similar to: Visualizing echo

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

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 am writing to gain a better understanding of the limitations of speex echo > cancellation, esp. with respect to the fixed point implementation. > If these limitations have been documented elsewhere already, please let me > know! Nothing officially documented, sorry. > I observe experimentally that when one or both of the echo or ref data for > speex_echo_cancel() have

Hi I have a question about the AEC and preprocessor. I have seen that when near end is talking for a longer time (about 10 s) without being interrupted by the far end the residual echo (or rather leak_estimate) increases making the preprocessor suppress the near end talk when it shouldn't. Is there a way to make the leakage estimate only update when far-end is present or similar in order to

Hi, Here are the next two patches for the data export. speex_get_psd should be applied after speex_get_agc_gain (sent in previous mail). It allows applications to get the power spectrum for the signal and the noise estimate. speex_get_prob should be applied last. It allows fetching the speech probability of the current frame (the value that the _PROB_START and _PROB_CONTINUE parameters are

hi List and Manuel, I have encounter the following problem with the function "lineplot.CI".? I'm running R 2.10.1, sciplot 1.0-7 on Win XP.? It seems like it's a scoping issue, but I couldn't figure it out. Thanks! ...Tao > lineplot.CI(x.factor = dose, response = len, data = ToothGrowth)??? ## fine > lineplot.CI(x.factor = dose, response = len, data = ToothGrowth,

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

Hi, Ref the disucussion on IRC yesterday; here's a patch which makes a bit more data from the analysis of the preprocessor and the echo canceller available. For the preprocessor: - Size of power spectrum. - Power spectrum and noise estimate of the previous frame. These are given as squared values, so sqrt() to get values in the 0->32767 range. - Current amplification level

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 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 think you'll have to modify Speex to get the functionality you're looking for. I've made a few simple modifications to the AGC to prevent it from 1) exceeding a specified level of amplification and 2) enable and disable adaptation, so I can freeze it at a certain level while speech is not detected. It's mostly just a matter of doing this at the end of speex_compute_agc():

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

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, I'm new. I tried to search out this answer but I suspect I was using the wrong terms, or simply not understanding some of the answers. Anyway here is my question: I want to have a 2x2 panel figure with 4 line graphs all in the same scale. Actually I have that. The thing I seem to be lacking is a way to Label each panel with a letter. I want it to look something like this:

I'm having with an echo or delay I connect to the PSTN with a x100p and then connect a std. phone to a FXS module on a TDM10B. The std phone is only 2-wire so I know this is not helping. (yes I have read the 2-wire 4-wire issue) I have tried many echocancel values. The best thing to help was rxgain and txgain. below is my current zapata.conf file All help would be grateful. I have tried

Hi: I'm using speex to perform echo cancellation in Windows. I'm aware of the problem about out of sync clocks in record and play sample rates in usual sound cards . In order to have an idea of how good is my echo cancelation working I would like to know if there is any #define thing i can pass to speex_echo_ctl to get the actual level of echo cancellation. If not, how can i extract that

Hi everyone, I've just released Speex 1.1.9. The main change in this release is the echo canceller work sponsored by Tipic Inc (http://www.tipic.com/). It is now possible to do acoustic echo cancellation and obtain good attenuation after a short adaptation time. This has been tested at 8 kHz, but it should also work at 16 khz and above, so give it a try. There were also some fixes to the

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 >

Attached is a snapshot of work-in-progress of a FFT based resampler. At the moment it works in floating point only, and only basic quality inspection has been done. Some benchmarks comparing the filter-based resampler at Q3 with the FFT resampler with overlap = in_len / 2, using 20ms chunks of data. (-O3 -ffast-math, FFTW3, gcc 4.3.0 on x86_64) 16=>48: 59us vs 19us 16=>44.1: 204us vs

On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > Alexander Chemeris wrote: > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > > > I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, > > > and there aren't any obvious artifacts. I also did a 16=>16 test, and the > > > results are delayed