TwoLAME

Based on tooLAME by Michael Cheng

All changes to the ISO source are licensed under the LGPL (see COPYING for details)

TwoLAME 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.
TwoLAME 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 TwoLAME; if not, write to the Free Software
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

INTRODUCTION

TwoLAME is an optimized MPEG Audio Layer 2 encoder. It is based heavily on:

For the latest version of TwoLAME, visit the project homepage: http://twolame.sourceforge.net/

INSTALLATION

Standard automate processs:

./configure
make
make install

USAGE

./twolame [options] <input> <output>

Input File

TwoLAME 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 twolame 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.
-E
        Store peak energy information at the end of each frame

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 [int]
        add DAB extensions with an XPAD length as specified.
-t [int]
        'talkativity' setting. 0 = no message. 3 = too much information

EXAMPLES

twolame sound.wav

This will encode sound.wav to sound.mp2 using the default bitrate of 192 kbps and using the default psychoacoustic model (model 3)

twolame -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.

twolame -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.

REFERENCE PAPERS

(Specifically Layer II 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


Last updated 19-Apr-2005 17:39:45 BST