[Speex-dev] About Sampling Rate Correction in acoustic echo
andreas.engel at bravis.eu
Thu Feb 10 02:21:41 PST 2011
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.
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.
> Thank you, Andreas Engel.
> I downloaded the white paper of the Fraunhofer Acoustic Echo Control.
> 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
> > 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?
> Speex-dev mailing list
> Speex-dev at xiph.org
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