CMU Artificial Intelligence Repository
GSM: Implementation of the 13 kbps GSM speech coding
standard.
areas/speech/systems/gsm/
This directory contains 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/sec.
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, this 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 modeled 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.
Origin:
svr-ftp.eng.cam.ac.uk:/comp.speech/source/ [129.169.24.20]
as the file gsm-1.0.4.tar.Z
Version: 1.0.4 (10-MAY-94)
Requires: C
Copying: Copyright (c) 1992 by Jutta Degener and Carsten Bormann,
Technische Universitaet Berlin
Use, copying, modification, and distribution permitted,
provided the copyright notice remains intact. Please
inform the authors of any novel uses, bugs, and
improvements of general interest.
CD-ROM: Prime Time Freeware for AI, Issue 1-1
Author(s): Jutta Degener
Carsten Bormann
Communications and Operating Systems Research Group
TU Berlin
Fax: +49.30.31425156
Tel: +49.30.31424315
Keywords:
Authors!Bormann, Authors!Degener, C!Code, GSM, Speech Coding,
Speech Coding, Speech Compression
References: ?
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