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.
- 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. 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.
|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).|
|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||AAC LTP (Long Term Prediction)||1999||contains AAC LC|
|5||SBR (Spectral Band Replication)||2003||used with AAC LC in the "High Efficiency AAC Profile" (HE-AAC v1)|
|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|
|12||TTSI (Text-To-Speech Interface)||1999|
|13||Main synthesis||1999||contains 'wavetable' sample-based synthesis and Algorithmic Synthesis and Audio Effects|
|14||'wavetable' sample-based synthesis||1999||based on SoundFont and DownLoadable Sounds, contains General MIDI|
|16||Algorithmic Synthesis and Audio Effects||1999|
|17||ER AAC LC||2000||Error Resilient|
|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|
|29||PS (Parametric Stereo)||2004 and 2006||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||also known as MPEG Spatial Audio Coding (SAC), it is a type of spatial audio coding (MPEG Surround was also defined in ISO/IEC 23003-1 in 2007)|
|34||MPEG-1/2 Layer-3||2005||also known as "MP3onMP4"|
|35||DST (Direct Stream Transfer)||2005||lossless audio coding, used on Super Audio CD|
|36||ALS (Audio Lossless Coding)||2006||lossless audio coding|
|37||SLS (Scalable Lossless Coding)||2006||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||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)|
|42||USAC (Unified Speech and Audio Coding)||2012||Unified Speech and audio Coding is defined in MPEG-D Part 3 (ISO/IEC 23003-3:2012)|
|43||SAOC (Spatial Audio Object Coding)||2010||note: Spatial Audio Object Coding is also the MPEG-D Part 2 standard (ISO/IEC 23003-2:2010)|
|44||LD MPEG Surround||2010||This object type conveys Low Delay MPEG Surround Coding side information (that was defined in MPEG-D Part 2 – ISO/IEC 23003-2
) in the MPEG-4 Audio framework.
|45||SAOC-DE||2013||Spatial Audio Object Coding Dialogue Enhancement|
|46||Audio Sync||2015||The audio synchronization tool provides capability of synchronizing multiple contents in multiple devices.|
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.
|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||2009|
|ALS Simple Profile||ALS||2010|
Audio storage and transport
|Multiplex||ISO/IEC 14496-1||MPEG-4 Multiplex scheme (M4Mux)|
|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. 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-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).
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. 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.
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.
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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 . 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 . 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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- Apple: MPEG-4: AAC
- "AAC" (VideoLAN WIKI)
- 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.
- A page comparing codecs including HE-AAC @64 kbit/s by listening tests. (Page is offline)
- Official MPEG web site
- 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
- RFC 4337 – MIME Type Registration for MPEG-4