similar to: AEC stops working in 1.2-rc1?

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.

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

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.

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 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

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

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

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

Hi all, I have been trying to implement audio echo cancellation using mdf.c from opal-2.2.2. Using testecho.c the output obtained is not satisfactory. It is cancelling the far end signal instead of the near end signal as per requirement. I have been moving in circles figuring how to approach the problem. Is there something wrong with the mdf that I am using? I would be really thankful if

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

Hi Ted, Thanks a lot for this analysis. > In FLOAT_DIVU() it hangs at the following: > while (a.m >= b.m) > { > e++; > a.m >>= 1; > } > for the case where a and b are both zero (yes, division by zero). > This happens from mdf.c: True, that needs to be fixed even after I fix the rest. > leak_estimate =

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

Hi, If I pass the same ref and the echo data to the echo cancellation API, I am expecting silence as output. I get back the original audio data. Is this correct? Thanks -Anurag Quoting ac2491@columbia.edu: > Hi Jean, > > I got the earlier problem fied with correct NN and tail values. > But > I dont see any echo being cancelled. To the echo cancel API I am > giving, audio

Jean-Marc, I have traced the second infinite loop further. When st->adapted becomes true (mdf.c line 623), the first Yf[i] value is 4, the leak_estimate is 0xd4e, the resulting r is 3. The first value in st->Rf is 0, so e is 1, and r is set to e>>1, or 0. A little later there is a divide by r, and there is the hang. It seems that the 0 in Rf[0] is the problem, but I am not

Probably the level of your signal is too low and/or you're just not letting it time to adapt. Jean-Marc Le mercredi 19 juillet 2006 ? 19:00 -0400, ac2491@columbia.edu a ?crit : > On closer looks and debugging I always end up in > > speex_echo_cancel function with comment > /* Temporary adaption rate if filter is not adapted correctly */ > > > Does this give any clue

Hi, What is the required size of the echo residue buffer in speex_echo_cancel function? The mdf.c source would suggest that it's frame_size + 1, while preprocessor.c source claims it's frame_size. Bug? Or am I missing something? I am using 1.2beta1. Thanks, Gregg

> 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 am using speex-1.2 beta1 version. When I am using sample files from http://www.surfnonstop.com/~bandit/speex/echocard1/ ,example 6, as input to mdf through testecho the output is fine However when I am recording the signals in echocancel.cxx in OpalEchoCanceler::ReceivedPacket taking speaker signal from echo_buf and mike signal from ref_buf, echo cancellation is not affective. Can