similar to: A few Speex AEC+preprocessor examples

Hi Andras, others, Andras Kadinger <bandit <at> surfnonstop.com> writes: > > > Daniel, > I recommend you to start from a simple case and gradually progress > towards your goal. > Can you make things work with the "Speex in a Disco" (Example 6) > testcase at http://ns.surfnonstop.com/~bandit/speex/echocard1/ ? > These files

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

Hi, Arun and me are working on a echo-canceler module for pulseaudio build around speex. When using the preprocessor's echo filter, specifically in combination with denoise and AGC, it's unclear whether these have to run before or after the AEC. Right now we've got one instance of speex_preprocessor doing AGC and denoise running before the echo canceler; and a second for echo

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

Andras Kadinger wrote: > I have now found a testcase where SVN AGC starts out good, but then it > keeps increasing the gain but never decreasing it, and eventually goes > out through the roof, during the same single recording. I've been doing massive changes in the svn preprocessor, so I'm not too surprised I broke the AGC, because I haven't updated it. Did it also do that in

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

2011.06.21. 19:22 keltez?ssel, Arun Raghavan ?rta: > Hi Andras, others, > > Andras Kadinger<bandit<at> surfnonstop.com> writes: >> >> Daniel, >> I recommend you to start from a simple case and gradually progress >> towards your goal. >> Can you make things work with the "Speex in a Disco" (Example 6) >>

On Tue, 2011-06-21 at 20:18 +0200, Kadinger Andr?s wrote: > 2011.06.21. 19:22 keltez?ssel, Arun Raghavan ?rta: > > Hi Andras, others, > > > > Andras Kadinger<bandit<at> surfnonstop.com> writes: > >> > >> Daniel, > >> I recommend you to start from a simple case and gradually progress > >> towards your goal. >

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

> 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 That's a good start indeed. > However when I am recording the signals in echocancel.cxx in > OpalEchoCanceler::ReceivedPacket taking speaker signal from echo_buf and > mike signal

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

Jean-Marc Valin ?rta: > Andras Kadinger wrote: >> I have now found a testcase where SVN AGC starts out good, but then it >> keeps increasing the gain but never decreasing it, and eventually goes >> out through the roof, during the same single recording. > > I've been doing massive changes in the svn preprocessor, so I'm not too > surprised I broke the AGC,

Perhaps you could add a warm-start to the AEC, such that the parameters start near the correct values on all but the first use? Stuart On 05/25/2011 08:22 AM, Andras Kadinger wrote: > 2011.05.25. 4:52 keltez?ssel, Arun Raghavan ?rta: >> On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: >>> The fact that the AEC takes a few seconds to converge is normal. The >> Do

On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: > The fact that the AEC takes a few seconds to converge is normal. The Do you think there might be a way to reduce this? > fact that it needs to completely re-converge in the middle of a call > probably indicates that something went "wrong" in the audio > capture/playback. For example, that could be an

I'm having trouble with the preprocessor's noise reduction feature. The basic issue is that it simply doesn't work very well. With my laptop (whose microphone is otherwise quite capable) I routinely hear transient background noise, typing, and other "quiet" sounds leaking through to the speex stream. Even worse, the AGC feature is blowing these things up into just awful

agc_gain seemed to fit with the idea of what I wanted to do, it was easy to understand its units and behavior, and freezing it produced the desired results. Also I wanted to cap it, so that's done at the same place, and that definitely works. All I want to do is be able to freeze AGC adaptation and put an upper bound on the AGC (for example, 2x amplification). Both of these things seem

I am trying to use the speex echo canceler and preprocessor with sipXtapi to develop a sip user agent. the sipXmedialib call flow graph uses 10ms frames and I am not sure what the implications are if I try to change this. The documentation seemed to indicate that a 20ms frame was recomended, but it didn't go into the consquences of using other frame sizes. We are going to use headsets

Echo cancellation works like a charm, but it seems to confuse the preprocessor a bit. If listening to background music (properly fed through the echo cancellator), the music is removed but the result is still detected as speech even if almost silence remains in the signal. Also, the AGC keeps adjusting to the minute remains in the signal, meaning that sooner or later it will amplify the

Jean-Marc Valin wrote: > Andy Ross wrote: > > Uh, production applications almost always require squelch, no? > > Some do, some don't. In general, distinguishing between a keyboard > and a speech transient is next to impossible based only on a few ms > of speech. That is true for distinguishing it by waveform, but not by amplitude. As I mentioned, these transients are

> No, the 1.2beta1 preprocessor works rather well. I have a different opinion, but glad you like it anyway :-) > In fact, I very much welcomed the change that the preprocessor in > 1.2beta1 now starts out with the assumption of there being no speech, > and only activating and adapting AGC when where there is new activity > detected above that. As a result, with 1.2beta1, there is