ISO/IEC 14496-3:2009

INTERNATIONAL ISO/IEC
STANDARD 14496-3
Fourth edition
2009-09-01


Information technology — Coding of
audio-visual objects —
Part 3:
Audio
Technologies de l'information — Codage des objets audiovisuels —
Partie 3: Codage audio




Reference number
ISO/IEC 14496-3:2009(E)
©
ISO/IEC 2009

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ISO/IEC 14496-3:2009(E)
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ISO/IEC 14496-3:2009(E)
Foreword
ISO (the International Organization for Standardization) and IEC (the International Electrotechnical
Commission) form the specialized system for worldwide standardization. National bodies that are members of
ISO or IEC participate in the development of International Standards through technical committees
established by the respective organization to deal with particular fields of technical activity. ISO and IEC
technical committees collaborate in fields of mutual interest. Other international organizations, governmental
and non-governmental, in liaison with ISO and IEC, also take part in the work. In the field of information
technology, ISO and IEC have established a joint technical committee, ISO/IEC JTC 1.
International Standards are drafted in accordance with the rules given in the ISO/IEC Directives, Part 2.
The main task of the joint technical committee is to prepare International Standards. Draft International
Standards adopted by the joint technical committee are circulated to national bodies for voting. Publication as
an International Standard requires approval by at least 75 % of the national bodies casting a vote.
Attention is drawn to the possibility that some of the elements of this document may be the subject of patent
rights. ISO and IEC shall not be held responsible for identifying any or all such patent rights.
ISO/IEC 14496-3 was prepared by Joint Technical Committee ISO/IEC JTC 1, Information technology,
Subcommittee SC 29, Coding of audio, picture, multimedia and hypermedia information.
This fourth edition cancels and replaces the third edition (ISO/IEC 14496-3:2005), which has been technically
revised. It also incorporates the Amendments ISO/IEC 14496-3:2005/Amd.1:2007,
ISO/IEC 14496-3:2005/Amd.2:2006, ISO/IEC 14496-3:2005/Amd.3:2006, ISO/IEC 14496-3:2005/Amd.5:2007,
ISO/IEC 14496-3:2005/Amd.8, ISO/IEC 14496-3:2005/Amd.9:2008, and the Technical Corrigenda
ISO/IEC 14496-3:2005/Cor.2:2008, ISO/IEC 14496-3:2005/Cor.3:2008, ISO/IEC 14496-3:2005/Cor.4:2008,
ISO/IEC 14496-3:2005/Cor.5:2008, ISO/IEC 14496-3:2005/Amd.2:2006/Cor.1:2006,
ISO/IEC 14496-3:2005/Amd.2:2006/Cor.2:2008, ISO/IEC 14496-3:2005/Amd.2:2006/Cor.3:2008,
ISO/IEC 14496-3:2005/Amd.3:2006/Cor.1:2008.
ISO/IEC 14496 consists of the following parts, under the general title Information technology — Coding of
audio-visual objects:
⎯ Part 1: Systems
⎯ Part 2: Visual
⎯ Part 3: Audio
⎯ Part 4: Conformance testing
⎯ Part 5: Reference software
⎯ Part 6: Delivery Multimedia Integration Framework (DMIF)
⎯ Part 7: Optimized reference software for coding of audio-visual objects [Technical Report]
⎯ Part 8: Carriage of ISO/IEC 14496 contents over IP networks
⎯ Part 9: Reference hardware description [Technical Report]
⎯ Part 10: Advanced Video Coding
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ISO/IEC 14496-3:2009(E)
⎯ Part 11: Scene description and application engine
⎯ Part 12: ISO base media file format
⎯ Part 13: Intellectual Property Management and Protection (IPMP) extensions
⎯ Part 14: MP4 file format
⎯ Part 15: Advanced Video Coding (AVC) file format
⎯ Part 16: Animation Framework eXtension (AFX)
⎯ Part 17: Streaming text format
⎯ Part 18: Font compression and streaming
⎯ Part 19: Synthesized texture stream
⎯ Part 20: Lightweight Application Scene Representation (LASeR) and Simple Aggregation Format (SAF)
⎯ Part 21: MPEG-J Graphics Framework eXtensions (GFX)
⎯ Part 22: Open Font Format
⎯ Part 23: Symbolic Music Representation
⎯ Part 24: Audio and systems interaction [Technical Report]
⎯ Part 25: 3D Graphics Compression Model
Audio conformance and 3D graphics conformance will form the subjects of the future Parts 26 and 27,
respectively.

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ISO/IEC 14496-3:2009(E)
0 Introduction
0.1 Overview
This part of ISO/IEC 14496 (MPEG-4 Audio) is a new kind of audio standard that integrates many different types of
audio coding: natural sound with synthetic sound, low bitrate delivery with high-quality delivery, speech with music,
complex soundtracks with simple ones, and traditional content with interactive and virtual-reality content. By
standardizing individually sophisticated coding tools as well as a novel, flexible framework for audio
synchronization, mixing, and downloaded post-production, the developers of the MPEG-4 Audio standard have
created new technology for a new, interactive world of digital audio.
MPEG-4, unlike previous audio standards created by ISO/IEC and other groups, does not target a single
application such as real-time telephony or high-quality audio compression. Rather, MPEG-4 Audio is a standard
that applies to every application requiring the use of advanced sound compression, synthesis, manipulation, or
playback. The subparts that follow specify the state-of-the-art coding tools in several domains; however, MPEG-4
Audio is more than just the sum of its parts. As the tools described here are integrated with the rest of the MPEG-4
standard, exciting new possibilities for object-based audio coding, interactive presentation, dynamic soundtracks,
and other sorts of new media, are enabled.
Since a single set of tools is used to cover the needs of a broad range of applications, interoperability is a natural
feature of systems that depend on the MPEG-4 Audio standard. A system that uses a particular coder — for
example a real-time voice communication system making use of the MPEG-4 speech coding toolset — can easily
share data and development tools with other systems, even in different domains, that use the same tool — for
example a voicemail indexing and retrieval system making use of MPEG-4 speech coding.
The remainder of this clause gives a more detailed overview of the capabilities and functioning of MPEG-4 Audio.
First a discussion of concepts, that have changed since the MPEG-2 Audio standards, is presented. Then the
MPEG-4 Audio toolset is outlined.
0.2 Concepts of MPEG-4 Audio
As with previous MPEG standards, MPEG-4 does not standardize methods for encoding sound. Thus, content
authors are left to their own decisions as to the best method of creating bitstream payloads. At the present time,
methods to automatically convert natural sound into synthetic or multi-object descriptions are not mature; therefore,
most immediate solutions will involve interactively-authoring the content stream in some way. This process is
similar to current schemes for MIDI-based and multi-channel mixdown authoring of soundtracks.
Many concepts in MPEG-4 Audio are different than those in previous MPEG Audio standards. For the benefit of
readers who are familiar with MPEG-1 and MPEG-2 we provide a brief overview here.
0.2.1 Audio storage and transport facilities
In all of the MPEG-4 tools for audio coding, the coding standard ends at the point of constructing access units that
contain the compressed data. The MPEG-4 Systems (ISO/IEC 14496-1) specification describes how to convert
these individually coded access units into elementary streams.
There is no standard transport mechanism of these elementary streams over a channel. This is because the broad
range of applications that can make use of MPEG-4 technology have delivery requirements that are too wide to
easily characterize with a single solution. Rather, what is standardized is an interface (the Delivery Multimedia
Interface Format, or DMIF, specified in ISO/IEC 14496-6) that describes the capabilities of a transport layer and the
communication between transport, multiplex, and demultiplex functions in encoders and decoders. The use of
DMIF and the MPEG-4 Systems specification allows transmission functions that are much more sophisticated than
are possible with previous MPEG standards.
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However, LATM and LOAS were defined to provide a low overhead audio multiplex and transport mechanism for
natural audio applications, which do not require sophisticated object-based coding or other functions provided by
MPEG-4 Systems.
Table 0.1 gives an overview about the multiplex, storage and transmission formats currently available for MPEG-4
Audio within the MPEG-4 framework:
Table 0.1 — Format overview
Format Functionality defined in Functionality originally Description
MPEG-4: defined in:
M4Mux ISO/IEC 14496-1 - MPEG-4 Multiplex scheme
(normative)
LATM ISO/IEC 14496-3 - Low Overhead Audio Transport
(normative) Multiplex
ADIF ISO/IEC 14496-3 ISO/IEC 13818-7 Audio Data Interchange Format,
(informative) (normative) (AAC only)
MP4FF ISO/IEC 14496-12 - MPEG-4 File Format
(normative)
ADTS ISO/IEC 14496-3 ISO/IEC 13818-7 Audio Data Transport Stream,
(informative) (normative, exemplarily) (AAC only)
LOAS ISO/IEC 14496-3 - Low Overhead Audio Stream, based
(normative, exemplarily) on LATM, three versions are
available:
AudioSyncStream()
EPAudioSyncStream()
AudioPointerStream()

To allow for a user on the remote side of a channel to dynamically control a server streaming MPEG-4 content,
MPEG-4 defines backchannel streams that can carry user interaction information.
0.2.2 MPEG-4 Audio supports low-bitrate coding
Previous MPEG Audio standards have focused primarily on transparent (undetectable) or nearly transparent coding
of high-quality audio at whatever bitrate was required to provide it. MPEG-4 provides new and improved tools for
this purpose, but also standardizes (and has tested) tools that can be used for transmitting audio at the low bitrates
suitable for Internet, digital radio, or other bandwidth-limited delivery. The new tools specified in MPEG-4 are the
state-of-the-art tools that support low-bitrate coding of speech and other audio.
0.2.3 MPEG-4 Audio is an object-based coding standard with multiple tools
Previous MPEG Audio standards provided a single toolset, with different configurations of that toolset specified for
use in various applications. MPEG-4 provides several toolsets that have no particular relationship to each other,
each with a different target function. The profiles of MPEG-4 Audio specify which of these tools are used together
for various applications.
Further, in previous MPEG standards, a single (perhaps multi-channel or multi-language) piece of content was
transmitted. In contrast, MPEG-4 supports a much more flexible concept of a soundtrack. Multiple tools may be
used to transmit several audio objects, and when using multiple tools together an audio composition system is
provided to create a single soundtrack from the several audio substreams. User interaction, terminal capability, and
speaker configuration may be used when determining how to produce a single soundtrack from the component
objects. This capability gives MPEG-4 significant advantages in quality and flexibility when compared to previous
audio standards.
0.2.4 MPEG-4 Audio provides capabilities for synthetic sound
In natural sound coding, an existing sound is compressed by a server, transmitted and decompressed at the
receiver. This type of coding is the subject of many existing standards for sound compression. In contrast, MPEG-4
standardizes a novel paradigm in which synthetic sound descriptions, including synthetic speech and synthetic
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ISO/IEC 14496-3:2009(E)
music, are transmitted and then synthesized into sound at the receiver. Such capabilities open up new areas of
very-low-bitrate but still very-high-quality coding.
0.2.5 MPEG-4 Audio provides capabilities for error robustness
Improved error robustness capabilities for all coding tools are provided through the error resilient bitstream payload
syntax. This tool supports advanced channel coding techniques, which can be adapted to the special needs of
given coding tools and a given communications channel. This error resilient bitstream payload syntax is mandatory
for all error resilient object types.
The error protection tool (EP tool) provides unequal error protection (UEP) for MPEG-4 Audio in conjunction with
the error resilient bitstream payload. UEP is an efficient method to improve the error robustness of source coding
schemes. It is used by various speech and audio coding systems operating over error-prone channels such as
mobile telephone networks or Digital Audio Broadcasting (DAB). The bits of the coded signal representation are
first grouped into different classes according to their error sensitivity. Then error protection is individually applied to
the different classes, giving better protection to more sensitive bits.
Improved error robustness for AAC is provided by a set of error resilience tools. These tools reduce the perceived
degradation of the decoded audio signal that is caused by corrupted bits in the bitstream payload.
0.2.6 MPEG-4 Audio provides capabilities for scalability
Previous MPEG Audio standards provided a single bitrate, single bandwidth toolset, with different configurations of
that toolset specified for use in various applications. MPEG-4 provides several bitrate and bandwidth options within
a single stream, providing a scalability functionality that permits a given stream to scale to the requirement of
different channels and applications or to be responsive to a given channel that has dynamic throughput
characteristics. The tools specified in MPEG-4 are the state-of-the-art tools providing scalable compression of
speech and audio signals.
0.3 The MPEG-4 Audio tool set
0.3.1 Speech coding tools
0.3.1.1 Overview
Speech coding tools are designed for the transmission and decoding of synthetic and natural speech.
Two types of speech coding tools are provided in MPEG-4. The natural speech tools allow the compression,
transmission, and decoding of human speech, for use in telephony, personal communication, and surveillance
applications. The synthetic speech tool provides an interface to text-to-speech synthesis systems; using synthetic
speech provides very-low-bitrate operation and built-in connection with facial animation for use in low-bitrate video
teleconferencing applications.
0.3.1.2 Natural speech coding
The MPEG-4 speech coding toolset covers the compression and decoding of natural speech sound at bitrates
ranging between 2 and 24 kbit/s. When variable bitrate coding is allowed, coding at even less than 2 kbit/s, for
example an average bitrate of 1,2 kbit/s, is also supported. Two basic speech coding techniques are used: One is a
parametric speech coding algorithm, HVXC (Harmonic Vector eXcitation Coding), for very low bit rates; and the
other is a CELP (Code Excited Linear Prediction) coding technique. The MPEG-4 speech coders target
applications ranging from mobile and satellite communications, to Internet telephony, to packaged media and
speech databases. It meets a wide range of requirements encompassing bitrate, functionality and sound quality.
MPEG-4 HVXC operates at fixed bitrates between 2,0 kbit/s and 4,0 kbit/s using a bitrate scalability technique. It
also operates at lower bitrates, typically 1,2 - 1,7 kbit/s, using a variable bitrate technique. HVXC provides
communications-quality to near-toll-quality speech in the 100 Hz – 3 800 Hz band at 8 kHz sampling rate. HVXC
also allows independent change of speed and pitch during decoding, which is a powerful functionality for fast
access to speech databases. HVXC functionalities include 2,0 - 4,0 kbit/s fixed bitrate modes and a 2,0 kbit/s
maximum variable bitrate mode.
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Error Resilient (ER) HVXC extends operation of the variable bitrate mode to 4,0 kbit/s to allow higher quality
variable rate coding. The ER HVXC therefore provides fixed bitrate modes of 2,0 - 4,0 kbit/s and a variable bitrate
of either less than 2,0 kbit/s or less than 4,0 kbit/s, both in scalable and non-scalable modes. In the variable bitrate
modes, non-speech parts are detected in unvoiced signals, and a smaller number of bits are used for these non-
speech parts to reduce the average bitrate. ER HVXC provides communications-quality to near-toll-quality speech
in the 100 Hz - 3800 Hz band at 8 kHz sampling rate. When the variable bitrate mode is allowed, operation at lower
average bitrate is possible. Coded speech using variable bitrate mode at typical bitrates of 1,5 kbit/s average, and
at typical bitrate of 3,0 kbit/s average has essentially the same quality as 2,0 kbit/s fixed rate and 4,0 kbit/s fixed
rate respectively. The functionality of pitch and speed change during decoding is supported for all modes. ER
HVXC has a bitstream payload syntax with the error sensitivity classes to be used with the EP-Tool, and some
error concealment functionality is supported for use in error-prone channels such as mobile communication
channels. The ER HVXC speech coder target applications range from mobile and satellite communications, to
Internet telephony, to packaged media and speech databases.
MPEG-4 CELP is a well-known coding algorithm with new functionality. Conventional CELP coders offer
compression at a single bit rate and are optimized for specific applications. Compression is one of the
functionalities provided by MPEG-4 CELP, but MPEG-4 also enables the use of one basic coder in multiple
applications. It provides scalability in bitrate and bandwidth, as well as the ability to generate bitstream payloads at
arbitrary bitrates. The MPEG-4 CELP coder supports two sampling rates, namely, 8 kHz and 16 kHz. The
associated bandwidths are 100 Hz – 3800 Hz for 8 kHz sampling and 50 Hz – 7000 Hz for 16 kHz sampling. The
silence compression tool comprises a voice activity detector (VAD), a discontinuous transmission (DTX) unit and a
comfort noise generator (CNG) module. The tool encodes/decodes the input signal at a lower bitrate during the
non-active-voice (silent) frames. During the active-voice (speech) frames, MPEG-4 CELP encoding and decoding
are used.
The silence compression tool reduces the average bitrate thanks to compression at a lower bitrate for silence. In
the encoder, a voice activity detector is used to distinguish between regions with normal speech activity and those
with silence or background noise. During normal speech activity, the CELP coding is used. Otherwise a silence
insertion descriptor (SID) is transmitted at a lower bitrate. This SID enables a comfort noise generator (CNG) in the
decoder. The amplitude and the spectral shape of this comfort noise are specified by energy and LPC parameters
in methods similar to those used in a normal CELP frame. These parameters are optionally re-transmitted in the
SID and thus can be updated as required.
MPEG has conducted extensive verification testing in realistic listening conditions in order to prove the efficacy of
the speech coding toolset.
0.3.1.3 Text-to-speech interface
Text-to-speech (TTS) capability is becoming a rather common media type and plays an important role in various
multimedia application areas. For instance, by using TTS functionality, multimedia content with narration can be
easily created without recording natural speech. Before MPEG-4, however, there was no way for a multimedia
content provider to easily give instructions to an unknown TTS system. With MPEG-4 TTS Interface, a single
common interface for TTS systems is standardized. This interface allows speech information to be transmitted in
the international phonetic alphabet (IPA), or in a textual (written) form of any language.
The MPEG-4 Hybrid/Multi-Level Scalable TTS Interface is a superset of the conventional TTS framework. This
extended TTS Interface can utilize prosodic information taken from natural speech in addition to input text and can
thus generate much higher-quality synthetic speech. The interface and its bitstream payload format is scalable in
terms of this added information; for example, if some parameters of prosodic information are not available, a
decoder can generate the missing parameters by rule. Normative algorithms for speech synthesis and text-to-
phoneme translation are not specified in MPEG-4, but to meet the goal that underlies the MPEG-4 TTS Interface, a
decoder should fully utilize all the provided information according to the user's requirement level.
As well as an interface to text-to-speech synthesis systems, MPEG-4 specifies a joint coding method for phonemic
information and facial animation (FA) parameters and other animation parameters (AP). Using this technique, a
single bitstream payload may be used to control both the text-to-speech interface and the facial animation visual
object decoder (see ISO/IEC 14496-2, Annex C). The functionality of this extended TTS thus ranges from
conventional TTS to natural speech coding and its application areas, from simple TTS to audio presentation with
TTS and motion picture dubbing with TTS.
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0.3.2 Audio coding tools
0.3.2.1 Overview
Audio coding tools are designed for the transmission and decoding of recorded music and other audio soundtracks.
0.3.2.2 General audio coding tools
MPEG-4 standardizes the coding of natural audio at bitrates ranging from 6 kbit/s up to several hundred kbit/s per
audio channel for mono, two-channel, and multi-channel-stereo signals. General high-quality compression is
provided by incorporating the MPEG-2 AAC standard (ISO/IEC 13818-7), with certain improvements, as MPEG-4
AAC. At 64 kbit/s/channel and higher ranges, this coder has been found in verification testing under rigorous
conditions to meet the criterion of “indistinguishable quality” as defined by the European Broadcasting Union.
General audio (GA) coding tools comprise the AAC tool set expanded by alternative quantization and coding
schemes (Twin-VQ and BSAC). The general audio coder uses a perceptual filterbank, a sophisticated masking
model, noise-shaping techniques, channel coupling, and noiseless coding and bit-allocation to provide the
maximum compression within the constraints of providing the highest possible quality. Psychoacoustic coding
standards developed by MPEG have represented the state-of-the-art in this technology since MPEG-1 Audio;
MPEG-4 General Audio coding continues this tradition.
For bitrates ranging from 6 kbit/s to 64 kbit/s per channel, the MPEG-4 standard provides extensions to the GA
coding tools, that allow the content author to achieve the highest quality coding at the desired bitrate. Furthermore,
various bit rate scalability options are available within the GA coder. The low-bitrate techniques and scalability
modes provided within this tool set have also been verified in formal tests by MPEG.
The MPEG-4 low delay coding functionality provides the ability to extend the usage of generic low bitrate audio
coding to applications requiring a very low delay in the encoding/decoding chain (e.g. full-duplex real-time
communications). In contrast to traditional low delay coders based on speech coding technology, the concept of
this low delay coder is based on general perceptual audio coding and is thus suitable for a wide range of audio
signals. Specifically, it is derived from the proven architecture of MPEG-2/4 Advanced Audio Coding (AAC) and all
capabilities for coding of 2 (stereo) or more sound channels (multi-channel) are available within the low delay
coder. To enable coding of general audio signals with an algorithmic delay not exceeding 20 ms at 48 kHz, it uses
a frame length of 512 or 480 samples (compared to the 1024 or 960 samples used in standard MPEG-2/4 AAC).
Also the size of the window used in the analysis and synthesis filterbank is reduced by a factor of 2. No block
switching is used to avoid the “look-ahead'” delay due to the block switching decision. To reduce pre-echo artefacts
in the case of transient signals, window shape switching is provided instead. For non-transient portions of the signal
a sine window is used, while a so-called low overlap window is used for transient portions. Use of the bit reservoir
is minimized in the encoder in order to reach the desired target delay. As one extreme case, no bit reservoir is used
at all.
The MPEG-4 BSAC is used in combination with the AAC coding tools and replaces the noiseless coding of the
quantized spectral data and the scalefactors. The MPEG-4 BSAC provides fine grain scalability in steps of 1 kbit/s
per audio channel, i.e. 2 kbit/s steps for a stereo signal. One base layer stream and many small enhancement layer
streams are used. To obtain fine step scalability, a bit-slicing scheme is applied to the quantized spectral data. First
the quantized spectral values are grouped into frequency bands. Each of these groups contains the quantized
spectral values in their binary representation. Then the bits of a group are processed in slices according to their
significance. Thus all most significant bits (MSB) of the quantized values in a group are processed first. These bit-
slices are then encoded using an arithmetic coding scheme to obtain entropy coding with minimal redundancy. In
order to implement fine grain scalability efficiently using MPEG-4 Systems tools, the fine grain audio data can be
grouped into large-step layers and these large-step layers can be further grouped by concatenating large-step
layers from several sub-frames. Furthermore, the configuration of the payload transmitted over an Elementary
Stream (ES) can be changed dynamically (by means of the MPEG-4 backchannel capability) depending on the
environment, such as network traffic or user interaction. This means that BSAC can allow for real-time adjustments
to the quality of service. In addition to fine grain scalability, it can improve the quality of an audio signal that is
decoded from a stream transmitted over an error-prone channel, such as a mobile communication network or
Digital Audio Broadcasting (DAB) channel.
MPEG-4 SBR (Spectral Band Replication) is a bandwidth extension tool used in combination with the AAC general
audio codec. When integrated into the MPEG AAC codec, a significant improvement of the performance is
available, which can be used to lower the bitrate or improve the audio quality. This is achieved by replicating the
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highband, i.e. the high frequency part of the spectrum. A small amount of data representing a parametric
description of the highband is encoded and used in the decoding process. The data rate is by far below the data
rate required when using conventional AAC coding of the highband.
0.3.2.3 Parametric audio coding tools
The parametric audio coding tool MPEG-4 HILN (Harmonic and Individual Lines plus Noise) codes non-speech
signals like music at bitrates of 4 kbit/s and higher using a parametric representation of the audio signal. The basic
idea of this technique is to decompose the input signal into audio objects which are described by appropriate
source models and represented by model parameters. Object models for sinusoids, harmonic tones, and noise are
utilized in the HILN coder. HILN allows independent change of speed and pitch during decoding.
The Parametric Audio Coding tools combine very low bitrate coding of general audio signals with the possibility of
modifying the playback speed or pitch during decoding without the need for an effects processing unit. In
combination with the speech and audio coding tools in MPEG-
...