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@@ -0,0 +1,224 @@ + +tooLAME - an optimized mpeg 1/2 layer 2 audio encoder +Copyright (C) 2002, 2003 Michael Cheng [mikecheng at NOT planckenergy.com] remove the NOT +http://www.planckenergy.com/ + +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 +MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +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 + + +********************* +INTRODUCTION +********************* + +tooLAME is an optimized Mpeg Audio 1/2 Layer 2 encoder. It is based heavily on + - the ISO dist10 code + - improvement to algorithms as part of the LAME project (www.sulaco.org/mp3) + - work by myself and other contributors (see CONTRIBUTORS) + +********************* +INSTALLATION +********************* + +1. edit Makefile + at least change the architecture type (ARCH) to suit your machine. +2. 'make' + +********************* +USAGE +********************* + + ./toolame [options] <input> <output> + +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. + + +Operation + -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. + +Misc + -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 + +********************* +EXAMPLES +********************* + +1. + 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) + +2. + 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. + +3. + 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. + +4. + cat sound.pcm | 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) + + +********************* +CONTRIBUTORS +********************* + +Dist10 code writers +LAME specific contributions + fht routines from Ron Mayer <mayer at acuson.com> + fht tweaking by Mathew Hendry <math at vissci.com> + window_subband & filter_subband from LAME circa v3.30 + (multiple LAME authors) + (before Takehiro's window/filter/mdct combination) + +Oliver Lietz <lietz at nanocosmos.de> + Tables now included in the exe! (yay! :) + +Patrick de Smet <pds at telin.rug.ac.be> + scale_factor calc speedup. + subband_quantization speedup + +Federico Grau <grauf at rfa.org> +Bill Eldridge <bill at hk.rfa.org> + option for "no padding" + +Nick Burch <gagravarr at SoftHome.net> + WAV file reading + os/2 Makefile mods. + +Phillipe Jouguet <philippe.jouguet at vdldiffusion.com> + DAB extensions + spelling, LSF using psyII, WAVE reading + +Henrik Herranen - leopold at vlsi.fi + (WAVE reading) + +Andreas Neukoetter - anti at webhome.de + (verbosity patch '-t' switch for transcode plugin) + +Sami Sallinen - sami.sallinen at g-cluster.com + (filter_subband loop unroll + psycho_i fix for "% 1408" calcs) + +Mike Cheng <mikecheng at NOT planckenergy.com> (remove the NOT) + Most of the rest + +********************* +REFERENCE PAPERS +********************* +(Specifically LayerII Papers) + +Kumar, M & Zubair, M., A high performance software implementation of mpeg audio +encoder, 1996, ICASSP Conf Proceedings (I think) + +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 + +Hyen-O et al, New Implementation techniques of a real-time mpeg-2 audio encoding +system. p2287, ICASSP 99. + +Imai, T., et al, MPEG-1 Audio real-time encoding system, IEEE Trans on Consumer +Electronics, v44, n3 1998. p888 + +Teh, D., et al, Efficient bit allocation algorithm for ISO/MPEG audio encoder, +Electronics Letters, v34, n8, p721 + +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 + +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 + +[mikecheng at NOT planckenergy.com] remove the NOT |