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TWOLAME(1)							    TWOLAME(1)

       twolame - an optimised MPEG Audio Layer 2 (MP2) encoder

       twolame [options] <infile> [outfile]

       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

   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-input
	   Specifies 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

       -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

       --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.

	   o   "a" auto	- choose mode automatically based on the input

	   o   "s" stereo

	   o   "d" dual	channel

	   o   "j" joint stereo

	   o   "m" mono

       -a, --downmix
	   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.

	       Sample Rate   Mono    Stereo
	       48000	     96	     192
	       44100	     96	     192
	       32000	     80	     160
	       24000	     48	     96
	       22050	     48	     96
	       16000	     32	     64

       -P, --psyc-mode <int>
	   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, --copyright
	   Turn	on Copyright flag in output bitstream.

       -o, --non-original
	   Turn	off Original flag in output bitstream.

	   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

       -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.

	   Don't send any messages to stderr, unless there is an error.	(Same
	   as --talkativity=0)

	   Only	display	a minimal number of messages while encoding. This
	   setting is quieter than the default talkativity setting. (Same as

	   Display an increased	number of messages on stderr. This setting is
	   useful to diagnose problems.	(Same as --talkativity=4)

       If encoding completes successfully, then	twolame	will return 0. However
       if encoding is not successful, then it will return one of the following

       o   1 (No encoding performed)

       o   2 (Error opening input file)

       o   4 (Error opening output file)

       o   6 (Error allocating memory)

       o   8 (Error in chosen encoding parameters)

       o   10 (Error reading input audio)

       o   12 (Error occurred while encoding)

       o   14 (Error writing output audio)

       This will encode	sound.wav to sound.mp2 using the default constant
       bitrate of 192 kbps and using the default psycho-acoustic model (model

	   twolame sound.wav

       Constant	bitrate	of 160kbps and joint stereo encoding, saved to file

	   twolame -b 160 -m j sound.aiff sound_160.mp2

       Encode sound.wav	to newfile.mp2 using psycho-acoustic model 2 and
       encoding	with variable bitrate:

	   twolame -P 2	-v sound.wav newfile.mp2

       Same as example above, except that the negative value of	the "-V"
       argument	means that the lower bitrates will be favoured over the	higher

	   twolame -P 2	-V -5 sound.wav	newfile.mp2

       Resample	audio file using sox and pipe straight through twolame:

	   sox sound_11025.aiff	-t raw -r 16000	| twolame -r -s	16000 -	- > out.mp2

       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.

       TwoLAME web site:

       lame(1),	mpg123(1), madplay(1), sox(1)

       Copyright (C) 2004-2018 The TwoLAME Project. Free use of	this software
       is granted under	the terms of the GNU Lesser General Public License

       Nicholas	J Humfrey <>

				  10/11/2019			    TWOLAME(1)


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