Provided by: lame_3.100-2build1_amd64 bug

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).

              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 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
              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.
              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.

              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
              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
              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.

       -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  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.
              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
              * 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 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 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

       LAME is able to embed ID3 v1, v1.1 or v2 tags inside the encoded MP3  file.   This  allows
       one 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

       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

       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 Rogério Brito.