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

[LiMaoquan2000]

Thank you, Andreas Engel.
I downloaded the white paper of the Fraunhofer Acoustic Echo Control.
http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf
It said 
> "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is
> modified so that the undesired echo components are removed from the signal transmitted to
> the far-end. The general approach is illustrated in figure 2."
This modification is a gain multiplication in each frequency instead of frequency domain
subtraction in common frequency domain echo cancellation. See ...
> As can be seen, both the loudspeaker and microphone signals are first transformed into the
> frequency domain by a spectral transform (ST). Based on these input signals, the control unit of
> the Acoustic Echo Control determines an optimum gain factor for each individual frequency
> band separately. These frequency dependent gain factors are also referred to as echo
> attenuation filters. Obviously, the gain factors are selected to be close to zero in circumstances
> where strong echo components need to be removed from the microphone signal. On the other
> hand, in the event of near-end speech only, it is set to one in order to leave the desired speech
> signal unchanged. After applying this echo attenuation filter to the spectral representation of
> the microphone signal, the echo-free signal is transformed back to the time domain by a
> corresponding inverse spectral transform (IST). In typical application scenarios, robust
> attenuation of the echo by 60 dB can be expected and achieved reliably.
So it likes some kinds of a Noice Reducer. It's kernel is not common time- or frequency-domain
adaptive filter. It is not sensitive to tiny frequency shift but the phase of the echo is lost.
So I doubt the quality of voice after processing especially in double talk.
> The calculation of the optimum gain factor is based on an estimate of the power spectrum of
> the echo signal captured by the microphone. The power spectrum of the echo is determined 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
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