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-
-tooLAME - an optimized mpeg 1/2 layer 2 audio encoder with
-ODR-DabMux compatible ZeroMQ output and JACK input
-
-Copyright (C) 2002, 2003 Michael Cheng [mikecheng at NOT planckenergy.com] remove the NOT
-http://www.planckenergy.com/
-
-Copyright (C) 2014, 2015 Matthias P. Braendli [matthias.braendli@mpb.li]
-http://opendigitalradio.org/
-
-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 Mike Cheng and other contributors (see CONTRIBUTORS)
-
-It can encode to a file, and also to a zeromq socket that can connect to ODR-DabMux
-See http://opendigitalradio.org
-
-*********************
-INSTALLATION
-*********************
-
-0. install zeromq 4.0.3 or newer, install cmake.
-   optional: install JACK and/or libvlc
-1. mkdir build
-2. cmake ..
-3. 'make'
-
-*********************
-USAGE
-*********************
-
-         ./toolame [options] <input> <output>
-
-Input
-    tooLAME parses AIFF and WAV files for file info
-    raw PCM is assumed if no header is found
-    for stdin use a -
-    for JACK input, use -j option, and specify the name
-    of the JACK port with <input>
-    for the libvlc based input, use -V, and specify the URL
-    with <input>
-
-Output
-    file is automatically renamed from *.* to *.mp2
-    for stdout use a -
-    for zeromq use tcp://<hostname>:<port> pointing to
-        a ODR-DabMux. You can put more than one endpoint,
-        separated by a semicolon. Mind that the shell might
-        interpret the semicolon: use quotes around the list
-        of endpoints to avoid this.
-
-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.
-        Don't use that for DAB encoding.
-
-
-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)
-
-5.
-    toolame -V -s 48 -L -b 128 http://your_stream_url tcp://localhost:9002
-
-    Use libvlc to read and decode the stream at the given url, and encode it at
-    48000Hz sample rate to the ZeroMQ input of ODR-DabMux running on localhost:9002
-
-
-*********************
-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
-
-Matthias P. Braendli <matthias@mpb.li>
-    ZeroMQ output for ODR-DabMux
-    PAD insertion for DLS and slides
-    JACK and libVLC input
-
-*********************
-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
-and
-[matthias.braendli@mpb.li]