Hi there!
Please, where I can learn about MP3 VBR and "Q0", "V0", "Q1", "V1"? I know what VBR is, but I know nothing about "Q0", etc...
Thanks in advance!
Some descriptions here:
https://wiki.hydrogenaud.io/index.php/LAME
https://svn.code.sf.net/p/lame/svn/trunk/lame/USAGE
Cool!!!
Thank you very much!
It is easier to find the answer in the html version (although most of the content should be equal or very similar)
https://svn.code.sf.net/p/lame/svn/trunk/lame/doc/html/detailed.html#q
https://svn.code.sf.net/p/lame/svn/trunk/lame/doc/html/detailed.html#V
Dear
@[JAZ] Thanks a lot! It's very useful! I really liked it!
Anyway, EZ CD Audio Converter (
program I used long before foobar2000) uses
Q0,
Q1,
Q2 and some articles above also use
V0,
V1,
V2... What's the difference???
Dear @[JAZ]
Thanks a lot! It's very useful! I really liked it!
Anyway, EZ CD Audio Converter (program I used long before foobar2000) uses Q0, Q1, Q2 and some articles above also use V0, V1, V2... What's the difference???
They are some of the input switches/ parameters to the lame encoder.
Here is the output from
lame.exe -?LAME 64bits version 3.100.1 (http://lame.sf.net)
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>
--swap-channel swap L/R channels
--ignorelength ignore file length in WAV header
--gain <arg> apply Gain adjustment in decibels, range -20.0 to +12.0
--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
--out-dir <dir> output dir, must exist
Input options for RAW PCM:
-r input is raw pcm
-s sfreq sampling frequency of input file (kHz) - default 44.1 kHz
--signed input is signed (default)
--unsigned input is unsigned
--bitwidth w input bit width is w (default 16)
-x force byte-swapping of input
--little-endian input is little-endian (default)
--big-endian input is big-endian
-a downmix from stereo to mono file for mono encoding
Operational options:
-m <mode> (j)oint, (s)imple, (f)orce, (d)ual-mono, (m)ono (l)eft (r)ight
default is (j)
joint = Uses 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 achieve 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
--quiet don't print anything on screen
--silent don't print anything on screen, but fatal errors
--brief print more useful information
--verbose print a lot of useful information
Noise shaping & psycho acoustic algorithms:
-q <arg> <arg> = 0...9. Default -q 3
-q 0: Highest quality, very slow
-q 9: Poor quality, but fast
-h Same as -q 2.
-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)
-Y lets LAME ignore noise in sfb21, like in CBR
(Default for V3 to V9.999)
-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
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
--buffer-constraint <constraint> available values for constraint:
default, strict, maximum
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, v2.3 tag)
--tv <id=value> user-defined frame specified by id and value (v2.3 tag)
syntax: --tv "TXXX=description=content"
--add-id3v2 force addition of version 2 tag
--id3v1-only add only a version 1 tag
--id3v2-only add only a version 2 tag
--id3v2-utf16 add following options in unicode text encoding
--id3v2-latin1 add following options in latin-1 text encoding
--space-id3v1 pad version 1 tag with spaces instead of nulls
--pad-id3v2 same as '--pad-id3v2-size 128'
--pad-id3v2-size <value> adds version 2 tag, pad with extra <value> 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
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