[Speex-dev] About Sampling Rate Correction in acoustic echo cancellation

Andreas Engel andreas.engel at bravis.eu
Wed Feb 9 05:44:43 PST 2011

I recently tested multiple echo cancellation solutions with asynchronous 
audio devices (webcam + loudspeakers). So far Fraunhofer's Echo Control 
was the only one giving good results. Although there might be some 
quality loss during double talk, echo is cancelled very well even with 
very loud playback. Unfortunately it is quite expensive.


Am 19.01.2011 11:44, schrieb LiMaoquan2000:
> Hi all,
> We have discussed so many about sampling rate asynchronous (or offset) 
> between rendering (D/A converter) and capturing (A/D converter) of 
> most PC soundcards. It seems all acoustic echo cancellers, include AEC 
> in speex, can not deal with this trouble, because it causes a drift of 
> echo path and also buffer overflow and underflow which jumps the delay 
> of echo path seriously.
> Unfortunately, this kind of sampling rate asynchronous exists in most 
> low-cost PC soundcards we have. So it is a big obstacle for us to make 
> an AEC algorithm practical.
> I have asked many people for help. It seems impossible to eliminate 
> this offset. Then I found something in microsoft msdn website. It 
> seems microsoft's AEC can deal with different sampling rate.
> > http://msdn.microsoft.com/en-us/library/ff536174%28VS.85%29.aspx
> > In Windows XP, the clock rate must be matched between the capture 
> and render streams. The AEC system filter implements no mechanism for 
> matching sample rates across devices. This limitation precludes using 
> AEC when the capture and render functions are performed by different 
> 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.
> There is also a IEEE paper, Adaptive Sampling Rate Correction for 
> Acoustic Echo Control in Voice-Over-IP, which introduced a complex 
> method to estimate the frequency offset and resynchronize the signals 
> using arbitrary sampling rate conversion. I wonder if it can provide 
> enough performance. Because I have also designed a sampling rate 
> converter. After tested the offset accurately, it can reduce the 
> offset to less than 0.1Hz, then the signal after resampling is send to 
> speex AEC. But there is still hearable echo even if it is far less 
> than that can be heared before resampling.
> Does anybody have any suggestion about practical acoustic echo 
> cancellation in low-cost soundcards? You know, most low-cost 
> soundcards have the problem of sampling rate asynchronous.
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