# Full command line switch reference

switch | parameter
`-a` | downmix stereo file to mono
`--athonly` | ATH only
`-b` | bitrate (8...320)
`-B` | max VBR bitrate (8...320)
`-c` | copyright
`--comp` | choose compression ratio
`-d` | block type control
`--decode` | decoding only
`-e` | de-emphasis (n, 5, c)
`-f` | fast mode
`-F` | strictly enforce the -b option
`--freeformat` | free format bitstream
`-g` | turn on MP3x graphical frame analyser (may not be available on your release)
`-h` | high quality
`--help` | help
`--highpass` | highpass filtering frequency in kHz
`--highpass-width` | width of highpass filtering in kHz
`-k` | full bandwidth
`--lowpass` | lowpass filtering frequency in kHz
`--lowpass-width` | width of lowpass filtering in kHz
`-m` | stereo mode (s, j, f, m)
`--mp3input` | mp3 input file
`--noath` | no ATH
`--nohist` | disable histogram display
`--nores` | disable bit reservoir
`--noshort` | disable short blocks frames
`-o` | non-original
`-p` | error protection
`--preset` | use built-in preset (phone, voice, fm, tape, hifi, cd, studio)
`-r` | input file is raw pcm
`--resample` | output sampling frequency in kHz
`-s` | sampling frequency in kHz
`-S` | silent operation
`-t` | disable VBR tag
`-V` | VBR quality setting (0...9)
`--voice` | voice encoding mode
`-x` | swapbytes

* `-a` downmix
  Mix the stereo input file to mono and encode as mono. 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 m" will always produce a mono .mp3 file from both mono and stereo input.
  The downmix is calculated as the sum of the left and right channel, attenuated by 6 dB.
* `--athonly` ATH only

  This option causes LAME to ignore the output of the psy-model and only use masking from the ATH (absolute threshold of hearing). Might be useful at very high bitrates or for testing the ATH.

* `-b` bitrate

  For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz)
  n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
  For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz)
  n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
  Default is 128 kbs for MPEG1 and 80 kbs for MPEG2.
  When used with variable bitrate encoding (VBR), -b specifies the minimum bitrate to be used. However, during silence, the smallest frame size available will be used.

* `-B` maximum VBR bitrate

  For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz)
  n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
  For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz)
  n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
  Specifies the maximum allowed bitrate when using VBR.
  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.

* `-c` copyright

  Mark the encoded file as being copyrighted.
* `--comp` choose compression ratio

  Instead of choosing bitrate, using this option, user can choose compression ratio to achieve.
* `-d` block type control

  Allows the left and right channels to use different block size types.
* `--decode` decoding only

  Use LAME for decoding.
  input=mp3 file, output=raw pcm
* `-e n/5/c` de-emphasis

  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.
* `-f` fast mode

  This switch forces the encoder to use the fast encoding mode. It will reduce a lot the quality of the output. This might be useful for real-time processing or streaming on slowest computers.
* `-F` strictly enforce the -b option

  This is mainly for use with hardware players that do not support low bitrate mp3.
* `--freeformat` free format bitstream

  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.
* `-h` high quality

  Enables specialized mid/side masking thresholds to be used in jstereo mode and also enables a more accurate but slightly slower quantization formula. It will sound better but is about 20% slower.
  Recommended setting.
  In the future there may be more quality settings, ranging from 0 (highest) to 9 (lowest) but right now there are just 3: 2,5 or 9. They are enabled with:

  -f 'fast mode' | quality 9
  default | quality 5
  -h 'high quality' | quality 2

* `--help` help

  Display a list of all available options.
* `--highpass` highpass filtering frequency in kHz

  Set an highpass filtering frequency. Frequencies below the specified one will be cutoff.
* `--highpass-width` width of highpass filtering in kHz

  Set the width of the highpass filter. The default value is 15% of highpass frequency.
* `-k` full bandwidth

  Tells the encoder to use full bandwidth. By default, the encoder uses some highpass filtering at low bitrates, in order to keep a good quality. Increasing the bandwidth from the default setting might produces ringing artefacts at low bitrates. Use with care!
* `--lowpass` lowpass filtering frequency in kHz

  Set a lowpass filtering frequency. Frequencies above the specified one will be cutoff.
* `--lowpass-width` width of lowpass filtering in kHz

  Set the width of the lowpass filter. The default value is 15% of lowpass frequency.
* `-m s/j/f/m` stereo mode

  Joint-stereo is the default mode for stereo files with VBR or fixed bitrates of 160kbs or less. At higher fixed bitrates, the default is stereo.
  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.
  joint stereo
  In this mode, the encoder will make use of a correlation between both channels. The signal will be matrixed into a sum ("mid") and difference ("side") signal. For quasi-mono signals, this will give a significant gain in encoding quality.
  This mode does not destroy phase information like IS stereo that may be used by other encoders and thus can be used to encode DOLBY ProLogic surround signals.
  forced joint stereo
  This mode will force MS joint stereo on all frames. It's 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.
  mono
  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.
* `--mp3input` mp3 input file

  Assume the input file is a MP3 file. Usefull for downsampling from one mp3 to another. As an example, it can be usefull 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 dont end in .mp3 you need to use this switch.

* `--noath` no ATH

  Disable any use of the ATH (absolute threshold of hearing) for masking.

* `--nohist` disable histogram display

  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.

* `--nores` disable bit reservoir

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

* `--noshort` disable short blocks frames

  Encode all frames using long blocks only. This could increase quality when encoding at very low bitrates, but can produce serious pre-echo artefacts.

* `-o` non-original

  Mark the encoded file as a copy.
* `-p` error protection

  Turn on CRC error protection.
* `--preset phone/voice/fm/tape/hifi/cd/studio` use built-in preset

  Use one of the built-in presets.
* `-r` input file is raw pcm

  Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo must be specified on the command line. 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.
* `--resample` output sampling frequency in kHz

  n = 16, 22.05, 24, 32, 44.1, 48
  Resample the input to the specified frequency. If not specified, LAME will automatically resample the input when using high compression ratios.

* `-s 16/22.05/24/32/44.1/48` sampling frequency

  Required only for raw PCM input files. Otherwise it will be determined from the header.
* `-S` silent operation

  Don't print progress report.
* `-t` disable VBR tag

  Disable writing of the VBR Tag (only valid in VBR mode).
  This tag in embedded in frame 0 of the MP3 file. It lets VBR aware players correctly seek and compute playing times of VBR files.
* `-V 0...9` VBR quality setting

  Enable VBR (Variable BitRate) and specifies the value of VBR quality.
  default=4
  0=highest quality.

* `--voice` voice encoding mode

  Use voice encoding mode. In the case of voice encoding, it will increase the quality a low bitrates, or reduce the file size when using VBR.
  However, be carefull to only use it on 44.1kHz files.

* `-x` swapbytes

  Swap bytes in the input file. For sorting out little endian/big endian type problems. If your encodings sounds like static, try this first.