Provided by: lame_3.99.3+repack1-1_amd64 bug

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 compression  artifacts.
              To  much  switching  between  mid/side  and  regular stereo can also sound bad.  To
              determine when to switch to mid/side stereo, LAME uses a  much  more  sophisticated
              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 calculated 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 channel, attenuated by 6
              dB.

              This option is only needed in the case of raw PCM stereo input (because LAME cannot
              determine the number of channels in the input file).  To encode a stereo PCM  input
              file as mono, use lame -m s -a.

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

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

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

              However, even if an mp3 decoder is required to 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 format:
              FreeAmp up to 440 kbps
              in_mpg123 up to 560 kbps
              l3dec up to 310 kbps
              LAME up to 560 kbps
              MAD up to 640 kbps

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

              If -t is used (disable wav header), LAME will  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 disabled in the build
              of LAME.

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

              When --decode is specified (decode to WAV), this flag will disable writing  of  the
              WAV  header.   The  output  will  be raw pcm, native endian format.  Use -x to swap
              bytes.

       --comp arg
              Instead of choosing bitrate, using this option, user can choose  compression  ratio
              to achieve.

       --scale n
       --scale-l n
       --scale-r n
              Scales  input  (every channel, only left channel or only right channel) by n.  This
              just multiplies the PCM data (after it has been converted to floating point) by n.

              n > 1: increase volume
              n = 1: no effect
              n < 1: reduce volume

              Use with care, since most MP3 decoders will 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  compressed  data  stream
              itself,  it  is  a  value stored in the header of a sound file.  Information on the
              purpose   of   ReplayGain   and   the   algorithms   used   is    available    from
              http://www.replaygain.org/.

              Only  the  "RadioGain"  Replaygain value is computed, it is stored in the LAME tag.
              The analysis is 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 compressed data stream
              itself, it is a value stored in the header of a sound  file.   Information  on  the
              purpose    of    ReplayGain   and   the   algorithms   used   is   available   from
              http://www.replaygain.org/.

              By default, LAME performs ReplayGain analysis on the input data  (after  the  user‐
              specified  volume  scaling).   This behavior might give slightly 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 determined  and  stored  in
                   the  file.   The  knowledge  of  the  peak  sample  can  be useful to decoders
                   (players) to prevent a  negative  effect  called  'clipping'  that  introduces
                   distortion 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 disabled in  the  build
              of  LAME.   (Note: if LAME is compiled without the MP3 decoder, ReplayGain analysis
              is performed on the input data after user-specified volume scaling).

              See also: --replaygain-fast, --noreplaygain --clipdetect

       --noreplaygain
              Disable ReplayGain analysis.

              By default ReplayGain analysis is enabled. This switch disables it.

              See also: --replaygain-fast, --replaygain-accurate

       --clipdetect
              Clipping detection.

              Enable --replaygain-accurate and print a message whether clipping  occurs  and  how
              far in dB the waveform is from full scale.

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

              See also: --replaygain-accurate

       --preset  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 producing  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 scalefactors 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 quality.

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

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

              Noise shaping will be disabled, but psycho 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 otherwise be used for encoding, and then will slightly reduce
              the sound quality.

       --nores
              Disable the bit reservoir.  Each frame will then become independent  from  previous
              ones, but the quality will be lower.

       --strictly-enforce-ISO
              With this option, LAME will enforce the 7680 bit limitation on total frame size.
              This  results in many wasted bits for high bitrate encodings but will ensure strict
              ISO compatibility.  This compatibility might be important for hardware players.

       Filter options:

       --lowpass freq
              Set a lowpass filtering frequency in kHz.  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 predicted.

       Average Bitrate (ABR)
              In  this  mode, you choose the encoder will maintain an average bitrate while using
              higher bitrates for the parts of your music that need more bits.  The  result  will
              be  of  higher  quality  than  CBR  encoding  but the average file size will remain
              predictable, so this mode is highly recommended over CBR.  This  encoding  mode  is
              similar  to  what  is  referred  as vbr in AAC or Liquid Audio (2 other compression
              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 specified bitrate.
              Depending on the bitrate entered, this preset will determine the  optimal  settings
              for  that  particular  situation.   While  this approach works, it is not nearly as
              flexible as VBR, and usually will not attain the same level of quality  as  VBR  at
              higher bitrates.

       The following options are also available for the corresponding profiles:

       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 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 sample.wav sample.mp3

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

              lame -h sample.wav sample.mp3

       Fixed bit rate jstereo 112 kbps encoding:

              lame -b 112 sample.wav sample.mp3

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

              lame -m s sample.wav sample.mp3

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

              lame -f sample.wav sample.mp3

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

              lame -h -V 6 sample.wav sample.mp3

       Streaming mono 22.05 kHz raw pcm, 24 kbps output:

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

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

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

       Encode with the 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 Rogério Brito.