This portion of the MultimediaWiki tracks an effort to get an open, freely-distributable, usable, and clear specification for the Advanced Audio Coding (AAC) format. The goal is to understand enough details about the format to create new decoder implementations that can handle production bitstreams starting with data packaged inside MPEG-4 files.
The homepage for libfaad has a Wiki that provides some decent details regarding the background coding concepts: http://www.audiocoding.com/modules/wiki/?page=AAC
AAC is a perceptual audio codec which means that it throws away certain information during the compression process, information that has been deemed less important.
Surface details of the format can be found at Wikipedia: http://en.wikipedia.org/wiki/Advanced_Audio_Coding
Conformance vectors can be obtained here: ftp://mpaudconf:email@example.com/
AAC is a variable bitrate (VBR) block-based codec where each block decodes to 1024 time-domain samples. Allegedly, each frame stands alone and does not depend on previous frames (whereas many perceptual audio codecs overlap data with the previous frame).
AAC includes a variety of profiles:
- low complexity (LC): reported to be the simplest (Apple iTunes files)
- main (MAIN): LC profile with backwards prediction
- sample-rate scalability (SRS): submitted by Sony and reportedly similar to ATRAC/3
- long term prediction (LTP): main profile with forward prediction
- high efficiency (HE, HE-AAC, aacPlus): uses Spectral Band Replication (SBR) and may use Parametric Stereo (PS)
- FAAD refers to another profile named LD, possibly the same as SRS
- provisions all over the libfaad source for error recovery (ER)
Done in most significant byte first, most significant bit first. Example:
5 bits: 2 (00010) 4 bits: 4 (0100) 4 bits: 2 (0010) 3 bits: 0 (000) Byte 1: 00010010 Byte 2: 00010000 00010010 00010000 [ 2 ][ 4 ][2 ]
Packaging/Encapsulation And Setup Data
There is a variety of methods for packaging AAC data from transport. 2 methods used in packaging raw streams are to use ADTS and ADIF headers. The libfaad knowledge base also makes reference to LATM and LOAS packaging.
Much AAC data is encapsulated in MPEG-4 files which is an extension of the QuickTime container format. the MPEG-4 file will have an audio 'trak' atom which will contain a 'stsd' description atom which will contain an 'mp4a' atom which will contain an 'esds' atom. Part of the esds atom contains the setup data for associated AAC stream. (TODO: need to document the precise format and method for obtaining the setup data.) This setup data is generally 2 bytes. This setup data has the following layout:
5 bits: object type 4 bits: frequency index if (frequency index == 15) 24 bits: frequency 4 bits: channel configuration 1 bit: frame length flag 1 bit: dependsOnCoreCoder 1 bit: extensionFlag
Object type and sampling frequency (index) are described in detail in the MPEG-4 Audio article.
frame length flag:
- 0: Each packet contains 1024 samples
- 1: Each packet contains 960 samples
Frames And Syntax Elements
In an MPEG-4 file, the AAC data is broken up into a series of variable length frames.
An AAC frame is comprised of blocks called syntax elements. Read the first 3 bits from the frame's bitstream to find the first element type. Decode the element. Proceed to read the first 3 bits of the next element and repeat the decoding process until the frame is depleted.
There are 8 different syntax elements:
- 0 SCE single channel element (codes a single audio channel)
- 1 CPE channel pair element (codes stereo signal)
- 2 CCE something to do with channel coupling, not implemented in libfaad2
- 3 LFE low-frequency effects? referenced as "special effects" in RTP doc
- 4 DSE data stream element (user data)
- 5 PCE program configuration element (describe bitstream)
- 6 FIL fill element (pad space/extension data)
- 7 END marks the end of the frame
This is an example layout for a 5.1 audio stream:
SCE CPE CPE LFE END
center - left/right - surround left/right - lfe - end
An ID within the respective CPE blocks indicates its channel assignments (front vs. surround).
First, let's list a few basic terms that FAAD2 uses throughout its decoding process:
- ics = individual channel stream, the basic audio unit that FAAD2 is concerned with
- ms = any parameter with this in its name deals with mid/side coding
- sfb = probably something to do with scale factors
- swb = scalefactor window band
- is = intensity stereo
As mentioned above, the ics is an important data structure in AAC decoding. These are its fields, according to FAAD2:
max_sfb num_swb num_window_groups num_windows window_sequence window_group_length window_shape scale_factor_grouping section_sfb_offset[8*15] swb_offset section_codebook[15*8] section_start[15*8] section_end[15*8] sfb_codebook[15*8] number_sections // number of sections in a group global_gain scale_factors // FAAD2 comment: [0..255]? ms_mask_present ms_used[MAX_WINDOW_GROUPS][MAX_SFB] // dimensions =  noise_used pulse_data_present tns_data_present gain_control_data_present predictor_data_present pulse_info pulse tns_info tns data structures for main profile, document later data structures for LTP, document later data structures for SSR, document later data structures for error resilience, document later
These pages detail the process for decoding the various syntax elements: