我現在用lame作為編碼工具,但是速度不夠.雖然我用了-q 9但是還是不行.請問有沒有好的辦法?或者是哪位大俠告訴我lame 的一些算法方面的事情,最好是MDCT和HUffman方面的.
萬分謝謝!
請個忙,好嗎?有關lame的問題,謝謝!
版主: DearHoney
這一些給你參考，不過我找不到有什麼方法可以減少CPU耗損....
LAME version 3.96 MMX (http://www.mp3dev.org/)
usage: lame [options] <infile> [outfile]
<infile> and/or <outfile> can be "-", which means stdin/stdout.
RECOMMENDED:
lame -h input.wav output.mp3
OPTIONS:
Input options:
-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)
--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
Operational options:
-m <mode> (s)tereo, (j)oint, (f)orce, (m)ono
default is (s) or (j) depending on bitrate
force = force ms_stereo on all frames.
-a downmix from stereo to mono file for mono encoding
--freeformat produce a free format bitstream
--decode input=mp3 file, output=wav
-t disable writing wav header when using --decode
--comp <arg> choose bitrate to achive a compression ratio of <arg>
--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>
--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
--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
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 use variable bitrate (VBR) (--vbr-old)
--vbr-old use old variable bitrate (VBR) routine
--vbr-new use new variable bitrate (VBR) routine
-V n quality setting for VBR. default n=4
0=high quality,bigger files. 9=smaller files
-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
ATH related:
--noath turns ATH down to a flat noise floor
--athshort ignore GPSYCHO for short blocks, use ATH only
--athonly ignore GPSYCHO completely, use ATH only
--athtype n selects between different ATH types [0-4]
--athlower x lowers ATH by x dB
--athaa-type n ATH auto adjust types 1-3, else no adjustment
--athaa-loudapprox n n=1 total energy or n=2 equal loudness curve
--athaa-sensitivity x activation offset in -/+ dB for ATH auto-adjustment
PSY related:
--short use short blocks when appropriate
--noshort do not use short blocks
--allshort use only short blocks
--cwlimit <freq> compute tonality up to freq (in kHz) default 8.8717
--notemp disable 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
--shortthreshold x,y short block switching threshold, x for L/R/M channel,
y for S channel
Noise Shaping related:
--substep n use pseudo substep noise shaping method types 0-2
experimental switches:
-X n[,m] selects between different noise measurements
n for long block, m for short. if m is omitted, m = n
-Y lets LAME ignore noise in sfb21, like in CBR
-Z [n] toggles the scalefac-scale and subblock gain feature on
if n is set and minus, only scalefac-scale is enabled
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:
-k keep ALL frequencies (disables all filters),
Can cause ringing and twinkling
--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> 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
--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, defau
lt)
3,4 = High priority (HIGH_PRIORITY_CLASS))
Note: Calling '--priority' without a parameter will select priority 0.
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 80 96 112 128 144 160
LAME version 3.96 MMX (http://www.mp3dev.org/)
usage: lame [options] <infile> [outfile]
<infile> and/or <outfile> can be "-", which means stdin/stdout.
RECOMMENDED:
lame -h input.wav output.mp3
OPTIONS:
Input options:
-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)
--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
Operational options:
-m <mode> (s)tereo, (j)oint, (f)orce, (m)ono
default is (s) or (j) depending on bitrate
force = force ms_stereo on all frames.
-a downmix from stereo to mono file for mono encoding
--freeformat produce a free format bitstream
--decode input=mp3 file, output=wav
-t disable writing wav header when using --decode
--comp <arg> choose bitrate to achive a compression ratio of <arg>
--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>
--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
--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
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 use variable bitrate (VBR) (--vbr-old)
--vbr-old use old variable bitrate (VBR) routine
--vbr-new use new variable bitrate (VBR) routine
-V n quality setting for VBR. default n=4
0=high quality,bigger files. 9=smaller files
-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
ATH related:
--noath turns ATH down to a flat noise floor
--athshort ignore GPSYCHO for short blocks, use ATH only
--athonly ignore GPSYCHO completely, use ATH only
--athtype n selects between different ATH types [0-4]
--athlower x lowers ATH by x dB
--athaa-type n ATH auto adjust types 1-3, else no adjustment
--athaa-loudapprox n n=1 total energy or n=2 equal loudness curve
--athaa-sensitivity x activation offset in -/+ dB for ATH auto-adjustment
PSY related:
--short use short blocks when appropriate
--noshort do not use short blocks
--allshort use only short blocks
--cwlimit <freq> compute tonality up to freq (in kHz) default 8.8717
--notemp disable 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
--shortthreshold x,y short block switching threshold, x for L/R/M channel,
y for S channel
Noise Shaping related:
--substep n use pseudo substep noise shaping method types 0-2
experimental switches:
-X n[,m] selects between different noise measurements
n for long block, m for short. if m is omitted, m = n
-Y lets LAME ignore noise in sfb21, like in CBR
-Z [n] toggles the scalefac-scale and subblock gain feature on
if n is set and minus, only scalefac-scale is enabled
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:
-k keep ALL frequencies (disables all filters),
Can cause ringing and twinkling
--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> 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
--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, defau
lt)
3,4 = High priority (HIGH_PRIORITY_CLASS))
Note: Calling '--priority' without a parameter will select priority 0.
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 80 96 112 128 144 160
Headphone : Audio-Technica ATH-W11JPN
Amp : Cute encore + Supplier
Interconnect : Furutech Sincere-2 + Apogee wyde eye
Source : E-MU 0404 + LITE DAC-AH
Speaker : AL AVS300B
-
kuang_tjktjvivi
- 初學者
- 文章: 5
- 註冊時間: 2004-09-06 19:30
http://jthz.com/~lame/kuang_tjktjvivi 寫:謝謝個位,只是我用的是p200.主要是要進行一些測試工作.但是明顯感覺用了-q 9的速度都不是很快.我聽說有lame 4.0但是找不到linux版本,所以現在很著急.請個位幫個忙!
真的謝謝了!!
-
kuang_tjktjvivi
- 初學者
- 文章: 5
- 註冊時間: 2004-09-06 19:30