acquires really loud clicks and/or

On Mon, Aug 11, 2008 at 12:34 PM, Jean-Marc Valin < jean-marc.valin at usherbrooke.ca> wrote: > OK, here's what happens. There is indeed a small difference between > beta3 and rc1, but the fundamental problem isn't there. I've attached > plots of the speaker signal (blue) alongside the mic signal (green). You > can see the delay is in the order of 1000 samples.

Greetings everyone, I was about to compare AEC performance between 1.1.12 and 1.2beta1 when I noticed something. If I configure (and compile) speex-1.1.12 with ./configure --enable-shared=no --enable-static=yes it compiles and works as expected: I can run a mic and speaker signal through testecho, it runs in a reasonable amount of time (about 23 secs for 3 minutes of audio) and I get back

I'll try to make some shorter samples later, but for now here are the ones I have tried with: http://www.surfnonstop.com/~bandit/speex/1.2beta1_AEC_garble/ The original recordings are in mic.raw and spk.raw. Jean-Marc Valin wrote: > You may have triggered an instability problem. Can you upload your files > somewhere so I can have a look at them? > > Jean-Marc > > Andras

Millions of ordinary people have systems that do not meet your stringent specifications. It does not matter what OS you or I use. It's the customers for whom this work is done. -----Original Message----- From: Jean-Marc Valin [mailto:Jean-Marc.Valin@USherbrooke.ca] Sent: Monday, July 02, 2007 6:35 PM To: Coffey, Michael Cc: speex-dev@xiph.org Subject: RE: [Speex-dev] Backup Echo Suppression

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) =

Jean-Marc, I think I can speak for most everyone here when I say we appreciate *very much* your code beyond the codec that helps people who use Speex in real applications. How far you want to take that code is up to you, of course. (And contributors, to a degree.) I don't think any reasonable person is expecting your code to solve everything. But when people are running up against

Believe me; I've "played with" priorities and buffering. Did you just say you have no idea how the Speex residual echo suppressor works? If that is the case, can you tell me where I could get some information about it? -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Friday, June 29, 2007 7:38 PM To: Coffey, Michael Cc:

I would like to add AEC to my program. Since the nature of my program is such that its playback and recording are asynchronous. I have no idea whether the testecho.c example will work for me. Anyone here has experience with the AEC in async mode? YUN TAO

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 >

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

On Sat, Aug 9, 2008 at 12:59 PM, Jean-Marc Valin < jean-marc.valin at usherbrooke.ca> wrote: > Hi Benny, > > Can you send me your pair of testecho input files that work well with > beta3 and not with rc1? I'll have a look. > > Thanks for the help. The files are on their way now, the upload will take few more minutes to complete. In the mean time let me explain more

What is the current status of the residual echo suppressor in the preprocessor? I ask this because I have occasional problems with the AEC, in which it fails to cancel echo. I think the problems have to do with samples being dropped by the audio IO system, which is effectively impossible to detect and counteract reliably. A colleague recommended using an echo suppression technique (which he

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

Hi Folks, I am having trouble using speex_echo_cancel. As a starting point, I am using the testecho.c source code. I compiled the 1.2beta1 version. I have not tried any other versions of speex. The document says that the order of arguments to Speex_echo_cancel is (echo_state, input_frame, echo_frame, output_frame, residue) where "input_frame" is as captured from mic 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

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

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.

When you say inverted, do you mean a) input and output exchanged with each other, or b) input or output having changed sign numerically (e.g. multiplied by -1)? Andras Jean-Marc Valin wrote: > Make sure that: > > 1) You're using svn head, which has many improvements > 2) The AEC inputs aren't inverted > 3) The delay between the near end and far end signals isn't too

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