similar to: Speex-dev Digest, Vol 67, Issue 15

Hello, I am using the speex AEC in a real time application. I have experienced that when the mic and the speakers tracks are on sync or with a small delay the AEC works very well. I understood that when they are out of sync the AEC cannot works, so what the "user" should do is to focus in order to have the tracks on sync. Since I am working in an environment where it is not rare to

I tried cross-correlation. It does locate the sync point and it's almost perfect. I guess now I have to make it more perfect. Thanks for the pointers. On the side note: I have a test environment. Where I created the stream such a way that I have one "captured stream" purely with echo (i.e. no one else is speaking) and one stream with echo + someone speaking. I confirmed that I

Is there an API in speex or have you implemented your own? I am currently trying to find one. But if you could point me to one it will reduce my search time. :) Many Thanks, Elston -----Original Message----- From: Marco Pierleoni [mailto:pierleoni.m at gmail.com] Sent: Tuesday, May 11, 2010 5:55 PM To: Elston Sa Cc: speex-dev at xiph.org Subject: Re: [Speex-dev] AEC - Echo is cancelled

The problem is that now I have two files very similar (one for mic, the other for speaker). If I play them together I can hardly say that they are two different file for most of the file length. When I try to cancel the echo, I obtain as output almost the same input. The update_foreground variable is always 0. I don't get the conditions which must be fulfilled in order to have

2010/5/11 Elston Sa <jose at rebaca.com>: > Is there an API in speex or have you implemented your own? I don't know if there is an API in speex for this, probably not. There is a test for AEC which does something very similar. I implemented it starting from a FFT library. It is quite easy to do. > > I am currently trying to find one. But if you could point me to one it will

I usually use the cross-correlation between the speaker signal and the mic signal. It works pretty well. The problem is the you need enough data in order to get a good S/N. I compute the cross-correlation between 1.5 second data of each signals. In this way the application detection of any anomalies is quite slow. I am looking for something faster, but I fear it will be something very application

Maybe you shifted the signal too much. If the mic signal is just one single sample *ahead* of the playback signal, then there's no cancellation possible (because the system is not causal). Be careful with that. Otherwise, if could simply be dropped frames. Jean-Marc On 2009-12-30 07:06, Marco Pierleoni wrote: > The problem is that now I have two files very similar (one for mic, the

Hi, Rhett, echo_diagnostic.m can produce an echo canceled file along with the possible warning messages. Can it cancel the echo properly? It may announce the "Drift estimate is ..." message if the recorded sound contains not only echo but also other sound like a near-end speech. In other cases, it may truly indicate a mismatch of the sampling frequency. Regards Kaiyu On Thu, Feb 6,

Marco, Thank you for your response. Is there a way I can determine if the synchronization is perfect or not? If not by how much it is off. - Elston -----Original Message----- From: Marco Pierleoni [mailto:pierleoni.m at gmail.com] Sent: Monday, May 10, 2010 3:46 PM To: Elston Sa Subject: Re: [Speex-dev] AEC - Echo is cancelled however..... Hi Helton, it is now several months that I am using

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

On Mar 15, 2010, at 8:46 AM, Jean-Marc Valin wrote: > If more than one speaker receives the *same* signal, it doesn't matter the > number of speakers. It only gets tricky when the speakers are playing slightly > different signals (e.g. from a stereo song). > Does "tricky" mean that the Speex AEC won't handle such situations well? Or just that you had to be

>> 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

Well, I am not the person to be asked about how Speex AEC works :) And I was referring speaker input as having direct access into what has been played thru speaker. (ASIO Soundcard provides interface for that) So if I am remebering correctly, with data from Mic and Speaker in sync, speex AEC should do the job. Tim _____ From: Guilherme Balena Versiani [mailto:guibv at

Hi All, I am trying to cross compile speex-1.1.12 to powerpc-405, i get a error after the make, speexec.lo error, please help me how to get rid of this error. On 2/9/07, speex-dev-request@xiph.org <speex-dev-request@xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev@xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

On 2010-03-16 14:22, Josh Gargus wrote: >> If more than one speaker receives the *same* signal, it doesn't >> matter the number of speakers. It only gets tricky when the >> speakers are playing slightly different signals (e.g. from a stereo >> song). > > Does "tricky" mean that the Speex AEC won't handle such situations > well? Or just that you

All I can say for now is that: 1) My diagnostic tool does funny things that need to be fixed 2) It was probably also getting confused by the clicks at the end of the files 3) There seems to be odd things with your recordings, though I can't say what that would be. Are you changing something during the recording by any chance? Note that if a person close to the mic of speaker moves, that

I understand that the capture/playback signals need to be sync'd for an AEC to adapt. I'm a little bit confused on the requirements of synchronous sampling between the near end (mic/speaker) and the far end (phone line). I have an embedded DSP system with mic and speaker getting 1msec packets containing 8 samples. We can watch the DSP and ISDN clock frames drift and every few minutes we

Hi All, I am trying to cross compile speex-1.1.12 to powerpc-405, i get a error after the make, speexec.lo error, please help me how to get rid of this error. On 2/9/07, speex-dev-request@xiph.org <speex-dev-request@xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev@xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

2011/4/20 Li Maoquan <limaoquan2000 at 126.com> > 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

On 2010-03-16 06:35, Greger Burman wrote: > Ok. Thanks J-M for clearing that up. > What if you mix up the polarity on one speaker (180 degree phase > change), would that matter? Not at all. It's still all linear. You can even apply a different equalizer to each speaker and it'll still be linear. Jean-Marc > cheers > Greger > > 2010/3/15 Jean-Marc Valin