gsmPPC

Short:        GSM speech compression (PPC), incl. source
Author:       Jutta Degener, Carsten Bormann, Andreas R. Kleinert (port)
Uploader:     Andreas_Kleinert t-online de
Type:         util/pack
Architecture: ppc-powerup

 GSM: lossy speech compression for WWW streaming audio
 -----------------------------------------------------

 This is a port of the GSM 06.10 (Release 1.0 Patchlevel 10)
 lossy speech compression library and the "toast" encoder/decoder
 tool.

 GSM is as "real" as other streaming audio standards, but it's
 free instead. There's already a "audio/x-gsm" MIME type defined
 (see http://itre.ncsu.edu/gsm/) and a GSM Java applet available
 from Vosaic (http://www.vosaic.com).

 There once already has been an ixemul port of GSM for 68k Amigas,
 done by Michael Cheng. The decoder is available under
 Aminet:util/pack/GSMToast.lha while Aminet:comm/tcp/unrealaudio.lha
 shows how to implement a streaming audio GSM mime type with
 Amiga browsers. Then, there's a realtime GSM player from Sinisa
 Kenic, which can be found under Aminet:comm/tcp/Gir#?.lha and
 does include some little tools for IFF conversion plus a small
 "littlegir" plugin for your web browser.

 For more information and further links, take a look at the GSM
 homepage under http://www.cs.tu-berlin.de/~jutta/toast.html


 About the powerUP (TM) PPC port:

  - all the changes have been documented in "src/changes.powerup"
  - there BTW shouldn't be a problem in generating another
    68k version (non-ixemul) with the supplied smakefile by doing
    only some minor adjustments
  - the ELF module can be found in the "bin" directory. To start
    it directly from Shell, make sure to have the ElfLoadSeg
    patch in your startup-sequence and set the "e" protection
    bit on the executable. Otherwise, please use SAS/C's
    RunElf tool for execution
  - in the "lib" directory there's the link library "libgsm.a",
    in case you'd like to add GSM support to your own PPC programs


 For a list of options type:

    RunElf toast.elf -help

--
ARK, 27/Apr/98


**********************************************************************
The original README says about GSM:
**********************************************************************

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

--
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.

**********************************************************************