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% lame [options] inputfile [outputfile] For more options, just type: % lame --help ======================================================================= Constant Bitrate Examples: ======================================================================= fixed bit rate jstereo 128 kbps encoding: % lame sample.wav sample.mp3 fixed bit rate jstereo 128 kbps encoding, higher quality: (recommended) % lame -h sample.wav sample.mp3 Fast encode, low quality (no noise shaping) % lame -f sample.wav sample.mp3 ======================================================================= Variable Bitrate Examples: ======================================================================= LAME has two types of variable bitrate: ABR and VBR. ABR is the type of variable bitrate encoding usually found in other MP3 encoders, Vorbis and AAC. The number of bits is determined by some metric (like perceptual entropy, or just the number of bits needed for a certain set of encoding tables), and it is not based on computing the actual encoding/quantization error. ABR should always give results equal or better than CBR: ABR: (--abr <x> means encode with an average bitrate of around x kbps) lame -h --abr 128 sample.wav sample.mp3 VBR is a true variable bitrate mode which bases the number of bits for each frame on the measured quantization error relative to the estimated allowed masking. VBR is currently under heavy development. It can on occasion result in too much compression, so it should be used with a minimum bitrate of 112 kbps. This will let LAME increase the bitrate for difficult-to-encode frames, but prevent LAME from being too aggressive for simple frames: Variable Bitrate (VBR): (use -V n to adjust quality/filesize) % lame -h -v -b 112 sample.wav sample.mp3 ======================================================================= LOW BITRATES ======================================================================= At lower bitrates, (like 24 kbps per channel), it is recommended that you use a 16 kHz sampling rate combined with lowpass filtering. LAME, as well as commercial encoders (FhG, Xing) will do this automatically. However, if you feel there is too much (or not enough) lowpass filtering, you may need to try different values of the lowpass cutoff and passband width (--resample, --lowpass and --lowpass-width options). ======================================================================= STREAMING EXAMPLES ======================================================================= % cat inputfile | lame [options] - - > output ======================================================================= Scripts are included (in the 'misc' subdirectory) to run lame on multiple files: bach script: mlame Run "mlame -?" for instructions. sh script: auenc Run auenc for instructions sh script: mugeco.sh Pearl script which will re-encode mp3 files and preserve id3 tags: lameid3.pl Windows scripts: lame4dos.bat Lame.vbs (and an HTML frontend: LameGUI.html) ======================================================================= options guide: ======================================================================= These options are explained in detail below. Quality related: -m m/s/j/f/a mode selection -k disable all filtering -d allow block types to differ between channels --athonly ignore psy-model output, only use masking from the ATH --voice (obsolete, try --preset voice instead) --noshort disable short blocks -q n Internal algorithm quality setting 0..9. 0 = slowest algorithms, but potentially highest quality 9 = faster algorithms, very poor quality -h same as -q2 -f same as -q7 Constant Bit Rate (CBR) -b n set bitrate (8, 16, 24, ..., 320) --freeformat produce a free format bitstream. User must also specify a bitrate with -b, between 8 and 640 kbps. Variable Bit Rate (VBR) -v VBR --vbr-old use old variable bitrate (VBR) routine (default) --vbr-new use new variable bitrate (VBR) routine -V n VBR quality setting (0=highest quality, 9=lowest) -b n specify a minimum allowed bitrate (8,16,24,...,320) -B n specify a maximum allowed bitrate (8,16,24,...,320) -F strictly enforce minimum bitrate -t disable VBR informational tag --nohist disable display of VBR bitrate histogram --abr n specify average bitrate desired Experimental (undocumented): may work better or worse: -X n try different quality measures (when comparing quantizations) -Y -Z Operational: -r assume input file is raw PCM -s n input sampling frequency in kHz (for raw PCM input files) --resample n output sampling frequency --mp3input input file is an MP3 file. decode using mpglib/mpg123 --ogginput input file is an Ogg Vorbis file. decode using libvorbis -x swap bytes of input file --scale <arg> multiply PCM input by <arg> --scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg> --scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg> -a downmix stereo input file to mono .mp3 -e n/5/c de-emphasis -p add CRC error protection -c mark the encoded file as copyrighted -o mark the encoded file as a copy -S don't print progress report, VBR histogram --strictly-enforce-ISO comply as much as possible to ISO MPEG spec --replaygain-fast compute RG fast but slightly inaccurately (default) --replaygain-accurate compute RG more accurately and find the peak sample --noreplaygain disable ReplayGain analysis --clipdetect enable --replaygain-accurate and print a message whether clipping occurs and how far the waveform is from full scale --decode assume input file is an mp3 file, and decode to wav. -t disable writing of WAV header when using --decode (decode to raw pcm, native endian format (use -x to swap)) --ogg Encode using Ogg Vorbis (.ogg) instead of mp3. ID3 tagging: --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> audio/song track number (1 to 255, creates v1.1 tag) --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 --space-id3v1 pad version 1 tag with spaces instead of nulls --pad-id3v2 pad version 2 tag with extra 128 bytes --genre-list print alphabetically sorted ID3 genre list and exit Note: A version 2 tag will NOT be added unless one of the input fields won't fit in a version 1 tag (e.g. the title string is longer than 30 characters), or the '--add-id3v2' or '--id3v2-only' options are used, or output is redirected to stdout. OS/2-specific options: --priority <type> sets the process priority options not yet described: --nores disable bit reservoir --noath disable ATH --athlower <n db> lower the ATH by n db. --athshort use only the ATH for short blocks --cwlimit <freq> specify range of tonality calculation --disptime --notemp disable temporal masking --lowpass --lowpass-width --highpass --highpass-width ======================================================================= Detailed description of all options in alphabetical order ======================================================================= ======================================================================= downmix ======================================================================= -a 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. ======================================================================= average bitrate encoding (aka Safe VBR) ======================================================================= --abr n turns on encoding with a targeted average bitrate of n kbps, allowing to use frames of different sizes. The allowed range of n is 8...320 kbps, you can use any integer value within that range. ======================================================================= ATH only ======================================================================= --athonly This option causes LAME to ignore the output of the psy-model and only use masking from the ATH. (absolute threshold of hearing) Using --athonly is NOT RECOMMENDED. It is designed for testing different ATH curves. ======================================================================= bitrate ======================================================================= -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 and MPEG-2.5 (sampling frequencies of 8, 11.025, 12, 16, 22.05 and 24 kHz) n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 The bitrate to be used. Default is 128 kbps MPEG1, 80 kbps MPEG2. When used with variable bitrate encodings (VBR), -b specifies the minimum bitrate to use. This is useful to prevent LAME VBR from using some very aggressive compression which can cause some distortion due to small flaws in the psycho-acoustic model. ======================================================================= max bitrate ======================================================================= -B n see also option "-b" for allowed bitrates. Maximum allowed bitrate when using VBR/ABR. Using -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. ABR/VBR modes minimize the use of the bit reservoir, and thus need to allow 320 kbps frames to get the same flexability as CBR streams. ======================================================================= copyright ======================================================================= -c mark the encoded file as copyrighted ======================================================================= clipping detection ======================================================================= --clipdetect 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 ======================================================================= block type control ======================================================================= -d Allows the left and right channels to use different block types. Normally this is not allowed, only because the FhG encoder does not seem to allow it either. If anyone finds a sample where -d produces better results, let me know. (mt@sulaco.org) ======================================================================= mpglib decode capability ======================================================================= --decode This just uses LAME's mpg123/mpglib interface to decode an MP3 file to a wav file. The input file can be any input type supported by encoding, including .mp3 (layers 1, 2 and 3) and .ogg. If -t is used (disable wav header), LAME will output raw pcm in native endian format (use -x to swap bytes). This option is not usable if the MP3 decoder was _explicitly_ disabled in the build of LAME. ======================================================================= de-emphasis ======================================================================= -e 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. ======================================================================= fast mode ======================================================================= -f Same as -q 7. NOT RECOMMENDED. Use when encoding speed is critical and encoding quality does not matter. Disable noise shaping. Psycho acoustics are used only for bit allocation and pre-echo detection. ======================================================================= strictly enforce VBR minimum bitrate ======================================================================= -F strictly enforce VBR minimum bitrate. With out this optioni, the minimum bitrate will be ignored for passages of analog silence. ======================================================================= free format bitstreams ======================================================================= --freeformat LAME will produce a fixed bitrate, free format bitstream. User must specify the desired bitrate in kbps, which can be any integer between 8 and 640. Not supported by most decoders. Complient decoders (of which there are few) are only required to support up to 320 kbps. Decoders which can handle free format: supports up to MAD 640 kbps "lame --decode" 550 kbps Freeamp: 440 kbps l3dec: 310 kbps ======================================================================= high quality ======================================================================= -h use some quality improvements. The same as -q 2. ======================================================================= keep all frequencies ======================================================================= -k keep all frequencies. (Disable all filters) LAME will automatically apply various types of lowpass filters. This is because the high frequency coefficients can take up a lot of bits that would be better used for lower, more important frequencies. -k will disable all lowpass filtering. Not recommended. ======================================================================= Modes: ======================================================================= -m m mono -m s stereo -m j joint stereo -m f forced mid/side stereo -m d dual (independent) channels -m i intensity stereo -m a auto MONO is the default mode for mono input files. If "-m m" is specified for a stereo input file, the two channels will be averaged into a mono signal. STEREO JOINT STEREO is the default mode for stereo files with fixed bitrates of 128 kbps or less. At higher fixed bitrates, the default is stereo. For VBR encoding, jstereo is the default for VBR_q >4, and stereo is the default for VBR_q <=4. You can override all of these defaults by specifing the mode on the command line. jstereo means the encoder can use (on a frame by frame bases) either regular stereo (just encode left and right channels independently) 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. This will effectively increase the bandwidth if the signal does not have too much stereo separation. Mid/side stereo is basically a trick to increase bandwidth. At 128 kbps, it is clearly worth while. At higher bitrates it is less useful. For truly mono content, use -m m, which will automatically down sample your input file to mono. This will produce 30% better results over -m j. 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. FORCED MID/SIDE STEREO forces all frames to be encoded mid/side stereo. It should only be used if you are sure every frame of the input file has very little stereo seperation. DUAL CHANNELS Not supported. INTENSITY STEREO AUTO Auto select should select (if input is stereo) 8 kbps Mono 16- 96 kbps Intensity Stereo (if available, otherwise Joint Stereo) 112-128 kbps Joint Stereo -mj 160-192 kbps -mj with variable mid/side threshold 224-320 kbps Independent Stereo -ms ======================================================================= MP3 input file ======================================================================= --mp3input Assume the input file is a MP3 file. LAME will decode the input file before re-encoding it. Since MP3 is a lossy format, this is not recommended in general. But it is useful for creating low bitrate mp3s from high bitrate mp3s. 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. ======================================================================= disable historgram display ======================================================================= --nohist By default, LAME will display a bitrate histogram while producing VBR mp3 files. This will disable that feature. ======================================================================= disable ReplayGain analysis ======================================================================= --noreplaygain By default ReplayGain analysis is enabled. This switch disables it. See also: --replaygain-accurate, --replaygain-fast ======================================================================= disable short blocks ======================================================================= --noshort Encode all frames using long blocks. NOT RECOMMENDED. For testing purposes only. ======================================================================= non-original ======================================================================= -o mark the encoded file as a copy ======================================================================= CRC error protection ======================================================================= -p turn on CRC error protection. Yes this really does work correctly in LAME. However, it takes 16 bits per frame that would otherwise be used for encoding. ======================================================================= algorithm quality selection ======================================================================= -q n 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 2: recommended. Same as -h. -q 0 and -q 1 are slow and may not produce significantly higher quality. -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. ======================================================================= input file is raw pcm ======================================================================= -r 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. Not supported if LAME is compiled to use LIBSNDFILE. ======================================================================= slightly more accurate ReplayGain analysis and finding the peak sample ======================================================================= --replaygain-accurate Enable decoding on the fly. Compute "Radio" ReplayGain on the decoded data stream. Find the peak sample of the decoded data stream and store it in the file. 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 behaviour 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 no 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 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 sound. Only the "Radio" 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 ======================================================================= fast ReplayGain analysis ======================================================================= --replaygain-fast Compute "Radio" ReplayGain of the input data stream after user-specified volume scaling and/or resampling. 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 "Radio" 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 ======================================================================= output sampling frequency in kHz ======================================================================= --resample n where n = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48 Output sampling frequency. Resample the input if necessary. If not specified, LAME may sometimes resample automatically when faced with extreme compression conditions (like encoding a 44.1 kHz input file at 32 kbps). To disable this automatic resampling, you have to use --resamle to set the output samplerate equal to the inptu samplerate. In that case, LAME will not perform any extra computations. ======================================================================= sampling frequency in kHz ======================================================================= -s n where n = sampling rate in kHz. Required for raw PCM input files. Otherwise it will be determined from the header information in the input file. LAME will automatically resample the input file to one of the supported MP3 samplerates if necessary. ======================================================================= silent operation ======================================================================= -S don't print progress report ======================================================================= scale ======================================================================= --scale <arg> Scales input by <arg>. This just multiplies the PCM data (after it has been converted to floating point) by <arg>. <arg> > 1: increase volume <arg> = 1: no effect <arg> < 1: reduce volume Use with care, since most MP3 decoders will truncate data which decodes to values greater than 32768. ======================================================================= strict ISO complience ======================================================================= --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. ======================================================================= disable VBR tag ======================================================================= -t Disable writing of the VBR Tag (only valid if -v flag is specified) 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. When '--decode' is specified (decode mp3 to wav), this flag will disable writing the WAV header. The output will be raw pcm, native endian format. Use -x to swap bytes. ======================================================================= variable bit rate (VBR) ======================================================================= -v Turn on VBR. There are several ways you can use VBR. I personally like using VBR to get files slightly bigger than 128 kbps files, where the extra bits are used for the occasional difficult-to-encode frame. For this, try specifying a minimum bitrate to use with VBR: lame -v -b 112 input.wav output.mp3 If the file is too big, use -V n, where n = 0...9 lame -v -V n -b 112 input.wav output.mp3 If you want to use VBR to get the maximum compression possible, and for this, you can try: lame -v input.wav output.mp3 lame -v -V n input.wav output.mp3 (to vary quality/filesize) ======================================================================= VBR quality setting ======================================================================= -V n n = 0...9. Specifies the value of VBR_q. default = 4, highest quality = 0, smallest files = 9 Using -V 5 or higher (lower quality) is NOT RECOMMENDED. ABR will produce better results. How is VBR_q used? The value of VBR_q influences two basic parameters of LAME's psycho acoustics: a) the absolute threshold of hearing b) the sample to noise ratio The lower the VBR_q value the lower the injected quantization noise will be. *NOTE* No psy-model is perfect, so there can often be distortion which is audible even though the psy-model claims it is not! Thus using a small minimum bitrate can result in some aggressive compression and audible distortion even with -V 0. Thus using -V 0 does not sound better than a fixed 256 kbps encoding. For example: suppose in the 1 kHz frequency band the psy-model claims 20 dB of distortion will not be detectable by the human ear, so LAME VBR-0 will compress that frequency band as much as possible and introduce at most 20 dB of distortion. Using a fixed 256 kbps framesize, LAME could end up introducing only 2 dB of distortion. If the psy-model was correct, they will both sound the same. If the psy-model was wrong, the VBR-0 result can sound worse. ======================================================================= voice encoding mode ======================================================================= --voice An experimental voice encoding mode. Tuned for 44.1 kHz input files. --voice is deprecated, use --preset voice instead ======================================================================= swapbytes ======================================================================= -x swap bytes in the input file (and output file when using --decode). For sorting out little endian/big endian type problems. If your encodings sound like static, try this first. ======================================================================= OS/2 process priority control ======================================================================= --priority <type> (OS/2 only) Sets the process priority for LAME while running under IBM OS/2. This can be very useful to avoid the system becoming slow and/or unresponsive. By setting LAME to run in a lower priority, you leave more time for the system to update basic processing (drawing windows, polling keyboard/mouse, etc). The impact in LAME's performance is minimal if you use priority 0 to 2. The valid parameters are: 0 = Low priority (IDLE, delta = 0) 1 = Medium priority (IDLE, delta = +31) 2 = Regular priority (REGULAR, delta = -31) 3 = High priority (REGULAR, delta = 0) 4 = Maximum priority (REGULAR, delta = +31) Note that if you call '--priority' without a parameter, then priority 0 will be assumed.