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.[1] It specifies audio coding methods. The first version of ISO/IEC 14496-3 was published in 1999.[2]

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.[3][4][5][6][7][8][9][10][11]

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.[7]


MPEG-4 Audio versions and editions[12]
Edition Release date Latest amendment Standard Description
First edition 1999 2001 ISO/IEC 14496-3:1999[2] also known as "MPEG-4 Audio Version 1"
2000 ISO/IEC 14496-3:1999/Amd 1:2000[13] also known as "MPEG-4 Audio Version 2", an Amendment to first edition[7][8]
Second edition 2001 2005 ISO/IEC 14496-3:2001[14]
Third edition 2005 2008 ISO/IEC 14496-3:2005[15]
Fourth edition 2009 2015 and under development[12] ISO/IEC 14496-3:2009[1][16]


MPEG-4 Part 3 contains following subparts:[16]

  • Subpart 1: Main (list of Audio Object Types, Profiles, Levels, interface to ISO/IEC 14496-1, MPEG-4 Audio transport stream, etc.)
  • Subpart 2: Speech coding – HVXC (Harmonic Vector eXcitation Coding)
  • Subpart 3: Speech coding – CELP (Code Excited Linear Prediction)
  • Subpart 4: General Audio Coding (GA) (Time/Frequency Coding) – AAC, TwinVQ, BSAC
  • Subpart 5: Structured Audio (SA)
  • Subpart 6: Text to Speech Interface (TTSI)
  • Subpart 7: Parametric Audio Coding – HILN (Harmonic and Individual Line plus Noise)
  • Subpart 8: Technical description of parametric coding for high quality audio (SSC, Parametric Stereo)
  • Subpart 9: MPEG-1/MPEG-2 Audio in MPEG-4
  • Subpart 10: Technical description of lossless coding of oversampled audio (MPEG-4 DST – Direct Stream Transfer)
  • Subpart 11: Audio Lossless Coding (ALS)
  • Subpart 12: Scalable Lossless Coding (SLS)

MPEG-4 Audio Object Types

MPEG-4 Audio includes a system for handling a diverse group of audio formats in a uniform manner. Each format is assigned a unique Audio Object Type to represent it.[17][18] Object Type is used to distinguish between different coding methods. It directly determines the MPEG-4 tool subset required to decode a specific object. The MPEG-4 profiles are based on the object types and each profile supports different list of object types.[18]

MPEG-4 Audio Object Types[7][9][17][19][20]
Object Type ID Audio Object Type First public release date Description
1 AAC Main 1999 contains AAC LC
2 AAC LC (Low Complexity) 1999 Used in the "AAC Profile". MPEG-4 AAC LC Audio Object Type is based on the MPEG-2 Part 7 Low Complexity profile (LC) combined with Perceptual Noise Substitution (PNS) (defined in MPEG-4 Part 3 Subpart 4).[4][21]
3 AAC SSR (Scalable Sample Rate) 1999 MPEG-4 AAC SSR Audio Object Type is based on the MPEG-2 Part 7 Scalable Sampling Rate profile (SSR) combined with Perceptual Noise Substitution (PNS) (defined in MPEG-4 Part 3 Subpart 4).[4][21]
4 AAC LTP (Long Term Prediction) 1999 contains AAC LC
5 SBR (Spectral Band Replication) 2003[22] used with AAC LC in the "High Efficiency AAC Profile" (HE-AAC v1)
6 AAC Scalable 1999
7 TwinVQ 1999 audio coding at very low bitrates
8 CELP (Code Excited Linear Prediction) 1999 speech coding
9 HVXC (Harmonic Vector eXcitation Coding) 1999 speech coding
10 (Reserved)
11 (Reserved)
12 TTSI (Text-To-Speech Interface) 1999
13 Main synthesis 1999 contains 'wavetable' sample-based synthesis[23] and Algorithmic Synthesis and Audio Effects
14 'wavetable' sample-based synthesis 1999 based on SoundFont and DownLoadable Sounds,[23] contains General MIDI
15 General MIDI 1999
16 Algorithmic Synthesis and Audio Effects 1999
17 ER AAC LC 2000 Error Resilient
18 (Reserved )
19 ER AAC LTP 2000 Error Resilient
20 ER AAC Scalable 2000 Error Resilient
21 ER TwinVQ 2000 Error Resilient
22 ER BSAC (Bit-Sliced Arithmetic Coding) 2000 It is also known as "Fine Granule Audio" or fine grain scalability tool. It is used in combination with the AAC coding tools and replaces the noiseless coding and the bitstream formatting of MPEG-4 Version 1 GA coder. Error Resilient
23 ER AAC LD (Low Delay) 2000 Error Resilient, used with CELP, ER CELP, HVXC, ER HVXC and TTSI in the "Low Delay Profile", (commonly used for real-time conversation applications)
24 ER CELP 2000 Error Resilient
25 ER HVXC 2000 Error Resilient
26 ER HILN (Harmonic and Individual Lines plus Noise) 2000 Error Resilient
27 ER Parametric 2000 Error Resilient
28 SSC (SinuSoidal Coding) 2004[24][25]
29 PS (Parametric Stereo) 2004[26] and 2006[27][28] used with AAC LC and SBR in the "HE-AAC v2 Profile". PS coding tool was defined in 2004 and Object Type defined in 2006.
30 MPEG Surround 2007[29] also known as MPEG Spatial Audio Coding (SAC), it is a type of spatial audio coding[30][31] (MPEG Surround was also defined in ISO/IEC 23003-1 in 2007[32])
31 (Reserved)
32 MPEG-1/2 Layer-1 2005[33]
33 MPEG-1/2 Layer-2 2005[33]
34 MPEG-1/2 Layer-3 2005[33] also known as "MP3onMP4"
35 DST (Direct Stream Transfer) 2005[34] lossless audio coding, used on Super Audio CD
36 ALS (Audio Lossless Coding) 2006[28] lossless audio coding
37 SLS (Scalable Lossless Coding) 2006[35] two-layer audio coding with lossless layer and lossy General Audio core/layer (e.g. AAC)
38 SLS non-core 2006 lossless audio coding without lossy General Audio core/layer (e.g. AAC)
39 ER AAC ELD (Enhanced Low Delay) 2008[36] Error Resilient
40 SMR (Symbolic Music Representation) Simple 2008 note: Symbolic Music Representation is also the MPEG-4 Part 23 standard (ISO/IEC 14496-23:2008)[37][38]
41 SMR Main 2008
42 USAC (Unified Speech and Audio Coding) (no SBR) 2012[39]
43 SAOC (Spatial Audio Object Coding) 2010[40][41] note: Spatial Audio Object Coding is also the MPEG-D Part 2 standard (ISO/IEC 23003-2:2010)[42]
44 LD MPEG Surround 2010[40][43] This object type conveys Low Delay MPEG Surround Coding side information (that was defined in MPEG-D Part 2 – ISO/IEC 23003-2[42]) in the MPEG-4 Audio framework.[44]
45 USAC[45] 2012[46] (it will be also defined in MPEG-D Part 3 – ISO/IEC 23003-3[47])

Audio Profiles

HE-AAC and HE-AAC v2
Hierarchical structure of AAC Profile, HE-AAC Profile and HE-AAC v2 Profile, and compatibility between them. The HE-AAC Profile decoder is fully capable of decoding any AAC Profile stream. Similarly the HE-AAC v2 decoder can handle all HE-AAC Profile streams as well as all AAC Profile streams. Based on the MPEG-4 Part 3 technical specification.[20]

The MPEG-4 Audio standard defines several profiles. These profiles are based on the object types and each profile supports different list of object types. Each profile may also have several levels, which limit some parameters of the tools present in a profile. These parameters usually are the sampling rate and the number of audio channels decoded at the same time.

MPEG-4 Audio Profiles[18][20]
Audio Profile Audio Object Types First public release date
AAC Profile AAC LC 2003
High Efficiency AAC Profile AAC LC, SBR 2003
HE-AAC v2 Profile AAC LC, SBR, PS 2006
Main Audio Profile AAC Main, AAC LC, AAC SSR, AAC LTP, AAC Scalable, TwinVQ, CELP, HVXC, TTSI, Main synthesis 1999
Scalable Audio Profile AAC LC, AAC LTP, AAC Scalable, TwinVQ, CELP, HVXC, TTSI 1999
Speech Audio Profile CELP, HVXC, TTSI 1999
Synthetic Audio Profile TTSI, Main synthesis 1999
High Quality Audio Profile AAC LC, AAC LTP, AAC Scalable, CELP, ER AAC LC, ER AAC LTP, ER AAC Scalable, ER CELP 2000
Low Delay Audio Profile CELP, HVXC, TTSI, ER AAC LD, ER CELP, ER HVXC 2000
Natural Audio Profile AAC Main, AAC LC, AAC SSR, AAC LTP, AAC Scalable, TwinVQ, CELP, HVXC, TTSI, ER AAC LC, ER AAC LTP, ER AAC Scalable, ER TwinVQ, ER BSAC, ER AAC LD, ER CELP, ER HVXC, ER HILN, ER Parametric 2000
Mobile Audio Internetworking Profile ER AAC LC, ER AAC Scalable, ER TwinVQ, ER BSAC, ER AAC LD 2000
HD-AAC Profile AAC LC, SLS[48] 2009[49]
ALS Simple Profile ALS 2010[41][50]

Audio storage and transport

Multiplex, storage and transmission formats for MPEG-4 Audio[16]
Standard Description
Multiplex ISO/IEC 14496-1 MPEG-4 Multiplex scheme (M4Mux)[51]
Multiplex ISO/IEC 14496-3 Low Overhead Audio Transport Multiplex (LATM)
Storage ISO/IEC 14496-3 (informative) Audio Data Interchange Format (ADIF) – only for AAC
Storage ISO/IEC 14496-12 MPEG-4 file format (MP4) / ISO base media file format
Transmission ISO/IEC 14496-3 (informative) Audio Data Transport Stream (ADTS) – only for AAC
Transmission ISO/IEC 14496-3 Low Overhead Audio Stream (LOAS), based on LATM

There is no standard for transport of elementary streams over a channel, because the broad range of MPEG-4 applications have delivery requirements that are too wide to easily characterize with a single solution.

The capabilities of a transport layer and the communication between transport, multiplex, and demultiplex functions are described in the Delivery Multimedia Integration Framework (DMIF) in ISO/IEC 14496-6.[16] A wide variety of delivery mechanisms exist below this interface, e.g., MPEG transport stream, Real-time Transport Protocol (RTP), etc.

Transport in Real-time Transport Protocol is defined in RFC 3016 (RTP Payload Format for MPEG-4 Audio/Visual Streams), RFC 3640 (RTP Payload Format for Transport of MPEG-4 Elementary Streams), RFC 4281 (The Codecs Parameter for "Bucket" Media Types) and RFC 4337 (MIME Type Registration for MPEG-4).

LATM and LOAS were defined for natural audio applications, which do not require sophisticated object-based coding or other functions provided by MPEG-4 Systems.

Bifurcation in the AAC technical standard

The Advanced Audio Coding in MPEG-4 Part 3 (MPEG-4 Audio) Subpart 4 was enhanced relative to the previous standard MPEG-2 Part 7 (Advanced Audio Coding), 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.

The MPEG-2 Part 7 standard (Advanced Audio Coding) was first published in 1997 and offers three default profiles:[52][53] Low Complexity profile (LC), Main profile and Scalable Sampling Rate profile (SSR).

The MPEG-4 Part 3 Subpart 4 (General Audio Coding) combined the profiles from MPEG-2 Part 7 with Perceptual Noise Substitution (PNS) and defined them as Audio Object Types (AAC LC, AAC Main, AAC SSR).[4]


High-Efficiency Advanced Audio Coding is an extension of AAC LC 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 Scalable Sample Rate was introduced by Sony to the MPEG-2 Part 7 and MPEG-4 Part 3 standards. It was first published in ISO/IEC 13818-7, Part 7: Advanced Audio Coding (AAC) in 1997.[52][53] 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 AAC LC 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 LC.

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.


  • 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 LC 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.


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.


In 2002, the MPEG-4 Audio Licensing Committee selected the Via Licensing Corporation as the Licensing Administrator for the MPEG-4 Audio patent pool.[3][54][55]

See also


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  3. ^ a b Business Wire (2002-12-02). "MPEG-4 Audio Licensing Committee Selects Via Licensing Corporation as Administrator; MPEG-4 Audio Licensing Committee Finalizing Terms for Audio Profile Licensing". The Free Library. Retrieved 2009-10-06.
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  5. ^ Karlheinz Brandenburg; Oliver Kunz; Akihiko Sugiyama. "MPEG-4 Natural Audio Coding – scalability in MPEG-4 natural audio". Archived from the original on 2010-02-28. Retrieved 2009-10-06.
  6. ^ D. Thom, H. Purnhagen, and the MPEG Audio Subgroup (October 1998). "MPEG Audio FAQ Version 9 – MPEG-4". Retrieved 2009-10-06.CS1 maint: Multiple names: authors list (link)
  7. ^ a b c d ISO/IEC JTC 1/SC 29/WG 11 (July 1999), ISO/IEC 14496-3:/Amd.1 – Final Committee Draft – MPEG-4 Audio Version 2 (PDF), retrieved 2009-10-07
  8. ^ a b Heiko Purnhagen (1999-06-07), An Overview of MPEG-4 Audio Version 2 (PDF), Heiko Purnhagen, retrieved 2009-10-07
  9. ^ a b Heiko Purnhagen (2001-06-01). "The MPEG-4 Audio Standard: Overview and Applications". Heiko Purnhagen. Retrieved 2009-10-07.
  10. ^ Heiko Purnhagen (2001-11-07). "The MPEG Audio Web Page – MPEG-4 Audio (ISO/IEC 14496-3)". Retrieved 2009-10-07.
  11. ^ Rob Koenen, ISO/IEC JTC1/SC29/WG11 (March 2002). "Overview of the MPEG-4 Standard". Retrieved 2009-10-06.
  12. ^ a b MPEG. "MPEG standards – Full list of standards developed or under development". Archived from the original on April 20, 2010. Retrieved 2009-10-31.
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  14. ^ ISO (2001). "ISO/IEC 14496-3:2001 - Information technology -- Coding of audio-visual objects -- Part 3: Audio". ISO. Retrieved 2009-10-14.
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  20. ^ a b c ISO/IEC JTC1/SC29/WG11/N7016 (2005-01-11), Text of ISO/IEC 14496-3:2001/FPDAM 4, Audio Lossless Coding (ALS), new audio profiles and BSAC extensions, archived from the original (DOC) on 2014-05-12, retrieved 2009-10-09
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  22. ^ ISO (2003). "Bandwidth extension, ISO/IEC 14496-3:2001/Amd 1:2003". ISO. Retrieved 2009-10-13.
  23. ^ a b Scheirer, Eric D.; Ray, Lee (1998). "Algorithmic and Wavetable Synthesis in the MPEG-4 Multimedia Standard". Audio Engineering Society Convention 105, 1998. 2.2 Wavetable synthesis with SASBF: The SASBF wavetable-bank format had a somewhat complex history of development. The original specification was contributed by E-Mu Systems and was based on their "SoundFont" format [15]. After integration of this component in the MPEG-4 reference software was complete, the MIDI Manufacturers Association (MMA) approached MPEG requesting that MPEG-4 SASBF be compatible with their "Downloaded Sounds" format [13]. E-Mu agreed that this compatibility was desirable, and so a new format was negotiated and designed collaboratively by all parties.
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  27. ^ 3GPP (2005-01-04). "ETSI TS 126 401 V6.1.0 (2004-12) - Universal Mobile Telecommunications System (UMTS)General audio codec audio processing functions; Enhanced aacPlus general audio codecGeneral description (3GPP TS 26.401 version 6.1.0 Release 6)". 3GPP. Retrieved 2009-10-13.
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External links

Advanced Audio Coding

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.

Audio Lossless Coding

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 (ISO/IEC 14496-3:2009) (4th edition) in August 2009.MPEG-4 ALS combines together a short-term predictor and a long term predictor. The short-term predictor is similar to FLAC in its operation - it is a quantized LPC predictor with a losslessly coded residual using Golomb Rice Coding or Block Gilbert Moore Coding (BGMC). The long term predictor is modeled by 5 long-term weighted residues, each with its own lag (delay). The lag can be hundreds of samples. This predictor improves the compression for sounds with rich harmonics (containing multiples of a single fundamental frequency, locked in phase) present in many musical instruments and human voice.


BSAC can stand for:

The British Screen Advisory Council

Bit Sliced Arithmetic Coding, audio coding from MPEG-4 Part 3

British South Africa Company

British Sub-Aqua Club

British Society for Antimicrobial Chemotherapy

Black Swamp Area Council

Benedictine Study and Arts Centre

Code-excited linear prediction

Code-excited linear prediction (CELP) is a speech coding algorithm originally proposed by M. R. Schroeder and B. S. Atal in 1985. At the time, it provided significantly better quality than existing low bit-rate algorithms, such as residual-excited linear prediction and linear predictive coding vocoders (e.g., FS-1015). Along with its variants, such as algebraic CELP, relaxed CELP, low-delay CELP and vector sum excited linear prediction, it is currently the most widely used speech coding algorithm. It is also used in MPEG-4 Audio speech coding. CELP is commonly used as a generic term for a class of algorithms and not for a particular codec.

Harmonic Vector Excitation Coding

Harmonic Vector Excitation Coding, abbreviated as HVXC is a speech coding algorithm specified in MPEG-4 Part 3 (MPEG-4 Audio) 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.It was published as subpart 2 of ISO/IEC 14496-3:1999 (MPEG-4 Audio) in 1999. An extended version of HVXC was published in MPEG-4 Audio Version 2 (ISO/IEC 14496-3:1999/Amd 1:2000).MPEG-4 Natural Speech Coding Tool Set uses two algorithms: HVXC and CELP (Code Excited Linear Prediction). HVXC is used at a low bit rate of 2 or 4 kbit/s. Higher bitrates than 4 kbit/s in addition to 3.85 kbit/s are covered by CELP.

MPEG-4 Part 14

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 official filename extension for MPEG-4 Part 14 files is .mp4. MPEG-4 Part 14 (formally ISO/IEC 14496-14:2003) is a standard specified as a part of MPEG-4.

Portable media players are sometimes advertised as "MP4 Players", although some are simply MP3 Players that also play AMV video or some other video format, and do not necessarily play the MPEG-4 Part 14 format.


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 SLS allows having both a lossy layer and a lossless correction layer similar to Wavpack Hybrid, OptimFROG DualStream and DTS-HD Master Audio, providing backwards compatibility to MPEG AAC-compliant bitstreams. MPEG-4 SLS can also work without a lossy layer (a.k.a. "SLS Non-Core"), in which case it will not be backwards compatible, Lossy compression of files is necessary for files that need to be streamed to the Internet or played in devices with limited storage.

With DRM, ripping of the lossless data or playback on non DRM-enabled devices could be disabled.

MPEG-4 SLS is not related in any way to MPEG-4 ALS (Audio Lossless Coding).

MPEG-4 Structured Audio

MPEG-4 Structured Audio is an ISO/IEC standard for describing sound. It was published as subpart 5 of MPEG-4 Part 3 (ISO/IEC 14496-3:1999) in 1999.It allows the transmission of synthetic music and sound effects at very low bit rates (from 0.01 to 10 kbit/s), and the description of parametric sound post-production for mixing multiple streams and adding effects to audio scenes. It does not standardize a particular set of synthesis methods, but a method for describing synthesis methods.

The sound descriptions generate audio when compiled (or interpreted) by a compliant decoder. MPEG-4 Structured Audio consists of the following major elements:

Structured Audio Orchestra Language (SAOL), an audio programming language. SAOL is historically related to Csound and other so-called Music-N languages. It was created by an MIT Media Lab grad student named Eric Scheirer while he was studying under Barry Vercoe during the 1990s.

Structured Audio Score Language (SASL) - is used to describe the manner in which algorithms described in SAOL are used to produce sound.

Structured Audio Sample Bank Format (SASBF) - allows for the transmission of banks of audio samples to be used in 'wavetable' sample-based synthesis (based on SoundFont and DownLoadable Sounds)

A normative Structured Audio scheduler description - it is the supervisory run-time element of the Structured Audio decoding process.

MIDI support - provides important backward-compatibility with existing content and authoring tools.MPEG-4 Structured Audio was cited by CNN as one of the top-25 innovations to arise at the Media Laboratory.

MPEG program stream

Program stream (PS or MPEG-PS) is a container format for multiplexing digital audio, video and more. The PS format is specified in MPEG-1 Part 1 (ISO/IEC 11172-1) and MPEG-2 Part 1, Systems (ISO/IEC standard 13818-1/ITU-T H.222.0). The MPEG-2 Program Stream is analogous and similar to ISO/IEC 11172 Systems layer and it is forward compatible.Program streams are used on DVD-Video discs and HD DVD video discs, but with some restrictions and extensions. The filename extensions are VOB and EVO respectively.

Nero Digital

Nero Digital is a brand name applied to a suite of MPEG-4-compatible video and audio compression codecs developed by Nero AG of Germany and Ateme of France. The audio codecs are integrated into the Nero Digital Audio+ audio encoding tool for Microsoft Windows, and the audio & video codecs are integrated into Nero's Recode DVD ripping software.

Nero certifies certain DVD player/recorder devices as Nero Digital compatible, and licenses the codec technology to integrated circuit manufacturers.The video codecs were developed by Ateme, and according to an interview with Nero AG developer Ivan Dimkovic, the audio codecs are improved versions of Dimkovic's older PsyTEL AAC Encoder. The audio codec is now available as a free stand-alone package called Nero AAC Codec.

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.

PlayStation Vita system software

The PlayStation Vita system software is the official firmware and operating system for the PlayStation Vita and PlayStation TV video game consoles. It uses the LiveArea as its graphical shell. The PlayStation Vita system software has one optional add-on component, the PlayStation Mobile Runtime Package. The system is built on a Unix-base which is derived from FreeBSD and NetBSD. The last version of the system software is 3.70, which was made available on January 14, 2019.

Spectral band replication

Spectral band replication (SBR) is a technology to enhance audio or speech codecs, especially at low bit rates and is based on harmonic redundancy in the frequency domain.

It can be combined with any audio compression codec: the codec itself transmits the lower and midfrequencies of the spectrum, while SBR replicates higher frequency content by transposing up harmonics from the lower and midfrequencies at the decoder. Some guidance information for reconstruction of the high-frequency spectral envelope is transmitted as side information.

When needed, it also reconstructs or adaptively mixes in noise-like information in selected frequency bands in order to faithfully replicate signals that originally contained no or fewer tonal components.

The SBR idea is based on the principle that the psychoacoustic part of the human brain tends to analyse higher frequencies with less accuracy; thus harmonic phenomena associated with the spectral band replication process needs only be accurate in a perceptual sense and not technically or mathematically exact.

Structured Audio Orchestra Language

Structured Audio Orchestra Language (SAOL) is an imperative, MUSIC-N programming language designed for describing virtual instruments, processing digital audio, and applying sound effects. It was published as subpart 5 of MPEG-4 Part 3 (ISO/IEC 14496-3:1999) in 1999.As part of the MPEG-4 international standard, SAOL is one of the key components of the MPEG-4 Structured Audio toolset, along with:

Structured Audio Score Language (SASL)

Structured Audio Sample Bank Format (SASBF)

The MPEG-4 SA scheduler

MIDI support


TwinVQ (transform-domain weighted interleave vector quantization) is an audio compression technique developed by Nippon Telegraph and Telephone Corporation (NTT) Human Interface Laboratories (now Cyber Space Laboratories) in 1994. The compression technique has been used in both standardized and proprietary designs.

MPEG-1 Parts
MPEG-2 Parts
MPEG-4 Parts
MPEG-7 Parts
MPEG-21 Parts
MPEG-D Parts
MPEG-G Parts
MPEG-H Parts

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