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

[LiMaoquan2000]

>> 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. 
>> 
> That one sounds much more interesting. If you don't have access to IEEE 
> papers, you can find it at the author's university site. They don't do 
> anything extraordinary, but they have thought through how you can track 
> the sample rate offset by LMS, and use a continuously variable rate 
> converter to allow for it. 

Yes. As you said, this is still not a good solution.

> I noticed that the Fraunhofer Institute is now selling a package to  
> address echo cancellation when the tx sample rate cannot be trusted to  
> exactly match the rx sample rate - primarily in VoIP conferencing  
> applications. They say they use the spectral envelope, and disregard the  
> phase. That sounds like its not a million miles from the spectral  
> subtraction a lot of noise suppression schemes use, and those aren't  
> great at getting high levels of suppression. However, they claim very  
> high levels of echo suppression. There must be more to what they do than  
> the blurb indicates. 

> It's relatively easy to getting high levels of echo suppression using   
> spectral subtraction methods. In fact even half-duplex cheap   
> hands-free phones can achieve that. The tricky part is not to distort   
> the "local" voice during double-talk. That's the hard part when you   
> can't rely on an adaptive filter. 

Does anybody know any kind of echo cancellation kernel which is not sensitive to different sampling rates?
At least I don't know.
But there is still a vivid example, AEC in MSN Messager, which is a real AEC, not a echo suppression.
It provides almost perfact echo cancellation even in double talk.
Why?
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