


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



LAME 3.98                 July 08, 2008                         1





lame(1)               LAME audio compressor               lame(1)


              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



LAME 3.98                 July 08, 2008                         2





lame(1)               LAME audio compressor               lame(1)


              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



LAME 3.98                 July 08, 2008                         3





lame(1)               LAME audio compressor               lame(1)


              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



LAME 3.98                 July 08, 2008                         4





lame(1)               LAME audio compressor               lame(1)


              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



LAME 3.98                 July 08, 2008                         5





lame(1)               LAME audio compressor               lame(1)


              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



LAME 3.98                 July 08, 2008                         6





lame(1)               LAME audio compressor               lame(1)


       --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




LAME 3.98                 July 08, 2008                         7





lame(1)               LAME audio compressor               lame(1)


       --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.




LAME 3.98                 July 08, 2008                         8





lame(1)               LAME audio compressor               lame(1)


       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.

       --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-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 con-
              sidered the "workhorse" VBR algorithm.

       --vbr-new
              Invokes  the  newest  VBR  algorithm.   During  the



LAME 3.98                 July 08, 2008                         9





lame(1)               LAME audio compressor               lame(1)


              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
              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.




LAME 3.98                 July 08, 2008                        10





lame(1)               LAME audio compressor               lame(1)


       -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).


       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.



LAME 3.98                 July 08, 2008                        11





lame(1)               LAME audio compressor               lame(1)


              -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 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.



LAME 3.98                 July 08, 2008                        12





lame(1)               LAME audio compressor               lame(1)


       -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.  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)



LAME 3.98                 July 08, 2008                        13





lame(1)               LAME audio compressor               lame(1)


       --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)

       --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



LAME 3.98                 July 08, 2008                        14





lame(1)               LAME audio compressor               lame(1)


              compiled without this.)


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



LAME 3.98                 July 08, 2008                        15





lame(1)               LAME audio compressor               lame(1)


              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 --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
       cbr  kbps





LAME 3.98                 July 08, 2008                        16





lame(1)               LAME audio compressor               lame(1)


       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
              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:




LAME 3.98                 July 08, 2008                        17





lame(1)               LAME audio compressor               lame(1)


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


       Encode with the fast standard preset:

              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 Rog‚rio Brito.































LAME 3.98                 July 08, 2008                        18


