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Master Member
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LAME MP3 Encoder
LAME 3.98 beta 7 April 6 2008
Robert Hegemann:
下載/更新履歷/原始碼 |
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Master Member
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說明列表
代碼:
LAME 32bits version 3.98 (beta 7, Apr 6 2008) (http://www.mp3dev.org/) usage: lame.exe [options] <infile> [outfile] <infile> and/or <outfile> can be "-", which means stdin/stdout. RECOMMENDED: lame -V2 input.wav output.mp3 OPTIONS: Input options: --scale <arg> scale input (multiply PCM data) by <arg> --scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg> --scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg> --mp1input input file is a MPEG Layer I file --mp2input input file is a MPEG Layer II file --mp3input input file is a MPEG Layer III file --nogap <file1> <file2> <...> gapless encoding for a set of contiguous files --nogapout <dir> output dir for gapless encoding (must precede --nogap) --nogaptags allow the use of VBR tags in gapless encoding Input options for RAW PCM: -r input is raw pcm -x force byte-swapping of input -s sfreq sampling frequency of input file (kHz) - default 44.1 kHz --bitwidth w input bit width is w (default 16) --signed input is signed (default) --unsigned input is unsigned --little-endian input is little-endian (default) --big-endian input is big-endian Operational options: -a downmix from stereo to mono file for mono encoding -m <mode> (j)oint, (s)imple, (f)orce, (d)dual-mono, (m)ono default is (j) or (s) depending on bitrate joint = joins the best possible of MS and LR stereo simple = force LR stereo on all frames force = force MS stereo on all frames. --preset type type must be "medium", "standard", "extreme", "insane", or a value for an average desired bitrate and depending on the value specified, appropriate quality settings will be used. "--preset help" gives more info on these --comp <arg> choose bitrate to achive a compression ratio of <arg> --replaygain-fast compute RG fast but slightly inaccurately (default) --replaygain-accurate compute RG more accurately and find the peak sample --noreplaygain disable ReplayGain analysis --clipdetect enable --replaygain-accurate and print a message whether clipping occurs and how far the waveform is from full scale --flush flush output stream as soon as possible --freeformat produce a free format bitstream --decode input=mp3 file, output=wav -t disable writing wav header when using --decode Verbosity: --disptime <arg>print progress report every arg seconds -S don't print progress report, VBR histograms --nohist disable VBR histogram display --silent don't print anything on screen --quiet don't print anything on screen --brief print more useful information --verbose print a lot of useful information Noise shaping & psycho acoustic algorithms: -q <arg> <arg> = 0...9. Default -q 5 -q 0: Highest quality, very slow -q 9: Poor quality, but fast -h Same as -q 2. Recommended. -f Same as -q 7. Fast, ok quality CBR (constant bitrate, the default) options: -b <bitrate> set the bitrate in kbps, default 128 kbps --cbr enforce use of constant bitrate ABR options: --abr <bitrate> specify average bitrate desired (instead of quality) VBR options: -V n quality setting for VBR. default n=4 0=high quality,bigger files. 9=smaller files -v the same as -V 4 --vbr-old use old variable bitrate (VBR) routine --vbr-new use new variable bitrate (VBR) routine (default) -b <bitrate> specify minimum allowed bitrate, default 32 kbps -B <bitrate> specify maximum allowed bitrate, default 320 kbps -F strictly enforce the -b option, for use with players that do not support low bitrate mp3 -t disable writing LAME Tag -T enable and force writing LAME Tag PSY related: --temporal-masking x x=0 disables, x=1 enables temporal masking effect --nssafejoint M/S switching criterion --nsmsfix <arg> M/S switching tuning [effective 0-3.5] --interch x adjust inter-channel masking ratio --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 experimental switches: -Y lets LAME ignore noise in sfb21, like in CBR MP3 header/stream options: -e <emp> de-emphasis n/5/c (obsolete) -c mark as copyright -o mark as non-original -p error protection. adds 16 bit checksum to every frame (the checksum is computed correctly) --nores disable the bit reservoir --strictly-enforce-ISO comply as much as possible to ISO MPEG spec Filter options: --lowpass <freq> frequency(kHz), lowpass filter cutoff above freq --lowpass-width <freq> frequency(kHz) - default 15% of lowpass freq --highpass <freq> frequency(kHz), highpass filter cutoff below freq --highpass-width <freq> frequency(kHz) - default 15% of highpass freq --resample <sfreq> sampling frequency of output file(kHz)- default=automatic 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[/total]> audio/song track number and (optionally) the total number of tracks on the original recording. (track and total each 1 to 255. just the track number creates v1.1 tag, providing a total forces v2.0). --tg <genre> audio/song genre (name or number in list) --ti <file> audio/song albumArt (jpeg/png/gif file, 128KB max, v2.3) --tv <id=value> user-defined frame specified by id and value (v2.3 tag) --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 --ignore-tag-errors ignore errors in values passed for tags Note: A version 2 tag will NOT be added unless one of the input fields won't fit in a version 1 tag (e.g. the title string is longer than 30 characters), or the '--add-id3v2' or '--id3v2-only' options are used, or output is redirected to stdout. MS-Windows-specific options: --priority <type> sets the process priority: 0,1 = Low priority (IDLE_PRIORITY_CLASS) 2 = normal priority (NORMAL_PRIORITY_CLASS, default) 3,4 = High priority (HIGH_PRIORITY_CLASS)) Note: Calling '--priority' without a parameter will select priority 0. Misc: --license print License information Platform specific: --noasm <instructions> disable assembly optimizations for mmx/3dnow/sse MPEG-1 layer III sample frequencies (kHz): 32 48 44.1 bitrates (kbps): 32 40 48 56 64 80 96 112 128 160 192 224 256 320 MPEG-2 layer III sample frequencies (kHz): 16 24 22.05 bitrates (kbps): 8 16 24 32 40 48 56 64 80 96 112 128 144 160 MPEG-2.5 layer III sample frequencies (kHz): 8 12 11.025 bitrates (kbps): 8 16 24 32 40 48 56 64 此文章於 2008-04-13 09:07 PM 被 A.C 編輯. |
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Master Member
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文章: 1,733
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preset help
代碼:
LAME 32bits version 3.98 (beta 7, Apr 6 2008) (http://www.mp3dev.org/) 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 presets: For VBR modes (generally highest quality): "--preset medium" This preset should provide near transparency to most people on most music. "--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 --preset switches): "--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. For ABR modes (high quality per given bitrate but not as high as VBR): "--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. The following options are also available for the corresponding profiles: <fast> standard <fast> extreme insane <cbr> (ABR Mode) - The ABR Mode is implied. To use it, simply specify a bitrate. For example: "--preset 185" activates this preset and uses 185 as an average kbps. "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 significant 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 useful in situations such as when streaming an mp3 over the internet may be important. For example: "--preset fast standard <input file> <output file>" or "--preset cbr 192 <input file> <output file>" or "--preset 172 <input file> <output file>" or "--preset extreme <input file> <output file>" A few aliases are available for ABR mode: phone => 16kbps/mono phon+/lw/mw-eu/sw => 24kbps/mono mw-us => 40kbps/mono voice => 56kbps/mono fm/radio/tape => 112kbps hifi => 160kbps cd => 192kbps studio => 256kbps |
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Master Member
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文章: 1,733
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LAME 3.98 beta 8
很快的,LAME 3.98 beta 8 出來了。
LAME 3.98 beta 8 April 13 2008 Robert Hegemann:
比較重要的是第一點更新,VBR 增加小數點設定,可更自由的選擇位元率。 |
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Master Member
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文章: 1,733
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用 VBR 模式轉檔的使用者,可以試試加上 Z2。VBR 的小數點範圍為 0~9.999999。
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*停權中*
加入日期: Mar 2008 您的住址: 潛水中
文章: 157
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我頭一次看到
能這樣拉網頁 東西真的學不完 感謝分享編碼器 ![]() . 此文章於 2008-07-05 12:48 PM 被 蝦米碗糕 編輯. |
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Amateur Member
![]() 加入日期: Nov 2001 您的住址: 台中、高雄
文章: 38
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http://www.free-codecs.com/download...Show_Filter.htm
幫貼 3.98正式版 DirectShow 用的 encoder... |
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Master Member
![]() ![]() ![]() ![]() 加入日期: Aug 2002 您的住址: Taipei&Taichung
文章: 1,637
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我也用beta好久了…
一直都是用EAC+Lame壓mp3,非常棒的encoder。 |
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