[Speex-dev] Speech switching in speakerphone?

[Jean-Marc Valin]

Johan Nilsson a écrit :
> We are using Speex AEC and preprocessor to handle the acoustic echo. The
> AEC typically contribute with ERLE of 15-20 dB which should be as
> expected from the algorithm. Additionally we need about 30 dB echo
> suppression which we hope to get from the preprocessor.

So far it seems like you're doing things right.

> However it has shown to be very hard to make the speech switching
> reliable under situations where the far-end signal is very strong
> compared to the near-end signal in the mic signal (rec). 

Can you explain what you mean here by "speech switching" and problem
you've encountered?

> We have
> converted the AEC and preprocessor code to matlab and are able to look
> at all signals. We can modify the Qcurve-function to bring more
> attention to the near-end signal or we can modify the echo_noise
> estimation (add gain and accumulation) to bring more attention to the
> residual echo suppression. However it is hard to optimize for both.

There's also a parameter to control the maximum amount of suppression
allowed:
SPEEX_PREPROCESS_SET_NOISE_SUPPRESS : noise suppression
SPEEX_PREPROCESS_SET_ECHO_SUPPRESS : echo suppression when there is no
local talk
SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE: echo suppression in double-talk


	Jean-Marc