Mpeg 4 Part 3

Information about Mpeg 4 Part 3

MPEG-4 is a suite of standards which has many "parts", where each part standardizes various entities related to multimedia, such as audio, video, and file formats. To know more about various parts and what they mean, please see the entry for MPEG-4.

MPEG-4 Part 3 (formally ISO/IEC 14496-3) is the third part of the ISO/IEC MPEG-4 international standard. It specifies audio coding methods.

Bifurcation in the AAC technical standard

The Advanced Audio Coding in MPEG-4 Part 3 was enhanced relative to the previous standard MPEG-2 Part 7, in order to provide better sound quality for a given encoding bitrate.

It is assumed that any Part 3 and Part 7 differences will be ironed out by the ISO standards body in the near future to avoid the possibility of future bitstream incompatibilities. At present there are no known player or codec incompatibilities due to the newness of the standard.

AAC's multiple codecs:

Low Complexity Advanced Audio Coding (LC-AAC)
High-Efficiency Advanced Audio Coding (HE-AAC)
Scalable Sample Rate Advanced Audio Coding (AAC-SSR)
Bit Sliced Arithmetic Coding (BSAC)
Long Term Predictor (LTP)

HE-AAC

HE-AAC is an extension of AAC using Spectral Band Replication (SBR), and Parametric Stereo (PS). It is designed to increase coding efficiency at low bitrates by using partial parametric representation of audio.

AAC-SSR

AAC Scalable Sample Rate was introduced by Sony to the MPEG-4 standard. The audio signal is first split into 4 bands using a 4 band polyphase quadrature filter bank. Then these 4 bands are further split using MDCTs with a size k of 32 or 256 samples. This is similar to normal MPEG-4 AAC which uses MDCTs with a size k of 128 or 1024 directly on the audio signal.

The advantage of this technique is that short block switching can be done separately for every PQF band. So high frequencies can be encoded using a short block to enhance temporal resolution, low frequencies can be still encoded with high spectral resolution. However, due to aliasing between the 4 PQF bands coding efficiencies around (1,2,3) * fs/8 is worse than normal MPEG-4 AAC.

MPEG-4 AAC-SSR is very similar to ATRAC and ATRAC-3.

Why AAC-SSR was introduced

The idea behind AAC-SSR was not only the advantage listed above, but also the possibility of reducing the data rate by removing 1, 2 or 3 of the upper PQF bands. A very simple bitstream splitter can remove these bands and thus reduce the bitrate and sample rate.

Example:

4 subbands: bitrate = 128 kbit/s, sample rate = 48 kHz, f_lowpass = 20 kHz
3 subbands: bitrate ~ 120 kbit/s, sample rate = 48 kHz, f_lowpass = 18 kHz
2 subbands: bitrate ~ 100 kbit/s, sample rate = 24 kHz, f_lowpass = 12 kHz
1 subband: bitrate ~ 65 kbit/s, sample rate = 12 kHz, f_lowpass = 6 kHz

Note: although possible, the resulting quality is much worse than typical for this bitrate. So for normal 64 kbit/s AAC a bandwidth of 14-16 kHz is achieved by using intensity stereo and reduced NMRs. This degrades audible quality less than transmitting 6 kHz bandwidth with perfect quality.

BSAC

Bit Sliced Arithmetic Coding is an MPEG-4 standard (ISO/IEC 14496-3 subpart 4) for scalable audio coding. BSAC uses an alternative noiseless coding to AAC, with the rest of the processing being identical to AAC. This support for scalability allows for nearly transparent sound quality at 64 kbit/s and graceful degradation at lower bit rates. BSAC coding is best performed in the range of 40 kbit/s to 64 kbit/s, though it operates in the range of 16 kbit/s to 64 kbit/s. The AAC-BSAC codec is used in Digital Multimedia Broadcasting (DMB) applications.

External links

Apple: MPEG-4: AAC
EBU subjective listening tests on low-bitrate audio codecs
AAC radio stations - Online radio stations in AAC format
Tuner2 - Directory of radio stations in AAC+ format at various bitrates
RadioFeeds UK & Ireland - Page containing plenty of terrestrial stations webcasting in AAC+ format.
http://www.rjamorim.com/test/64test/results.html A page comparing codecs including He-AAC @64 kbit/s by listening tests.

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Multimedia compression formats

Video compression formats

ISO/IEC

MJPEG MPEG-1 MPEG-2 MPEG-4 ASP MPEG-4/AVC

ITU-T

H.261 H.262 H.263 H.264

Others

AVS Bink Dirac Indeo MJPEG RealVideo Theora VC-1 VP6 VP7 WMV

Audio compression formats

ISO/IEC

MPEG-1 Layer III (MP3) MPEG-1 Layer II MPEG-1 Layer I AAC HE-AAC HE-AAC v2 aacPlus v2

ITU-T

G.711 G.722 G.722.1 G.722.2 G.723 G.723.1 G.726 G.728 G.729 G.729.1 G.729a

Others

AC3 Apple Lossless ATRAC FLAC iLBC Monkey's Audio μ-law Musepack Nellymoser OptimFROG RealAudio SHN Speex Vorbis WavPack WMA TAK

Image compression formats

ISO/IEC/ITU-T

JPEG JPEG 2000 lossless JPEG JBIG JBIG2 PNG WBMP

Others

BMP GIF ILBM PCX PGF TGA TIFF HD Photo

Media container formats

General

3GP ASF AVI DMF DPX FLV Matroska MP4 MXF NUT Ogg Ogg Media QuickTime RealMedia VOB

Audio only

AIFF AU WAV

See for methods and for codecs

MPEG-4 is a standard used primarily to compress audio and visual (AV) digital data. Introduced in late 1998, it is the designation for a group of audio and video coding standards and related technology agreed upon by the ISO/IEC Moving Picture Experts Group (MPEG) under the formal
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International Organization for Standardization (Organisation internationale de normalisation), widely known as ISO, is an international standard-setting body composed of representatives from various national standards organizations.
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The International Electrotechnical Commission^[1] (IEC) is a not-for-profit, non-governmental international standards organization that prepares and publishes International Standards for all electrical, electronic and related technologies – collectively known
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Sound is a disturbance of mechanical energy that propagates through matter as a wave (through fluids as a compression wave, and through solids as both compression and shear waves).
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The term coding has the following meanings:

In communications systems, the altering of the characteristics of a signal to make the signal more suitable for an intended application, such as optimizing the signal for transmission, improving transmission quality and fidelity,

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Advanced Audio Coding

File extension: .m4a, .m4b, .m4p, .m4v, .aac, .3gp, .mp4
Type of format: Lossy compression
Container for: Audio

Advanced Audio Coding (AAC) is a standardized, lossy compression and encoding scheme for
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MPEG-2 is a standard for "the generic coding of moving pictures and associated audio information".^[1] It describes a combination of lossy video compression and lossy audio compression (audio data compression) methods which permit storage and transmission of movies using
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Sound quality generally is the quality of the audio output from various electronic devices.

Sound quality can be defined as the degree of accuracy with which a device records or emits the original sound waves.
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High Efficiency AAC (HE-AAC) is a lossy data compression scheme for digital audio. It is an extension of Low Complexity AAC (AAC LC) optimized for low-bitrate applications such as streaming audio.
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Spectral band replication (SBR) is a technology to enhance audio or speech codecs, especially at low bit rates.

How it works

It can be combined with any audio compression codec: the codec itself transmits the lower frequencies of the spectrum, while SBR synthesizes
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Parametric Stereo is a feature used in Advanced Audio Coding to further enhance efficiency in low bandwidth stereo media. It, along with Spectral Band Replication, is part of HE-AAC v2. An HE-AAC v1 decoder will only give mono sound when decoding an AAC HE v2 bitstream.
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A polyphase quadrature filter, or PQF, is a filter bank which splits an input signal into a given number N (mostly a power of 2) of equidistant sub-bands. These sub-bands are subsampled by a factor of N, so they are critically sampled.
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The letters MDCT may stand for:

Modified discrete cosine transform
Multidetector computed tomography

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The letters MDCT may stand for:

Modified discrete cosine transform
Multidetector computed tomography

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This article or section may contain original research or unverified claims.

Please help Wikipedia by adding references. See the for details.
This article has been tagged since September 2007.

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Fault-tolerance or graceful degradation is the property that enables a system (often computer-based) to continue operating properly in the event of the failure of (or one or more faults within) some of its components.
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Digital Multimedia Broadcasting (DMB) is a digital radio transmission system for sending multimedia (radio, TV, and datacasting) to mobile devices such as mobile phones.
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MPEG-4 Part 2 is a video compression technology developed by MPEG. It belongs to the MPEG-4 ISO/IEC standard (ISO/IEC 14496-2). It is a Discrete Cosine Transform compression standard, similar to previous standards such as MPEG-1 and MPEG-2.
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MP4 (MPEG-4 Part 14)

File extension: .mp4
MIME type: video/mp4
Type code: mpg4
Developed by: ISO
Type of format: Media container
Container for: Audio, video, text
Extended from: Quicktime .
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Digital rights management (DRM) is an umbrella term that refers to access control technologies used by publishers and other copyright holders to limit usage of digital media or devices.
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Multimedia (Lat. Multum + Medium) is media that uses multiple forms of information content and information processing (e.g. text, audio, graphics, animation, video, interactivity) to inform or entertain the (user) audience.
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data compression or source coding is the process of encoding information using fewer bits (or other information-bearing units) than an un-encoded representation would use through use of specific encoding schemes.
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Video compression refers to reducing the quantity of data used to represent video images, and this is almost always coupled with the goal of retaining as much of the original's quality as possible.
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Mpeg 4 Part 3

Information about Mpeg 4 Part 3

Bifurcation in the AAC technical standard

HE-AAC

AAC-SSR

Why AAC-SSR was introduced

BSAC

See also

External links

How it works