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diff --git a/lib/liblame/doc/man/lame.1 b/lib/liblame/doc/man/lame.1 new file mode 100644 index 0000000000..27f864c724 --- /dev/null +++ b/lib/liblame/doc/man/lame.1 @@ -0,0 +1,1145 @@ +.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: |