Advanced Audio Coding

Advanced Audio Coding

Advanced Audio Coding (AAC) is a standardized, lossy compression and encoding scheme for digital audio. Designed to be the successor of the MP3 format, AAC generally achieves better sound quality than MP3 at many bit rates.

AAC has been standardized by ISO and IEC, as part of the MPEG-2 & MPEG-4 specifications. The MPEG-2 standard contains several audio coding methods, including the MP3 coding scheme. AAC is able to include 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 15 low frequency enhancement (LFE, limited to 120 Hz) channels and up to 15 data streams. AAC is able to achieve indistinguishable audio quality at data rates of 320 kbit/s (64kbit/s/channel) for five channels. The quality is close to CD also at 96 kbit/s (48kbit/s/channel) for stereo.

AAC's best known use is as the default audio format of Apple's iPhone, iPod, iTunes, and the format used for all iTunes Store audio (with extensions for proprietary digital rights management).

AAC is also the standard audio format for Sony’s PlayStation 3 and is supported by Sony's Playstation Portable, latest generation of Sony Walkman, Walkman Phones from Sony Ericsson, Nintendo's Wii (with the Photo Channel 1.1 update installed for Wii consoles purchased before late 2007),The nintendo DSi, and the MPEG-4 video standard.

HE-AAC is part of digital radio standards like DAB+ and Digital Radio Mondiale.


AAC was developed with the cooperation and contributions of companies including Fraunhofer IIS, AT&T Bell Laboratories, Dolby, Sony Corporation and Nokia, and was officially declared an international standard by the Moving Pictures Experts Group in April 1997. MPEG-2 AAC LC profile consists of a base format very much like AT&T's PAC coding format , with the addition of TNS, the Dolby Kaiser Window described below, a nonuniform quantizer, and a reworking of the bitstream format to handle up to 16 stereo, 16 mono, 16 LFE, and 16 commentary channels in one bitstream. The Main profile adds a set of recursive predictors that are calculated on each tap of the filterbank. The SSR uses a 4-band PQMF filterbank, with four shorter filterbanks following, in order to allow for scalable sampling rates.


It is specified both as Part 7 of the MPEG-2 standard, and Part 3 of the MPEG-4 standard. As such, it can be referred to as MPEG-2 Part 7 and MPEG-4 Part 3 depending on its implementation, however it is most often referred to as MPEG-4 AAC, or AAC for short.

AAC was first specified in the standard MPEG-2 Part 7 (known formally as ISO/IEC 13818-7:1997) in 1997 as a new "part" (distinct from ISO/IEC 13818-3) in the MPEG-2 family of international standards.

It was updated in MPEG-4 Part 3 (known formally as ISO/IEC 14496-3:1999) in 1999. The reference software is specified in MPEG-4 Part 4 and the conformance bit-streams are specified in MPEG-4 Part 5. A notable addition in this version of the standard is Perceptual Noise Substitution (PNS).

HE-AAC (AAC with SBR) was first standardized in ISO/IEC 14496-3:2001/Amd.1. HE-AAC v2 (AAC with Parametric Stereo) was first specified in ISO/IEC 14496-3:2001/Amd.4.

The current version of the AAC standard is ISO/IEC 14496-3:2005 (with 14496-3:2005/Amd.2. for HE-AAC v2)

AacPlus v2 is also standardized by ETSI (European Telecommunications Standards Institute) as TS 102005.

The MPEG4 standard also contains other ways of compressing sound. These are low bit rate and generally used for speech.

AAC’s improvements over MP3

AAC was designed to fix many of the serious performance flaws in the MP3 format (which was specified in MPEG-1 and MPEG-2) by the ISO/IEC in 11172-3 and 13818-3.

Improvements include:

  • More sample frequencies (from 8 kHz to 96 kHz) than MP3 (16 kHz to 48 kHz)
  • Up to 48 channels (MP3 supports up to two channels in MPEG-1 mode and up to 5.1 channels in MPEG-2 mode)
  • Arbitrary bit-rates and variable frame length. Standardized constant bit rate with bit reservoir.
  • Higher efficiency and simpler filterbank (rather than MP3's hybrid coding, AAC uses a pure MDCT)
  • Higher coding efficiency for stationary signals (AAC uses a blocksize of 1024 samples, allowing more efficient coding than MP3's 576 sample blocks)
  • Higher coding accuracy for transient signals (AAC uses a blocksize of 128 samples, allowing more accurate coding than MP3's 192 sample blocks)
  • Can use Kaiser-Bessel derived window function to eliminate spectral leakage at the expense of widening the main lobe
  • Much better handling of audio frequencies above 16 kHz
  • More flexible joint stereo (separate for every scale band)
  • Adds additional modules (tools) to increase compression efficiency: TNS, Backwards Prediction, PNS etc... These modules can be combined to constitute different encoding profiles.

Overall, the AAC format allows developers more flexibility to design codecs than MP3 does, and corrects many of the unfortunate design choices made in the original MPEG 1 audio specification. This increased flexibility often leads to more concurrent encoding strategies and, as a result, to more efficient compression. However in terms of whether AAC is better than MP3, the advantages of AAC are not entirely decisive, and the MP3 specification, while outdated, has proven surprisingly robust in spite of considerable flaws. AAC and HE-AAC are universally accepted as better than MP3 at low bitrates (typically less than 128 kbit/s). This is especially true at very low bitrates where the superior stereo coding, pure MDCT, and more optimal transform window sizes leave MP3 unable to compete. However, as bitrate increases, the efficiency of an audio format becomes less important relative to the efficiency of the encoder's implementation, and the intrinsic advantage AAC holds over MP3 no longer dominates audio quality.

How AAC works

AAC is a wideband audio coding algorithm that exploits two primary coding strategies to dramatically reduce the amount of data needed to represent high-quality digital audio.

  1. Signal components that are perceptually irrelevant are discarded;
  2. Redundancies in the coded audio signal are eliminated.

The actual encoding process consists of the following steps:

  • The signal is converted from time-domain to frequency-domain using forward modified discrete cosine transform (MDCT). This is done by using filter banks that takes appropriate amount of time samples and convert them to frequency samples.
  • The frequency domain signal is quantized based on psychoacoustics model and encoded.
  • Internal error correction codes are added;
  • The signal is stored or transmitted.
  • In order to prevent corrupt samples, a modern implementation of the Luhn mod N algorithm is applied to each frame

The MPEG-4 audio standard does not define a single or small set of highly efficient compression schemes but rather a complex toolbox to perform a wide range of operations from low bitrate speech coding to high-quality audio coding and music synthesis.

  • The MPEG-4 audio coding algorithm family spans the range from low bitrate speech encoding (down to 2 kbit/s) to high-quality audio coding (at 64 kbit/s per channel and higher).
  • AAC offers sampling frequencies between 8 kHz and 96 kHz and any number of channels between 1 and 48.
  • In contrast to MP3's hybrid filter bank, AAC uses the modified discrete cosine transform (MDCT) together with the increased window lengths of 1024 points.

AAC encoders can switch dynamically between a single MDCT block of length 1024 points or 8 blocks of 128 points.

  • If a signal change or a transient occurs, 8 shorter windows of 128 points each are chosen for their better temporal resolution.
  • By default, the longer 1024-point window is otherwise used because the increased frequency resolution allows for a more sophisticated psychoacoustic model, resulting in improved coding efficiency.

Modular encoding

AAC takes a modular approach to encoding. Depending on the complexity of the bitstream to be encoded, the desired performance and the acceptable output, implementers may create profiles to define which of a specific set of tools they want to use for a particular application. The standard offers four default profiles:

  • Low Complexity (LC) - the simplest and most widely used and supported;
  • Main Profile (MAIN) - like the LC profile, with the addition of backwards prediction;
  • Sample-Rate Scalable (SRS), a.k.a. Scalable Sample Rate (MPEG-4 AAC-SSR);
  • Long Term Prediction (LTP); added in the MPEG-4 standard – an improvement of the MAIN profile using a forward predictor with lower computational complexity.

Depending on the AAC profile and the MP3 encoder, 96 kbit/s AAC can give nearly the same or better perceptual quality as 128 kbit/s MP3.

AAC error protection toolkit

Applying error protection enables error correction up to a certain extent. Error correcting codes are usually applied equally to the whole payload. However since different parts of an AAC payload show different sensitivity to transmission errors, this would not be a very efficient approach.

The AAC payload can be subdivided into parts with different error sensitivities.

  • Independent error correcting codes can be applied to any of these parts using the Error Protection (EP) tool defined in MPEG-4 Audio standard.
  • This toolkit provides the error correcting capability to the most sensitive parts of the payload in order to keep the additional overhead low.
  • The toolkit is backwardly compatible simpler and pre-existing AAC decoders. A great deal of the tool kit's error correction functions are based around spreading information about the audio signal more evenly in the datastream.

Error Resilient (ER) AAC

Error Resilience (ER) techniques can be used to make the coding scheme itself more robust against errors.

For AAC, three custom-tailored methods were developed and defined in MPEG-4 Audio

  • Huffman Codeword Reordering (HCR) to avoid error propagation within spectral data;
  • Virtual Codebooks (VCB11) to detect serious errors within spectral data;
  • Reversible Variable Length Code (RVLC) to reduce error propagation within scale factor data.

AAC Low Delay

The MPEG-4 Low Delay Audio Coder (AAC-LD) is designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the MPEG-2 Advanced Audio Coding (AAC) format.

Licensing and Patents

No licenses or payments are required to be able to stream or distribute content in AAC format. This reason alone makes AAC a much more attractive format to distribute content than MP3, particularly for streaming content (such as Internet radio).

However, a patent license is required for all manufacturers or developers of AAC codecs . It is for this reason FOSS implementations such as FAAC and FAAD are distributed in source form only, in order to avoid patent infringement. (See below under Products that support AAC, Software.)

Products that support AAC

HDTV Standards

Japanese ISDB-T

In December 2003, Japan started broadcasting terrestrial DTV ISDB-T standard that implements Mpeg2 video and Mpeg2 AAC audio. In April 2006 Japan started broadcasting the ISDB-T mobile sub-program, called 1Seg, that was the first implementation of video H.264AVC with audio HE-AAC in Terrestrial HDTV broadcasting service on the planet.

International ISDB-Tb

In December 2007, Brazil started broadcasting terrestrial DTV standard called International ISDB-Tb that implements video coding H.264AVC with audio AAC-LC on main program(single or multi) and video H.264AVC with audio HE-AACv2 in the 1Seg mobile sub-program.


iTunes and iPod

In April 2003, Apple Computer brought mainstream attention to AAC by announcing that its iTunes and iPod products would support songs in MPEG-4 AAC format (via a firmware update for older iPods). Customers could download music in a proprietary Digital Rights Management (DRM)-restricted form of AAC (see FairPlay) via the iTunes Store or create files without DRM from their own CDs using iTunes. In later years, Apple began offering music videos and movies, which also use AAC for audio encoding.

On May 29, 2007, Apple began selling songs and music videos free of DRM from participating record labels. These files mostly adhere to the AAC standard and are playable on many non-Apple products but they do include custom iTunes information such as album artwork and a purchase receipt, so as to identify the customer in case the file is leaked out onto peer-to-peer networks, it is possible however to remove these custom tags to restore interoperability with players that conform strictly to the AAC specification.

iTunes supports a "Variable bit rate" (VBR) encoding option which encodes AAC tracks in an "Average bit rate" (ABR) scheme. As of October 2007, Apple has not added support for HE-AAC which is fully part of the MP4 standard or true VBR encoding to iTunes.

Other Portable Players

Mobile phones

For a number of years, many mobile phones from manufacturers such as Nokia, Motorola, Samsung, Sony Ericsson, BenQ-Siemens and Philips have supported AAC playback. The first such phone was the Nokia 5510 released in 2002 which also plays MP3s. However this phone was a commercial failure and such phones with integrated music players did not gain mainstream popularity until 2005 when the trend of having AAC as well as MP3 support continued. Most new smartphones and music-themed phones support playback of these formats.

  • Sony Ericsson phones support various AAC formats in MP4 container. AAC-LC is supported in all phones beginning with K700, phones beginning with W550 have support of HE-AAC. The latest devices such as the P990, K610, W890i and later support HE-AAC v2.
  • Nokia XpressMusic and other new generation Nokia multimedia phones: also support AAC format.
  • BlackBerry: RIM’s latest series of Smartphones such as the 8100 ("Pearl") and 8800 support AAC.
  • Apple's iPhone supports AAC and FairPlay protected AAC files used as the default encoding format in the iTunes store.

Other devices

  • Palm OS PDAs: Many Palm OS based PDAs and smartphones can play AAC and HE-AAC with the 3rd party software Pocket Tunes. Version 4.0, released in December 2006, added support for native AAC and HE-AAC files. The AAC codec for TCPMP, a popular video player, was withdrawn after version 0.66 due to patent issues, but can still be downloaded from sites other than CorePlayer, the commercial follow-on to TCPMP, includes AAC support. Other PalmOS programs supporting AAC include Kinoma Player and AeroPlayer.
  • Microsoft Windows Mobile platforms support AAC either by the native Windows Media Player or by third-party products (TCPMP, CorePlayer)
  • Epson supports AAC playback in the P-2000 and P-4000 Multimedia/Photo Storage Viewers. This support is not available with their older models, however.
  • Vosonic supports AAC recording and playback in the VP8350, VP8360 and VP8390 MultiMedia Viewers.
  • The Sony Reader portable eBook plays M4A files containing AAC, and displays metadata created by iTunes. Other Sony products, including the A and E series Network Walkmans, support AAC with firmware updates (released May 2006) while the S series supports it out of the box.
  • Nearly every major car stereo manufacturer offers models that will play back .m4a files recorded onto CD in a data format. This includes Pioneer, Sony, Alpine, Kenwood, Clarion, Panasonic, and JVC.
  • The Sonos Digital Media Player supports playback of AAC files.
  • The Roku SoundBridge network audio player supports playback of AAC encoded files.
  • The Squeezebox network audio player (made by Slim Devices, a Logitech company) supports playback of AAC files.
  • The PlayStation 3 supports encoding and decoding of AAC files.
  • The Xbox 360 supports streaming of AAC through the Zune software, and off supported iPods connected through the USB port
  • The Wii video game console supports AAC files through version 1.1 of the Photo Channel as of December 11, 2007. All AAC profiles and bitrates are supported as long as it is in the .m4a file extension. This update removed MP3 compatibility, but users who have installed this may freely downgrade to the old version if they wish.
  • The new DSi (an updated version of Nintendo's popular DS handheld) will be able to natively play AAC files from its new SD card slot.
  • The Livescribe Pulse Smartpen records and stores audio in AAC format. The audio files can be replayed using the pen's integrated speaker, attached headphones, or on a computer using the Livescribe Desktop software. The AAC files are stored in the user's "My Documents" folder of the Windows OS and can be distributed and played without specialized hardware or software from Livescribe.


The Rockbox Open source firmware (available for multiple portable players) also offers support for AAC to varying degrees, depending on the model of player and the AAC profile.

Optional iPod Support (playback of unprotected AAC files) for the Xbox 360 is available as a free download from Xbox Live.

Other software media players

Almost all current computer media players include built-in decoders for AAC, or can utilize a library to decode it. On Microsoft Windows, DirectShow can be utilized this way with the corresponding filters to enable AAC playback in any DirectShow based player. Software player applications of particular note include:

Some of these players (e.g., foobar2000, Winamp, and VLC) also support the decoding of raw or MP4-contained AAC streamed over HTTP using the SHOUTcast protocol. Plug-ins for Winamp and foobar2000 enable the creation of such streams.

Nero Digital Audio

In May 2006, Nero AG released an AAC encoding tool free of charge, Nero Digital Audio , which is capable of encoding LC-AAC, HE-AAC and HE-AAC v2 streams. The tool is a Command Line Interface tool only, and a separate utility is included to decode to PCM WAV.

Various tools including the foobar2000 audio player and MeGUI can provide a GUI for the encoder.


FAAC and FAAD2 stand for Freeware Advanced Audio Coder and Decoder 2 respectively, collectively make up an open source implementation of AAC.

Extensions and improvements

Some extensions have been added to the original AAC standard:

Container formats

In addition to the MP4 container format for storage, AAC audio data may be packaged in a more basic format called Audio Data Interchange Format (ADIF), consisting of a single header followed by the raw AAC audio data blocks. Alternatively, it may be packaged in a streaming format called Audio Data Transport Stream (ADTS), consisting of a series of frames, each frame having a header followed by the AAC audio data. Both formats are defined in MPEG-2 part 7, but are only considered informative by MPEG-4, so an MPEG-4 decoder does not need to support either format. Two more formats are defined in MPEG-4 part 3: Low-overhead MPEG-4 Audio Transport Multiplex (LATM), which provides a way to combine separate audio payloads, and Low Overhead Audio Stream (LOAS), a self-synchronizing streaming format.

See also


External links

Search another word or see advanced audio codingon Dictionary | Thesaurus |Spanish
Copyright © 2015, LLC. All rights reserved.
  • Please Login or Sign Up to use the Recent Searches feature