questzdoom/Projects/Android/jni/SupportLibs/timidity/lame/USAGE

% lame [options] inputfile [outputfile]

For more options, just type:
% lame --help


=======================================================================
Constant Bitrate Examples:
=======================================================================
fixed bit rate jstereo 128 kbps encoding:
% lame sample.wav  sample.mp3      

fixed bit rate jstereo 128 kbps encoding, higher quality:  (recommended)
% lame -h sample.wav  sample.mp3      

Fast encode, low quality  (no noise shaping)
% lame -f sample.wav  sample.mp3     

=======================================================================
Variable Bitrate Examples:
=======================================================================
LAME has two types of variable bitrate: ABR and VBR.

ABR is the type of variable bitrate encoding usually found in other
MP3 encoders, Vorbis and AAC.  The number of bits is determined by
some metric (like perceptual entropy, or just the number of bits
needed for a certain set of encoding tables), and it is not based on
computing the actual encoding/quantization error.  ABR should always
give results equal or better than CBR:

ABR:   (--abr <x> means encode with an average bitrate of around x kbps)
lame -h --abr 128  sample.wav sample.mp3


VBR is a true variable bitrate mode which bases the number of bits for
each frame on the measured quantization error relative to the
estimated allowed masking. There are 10 compression levels defined, 
ranging from 0=lowest compression to 9 highest compression. The resulting
filesizes depend on the input material. On typical music you can expect
-V5 resulting in files averaging 132 kbps, -V2 averaging 200 kbps.

Variable Bitrate (VBR): (use -V n to adjust quality/filesize)
% lame -V2 sample.wav sample.mp3



=======================================================================
LOW BITRATES
=======================================================================
At lower bitrates, (like 24 kbps per channel), it is recommended that
you use a 16 kHz sampling rate combined with lowpass filtering.  LAME,
as well as commercial encoders (FhG, Xing) will do this automatically.
However, if you feel there is too much (or not enough) lowpass
filtering, you may need to try different values of the lowpass cutoff
and passband width (--resample, --lowpass and --lowpass-width options).


=======================================================================
STREAMING EXAMPLES
=======================================================================

% cat inputfile | lame [options] - - > output




=======================================================================
Scripts are included (in the 'misc' subdirectory)
to run lame on multiple files:

bach script:  mlame     Run "mlame -?" for instructions.
sh script:    auenc     Run auenc for instructions
sh script:    mugeco.sh

Pearl script which will re-encode mp3 files and preserve id3 tags:
lameid3.pl 

Windows scripts:
lame4dos.bat  
Lame.vbs   (and an HTML frontend: LameGUI.html)


=======================================================================
options guide:
=======================================================================
These options are explained in detail below.


Quality related:

-m m/s/j/f/a   mode selection
-q n           Internal algorithm quality setting 0..9. 
               0 = slowest algorithms, but potentially highest quality
               9 = faster algorithms, very poor quality
-h             same as -q2
-f             same as -q7


Constant Bit Rate (CBR)
-b  n          set bitrate (8, 16, 24, ..., 320)
--freeformat   produce a free format bitstream.  User must also specify
               a bitrate with -b, between 8 and 640 kbps.

Variable Bit Rate (VBR)
-v             VBR
--vbr-old      use old variable bitrate (VBR) routine
--vbr-new      use new variable bitrate (VBR) routine (default)
-V n           VBR quality setting  (0=highest quality, 9=lowest)
-b  n          specify a minimum allowed bitrate (8,16,24,...,320)
-B  n          specify a maximum allowed bitrate (8,16,24,...,320)
-F             strictly enforce minimum bitrate
-t             disable VBR informational tag 
--nohist       disable display of VBR bitrate histogram

--abr n        specify average bitrate desired


Operational:

-r              assume input file is raw PCM
-s  n           input sampling frequency in kHz (for raw PCM input files)
--resample n    output sampling frequency
--mp3input      input file is an MP3 file.  decode using mpglib/mpg123
--ogginput      input file is an Ogg Vorbis file.  decode using libvorbis
-x              swap bytes of input file
--scale <arg>   multiply PCM input by <arg>
--scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg>
--scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg>
-a              downmix stereo input file to mono .mp3
-e  n/5/c       de-emphasis
-p              add CRC error protection
-c              mark the encoded file as copyrighted
-o              mark the encoded file as a copy
-S              don't print progress report, VBR histogram
--strictly-enforce-ISO   comply as much as possible to ISO MPEG spec
--replaygain-fast   compute RG fast but slightly inaccurately (default)
--replaygain-accurate   compute RG more accurately and find the peak sample
--noreplaygain  disable ReplayGain analysis
--clipdetect    enable --replaygain-accurate and print a message whether
                clipping occurs and how far the waveform is from full scale

--decode        assume input file is an mp3 file, and decode to wav.
-t              disable writing of WAV header when using --decode
                (decode to raw pcm, native endian format (use -x to swap))



ID3 tagging:

--tt <title>    audio/song title (max 30 chars for version 1 tag)
--ta <artist>   audio/song artist (max 30 chars for version 1 tag)
--tl <album>    audio/song album (max 30 chars for version 1 tag)
--ty <year>     audio/song year of issue (1 to 9999)
--tc <comment>  user-defined text (max 30 chars for v1 tag, 28 for v1.1)
--tn <track>    audio/song track number (1 to 255, creates v1.1 tag)
--tg <genre>    audio/song genre (name or number in list)
--add-id3v2     force addition of version 2 tag
--id3v1-only    add only a version 1 tag
--id3v2-only    add only a version 2 tag
--space-id3v1   pad version 1 tag with spaces instead of nulls
--pad-id3v2     same as '--pad-id3v2-size 128'
--pad-id3v2-size <num> adds version 2 tag, pad with extra <num> bytes
--genre-list    print alphabetically sorted ID3 genre list and exit

Note: A version 2 tag will NOT be added unless one of the input fields
won't fit in a version 1 tag (e.g. the title string is longer than 30
characters), or the '--add-id3v2' or '--id3v2-only' options are used,
or output is redirected to stdout.

Windows and OS/2-specific options:
    --priority <type>     sets the process priority


options not yet described:
--nores            disable bit reservoir
--disptime

--lowpass
--lowpass-width
--highpass
--highpass-width





=======================================================================
Detailed description of all options in alphabetical order
=======================================================================


=======================================================================
downmix
=======================================================================
-a  

mix the stereo input file to mono and encode as mono.  

This option is only needed in the case of raw PCM stereo input 
(because LAME cannot determine the number of channels in the input file).
To encode a stereo PCM input file as mono, use "lame -m s -a"

For WAV and AIFF input files, using "-m m" will always produce a
mono .mp3 file from both mono and stereo input.


=======================================================================
average bitrate encoding (aka Safe VBR)
=======================================================================
--abr n

turns on encoding with a targeted average bitrate of n kbps, allowing
to use frames of different sizes.  The allowed range of n is 8...320 
kbps, you can use any integer value within that range.





=======================================================================
bitrate
=======================================================================
-b  n

For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz)
n =   32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320

For MPEG-2 (sampling frequencies of 16, 22.05 and 24 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160

For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64


The bitrate to be used.  Default is 128 kbps MPEG1, 80 kbps MPEG2.

When used with variable bitrate encodings (VBR), -b specifies the
minimum bitrate to use.  This is useful only if you need to circumvent
a buggy hardware device with strange bitrate constrains.


=======================================================================
max bitrate
=======================================================================
-B  n

see also option "-b" for allowed bitrates.

Maximum allowed bitrate when using VBR/ABR.

Using -B is NOT RECOMMENDED.  A 128 kbps CBR bitstream, because of the
bit reservoir, can actually have frames which use as many bits as a
320 kbps frame.  ABR/VBR modes minimize the use of the bit reservoir, and
thus need to allow 320 kbps frames to get the same flexability as CBR
streams.  This is useful only if you need to circumvent a buggy hardware
device with strange bitrate constrains.




=======================================================================
copyright
=======================================================================
-c   

mark the encoded file as copyrighted



=======================================================================
clipping detection
=======================================================================
--clipdetect

Enable --replaygain-accurate and print a message whether clipping 
occurs and how far in dB the waveform is from full scale.

This option is not usable if the MP3 decoder was _explicitly_ disabled 
in the build of LAME.

See also: --replaygain-accurate



=======================================================================
mpglib decode capability
=======================================================================
--decode 

This just uses LAME's mpg123/mpglib interface to decode an MP3 file to
a wav file.  The input file can be any input type supported by
encoding, including .mp3 (layers 1, 2 and 3) and .ogg.  

If -t is used (disable wav header), LAME will output
raw pcm in native endian format (use -x to swap bytes).

This option is not usable if the MP3 decoder was _explicitly_ disabled
in the build of LAME.


=======================================================================
de-emphasis
=======================================================================
-e  n/5/c   

  n = (none, default)
  5 = 0/15 microseconds
  c = citt j.17

All this does is set a flag in the bitstream.  If you have a PCM
input file where one of the above types of (obsolete) emphasis has
been applied, you can set this flag in LAME.  Then the mp3 decoder
should de-emphasize the output during playback, although most 
decoders ignore this flag.

A better solution would be to apply the de-emphasis with a standalone
utility before encoding, and then encode without -e.  



=======================================================================
fast mode
=======================================================================
-f   

Same as -q 7.  

NOT RECOMMENDED.  Use when encoding speed is critical and encoding
quality does not matter.  Disable noise shaping.  Psycho acoustics are
used only for bit allocation and pre-echo detection.

=======================================================================
strictly enforce VBR minimum bitrate
=======================================================================
-F   

strictly enforce VBR minimum bitrate.  With out this optioni, the minimum
bitrate will be ignored for passages of analog silence.



=======================================================================
free format bitstreams
=======================================================================
--freeformat   

LAME will produce a fixed bitrate, free format bitstream.
User must specify the desired bitrate in kbps, which can
be any integer between 8 and 640.

Not supported by most decoders.  Complient decoders (of which there
are few) are only required to support up to 320 kbps.  

Decoders which can handle free format:

                     supports up to
MAD                      640 kbps
"lame --decode"          550 kbps  
Freeamp:                 440 kbps
l3dec:                   310 kbps





=======================================================================
high quality
=======================================================================
-h

use some quality improvements.  The same as -q 2.



=======================================================================
Modes:
=======================================================================

-m m           mono
-m s           stereo
-m j           joint stereo
-m f           forced mid/side stereo
-m d           dual (independent) channels
-m i           intensity stereo
-m a           auto

MONO is the default mode for mono input files.  If "-m m" is specified
for a stereo input file, the two channels will be averaged into a mono
signal.  

STEREO

JOINT STEREO is the default mode for stereo files with fixed bitrates of
128 kbps or less.  At higher fixed bitrates, the default is stereo.
For VBR encoding, jstereo is the default for VBR_q >4, and stereo
is the default for VBR_q <=4.  You can override all of these defaults
by specifing the mode on the command line.  

jstereo means the encoder can use (on a frame by frame bases) either
regular stereo (just encode left and right channels independently)
or mid/side stereo.  In mid/side stereo, the mid (L+R) and side (L-R)
channels are encoded, and more bits are allocated to the mid channel
than the side channel.  This will effectively increase the bandwidth
if the signal does not have too much stereo separation.  

Mid/side stereo is basically a trick to increase bandwidth.  At 128 kbps,
it is clearly worth while.  At higher bitrates it is less useful.

For truly mono content, use -m m, which will automatically down
sample your input file to mono.  This will produce 30% better results
over -m j.  

Using mid/side stereo inappropriately can result in audible
compression artifacts.  To much switching between mid/side and regular
stereo can also sound bad.  To determine when to switch to mid/side
stereo, LAME uses a much more sophisticated algorithm than that
described in the ISO documentation.

FORCED MID/SIDE STEREO forces all frames to be encoded mid/side stereo.  It 
should only be used if you are sure every frame of the input file
has very little stereo seperation.  

DUAL CHANNELS   Not supported.

INTENSITY STEREO

AUTO

Auto select should select (if input is stereo)
          8 kbps   Mono
     16- 96 kbps   Intensity Stereo (if available, otherwise Joint Stereo)
    112-128 kbps   Joint Stereo -mj
    160-192 kbps   -mj with variable mid/side threshold
    224-320 kbps   Independent Stereo -ms



=======================================================================
MP3 input file
=======================================================================
--mp3input

Assume the input file is a MP3 file.  LAME will decode the input file
before re-encoding it.  Since MP3 is a lossy format, this is 
not recommended in general.  But it is useful for creating low bitrate
mp3s from high bitrate mp3s.  If the filename ends in ".mp3" LAME will assume
it is an MP3.  For stdin or MP3 files which dont end in .mp3 you need
to use this switch.


=======================================================================
disable historgram display
=======================================================================
--nohist

By default, LAME will display a bitrate histogram while producing
VBR mp3 files.  This will disable that feature.


=======================================================================
disable ReplayGain analysis
=======================================================================
--noreplaygain

By default ReplayGain analysis is enabled. This switch disables it.

See also: --replaygain-accurate, --replaygain-fast


=======================================================================
non-original
=======================================================================
-o   

mark the encoded file as a copy



=======================================================================
CRC error protection
=======================================================================
-p  

turn on CRC error protection.  
Yes this really does work correctly in LAME.  However, it takes 
16 bits per frame that would otherwise be used for encoding.


=======================================================================
algorithm quality selection
=======================================================================
-q n  

Bitrate is of course the main influence on quality.  The higher the
bitrate, the higher the quality.  But for a given bitrate,
we have a choice of algorithms to determine the best
scalefactors and huffman encoding (noise shaping).

-q 0:  use slowest & best possible version of all algorithms.

-q 2:  recommended.  Same as -h.  -q 0 and -q 1 are slow and may not produce 
       significantly higher quality.  

-q 5:  default value.  Good speed, reasonable quality

-q 7:  same as -f.  Very fast, ok quality.  (psycho acoustics are
       used for pre-echo & M/S, but no noise shaping is done.  

-q 9:  disables almost all algorithms including psy-model.  poor quality.



=======================================================================
input file is raw pcm
=======================================================================
-r  

Assume the input file is raw pcm.  Sampling rate and mono/stereo/jstereo
must be specified on the command line.  Without -r, LAME will perform
several fseek()'s on the input file looking for WAV and AIFF headers.

Not supported if LAME is compiled to use LIBSNDFILE.



=======================================================================
slightly more accurate ReplayGain analysis and finding the peak sample
=======================================================================
--replaygain-accurate

Enable decoding on the fly. Compute "Radio" ReplayGain on the decoded 
data stream. Find the peak sample of the decoded data stream and store 
it in the file.


ReplayGain analysis does _not_ affect the content of a compressed data
stream itself, it is a value stored in the header of a sound file. 
Information on the purpose of ReplayGain and the algorithms used is 
available from http://www.replaygain.org/

By default, LAME performs ReplayGain analysis on the input data (after
the user-specified volume scaling). This behaviour might give slightly 
inaccurate results because the data on the output of a lossy 
compression/decompression sequence differs from the initial input data. 
When --replaygain-accurate is specified the mp3 stream gets decoded on
the fly and the analysis is performed on the decoded data stream. 
Although theoretically this method gives more accurate results, it has
several disadvantages:
  * tests have shown that the difference between the ReplayGain values 
    computed on the input data and decoded data is usually no greater 
    than 0.5dB, although the minimum volume difference the human ear 
    can perceive is about 1.0dB
  * decoding on the fly significantly slows down the encoding process
The apparent advantage is that:
  * with --replaygain-accurate the peak sample is determined and 
    stored in the file. The knowledge of the peak sample can be useful
    to decoders (players) to prevent a negative effect called 'clipping'
    that introduces distortion into sound.
    

Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag. 
The analysis is performed with the reference volume equal to 89dB. 
Note: the reference volume has been changed from 83dB on transition
from version 3.95 to 3.95.1.

This option is not usable if the MP3 decoder was _explicitly_ disabled 
in the build of LAME. (Note: if LAME is compiled without the MP3 decoder, 
ReplayGain analysis is performed on the input data after user-specified 
volume scaling).

See also: --replaygain-fast, --noreplaygain, --clipdetect


=======================================================================
fast ReplayGain analysis
=======================================================================
--replaygain-fast

Compute "Radio" ReplayGain of the input data stream after user-specified
volume scaling and/or resampling.

ReplayGain analysis does _not_ affect the content of a compressed data
stream itself, it is a value stored in the header of a sound file. 
Information on the purpose of ReplayGain and the algorithms used is 
available from http://www.replaygain.org/

Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag. 
The analysis is performed with the reference volume equal to 89dB. 
Note: the reference volume has been changed from 83dB on transition
from version 3.95 to 3.95.1.

This switch is enabled by default.

See also: --replaygain-accurate, --noreplaygain



=======================================================================
output sampling frequency in kHz
=======================================================================
--resample  n

where n = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48

Output sampling frequency.  Resample the input if necessary.  

If not specified, LAME may sometimes resample automatically 
when faced with extreme compression conditions (like encoding
a 44.1 kHz input file at 32 kbps).  To disable this automatic
resampling, you have to use --resamle to set the output samplerate
equal to the inptu samplerate.  In that case, LAME will not
perform any extra computations.



=======================================================================
sampling frequency in kHz
=======================================================================
-s  n

where n = sampling rate in kHz.

Required for raw PCM input files.  Otherwise it will be determined
from the header information in the input file.

LAME will automatically resample the input file to one of the
supported MP3 samplerates if necessary.


=======================================================================
silent operation
=======================================================================
-S

don't print progress report

=======================================================================
scale
=======================================================================
--scale <arg>

Scales input by <arg>.  This just multiplies the PCM data
(after it has been converted to floating point) by <arg>.  

<arg> > 1:  increase volume
<arg> = 1:  no effect
<arg> < 1:  reduce volume

Use with care, since most MP3 decoders will truncate data
which decodes to values greater than 32768.  


=======================================================================
strict ISO complience
=======================================================================
--strictly-enforce-ISO   

With this option, LAME will enforce the 7680 bit limitation on
total frame size.  This results in many wasted bits for
high bitrate encodings.


=======================================================================
disable VBR tag
=======================================================================
-t              

Disable writing of the VBR Tag (only valid if -v flag is
specified) This tag in embedded in frame 0 of the MP3 file.  It lets
VBR aware players correctly seek and compute playing times of VBR
files.

When '--decode' is specified (decode mp3 to wav), this flag will 
disable writing the WAV header.  The output will be raw pcm,
native endian format.  Use -x to swap bytes.



=======================================================================
variable bit rate  (VBR)
=======================================================================
-v

Turn on VBR.  There are several ways you can use VBR.  I personally
like using VBR to get files slightly bigger than 128 kbps files, where
the extra bits are used for the occasional difficult-to-encode frame.
For this, try specifying a minimum bitrate to use with VBR:

lame -v      -b 112  input.wav output.mp3

If the file is too big, use -V n, where n = 0...9

lame -v -V n -b 112  input.wav output.mp3


If you want to use VBR to get the maximum compression possible,
and for this, you can try:  

lame -v  input.wav output.mp3
lame -v -V n input.wav output.mp3         (to vary quality/filesize)






=======================================================================
VBR quality setting
=======================================================================
-V n       

n = 0...9.  Specifies the value of VBR_q.
default = 4,  highest quality = 0, smallest files = 9

Using -V 6 or higher (lower quality) is NOT RECOMMENDED.  
ABR will produce better results.  


How is VBR_q used?

The value of VBR_q influences two basic parameters of LAME's psycho
acoustics:
 a) the absolute threshold of hearing
 b) the sample to noise ratio
The lower the VBR_q value the lower the injected quantization noise
will be.
 
*NOTE* No psy-model is perfect, so there can often be distortion which
is audible even though the psy-model claims it is not!  Thus using a
small minimum bitrate can result in some aggressive compression and
audible distortion even with -V 0.  Thus using -V 0 does not sound
better than a fixed 256 kbps encoding.  For example: suppose in the 1 kHz
frequency band the psy-model claims 20 dB of distortion will not be
detectable by the human ear, so LAME VBR-0 will compress that
frequency band as much as possible and introduce at most 20 dB of
distortion.  Using a fixed 256 kbps framesize, LAME could end up
introducing only 2 dB of distortion.  If the psy-model was correct,
they will both sound the same.  If the psy-model was wrong, the VBR-0
result can sound worse.


=======================================================================
swapbytes   
=======================================================================
-x

swap bytes in the input file (and output file when using --decode).
For sorting out little endian/big endian type problems.  If your
encodings sound like static, try this first.

=======================================================================
Window and OS/2 process priority control   
=======================================================================
--priority <type>

(Windows and OS/2 only)

Sets the process priority for LAME while running under IBM OS/2.
This can be very useful to avoid the system becoming slow and/or
unresponsive. By setting LAME to run in a lower priority, you leave
more time for the system to update basic processing (drawing windows,
polling keyboard/mouse, etc). The impact in LAME's performance is 
minimal if you use priority 0 to 2.

The valid parameters are:

     0 = Low priority (IDLE, delta = 0)
     1 = Medium priority (IDLE, delta = +31)
     2 = Regular priority (REGULAR, delta = -31)
     3 = High priority (REGULAR, delta = 0)
     4 = Maximum priority (REGULAR, delta = +31) 

Note that if you call '--priority' without a parameter, then 
priority 0 will be assumed.