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