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...atching sample rates across devices. ............. In Windows XP SP1, Windows Server 2003, and later, this limitation does not exist. The AEC system filter correctly handles mismatches between the clocks for the capture and render streams, and separate devices can be used for capture and rendering. Maoquan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20110412/9b10eb0a/attachment.htm

...re the input and output sample rates are known up front. The routine than interpolates between the jitter. This should solve the problem. The crystals used to clock the input and output have very fine tolerances on most standard audio cards. Vas ________________________________________ From: Li Maoquan [limaoquan2000 at 126.com] Sent: Tuesday, April 12, 2011 2:48 PM To: Shridhar, Vasant Cc: speex-dev Subject: Re:RE: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams? Hi Shridhar, Sample rate conversion is not enough to solve this...

...h is omitted in the microsoft paper (Challenges and Solutions for Designing Software AEC on Personal Computers) Let me call is paper-Challenges. The resampling in the paper-Drift is too coarse. Maybe we can combine the RSO detection of paper-Drift to the paper-Challenges? What's your opnion? Maoquan > I don't know if this has only recently been put on line, but I never > noticed it until today - > www.iwaenc.org/proceedings/*2008*/contents/papers/9044.pdf > > That paper is from people at MS describing, in some detail, what the > Windows kernel echo canceller does to hand...

...d. After all, frequency step of sample rate conversion is limited, mismatch is still exist after resampling. Someone told me that capture and render codec have different clock generator which shift independently. And LMS algorithm is very sensitive to the difference between sample rates. Sincerely Maoquan At 2011-04-12 21:46:26?"Shridhar, Vasant" <vasant.shridhar at harman.com> wrote: I would imagine that it is handle through basic asynchronous sample rate conversion. There is a lot of literature out there on the different techniques to do this. A common method is sinc interpol...

...0Hz and ERLE will increase only when frequence difference is very close to 0Hz. These results are under the environment without double talk. Have GIPS and Microsoft some secret high efficient method? They AECs converge very quickly, I could hardly hear any echo in the process. How can they do it? Maoquan > > ---------------------------------------------------------------------- > > On 04/13/2011 02:58 AM, Shridhar, Vasant wrote: > I am doing this right now with no problem. I am not using speex for this at the moment though. Group delay is the biggest problem. I implemented a...

...shown perfect AEC result are the only sound cards > which keep the same sampling and playing rate. > > Then, what can I do to solve the problem? Could you please give me any advice? > BTW: I am working on Windows XP platform and I have tried DirectSound. > > Your Sincerely > Li Maoquan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20100610/f6d18531/attachment.htm

...Noisy timing measurements: Modern audio hardware provides timing data in order to synchronize m[i] and s[i]. The information is always noisy, due to limited numerical precision, data transfer delay, multithreading, etc." m[i] is microphone signal after ADC, s[i] is speaker signal before DAC. Maoquan > > Steve > > On 04/14/2011 07:26 PM, LiMaoquan2000 wrote: > > Hi All, > > Many Thanks to Underwood for her excellent review of our big trouble > > which prevent LMS-based AEC algorithms to be used in most computer. > > Maybe it can be summaried as follows: >...

...ricks, to make the echo perceptually tolerable - an approach which has historically worked pretty well (e.g. the DSP Group solution from the 90s). At least one person reported, on this list, that their solution is the best around. > Vas > ________________________________________ > From: Li Maoquan [limaoquan2000 at 126.com] > Sent: Tuesday, April 12, 2011 2:48 PM > To: Shridhar, Vasant > Cc: speex-dev > Subject: Re:RE: [Speex-dev] Anyone knows how microsoft AEC can deal with mismatches between clocks of capture and render streams? > > Hi Shridhar, > > Sample rate...

...termined by > applying an adaptive estimate of the acoustic echo path to the known power spectrum of the > loudspeaker signal There is still a question. Which algorithm is this adaptive echo power spectrum estimation based on? Is this algorithm not sensitive to frequency difference? Sincerely Maoquan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20110210/a558c816/attachment.htm

...und cards of the computers which shown perfect AEC result are the only sound cards which keep the same sampling and playing rate. Then, what can I do to solve the problem? Could you please give me any advice? BTW: I am working on Windows XP platform and I have tried DirectSound. Your Sincerely Li Maoquan

...you can also find that AC97 controller and all Codecs share the same SYNC and BIT_CLK signal. So anyone could give me more details for the reason of mismatch between sample rates of of capturing and rendering? Which is a HUGE OBSTACLE for all LMS based acoustic echo cancellers, such as speex AEC. Maoquan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20110412/44fd668b/attachment.htm

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

...of the acoustic echo path to the known > power spectrum of the > > loudspeaker signal > > There is still a question. Which algorithm is this adaptive echo power > spectrum estimation based on? > Is this algorithm not sensitive to frequency difference? > > Sincerely > Maoquan > > > _______________________________________________ > Speex-dev mailing list > Speex-dev at xiph.org > http://lists.xiph.org/mailman/listinfo/speex-dev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attac...

On 04/14/2011 07:26 PM, LiMaoquan2000 wrote: > Hi All, > Many Thanks to Underwood for her excellent review of our big trouble > which prevent LMS-based AEC algorithms to be used in most computer. > Maybe it can be summaried as follows: > 1. Different sample rate of sampling and rendering does exists in most > l...

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, yo...

>>>> Hi, >>> >>> I have a scenario in a mobile VoIP app that requires echo cancellation but >>> is somewhat different from what's described in the docs. >>> >>> Audio is received from and sent to the network at 8000Hz. Each packet >>> contains 160 samples worth a playback of 20ms. >>> >>> But the hardware

>> 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