Short: Lame Ain't an MP3 Encoder Author: Mark Taylor Uploader: Diego Casorran Type: mus/conv Version: 3.93.1 Requires: 68060 - sndfile.library Architecture: m68k-amigaos NAME lame - create mp3 audio files SYNOPSIS lame [options] DESCRIPTION LAME is a program which can be used to create compressed audio files. (Lame ain't an MP3 encoder). These audio files can be played back by popular MP3 players such as mpg123 or madplay. To read from stdin, use "-" for . To write to stdout, use a "-" for . OPTIONS Input options: -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 sev- eral fseek()'s on the input file looking for WAV and AIFF headers. Might not be available on your release. -x Swap bytes in the input file or output file when using --decode. For sorting out little endian/big endian type prob- lems. If your encodings sounds like static, try this first. -s sfreq sfreq = 8/11.025/12/16/22.05/24/32/44.1/48 Required only for raw PCM input files. Otherwise it will be determined from the header of the input file. LAME will automatically resample the input file to one of the supported MP3 samplerates if necessary. --bitwidth n Input bit width. 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. --mp1input Assume the input file is a MPEG Layer I file. If the filename ends in ".mp1" or ".mpg" LAME will assume it is a MPEG Layer I file. For stdin or Layer I files which do not end in .mp1 or .mpg you need to use this switch. October 13, 2001 1 --mp2input Assume the input file is a MPEG Layer II (ie MP2) file. If the filename ends in ".mp2" LAME will assume it is a MPEG Layer II file. For stdin or Layer II files which do not end in .mp2 you need to use this switch. --mp3input Assume the input file is a MP3 file. 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 do not end in .mp3 you need to use this switch. --nogap file1 file2 ... gapless encoding for a set of contiguous files --nogapout dir output dir for gapless encoding (must precede --nogap) Operational options: -m mode mode = s, j, f, d, m Joint-stereo is the default mode for stereo files with VBR when -V is more than 4 or fixed bitrates of 160kbs or less. At higher fixed bitrates or higher VBR settings, the default is stereo. (s)tereo In this mode, the encoder makes no use of poten- tially existing correlations between the two input channels. It can, however, negotiate the bit demand between both channel, i.e. give one channel more bits if the other contains silence or needs less bits because of a lower complexity. (j)oint stereo In this mode, the encoder will make use of a corre- lation 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 sig- nal does not have too much stereo separation, thus giving a significant gain in encoding quality. Using mid/side stereo inappropriately can result in October 13, 2001 2 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 algo- rithm than that described in the ISO documentation, and thus is safe to use in joint stereo mode. (f)orced joint stereo This mode will force MS joint 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 sep- aration. (d)ual channels In this mode, the 2 channels will be totally inden- pendently 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 encod- ing. (mo)no 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. -a Mix the stereo input file to mono and encode as mono. The downmix is calculated as the sum of the left and right channel, attenuated by 6 dB. This option is only needed in the case of raw PCM stereo input (because LAME cannot determine the number of channels in the input file). To encode a stereo PCM input file as mono, use lame -m s -a. For WAV and AIFF input files, using -m -I m will always produce a mono .mp3 file from both mono and stereo input. -d Allows the left and right channels to use different block size types. --freeformat Produces a free format bitstream. With this option, you can use -b with any bitrate higher than 8 kbps. However, even if an mp3 decoder is required to sup- port free bitrates at least up to 320 kbps, many October 13, 2001 3 players are unable to deal with it. Tests have shown that the following decoders sup- port free format: FreeAmp up to 440 kbps in_mpg123 up to 560 kbps l3dec up to 310 kbps LAME up to 560 kbps MAD up to 640 kbps --decode Uses LAME for decoding to a wav file. The input file can be any input type supported by encoding, including layer I,II,III (MP3) and OGG files. In case of MPEG files, LAME uses a bugfixed version of mpglib for decoding. If -t is used (disable wav header), LAME will out- put raw pcm in native endian format. You can use -x to swap bytes order. -t Disable writing of the INFO Tag on encoding. This tag in embedded in frame 0 of the MP3 file. It includes some information about the encoding options of the file, and in VBR it lets VBR aware players correctly seek and compute playing times of VBR files. When --decode is specified (decode to WAV), this flag will disable writing of the WAV header. The output will be raw pcm, native endian format. Use -x to swap bytes. --comp arg Instead of choosing bitrate, using this option, user can choose compression ratio to achieve. --scale n --scale-l n --scale-r n Scales input (every channel, only left channel or only right channel) by n. This just multiplies the PCM data (after it has been converted to floating point) by n. n > 1: increase volume n = 1: no effect n < 1: reduce volume Use with care, since most MP3 decoders will trun- cate data which decodes to values greater than 32768. October 13, 2001 4 --preset [fast] type | [cbr] kbps Use one of the built-in presets. Have a look at the PRESETS section below. Warning: with the current version fast presets might result in too high bitrate compared to regu- lar presets. --preset help gives more infos about the the used options in these presets. --alt-preset [fast] type | [cbr] kbps Use one of the built-in presets. This option is deprecated and offers the same as the --preset option above. Do not use it anymore, it will go away in a later version. --r3mix Uses r3mix VBR preset. See http://www.r3mix.net/ for more details. --noasm type Disable specific assembly optimizations ( mmx / 3dnow / sse ). Quality will not increase, only speed will be reduced. If you have problems run- ning Lame on a Cyrix/Via processor, disabling mmx optimizations might solve your problem. Verbosity: --disptime n Set the delay in seconds between two display updates. --nohist By default, LAME will display a bitrate histogram while producing VBR mp3 files. This will disable that feature. Histogram display might not be available on your release. -S --silent --quiet Do not print anything on the screen. --verbose Print a lot of information on the screen. --help Display a list of available options. October 13, 2001 5 Noise shaping & psycho acoustic algorithms: -q qual 0 <= qual <= 9 Bitrate is of course the main influence on quality. The higher the bitrate, the higher the quality. But for a given bitrate, we have a choice of algo- rithms to determine the best scalefactors and huff- man encoding (noise shaping). -q 0: use slowest & best possible version of all algo- rithms. -q 0 and -q 1 are slow and may not produce significantly higher quality. -q 2: recommended. Same as -h. -q 5: default value. Good speed, reasonable quality. -q 7: same as -f. Very fast, ok quality. Psycho acous- tics are used for pre-echo & M/S, but no noise shaping is done. -q 9: disables almost all algorithms including psy-model. Poor quality. -h Use some quality improvements. Encoding will be slower, but the result will be of higher quality. The behaviour is the same as the -q 2 switch. This switch is always enabled when using VBR. -f This switch forces the encoder to use a faster encoding mode, but with a lower quality. The behaviour is the same as the -q 7 switch. Noise shaping will be disabled, but psycho acous- tics will still be computed for bit allocation and pre-echo detection. CBR (constant bitrate, the default) options: -b n For MPEG1 (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 MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz) October 13, 2001 6 n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 Default is 128 for MPEG1 and 64 for MPEG2. --cbr enforce use of constant bitrate ABR (average bitrate) options: --abr n Turns on encoding with a targeted average bitrate of n kbits, allowing to use frames of different sizes. The allowed range of n is 8 - 310, you can use any integer value within that range. It can be combined with the -b and -B switches like: lame --abr 123 -b 64 -B 192 a.wav a.mp3 which would limit the allowed frame sizes between 64 and 192 kbits. The use of -B is NOT RECOMMENDED. A 128 kbps CBR bitstream, because of the bit reservoir, can actu- ally have frames which use as many bits as a 320 kbps frame. VBR modes minimize the use of the bit reservoir, and thus need to allow 320 kbps frames to get the same flexibility as CBR streams. VBR (variable bitrate) options: -v use variable bitrate (--vbr-old) --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 con- sidered the "workhorse" VBR algorithm. --vbr-new Invokes the newest VBR algorithm. During the development of version 3.90, considerable tuning was done on this algorithm, and it is now consid- ered to be on par with the original --vbr-old. It has the added advantage of being very fast (over twice as fast as --vbr-old). -V n 0 <= n <= 9 Enable VBR (Variable BitRate) and specifies the value of VBR quality (default = 4). 0 = highest quality. ABR and VBR options: October 13, 2001 7 -b bitrate For MPEG1 (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 MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz) n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 Specifies the minimum bitrate to be used. However, in order to avoid wasted space, the smallest frame size available will be used during silences. -B bitrate For MPEG1 (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 MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz) n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 Specifies the maximum allowed bitrate. Note: If you own an mp3 hardware player build upon a MAS 3503 chip, you must set maximum bitrate to no more than 224 kpbs. -F Strictly enforce the -b option. This is mainly for use with hardware players that do not support low bitrate mp3. Without this option, the minimum bitrate will be ignored for passages of analog silence, i.e. when the music level is below the absolute threshold of human hearing (ATH). ATH related: --noath Disable any use of the ATH (absolute threshold of hearing) for masking. Normally, humans are unable to hear any sound below this threshold. --athshort Ignore psychoacoustic model for short blocks, use ATH only. October 13, 2001 8 --athonly This option causes LAME to ignore the output of the psy-model and only use masking from the ATH (abso- lute threshold of hearing). Might be useful at very high bitrates or for testing the ATH. --athtype shape The Absolute Threshold of Hearing is the minimum threshold under which humans are unable to hear any sound. In the past, LAME was using ATH shape 0 which is the Painter & Spanias formula. Tests have shown that this formula is innacurate for the 13 - 22 kHz area, leading to audible artifacts in some cases. Shape 1 was thus implemented, which is over sensi- tive, leading to very high bitrates. Shape 2 formula was accurately modelized from real data in order to reach optimal quality while not wasting bitrate. In CBR and ABR modes, LAME uses ATH shape 2 by default, VBR selects one depending on the specified parameter to the -V option. --athlower n Lower the ATH (absolute threshold of hearing) by n dB. Normally, humans are unable to hear any sound below this threshold, but for music recorded at very low level this option might be usefull. --athaa-type n ATH auto adjust types 1 - 3, else no adjustment --athaa-sensitivity x activation offset in -/+ dB for ATH auto-adjustment PSY related: --short Let LAME use short blocks when appropriate. It is the default setting. --noshort Encode all frames using long blocks only. This could increase quality when encoding at very low bitrates, but can produce serious pre-echo arte- facts. --allshort Use only short blocks, no long ones. --cwlimit freq Compute tonality up to freq (in kHz). Default set- ting is 8.8717. October 13, 2001 9 --notemp Do not make use of the temporal masking effect. --nspsytune Experimental PSY tunings by Naoki Shibata --nssafejoint M/S switching criterion --nsmsfix arg M/S switching tuning [effective 0-3.5] --ns-bass x Adjust masking for sfbs 0 - 6 (long) 0 - 5 (short) --ns-alto x Adjust masking for sfbs 7 - 13 (long) 6 - 10 (short) --ns-treble x Adjust masking for sfbs 14 - 21 (long) 11 - 12 (short) --ns-sfb21 x Change ns-treble by x dB for sfb21 Experimantal options: -X n 0 <= n <= 7 When LAME searches for a "good" quantization, it has to compare the actual one with the best one found so far. The comparison says which one is better, the best so far or the actual. The -X parameter selects between different approaches to make this decision, -X0 beeing the default mode: -X0 The criterions are (in order of importance): * less distorted scalefactor bands * the sum of noise over the thresholds is lower * the total noise is lower -X1 The actual is better if the maximum noise over all scalefactor bands is less than the best so far. -X2 The actual is better if the total sum of noise is lower than the best so far. -X3 October 13, 2001 10 The actual is better if the total sum of noise is lower than the best so far and the maximum noise over all scalefactor bands is less than the best so far plus 2dB. -X4 Not yet documented. -X5 The criterions are (in order of importance): * the sum of noise over the thresholds is lower * the total sum of noise is lower -X6 The criterions are (in order of importance): * the sum of noise over the thresholds is lower * the maximum noise over all scalefactor bands is lower * the total sum of noise is lower -X7 The criterions are: * less distorted scalefactor bands or * the sum of noise over the thresholds is lower -Y lets LAME ignore noise in sfb21, like in CBR -Z toggles the scalefac feature on MP3 header/stream options: -e emp emp = n, 5, c n = (none, default) 5 = 0/15 microseconds c = citt j.17 All this does is set a flag in the bitstream. If you have a PCM input file where one of the above types of (obsolete) emphasis has been applied, you can set this flag in LAME. Then the mp3 decoder should de-emphasize the output during playback, although most decoders ignore this flag. A better solution would be to apply the de-emphasis with a standalone utility before encoding, and then encode without -e. -c Mark the encoded file as being copyrighted. -o Mark the encoded file as being a copy. October 13, 2001 11 -p Turn on CRC error protection. It will add a cyclic redundancy check (CRC) code in each frame, allowing to detect transmission errors that could occur on the MP3 stream. However, it takes 16 bits per frame that would otherwise be used for encoding, and then will slightly reduce the sound quality. --nores Disable the bit reservoir. Each frame will then become independent from previous ones, but the quality will be lower. --strictly-enforce-ISO With this option, LAME will enforce the 7680 bit limitation on total frame size. This results in many wasted bits for high bitrate encodings but will ensure strict ISO compatibility. This compatibility might be important for hardware players. Filter options: -k Tells the encoder to use full bandwidth and to dis- able all filters. By default, the encoder uses some highpass filtering at low bitrates, in order to keep a good quality by giving more bits to more important frequencies. Increasing the bandwidth from the default setting might produce ringing artefacts at low bitrates. Use with care! --lowpass freq Set a lowpass filtering frequency in kHz. Frequen- cies above the specified one will be cutoff. --lowpass-width freq Set the width of the lowpass filter. The default value is 15% of the lowpass frequency. --highpass freq Set an highpass filtering frequency in kHz. Fre- quencies below the specified one will be cutoff. --highpass-width freq Set the width of the highpass filter in kHz. The default value is 15% of the highpass frequency. --resample sfreq sfreq = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48 Select ouptut sampling frequency (only supported for encoding). If not specified, LAME will automatically resample October 13, 2001 12 the input when using high compression ratios. ID3 tag options: --tt title audio/song title (max 30 chars for version 1 tag) --ta artist audio/song artist (max 30 chars for version 1 tag) --tl album audio/song album (max 30 chars for version 1 tag) --ty year audio/song year of issue (1 to 9999) --tc comment user-defined text (max 30 chars for v1 tag, 28 for v1.1) --tn track 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 Analysis options: -g run graphical analysis on . can also be a .mp3 file. (This feature is a compile time option. Your binary may for speed reasons be compiled without this.) October 13, 2001 13 ID3 TAGS LAME is able to embed ID3 v1, v1.1 or v2 tags inside the encoded MP3 file. This allows to have some usefull infor- mation about the music track included inside the file. Those data can be read by most MP3 players. Lame will smartly choose wich 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 wich are unable to read ID3 v2 tags. ENCODING MODES LAME is able to encode your music using one of its 3 encoding modes: constant bitrate (CBR), average bitrate (ABR) and variable bitrate (VBR). Constant Bitrate (CBR) This is the default encoding mode, and also the most basic. In this mode, the bitrate will be the same for the whole file. It means that each part of your mp3 file will be using the same number of bits. The musical passage beeing a difficult one to encode or an easy one, the encoder will use the same bitrate, so the quality of your mp3 is vari- able. 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 accu- rately predicted. Average Bitrate (ABR) In this mode, you choose the encoder will maintain an average bitrate while using higher bitrates for the parts of your music that need more bits. The result will be of higher quality than CBR encoding but the average file size will remain predictible, so this mode is highly recommended over CBR. This encoding mode is similar to what is reffered as vbr in AAC or Liquid Audio (2 other compression tech- nologies). 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 qual- ity level that you want to reach, but the inconve- nient is that the final file size is totally unpre- dictible. October 13, 2001 14 PRESETS The --preset switches are designed to provide the highest possible quality. They have for the most part been subject to and tuned via rigorous double blind listening tests to verify and achieve this objective. These are continually updated to coincide with the latest developments that occur and as a result should provide you with nearly the best quality currently possible from LAME. To activate these prests: For VBR modes (generally highest quality): --preset standard This preset should generally be transparent to most people on most music and is already quite high in quality. --preset extreme If you have extremely good hearing and similar equipment, this preset will generally provide slightly higher quality than the standard mode. For CBR 320kbps (highest quality possible from the --pre- set switches): --preset insane This preset will usually be overkill for most peo- ple and most situations, but if you must have the absolute highest quality with no regard to file- size, this is the way to go. For ABR modes (high quality per given bitrate but not as high as VBR): --preset kbps Using this preset will usually give you good qual- ity at a specified bitrate. Depending on the bitrate entered, this preset will determine the optimal settings for that particular situation. While this approach works, it is not nearly as flexible as VBR, and usually will not attain the same level of quality as VBR at higher bitrates. The following options are also available for the corre- sponding profiles: fast standard|extreme|insane cbr kbps October 13, 2001 15 fast Enables the new fast VBR for a particular profile. The disadvantage to the speed switch is that often times the bitrate will be slightly higher than with the normal mode and quality may be slightly lower also. cbr If you use the ABR mode (read above) with a signif- icant bitrate such as 80, 96, 112, 128, 160, 192, 224, 256, 320, you can use the cbr option to force CBR mode encoding instead of the standard ABR mode. ABR does provide higher quality but CBR may be use- ful in situations such as when streaming an MP3 over the internet may be important. EXAMPLES Fixed bit rate jstereo 128kbs encoding: lame sample.wav sample.mp3 Fixed bit rate jstereo 128 kbps encoding, highest quality (recommended): lame -h sample.wav sample.mp3 Fixed bit rate jstereo 112 kbps encoding: lame -b 112 sample.wav sample.mp3 To disable joint stereo encoding (slightly faster, but less quality at bitrates <= 128 kbps): lame -m s sample.wav sample.mp3 Fast encode, low quality (no psycho-acoustics): lame -f sample.wav sample.mp3 Variable bitrate (use -V n to adjust quality/filesize): lame -h -V 6 sample.wav sample.mp3 Streaming mono 22.05 kHz raw pcm, 24 kbps output: cat inputfile | lame -r -m m -b 24 -s 22.05 - - > output October 13, 2001 16 Streaming mono 44.1 kHz raw pcm, with downsampling to 22.05 kHz: cat inputfile | lame -r -m m -b 24 --resample 22.05 - - > output Encode with the fast standard preset: lame --preset fast standard sample.wav sample.mp3 BUGS Probably there are some. SEE ALSO mpg123 (1) , madplay (1) , sox (1) AUTHORS LAME originally developed by Mike Cheng and now maintained by Mark Taylor. GPSYCHO psycho-acoustic model by Mark Taylor. ( http://www.mp3dev.org/). mpglib by Michael Hipp Manual page by William Schelter, Nils Faerber, Alexander Leidinger ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°` `°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø Latest update of this package can be found at http://amiga.sourceforge.net/ ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°`ø°` `°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø`°ø ·············································A·r·c·h·i·v·e··C·o·n·t·e·n·t·s·· LhA Freeware Version 2.2 Copyright © 1991-94 by Stefan Boberg. Copyright © 1998-2000 by Jim Cooper and David Tritscher. Listing of archive 'lame-3.93.1.lha': Original Packed Ratio Date Time Name -------- ------- ----- --------- -------- ------------- 442351 134472 69.6% 01-Dec-02 16:29:26 +ChangeLog 25292 9301 63.2% 29-Nov-00 15:44:48 +COPYING 5406 1761 67.4% 24-Oct-01 14:32:24 +basic.html 3591 1801 49.8% 15-Oct-02 09:32:00 +contributors.html 1842 767 58.3% 24-Oct-01 14:32:24 +examples.html 74810 24058 67.8% 01-Dec-02 16:09:22 +history.html 7211 2398 66.7% 13-Oct-01 11:44:50 +id3.html 2200 906 58.8% 01-Dec-02 13:15:32 +index.html 708 275 61.1% 04-Dec-00 14:32:36 +lame.css 707 423 40.1% 19-Dec-00 13:07:10 +LICENSE 10425 9453 9.3% 17-Nov-02 10:04:36 +testcase.mp3 100044 94371 5.6% 29-Nov-00 15:44:48 +testcase.wav 5351 2685 49.8% 06-Apr-02 07:22:58 +TODO 25632 8353 67.4% 22-Jan-02 12:20:40 +USAGE 2303 1093 52.5% 13-Oct-01 11:44:50 +modes.html 6835 2674 60.8% 24-Oct-01 14:32:24 +node6.html 3183 1423 55.2% 17-Nov-02 09:06:56 +presets.html 43403 10307 76.2% 03-Sep-02 11:08:24 +switchs.html 445912 202335 54.6% 24-Mar-03 05:22:10 +LAME 25905 9663 62.6% 17-Nov-02 09:20:42 +lame.1 -------- ------- ----- --------- -------- 1233111 518519 57.9% Operation successful. _____________________________ .Readme created with: MRea \ ============================================================================== >»>»>»>»> Some additional info about this archive: Source: http://prdownloads.sf.net/amiga/lame-3.93.1.lha?download FileSize: 519552 Bytes CRC: 8906A867 MD5: 1826BE2FE2E1A4700BF3601241447CF5 SHA: A2AA7C5DED0890A939868D6B0B021B051B820346 ==============================================================================