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. **********************************************************************