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 Readme data for Root » Audio » Convert » faac.lha

Description: Freeware Advanced Audio Coder
Install: faac.lha
Size: 539kb 5
Version: 1.28
Date: 02 Jul 10
Author: Michael Trebilcock (OS4 compile)
Submitter: Michael Trebilcock
Category: audio/convert
License: LGPL
Distribute: yes
Min OS Version: 4.0
Changes: faac-shared, uses libfaac.so and also includes MP4 container support.

FAAC is an Advanced Audio Coder (MPEG2-AAC, MPEG4-AAC).

The goal of FAAC is to explore the possibilities of AAC and exceed the quality
of the currently best MP3 encoders.

Freeware Advanced Audio Coder
FAAC 1.28

Usage: faac [options] infiles ...

Quality-related options:
  -q <quality>	Set default variable bitrate (VBR) quantizer quality in percent.
		(default: 100, averages at approx. 120 kbps VBR for a normal
		stereo input file with 16 bit and 44.1 kHz sample rate; max.
		value 500, min. 10).
  -b <bitrate>	Set average bitrate (ABR) to approximately <bitrate> kbps.
		(max. value 152 kbps/stereo with a 16 kHz cutoff, can be raised
		with a higher -c setting).
  -c <freq>	Set the bandwidth in Hz (default: automatic, i.e. adapts
		maximum value to input sample rate).

Input/output options:
  -		<stdin/stdout>: If you simply use a hyphen/minus sign instead
		of an input file name, FAAC can encode directly from stdin,
		thus enabling piping from other applications and utilities. The
		same works for stdout as well, so FAAC can pipe its output to
		other apps such as a server.
  -o X		Set output file to X (only for one input file)
		only for one input file; you can use *.aac, *.mp4, *.m4a or
		*.m4b as file extension, and the file format will be set
		automatically to ADTS or MP4).
  -P		Raw PCM input mode (default: off, i.e. expecting a WAV header;
		necessary for input files or bitstreams without a header; using
		only -P assumes the default values for -R, -B and -C in the
		input file).
  -R		Raw PCM input sample rate in Hz (default: 44100 Hz, max. 96 kHz)
  -B		Raw PCM input sample size (default: 16, also possible 8, 24, 32
		bit fixed or float input).
  -C		Raw PCM input channels (default: 2, max. 33 + 1 LFE).
  -X		Raw PCM swap input bytes (default: bigendian).
  -I <C[,LFE]>	Input multichannel configuration (default: 3,4 which means
		Center is third and LFE is fourth like in 5.1 WAV, so you only
		have to specify a different position of these two mono channels
		in your multichannel input files if they haven't been reordered
		already).

MP4 specific options:
  -w		Wrap AAC data in MP4 container. (default for *.mp4, *.m4a and
		*.m4b)
  -s		Optimize MP4 container layout after encoding.
  --artist X	Set artist to X
  --writer X	Set writer/composer to X
  --title X	Set title/track name to X
  --genre X	Set genre to X
  --album X	Set album/performer to X
  --compilation	Mark as compilation
  --track X	Set track to X (number/total)
  --disc X	Set disc to X (number/total)
  --year X	Set year to X
  --cover-art X	Read cover art from file X
		Supported image formats are GIF, JPEG, and PNG.
  --comment X	Set comment to X

Expert options, only for testing purposes:
  --tns  	Enable coding of TNS, temporal noise shaping.
  --no-midside	Don't use mid/side coding.
  --mpeg-vers X	Force AAC MPEG version, X can be 2 or 4
  --obj-type X	AAC object type. (LC (Low Complexity, default), Main or LTP
		(Long Term Prediction)
  --shortctl X	Enforce block type (0 = both (default); 1 = no short; 2 = no
		long).
  -r		Generate raw AAC bitstream (i.e. without any headers).
		Not advised!!!, RAW AAC files are practically useless!!!

Documentation:
  --license	Show the FAAC license.
  --help	Show this abbreviated help.
  --long-help	Show complete help.

  More tips can be found in the audiocoding.com Knowledge Base at
  <http://www.audiocoding.com/wiki/>






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