similar to: Re: aec

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

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 >

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

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

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.

When I ran test 4 as originally described there is substantial echo cancellation (but not as good as when the files are perfectly aligned). When I invert the inputs, there is no noticeable cancellation. I'm using testecho with the preprocess line commented out. Preprocess seems to work very well at cleaning up the residual echo when mdf does its job, so I'm just focusing on testing mdf.

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

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

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

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

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

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.

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.

Jean Marc & list, Following up on this- there seems to be an issue with this calculation in mdf.c (Smooth echo energy estimate over time) Pey += Eh*Yh; Pyy += Yh*Yh; it goes too large for floating point, which seems to be caused by extremely large values in st->Rf[j], st->Yf[j] and smaller values in st->Eh[j], st->Yh[j]- things were relatively quiet

Hello? I am a new user of speex.I am currently working on speex frames and I have some questions. I am using narrowband and long tail length, and it works very well with speex test DEMO. But it is very difficult to have speaker input in perfect sync with mic input. Speex does not work at all. Any suggestion? Regards? -------------- next part -------------- An HTML attachment was scrubbed...

Hello Jean-Marc and all, I?ve got a problem with speex_echo_cancel usage. I only managed to get an attenuation of 5-8 db using testecho.c and it depends heavily on the playback/capture delay. I used clear speech as far-end input and same speech with a bit of reverb, attenuation and delay as mic nput. I also tried using played and recorded speech for mic input. I tried different constant delays

Hi, For my VoIP application machine A sends speex encoded audio of to machine B and vice versa at. Data is captured in PCM 8Khz, 16 bit and then encoded using speex 1.1.12 The packet A played and the packet A captured through mic are the input to speex echo canceller. So I am trying to remove traces of packet A played from the captured data. I have followed example testecho.c All I hear is some