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 better sound quality than MP3 at the same bit rate. The confusingly named AAC+ (HE-AAC) does so only at low bit rates and less so at high ones.
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 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 is the default or standard audio format for YouTube, 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.
|Advanced Audio Coding|
|Filename extension||MPEG/3GPP container
|Internet media type|
|Developed by||Bell Labs, Fraunhofer Institute, Dolby Labs, Sony and Nokia|
|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)|
AAC was developed with the cooperation and contributions of companies including AT&T Bell Laboratories, Fraunhofer IIS, Dolby Laboratories, Sony Corporation and Nokia. 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 Dolby 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 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.
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.
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. However, in terms of whether AAC is better than MP3, the advantages of AAC are not entirely decisive, and the MP3 specification, although antiquated, has proven surprisingly robust in spite of considerable flaws. AAC and HE-AAC are better than MP3 at low bit rates (typically less than 128 kilobits per second.) 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-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, Enhanced Low Delay and Enhanced Low Delay v2 (AAC-LD, AAC-ELD, 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 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.)
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.
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. 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. 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. 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 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.
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.
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.
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.
Which encoder provides the best quality? ... the likely answer is: libfdk_aac
The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) 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 (ISO/IEC 14496-3:1999/Amd 1:2000) and in its later revisions.ADIF
ADIF or Adif may refer to:
Audio Data Interchange Format (ADIF), a file format to exchange Advanced Audio Coding (AAC) data; see Advanced Audio Coding § Container formats
Adif (Administrador de Infraestructuras Ferroviarias), the state company responsible for railway infrastructure (track and stations) in Spain
Administración de Infraestructuras Ferroviarias Sociedad del Estado (ADIF), the state company responsible for railway infrastructure in ArgentinaCoding Technologies
Coding Technologies AB was a Swedish technology company that pioneered the use of spectral band replication in Advanced Audio Coding. Its MPEG-2 AAC-derived codec, called aacPlus, was published in 2001 and submitted to the MPEG for standardization. The codec would become the MPEG-4 High-Efficiency AAC (HE-AAC) profile in 2003. XM Satellite Radio used aacPlus for its streams. aacPlus with Parametric stereo, called enhanced aacPlus, would become MPEG-4 HE-AACv2. Coding Technologies was acquired by Dolby Laboratories in 2007 for $250 million in cash.The company was founded in Stockholm, Sweden, in 1997 by Lars Liljeryd. A German subsidiary was formed in 2000 as Coding Technologies GmbH with support from the research organization Fraunhofer IIS. The company also had offices in the United States and China.
Lars Liljeryd, Kristofer Kjörling, and Martin Dietz received the IEEE Masaru Ibuka Consumer Electronics Award in 2013 for their work at Coding Technologies, developing and marketing SBR-based audio coding.Digital television
Digital television (DTV) is the transmission of television signals, including the sound channel, using digital encoding, in contrast to the earlier television technology, analog television, in which the video and audio are carried by analog signals. It is an innovative advance that represents the first significant evolution in television technology since color television in the 1950s. Digital TV transmits in a new image format called high definition television (HDTV), with greater resolution than analog TV, in a wide screen aspect ratio similar to recent movies in contrast to the narrower screen of analog TV. It makes more economical use of scarce radio spectrum space; it can transmit multiple channels, up to 7, in the same bandwidth occupied by a single channel of analog television, and provides many new features that analog television cannot. A transition from analog to digital broadcasting began around 2006. Different digital television broadcasting standards have been adopted in different parts of the world; below are the more widely used standards:
Digital Video Broadcasting (DVB) uses coded orthogonal frequency-division multiplexing (OFDM) modulation and supports hierarchical transmission. This standard has been adopted in Europe, Africa, Asia, Australia, total about 60 countries.
Advanced Television System Committee (ATSC) uses eight-level vestigial sideband (8VSB) for terrestrial broadcasting. This standard has been adopted by 6 countries: United States, Canada, Mexico, South Korea, Dominican Republic and Honduras.
Integrated Services Digital Broadcasting (ISDB) is a system designed to provide good reception to fixed receivers and also portable or mobile receivers. It utilizes OFDM and two-dimensional interleaving. It supports hierarchical transmission of up to three layers and uses MPEG-2 video and Advanced Audio Coding. This standard has been adopted in Japan and the Philippines. ISDB-T International is an adaptation of this standard using H.264/MPEG-4 AVC that been adopted in most of South America and is also being embraced by Portuguese-speaking African countries.
Digital Terrestrial Multimedia Broadcasting (DTMB) adopts time-domain synchronous (TDS) OFDM technology with a pseudo-random signal frame to serve as the guard interval (GI) of the OFDM block and the training symbol. The DTMB standard has been adopted in the People's Republic of China, including Hong Kong and Macau.
Digital Multimedia Broadcasting (DMB) is a digital radio transmission technology developed in South Korea as part of the national IT project for sending multimedia such as TV, radio and datacasting to mobile devices such as mobile phones, laptops and GPS navigation systems.FAAC
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 (LC, Main, LTP for encoding and SBR, PS, ER, LD for decoding), file formats (ADTS AAC, raw AAC, MP4), 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.FAAC and FAAD2, being distributed in C source code form, can be compiled on various platforms and are distributed free of charge. FAAD2 is free software. FAAC contains some code which is published as Free Software, but as a whole it is only distributed under a proprietary license.
FAAC was originally written by Menno Bakker.Fraunhofer FDK AAC
Fraunhofer FDK AAC (Full title Fraunhofer FDK AAC Codec Library for Android) is an open-source software library for encoding and decoding Advanced Audio Coding (AAC) format audio, developed by Fraunhofer IIS, and included as part of Android. It supports several Audio Object Types including MPEG-2 and MPEG-4 AAC LC, HE-AAC (AAC LC + SBR), HE-AACv2 (LC + SBR + PS) as well AAC-LD (low delay) and AAC-ELD (enhanced low delay) for real-time communication. The encoding library supports sample rates up to 96 kHz and up to eight channels (7.1 surround).The Android-targeted implementation of the Fraunhofer AAC encoder uses fixed-point math and is optimized for low-delay encoding on embedded devices/mobile phones. The library is currently limited to 16-bit PCM input. Other versions of the Fraunhofer encoder, like the one included in Winamp, are optimized for encoding music on desktop-class processors. Those versions of the encoder, however, are not open source and require a commercial license. The license included by Fraunhofer in the FDK library source code allows redistribution in source or binary forms, but does not license patented technologies described by the code. The license states that the library may only be distributed as authorized by patent licenses. Due to this restriction, along with a limitation on charging for the library, Debian considers it non-free. It was classified as free by Fedora after a review by the legal department at Red Hat. The FSF also considers it to be free, though discourages its use due to the explicit lack of a patent grant.The FDK AAC encoder employs a more aggressive default low-pass filter than is used in other codecs. Higher frequencies are removed so that more bits are available to better describe sounds of lower frequencies, improving the overall quality for most combinations of recordings and listeners. In some, not completely rare, combinations the missing high frequencies are noticeable. The library allows overriding the low-pass filter setting, and in the highest VBR mode effectively applies no filter at all.A cross-platform source distribution is maintained by Martin Storsjö as part of the opencore-amr project under the name fdk-aac. The code compiles into a shared library, libfdk-aac. The media frameworks FFmpeg and Libav support audio encoding through libfdk-aac.High-Efficiency Advanced Audio Coding
Not to be confused with MPEG-4 SLS, which is branded HD-AAC.
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. HE-AAC version 1 profile (HE-AAC v1) uses spectral band replication (SBR) to enhance the compression efficiency in the frequency domain. HE-AAC version 2 profile (HE-AAC v2) couples SBR with Parametric Stereo (PS) to enhance the compression efficiency of stereo signals. It is a standardized and improved version of the AACplus codec.
HE-AAC is used in digital radio standards like DAB+ and Digital Radio Mondiale.MP3 blog
An MP3 blog is a type of blog in which the creator makes music files, normally in the MP3 format, available for download. They are also known as musicblogs, audioblogs or soundblogs (the latter two can also mean podcasts). MP3 blogs have become increasingly popular since 2003. The music posted ranges from hard-to-find rarities that have not been issued in many years to more contemporary offerings, and selections are often restricted to a particular musical genre or theme. Some musicblogs offer music in Advanced Audio Coding (AAC) or Ogg formats.MP3 player
An MP3 player or Digital Audio Player is an electronic device that can play digital audio files. It is a type of Portable Media Player. The term 'MP3 player' is a misnomer, as most players play more than the MP3 file format.
Since the MP3 format is widely used, almost all players can play that format. In addition, there are many other digital audio formats. Some formats are proprietary, such as Windows Media Audio (WMA), and Advanced Audio Coding (AAC). Some of these formats also may incorporate digital rights management (DRM), such as WMA DRM, which are often part of paid download sites. Other formats are patent-free or otherwise open, such as MP3, Vorbis, FLAC, and Speex (the latter three part of the Ogg open multimedia project).MPEG-2
MPEG-2 (a.k.a. H.222/H.262 as defined by the ITU) is a standard for "the generic coding of moving pictures and associated audio information". It describes a combination of lossy video compression and lossy audio data compression methods, which permit storage and transmission of movies using currently available storage media and transmission bandwidth. While MPEG-2 is not as efficient as newer standards such as H.264/AVC and H.265/HEVC, backwards compatibility with existing hardware and software means it is still widely used, for example in over-the-air digital television broadcasting and in the DVD-Video standard.MPEG-2 Part 3
Part 3 of the MPEG-2 standard (formally known as ISO/IEC 13818-3, also known as MPEG-2 Audio or MPEG-2 BC) defines audio coding:
MPEG Multichannel - It enhances MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel. This method is backwards-compatible (also known as MPEG-2 BC), allowing MPEG-1 audio decoders to decode the two main stereo components of the presentation.
MPEG-2 Part 3 also defined additional bit rates and sample rates for MPEG-1 Audio Layer I, MPEG-1 Audio Layer II and MPEG-1 Audio Layer III (a.k.a. MP3).The MPEG-2 Part 3 should not be confused with MPEG-2 Part 7: AAC a.k.a. MPEG-2 NBC (Non-Backward Compatible) - the MPEG-2 Advanced Audio Coding with support for multichannel encoding (up to 48 channels).MPEG-4 Part 3
MPEG-4 Part 3 or MPEG-4 Audio (formally ISO/IEC 14496-3) 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.The MPEG-4 Part 3 consists of a variety of audio coding technologies – from lossy speech coding (HVXC, CELP), general audio coding (AAC, TwinVQ, BSAC), lossless audio compression (MPEG-4 SLS, Audio Lossless Coding, MPEG-4 DST), a Text-To-Speech Interface (TTSI), Structured Audio (using SAOL, SASL, MIDI) and many additional audio synthesis and coding techniques.MPEG-4 Audio does not target a single application such as real-time telephony or high-quality audio compression. It applies to every application which requires the use of advanced sound compression, synthesis, manipulation, or playback.
MPEG-4 Audio is a new type of audio standard that integrates numerous different types of audio coding: natural sound and synthetic sound, low bitrate delivery and high-quality delivery, speech and music, complex soundtracks and simple ones, traditional content and interactive content.Marina Bosi
Marina Bosi is a Consulting Professor at Stanford University's Center for Computer Research in Music and Acoustics (CCRMA). Originally a flutist and flute teacher, she is known for her work on digital audio coding formats.MediaCoder
MediaCoder is a proprietary transcoding program for Microsoft Windows, developed by Stanley Huang since 2005.Mora (music store)
mora (モーラ, mōra) is an online music and video store for the Japanese market. It is integrated into the Japanese version of Sony's SonicStage software and is now the official store for their Walkman devices. Up until October 1, 2012, music purchased from mora was exclusively in Sony's proprietary ATRAC3 format with OpenMG DRM. A partner store called
mora win (モーラ ウィン, mōra win) was also in operation, using Windows Media Audio codec and Windows Media DRM encryption, and integrated into the Japanese version of Microsoft's Windows Media Player 11 as the "recommended store" in Japan.On October 1, 2012, mora underwent a rehaul into a DRM free store, selling songs and music videos in Advanced Audio Coding formats. The mora win service was also eliminated in the restructuring, with the DRM free store being integrated into the Media Go service in Windows 8. One year later on October 17, 2013, mora began offering songs and albums as 24bit/96kHz FLAC files.mora is operated by Label Gate Co., Ltd., a joint venture of 17 Japanese record companies including Sony Music Entertainment Japan, Avex Group, and Universal Music Japan.Nero AAC Codec
Nero AAC Codec is a set of software tools for encoding and decoding Advanced Audio Coding (AAC) format audio, and editing MPEG-4 metadata. It was developed and distributed by Nero AG, and is available at no cost for Windows and Linux for non-commercial use. The codec was originally part of Nero Digital, but was later released as a stand-alone package.
Nero's AAC encoder has been very competitive when tested against other encoders in scientific listening tests, for a time, second only to Apple's AAC encoder.In 2006, Chip Magazine (Germany) found that AAC files encoded with the Nero AAC encoder would consume as little as half of the space on a portable music player when compared to MP3 files of similar audio quality.Parametric Stereo
Parametric Stereo (PS) is lossy audio compression algorithm and a feature and an Audio Object Type (AOT) defined and used in MPEG-4 Part 3 (MPEG-4 Audio) to further enhance efficiency in low bandwidth stereo media. Advanced Audio Coding Low Complexity (AAC LC) combined with Spectral Band Replication (SBR) and Parametric Stereo (PS) was defined as HE-AAC v2. An HE-AAC v1 decoder will only give mono sound when decoding an HE-AAC v2 bitstream. Parametric Stereo performs sparse coding in the spatial domain, somewhat similar to what SBR does in the frequency domain.SHOUTcast
SHOUTcast DNAS is cross-platform proprietary software for streaming media over the Internet. The software, developed by Nullsoft, is available free of charge. It allows digital audio content, primarily in MP3 or High-Efficiency Advanced Audio Coding format, to be broadcast to and from media player software, enabling the creation of Internet radio "stations".
The most common use of SHOUTcast is for creating or listening to Internet audio broadcasts; however, video streams exist as well. Some traditional radio stations use SHOUTcast to extend their presence onto the Web.
SHOUTcast Radio is a related website which provides a directory of SHOUTcast stations.Samsung A767 Propel
The Samsung SGH-A767, more commonly known as the Samsung Propel, is a mobile phone by Samsung Telecommunications. It features a full QWERTY keyboard that slides out from under the phone. It comes in white, blue, red, and green, and is one of AT&T's most popular cell phones. It was designed as a quick texting phone along with the Pantech Matrix, the Pantech Slate, and the UT Starcom Quickfire.
The Samsung SGH-A767's QWERTY keyboard makes texting and any other task that requires using letters, numbers, and symbols, much easier and faster to complete. However, the size of the keyboard can be a drawback for people with large hands because the keys are flush mounted and close to each other. Another downside is the screen resolution, which only has 65,000 colors. Along with the standard features of most phones, the Samsung SGH-A767 has GPS capability, with both audible and visual aide which is Java based. It is, however, a separately licensed product, requiring both downloading the program and paying extra for data services.Music and other multimedia capabilities are included with the Samsung SGH-A767. The microSD slot allows you to transfer music and pictures from a computer to your phone, while adding extra memory. The AT&T Music feature allows you to download and play music and ringtones. Recorded audio and playlists can also be played. Supported audio file extensions are AAC (Advanced Audio Coding), AAC+, eAAC+, MP3, WMA (Windows Media Audio), 3GP, MPEG, MP4, and M4A.
See Compression methods for techniques and Compression software for codecs
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