similar to: Acoustic echo cancellation

The echo suppression in SVN is shaping up quite nicely! I have also observed some - occasional, apparently data-dependent - misbehaviour in AGC. http://www.surfnonstop.com/~bandit/speex/echocard1/

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. >

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) >>

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

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

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

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

Simply to say, in a quiet room, you can play a impulse signal and then find it's impulse response signal from the microphone. For example, if the delay between the impulse signal and its response signal range from 500 to 3000 cycles, you can buffer the far-end signal to 0-300 cycles and set the filter length to 4000. It is also called to align far-end signal and near-end signal. BTW: Speex

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

2011.06.21. 22:57 keltez?ssel, Arun Raghavan ?rta: > On Tue, 2011-06-21 at 11:39 -0700, Arun Raghavan wrote: > [...] >> I'm also running this on x86 (x86_64, technically), and it's all >> floating-point, so I guess this is a regression somewhere. Will try to >> see if I can run it without any optimisations if possible, which I >> assume should serve as an

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

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

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

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

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 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, Andras, Thanks for the comments. Yes, I am aware of those issues. I probably should have been more accurate on my usage of terms. Actually in my project, the unit collection is a mixture of diphones and words. However seems to me, these synthesizer specific issue is irrelevant to my question about speex. As you said, i merely use speex as storage methods. All I ask for is to get the

Ok, let me first explain why 5ms matters, even they are 0's, in my particular application. I am working on a speech synthesis system. The basic idea is concatenating pre-recorded phonemes or words into longer sentences. So any missing or extra samples, even it is as short as 5~10ms, cause very noticeable discontinuities. I want to use speex to compress/decompress those pre-recorded

Hi All, Does anyone know, if there is a tool, which is doing the follwing: - Testprogram on host A establishes a sip connection to testprogramm on host B - Testprogram on host A plays a tone and Testprogram B verifies, if tone is playing correctly (without any interruptions) Thank You.

On 06/22/2011 04:57 AM, Arun Raghavan wrote: > On Tue, 2011-06-21 at 11:39 -0700, Arun Raghavan wrote: > [...] >> I'm also running this on x86 (x86_64, technically), and it's all >> floating-point, so I guess this is a regression somewhere. Will try to >> see if I can run it without any optimisations if possible, which I >> assume should serve as an adequate