similar to: Re: aec

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

Are you sure you're not just inverting the two inputs? Jean-Marc On Wed, 2005-11-09 at 22:16 -0800, Jason Harper wrote: > 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

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

This kind of behaviour is odd. One of the reason could be the fact that you're using a really long impulse response. Try syncing your signals and making the tail length more in the order of 100 ms to 300 ms. Jean-Marc Le dimanche 06 novembre 2005 ? 21:25 -0800, Jason Harper a ?crit : > Thanks for alerting me to the new changes. I just > tried the latest code from SVN, but

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

Thanks for taking a look. There was no motion; however you are right about sampling from a different card. The speaker is connected to the Sound Blaster card, while the microphone is part of a USB webcam. I don't think that this is likely to be too unusual a configuration among users. I can retry the test using a sound card microphone to see if there is a difference. If it turns out that

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 somewhere in some of the OS resampling algorithms, but I've seen that on many machines.

I wasn't implying that anyone do anything about it, just that's it a real problem. Unfortunately, most of the crappy sound cards are the ones that ship with your typical PC, so it's just something that people should be aware of. The solution is pretty straightforward -- just resample the audio data in real time using a reference clock. -----Original Message----- From: Jean-Marc

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

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 >

To everyone on the list: do *NOT* attempt to do echo cancellation with signals sampled using different clocks. This will *NOT* work. Just a 0.1% difference between the two sampling rates (it's sometimes worse than that) means that the impulse response drifts by 8 samples every second. There's just no way to efficiently track this. Or at least no way that doesn't involve something 100x

I've recently made changes to the AEC. Please try the code in SVN and see if it works better. Jean-Marc Le jeudi 03 novembre 2005 ? 22:36 -0800, Jason Harper a ?crit : > 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 >

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.

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. That's way too much to do anything useful because the tail doesn't even "see" the echo. You need to reduce

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

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.

Hi Jean-Marc, I tried with both testecho and my test program, and for some reason it just doesn't cancel any echoes with the 1.2-rc1. The testecho from beta3 binaries works fine, and also if I replaced mdf.c in 1.2-rc1 with mdf.c from beta3 and use my test program, it will work again. This happens for both 8KHz and 16KHz. Any ideas? I could upload the test samples and results if needed.

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

You may have triggered an instability problem. Can you upload your files somewhere so I can have a look at them? Jean-Marc Andras Kadinger a ?crit : > 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

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