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Yes. See the introduction to MPEG given in part 2 of this FAQ.
A lossless compressor for 8bit and 16bit audio data (.au) is available
in ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/shorten.tar.gz
or http://www.softsound.com/ShortenDownload.html
Shorten works by using Huffman coding of prediction residuals.
Compression is generally better than that obtained by applying general
purpose compression utilities to audio files. Also supports lossy
compression. Contact: Tony Robinson <ajr@eng.cam.ac.uk>.
Benchmarks of shorten and other lossless audio compression programs
are in http://www.firstpr.com.au/audiocomp/lossless/
Audio software is available in subdirectories of
ftp://sunsite.unc.edu/pub/electronic-publications/IUMA/audio_utils/ :
- An MPEG audio player is in mpeg_players/Workstations/maplay1_2.tar.Z.
- The sources of the XING MPEG audio player for Windows is in
mpeg_players/Windows/mpgaudio.zip.
- An encoder/decoder is in converters/source/mpegaudio.tar.Z.
MSDOS audio software is available in
ftp://ftp.simtel.net/pub/simtelnet/msdos/sound/
In particular, MPEG-2 audio software is in ampegsrc.zip and ampeg43.zip.
MPEG audio files are available in ftp://ftp.iuma.com and http://www.iuma.com/
The site http://www.mp3tech.com is dedicated to the MP3 audio compression
standard. It has information about the MP3 standard, audio compression
techniques, tests, sources, etc...
Copied from the comp.dsp FAQ posted by guido@cwi.nl (Guido van Rossum):
Strange though it seems, audio data is remarkably hard to compress
effectively. For 8-bit data, a Huffman encoding of the deltas between
successive samples is relatively successful. For 16-bit data,
companies like Sony and Philips have spent millions to develop
proprietary schemes.
Public standards for voice compression are slowly gaining popularity,
e.g. CCITT G.721 and G.723 (ADPCM at 32 and 24 kbits/sec). (ADPCM ==
Adaptive Delta Pulse Code Modulation.) Free source code for a *fast*
32 kbits/sec ADPCM (lossy) algorithm is available by ftp from ftp.cwi.nl
as /pub/audio/adpcm.shar. (** NOTE: if you are using v1.0, you should get
v1.1, released 17-Dec-1992, which fixes a serious bug -- the quality
of v1.1 is claimed to be better than uLAW **)
(Note that U-LAW and silence detection can also be considered
compression schemes.)
Information and source code for adpcm are available in
http://www.rss.rockwell.com/techinfo/pc/adpcm/adpcm.html
Source for Sun's free implementation of CCITT compression types G.711,
G.721 and G.723 is in ftp://ftp.cwi.nl/pub/audio/ccitt-adpcm.tar.gz
You can get a G.721/722/723 package by email to teledoc@itu.arcom.ch, with
GET ITU-3022
as the *only* line in the body of the message.
A note on u-law from Markus Kuhn <mskuhn@immd4.informatik.uni-erlangen.de>:
u-law (more precisely (greek mu)-law or 5-law if you have an 8-bit
ISO terminal) is more an encoding then a compression method,
although a 12 to 8 bit reduction is normally part of the encoding.
The official definition is CCITT recommendation G.711. If you want
to know how to get CCITT documents, check the Standards FAQ
posted to news.answers or get the file standards-faq by ftp in
directory ftp://rtfm.mit.edu/pub/usenet/news.answers/
See also the comp.dsp FAQ for more information on:
- The U.S. DoD's Federal-Standard-1016 based 4800 bps code excited linear
prediction voice coder version 3.2a (CELP 3.2a)
- The U.S. DoD's Federal-Standard-1015/NATO-STANAG-4198 based 2400 bps
linear prediction coder version 53 (LPC-10e v53)
- Realtime DSP code and hardware for FS-1015 and FS-1016
The comp.dsp FAQ is in comp.dsp with subject "FAQ: Audio File Formats" and in
ftp://rtfm.mit.edu/pub/usenet/news.answers/audio-fmts/
CELP C code for Sun SPARCs is in ftp://ftp.super.org/pub/speech/
An LPC10 speech coder is in ftp://ftp.super.org/pub/speech/ ;
a derived version is available from http://www.arl.wustl.edu/~jaf/lpc/
Source code for ITU-T (CCITT) G.728 Low Delay CELP speech compression
is in ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources/
Recommended reading:
Digital Coding of Waveforms: Principles and Applications to Speech and
Video. N. S. Jayant and Peter Noll. Prentice-Hall, 1984, ISBN
0-13-211913-7.
Information on GSM sound compression is available at
http://ccnga.uwaterloo.ca/~jscouria/gsm.html
from Markus Kuhn <mskuhn@immd4.informatik.uni-erlangen.de>:
One highest quality sound compression format is called ASPEC and has
been developed by a team at the Frauenhofer Institut in Erlangen (Germany)
and others.
ASPEC produces CD like quality and offers several bitrates, one is
128 kbit/s. It is a lossy algorithm that throws away frequencies that
aren't registered in the human cochlea in addition to sophisticated
entropy coding. The 64 kbit/s ASPEC variant might soon bring hifi
quality ISDN phone connections. It has been implemented on standard DSPs.
The Layer 3 MPEG audio compression standard now contains what is officially
called the best parts of the ASPEC and MUSICAM algorithms. A reference is:
K.Brandenburg, G.Stoll, Y.F.Dehery, J.D.Johnston, L.v.d.Kerkhof,
E.F.Schroeder: "The ISO/MPEG-Audio Codec: A Generic Standard for Coding
of High Quality Digital Audio",
92nd. AES-convention, Vienna 1992, preprint 3336
from Jutta Degener <jutta@cs.tu-berlin.de> and Carsten Bormann
<cabo@cs.tu-berlin.de>:
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).
Version 1.0 of the implementation is available per anonymous ftp from
ftp.cs.tu-berlin.de in the directory /pub/local/kbs/tubmik/gsm/ ;
more information about the library can be found on the World-Wide Web
at http://www.cs.tu-berlin.de/~jutta/toast.html .
Questions and bug reports should be directed to jutta@cs.tu-berlin.de
and cabo@informatik.uni-bremen.de .
from Nicola Ferioli <ser1509@cdc835.cdc.polimi.it>:
ftp://ftp.simtel.net/pub/simtelnet/msdos/sound/vocpak20.zip
Lossless 8-bit sound file compressor
VOCPACK is a compressor/decompressor for 8-bit digital sound using a
lossless algorithm; it is useful to save disk space without degrading
sound quality. It can compress signed and unsigned data, sampled at any
rate, mono or stereo. Since the method used is not lossy, it isn't
necessary to strip file headers before compressing.
VOCPACK was developed for use with .VOC (SoundBlaster) and .WAV (Windows)
files, but any 8-bit sound can be compressed since the program takes no
assumptions about the file structure.
The typical compression ratio obtained goes from 0,8 for files sampled at
11 KHz to 0,4 for 44 Khz files. The best results are obtained with 44 KHz
sounds (mono or stereo): general-purpose archivers create files that can be
twice longer than the output of VOCPACK. You can obtain smaller values
using lossy compressors but if your goal is to keep the sound quality
unaltered you should use a lossless program like VOCPACK.
from Harald Popp <popp@iis.fhg.de>:
new version 1.0 of ISO/MPEG1 Audio Layer 3 Shareware available
major improvements of the new version:
- encoder works twice as fast
- improved file handling for encoder including .WAV files
You may download the shareware from fhginfo.fhg.de (153.96.1.4)
from the directory /pub/layer3
The source code for the MPEG1 audio decoder layer 1, 2 and 3 is
now available on fhginfo.fhg.de (153.96.1.4) in /pub/layer3/public_c.
There are two files:
mpeg1_iis.tar.Z (Unix: lines seperated by line feed only)
mpeg1iis.zip (PC: lines seperated by carriage return and line feed)
For more information about this product and MPEG Audio Layer 3, see
the document "Informations about MPEG Audio Layer-3" maintained by
Juergen Zeller <zeller@iis.fhg.de>, available in
ftp://fhginfo.fhg.de/pub/layer3/
from Monty <xiphmont@athena.mit.edu>:
OggSquish is a compression package designed to reduce the file size of
digitized 8 and 16 bit audio samples (or samples of any periodic
data). OggSquish will operate on files sampled at any speed, but it is
designed to work with very high quality samples, for example, CD
quality samples.
[OggSquish is now at http://www.xiph.com/OggSquish/index.html or
http://world.std.com/~xiph/OggSquish/ or
http://web.mit.edu/afs/sipb/user/xiphmont/OggSquish/html-pages/ ]
from Dmitrij V. Schmunk <shmunk@csd.inp.nsk.su>:
Take a look at http://www.inp.nsk.su/~shmunk/
This compressor gives you about 2-3 times better compression
for 44.1kHz stereo sound then MPEG layer-3.
from Dennis Lee <denlee@ecf.utoronto.ca>
WA incorporates lossless audio codecs (similar to SHORTEN and OggSquish)
into an easy to use archiver program. WA only supports compression of the
popular ".WAV" audio format. To the author's knowledge, WA can compress
waveform data better than any existing software. With default settings, WA
also compresses faster than PKZIP, so it is convenient to use. This
software can be found at http://www.ecf.utoronto.ca/~denlee/software.html
from Jack Berlin <jberlin@jpg.com>:
Pegasus has a new lossless sound compressor based on our patent pending
arithmetic coder. Currently beats Shorten in all cases. Trial app for
Windows: ftp://207.69.208.43/jpg.com/SPSEXE.ZIP ($39 to register)
http://www.pegasusimaging.com/sound.html
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Last Update May 13 2007 @ 00:22 AM