Blame doc/man/lame.1

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.TH lame 1 "December 08, 2013" "LAME 3.99" "LAME audio compressor"
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.SH NAME
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lame \- create mp3 audio files
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.SH SYNOPSIS
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lame [options] <infile> <outfile>
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.SH DESCRIPTION
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.PP
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LAME is a program which can be used to create compressed audio files.
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(Lame ain't an MP3 encoder).
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These audio files can be played back by popular MP3 players such as
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mpg123 or madplay.
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To read from stdin, use "\-" for <infile>.
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To write to stdout, use "\-" for <outfile>.
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.SH OPTIONS
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Input options:
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.TP
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.B \-r
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Assume the input file is raw pcm.
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Sampling rate and mono/stereo/jstereo must be specified on the command line.
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For each stereo sample, LAME expects the input data to be ordered left channel
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first, then right channel. By default, LAME expects them to be signed integers
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with a bitwidth of 16 and stored in little-endian.
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Without
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.B \-r,
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LAME will perform several
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.I fseek()'s
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on the input file looking for WAV and AIFF headers.
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.br
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Might not be available on your release. 
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.TP
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.B \-x
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Swap bytes in the input file (or output file when using
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.B \-\-decode).
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.br
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For sorting out little endian/big endian type problems.
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If your encodings sounds like static,
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try this first.
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.br
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Without using
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.B \-x,
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LAME will treat input file as native endian.
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.TP
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.BI \-s " sfreq"
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.I sfreq
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= 8/11.025/12/16/22.05/24/32/44.1/48
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Required only for raw PCM input files.
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Otherwise it will be determined from the header of the input file.
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LAME will automatically resample the input file to one of the supported
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MP3 samplerates if necessary.
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.TP
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.BI \-\-bitwidth " n"
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Input bit width per sample.
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.br
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.I n
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= 8, 16, 24, 32 (default 16)
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Required only for raw PCM input files.
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Otherwise it will be determined from the header of the input file.
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.TP
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.BI \-\-signed
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Instructs LAME that the samples from the input are signed (the default
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for 16, 24 and 32 bits raw pcm data).
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Required only for raw PCM input files.
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.TP
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.BI \-\-unsigned
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Instructs LAME that the samples from the input are unsigned (the default
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for 8 bits raw pcm data, where 0x80 is zero).
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Required only for raw PCM input files
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and only available at bitwidth 8.
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.TP
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.BI \-\-little-endian
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Instructs LAME that the samples from the input are in little-endian form.
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Required only for raw PCM input files.
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.TP
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.BI \-\-big-endian
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Instructs LAME that the samples from the input are in big-endian form.
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Required only for raw PCM input files.
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.TP
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.B \-\-mp1input
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Assume the input file is a MPEG Layer I (ie MP1) file.
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.br
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If the filename ends in ".mp1" LAME will assume it is a MPEG Layer I file.
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For stdin or Layer I files which do not end in .mp1 you need to use
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this switch. 
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.TP
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.B \-\-mp2input
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Assume the input file is a MPEG Layer II (ie MP2) file.
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.br
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If the filename ends in ".mp2" LAME will assume it is a MPEG Layer II file.
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For stdin or Layer II files which do not end in .mp2 you need to use
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this switch. 
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.TP
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.B \-\-mp3input
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Assume the input file is a MP3 file.
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.br
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Useful for downsampling from one mp3 to another.
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As an example,
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it can be useful for streaming through an IceCast server.
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.br
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If the filename ends in ".mp3" LAME will assume it is an MP3.
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For stdin or MP3 files which do not end in .mp3 you need to use this switch. 
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.TP
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.BI \-\-nogap " file1 file2 ..."
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gapless encoding for a set of contiguous files
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.TP
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.BI \-\-nogapout " dir"
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output dir for gapless encoding (must precede \-\-nogap)
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.TP
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.BI \-\-out-dir " dir"
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If no explicit output file is specified, a file will be written at given path.
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Ignored when using piped/streamed input
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.PP
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Operational options:
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.TP
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.BI \-m  " mode"
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.I mode
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= s, j, f, d, m, l, r
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Joint-stereo is the default mode for stereo files. 
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.B (s)imple stereo (Forced LR)
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.br
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In this mode,
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the encoder makes no use of potentially existing correlations between
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the two input channels.
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It can,
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however,
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negotiate the bit demand between both channel,
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i.e. give one channel more bits if the other contains silence or needs
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less bits because of a lower complexity.
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.B (j)oint stereo
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.br
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In this mode, 
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the encoder can use (on a frame by frame basis) either L/R stereo or mid/side stereo.
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In mid/side stereo,
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the mid (L+R) and side (L-R) channels are encoded,
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and more bits are allocated to the mid channel than the side channel.
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When there isn't too much stereo separation, this effectively increases the bandwidth,
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so having higher quality with the same amount of bits.
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Using mid/side stereo inappropriately can result in audible compression artifacts.
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Too much switching between mid/side and regular stereo can also sound bad.
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To determine when to switch to mid/side stereo,
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LAME uses a much more sophisticated algorithm than the one described in the ISO documentation.
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.B (f)orced MS stereo 
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.br
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Forces all frames to be encoded with mid/side stereo. It should be used only if you are sure that every frame of the input file has very little stereo separation.
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.B (d)ual channel
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.br
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In this mode,
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the 2 channels will be totally independently encoded.
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Each channel will have exactly half of the bitrate.
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This mode is designed for applications like dual languages
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encoding (for example: English in one channel and French in the other).
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Using this encoding mode for regular stereo files will result in a
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lower quality encoding.
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.B (m)ono
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.br
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The input will be encoded as a mono signal.
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If it was a stereo signal,
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it will be downsampled to mono.
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The downmix is calculated as the sum of the left and right channel,
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attenuated by 6 dB.
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Also note that, if using a stereo RAW PCM stream, you need to use the -a parameter.
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.B (l)eft channel only
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.br
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The input will be encoded as a mono signal.
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If it was a stereo signal,
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the left channel will be encoded only.
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.B (r)ight channel only
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.br
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The input will be encoded as a mono signal.
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If it was a stereo signal,
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the right channel will be encoded only.
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.TP
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.B \-a
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Mix the stereo input file to mono and encode as mono.
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.br
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The downmix is calculated as the sum of the left and right channel,
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attenuated by 6 dB. 
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This option is only needed in the case of raw PCM stereo input
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(because LAME cannot determine the number of channels in the input file).
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To encode a stereo RAW PCM input file as mono,
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use
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.B lame \-a \-m m
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For WAV and AIFF input files,
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using
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.B \-m m
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will always produce a mono .mp3 file from both mono and stereo input. 
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.TP
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.B \-\-freeformat
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Produces a free format bitstream.
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With this option,
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you can use
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.B \-b
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with any bitrate higher than 8 kbps.
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However,
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even if an mp3 decoder is required to support free bitrates at
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least up to 320 kbps,
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many players are unable to deal with it.
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Tests have shown that the following decoders support free format:
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.br
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.B in_mpg123
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up to 560 kbps
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.br
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.B l3dec
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up to 310 kbps
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.br
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.B LAME
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up to 640 kbps
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.br
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.B MAD
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up to 640 kbps
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.TP
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.B \-\-decode
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Uses LAME for decoding to a wav file.
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The input file can be any input type supported by encoding,
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including layer II files.
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LAME uses a fork of mpglib known as HIP for decoding.
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If
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.B \-t
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is used (disable wav header),
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LAME will output raw pcm in native endian format.
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You can use
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.B \-x
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to swap bytes order.
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This option is not usable if the MP3 decoder was
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.B explicitly
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disabled in the build of LAME.
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.TP
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.BI \-t
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Disable writing of the INFO Tag on encoding.
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.br
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This tag is embedded in frame 0 of the MP3 file.
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It includes some information about the encoding options of the file,
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and in VBR it lets VBR aware players correctly seek and compute
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playing times of VBR files.
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When
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.B \-\-decode
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is specified (decode to WAV),
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this flag will disable writing of the WAV header.
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The output will be raw pcm,
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native endian format.
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Use
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.B \-x
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to swap bytes.
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.TP
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.BI \-\-comp " arg"
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Instead of choosing bitrate,
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using this option,
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user can choose compression ratio to achieve.
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.TP
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.BI \-\-scale " n"
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.PD 0
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.TP
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.BI \-\-scale\-l " n"
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.TP
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.BI \-\-scale\-r " n"
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Scales input (every channel, only left channel or only right channel) by
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.I n.
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This just multiplies the PCM data (after it has been converted to floating
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point) by
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.I n. 
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.I n
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> 1: increase volume
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.br
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.I n
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= 1: no effect
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.br
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.I n
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< 1: reduce volume
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Use with care,
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since most MP3 decoders will truncate data which decodes to values
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greater than 32768.
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.PD
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.TP
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.B \-\-replaygain\-fast
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Compute ReplayGain fast but slightly inaccurately.
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This computes "Radio" ReplayGain on the input data stream after
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user\(hyspecified volume\(hyscaling and/or resampling.
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The ReplayGain analysis does
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.I not
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affect the content of a compressed data stream itself,
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it is a value stored in the header of a sound file.
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Information on the purpose of ReplayGain and the algorithms used is
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available from
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.B http://www.replaygain.org/.
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Only the "RadioGain" Replaygain value is computed,
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it is stored in the LAME tag.
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The analysis is performed with the reference
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volume equal to 89dB.
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Note: the reference volume has been changed from 83dB on transition from
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version 3.95 to 3.95.1.
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This switch is enabled by default.
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See also:
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.B \-\-replaygain\-accurate, \-\-noreplaygain
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.TP
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.B \-\-replaygain\-accurate
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Compute ReplayGain more accurately and find the peak sample.
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This computes "Radio" ReplayGain on the decoded data stream,
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finds the peak sample by decoding on the fly the encoded data stream and stores it in the file. 
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The ReplayGain analysis does
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.I not
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affect the content of a compressed data stream itself,
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it is a value stored in the header of a sound file.
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Information on the purpose of ReplayGain and the algorithms used is
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available from
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.B http://www.replaygain.org/.
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By default, LAME performs ReplayGain analysis on the input data
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(after the user\(hyspecified volume scaling).
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This behavior might give slightly inaccurate results
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because the data on the output of a lossy compression/decompression sequence
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differs from the initial input data.
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When
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.B \-\-replaygain-accurate
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is specified the mp3 stream gets decoded on the fly and the analysis is
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performed on the decoded data stream.
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Although theoretically this method gives more accurate results,
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it has several disadvantages:
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.RS 8
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.IP "*" 4
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tests have shown that the difference between the ReplayGain values computed
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on the input data and decoded data is usually not greater than 0.5dB,
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although the minimum volume difference the human ear can perceive is
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about 1.0dB
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.IP "*" 4
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decoding on the fly significantly slows down the encoding process
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.RE
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.RS 7
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The apparent advantage is that:
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.RE
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.RS 8
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.IP "*" 4
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with
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.B \-\-replaygain-accurate
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the real peak sample is determined and stored in the file.
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The knowledge of the peak sample can be useful to decoders (players)
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to prevent a negative effect called 'clipping' that introduces distortion
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into the sound.
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.RE
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.RS 7
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Only the "RadioGain" ReplayGain value is computed,
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it is stored in the LAME tag.
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The analysis is performed with the reference
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volume equal to 89dB.
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Note: the reference volume has been changed from 83dB on transition from
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version 3.95 to 3.95.1.
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This option is not usable if the MP3 decoder was
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.B explicitly
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disabled in the build of LAME.
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(Note: if LAME is compiled without the MP3 decoder,
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ReplayGain analysis is performed on the input data after user-specified
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volume scaling).
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See also:
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.B \-\-replaygain-fast, \-\-noreplaygain \-\-clipdetect
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.RE
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.TP
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.B \-\-noreplaygain
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Disable ReplayGain analysis.
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By default ReplayGain analysis is enabled. This switch disables it.
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See also:
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.B \-\-replaygain-fast, \-\-replaygain-accurate
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.TP
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.B \-\-clipdetect
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Clipping detection.
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Enable
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.B \-\-replaygain-accurate
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and print a message whether clipping occurs and how far in dB the waveform
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is from full scale.
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This option is not usable if the MP3 decoder was
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.B explicitly
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disabled in the build of LAME.
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See also:
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.B \-\-replaygain-accurate
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.TP
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.B \-\-preset " type | [cbr] kbps"
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Use one of the built-in presets.
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Have a look at the PRESETS section below.
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.B \-\-preset help
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gives more infos about the the used options in these presets.
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.TP
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.B \-\-noasm " type"
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Disable specific assembly optimizations (
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.B mmx
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/
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.B 3dnow
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/
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.B sse
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).
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Quality will not increase, only speed will be reduced.
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If you have problems running Lame on a Cyrix/Via processor,
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disabling mmx optimizations might solve your problem.
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.PP
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Verbosity:
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.TP
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.BI \-\-disptime " n"
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Set the delay in seconds between two display updates. 
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.TP
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.B \-\-nohist
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By default,
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LAME will display a bitrate histogram while producing VBR mp3 files.
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This will disable that feature.
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.br
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Histogram display might not be available on your release. 
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.TP
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.B -S
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.PD 0
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.TP
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.B \-\-silent
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.TP
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.B \-\-quiet
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Do not print anything on the screen.
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.PD
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.TP
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.B \-\-verbose
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Print a lot of information on the screen.
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.TP
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.B \-\-help
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Display a list of available options.
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.PP
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Noise shaping & psycho acoustic algorithms:
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.TP
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.BI -q " qual"
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0 <=
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.I qual
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<= 9
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Bitrate is of course the main influence on quality.
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The higher the bitrate,
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the higher the quality.
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But for a given bitrate,
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we have a choice of algorithms to determine the best scalefactors
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and Huffman encoding (noise shaping).
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For CBR and ABR, the following table applies:
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.B -q 0:
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.br
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Use the best algorithms (Best Huffman coding search, full outer loop, and the highest precision of several parameters).
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.B -q 1 to q 4:
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.br
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Similar to -q 0 without the full outer loop and decreasing precision of parameters the further from q0. -q 3 is the default.
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.B -q 5 and -q 6:
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.br
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Same as -q 7, but enables noise shaping and increases subblock gain
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.B -q 7 to -q 9:
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.br
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Same as -f. Very fast, OK quality. Psychoacoustics are used for pre-echo and mid/side stereo, but no noise-shaping is done.
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For the default VBR mode since LAME 3.98, the following table applies :
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.B -q 0 to -q 4:
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.br
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include all features of the other modes and additionally use the best search when applying Huffman coding.
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.B -q 5 and -q 6:
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.br
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include all features of -q7, calculate and consider actual quantisation noise, and additionally enable subblock gain.
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.B -q 7 to -q 9
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.br
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This level uses a psymodel but does not calculate quantisation noise when encoding: it takes a quick guess.
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.TP
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.B -h
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Alias of 
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.B -q 2
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.TP
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.B -f
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Alias of
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.B -q 7
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.PP
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CBR (constant bitrate, the default) options:
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.TP
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.BI -b  " n"
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For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz)
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.br
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.I n
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= 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320
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For MPEG-2 (sampling frequencies of 16, 22.05 and 24 kHz)
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.br
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.I n
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= 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160
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For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)
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.br
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.I n
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= 8, 16, 24, 32, 40, 48, 56, 64
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Default is 128 for MPEG1 and 64 for MPEG2 and 32 for MPEG2.5
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 (64, 32 and 16 respectively in case of mono). 
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.TP
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.BI \-\-cbr
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enforce use of constant bitrate. Used to disable VBR or ABR encoding even if their settings
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are enabled.
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.PP
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ABR (average bitrate) options:
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.TP
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.BI \-\-abr " n"
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Turns on encoding with a targeted average bitrate of n kbits,
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allowing to use frames of different sizes.
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The allowed range of
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.I n
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is 8 - 310,
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you can use any integer value within that range.
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It can be combined with the
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.B -b
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and
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.B -B
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switches like:
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.B lame \-\-abr
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.I 123
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.B -b
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.I 64
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.B -B
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.I 192 a.wav a.mp3
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which would limit the allowed frame sizes between 64 and 192 kbits.
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The use of
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.B -B
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is NOT RECOMMENDED.
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A 128 kbps CBR bitstream,
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because of the bit reservoir,
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can actually have frames which use as many bits as a 320 kbps frame.
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VBR modes minimize the use of the bit reservoir,
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and thus need to allow 320 kbps frames to get the same flexibility
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as CBR streams. 
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.PP
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VBR (variable bitrate) options:
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.TP
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.B -v
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use variable bitrate
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.B (\-\-vbr-new)
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.TP
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.B \-\-vbr-old
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Invokes the oldest,
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most tested VBR algorithm.
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It produces very good quality files,
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though is not very fast.
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This has,
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up through v3.89,
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been considered the "workhorse" VBR algorithm.
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.TP
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.B \-\-vbr-new
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Invokes the newest VBR algorithm.
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During the development of version 3.90,
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considerable tuning was done on this algorithm,
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and it is now considered to be on par with the original
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.B \-\-vbr-old. 
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It has the added advantage of being very fast (over twice as fast as
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.B \-\-vbr-old
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). This is the default since 3.98.
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.TP
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.BI -V " n"
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0 <=
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.I n
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<= 9.999
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.br
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Enable VBR (Variable BitRate) and specifies the value of VBR quality
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(default = 4). Decimal values can be specified, like 4.51.
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.br
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0 = highest quality.
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.PP
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ABR and VBR options:
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.TP
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.BI -b " bitrate"
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For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz)
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.br
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.I n
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= 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320
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For MPEG-2 (sampling frequencies of 16, 22.05 and 24 kHz)
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.br
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.I n
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= 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160
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For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)
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.br
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.I n
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= 8, 16, 24, 32, 40, 48, 56, 64
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Specifies the minimum bitrate to be used.
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However,
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in order to avoid wasted space,
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the smallest frame size available will be used during silences. 
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.TP
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.BI -B " bitrate"
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For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz)
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.br
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.I n
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= 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320
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For MPEG-2 (sampling frequencies of 16, 22.05 and 24 kHz)
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.br
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.I n
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= 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160
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For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)
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.br
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.I n
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= 8, 16, 24, 32, 40, 48, 56, 64
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Specifies the maximum allowed bitrate.
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Note: If you own an mp3 hardware player build upon a MAS 3503 chip,
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you must set maximum bitrate to no more than 224 kpbs. 
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.TP
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.B -F
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Strictly enforce the
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.B -b
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option.
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.br
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This is mainly for use with hardware players that do not support low
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bitrate mp3.
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Without this option,
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the minimum bitrate will be ignored for passages of analog silence,
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i.e. when the music level is below the absolute threshold of
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human hearing (ATH). 
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.PP
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Experimental options:
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.TP
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.BI -X " n"
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0 <=
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.I n
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<= 7
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When LAME searches for a "good" quantization,
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it has to compare the actual one with the best one found so far. 
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The comparison says which one is better,
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the best so far or the actual.
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The
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.B -X
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parameter selects between different approaches to make this decision,
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.B -X0
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being the default mode:
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.B -X0 
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.br
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The criteria are (in order of importance):
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.br
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* less distorted scalefactor bands
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.br
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* the sum of noise over the thresholds is lower
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.br
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* the total noise is lower
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.B -X1
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.br
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The actual is better if the maximum noise over all scalefactor bands is
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less than the best so far.
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.B -X2
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.br
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The actual is better if the total sum of noise is lower than the best so
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far.
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.B -X3
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.br
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The actual is better if the total sum of noise is lower than the best so
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far and the maximum noise over all scalefactor bands is less than the
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best so far plus 2dB.
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.B -X4
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.br
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Not yet documented.
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.B -X5
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.br
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The criteria are (in order of importance):
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.br
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* the sum of noise over the thresholds is lower 
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.br
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* the total sum of noise is lower
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.B -X6 
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.br
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The criteria are (in order of importance):
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.br
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* the sum of noise over the thresholds is lower
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.br
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* the maximum noise over all scalefactor bands is lower
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.br
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* the total sum of noise is lower
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.B -X7 
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.br
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The criteria are:
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.br
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* less distorted scalefactor bands
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.br
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or
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.br
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* the sum of noise over the thresholds is lower 
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.TP
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.B -Y
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lets LAME ignore noise in sfb21, like in CBR
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.PP
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MP3 header/stream options:
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.TP
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.BI -e " emp"
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.I emp
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= n, 5, c
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n = (none, default)
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.br
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5 = 0/15 microseconds
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.br
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c = citt j.17
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All this does is set a flag in the bitstream.
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If you have a PCM input file where one of the above types of
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(obsolete) emphasis has been applied,
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you can set this flag in LAME.
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Then the mp3 decoder should de-emphasize the output during playback,
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although most decoders ignore this flag.
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A better solution would be to apply the de-emphasis with a standalone
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utility before encoding,
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and then encode without
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.B -e. 
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.TP
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.B -c
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Mark the encoded file as being copyrighted.
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.TP
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.B -o
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Mark the encoded file as being a copy. 
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.TP
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.B -p
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Turn on CRC error protection.
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.br
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It will add a cyclic redundancy check (CRC) code in each frame,
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allowing to detect transmission errors that could occur on the
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MP3 stream.
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However,
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it takes 16 bits per frame that would otherwise be used for encoding,
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and then will slightly reduce the sound quality. 
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.TP
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.B \-\-nores
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Disable the bit reservoir.
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Each frame will then become independent from previous ones,
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but the quality will be lower. 
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.TP
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.B \-\-strictly-enforce-ISO
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With this option,
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LAME will enforce the 7680 bit limitation on total frame size.
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.br
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This results in many wasted bits for high bitrate encodings but will
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ensure strict ISO compatibility.
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This compatibility might be important for hardware players.
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.PP
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Filter options:
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.TP
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.BI \-\-lowpass " freq"
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Set a lowpass filtering frequency in kHz.
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Frequencies above the specified one will be cutoff. 
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.TP
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.BI \-\-lowpass-width " freq"
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Set the width of the lowpass filter.
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The default value is 15% of the lowpass frequency. 
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.TP
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.BI \-\-highpass " freq"
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Set an highpass filtering frequency in kHz.
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Frequencies below the specified one will be cutoff. 
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.TP
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.BI \-\-highpass-width " freq"
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Set the width of the highpass filter in kHz.
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The default value is 15% of the highpass frequency.
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.TP
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.BI \-\-resample " sfreq"
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.I sfreq
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= 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48
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.br
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Select output sampling frequency (only supported for encoding).
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.br
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If not specified,
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LAME will automatically resample the input when using high compression ratios.
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.PP
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ID3 tag options:
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.TP
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.BI \-\-tt " title"
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audio/song title (max 30 chars for version 1 tag)
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.TP
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.BI \-\-ta " artist"
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audio/song artist (max 30 chars for version 1 tag)
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.TP
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.BI \-\-tl " album"
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audio/song album (max 30 chars for version 1 tag)
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.TP
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.BI \-\-ty " year"
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audio/song year of issue (1 to 9999)
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.TP
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.BI \-\-tc " comment"
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user-defined text (max 30 chars for v1 tag, 28 for v1.1)
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.TP
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.BI \-\-tn " track[/total]"
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audio/song track number and (optionally) the total number of tracks on
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the original recording. (track and total each 1 to 255. Providing
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just the track number creates v1.1 tag, providing a total forces v2.0).
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.TP
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.BI \-\-tg " genre"
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audio/song genre (name or number in list)
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.TP
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.BI \-\-tv " id=value"
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Text or URL frame specified by id and value (v2.3 tag). User defined frame. Syntax: \-\-tv "TXXX=description=content"
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.TP
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.B \-\-add-id3v2
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force addition of version 2 tag
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.TP
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.B \-\-id3v1-only
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add only a version 1 tag
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.TP
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.B \-\-id3v2-only
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add only a version 2 tag
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.TP
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.B \-\-id3v2-latin1
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add following options in ISO-8859-1 text encoding.
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.TP
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.B \-\-id3v2-utf16
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add following options in unicode text encoding.
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.TP
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.B \-\-space-id3v1
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pad version 1 tag with spaces instead of nulls
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.TP
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.B \-\-pad-id3v2
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same as \-\-pad-id3v2-size 128
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.TP
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.B \-\-pad-id3v2-size "num"
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adds version 2 tag, pad with extra "num" bytes
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.TP
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.B \-\-genre-list
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print alphabetically sorted ID3 genre list and exit
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.TP
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.B \-\-ignore-tag-errors
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ignore errors in values passed for tags, use defaults in case an error occurs
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.PP
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Analysis options:
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.TP
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.B \-g
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run graphical analysis on <infile>.
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<infile> can also be a .mp3 file.
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(This feature is a compile time option.
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Your binary may for speed reasons be compiled without this.)
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.SH ID3 TAGS
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LAME is able to embed ID3 v1,
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v1.1 or v2 tags inside the encoded MP3 file.
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This allows to have some useful information about the music track
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included inside the file.
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Those data can be read by most MP3 players.
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Lame will smartly choose which tags to use.
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It will add ID3 v2 tags only if the input comments won't fit in v1
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or v1.1 tags,
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i.e. if they are more than 30 characters.
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In this case,
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both v1 and v2 tags will be added,
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to ensure reading of tags by MP3 players which are unable to read ID3 v2 tags.
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.SH ENCODING MODES
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LAME is able to encode your music using one of its 3 encoding modes:
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constant bitrate (CBR), average bitrate (ABR) and variable bitrate (VBR).
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.TP
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.B Constant Bitrate (CBR)
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This is the default encoding mode,
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and also the most basic.
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In this mode,
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the bitrate will be the same for the whole file.
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It means that each part of your mp3 file will be using the same
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number of bits.
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The musical passage being a difficult one to encode or an easy one,
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the encoder will use the same bitrate,
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so the quality of your mp3 is variable.
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Complex parts will be of a lower quality than the easiest ones.
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The main advantage is that the final files size won't change and
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can be accurately predicted.
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.TP
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.B Average Bitrate (ABR)
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In this mode,
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you choose the encoder will maintain an average bitrate while using
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higher bitrates for the parts of your music that need more bits.
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The result will be of higher quality than CBR encoding but the
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average file size will remain predictable,
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so this mode is highly recommended over CBR.
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This encoding mode is similar to what is referred as vbr in AAC or
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Liquid Audio (2 other compression technologies).
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.TP
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.B Variable bitrate (VBR)
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In this mode,
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you choose the desired quality on a scale from 9 (lowest
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quality/biggest distortion) to 0 (highest quality/lowest distortion).
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Then encoder tries to maintain the given quality in the whole file by
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choosing the optimal number of bits to spend for each part of your music.
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The main advantage is that you are able to specify the quality level that
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you want to reach,
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but the inconvenient is that the final file size is totally unpredictable.
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.SH PRESETS
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The
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.B \-\-preset
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switches are aliases over LAME settings.
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To activate these presets:
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.PP
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For VBR modes (generally highest quality):
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.TP
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.B \-\-preset medium
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This preset should provide near transparency to most people on most music.
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.TP
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.B \-\-preset standard
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This preset should generally be transparent to most people on most music and
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is already quite high in quality.
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.TP
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.B \-\-preset extreme
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If you have extremely good hearing and similar equipment,
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this preset will generally provide slightly higher quality than the
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.B standard
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mode.
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.PP
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For CBR 320kbps (highest quality possible from the
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.B \-\-preset
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switches):
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.TP
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.B \-\-preset insane
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This preset will usually be overkill for most people and most situations,
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but if you must have the absolute highest quality with no regard to filesize,
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this is the way to go.
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.PP
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For ABR modes (high quality per given bitrate but not as high as VBR):
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.TP
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.B \-\-preset " kbps"
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Using this preset will usually give you good quality at a specified bitrate.
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Depending on the bitrate entered,
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this preset will determine the optimal settings for that particular situation.
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While this approach works,
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it is not nearly as flexible as VBR,
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and usually will not attain the same level of quality as VBR at higher bitrates.
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.TP
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.B cbr
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If you use the ABR mode (read above) with a significant bitrate such as 80,
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96,
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112,
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128,
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160,
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192,
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224,
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256,
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320,
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you can use the
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.B \-\-preset cbr " kbps"
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option to force CBR mode encoding instead of the standard ABR mode.
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ABR does provide higher quality but CBR may be useful in situations such as when
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streaming an MP3 over the Internet may be important.
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.SH EXAMPLES
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.LP
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Fixed bit rate jstereo 128kbs encoding:
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.IP
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.B lame \-b
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.I 128 sample.wav sample.mp3
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.LP
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Fixed bit rate jstereo 128 kbps encoding, highest quality:
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.IP
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.B lame \-q 0 \-b
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.I 128 sample.wav sample.mp3
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.LP
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To disable joint stereo encoding (slightly faster,
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but less quality at bitrates <= 128 kbps):
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.IP
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.B lame \-m
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.I s sample.wav sample.mp3
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.LP
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Variable bitrate (use \-V n to adjust quality/filesize):
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.IP
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.B lame \-V
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.I 2 sample.wav sample.mp3
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.LP
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Streaming mono 22.05 kHz raw pcm, 24 kbps output:
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.IP
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.B cat
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.I inputfile
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.B | lame \-r \-m
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.I m
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.B \-b
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.I 24
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.B \-s
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.I 22.05 \- \-
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.B >
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.I output
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.LP
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Streaming mono 44.1 kHz raw pcm,
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with downsampling to 22.05 kHz:
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.IP
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.B cat
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.I inputfile
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.B | lame \-r \-m
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.I m
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.B \-b
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.I 24
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.B \-\-resample
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.I 22.05 \- \-
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.B >
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.I output
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.LP
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Encode with the
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.B standard
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preset:
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.IP
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.B lame \-\-preset standard
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.I sample.wav sample.mp3
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.SH BUGS
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.PP
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Probably there are some.
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.SH SEE ALSO
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.BR mpg123 (1) ,
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.BR madplay (1) ,
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.BR sox (1)
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.SH AUTHORS
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.nf
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LAME originally developed by Mike Cheng and now maintained by
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Mark Taylor, and the LAME team.
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GPSYCHO psycho-acoustic model by Mark Taylor.
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(See http://www.mp3dev.org/).
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mpglib by Michael Hipp
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Manual page by William Schelter, Nils Faerber, Alexander Leidinger,
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and Rog\['e]rio Brito.
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.\" Local Variables:
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.\" mode: nroff
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.\" End: