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I can only evaluate this with my subjective point of view. I had a
special test scenario doing chat with cheap webcam microphones and
loudspeakers. Fraunhofers solution was the only one that could
eliminate the echo. In double talk the quality gets lower but is
still very good. You might want to ask Fraunhofer for a demo version
to test for yourself.<br>
I have no details on the algorithms being used, I only know that it
is patented. Since it is a commercial product I doubt they are
willing to share any more details than they already do in that
paper. But the algorithm is obviously very insensitive to frequency
drift.<br>
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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 moz-do-not-send="true"
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>
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