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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 "-" for <outfile>.

OPTIONS
       Input options:

       -r     Assume  the  input  file	is  raw	 pcm.	 Sampling   rate   and
	      mono/stereo/jstereo  must	be specified on	the command line.  For
	      each stereo sample, LAME expects the input data  to  be  ordered
	      left channel first, then right channel. By default, LAME expects
	      them to be signed	integers with a	bitwidth of 16 and  stored  in
	      little-endian.   Without -r, LAME	will perform several fseek()'s
	      on the input file	looking	for WAV	and AIFF headers.
	      Might not	be available on	your release.

       -x     Swap bytes in the	input file (or output file  when  using	 --de-
	      code).
	      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 de-
	      termined 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 de-
	      termined 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-en-
	      dian form.

	      Required only for	raw PCM	input files.

       --big-endian
	      Instructs	LAME that the samples from the input are in big-endian
	      form.

	      Required only for	raw PCM	input files.

       --mp1input
	      Assume the input file is a MPEG Layer I (ie MP1) file.
	      If the filename ends in ".mp1" LAME will assume  it  is  a  MPEG
	      Layer  I	file.	For stdin or Layer I files which do not	end in
	      .mp1 you need to use this	switch.

       --mp2input
	      Assume the input file is a MPEG Layer II (ie MP2)	file.
	      If the filename ends in ".mp2" LAME will assume  it  is  a  MPEG
	      Layer  II	file.  For stdin or Layer II files which do not	end in
	      .mp2 you need to use this	switch.

       --mp3input
	      Assume the input file is a MP3 file.
	      Useful for downsampling from one mp3 to another.	As an example,
	      it can be	useful for streaming through an	IceCast	server.
	      If  the  filename	 ends in ".mp3"	LAME will assume it is an MP3.
	      For stdin	or MP3 files which do not end in .mp3 you need to  use
	      this switch.

       --nogap file1 file2 ...
	      gapless encoding for a set of contiguous files

       --nogapout dir
	      output dir for gapless encoding (must precede --nogap)

       --out-dir dir
	      If  no explicit output file is specified,	a file will be written
	      at given path.  Ignored when using piped/streamed	input

       Operational options:

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

	      Joint-stereo is the default mode for stereo files.

	      (s)imple stereo (Forced LR)
	      In this mode, the	encoder	makes no use of	 potentially  existing
	      correlations  between  the two input channels.  It can, however,
	      negotiate	the bit	demand between both  channel,  i.e.  give  one
	      channel  more  bits  if the other	contains silence or needs less
	      bits because of a	lower complexity.

	      (j)oint stereo
	      In this mode, the	encoder	can use	(on a frame  by	 frame	basis)
	      either  L/R  stereo or mid/side stereo.  In mid/side stereo, the
	      mid (L+R)	and side (L-R) channels	are encoded, and more bits are
	      allocated	 to the	mid channel than the side channel.  When there
	      isn't too	much stereo separation,	this effectively increases the
	      bandwidth,  so  having  higher  quality  with the	same amount of
	      bits.

	      Using mid/side stereo inappropriately can	result in audible com-
	      pression	artifacts.   Too  much	switching between mid/side and
	      regular stereo can also sound bad.  To determine when to	switch
	      to  mid/side  stereo,  LAME uses a much more sophisticated algo-
	      rithm than the one described in the ISO documentation.

	      (f)orced MS stereo
	      Forces all frames	to be encoded with mid/side stereo. It	should
	      be  used only if you are sure that every frame of	the input file
	      has very little stereo separation.

	      (d)ual channel
	      In this mode, the	2 channels will	be totally  independently  en-
	      coded.   Each  channel  will  have  exactly half of the bitrate.
	      This mode	is designed for	applications like dual	languages  en-
	      coding  (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.  Also	note that, if using a stereo RAW PCM stream, you  need
	      to use the -a parameter.

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

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

       -a     Mix the stereo input file	to mono	and encode as mono.
	      The downmix is calculated	as the sum of the left and right 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 in-
	      put file).  To encode a stereo RAW PCM input file	as  mono,  use
	      lame -a -m m

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

       --freeformat
	      Produces a free format bitstream.	 With this option, you can use
	      -b with any bitrate higher than 8	kbps.

	      However,	even if	an mp3 decoder is required to support free bi-
	      trates 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:
	      in_mpg123	up to 560 kbps
	      l3dec up to 310 kbps
	      LAME up to 640 kbps
	      MAD up to	640 kbps

       --decode
	      Uses LAME	for decoding to	a wav file.  The input file can	be any
	      input  type  supported  by  encoding,  including layer II	files.
	      LAME uses	a fork of mpglib known as HIP for decoding.

	      If -t is used (disable wav header), LAME will output raw pcm  in
	      native endian format.  You can use -x to swap bytes order.

	      This option is not usable	if the MP3 decoder was explicitly dis-
	      abled in the build of LAME.

       -t     Disable writing of the INFO Tag on encoding.
	      This tag is embedded in frame 0 of the MP3  file.	  It  includes
	      some  information	about the encoding options of the file,	and in
	      VBR it lets VBR aware players correctly seek and compute playing
	      times of VBR files.

	      When  --decode is	specified (decode to WAV), this	flag will dis-
	      able writing of the WAV header.  The output will be raw pcm, na-
	      tive 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  computes  "Radio"  ReplayGain  on the decoded data stream,
	      finds the	peak sample by decoding	on the fly  the	 encoded  data
	      stream and stores	it in the file.

	      The  ReplayGain  analysis	 does not affect the content of	a 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.

       --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).

	      For CBR and ABR, the following table applies:

	      -q 0:
	      Use  the best algorithms (Best Huffman coding search, full outer
	      loop, and	the highest precision of several parameters).

	      -q 1 to q	4:
	      Similar to -q 0 without the full outer loop and decreasing  pre-
	      cision of	parameters the further from q0.	-q 3 is	the default.

	      -q 5 and -q 6:
	      Same  as	-q 7, but enables noise	shaping	and increases subblock
	      gain

	      -q 7 to -q 9:
	      Same as -f. Very fast, OK	quality. Psychoacoustics are used  for
	      pre-echo and mid/side stereo, but	no noise-shaping is done.

	      For  the	default	 VBR mode since	LAME 3.98, the following table
	      applies :

	      -q 0 to -q 4:
	      include all features of the other	modes and additionally use the
	      best search when applying	Huffman	coding.

	      -q 5 and -q 6:
	      include all features of -q7, calculate and consider actual quan-
	      tisation noise, and additionally enable subblock gain.

	      -q 7 to -q 9
	      This level uses a	psymodel but does not  calculate  quantisation
	      noise when encoding: it takes a quick guess.

       -h     Alias of -q 2

       -f     Alias of -q 7

       CBR (constant bitrate, the default) options:

       -b n   For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz)
	      n	 =  32,	 40, 48, 56, 64, 80, 96, 112, 128, 160,	192, 224, 256,
	      320

	      For MPEG-2 (sampling frequencies of 16, 22.05 and	24 kHz)
	      n	= 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144,	160

	      For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)
	      n	= 8, 16, 24, 32, 40, 48, 56, 64

	      Default is 128 for MPEG1 and 64 for MPEG2	and 32 for MPEG2.5
	       (64, 32 and 16 respectively in case of mono).

       --cbr  enforce use of constant bitrate. Used to disable VBR or ABR  en-
	      coding even if their settings are	enabled.

       ABR (average bitrate) options:

       --abr n
	      Turns  on	 encoding  with	a targeted average bitrate of n	kbits,
	      allowing to use frames of	different sizes.  The allowed range of
	      n	is 8 - 310, you	can use	any integer value within that range.

	      It  can be combined with the -b and -B switches like: lame --abr
	      123 -b 64	-B 192 a.wav a.mp3 which would limit the allowed frame
	      sizes between 64 and 192 kbits.

	      The use of -B is NOT RECOMMENDED.	 A 128 kbps CBR	bitstream, be-
	      cause of the bit reservoir, can actually have frames  which  use
	      as many bits as a	320 kbps frame.	 VBR modes minimize the	use of
	      the bit reservoir, and thus need to allow	320 kbps frames	to get
	      the same flexibility as CBR streams.

       VBR (variable bitrate) options:

       -v     use variable bitrate (--vbr-new)

       --vbr-old
	      Invokes the oldest, most tested VBR algorithm.  It produces very
	      good quality files, though is  not  very	fast.	This  has,  up
	      through v3.89, been considered the "workhorse" VBR algorithm.

       --vbr-new
	      Invokes  the  newest  VBR	 algorithm.  During the	development of
	      version 3.90, considerable tuning	was done  on  this  algorithm,
	      and  it  is now considered to be on par with the original	--vbr-
	      old.  It has the added advantage of being	very fast (over	 twice
	      as fast as --vbr-old ). This is the default since	3.98.

       -V n   0	<= n <=	9.999
	      Enable  VBR  (Variable  BitRate)	and specifies the value	of VBR
	      quality (default = 4). Decimal values  can  be  specified,  like
	      4.51.
	      0	= highest quality.

       ABR and VBR options:

       -b bitrate
	      For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz)
	      n	 =  32,	 40, 48, 56, 64, 80, 96, 112, 128, 160,	192, 224, 256,
	      320

	      For MPEG-2 (sampling frequencies of 16, 22.05 and	24 kHz)
	      n	= 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144,	160

	      For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)
	      n	= 8, 16, 24, 32, 40, 48, 56, 64

	      Specifies	the minimum bitrate to be used.	 However, in order  to
	      avoid  wasted  space,  the smallest frame	size available will be
	      used during silences.

       -B bitrate
	      For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz)
	      n	= 32, 40, 48, 56, 64, 80, 96, 112, 128,	160,  192,  224,  256,
	      320

	      For MPEG-2 (sampling frequencies of 16, 22.05 and	24 kHz)
	      n	= 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144,	160

	      For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)
	      n	= 8, 16, 24, 32, 40, 48, 56, 64

	      Specifies	the maximum allowed bitrate.

	      Note:  If	 you  own an mp3 hardware player build upon a MAS 3503
	      chip, you	must set maximum bitrate to no more than 224 kpbs.

       -F     Strictly enforce the -b option.
	      This is mainly for use with hardware players that	do not support
	      low bitrate mp3.

	      Without  this  option,  the  minimum bitrate will	be ignored for
	      passages of analog silence, i.e. when the	music level  is	 below
	      the absolute threshold of	human hearing (ATH).

       Experimental options:

       -X n   0	<= n <=	7

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

       --tv id=value
	      Text or URL frame	specified by id	and value (v2.3	tag). User de-
	      fined frame. Syntax: --tv	"TXXX=description=content"

       --add-id3v2
	      force addition of	version	2 tag

       --id3v1-only
	      add only a version 1 tag

       --id3v2-only
	      add only a version 2 tag

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

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

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

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

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

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

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

       Analysis	options:

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

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

       cbr    If  you use the ABR mode (read above) with a significant bitrate
	      such as 80, 96, 112, 128,	160, 192, 224, 256, 320, you  can  use
	      the --preset cbr	kbps option to force CBR mode encoding instead
	      of the standard ABR mode.	 ABR does provide higher  quality  but
	      CBR  may	be  useful in situations such as when streaming	an MP3
	      over the Internet	may be important.

EXAMPLES
       Fixed bit rate jstereo 128kbs encoding:

	      lame -b 128 sample.wav sample.mp3

       Fixed bit rate jstereo 128 kbps encoding, highest quality:

	      lame -q 0	-b 128 sample.wav sample.mp3

       To disable joint	stereo encoding	(slightly faster, but less quality  at
       bitrates	<= 128 kbps):

	      lame -m s	sample.wav sample.mp3

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

	      lame -V 2	sample.wav sample.mp3

       Streaming mono 22.05 kHz	raw pcm, 24 kbps output:

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

       Streaming mono 44.1 kHz raw pcm,	with downsampling to 22.05 kHz:

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

       Encode with the standard	preset:

	      lame --preset standard sample.wav	sample.mp3

BUGS
       Probably	there are some.

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

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

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

       mpglib by Michael Hipp

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

LAME 3.99		       December	08, 2013		       lame(1)

NAME | SYNOPSIS | DESCRIPTION | OPTIONS | ID3 TAGS | ENCODING MODES | PRESETS | EXAMPLES | BUGS | SEE ALSO | AUTHORS

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