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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



 NAME
      lame - create mp3 audio files

 SYNOPSIS
      lame [options] <infile> <outfile>

 DESCRIPTION
      LAME is a program which can be used to create compressed audio files.
      (Lame ain't an MP3 encoder).  These audio files can be played back by
      popular MP3 players such as mpg123 or madplay.  To read from stdin,
      use "-" for <infile>.  To write to stdout, use "-" for <outfile>.

 OPTIONS
      Input options:

      -r   Assume the input file is raw pcm.  Sampling rate and
           mono/stereo/jstereo must be specified on the command line.  For
           each stereo sample, LAME expects the input data to be ordered
           left channel first, then right channel. By default, LAME expects
           them to be signed integers with a bitwidth of 16 and stored in
           little-endian.  Without -r, LAME will perform several fseek()'s
           on the input file looking for WAV and AIFF headers.
           Might not be available on your release.

      -x   Swap bytes in the input file (or output file when using
           --decode).
           For sorting out little endian/big endian type problems.  If your
           encodings sounds like static, try this first.
           Without using -x, LAME will treat input file as native endian.

      -s sfreq
           sfreq = 8/11.025/12/16/22.05/24/32/44.1/48

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

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

      --bitwidth n
           Input bit width per sample.
           n = 8, 16, 24, 32 (default 16)

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

      --signed
           Instructs LAME that the samples from the input are signed (the
           default for 16, 24 and 32 bits raw pcm data).



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           Required only for raw PCM input files.

      --unsigned
           Instructs LAME that the samples from the input are unsigned (the
           default for 8 bits raw pcm data, where 0x80 is zero).

           Required only for raw PCM input files and only available at
           bitwidth 8.

      --little-endian
           Instructs LAME that the samples from the input are in little-
           endian form.

           Required only for raw PCM input files.

      --big-endian
           Instructs LAME that the samples from the input are in big-endian
           form.

           Required only for raw PCM input files.

      --mp1input
           Assume the input file is a MPEG Layer I (ie MP1) file.
           If the filename ends in ".mp1" LAME will assume it is a MPEG
           Layer I file.  For stdin or Layer I files which do not end in
           .mp1 you need to use this switch.

      --mp2input
           Assume the input file is a MPEG Layer II (ie MP2) file.
           If the filename ends in ".mp2" LAME will assume it is a MPEG
           Layer II file.  For stdin or Layer II files which do not end in
           .mp2 you need to use this switch.

      --mp3input
           Assume the input file is a MP3 file.
           Useful for downsampling from one mp3 to another.  As an example,
           it can be useful for streaming through an IceCast server.
           If the filename ends in ".mp3" LAME will assume it is an MP3.
           For stdin or MP3 files which do not end in .mp3 you need to use
           this switch.

      --nogap file1 file2 ...
           gapless encoding for a set of contiguous files

      --nogapout dir
           output dir for gapless encoding (must precede --nogap)

      --out-dir dir
           If no explicit output file is specified, a file will be written



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           at given path.  Ignored when using piped/streamed input


      Operational options:

      -m mode
           mode = s, j, f, d, m, l, r

           Joint-stereo is the default mode for stereo files.

           (s)imple stereo (Forced LR)
           In this mode, the encoder makes no use of potentially existing
           correlations between the two input channels.  It can, however,
           negotiate the bit demand between both channel, i.e. give one
           channel more bits if the other contains silence or needs less
           bits because of a lower complexity.

           (j)oint stereo
           In this mode, the encoder can use (on a frame by frame basis)
           either L/R stereo 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.  When there
           isn't too much stereo separation, this effectively increases the
           bandwidth, so having higher quality with the same amount of bits.

           Using mid/side stereo inappropriately can result in audible
           compression artifacts.  Too 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 the one described in the ISO documentation.

           (f)orced MS stereo
           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.

           (d)ual channel
           In this mode, the 2 channels will be totally independently
           encoded.  Each channel will have exactly half of the bitrate.
           This mode is designed for applications like dual languages
           encoding (for example: English in one channel and French in the
           other).  Using this encoding mode for regular stereo files will
           result in a lower quality encoding.

           (m)ono
           The input will be encoded as a mono signal.  If it was a stereo
           signal, it will be downsampled to mono.  The downmix is
           calculated as the sum of the left and right channel, attenuated
           by 6 dB.  Also note that, if using a stereo RAW PCM stream, you



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           need to use the -a parameter.

           (l)eft channel only
           The input will be encoded as a mono signal.  If it was a stereo
           signal, the left channel will be encoded only.

           (r)ight channel only
           The input will be encoded as a mono signal.  If it was a stereo
           signal, the right channel will be encoded only.


      -a   Mix the stereo input file to mono and encode as mono.
           The downmix is calculated as the sum of the left and right
           channel, attenuated by 6 dB.

           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 RAW PCM input file as mono, use
           lame -a -m m

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

      --freeformat
           Produces a free format bitstream.  With this option, you can use
           -b with any bitrate higher than 8 kbps.

           However, even if an mp3 decoder is required to support free
           bitrates at least up to 320 kbps, many players are unable to deal
           with it.

           Tests have shown that the following decoders support free format:
           in_mpg123 up to 560 kbps
           l3dec up to 310 kbps
           LAME up to 640 kbps
           MAD up to 640 kbps

      --decode
           Uses LAME for decoding to a wav file.  The input file can be any
           input type supported by encoding, including layer II files.  LAME
           uses a fork of mpglib known as HIP for decoding.

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

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

      -t   Disable writing of the INFO Tag on encoding.



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           This tag is embedded in frame 0 of the MP3 file.  It includes
           some information about the encoding options of the file, and in
           VBR it lets VBR aware players correctly seek and compute playing
           times of VBR files.

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

      --comp arg
           Instead of choosing bitrate, using this option, user can choose
           compression ratio to achieve.

      --scale n
      --scale-l n
      --scale-r n
           Scales input (every channel, only left channel or only right
           channel) by n. This just multiplies the PCM data (after it has
           been converted to floating point) by n.

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

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

      --replaygain-fast
           Compute ReplayGain fast but slightly inaccurately.

           This computes "Radio" ReplayGain on the input data stream after
           user-specified volume-scaling and/or resampling.

           The 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 "RadioGain" 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

      --replaygain-accurate
           Compute ReplayGain more accurately and find the peak sample.



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           This computes "Radio" ReplayGain on the decoded data stream,
           finds the peak sample by decoding on the fly the encoded data
           stream and stores it in the file.

           The 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 behavior 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 not 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 real 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 the sound.

             Only the "RadioGain" 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

      --noreplaygain



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           Disable ReplayGain analysis.

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

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

      --clipdetect
           Clipping detection.

           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

      --preset  type | [cbr] kbps
           Use one of the built-in presets.

           Have a look at the PRESETS section below.

           --preset help gives more infos about the the used options in
           these presets.

      --noasm  type
           Disable specific assembly optimizations ( mmx / 3dnow / sse ).
           Quality will not increase, only speed will be reduced.  If you
           have problems running Lame on a Cyrix/Via processor, disabling
           mmx optimizations might solve your problem.


      Verbosity:

      --disptime n
           Set the delay in seconds between two display updates.

      --nohist
           By default, LAME will display a bitrate histogram while producing
           VBR mp3 files.  This will disable that feature.
           Histogram display might not be available on your release.

      -S
      --silent
      --quiet
           Do not print anything on the screen.

      --verbose



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           Print a lot of information on the screen.

      --help
           Display a list of available options.


      Noise shaping & psycho acoustic algorithms:

      -q qual
           0 <= qual <= 9

           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).

           For CBR and ABR, the following table applies:

           -q 0:
           Use the best algorithms (Best Huffman coding search, full outer
           loop, and the highest precision of several parameters).

           -q 1 to q 4:
           Similar to -q 0 without the full outer loop and decreasing
           precision of parameters the further from q0. -q 3 is the default.

           -q 5 and -q 6:
           Same as -q 7, but enables noise shaping and increases subblock
           gain

           -q 7 to -q 9:
           Same as -f. Very fast, OK quality. Psychoacoustics are used for
           pre-echo and mid/side stereo, but no noise-shaping is done.

           For the default VBR mode since LAME 3.98, the following table
           applies :

           -q 0 to -q 4:
           include all features of the other modes and additionally use the
           best search when applying Huffman coding.

           -q 5 and -q 6:
           include all features of -q7, calculate and consider actual
           quantisation noise, and additionally enable subblock gain.

           -q 7 to -q 9
           This level uses a psymodel but does not calculate quantisation
           noise when encoding: it takes a quick guess.




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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



      -h   Alias of -q 2

      -f   Alias of -q 7



      CBR (constant bitrate, the default) options:

      -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

           Default is 128 for MPEG1 and 64 for MPEG2 and 32 for MPEG2.5
            (64, 32 and 16 respectively in case of mono).

      --cbr
           enforce use of constant bitrate. Used to disable VBR or ABR
           encoding even if their settings are enabled.


      ABR (average bitrate) options:

      --abr n
           Turns on encoding with a targeted average bitrate of n kbits,
           allowing to use frames of different sizes.  The allowed range of
           n is 8 - 310, you can use any integer value within that range.

           It can be combined with the -b and -B switches like: lame --abr
           123 -b 64 -B 192 a.wav a.mp3 which would limit the allowed frame
           sizes between 64 and 192 kbits.

           The use of -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.  VBR modes minimize the use of
           the bit reservoir, and thus need to allow 320 kbps frames to get
           the same flexibility as CBR streams.


      VBR (variable bitrate) options:

      -v   use variable bitrate (--vbr-new)

      --vbr-old
           Invokes the oldest, most tested VBR algorithm.  It produces very



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           good quality files, though is not very fast.  This has, up
           through v3.89, been considered the "workhorse" VBR algorithm.

      --vbr-new
           Invokes the newest VBR algorithm.  During the development of
           version 3.90, considerable tuning was done on this algorithm, and
           it is now considered to be on par with the original --vbr-old. It
           has the added advantage of being very fast (over twice as fast as
           --vbr-old ). This is the default since 3.98.

      -V n 0 <= n <= 9.999
           Enable VBR (Variable BitRate) and specifies the value of VBR
           quality (default = 4). Decimal values can be specified, like
           4.51.
           0 = highest quality.


      ABR and VBR options:

      -b bitrate
           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

           Specifies the minimum bitrate to be used.  However, in order to
           avoid wasted space, the smallest frame size available will be
           used during silences.

      -B bitrate
           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

           Specifies the maximum allowed bitrate.

           Note: If you own an mp3 hardware player build upon a MAS 3503
           chip, you must set maximum bitrate to no more than 224 kpbs.

      -F   Strictly enforce the -b option.



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           This is mainly for use with hardware players that do not support
           low bitrate mp3.

           Without this option, the minimum bitrate will be ignored for
           passages of analog silence, i.e. when the music level is below
           the absolute threshold of human hearing (ATH).


      Experimental options:

      -X n 0 <= n <= 7

           When LAME searches for a "good" quantization, it has to compare
           the actual one with the best one found so far. The comparison
           says which one is better, the best so far or the actual.  The -X
           parameter selects between different approaches to make this
           decision, -X0 being the default mode:

           -X0
           The criteria are (in order of importance):
           * less distorted scalefactor bands
           * the sum of noise over the thresholds is lower
           * the total noise is lower

           -X1
           The actual is better if the maximum noise over all scalefactor
           bands is less than the best so far.

           -X2
           The actual is better if the total sum of noise is lower than the
           best so far.

           -X3
           The actual is better if the total sum of noise is lower than the
           best so far and the maximum noise over all scalefactor bands is
           less than the best so far plus 2dB.

           -X4
           Not yet documented.

           -X5
           The criteria are (in order of importance):
           * the sum of noise over the thresholds is lower
           * the total sum of noise is lower

           -X6
           The criteria are (in order of importance):
           * the sum of noise over the thresholds is lower
           * the maximum noise over all scalefactor bands is lower



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           * the total sum of noise is lower

           -X7
           The criteria are:
           * less distorted scalefactor bands
           or
           * the sum of noise over the thresholds is lower

      -Y   lets LAME ignore noise in sfb21, like in CBR


      MP3 header/stream options:

      -e emp
           emp = 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.

      -c   Mark the encoded file as being copyrighted.

      -o   Mark the encoded file as being a copy.

      -p   Turn on CRC error protection.
           It will add a cyclic redundancy check (CRC) code in each frame,
           allowing to detect transmission errors that could occur on the
           MP3 stream.  However, it takes 16 bits per frame that would
           otherwise be used for encoding, and then will slightly reduce the
           sound quality.

      --nores
           Disable the bit reservoir.  Each frame will then become
           independent from previous ones, but the quality will be lower.

      --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 but
           will ensure strict ISO compatibility.  This compatibility might



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           be important for hardware players.


      Filter options:

      --lowpass freq
           Set a lowpass filtering frequency in kHz.  Frequencies above the
           specified one will be cutoff.

      --lowpass-width freq
           Set the width of the lowpass filter.  The default value is 15% of
           the lowpass frequency.

      --highpass freq
           Set an highpass filtering frequency in kHz.  Frequencies below
           the specified one will be cutoff.

      --highpass-width freq
           Set the width of the highpass filter in kHz.  The default value
           is 15% of the highpass frequency.

      --resample sfreq
           sfreq = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48
           Select output sampling frequency (only supported for encoding).
           If not specified, LAME will automatically resample the input when
           using high compression ratios.


      ID3 tag options:

      --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[/total]
           audio/song track number and (optionally) the total number of
           tracks on the original recording. (track and total each 1 to 255.
           Providing just the track number creates v1.1 tag, providing a



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           total forces v2.0).

      --tg genre
           audio/song genre (name or number in list)

      --tv id=value
           Text or URL frame specified by id and value (v2.3 tag). User
           defined frame. Syntax: --tv "TXXX=description=content"

      --add-id3v2
           force addition of version 2 tag

      --id3v1-only
           add only a version 1 tag

      --id3v2-only
           add only a version 2 tag

      --id3v2-latin1
           add following options in ISO-8859-1 text encoding.

      --id3v2-utf16
           add following options in unicode text encoding.

      --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

      --ignore-tag-errors
           ignore errors in values passed for tags, use defaults in case an
           error occurs


      Analysis options:

      -g   run graphical analysis on <infile>.  <infile> can also be a .mp3
           file.  (This feature is a compile time option.  Your binary may
           for speed reasons be compiled without this.)


 ID3 TAGS



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



      LAME is able to embed ID3 v1, v1.1 or v2 tags inside the encoded MP3
      file.  This allows to have some useful information about the music
      track included inside the file.  Those data can be read by most MP3
      players.

      Lame will smartly choose which tags to use.  It will add ID3 v2 tags
      only if the input comments won't fit in v1 or v1.1 tags, i.e. if they
      are more than 30 characters.  In this case, both v1 and v2 tags will
      be added, to ensure reading of tags by MP3 players which are unable to
      read ID3 v2 tags.


 ENCODING MODES
      LAME is able to encode your music using one of its 3 encoding modes:
      constant bitrate (CBR), average bitrate (ABR) and variable bitrate
      (VBR).

      Constant Bitrate (CBR)
           This is the default encoding mode, and also the most basic.  In
           this mode, the bitrate will be the same for the whole file.  It
           means that each part of your mp3 file will be using the same
           number of bits.  The musical passage being a difficult one to
           encode or an easy one, the encoder will use the same bitrate, so
           the quality of your mp3 is variable.  Complex parts will be of a
           lower quality than the easiest ones.  The main advantage is that
           the final files size won't change and can be accurately
           predicted.

      Average Bitrate (ABR)
           In this mode, you choose the encoder will maintain an average
           bitrate while using higher bitrates for the parts of your music
           that need more bits.  The result will be of higher quality than
           CBR encoding but the average file size will remain predictable,
           so this mode is highly recommended over CBR.  This encoding mode
           is similar to what is referred as vbr in AAC or Liquid Audio (2
           other compression technologies).

      Variable bitrate (VBR)
           In this mode, you choose the desired quality on a scale from 9
           (lowest quality/biggest distortion) to 0 (highest quality/lowest
           distortion).  Then encoder tries to maintain the given quality in
           the whole file by choosing the optimal number of bits to spend
           for each part of your music.  The main advantage is that you are
           able to specify the quality level that you want to reach, but the
           inconvenient is that the final file size is totally
           unpredictable.


 PRESETS



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



      The --preset switches are aliases over LAME settings.

      To activate these presets:

      For VBR modes (generally highest quality):

      --preset medium
           This preset should provide near transparency to most people on
           most music.

      --preset standard
           This preset should generally be transparent to most people on
           most music and is already quite high in quality.

      --preset extreme
           If you have extremely good hearing and similar equipment, this
           preset will generally provide slightly higher quality than the
           standard mode.

      For CBR 320kbps (highest quality possible from the --preset switches):

      --preset insane
           This preset will usually be overkill for most people and most
           situations, but if you must have the absolute highest quality
           with no regard to filesize, this is the way to go.

      For ABR modes (high quality per given bitrate but not as high as VBR):

      --preset  kbps
           Using this preset will usually give you good quality at a
           specified bitrate.  Depending on the bitrate entered, this preset
           will determine the optimal settings for that particular
           situation.  While this approach works, it is not nearly as
           flexible as VBR, and usually will not attain the same level of
           quality as VBR at higher bitrates.

      cbr  If you use the ABR mode (read above) with a significant bitrate
           such as 80, 96, 112, 128, 160, 192, 224, 256, 320, you can use
           the --preset cbr  kbps option to force CBR mode encoding instead
           of the standard ABR mode.  ABR does provide higher quality but
           CBR may be useful in situations such as when streaming an MP3
           over the Internet may be important.



 EXAMPLES
      Fixed bit rate jstereo 128kbs encoding:

           lame -b 128 sample.wav sample.mp3



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 lame(1)                          LAME 3.99                          lame(1)
 LAME audio compressor                                 LAME audio compressor

                              December 08, 2013



           Fixed bit rate jstereo 128 kbps encoding, highest quality:

           lame -q 0 -b 128 sample.wav sample.mp3

           To disable joint stereo encoding (slightly faster, but less
           quality at bitrates <= 128 kbps):

           lame -m s sample.wav sample.mp3

           Variable bitrate (use -V n to adjust quality/filesize):

           lame -V 2 sample.wav sample.mp3

           Streaming mono 22.05 kHz raw pcm, 24 kbps output:

           cat inputfile | lame -r -m m -b 24 -s 22.05 - - > output

           Streaming mono 44.1 kHz raw pcm, with downsampling to 22.05 kHz:

           cat inputfile | lame -r -m m -b 24 --resample 22.05 - - > output

           Encode with the standard preset:

           lame --preset standard sample.wav sample.mp3


 BUGS
      Probably there are some.

 SEE ALSO
      mpg123(1), madplay(1), sox(1)

 AUTHORS
      LAME originally developed by Mike Cheng and now maintained by
      Mark Taylor, and the LAME team.

      GPSYCHO psycho-acoustic model by Mark Taylor.
      (See http://www.mp3dev.org/).

      mpglib by Michael Hipp

      Manual page by William Schelter, Nils Faerber, Alexander Leidinger,
      and Rog['e]rio Brito.









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