FAAC For Linux

The FAAC project includes the AAC encoder FAAC and decoder FAAD2.

FAAC supports several MPEG-4 object types (LC, LTP, HE AAC, Main, PS) and file formats (raw AAC, MP4, ADTS AAC), multichannel and gapless en/decoding as well as MP4 metadata tags.

The codecs are compatible with standard-compliant audio applications using one or more of these profiles.

1. Make sure you have autoconf, automake and libtool installed. For MP4 support, you must have libmp4v2 (included in faad2) installed. 2. cd to FAAC source dir 3. Run:

./bootstrap ./configure make make install

faac [options]

Options:

-a X Set average bitrate to approximately X kbps per channel (i.e. using -a 64 averages at 128 kbps/stereo). -c < bandwidth > Set the bandwidth in Hz (default value depends on sample rate) -q < quality > Set quantizer quality (default: 100, averages at approx. 128 kbps VBR for a normal stereo input file at 16 bit and 44.1 kHz sample rate). --tns Enable TNS coding. --notns Disable TNS coding. -n Disable mid/side coding. -m X AAC MPEG version, X can be 2 or 4 (default: MPEG-2, so for the sake of interoperability with non-standard compliant players like QuickTime 6 you should set it to "4"). -o X AAC object type, X can be LC, MAIN or LTP (default: LC, for the same reason as with the MPEG version don't use Main or LTP). -r RAW AAC output file (i.e. without ADTS headers). -P Raw PCM input mode. -R Raw PCM input sample rate in Hz (default: 44100 Hz). -B Raw PCM input bit depth (default: 16 bits, also possible 8 bits). -C Raw PCM input channels (default: 2). - < stdin > If you simply use a hyphen/minus sign instead of an input file name, FAAC can encode directly from stdin, thus enabling piping within other applications like foobar2000 or mp4live.

Note: VBR output bitrate depends on -q AND -c, so you should only vary the default setting -q 100 -c 16000 if you know what you're doing and/or want to experiment with other cutoff frequencies at a given quality setting.

The ABR setting with -a is an approximate average bitrate that does not use a bit reservoir, i.e -a 64 and -q 100 at 44.1 kHz will result in exactly the same output file.

The following list should give some orientation for useful -q and -c settings, based on FAAC v1.17. The resulting VBR bitrates are referring to an average sounding stereo file with 16bit, 44.1 kHz, i.e. ct_reference.wav in this case. Multiplexing these AAC files to MP4 with e.g. mp4creator will result in a ~3 kbps lower bitrate because of the stripped ADTS headers:

-q 130 -c 22000 -m 4 (~218 kbps) -q 120 -c 20000 -m 4 (~194 kbps) -q 110 -c 18000 -m 4 (~158 kbps) -q 100 -c 16000 -m 4 (~129 kbps) -q 90 -c 14000 -m 4 (~103 kbps) -q 80 -c 12000 -m 4 (~79 kbps) -q 70 -c 10000 -m 4 (~62 kbps)

The added -m 4 switch does not change the bitrate or sound of course, but is recommended for most AAC/MP4 players that use an updated FAAD2-based plugin from this year (Winamp 2.x, foobar2000 etc.) or can't decode MPEG-2 AAC LC files like QuickTime 6. Philips Expanium users should not use this switch, because their CD portable does not know MPEG-4 AAC files.