Advanced Audio Coding

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Advanced Audio Coding (AAC), also known as MPEG-2 Part 7, and also MPEG-4 Part 3 in a slightly modified form, is a digital audio encoding and lossy compression format. AAC was declared an international standard by the MPEG group by the end of April 1997. It was developed with contributions by Dolby, Fraunhofer (FhG), AT&T, Sony and Nokia.

AAC was designed as an improved-performance codec relative to MP3 (which was specified in MPEG-1) and MPEG-2 Part 3 (which is also known as "MPEG-2 Audio" or ISO/IEC 13818-3).

AAC was promoted as the successor to MP3 for audio coding at medium to high bit rates, though has yet to overtake the MP3 format or Microsoft's rival WMA format in terms of popularity or availability within the portable audio devices market.

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 convey high-quality digital audio.

1. Signal components that are perceptually irrelevant are discarded.
2. Redundancies in the coded audio signal are eliminated.
3. The signal is processed by a modified discrete cosine transform (MDCT) according to its complexity.
4. Internal error correction codes are added.
5. The signal is stored or transmitted.

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

• The MPEG-4 audio coding algorithm family spans the range from low-bit-rate 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 2,048 points. AAC is much more capable of encoding audio with streams of complex pulses and square waves than MP3 or Musicam.

AAC can be switched dynamically between MDCT block lengths of 2,048 points to 256 points.

• If a single change or transient occurs, the short window of 256 points is chosen for better resolution.
• By default the longer 2,048-point window is used to improve the coding efficiency because of better frequency resolution.

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 use for a particular application. The standard offers four default profiles:

• Low Complexity Profile (LC) - the simplest and most widely used and supported.
• Main Profile (MAIN), which expands upon LC with backwards prediction.
• Sample-Rate Scalable (SRS) is also called Scalable Sample Rate (MPEG-4 AAC-SSR) in line with the standard nomenclature.
• Long Term Prediction (LTP), added in MPEG-4, 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 perceptional quality as 128 kbit/s MP3.

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. The codec is closely derived from MPEG-2 Advanced Audio Coding (AAC).

• The most stringent requirements are a maximum algorithmic delay of only 20 ms and a good audio quality for all kind of audio signals including speech and music.
• In this way, the AAC LD coding scheme bridges the gap between speech coding schemes and high quality audio coding schemes.

Two-way communication with AAC LD is possible on usual analog telephone lines and via ISDN connections. Compared to known speech coders, the codec is capable of coding both music and speech signals with good quality. Unlike speech coders, however, the achieved coding quality scales up with bitrate. Transparent quality can be achieved.

AAC LD can also process stereo signals by using the advanced stereo coding tools of AAC. Thus it is possible to transmit a stereo signal with a bandwidth of 7 kHz via one ISDN line or with a bandwidth of 15 kHz via two ISDN lines.

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.

But 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. This provides the error correcting capability just the most sensitive parts of the payload in order to keep the additional overhead low.

Error Resilient AAC
Error resilience 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's improvements over MP3

Some of its advances:

• Sample frequencies from 8 kHz to 96 kHz (official MP3: 16 kHz to 48 kHz)
• Up to 48 channels
• Higher efficiency and simpler filterbank (hybrid → pure MDCT)
• Higher coding efficiency for stationary signals (blocksize: 576 → 1024 samples)
• Higher coding efficiency for transient signals (blocksize: 192 → 128 samples)
• Much better handling of frequencies above 16 kHz
• More flexible joint stereo (separate for every scale band)

The result is a specification that allows developers more flexibility to design codecs that offer efficient compression compared to MP3. However, the advantages are not entirely decisive, and the MP3 specification, while outdated, has proven surprisingly robust. Although AAC and HE-AAC completely dominate MP3 at very low bitrates, at medium to higher bitrates the two formats are more comparable. In the future as developers learn to better exploit the AAC format, AAC is expected to gain additional ground and perhaps overtake MP3.

AAC ISO standard

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

Products that support AAC

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), and that customers could download popular songs in a copyright-protected form (see FairPlay) via the iTunes Music Store.

Apple has added support for VBR encoding of AAC tracks in iTunes v5.0. They have also added certain enhancements in higher-end iPods such as chapters (bookmarks that can incorporate web links and pictures set to appear at certain times during playback of audio books and podcasts) which are not features of AAC itself, but of the Apple proprietary file-format that wraps the AAC bitstream.

Nero Digital Audio

In May 2006, Nero AG released a free AAC encoding tool entitled "Nero Digital Audio", which is capable of encoding LC, HE and HEv2 AAC streams. The tool is a Command Line Interface tool only, and a separate decoder utility is included to convert the files to .wav.

Other Media Players

Almost all current computer media players include built in decoders for AAC, or can utilize DirectShow filters to decode it. Programs of particular note include:

• Winamp, which includes an AAC encoder that supports LC and HE encoding.
• The decoder FAAD (Free AAC Decoder), which is used in many players to decode all forms of AAC audio.
• mplayer or its components are often used as AAC decoders on linux

Portable Devices

For a number of years, many mobile (cell) phones from the big manufacturers such as Nokia, Motorola, Samsung and Sony Ericsson have supported AAC playback. During 2005, the buzz around music on mobile phones increased dramatically. Many manufacturers announced dedicated music phones, such as the Sony Ericsson S700i, Sony Ericsson W600/Sony Ericsson W550, Sony Ericsson K750i/Sony Ericsson W800, Sony Ericsson W810, Sony Ericsson W900i, Nokia N91, Nokia 3250, Nokia 3300, Nokia N70, Nokia 6270, Samsung SGH-i300, Motorola ROKR E1, Motorola RAZR V3i, Motorola RAZR V3x, Motorola SLVR L7, Siemens M75, Siemens CX75, Siemens EL71 - all with AAC playback as standard. This trend towards supporting AAC continues with the ever increasing number of advanced phones on the market today, with most high-end phone models capable of AAC playback. (Caveat Emptor Note: Although some of these portable devices may be able to play back AAC, many do not have support for the Apple AAC wrapper. Therefore they fail to read Title/Artist metadata embedded in AAC files encoded with Apple's iTunes product and will not display that information in their media players. The Sony Ericsson Walkman phones do not, for example.)

Also, the PlayStation Portable has had support for AAC files since the version 2.0 firmware update (released August 2005), but with a filename extension of .MP4 only. Other extensions, such as iTunes .M4A could be played by simply renaming the files to .MP4. This restriction was removed with the 2.7 firmware update (released April 2006). Other Sony products, including the A and E series network Walkmans, support AAC with firmware updates (released May 2006).

Epson supports AAC playback in the P-2000 and P-4000 Multimedia / Photo Storage Viewers. This support is not available in their older models, however.

Extensions & improvements

Some technology extensions have been added to the AAC standard

• MPEG-4 SLS- Scalable to lossless audio standard
• High Efficiency AAC (HE-AAC) - SBR technology has been applied to AAC, and was incorporated into the standard to form High Efficiency AAC v1; see MPEG-4 Part 3 for further details.
• aacPlus v2 or Enhanced aacPlus is another recent introduction to AAC from the use of Parametric Stereo . These technologies significantly improve the performance of AAC at lower bitrates, and are used, for example, in Digital Radio Mondiale.
• The codec design was further improved in MPEG-4 Part 3, known formally as ISO/IEC 14496-3, with the addition of Perceptual Noise Substitution (PNS) and a Long Term Predictor (LTP).

Although the AAC codec specified in MPEG-2 Part 7 and the AAC specified in MPEG-4 Part 3 are somewhat different, they are both informally known as AAC.

See also

• Audio data compression
• The *.m4a (MPEG-4 audio) file extension
• Comparison of audio codecs

External links

• [Apple's page on MPEG-4 AAC]
• [EE Times article on AAC]
• [Fraunhofer MPEG-2 AAC Information]
• [International Organization for Standardization]
• [AAC Licensing]
• [Open Source AAC codec] FAAC (encoder) and FAAD2 (decoder)
• [Roberto's public listening tests] - blind, controlled listening tests of lossy compression formats including AAC
• [List of AAC and aacPlus resources]

 

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