Amiga Plus 2000 #4

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% lame [options] inputfile [outputfile] ======================================================================= Examples: ======================================================================= fixed bit rate jstereo 128kbs encoding: % lame sample.wav sample.mp3 fixed bit rate jstereo 128kbs encoding, highest quality: (recommended) % lame -h sample.wav sample.mp3 To disable joint stereo encoding (slightly faster, but less quality at bitrates<=128kbs) % lame -m s sample.wav sample.mp3 Fast encode, low quality (no psycho-acoustics) % lame -f sample.wav sample.mp3 Variable Bitrate (VBR): (use -V n to adjust quality/filesize) % lame -h -v sample.wav sample.mp3 Note: VBR is currently under heavy development. Right now it can often result in too much compression. I would recommend using VBR with a minimum bitrate of 112kbs. This will let LAME increase the bitrate for difficult-to-encode frames, but prevent LAME from being too aggressive for simple frames: % lame -h -v -b 112 sample.wav sample.mp3 ======================================================================= LOW BITRATES ======================================================================= At lower bitrates, (like 24kbs per channel), it is recommended that you use a 16kHz 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 (--lowpass and --lowpass-width options). ======================================================================= STREAMING EXAMPLES ======================================================================= Streaming mono 22.05kHz raw pcm, 24kbs output: % cat inputfile | lame -r -m m -b 24 -s 22.05 - - > output Streaming mono 22.05kHz raw pcm, with downsampling to 16kHz: % cat inputfile | sox -t raw -x -w -s -c 1 -r 22050 - -t raw -x -w -s -c 1 -r 16000 - resample 0.66 | lame -r -m m -b 24 -s 16 - - > output You may or may not need one or both of the "-x" (swap bytes) option in Sox. ======================================================================= For more options, just type: % lame --help Scripts are included to run lame on multiple files: bach script: mlame Run "mlame -h" for instructions. sh script: auenc Run auenc for instructions ======================================================================= options guide: ======================================================================= These options are explained in detail below. Quality related: -m m/s/j/f 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 experimental voice encoding mode --noshort disable short blocks Constant Bit Rate (CBR) -b n set bitrate (8,16,24,...,320) -h higher quality but slower -f disable psycho-acoustics. Encoding much faster but lower quality --freeformat produce a free format bitstream. User must also specify a bitrate with -b, between 8 and 320 kbs. Variable Bit Rate (VBR) -v VBR -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 Xing VBR informational tag --nohist disable display of VBR bitrate histogram 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 --decode assume input file is an mp3 file, and decode to 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 -x swap bytes of input file -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 -g run MP3x, the graphical frame analyzer --strictly-enforce-ISO comply as much as possible to ISO MPEG spec id3 tagging: --tt "title" title of song (max 30 chars) --ta "artist" artist who did the song (max 30 chars) --tl "album" album where it came from (max 30 chars) --ty "year" year in which the song/album was made (max 4 chars) --tc "comment" additional info (max 30 chars) --tg "genre" genre of song (name or number) options not yet described: --nores disable bit reservoir --noath disable ATH --cwlimit <freq> specify range of tonality calculation --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. ======================================================================= ATH only ======================================================================= --athonly This option causes LAME to ignore the output of the psy-model and only use masking from the ATH. Might be useful at very high bitrates or for testing the ATH. ======================================================================= bitrate ======================================================================= -b n For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz) n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320 For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz) n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160 The bitrate to be used. Default is 128kbs MPEG1, 80kbs 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 For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz) n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320 For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz) n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160 Maximum allowed bitrate when using VBR. ======================================================================= copyright ======================================================================= -c mark the encoded file as copyrighted ======================================================================= 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 raw pcm file. The input file must be an MP3 file, and the raw pcm data will be written to the output file in native endian format. Under linux, on i386, to convert the output file to a wav, use: % lame --decode input.mp3 output.pcm % sox -c 2 -t raw -r 44100 -s -w output.pcm output.wav (assuming output.pcm came from a stereo, 44.1khz mp3 file) ======================================================================= 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 disable psycho-acoustics. Encoding much faster but lower quality ======================================================================= 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 kbs, which can be any integer between 8 and 320. Not supported by many decoders. ======================================================================= graphical frame analyzer ======================================================================= -g run MP3x, the graphical frame analyzer analysis on the inputfile. The inputfile can be either an .mp3 file or uncompressed audio file. MP3x support must be compiled into LAME, and requires GTK 1.2. Documentation is under the About pull down menu. ======================================================================= high quality ======================================================================= -h use (maybe) some quality improvements LAME 3.21 and up: -h enables specialized mid/side masking thresholds to be used in jstereo mode. Will sound better in jstereo mode but is 20% slower. No effect for mono files. LAME 3.58beta and up: -h also enables a more accurate but slightly slower quantization formula. ======================================================================= sfb=21 cutoff ======================================================================= -k keep all frequencies. (Disable all filters) Without -k, 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. ======================================================================= Modes: ======================================================================= -m m mono. -m s stereo -m j jstereo -m f forced mid/side stereo 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. jstereo is the default mode for stereo files with fixed bitrates of 128kbs 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 128kbs, it is clearly worth while. At higher bitrates it is less usefull. 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. -m f 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. ======================================================================= MP3 input file ======================================================================= --mp3input Assume the input file is a MP3 file. Usefull for downsampling from one mp3 to another. 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 short blocks ======================================================================= --noshort Encode all frames using long blocks. ======================================================================= 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. ======================================================================= 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. ======================================================================= output sampling frequency in kHZ ======================================================================= --resample n where n = 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 44.1khz input file at 16kbs). ======================================================================= 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 ======================================================================= 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 Xing VBR tag ======================================================================= -t Disable writing of the Xing 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. ======================================================================= 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 128kbs 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 wan 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 very quality/filesize) ======================================================================= VBR quality setting ======================================================================= -V n n=0..9. Specifies the value of VBR_q. default=4. 0=highest quality. How is VBR_q used? OVER = number of scalefactor bands with distortion that exceeds the allowed distortion given by the masking thresholds. OVER is computed by outer_loop, and the masking thresholds are computed by the psycho-acoustic model. VBR_q = the minimum value of OVER which is to be allowed. LAME will choose the smallest bitrate for which OVER <= VBR_q. (a minimum allowed bitrate can be set with -b. default=64kbs) If the frame contains short blocks, then the minimum bitrate is made much larger since the OVER does not adequately measure distortion caused by pre-echo. LAME uses bitrates of at least 160kbs for short blocks to make sure they sound good. *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 256kbs encoding. For example: suppose in the 1kHz frequency band the psy-model claims 20db 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 20db of distortion. Using a fixed 256kbit framesize, LAME could end up introducing only 2db 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 44.1kHz input files. ======================================================================= swapbytes ======================================================================= -x swap bytes in the input file. for sorting out little endian/big endian type problems. If your encodings sound like static, try this first.