


lame(1)               LAME audio compressor               lame(1)


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 a "-" 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.  With-
              out  -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 prob-
              lems.  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



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              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.

       --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)


       Operational options:

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



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              Joint-stereo is the default mode for  stereo  files
              with  VBR  when -V is more than 4 or fixed bitrates
              of 160kbs or less.  At  higher  fixed  bitrates  or
              higher VBR settings, the default is stereo.

              (s)imple stereo
              In  this  mode,  the encoder makes no use of poten-
              tially 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 will make use of a corre-
              lation  between  both channels.  The signal will be
              matrixed into a sum ("mid"), computed by  L+R,  and
              difference  ("side")  signal,  computed by L-R, and
              more bits are allocated to the mid  channel.   This
              will effectively increase the bandwidth if the sig-
              nal does not have too much stereo separation,  thus
              giving a significant gain in encoding quality.

              Using mid/side stereo inappropriately can result in
              audible compression artifacts.  To  much  switching
              between  mid/side and regular stereo can also sound
              bad.  To  determine  when  to  switch  to  mid/side
              stereo,  LAME  uses a much more sophisticated algo-
              rithm than that described in the ISO documentation,
              and thus is safe to use in joint stereo mode.

              (f)orced MS stereo
              This  mode  will force MS stereo on all frames.  It
              is slightly faster than joint stereo, but it should
              be  used  only  if you are sure that every frame of
              the input file has very little stereo separation.

              (d)ual mono
              In this mode, the 2 channels will be totally  inde-
              pendently  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  encod-
              ing.

              (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.

       -a     Mix the stereo input file to  mono  and  encode  as



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              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 PCM input file as mono, use lame -m s -a.

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

       -d     Allows the left and right channels to use different
              block size types.

       --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 sup-
              port  free  bitrates  at least up to 320 kbps, many
              players are unable to deal with it.

              Tests have shown that the following  decoders  sup-
              port free format:
              FreeAmp up to 440 kbps
              in_mpg123 up to 560 kbps
              l3dec up to 310 kbps
              LAME up to 560 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 bugfixed
              version of mpglib for decoding.

              If -t is used (disable wav header), LAME will  out-
              put  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  in  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



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              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  trun-
              cate  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 per-
              formed 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  enables decoding on the fly, computes "Radio"
              ReplayGain on the decoded data  stream,  finds  the
              peak  sample  of the decoded data stream and stores



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              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 inac-
              curate 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  dif-
                   ference  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 sam-
                   ple 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 dis-
                   tortion into the sound.

              Only the "RadioGain" ReplayGain value is  computed,
              it is stored in the LAME tag.  The analysis is per-
              formed 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



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       --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 wave-
              form 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  [fast] 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.

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

       --noasm  type
              Disable specific assembly  optimizations  (  mmx  /
              3dnow  /  sse  ).   Quality will not increase, only
              speed will be reduced.  If you have  problems  run-
              ning  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




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       --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  algo-
              rithms to determine the best scalefactors and Huff-
              man encoding (noise shaping).

              -q 0:
              use slowest & best possible version  of  all  algo-
              rithms.  -q 0 and -q 1 are slow and may not produce
              significantly higher quality.

              -q 2:
              recommended.  Same as -h.

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

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

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

       -h     Use  some  quality  improvements.  Encoding will be
              slower, but the result will be of  higher  quality.
              The behavior is the same as the -q 2 switch.
              This switch is always enabled when using VBR.

       -f     This  switch  forces  the  encoder  to use a faster
              encoding mode,  but  with  a  lower  quality.   The
              behavior is the same as the -q 7 switch.

              Noise  shaping  will be disabled, but psycho acous-
              tics will still be computed for bit allocation  and
              pre-echo detection.




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       CBR (constant bitrate, the default) options:

       -b n   For  MPEG1 (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 MPEG2 (sampling frequencies of 16, 22.05 and 24
              kHz)
              n = 8, 16, 24, 32, 40, 48, 56,  64,  80,  96,  112,
              128, 144, 160

              Default is 128 for MPEG1 and 64 for MPEG2.

       --cbr  enforce use of constant bitrate


       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  actu-
              ally  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-old)

       --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  con-
              sidered 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 consid-
              ered to be on par with the original --vbr-old.   It
              has  the  added  advantage of being very fast (over



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              twice as fast as --vbr-old).

       -V n   0 <= n <= 9
              Enable VBR (Variable  BitRate)  and  specifies  the
              value  of  VBR  quality (default = 4).  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.




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              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).


       PSY related:

       --nssafejoint
              M/S switching criterion

       --nsmsfix arg
              M/S switching tuning [effective 0-3.5]

       --ns-bass x
              Adjust masking for sfbs  0 -   6  (long)   0  -   5
              (short)

       --ns-alto x
              Adjust  masking  for  sfbs   7  - 13 (long)  6 - 10
              (short)

       --ns-treble x
              Adjust masking for sfbs 14 -  21  (long)  11  -  12
              (short)

       --ns-sfb21 x
              Change ns-treble by x dB for sfb21


       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 criterions 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.



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

              -X6
              The criterions 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 criterions 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.



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              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.  Frequen-
              cies 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.   Fre-
              quencies 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)



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       --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)

       --add-id3v2
              force addition of version 2 tag

       --id3v1-only
              add only a version 1 tag

       --id3v2-only
              add only a version 2 tag

       --space-id3v1
              pad version 1 tag with spaces instead of nulls

       --pad-id3v2
              same as --pad-id3v2-size 128

       --pad-id3v2-size num
              adds version 2 tag, pad with extra "num" bytes

       --genre-list
              print alphabetically sorted ID3 genre list and exit

       --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.)





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lame(1)               LAME audio compressor               lame(1)


ID3 TAGS
       LAME is able to embed ID3 v1, v1.1 or v2 tags  inside  the
       encoded  MP3 file.  This allows to have some useful infor-
       mation 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  vari-
              able.   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 accu-
              rately 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 tech-
              nologies).

       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 qual-
              ity  level that you want to reach, but the inconve-
              nient is that the final file size is totally unpre-
              dictable.



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lame(1)               LAME audio compressor               lame(1)


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  --pre-
       set switches):

       --preset insane
              This  preset will usually be overkill for most peo-
              ple and most situations, but if you must  have  the
              absolute  highest  quality  with no regard to file-
              size, 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 qual-
              ity 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.

       The  following  options  are also available for the corre-
       sponding profiles:

       fast standard|extreme|insane
       cbr  kbps


       fast   Enables the new fast VBR for a particular  profile.

       cbr    If you use the ABR mode (read above) with a signif-
              icant bitrate such as 80, 96, 112, 128,  160,  192,
              224,  256, 320, you can use the cbr option to force



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lame(1)               LAME audio compressor               lame(1)


              CBR mode encoding instead of the standard ABR mode.
              ABR does provide higher quality but CBR may be use-
              ful in situations such as  when  streaming  an  MP3
              over the Internet may be important.



EXAMPLES
       Fixed bit rate jstereo 128kbs encoding:

              lame sample.wav sample.mp3


       Fixed  bit rate jstereo 128 kbps encoding, highest quality
       (recommended):

              lame -h sample.wav sample.mp3


       Fixed bit rate jstereo 112 kbps encoding:

              lame -b 112 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


       Fast encode, low quality (no psycho-acoustics):

              lame -f sample.wav sample.mp3


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

              lame -h -V 6 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 fast standard preset:



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lame(1)               LAME audio compressor               lame(1)


              lame --preset fast 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 Rogrio Brito.





































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