home
***
CD-ROM
|
disk
|
FTP
|
other
***
search
/
Dream 52
/
Amiga_Dream_52.iso
/
Amiga
/
Workbench
/
Archivers
/
gsmPPC.lha
/
gsmPPC
/
gsmPPC.readme
< prev
next >
Wrap
Text File
|
1998-04-27
|
4KB
|
93 lines
Short: GSM speech compression (PPC), incl. source
Type: util/pack
Uploader: Andreas_Kleinert@t-online.de
Author: Jutta Degener, Carsten Bormann, Andreas R. Kleinert (port)
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
--
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.
**********************************************************************