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 "-" 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, l, r

	      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.

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

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


       -a     Mix the stereo input file to mono and encode as mono.
	      The downmix is calculated as the sum of the left and right 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  usu-
		   ally	 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	 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	 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).


       Experimental options:

       -X n   0 <= n <= 7

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

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

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

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

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

	      -X4
	      Not yet documented.

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

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

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

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


       MP3 header/stream options:

       -e emp emp = n, 5, c

	      n = (none, default)
	      5 = 0/15 microseconds
	      c = citt j.17

	      All this does is set a flag in the bitstream.  If you have a PCM
	      input  file  where one of the above types of (obsolete) emphasis
	      has been applied, you can set this flag in LAME.	Then  the  mp3
	      decoder should de-emphasize the output during playback, although
	      most decoders ignore this flag.

	      A better solution would be  to  apply  the  de-emphasis  with  a
	      standalone utility before encoding, and then encode without -e.

       -c     Mark the encoded file as being copyrighted.

       -o     Mark the encoded file as being a copy.

       -p     Turn on CRC error protection.
	      It  will add a cyclic redundancy check (CRC) code in each frame,
	      allowing to detect transmission errors that could occur  on  the
	      MP3 stream.  However, it takes 16 bits per frame that would 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

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

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

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

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

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

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

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


       Analysis options:

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


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

       standard|extreme
       cbr  kbps

       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 standard preset:

	      lame --preset standard sample.wav sample.mp3


BUGS
       Probably there are some.

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

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

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

       mpglib by Michael Hipp

       Manual page by William Schelter, Nils Faerber, Alexander Leidinger,
       and Rogerio Brito.



LAME 3.99			 July 08, 2008			       lame(1)