Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. Designed to be the successor of the MP3 format, AAC generally achieves higher sound quality than MP3 at the same bit rate.
AAC has been standardized by ISO and IEC, as part of the MPEG-2 and MPEG-4 specifications. Part of AAC, HE-AAC ("AAC+"), is part of MPEG-4 Audio and also adopted into digital radio standards DAB+ and Digital Radio Mondiale, as well as mobile television standards DVB-H and ATSC-M/H.
AAC supports the inclusion of 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 16 low-frequency effects (LFE, limited to 120 Hz) channels, up to 16 "coupling" or dialog channels, and up to 16 data streams. The quality for stereo is satisfactory to modest requirements at 96 kbit/s in joint stereo mode; however, hi-fi transparency demands data rates of at least 128 kbit/s (VBR). Tests of MPEG-4 audio have shown that AAC meets the requirements referred to as "transparent" for the ITU at 128 kbit/s for stereo, and 320 kbit/s for 5.1 audio. AAC uses a purely modified discrete cosine transform (MDCT) algorithm, giving it higher compression efficiency than MP3, which uses a hybrid coding algorithm that is part MDCT and part FFT.
AAC is the default or standard audio format for iPhone, iPod, iPad, Nintendo DSi, Nintendo 3DS, iTunes, DivX Plus Web Player, PlayStation 3, and various Nokia Series 40 phones. It is supported on PlayStation Vita, Wii (with the Photo Channel 1.1 update installed), Sony Walkman MP3 series and later, Android and BlackBerry. AAC is also supported by manufacturers of in-dash car audio systems.
The discrete cosine transform (DCT), a type of transform coding for lossy compression, was proposed by Nasir Ahmed in 1972, and developed by Ahmed with T. Natarajan and K. R. Rao in 1973, publishing their results in 1974. This led to the development of the modified discrete cosine transform (MDCT), proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987, following earlier work by Princen and Bradley in 1986. The MP3 audio coding standard introduced in 1994 used a hybrid coding algorithm that is part MDCT and part FFT. AAC uses a purely MDCT algorithm, giving it higher compression efficiency than MP3.
AAC was developed with the cooperation and contributions of companies including Bell Labs, Fraunhofer IIS, Dolby Laboratories, LG Electronics, NEC, NTT Docomo, Panasonic, Sony Corporation, ETRI, JVC Kenwood, Philips, Microsoft, and NTT. It was officially declared an international standard by the Moving Picture Experts Group in April 1997. It is specified both as Part 7 of the MPEG-2 standard, and Subpart 4 in Part 3 of the MPEG-4 standard.
In 1997, AAC was first introduced as MPEG-2 Part 7, formally known as ISO/IEC 13818-7:1997. This part of MPEG-2 was a new part since MPEG-2 already included MPEG-2 Part 3, formally known as ISO/IEC 13818-3: MPEG-2 BC (Backwards Compatible). Therefore, MPEG-2 Part 7 is also known as MPEG-2 NBC (Non-Backward Compatible), because it is not compatible with the MPEG-1 audio formats (MP1, MP2, and MP3).
MPEG-2 Part 7 defined three profiles: Low-Complexity profile (AAC-LC / LC-AAC), Main profile (AAC Main), and Scalable Sampling Rate profile (AAC-SSR). AAC-LC profile consists of a base format very much like AT&T's Perceptual Audio Coding (PAC) coding format, with the addition of temporal noise shaping (TNS), the Kaiser window (described below), a non-uniform quantizer, and a reworking of the bitstream format to handle up to 16 stereo channels, 16 mono channels, 16 low-frequency effect (LFE) channels 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.
In 1999, MPEG-2 Part 7 was updated and included in the MPEG-4 family of standards and became known as MPEG-4 Part 3, MPEG-4 Audio, or ISO/IEC 14496-3:1999. This update included several improvements. One of these improvements was the addition of Audio Object Types which are used to allow interoperability with a diverse range of other audio formats such as TwinVQ, CELP, HVXC, Text-To-Speech Interface, and MPEG-4 Structured Audio. Another notable addition in this version of the AAC standard is Perceptual Noise Substitution (PNS). In that regard, the AAC profiles (AAC-LC, AAC Main, and AAC-SSR profiles) are combined with perceptual noise substitution and are defined in the MPEG-4 audio standard as Audio Object Types. MPEG-4 Audio Object Types are combined in four MPEG-4 Audio profiles: Main (which includes most of the MPEG-4 Audio Object Types), Scalable (AAC LC, AAC LTP, CELP, HVXC, TwinVQ, Wavetable Synthesis, TTSI), Speech (CELP, HVXC, TTSI) and Low Rate Synthesis (Wavetable Synthesis, TTSI).
The reference software for MPEG-4 Part 3 is specified in MPEG-4 Part 5 and the conformance bit-streams are specified in MPEG-4 Part 4. MPEG-4 Audio remains backward-compatible with MPEG-2 Part 7.
The MPEG-4 Audio Version 2 (ISO/IEC 14496-3:1999/Amd 1:2000) defined new audio object types: the low delay AAC (AAC-LD) object type, bit-sliced arithmetic coding (BSAC) object type, parametric audio coding using harmonic and individual line plus noise and error-resilient (ER) versions of object types. It also defined four new audio profiles: High-Quality Audio Profile, Low Delay Audio Profile, Natural Audio Profile, and Mobile Audio Internetworking Profile.
The HE-AAC Profile (AAC LC with SBR) and AAC Profile (AAC LC) were first standardized in ISO/IEC 14496-3:2001/Amd 1:2003. The HE-AAC v2 Profile (AAC LC with SBR and Parametric Stereo) was first specified in ISO/IEC 14496-3:2005/Amd 2:2006. The Parametric Stereo audio object type used in HE-AAC v2 was first defined in ISO/IEC 14496-3:2001/Amd 2:2004.
The current version of the AAC standard is defined in ISO/IEC 14496-3:2009.
AAC+ v2 is also standardized by ETSI (European Telecommunications Standards Institute) as TS 102005.
The MPEG-4 Part 3 standard also contains other ways of compressing sound. These include lossless compression formats, synthetic audio and low bit-rate compression formats generally used for speech.
Advanced Audio Coding is designed to be the successor of the MPEG-1 Audio Layer 3, known as MP3 format, which was specified by ISO/IEC in 11172-3 (MPEG-1 Audio) and 13818-3 (MPEG-2 Audio).
Blind tests in the late 1990s showed that AAC demonstrated greater sound quality and transparency than MP3 for files coded at the same bit rate.
Overall, the AAC format allows developers more flexibility to design codecs than MP3 does, and corrects many of the 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. This is especially true at very low bit rates where the superior stereo coding, pure MDCT, and better transform window sizes leave MP3 unable to compete.
While the MP3 format has near-universal hardware and software support, primarily because MP3 was the format of choice during the crucial first few years of widespread music file-sharing/distribution over the internet, AAC is a strong contender due to some unwavering industry support.