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- From: simon@lia.di.epfl.ch (Simon Leinen)
- Newsgroups: comp.speech,alt.binaries.sounds.d
- Subject: Re: Looking for GSM 06.10 spec/implementation
- Message-ID: <SIMON.92Nov8214511@liasg2.epfl.ch>
- Date: 8 Nov 92 20:45:11 GMT
- References: <1992Nov8.194321.26097@CSD-NewsHost.Stanford.EDU>
- Sender: news@disuns2.epfl.ch
- Followup-To: comp.speech
- Organization: DI-LIA -- Ecole Polytechnique Federale de Lausanne
- Lines: 55
- Nntp-Posting-Host: liasg2.epfl.ch
- In-reply-to: minakami@Xenon.Stanford.EDU's message of 8 Nov 92 19:43:21 GMT
- X-Md4-Signature: d8f3c99e953539231960ae0e34f4fa90
-
- minakami@Xenon.Stanford.EDU (Michael K. Minakami) writes:
-
- Hi, does anyone out there have the GSM 06.10 specification in
- softcopy, or alternatively some C code implementing it?
-
- An implementation can be FTPed from
- tub.cs.tu-berlin.de:/pub/tubmik/gsm-1.0.tar.Z
- +/pub/tubmik/gsm-1.0-patch1
-
- or as a faster but not always up-to-date alternative:
- liasun3.epfl.ch:/pub/audio/gsm-1.0pl1.tar.Z
-
- See the README file after the signature.
- --
- Simon Leinen.
- Laboratoire d'Intelligence Artificielle
- Ecole Polytechnique Federale de Lausanne
-
- /*
- * 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.
- */
-
- 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.
-
- 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
-