NAME
twolame - an optimised MPEG Audio Layer 2 (MP2) encoderSYNOPSIS
twolame [options] <infile> [outfile]DESCRIPTION
TwoLAME is an optimised MPEG Audio Layer 2 (MP2) encoder based on tooLAME by Mike Cheng, which in turn is based upon the ISO dist10 code and portions of LAME. Encoding is performed by the libtwolame library backend.OPTIONS
Input File
twolame uses libsndfile for reading the input sound file, so the input file can be in any format supported by libsndfile. To read raw PCM audio from STDIN, then use - as the input filename.Output File
If no output filename is specified, then suffix of the input filename is automatically changed to .mp2. To write the encoded audio to STDOUT then use - as the output filename.Input Options
-r, --raw-inputSpecifies that input is raw signed PCM audio.
If audio is stereo, than audio samples are interleaved between the two
channels.
-x, --byte-swap
Force byte-swapping of the input. Endian
detection is performed automatically by libsndfile, so this option
shouldn’t normally be needed.
-s, --samplerate <int>
If inputting raw PCM sound, you must specify
the sample rate of the audio in Hz. Valid sample rates: 16000, 22050, 24000,
32000, 44100, 48000Hz. Default sample rate is 44100Hz.
--samplesize <int>
Specifies the sample size (in bits) of the raw
PCM audio. Valid sample sizes: 8, 16, 24, 32. Default sample size is
16-bit.
-N, --channels <int>
If inputting raw PCM sound, you must specify
the number of channels in the input audio. Default number of channels is
2.
-g, --swap-channels
Swap the Left and Right channels of a stereo
input file.
--scale <float>
Scale the input audio prior to encoding. All
of the input audio is multiplied by specified value. Value between 0 and 1
will reduce the audio gain, and a value above 1 will increase the gain of the
audio.
--scale-l <float>
Same as --scale, but only affects the left
channel.
--scale-r <float>
Same as --scale, but only affects the right
channel.
Output Options
-m, --mode <char>Choose the mode of the resulting audio.
Default is auto.
-a, --downmix
•"a" auto - choose mode
automatically based on the input
•"s" stereo
•"d" dual channel
•"j" joint stereo
•"m" mono
If the input file is stereo then, downmix the
left and right input channels into a single mono channel.
-b, --bitrate <int>
Sets the total bitrate (in kbps) for the
output file. The default bitrate depends on the number of input channels and
samplerate.
-P, --psyc-mode <int>
------------------------------ Sample Rate Mono Stereo ------------------------------ 48000 96 192 44100 96 192 32000 80 160 24000 48 96 22050 48 96 16000 32 64 ------------------------------
Choose the psycho-acoustic model to use (-1 to
4). Model number -1 is turns off psycho-acoustic modelling and uses fixed
default values instead. Please see the file psycho for a full
description of each of the models available. Default model is 3.
-v, --vbr
Enable VBR mode. See vbr documentation
file for details. Default VBR level is 5.0.
-V, --vbr-level <float>
Enable VBR mode and set quality level. The
higher the number the better the quality. Maximum range is -50 to 50 but
useful range is -10 to 10. See vbr documentation file for
details.
-l, --ath <float>
Set the ATH level. Default level is 0.0.
-q, --quick <int>
Enable quick mode. Only re-calculate
psycho-acoustic model every specified number of frames.
-S, --single-frame
Enables single frame mode: only a single frame
of MPEG audio is output and then the program terminates.
Miscellaneous Options
-c, --copyrightTurn on Copyright flag in output
bitstream.
-o, --non-original
Turn off Original flag in output
bitstream.
--original
Turn on Original flag in output
bitstream.
-p, --protect
Enable CRC error protection in output
bitstream. An extra 16-bit checksum is added to frames.
-d, --padding
Turn on padding in output bitstream.
-R, --reserve <int>
Reserve specified number of bits in the each
from of the output bitstream.
-e, --deemphasis <char>
Set the de-emphasis type (n/c/5). Default is
none.
-E, --energy
Turn on energy level extensions.
Verbosity Options
-t, --talkativity <int>Set the amount of information to be displayed
on stderr (0 to 10). Default is 2.
--quiet
Don’t send any messages to stderr,
unless there is an error. (Same as --talkativity=0)
--brief
Only display a minimal number of messages
while encoding. This setting is quieter than the default talkativity setting.
(Same as --talkativity=1)
--verbose
Display an increased number of messages on
stderr. This setting is useful to diagnose problems. (Same as
--talkativity=4)
RETURN CODES
If encoding completes successfully, then twolame will return 0. However if encoding is not successful, then it will return one of the following codes.•1 (No encoding performed)
•2 (Error opening input file)
•4 (Error opening output file)
•6 (Error allocating memory)
•8 (Error in chosen encoding
parameters)
•10 (Error reading input audio)
•12 (Error occurred while
encoding)
•14 (Error writing output audio)
EXAMPLES
This will encode sound.wav to sound.mp2 using the default constant bitrate of 192 kbps and using the default psycho-acoustic model (model 3):twolame sound.wav
twolame -b 160 -m j sound.aiff sound_160.mp2
twolame -P 2 -v sound.wav newfile.mp2
twolame -P 2 -V -5 sound.wav newfile.mp2
sox sound_11025.aiff -t raw -r 16000 | twolame -r -s 16000 - - > out.mp2
AUTHORS
The twolame frontend was (re)written by Nicholas J Humfrey. The libtwolame library is based on toolame by Mike Cheng. For a full list of authors, please see the AUTHORS file.RESOURCES
TwoLAME web site: http://www.twolame.org/SEE ALSO
lame(1), mpg123(1), madplay(1), sox(1)COPYING
Copyright © 2004-2018 The TwoLAME Project. Free use of this software is granted under the terms of the GNU Lesser General Public License (LGPL).AUTHOR
Nicholas J Humfrey <[email protected]>Author.
| 10/11/2019 |