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authorMatthias P. Braendli <matthias.braendli@mpb.li>2016-02-15 02:44:20 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2016-02-15 02:44:20 +0100
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+<title>TooLAME: MPEG 1/2 Layer II Audio Encoder</title>
+<!-- BODY { BACKGROUND: #FFFFFF; COLOR: #000000; FONT-SIZE: 10pt; FONT-FAMILY: verdana, sans-serif }
+ A { COLOR: #111177; TEXT-DECORATION: none }
+ TD { font-size: medium; font-weight:normal }
+<table border = 0 width="75%" align="center"><tr><td>
+<h2>tooLAME - an optimized mpeg 1/2 layer 2 audio encoder</h2>
+Copyright (C) 2002, 2003 Michael Cheng [mikecheng at NOT planckenergy com] remove the NOT
+<LI> Introduction
+<LI> Usage
+<LI> Examples
+<LI> Contributors
+<LI> References
+All changes to the ISO source are licensed under the LGPL
+(see LGPL.txt for details)
+tooLAME is free software; you can redistribute it and/or
+modify it under the terms of the GNU Lesser General Public
+License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version.
+tooLAME is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+Lesser General Public License for more details.
+You should have received a copy of the GNU Lesser General Public
+License along with tooLAME; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+tooLAME is an optimized Mpeg Audio 1/2 Layer 2 encoder. It is based heavily on
+ <LI>the ISO dist10 code
+ <LI>improvement to algorithms as part of the LAME project (www.sulaco.org/mp3)
+ <LI>work by myself and other contributors (see CONTRIBUTORS)
+<LI> edit Makefile (at least change the architecture type [ARCH] to suit your machine)
+<LI> 'make'
+ ./toolame [options] [input file] [output file]
+Input File
+ tooLAME parses AIFF and WAV files for file info
+ raw PCM is assumed if no header is found
+ for stdin use a -
+Output File
+ file is automatically renamed from *.* to *.mp2
+ for stdout use a -
+Input Options
+ -s [int]
+ if inputting raw PCM sound, you must specify the sample rate
+ default sample rate is 44.1khz.
+ -a
+ downmix from stereo to mono
+ if the incoming file is stereo, combine the audio into
+ a single channel
+ -x
+ force byte-swapping of the input. (current endian detection is dodgy,
+ so if toolame produces only noise, use -x )
+ -g
+ swap the LR channels of a stereo file
+Output Options
+ -m [char]
+ the encoding mode (default 'j')
+ 's' stereo
+ 'd' dual channel
+ 'j' joint stereo
+ 'm' mono
+ -p [int]
+ which psy model to use (default '1')
+ Different models for the psychoacoustics
+ Models: -1 to 4
+ -b [int]
+ the total bitrate
+ For 48/44.1/32kHz default = 192
+ For 24/22.05/16kHz default = 96
+ -v [int]
+ Switch on VBR mode.
+ The higher the number the better the quality.
+ Useful range -10 to 10.
+ See README.VBR for details.
+ -f
+ fast mode turns off calculation of the psychoacoustic model.
+ Instead a set of default values are assumed
+ -q [int]
+ quick mode calculates the psy model every 'num' frames.
+ -d emp
+ de-emphasis (default 'n')
+ -c
+ mark as copyright
+ -o
+ mark as original
+ -e
+ add error protection
+ -r
+ force padding bits off
+ -D
+ add DAB extensions
+ -t [int]
+ 'talkativity' setting. 0 = no message. 3 = too much information
+ (-t 20 will probably flood you off your terminal)
+ toolame sound.wav
+ This will encode sound.wav to sound.mp2 using the default bitrate of 192 kbps
+ and using the default psychoacoustic model (model 1)</p>
+ toolame -p 2 -v 5 sound.wav newfile.mp2
+ Encode sound.wav to newfile.mp2 using psychoacoustic model 2 and encoding
+ with variable bitrate. The high value of the "-v" argument means that
+ the encoding will tend to favour higher bitrates.</p>
+ toolame -p 2 -v -5 sound.wav newfile.mp2
+ Same as example above, except that the negative value of the "-v" argument
+ means that the lower bitrates will be favoured over the higher ones.</p>
+ cat sound.raw | toolame -s 22050 -f -b 96 - newfile.mp2
+ Toolame is encoding from stdin at a bitrate of 96kbps and is using the
+ 'fast' mode which means that no psychoacoustic modelling is done. The
+ input file is raw pcm so the sample rate needs to be specified (22050Hz)</p>
+<LI>Dist10 code writers
+<LI>LAME specific contributions
+ <LI>fht routines from Ron Mayer mayer at acuson.com
+ <LI>fht tweaking by Mathew Hendry math at vissci.com
+ <LI>window_subband & filter_subband from LAME circa v3.30
+ (multiple LAME authors)
+ (before Takehiro's window/filter/mdct combination)
+<LI> Oliver Lietz - lietz at nanocosmos.de - Tables included in the exe
+<LI> Patrick de Smet - pds at telin.rug.ac.be - scale_factor calc speedup. subband_quantization speedup
+<LI> Federico Grau - grauf at rfa.org - and Bill Eldridge - bill at hk.rfa.org - option for "no padding"
+<LI> Nick Burch - gagravarr at SoftHome.net - WAV file reading, os/2 Makefile mods.
+<LI> Phillipe Jouguet - philippe.jouguet at vdldiffusion.com - DAB extensions. spelling, LSF using psyII, WAVE reading [02j]
+<LI> Henrik Herranen - leopold at vlsi.fi - fixed WAVE reading [02j]
+<LI> Andreas Neukoetter - anti at webhome.de - verbosity patch '-t' switch for transcode plugin [02j]
+<LI> Sami Sallinen - sami.sallinen at g-cluster.com - filter_subband loop unroll, psycho_i fix for "% 1408" calcs [02j]
+<LI> Ricardo Schelp - ricardoschelp at arnet.com.ar - merged window/filter subband for a nice speedup [02k]
+<LI> Nicolas Croiset - ncroiset at vdl.fr - DAB length control, ignore 4GB limit when reading from stdin, fixed bitstream ending to allow concatenation of mp2 files, fixes for psycho1 model [02k]
+<LI> Mike Cheng mikecheng at NOT planckenergy.com - Most of the rest
+Kumar, M & Zubair, M., A high performance software implementation of mpeg audio
+encoder, 1996, ICASSP Conf Proceedings (I think)</p>
+Fischer, K.A., Calculation of the psychoacoustic simultaneous masked threshold
+based on MPEG/Audio Encoder Model One, ICSI Technical Report, 1997
+ftp://ftp.icsi.berkeley.edu/pub/real/kyrill/PsychoMpegOne.tar.Z </p>
+Hyen-O et al, New Implementation techniques of a real-time mpeg-2 audio encoding
+system. p2287, ICASSP 99.</p>
+Imai, T., et al, MPEG-1 Audio real-time encoding system, IEEE Trans on Consumer
+Electronics, v44, n3 1998. p888</p>
+Teh, D., et al, Efficient bit allocation algorithm for ISO/MPEG audio encoder,
+Electronics Letters, v34, n8, p721</p>
+Murphy, C & Anandakumar, K, Real-time MPEG-1 audio coding and decoding on a DSP
+Chip, IEEE Trans on Consumer Electronics, v43, n1, 1997 p40</p>
+Hans, M & Bhaskaran, V., A compliant MPEG-1 layer II audio decoder with 16-B
+arithmetic operations, IEEE Signal Proc Letters v4 n5 1997 p121</p>
+[mikecheng at NOT planckenergy com] remove the NOT</p>