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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.	  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.

       --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

	      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	 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 will make use of a correlation 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 signal does  not  have
	      too  much	 stereo	 separation, thus giving a significant gain in
	      encoding quality.

	      Using mid/side stereo inappropriately can result in audible com-
	      pression artifacts.  To much switching between mid/side and reg-
	      ular stereo can also sound bad.  To determine when to switch  to
	      mid/side	stereo,	 LAME uses a much more sophisticated algorithm
	      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	 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	calcu-
	      lated  as the sum of the left and right channel, attenuated by 6
	      dB.

       -a     Mix the stereo input file to mono and encode as mono.
	      The downmix is calculated as the sum of the left and right chan-
	      nel, 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 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 for-
	      mat:
	      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 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 dis-
	      abled 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 flag will dis-
	      able 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  com-
	      pressed  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.replay-
	      gain.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 enables decoding on the fly, computes "Radio" ReplayGain on
	      the decoded data stream, finds the peak sample  of  the  decoded
	      data stream and stores it in the file.

	      The  ReplayGain  analysis	 does not affect the content of a com-
	      pressed 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.replay-
	      gain.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	deter-
		   mined  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 distor-
		   tion 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 dis-
	      abled 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 clip-
	      ping occurs and how far in dB the waveform is from full scale.

	      This option is not usable if the MP3 decoder was explicitly dis-
	      abled 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 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 produc-
	      ing 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  scalefac-
	      tors and Huffman encoding (noise shaping).

	      -q 0:
	      use slowest & best possible version of all algorithms.  -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 acoustics are used
	      for pre-echo & M/S, but no noise shaping is done.

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

       -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 acoustics will still
	      be computed for bit allocation and pre-echo detection.


       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 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).

       -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.

	      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.

	      -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 stan-
	      dalone 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 oth-
	      erwise be used for encoding, and then will slightly  reduce  the
	      sound quality.

       --nores
	      Disable the bit reservoir.  Each frame will then become indepen-
	      dent 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)

       --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.)


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	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 pre-
	      dicted.

       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 speci-
	      fied bitrate.  Depending on the  bitrate	entered,  this	preset
	      will  determine  the optimal settings for that particular situa-
	      tion.  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  corresponding  pro-
       files:

       fast standard|extreme
       cbr  kbps


       fast   Enables the new fast VBR for a particular profile.

       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  cbr	option to force CBR mode encoding instead of the stan-
	      dard 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 sample.wav sample.mp3


       Fixed bit rate jstereo  128  kbps  encoding,  highest  quality  (recom-
       mended):

	      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:

	      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 Rogerio Brito.



LAME 3.98			 July 08, 2008			       lame(1)
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