[Speex-dev] Speech switching in speakerphone?

[Johan Nilsson]

Hi All

We are working on an ARM-based speaker phone application where the speaker and microphone is placed roughly 8 cm from each other (similar to this one: http://www.voipsupply.com/images/CHAT50SPLASH.jpg).

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.

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). 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.

My question is if anyone have any tips of how to make speex preprocessor work in systems like ours (with speaker and mic close to each other) or if anyone have a reference of success/failure of such implementation.

Best Regards
Johan


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