voice-gateway-service/pjproject/third_party/gsm

GSM 06.10 13 kbit/s RPE/LTP speech compression available
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The Communications and Operating Systems Research Group (KBS) at the
Technische Universitaet Berlin is currently working on a set of
UNIX-based tools for computer-mediated telecooperation that will be
made freely available.

As part of this effort we are publishing an implementation of the
European GSM 06.10 provisional standard for full-rate speech
transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse
excitation/long term prediction) coding at 13 kbit/s.

GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling
rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility
with typical UNIX applications, our implementation turns frames of 160
16-bit linear samples into 33-byte frames (1650 Bytes/s).
The quality of the algorithm is good enough for reliable speaker
recognition; even music often survives transcoding in recognizable
form (given the bandwidth limitations of 8 kHz sampling rate).

The interfaces offered are a front end modelled after compress(1), and
a library API. Compression and decompression run faster than realtime
on most SPARCstations. The implementation has been verified against the
ETSI standard test patterns.

Jutta Degener (jutta@cs.tu-berlin.de)
Carsten Bormann (cabo@cs.tu-berlin.de)

Communications and Operating Systems Research Group, TU Berlin
Fax: +49.30.31425156, Phone: +49.30.31424315

--
Copyright 1992 by Jutta Degener and Carsten Bormann, Technische
Universitaet Berlin. See the accompanying file "COPYRIGHT" for
details. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.