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Archive-name: auto/comp.speech/GSM-06-10-13-kbit-s-RPE-LTP-speech-compression-available

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.

Version 1.0 of the implementation is available per anonymous ftp from
tub.cs.tu-berlin.de as /pub/tubmik/gsm-1.0.tar.Z.  Questions and bug
reports should be directed to toast@tub.cs.tu-berlin.de.  
Note that the distribution is not available via E-mail (please use one
of the ftp-via-E-mail servers).

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