|Filename extension||MPEG/3GPP container|
|Internet media type|
|Developed by||Bell, Fraunhofer, Dolby, Sony, Nokia, LG Electronics, NEC, NTT Docomo, Panasonic|
|Type of format||Audio compression format, lossy compression|
|Contained by||MPEG-4 Part 14, 3GP and 3G2, ISO base media file format and Audio Data Interchange Format (ADIF)|
|Standard|| ISO/IEC 13818-7,|
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 is adopted into digital radio standards DAB+ and Digital Radio Mondiale, and mobile television standards DVB-H and ATSC-M/H.
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 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.[ citation needed ] 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.
AAC is the default or standard audio format for iPhone, iPod, iPad, Nintendo DSi, Nintendo 3DS, Apple Music, iTunes, DivX Plus Web Player, PlayStation 4 and various Nokia Series 40 phones. It is supported on PlayStation Vita, Wii, Sony Walkman MP3 series and later, Android and BlackBerry. AAC is also supported by manufacturers of in-dash car audio systems.[ when? ][ vague ] It is also one of the audio formats used on Spotify.
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 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, 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.
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:
The actual encoding process consists of the following steps:
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 bit rate speech coding to high-quality audio coding and music synthesis.
AAC encoders can switch dynamically between a single MDCT block of length 1024 points or 8 blocks of 128 points (or between 960 points and 120 points, respectively).
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 MPEG-2 Part 7 standard (Advanced Audio Coding) was first published in 1997 and offers three default profiles:
The MPEG-4 Part 3 standard (MPEG-4 Audio) defined various new compression tools (a.k.a. Audio Object Types) and their usage in brand new profiles. AAC is not used in some of the MPEG-4 Audio profiles. The MPEG-2 Part 7 AAC LC profile, AAC Main profile and AAC SSR profile are combined with Perceptual Noise Substitution and defined in the MPEG-4 Audio standard as Audio Object Types (under the name AAC LC, AAC Main and AAC SSR). These are combined with other Object Types in MPEG-4 Audio profiles.Here is a list of some audio profiles defined in the MPEG-4 standard:
One of many improvements in MPEG-4 Audio is an Object Type called Long Term Prediction (LTP), which is an improvement of the Main profile using a forward predictor with lower computational complexity.
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.
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
The audio coding standards MPEG-4 Low Delay (AAC-LD), Enhanced Low Delay (AAC-ELD), and Enhanced Low Delay v2 (AAC-ELDv2) as defined in ISO/IEC 14496-3:2009 and ISO/IEC 14496-3:2009/Amd 3 are designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. They are closely derived from the MPEG-2 Advanced Audio Coding (AAC) format.AAC-ELD is recommended by GSMA as super-wideband voice codec in the IMS Profile for High Definition Video Conference (HDVC) Service.
No licenses or payments are required for a user to stream or distribute content in AAC format.This reason alone might have made AAC a more attractive format to distribute content than its predecessor MP3, particularly for streaming content (such as Internet radio) depending on the use case.
However, a patent license is[ when? ] required for all manufacturers or developers of AAC codecs. For this reason, free and open source software implementations such as FFmpeg and FAAC may be distributed in source form only, in order to avoid patent infringement. (See below under Products that support AAC, Software.)
The AAC patent holders include Bell Labs, Dolby, Fraunhofer, LG Electronics, NEC, NTT Docomo, Panasonic, Sony Corporation,ETRI, JVC Kenwood, Philips, Microsoft, and NTT.
Some extensions have been added to the first AAC standard (defined in MPEG-2 Part 7 in 1997):
In addition to the MP4, 3GP and other container formats based on ISO base media file format for file storage, AAC audio data was first packaged in a file for the MPEG-2 standard using Audio Data Interchange Format (ADIF),consisting of a single header followed by the raw AAC audio data blocks. However, if the data is to be streamed within an MPEG-2 transport stream, a self-synchronizing format called an Audio Data Transport Stream (ADTS) is used, consisting of a series of frames, each frame having a header followed by the AAC audio data. This file and streaming-based format 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. These containers, as well as a raw AAC stream, may bear the .aac file extension. MPEG-4 Part 3 also defines its own self-synchronizing format called a Low Overhead Audio Stream (LOAS) that encapsulates not only AAC, but any MPEG-4 audio compression scheme such as TwinVQ and ALS. This format is what was defined for use in DVB transport streams when encoders use either SBR or parametric stereo AAC extensions. However, it is restricted to only a single non-multiplexed AAC stream. This format is also referred to as a Low Overhead Audio Transport Multiplex (LATM), which is just an interleaved multiple stream version of a LOAS.
This section needs additional citations for verification .(September 2017)
In December 2003, Japan started broadcasting terrestrial DTV ISDB-T standard that implements MPEG-2 video and MPEG-2 AAC audio. In April 2006 Japan started broadcasting the ISDB-T mobile sub-program, called 1seg, that was the first implementation of video H.264/AVC with audio HE-AAC in Terrestrial HDTV broadcasting service on the planet.
In December 2007, Brazil started broadcasting terrestrial DTV standard called International ISDB-Tb that implements video coding H.264/AVC with audio AAC-LC on main program (single or multi) and video H.264/AVC with audio HE-AACv2 in the 1seg mobile sub-program.
The ETSI, the standards governing body for the DVB suite, supports AAC, HE-AAC and HE-AAC v2 audio coding in DVB applications since at least 2004. [ citation needed ]DVB broadcasts which use the H.264 compression for video normally use HE-AAC for audio.
In April 2003, Apple 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 closed-source 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.[ citation needed ] As of January 6, 2009, nearly all music on the USA regioned iTunes Store became DRM-free, with the remainder becoming DRM-free by the end of March 2009.
iTunes supports a "Variable Bit Rate" (VBR) encoding option which encodes AAC tracks in an "Average Bit Rate" (ABR) scheme.[ citation needed ] As of September 2009, Apple has added support for HE-AAC (which is fully part of the MP4 standard) only for radio streams, not file playback, and iTunes still lacks support for true VBR encoding. The underlying QuickTime API does offer a true VBR encoding profile however.
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[ citation needed ] 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.
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 used this way with the corresponding filters to enable AAC playback in any DirectShow based player. Mac OS X supports AAC via the QuickTime libraries.
Adobe Flash Player, since version 9 update 3, can also play back AAC streams.Since Flash Player is also a browser plugin, it can play AAC files through a browser as well.
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.
The following is a non-comprehensive list of other software player applications:
Some of these players (e.g., foobar2000, Winamp, and VLC) also support the decoding of ADTS (Audio Data Transport Stream) using the SHOUTcast protocol. Plug-ins for Winamp and foobar2000 enable the creation of such streams.
In May 2006, Nero AG released an AAC encoding tool free of charge, Nero Digital Audio (the AAC codec portion has become Nero AAC Codec),which is capable of encoding LC-AAC, HE-AAC and HE-AAC v2 streams. The tool is a Command Line Interface tool only. A separate utility is also included to decode to PCM WAV.
Various tools including the foobar2000 audio player and MediaCoder can provide a GUI for this encoder.
FAAC and FAAD2 stand for Freeware Advanced Audio Coder and Decoder 2 respectively. FAAC supports audio object types LC, Main and LTP.FAAD2 supports audio object types LC, Main, LTP, SBR and PS. Although FAAD2 is free software, FAAC is not free software.
A Fraunhofer-authored open-source encoder/decoder included in Android has been ported to other platforms. It is the recommended AAC encoder of FFmpeg.[ citation needed ]
The native AAC encoder created in FFmpeg's libavcodec, and forked with Libav, was considered experimental and poor. A significant amount of work was done for the 3.0 release of FFmpeg (February 2016) to make its version usable and competitive with the rest of the AAC encoders.Libav has not merged this work and continues to use the older version of the AAC encoder. These encoders are LGPL-licensed open-source and can be built for any platform that the FFmpeg or Libav frameworks can be built.
Both FFmpeg and Libav can use the Fraunhofer FDK AAC library via libfdk-aac, and while the FFmpeg native encoder has become stable and good enough for common use, FDK is still considered the highest quality encoder available for use with FFmpeg.Libav also recommends using FDK AAC if it is available.
MP3 is a coding format for digital audio developed largely by the Fraunhofer Society in Germany, with support from other digital scientists in the US and elsewhere. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended — defining additional bit-rates and support for more audio channels — as the third audio format of the subsequent MPEG-2 standard. A third version, known as MPEG 2.5 — extended to better support lower bit rates — is commonly implemented, but is not a recognized standard.
MPEG-4 is a method of defining compression of audio and visual (AV) digital data. It was introduced in late 1998 and designated a standard for a group of audio and video coding formats and related technology agreed upon by the ISO/IEC Moving Picture Experts Group (MPEG) under the formal standard ISO/IEC 14496 – Coding of audio-visual objects. Uses of MPEG-4 include compression of AV data for Internet video and CD distribution, voice and broadcast television applications. The MPEG-4 standard was developed by a group led by Touradj Ebrahimi and Fernando Pereira.
MPEG-4 Part 3 or MPEG-4 Audio is the third part of the ISO/IEC MPEG-4 international standard developed by Moving Picture Experts Group. It specifies audio coding methods. The first version of ISO/IEC 14496-3 was published in 1999.
3GP is a multimedia container format defined by the Third Generation Partnership Project (3GPP) for 3G UMTS multimedia services. It is used on 3G mobile phones but can also be played on some 2G and 4G phones.
Harmonic Vector Excitation Coding, abbreviated as HVXC is a speech coding algorithm specified in MPEG-4 Part 3 standard for very low bit rate speech coding. HVXC supports bit rates of 2 and 4 kbit/s in the fixed and variable bit rate mode and sampling frequency 8 kHz. It also operates at lower bitrates, such as 1.2 - 1.7 kbit/s, using a variable bit rate technique. The total algorithmic delay for the encoder and decoder is 36 ms.
High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio defined as an MPEG-4 Audio profile in ISO/IEC 14496-3. It is an extension of Low Complexity AAC (AAC-LC) optimized for low-bitrate applications such as streaming audio. The usage profile HE-AAC v1 uses spectral band replication (SBR) to enhance the modified discrete cosine transform (MDCT) compression efficiency in the frequency domain. The usage profile HE-AAC v2 couples SBR with Parametric Stereo (PS) to further enhance the compression efficiency of stereo signals.
TwinVQ is an audio compression technique developed by Nippon Telegraph and Telephone Corporation (NTT) Human Interface Laboratories in 1994. The compression technique has been used in both standardized and proprietary designs.
MPEG-4 Part 2, MPEG-4 Visual is a video compression format developed by the Moving Picture Experts Group (MPEG). It belongs to the MPEG-4 ISO/IEC standards. It is a discrete cosine transform (DCT) compression standard, similar to previous standards such as MPEG-1 Part 2 and H.262/MPEG-2 Part 2.
FAAC or Freeware Advanced Audio Coder is a software project which includes the AAC encoder FAAC and decoder FAAD2. It supports MPEG-2 AAC as well as MPEG-4 AAC. It supports several MPEG-4 Audio object types, file formats, multichannel and gapless encoding/decoding and MP4 metadata tags. The encoder and decoder is compatible with standard-compliant audio applications using one or more of these object types and facilities. It also supports Digital Radio Mondiale.
QuickTime File Format (QTFF) is a computer file format used natively by the QuickTime framework.
MPEG-4 Audio Lossless Coding, also known as MPEG-4 ALS, is an extension to the MPEG-4 Part 3 audio standard to allow lossless audio compression. The extension was finalized in December 2005 and published as ISO/IEC 14496-3:2005/Amd 2:2006 in 2006. The latest description of MPEG-4 ALS was published as subpart 11 of the MPEG-4 Audio standard in August 2009.
MPEG-4 SLS, or MPEG-4 Scalable to Lossless as per ISO/IEC 14496-3:2005/Amd 3:2006 (Scalable Lossless Coding), is an extension to the MPEG-4 Part 3 (MPEG-4 Audio) standard to allow lossless audio compression scalable to lossy MPEG-4 General Audio coding methods (e.g., variations of AAC). It was developed jointly by the Institute for Infocomm Research (I2R) and Fraunhofer, which commercializes its implementation of a limited subset of the standard under the name of HD-AAC. Standardization of the HD-AAC profile for MPEG-4 Audio is under development (as of September 2009).
MPEG-4 Part 14 or MP4 is a digital multimedia container format most commonly used to store video and audio, but it can also be used to store other data such as subtitles and still images. Like most modern container formats, it allows streaming over the Internet. The only filename extension for MPEG-4 Part 14 files as defined by the specification is .mp4. MPEG-4 Part 14 is a standard specified as a part of MPEG-4.
The MPEG-4 Low Delay Audio Coder is audio compression standard 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) standard. It was published in MPEG-4 Audio Version 2 and in its later revisions.
MPEG Surround, also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo audio services to multi-channel audio in a backwards compatible fashion. The total bit rates used for the core and the MPEG Surround data are typically only slightly higher than the bit rates used for coding of the core. MPEG Surround adds a side-information stream to the core bit stream, containing spatial image data. Legacy stereo playback systems will ignore this side-information while players supporting MPEG Surround decoding will output the reconstructed multi-channel audio.
G.719 is an ITU-T standard audio coding format providing high quality, moderate bit rate wideband audio coding at low computational load. It was produced through a collaboration between Polycom and Ericsson.
ISO/IEC base media file format defines a general structure for time-based multimedia files such as video and audio. The identical text is published as ISO/IEC 15444-12.
Unified Speech and Audio Coding (USAC) is an audio compression format and codec for both music and speech or any mix of speech and audio using very low bit rates between 12 and 64 kbit/s. It was developed by Moving Picture Experts Group (MPEG) and was published as an international standard ISO/IEC 23003-3 and also as an MPEG-4 Audio Object Type in ISO/IEC 14496-3:2009/Amd 3 in 2012.
An audio coding format is a content representation format for storage or transmission of digital audio. Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.
|journal=(help) Presented at the 115th Convention of the Audio Engineering Society, 10–13 October 2003.
Which encoder provides the best quality? ... the likely answer is: libfdk_aac