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
              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 Rogerio Brito.



LAME 3.99                        July 08, 2008                         lame(1)
