| .TH lame 1 "July 08, 2008" "LAME 3.98" "LAME audio compressor" |
| .SH NAME |
| lame \- create mp3 audio files |
| .SH SYNOPSIS |
| lame [options] <infile> <outfile> |
| .SH DESCRIPTION |
| .PP |
| 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 a "\-" for <outfile>. |
| .SH OPTIONS |
| Input options: |
| .TP |
| .B \-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 |
| .B \-r, |
| LAME will perform several |
| .I fseek()'s |
| on the input file looking for WAV and AIFF headers. |
| .br |
| Might not be available on your release. |
| .TP |
| .B \-x |
| Swap bytes in the input file or output file when using |
| .B \-\-decode. |
| .br |
| For sorting out little endian/big endian type problems. |
| If your encodings sounds like static, |
| try this first. |
| .br |
| Without using |
| .B \-x, |
| LAME will treat input file as native endian. |
| .TP |
| .BI \-s " sfreq" |
| .I 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. |
| .TP |
| .BI \-\-bitwidth " n" |
| Input bit width per sample. |
| .br |
| .I 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. |
| .TP |
| .BI \-\-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. |
| .TP |
| .BI \-\-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. |
| .TP |
| .BI \-\-little-endian |
| Instructs LAME that the samples from the input are in little-endian form. |
| |
| Required only for raw PCM input files. |
| .TP |
| .BI \-\-big-endian |
| Instructs LAME that the samples from the input are in big-endian form. |
| |
| Required only for raw PCM input files. |
| .TP |
| .B \-\-mp2input |
| Assume the input file is a MPEG Layer II (ie MP2) file. |
| .br |
| 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. |
| .TP |
| .B \-\-mp3input |
| Assume the input file is a MP3 file. |
| .br |
| Useful for downsampling from one mp3 to another. |
| As an example, |
| it can be useful for streaming through an IceCast server. |
| .br |
| 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. |
| .TP |
| .BI \-\-nogap " file1 file2 ..." |
| gapless encoding for a set of contiguous files |
| .TP |
| .BI \-\-nogapout " dir" |
| output dir for gapless encoding (must precede \-\-nogap) |
| |
| .PP |
| Operational options: |
| .TP |
| .BI \-m " mode" |
| .I mode |
| = s, j, f, d, m |
| |
| Joint-stereo is the default mode for stereo files with VBR when |
| .B \-V |
| is more than 4 or fixed bitrates of 160kbs or less. |
| At higher fixed bitrates or higher VBR settings, |
| the default is stereo. |
| |
| .B (s)imple stereo |
| .br |
| 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. |
| |
| .B (j)oint stereo |
| .br |
| 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. |
| |
| .B (f)orced MS stereo |
| .br |
| 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. |
| |
| .B (d)ual mono |
| .br |
| 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. |
| |
| .B (m)ono |
| .br |
| 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. |
| .TP |
| .B \-a |
| Mix the stereo input file to mono and encode as mono. |
| .br |
| 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 |
| .B lame \-m |
| .I s |
| .B \-a. |
| |
| For WAV and AIFF input files, |
| using |
| .B \-m |
| will always produce a mono .mp3 file from both mono and stereo input. |
| .TP |
| .B \-d |
| Allows the left and right channels to use different block size types. |
| .TP |
| .B \-\-freeformat |
| Produces a free format bitstream. |
| With this option, |
| you can use |
| .B \-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: |
| .br |
| .B FreeAmp |
| up to 440 kbps |
| .br |
| .B in_mpg123 |
| up to 560 kbps |
| .br |
| .B l3dec |
| up to 310 kbps |
| .br |
| .B LAME |
| up to 560 kbps |
| .br |
| .B MAD |
| up to 640 kbps |
| .TP |
| .B \-\-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 |
| .B \-t |
| is used (disable wav header), |
| LAME will output raw pcm in native endian format. |
| You can use |
| .B \-x |
| to swap bytes order. |
| |
| This option is not usable if the MP3 decoder was |
| .B explicitly |
| disabled in the build of LAME. |
| .TP |
| .BI \-t |
| Disable writing of the INFO Tag on encoding. |
| .br |
| 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 |
| .B \-\-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 |
| .B \-x |
| to swap bytes. |
| .TP |
| .BI \-\-comp " arg" |
| Instead of choosing bitrate, |
| using this option, |
| user can choose compression ratio to achieve. |
| .TP |
| .BI \-\-scale " n" |
| .PD 0 |
| .TP |
| .BI \-\-scale\-l " n" |
| .TP |
| .BI \-\-scale\-r " n" |
| Scales input (every channel, only left channel or only right channel) by |
| .I n. |
| This just multiplies the PCM data (after it has been converted to floating |
| point) by |
| .I n. |
| |
| .I n |
| > 1: increase volume |
| .br |
| .I n |
| = 1: no effect |
| .br |
| .I n |
| < 1: reduce volume |
| |
| Use with care, |
| since most MP3 decoders will truncate data which decodes to values |
| greater than 32768. |
| .PD |
| .TP |
| .B \-\-replaygain\-fast |
| Compute ReplayGain fast but slightly inaccurately. |
| |
| This computes "Radio" ReplayGain on the input data stream after |
| user\(hyspecified volume\(hyscaling and/or resampling. |
| |
| The ReplayGain analysis does |
| .I 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 |
| .B 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: |
| .B \-\-replaygain\-accurate, \-\-noreplaygain |
| .TP |
| .B \-\-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 |
| .I 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 |
| .B http://www.replaygain.org/. |
| |
| |
| By default, LAME performs ReplayGain analysis on the input data |
| (after the user\(hyspecified 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 |
| .B \-\-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: |
| .RS 8 |
| .IP "*" 4 |
| 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 |
| .IP "*" 4 |
| decoding on the fly significantly slows down the encoding process |
| .RE |
| .RS 7 |
| |
| The apparent advantage is that: |
| .RE |
| .RS 8 |
| .IP "*" 4 |
| with |
| .B \-\-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. |
| .RE |
| .RS 7 |
| |
| 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 |
| .B 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: |
| .B \-\-replaygain-fast, \-\-noreplaygain \-\-clipdetect |
| .RE |
| .TP |
| .B \-\-noreplaygain |
| Disable ReplayGain analysis. |
| |
| By default ReplayGain analysis is enabled. This switch disables it. |
| |
| See also: |
| .B \-\-replaygain-fast, \-\-replaygain-accurate |
| .TP |
| .B \-\-clipdetect |
| Clipping detection. |
| |
| Enable |
| .B \-\-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 |
| .B explicitly |
| disabled in the build of LAME. |
| |
| See also: |
| .B \-\-replaygain-accurate |
| .TP |
| .B \-\-preset " [fast] type | [cbr] kbps" |
| Use one of the built-in presets. |
| |
| Have a look at the PRESETS section below. |
| |
| .B \-\-preset help |
| gives more infos about the the used options in these presets. |
| .TP |
| .B \-\-preset " [fast] type | [cbr] kbps" |
| Use one of the built-in presets. |
| .TP |
| .B \-\-noasm " type" |
| Disable specific assembly optimizations ( |
| .B mmx |
| / |
| .B 3dnow |
| / |
| .B 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. |
| |
| .PP |
| Verbosity: |
| .TP |
| .BI \-\-disptime " n" |
| Set the delay in seconds between two display updates. |
| .TP |
| .B \-\-nohist |
| By default, |
| LAME will display a bitrate histogram while producing VBR mp3 files. |
| This will disable that feature. |
| .br |
| Histogram display might not be available on your release. |
| .TP |
| .B -S |
| .PD 0 |
| .TP |
| .B \-\-silent |
| .TP |
| .B \-\-quiet |
| Do not print anything on the screen. |
| .PD |
| .TP |
| .B \-\-verbose |
| Print a lot of information on the screen. |
| .TP |
| .B \-\-help |
| Display a list of available options. |
| |
| .PP |
| Noise shaping & psycho acoustic algorithms: |
| .TP |
| .BI -q " qual" |
| 0 <= |
| .I 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). |
| |
| .B -q 0: |
| .br |
| use slowest & best possible version of all algorithms. |
| .B -q 0 |
| and |
| .B -q 1 |
| are slow and may not produce significantly higher quality. |
| |
| .B -q 2: |
| .br |
| recommended. |
| Same as |
| .B -h. |
| |
| .B -q 5: |
| .br |
| default value. |
| Good speed, |
| reasonable quality. |
| |
| .B -q 7: |
| .br |
| same as |
| .B -f. |
| Very fast, |
| ok quality. |
| Psycho acoustics are used for pre-echo & M/S, |
| but no noise shaping is done. |
| |
| .B -q 9: |
| .br |
| disables almost all algorithms including psy-model. |
| Poor quality. |
| .TP |
| .B -h |
| Use some quality improvements. |
| Encoding will be slower, |
| but the result will be of higher quality. |
| The behavior is the same as the |
| .B -q 2 |
| switch. |
| .br |
| This switch is always enabled when using VBR. |
| .TP |
| .B -f |
| This switch forces the encoder to use a faster encoding mode, |
| but with a lower quality. |
| The behavior is the same as the |
| .B -q 7 |
| switch. |
| |
| Noise shaping will be disabled, |
| but psycho acoustics will still be computed for bit allocation |
| and pre-echo detection. |
| |
| .PP |
| CBR (constant bitrate, the default) options: |
| .TP |
| .BI -b " n" |
| For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz) |
| .br |
| .I 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) |
| .br |
| .I 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) |
| .br |
| .I n |
| = 8, 16, 24, 32, 40, 48, 56, 64 |
| |
| Default is 128 for MPEG1 and 64 for MPEG2. |
| .TP |
| .BI \-\-cbr |
| enforce use of constant bitrate |
| |
| .PP |
| ABR (average bitrate) options: |
| .TP |
| .BI \-\-abr " n" |
| Turns on encoding with a targeted average bitrate of n kbits, |
| allowing to use frames of different sizes. |
| The allowed range of |
| .I n |
| is 8 - 310, |
| you can use any integer value within that range. |
| |
| It can be combined with the |
| .B -b |
| and |
| .B -B |
| switches like: |
| .B lame \-\-abr |
| .I 123 |
| .B -b |
| .I 64 |
| .B -B |
| .I 192 a.wav a.mp3 |
| which would limit the allowed frame sizes between 64 and 192 kbits. |
| |
| The use of |
| .B -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. |
| |
| .PP |
| VBR (variable bitrate) options: |
| .TP |
| .B -v |
| use variable bitrate |
| .B (\-\-vbr-new) |
| .TP |
| .B \-\-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. |
| .TP |
| .B \-\-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 |
| .B \-\-vbr-old. |
| It has the added advantage of being very fast (over twice as fast as |
| .B \-\-vbr-old). |
| .TP |
| .BI -V " n" |
| 0 <= |
| .I n |
| <= 9 |
| .br |
| Enable VBR (Variable BitRate) and specifies the value of VBR quality |
| (default = 4). |
| 0 = highest quality. |
| |
| .PP |
| ABR and VBR options: |
| .TP |
| .BI -b " bitrate" |
| For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz) |
| .br |
| .I 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) |
| .br |
| .I 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) |
| .br |
| .I 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. |
| .TP |
| .BI -B " bitrate" |
| For MPEG-1 (sampling frequencies of 32, 44.1 and 48 kHz) |
| .br |
| .I 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) |
| .br |
| .I 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) |
| .br |
| .I 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. |
| .TP |
| .B -F |
| Strictly enforce the |
| .B -b |
| option. |
| .br |
| 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). |
| |
| .PP |
| PSY related: |
| .TP |
| .B \-\-nssafejoint |
| M/S switching criterion |
| .TP |
| .BI \-\-nsmsfix " arg" |
| M/S switching tuning [effective 0-3.5] |
| .TP |
| .BI \-\-ns-bass " x" |
| Adjust masking for sfbs 0 - 6 (long) 0 - 5 (short) |
| .TP |
| .BI \-\-ns-alto " x" |
| Adjust masking for sfbs 7 - 13 (long) 6 - 10 (short) |
| .TP |
| .BI \-\-ns-treble " x" |
| Adjust masking for sfbs 14 - 21 (long) 11 - 12 (short) |
| .TP |
| .BI \-\-ns-sfb21 " x" |
| Change ns-treble by x dB for sfb21 |
| |
| .PP |
| Experimental options: |
| .TP |
| .BI -X " n" |
| 0 <= |
| .I 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 |
| .B -X |
| parameter selects between different approaches to make this decision, |
| .B -X0 |
| being the default mode: |
| |
| .B -X0 |
| .br |
| The criterions are (in order of importance): |
| .br |
| * less distorted scalefactor bands |
| .br |
| * the sum of noise over the thresholds is lower |
| .br |
| * the total noise is lower |
| |
| .B -X1 |
| .br |
| The actual is better if the maximum noise over all scalefactor bands is |
| less than the best so far. |
| |
| .B -X2 |
| .br |
| The actual is better if the total sum of noise is lower than the best so |
| far. |
| |
| .B -X3 |
| .br |
| 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. |
| |
| .B -X4 |
| .br |
| Not yet documented. |
| |
| .B -X5 |
| .br |
| The criterions are (in order of importance): |
| .br |
| * the sum of noise over the thresholds is lower |
| .br |
| * the total sum of noise is lower |
| |
| .B -X6 |
| .br |
| The criterions are (in order of importance): |
| .br |
| * the sum of noise over the thresholds is lower |
| .br |
| * the maximum noise over all scalefactor bands is lower |
| .br |
| * the total sum of noise is lower |
| |
| .B -X7 |
| .br |
| The criterions are: |
| .br |
| * less distorted scalefactor bands |
| .br |
| or |
| .br |
| * the sum of noise over the thresholds is lower |
| .TP |
| .B -Y |
| lets LAME ignore noise in sfb21, like in CBR |
| |
| .PP |
| MP3 header/stream options: |
| .TP |
| .BI -e " emp" |
| .I emp |
| = n, 5, c |
| |
| n = (none, default) |
| .br |
| 5 = 0/15 microseconds |
| .br |
| 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 |
| .B -e. |
| .TP |
| .B -c |
| Mark the encoded file as being copyrighted. |
| .TP |
| .B -o |
| Mark the encoded file as being a copy. |
| .TP |
| .B -p |
| Turn on CRC error protection. |
| .br |
| 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. |
| .TP |
| .B \-\-nores |
| Disable the bit reservoir. |
| Each frame will then become independent from previous ones, |
| but the quality will be lower. |
| .TP |
| .B \-\-strictly-enforce-ISO |
| With this option, |
| LAME will enforce the 7680 bit limitation on total frame size. |
| .br |
| This results in many wasted bits for high bitrate encodings but will |
| ensure strict ISO compatibility. |
| This compatibility might be important for hardware players. |
| |
| .PP |
| Filter options: |
| .TP |
| .BI \-\-lowpass " freq" |
| Set a lowpass filtering frequency in kHz. |
| Frequencies above the specified one will be cutoff. |
| .TP |
| .BI \-\-lowpass-width " freq" |
| Set the width of the lowpass filter. |
| The default value is 15% of the lowpass frequency. |
| .TP |
| .BI \-\-highpass " freq" |
| Set an highpass filtering frequency in kHz. |
| Frequencies below the specified one will be cutoff. |
| .TP |
| .BI \-\-highpass-width " freq" |
| Set the width of the highpass filter in kHz. |
| The default value is 15% of the highpass frequency. |
| .TP |
| .BI \-\-resample " sfreq" |
| .I sfreq |
| = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48 |
| .br |
| Select output sampling frequency (only supported for encoding). |
| .br |
| If not specified, |
| LAME will automatically resample the input when using high compression ratios. |
| |
| .PP |
| ID3 tag options: |
| .TP |
| .BI \-\-tt " title" |
| audio/song title (max 30 chars for version 1 tag) |
| .TP |
| .BI \-\-ta " artist" |
| audio/song artist (max 30 chars for version 1 tag) |
| .TP |
| .BI \-\-tl " album" |
| audio/song album (max 30 chars for version 1 tag) |
| .TP |
| .BI \-\-ty " year" |
| audio/song year of issue (1 to 9999) |
| .TP |
| .BI \-\-tc " comment" |
| user-defined text (max 30 chars for v1 tag, 28 for v1.1) |
| .TP |
| .BI \-\-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). |
| .TP |
| .BI \-\-tg " genre" |
| audio/song genre (name or number in list) |
| .TP |
| .B \-\-add-id3v2 |
| force addition of version 2 tag |
| .TP |
| .B \-\-id3v1-only |
| add only a version 1 tag |
| .TP |
| .B \-\-id3v2-only |
| add only a version 2 tag |
| .TP |
| .B \-\-space-id3v1 |
| pad version 1 tag with spaces instead of nulls |
| .TP |
| .B \-\-pad-id3v2 |
| same as \-\-pad-id3v2-size 128 |
| .TP |
| .B \-\-pad-id3v2-size "num" |
| adds version 2 tag, pad with extra "num" bytes |
| .TP |
| .B \-\-genre-list |
| print alphabetically sorted ID3 genre list and exit |
| .TP |
| .B \-\-ignore-tag-errors |
| ignore errors in values passed for tags, use defaults in case an error occurs |
| |
| .PP |
| Analysis options: |
| .TP |
| .B \-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.) |
| |
| .SH 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. |
| |
| .SH 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). |
| .TP |
| .B 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. |
| .TP |
| .B 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). |
| .TP |
| .B 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. |
| |
| .SH PRESETS |
| The |
| .B \-\-preset |
| switches are aliases over LAME settings. |
| |
| To activate these presets: |
| .PP |
| For VBR modes (generally highest quality): |
| .TP |
| .B \-\-preset medium |
| This preset should provide near transparency to most people on most music. |
| .TP |
| .B \-\-preset standard |
| This preset should generally be transparent to most people on most music and |
| is already quite high in quality. |
| .TP |
| .B \-\-preset extreme |
| If you have extremely good hearing and similar equipment, |
| this preset will generally provide slightly higher quality than the |
| .B standard |
| mode. |
| .PP |
| For CBR 320kbps (highest quality possible from the |
| .B \-\-preset |
| switches): |
| .TP |
| .B \-\-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. |
| .PP |
| For ABR modes (high quality per given bitrate but not as high as VBR): |
| .TP |
| .B \-\-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. |
| .PP |
| The following options are also available for the corresponding profiles: |
| .PP |
| .B fast standard|extreme |
| .br |
| .B cbr " kbps" |
| .PP |
| .TP |
| .B fast |
| Enables the new fast VBR for a particular profile. |
| .TP |
| .B 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 |
| .B 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. |
| |
| |
| .SH EXAMPLES |
| .LP |
| Fixed bit rate jstereo 128kbs encoding: |
| .IP |
| .B lame |
| .I sample.wav sample.mp3 |
| |
| .LP |
| Fixed bit rate jstereo 128 kbps encoding, highest quality (recommended): |
| .IP |
| .B lame \-h |
| .I sample.wav sample.mp3 |
| |
| .LP |
| Fixed bit rate jstereo 112 kbps encoding: |
| .IP |
| .B lame \-b |
| .I 112 sample.wav sample.mp3 |
| |
| .LP |
| To disable joint stereo encoding (slightly faster, |
| but less quality at bitrates <= 128 kbps): |
| .IP |
| .B lame \-m |
| .I s sample.wav sample.mp3 |
| |
| .LP |
| Fast encode, |
| low quality (no psycho-acoustics): |
| .IP |
| .B lame \-f |
| .I sample.wav sample.mp3 |
| |
| .LP |
| Variable bitrate (use \-V n to adjust quality/filesize): |
| .IP |
| .B lame \-h \-V |
| .I 6 sample.wav sample.mp3 |
| |
| .LP |
| Streaming mono 22.05 kHz raw pcm, 24 kbps output: |
| .IP |
| .B cat |
| .I inputfile |
| .B | lame \-r \-m |
| .I m |
| .B \-b |
| .I 24 |
| .B \-s |
| .I 22.05 \- \- |
| .B > |
| .I output |
| |
| .LP |
| Streaming mono 44.1 kHz raw pcm, |
| with downsampling to 22.05 kHz: |
| .IP |
| .B cat |
| .I inputfile |
| .B | lame \-r \-m |
| .I m |
| .B \-b |
| .I 24 |
| .B \-\-resample |
| .I 22.05 \- \- |
| .B > |
| .I output |
| |
| .LP |
| Encode with the |
| .B fast standard |
| preset: |
| .IP |
| .B lame \-\-preset fast standard |
| .I sample.wav sample.mp3 |
| |
| .SH BUGS |
| .PP |
| Probably there are some. |
| .SH SEE ALSO |
| .BR mpg123 (1) , |
| .BR madplay (1) , |
| .BR sox (1) |
| .SH AUTHORS |
| .nf |
| 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 Rog\['e]rio Brito. |
| .\" Local Variables: |
| .\" mode: nroff |
| .\" End: |