Advanced Audio Coding

Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. It was developed by Dolby, AT&T, Fraunhofer and Sony, originally as part of the MPEG-2 specification but later improved under MPEG-4. AAC was designed to be the successor of the MP3 format (MPEG-2 Audio Layer III) and generally achieves higher sound quality than MP3 at the same bit rate. AAC encoded audio files are typically packaged in an MP4 container most commonly using the filename extension .m4a.

The basic profile of AAC (both MPEG-4 and MPEG-2) is called AAC-LC (Low Complexity). It is widely supported in the industry and has been adopted as the default or standard audio format on products including Apple's iTunes Store, Nintendo's Wii, DSi and 3DS and Sony's PlayStation 3. It is also further supported on various other devices and software such as iPhone, iPod, PlayStation Portable and Vita, PlayStation 5, Android and older cell phones, digital audio players like Sony Walkman and SanDisk Clip, media players such as VLC, Winamp and Windows Media Player, various in-dash car audio systems, and is used on Spotify, Google Nest, Amazon Alexa.^{[citation needed]} Apple Music, and YouTube web streaming services. AAC has been further extended into HE-AAC (High Efficiency, or AAC+), which improves efficiency over AAC-LC. Another variant is AAC-LD (Low Delay).

AAC supports 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^[which?] 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 384 kbit/s for 5.1 audio. AAC uses only a 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.

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 1992 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. Development further advanced when Lars Liljeryd introduced a method that radically shrank the amount of information needed to store the digitized form of a song or speech.

AAC was developed with cooperation between AT&T Labs, Dolby, Fraunhofer IIS (who developed MP3) and Sony Corporation. AAC 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. Further companies have contributed to development in later years including Bell Labs, LG Electronics, NEC, Nokia, Panasonic, ETRI, JVC Kenwood, Philips, Microsoft, and NTT.

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 nonuniform 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, speech synthesis 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).