AAC for film audio questions

[elenhil]

I used to encode film audio tracks with Vorbis at 96 kBit/sec and recieve a decent (for me. at least) sound in about 80 MB file per average 2 hour film.

Now with all this fuss about AAC I don't know whether Vorbis is still the best choice for such tasks at this bitrate. Can anyone give his/her opinion? Is AAC better at 96 kBit/s (and at the relatively same filesize) than Vorbis? Or can it produce higher bitrates with the same resulting size as 96 Vorbis?

[stephanV]

This is a test done at 80 kbps.

http://forum.doom9.org/showthread.php?t=99009

Or can it produce higher bitrates with the same resulting size as 96 Vorbis?
That is of course never possible. filesize = file length * bit rate

If I were you, I would just keep doing what you were doing. The fuss about AAC is just mainly that: fuss. :)

[elenhil]

Thank you, stephanV, you sure did comfort me.

Still, it would be interesting if someone could tell at what bitrate AAC can approach Vorbis's sound quality. The reason is simple: Vorbis lacks adequate multi-channel support. One could cosider trading Vorbis for multi-channel AAC (out of sheer desire of the 5.1 sound) if he knew what bitrate (and filesize) it would take to produce a multi-channel AAC soundtrack of a quality comparable to 96 kBit/s Vorbis...

[stephanV]

Nero is currently the only multichannel AAC encoder available I believe, but Nero appears not to be not a good option for AAC right now.

If I'd care about multichannel audio (which I don't) I would seriously opt to just keep the ac3 sound and not recode it at all. I have doubts if the filesize gain would be worth the effort and the possible loss in quality. But that is something everybody has to decide for him/herself. :)

[[)370|\|470!2]

As an option(i'm doing so sometimes): Create dolby pro logic II stereo stream from 5.1 AC3, then encode to vorbis or aac(personally i would prefer aac because it's more flexible
imho and
also can be muxed into an avi). To make dolby pro logic II stream, you may use azid.exe or
BeSweet.

[elenhil]

As an option(i'm doing so sometimes): Create dolby pro logic II stereo stream from 5.1 AC3, then encode to vorbis or aac
I have no standalone DPL decoder, so unless there was a software one capable of tackling Vorbis or AAC. Moreover, Vorbis needs to be encoded at much greater bitrate than 96 to maintain DPL signal :(

[Inc]

Isn't Faac capable encoding 5.1 Ch ?
It even supports input via stdin so building pipes using makeavis.exe (if audio via stdout id supported) or ffmpeg.exe (ac3 decoding) maybe could work.

Freeware Advanced Audio Coder
FAAC 1.24

Usage: FAAC.EXE [options] infiles ...


Quality-related options:

-q <quality> Set default variable bitrate (VBR) quantizer quality in % (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 (default:
VBR mode; max. value 152 kbps/stereo with a 16 kHz cutoff, can be raised with a
higher -c setting).

-c <freq> Set the lowpass cutoff in Hz (default: automatic, i.e. adapts maximum
value to input sample rate according to the Nyquist theorem; changes with different
-q settings, but can also be set manually; 0 means no cutoff, only for testing
purposes).


Input/output options:

- <stdin/out> 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. The same works for stdout as well, so FAAC can pipe its output to
programs like mp4live (streaming live AAC content).

-o X Set output file name to X (only for one input file; you can use *.aac, *.mp4,
*.m4a or *.m4b as file extension, and the correct file format MPEG-2 AAC ADTS or
MPEG-4 AAC in the MP4 container will be chosen automatically then).

-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 Swap raw PCM input byte order (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 file or bitstream
if they haven't been reordered already, e.g. in the decoder).


MP4 specific options (muxing and tagging):

-w Wrap AAC data in MP4 container (default: *.m4a file extension, *.mp4 or *.m4b
also possible; enables direct MP4 file writing without a separate file muxer;
necessary for gapless MP4 files and for multiplexing FAAC output with MPEG-4
Systems content in MP4Box from the GPAC project).

--artist "X" Set artist to X, always use quotation marks around tags with spaces.

--writer "X" Set writer to X.

--title "X" Set title to X.

--genre "X" Set genre to X.

--album "X" Set album to X.

--track "X" Set track to X (number/total).

--disc "X" Set disc to X (number/total).

--year "X" Set year to X.

--comment "X" Set comment to X.

--cover-art "X" Read cover art from file X (including path), supported image formats
are GIF, JPG and PNG.

--compilation Set compilation flag.


Expert options:

--mpeg-vers X AAC MPEG version, X can be 2 or 4 (default: MPEG-2 for *.aac files,
using the -w switch or the *.mp4/*.m4a/*.m4b file extension automatically defaults
to MPEG-4).

--obj-type X AAC object type, X can be LC (Low Complexity, default), Main or LTP
(Long Term Prediction).

-r Raw AAC output mode (only for testing purposes, creates AAC bitstream without
ADTS or other headers similar to the one in a MP4 container, so AAC file properties
will not be detected correctly in FAAD2 then).

--no-tns Disable Temporal Noise Shaping.

--shortctl X Enforce block type (default: both; 1 = short only; 2 = long only).

--no-midside Don't use M/S matrixing.


Documentation:

--license Show the FAAC license.

--help Show abbreviated help.

--long-help Show complete help.Freeware Advanced Audio Coder
FAAC 1.24

Usage: FAAC.EXE [options] infiles ...


Quality-related options:

-q <quality> Set default variable bitrate (VBR) quantizer quality in % (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 (default:
VBR mode; max. value 152 kbps/stereo with a 16 kHz cutoff, can be raised with a
higher -c setting).

-c <freq> Set the lowpass cutoff in Hz (default: automatic, i.e. adapts maximum
value to input sample rate according to the Nyquist theorem; changes with different
-q settings, but can also be set manually; 0 means no cutoff, only for testing
purposes).


Input/output options:

- <stdin/out> 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. The same works for stdout as well, so FAAC can pipe its output to
programs like mp4live (streaming live AAC content).

-o X Set output file name to X (only for one input file; you can use *.aac, *.mp4,
*.m4a or *.m4b as file extension, and the correct file format MPEG-2 AAC ADTS or
MPEG-4 AAC in the MP4 container will be chosen automatically then).

-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 Swap raw PCM input byte order (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 file or bitstream
if they haven't been reordered already, e.g. in the decoder).


MP4 specific options (muxing and tagging):

-w Wrap AAC data in MP4 container (default: *.m4a file extension, *.mp4 or *.m4b
also possible; enables direct MP4 file writing without a separate file muxer;
necessary for gapless MP4 files and for multiplexing FAAC output with MPEG-4
Systems content in MP4Box from the GPAC project).

--artist "X" Set artist to X, always use quotation marks around tags with spaces.

--writer "X" Set writer to X.

--title "X" Set title to X.

--genre "X" Set genre to X.

--album "X" Set album to X.

--track "X" Set track to X (number/total).

--disc "X" Set disc to X (number/total).

--year "X" Set year to X.

--comment "X" Set comment to X.

--cover-art "X" Read cover art from file X (including path), supported image formats
are GIF, JPG and PNG.

--compilation Set compilation flag.


Expert options:

--mpeg-vers X AAC MPEG version, X can be 2 or 4 (default: MPEG-2 for *.aac files,
using the -w switch or the *.mp4/*.m4a/*.m4b file extension automatically defaults
to MPEG-4).

--obj-type X AAC object type, X can be LC (Low Complexity, default), Main or LTP
(Long Term Prediction).

-r Raw AAC output mode (only for testing purposes, creates AAC bitstream without
ADTS or other headers similar to the one in a MP4 container, so AAC file properties
will not be detected correctly in FAAD2 then).

--no-tns Disable Temporal Noise Shaping.

--shortctl X Enforce block type (default: both; 1 = short only; 2 = long only).

--no-midside Don't use M/S matrixing.


Documentation:

--license Show the FAAC license.

--help Show abbreviated help.

--long-help Show complete help.

[stephanV]

Have you ever compare quality between FAAC and, say, Vorbis?

[[)370|\|470!2]

I'd assume, that @ 64kbps/channel there won't be a huge quality gap between faac and vorbis.

[elenhil]

Still, maybe there IS a software DPL decoder for formats outside of MP3?

[[)370|\|470!2]

Still, maybe there IS a software DPL decoder for formats outside of MP3?

I'm using powerdvd6 to play files (avi with xvid/x264/aac dplII).
Though it plays these w/o problems, i'm unsure if it really processes dplII.

[stephanV]

I'd assume, that @ 64kbps/channel there won't be a huge quality gap between faac and vorbis.
@ 64 kbps per channel there is little reason to re-encode ac3...

[[)370|\|470!2]

@ 64 kbps per channel there is little reason to re-encode ac3...

True :P

[Inc]

Have you ever compare quality between FAAC and, say, Vorbis?
Not really as Im "just" changing my mind to mp4. The last years I was mpeg2/ac3 addicted. But for shure I always like to test "for free" things.

[elenhil]

Still, can anyone tell at what bitrate does AAC approach Vorbis @96 quality?

[[)370|\|470!2]

Compare yourself - is teh best way to know^^
iTunes aac @112 kbps(stereo) sounds quite reasonable to me.