ISO/IEC 13818-7:2006

INTERNATIONAL ISO/IEC
STANDARD 13818-7
Fourth edition
2006-01-15
Information technology — Generic coding
of moving pictures and associated audio
information —
Part 7:
Advanced Audio Coding (AAC)
Technologies de l'information — Codage générique des images
animées et du son associé —
Partie 7: Codage du son avancé (AAC)

Reference number
©
ISO/IEC 2006
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©  ISO/IEC 2006
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ii © ISO/IEC 2006 – All rights reserved

Contents Page
Foreword.v
Introduction .vi
1 Scope .1
1.1 General.1
1.2 MPEG-2 AAC Tools Overview.1
2 Normative References.7
3 Terms and Definitions .7
4 Symbols and Abbreviations .14
4.1 Arithmetic Operators.14
4.2 Logical Operators .15
4.3 Relational Operators .15
4.4 Bitwise Operators .16
4.5 Assignment .16
4.6 Mnemonics .16
4.7 Constants .16
5 Method of Describing Bitstream Syntax .16
6 Syntax .18
6.1 Audio Data Interchange Format, ADIF.18
6.2 Audio Data Transport Stream, ADTS.19
6.3 Raw Data.21
7 Profiles and Profile Interoperability.33
7.1 Profiles.33
7.2 Profile Interoperability.35
8 Overall Data Structure.36
8.1 AAC Interchange Formats .36
8.2 Raw Data.41
8.3 Single Channel Element (SCE), Channel Pair Element (CPE) and Individual Channel
Stream (ICS) .45
8.4 Low Frequency Enhancement Channel (LFE) .51
8.5 Program Config Element (PCE).51
8.6 Data Stream Element (DSE) .56
8.7 Fill Element (FIL).56
8.8 Extension Payload.57
8.9 Tables.61
8.10 Figures .70
9 Noiseless Coding.70
9.1 Tool Description.70
9.2 Definitions .71
9.3 Decoding Process.73
9.4 Tables.76
10 Quantization .76
10.1 Tool Description.76
10.2 Definitions .76
10.3 Decoding Process.76
11 Scalefactors.77
11.1 Tool Description.77
© ISO/IEC 2006 – All rights reserved iii

11.2 Definitions. 77
11.3 Decoding Process. 78
12 Joint Coding . 79
12.1 M/S Stereo. 79
12.2 Intensity Stereo . 80
12.3 Coupling Channel . 82
13 Prediction. 86
13.1 Tool Description. 86
13.2 Definitions. 86
13.3 Decoding Process. 87
13.4 Diagrams. 93
14 Temporal Noise Shaping (TNS) . 93
14.1 Tool Description. 93
14.2 Definitions. 94
14.3 Decoding Process. 94
15 Filterbank and Block Switching. 96
15.1 Tool Description. 96
15.2 Definitions. 96
15.3 Decoding Process. 97
16 Gain Control. 101
16.1 Tool Description. 101
16.2 Definitions. 102
16.3 Decoding Process. 102
16.4 Diagrams. 109
16.5 Tables. 109
Annex A (normative) Huffman Codebook Tables. 111
Annex B (informative) Information on Unused Codebooks . 130
Annex C (informative) Encoder . 131
Annex D (informative) Patent Holders . 189
Annex E (informative) Registration Procedure. 190
Annex F (informative) Registration Application Form . 192
Annex G (informative) Registration Authority . 193
Bibliography . 194

iv © ISO/IEC 2006 – All rights reserved

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 13818-7 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 13818-7:2004), which has been technically
revised. It also incorporates the Technical Corrigendum ISO/IEC 13818-7:2004/Cor.1:2005.
ISO/IEC 13818 consists of the following parts, under the general title Information technology — Generic
coding of moving pictures and associated audio information:
— Part 1: Systems
— Part 2: Video
— Part 3: Audio
— Part 4: Conformance testing
— Part 5: Software simulation [Technical Report]
— Part 6: Extensions for DSM-CC
— Part 7: Advanced Audio Coding (AAC)
— Part 9: Extension for real time interface for systems decoders
— Part 10: Conformance extensions for Digital Storage Media Command and Control (DSM-CC)
— Part 11: IPMP on MPEG-2 systems

© ISO/IEC 2006 – All rights reserved v

Introduction
The standardization body ISO/IEC JTC 1/SC 29/WG 11, also known as the Moving Pictures Experts Group
(MPEG), was established in 1988 to specify digital video and audio coding schemes at low data rates. MPEG
completed its first phase of audio specifications (MPEG-1) in November 1992, ISO/IEC 11172-3. In its second
phase of development, the MPEG Audio subgroup defined a multichannel extension to MPEG-1 audio that is
backwards compatible with existing MPEG-1 systems (MPEG-2 BC) and defined an audio coding standard at
lower sampling frequencies than MPEG-1, ISO/IEC 13818-3.
The International Organization for Standardization (ISO) and International Electrotechnical Commission (IEC)
draw attention to the fact that it is claimed that compliance with this document may involve the use of patents.
The ISO and IEC take no position concerning the evidence, validity and scope of this patent right.
The holder of this patent right has assured the ISO and IEC that he is willing to negotiate licences under
reasonable and non-discriminatory terms and conditions with applicants throughout the world. In this respect,
the statement of the holder of this patent right is registered with the ISO and IEC. Information may be obtained
from the companies listed in Annex D.
Attention is drawn to the possibility that some of the elements of this document may be the subject of patent
rights other than those identified in Annex D. ISO and IEC shall not be held responsible for identifying any or
all such patent rights.
vi © ISO/IEC 2006 – All rights reserved

INTERNATIONAL STANDARD ISO/IEC 13818-7:2006(E)

Information technology — Generic coding of moving pictures
and associated audio information —
Part 7:
Advanced Audio Coding (AAC)
1 Scope
1.1 General
This International Standard describes the MPEG-2 audio non-backwards compatible standard called MPEG-2
Advanced Audio Coding, AAC [1], a higher quality multichannel standard than achievable while requiring
MPEG-1 backwards compatibility. This MPEG-2 AAC audio standard allows for ITU-R “indistinguishable”
quality according to [2] at data rates of 320 kbit/s for five full-bandwidth channel audio signals.
The AAC decoding process makes use of a number of required tools and a number of optional tools. Table 1
lists the tools and their status as required or optional. Required tools are mandatory in any possible profile.
Optional tools may not be required in some profiles.
Table 1 — AAC decoder tools
Tool Name Required / Optional
Bitstream Formatter Required
Noiseless Decoding Required
Inverse quantization Required
Rescaling Required
M/S Optional
Prediction Optional
Intensity Optional
Dependently switched coupling Optional
TNS Optional
Filterbank / block switching Required
Gain control Optional
Independently switched coupling Optional

1.2 MPEG-2 AAC Tools Overview
The basic structure of the MPEG-2 AAC system is shown in Figure 1 and Figure 2. As is shown in Table 1,
there are both required and optional tools in the decoder. The data flow in this diagram is from left to right, top
to bottom. The functions of the decoder are to find the description of the quantized audio spectra in the
bitstream, decode the quantized values and other reconstruction information, reconstruct the quantized
spectra, process the reconstructed spectra through whatever tools are active in the bitstream in order to arrive
at the actual signal spectra as described by the input bitstream, and finally convert the frequency domain
spectra to the time domain, with or without an optional gain control tool. Following the initial reconstruction and
scaling of the spectrum reconstruction, there are many optional tools that modify one or more of the spectra in
order to provide more efficient coding. For each of the optional tools that operate in the spectral domain, the
option to “pass through” is retained, and in all cases where a spectral operation is omitted, the spectra at its
input are passed directly through the tool without modification.
© ISO/IEC 2006 – All rights reserved 1

The input to the bitstream demultiplexer tool is the MPEG-2 AAC bitstream. The demultiplexer separates the
parts of the MPEG-AAC data stream into the parts for each tool, and provides each of the tools with the
bitstream information related to that tool.
The outputs from the bitstream demultiplexer tool are:
• The sectioning information for the noiselessly coded spectra,
• The noiselessly coded spectra,
• The M/S decision information (optional),
• The predictor state information (optional),
• The intensity stereo control information and coupling channel control information (both optional),
• The temporal noise shaping (TNS) information (optional),
• The filterbank control information, and
• The gain control information (optional).
The noiseless decoding tool takes information from the bitstream demultiplexer, parses that information,
decodes the Huffman coded data, and reconstructs the quantized spectra and the Huffman and DPCM coded
scalefactors.
The inputs to the noiseless decoding tool are:
• The sectioning information for the noiselessly coded spectra, and
• The noiselessly coded spectra.
The outputs of the Noiseless Decoding tool are:
• The decoded integer representation of the scalefactors, and
• The quantized values for the spectra.
The inverse quantizer tool takes the quantized values for the spectra, and converts the integer values to the
non-scaled, reconstructed spectra. This quantizer is a non-uniform quantizer.
The input to the Inverse Quantizer tool is:
• The quantized values for the spectra.
The output of the inverse quantizer tool is:
• The un-scaled, inversely quantized spectra.
The rescaling tool converts the integer representation of the scalefactors to the actual values, and multiplies
the un-scaled inversely quantized spectra by the relevant scalefactors.
The inputs to the rescaling tool are:
• The decoded integer representation of the scalefactors, and
• The un-scaled, inversely quantized spectra.
The output from the scalefactors tool is:
• The scaled, inversely quantized spectra.
2 © ISO/IEC 2006 – All rights reserved

The M/S tool converts spectra pairs from Mid/Side to Left/Right under control of the M/S decision information
in order to improve coding efficiency.
The inputs to the M/S tool are:
• The M/S decision information, and
• The scaled, inversely quantized spectra related to pairs of channels.
The output from the M/S tool is:
• The scaled, inversely quantized spectra related to pairs of channels, after M/S decoding.
Note The scaled, inversely quantized spectra of individually coded channels are not processed by the M/S block, rather
they are passed directly through the block without modification. If the M/S block is not active, all spectra are passed
through this block unmodified.
The prediction tool reverses the prediction process carried out at the encoder. This prediction process re-
inserts the redundancy that was extracted by the prediction tool at the encoder, under the control of the
predictor state information. This tool is implemented as a second order backward adaptive predictor. The
inputs to the prediction tool are:
• The predictor state information, and
• The scaled, inversely quantized spectra.
The output from the prediction tool is:
• The scaled, inversely quantized spectra, after prediction is applied.
Note If the prediction is disabled, the scaled, inversely quantized spectra are passed directly through the block without
modification.
The intensity stereo tool implements intensity stereo decoding on pairs of spectra.
The inputs to the intensity stereo tool are:
• The inversely quantized spectra, and
• The intensity stereo control information.
The output from the intensity stereo tool is:
• The inversely quantized spectra after intensity channel decoding.
Note The scaled, inversely quantized spectra of individually coded channels are passed directly through this tool without
modification, if intensity stereo is not indicated. The intensity stereo tool and M/S tool are arranged so that the operation of
M/S and intensity stereo are mutually exclusive on any given scalefactor band and group of one pair of spectra.
The coupling tool for dependently switched coupling channels adds the relevant data from dependently
switched coupling channels to the spectra, as directed by the coupling control information.
The inputs to the coupling tool are:
• The inversely quantized spectra, and
• The coupling control information.
The output from the coupling tool is:
• The inversely quantized spectra coupled with the dependently switched coupling channels.
© ISO/IEC 2006 – All rights reserved 3

Note The scaled, inversely quantized spectra are passed directly through this tool without modification, if coupling is not
indicated. Depending on the coupling control information, dependently switched coupling channels might either be coupled
before or after the TNS processing.
The coupling tool for independently switched coupling channels adds the relevant data from independently
switched coupling channels to the time signal, as directed by the coupling control information.
The inputs to the coupling tool are:
• The time signal as output by the filterbank, and
• The coupling control information.
The output from the coupling tool is:
• The time signal coupled with the independently switched coupling channels.
Note The time signal is passed directly through this tool without modification, if coupling is not indicated.
The temporal noise shaping (TNS) tool implements a control of the fine time structure of the coding noise. In
the encoder, the TNS process has flattened the temporal envelope of the signal to which it has been applied.
In the decoder, the inverse process is used to restore the actual temporal envelope(s), under control of the
TNS information. This is done by applying a filtering process to parts of the spectral data.
The inputs to the TNS tool are:
• The inversely quantized spectra, and
• The TNS information.
The output from the TNS block is:
• The inversely quantized spectra.
Note If this block is disabled, the inversely quantized spectra are passed through without modification.
The filterbank / block switching tool applies the inverse of the frequency mapping that was carried out in the
encoder. An inverse modified discrete cosine transform (IMDCT) is used for the filterbank tool. The IMDCT
can be configured to support either one set of 128 or 1024, or four sets of 32 or 256 spectral coefficients.
The inputs to the filterbank tool are:
• The inversely quantized spectra, and
• The filterbank control information.
The output(s) from the filterbank tool is (are):
• The time domain reconstructed audio signal(s).
When present, the gain control tool applies a separate time domain gain control to each of four frequency
bands that have been created by the gain control PQF filterbank in the encoder. Then, it assembles four
frequency bands and reconstructs the time waveform through the gain control tool’s filterbank.
The inputs to the gain control tool are:
• The time domain reconstructed audio signal(s), and
• The gain control information.
4 © ISO/IEC 2006 – All rights reserved

The output(s) from the gain control tool is (are):
• The time domain reconstructed audio signal(s).
If the gain control tool is not active, the time domain reconstructed audio signal(s) are passed directly from the
filterbank tool to the output of the decoder. This tool is used for the scalable sampling rate (SSR) profile only.

input time signal
Legend:
data
control
AAC
gain control
psychoacoustic
model
window length block
decision switching
filterbank
threshold
TNS
calculation
coded
audio
intensity
stream
bitstream
formatter
spectral
prediction
processing
M/S
scaling
quantization
quantization
and noiseless
coding
Huffman coding
Figure 1 — MPEG-2 AAC Encoder Block Diagram

© ISO/IEC 2006 – All rights reserved 5

Legend:
data
control
Huffman decoding
noiseless
inverse
decoding and
quantization
inverse
quantization
rescaling
M/S
bitstream
deformatter
prediction
coded
audio
intensity
stream
spectral
dependently
processing
switched
coupling
TNS
dependently
switched
coupling
block
switching
filterbank
AAC
gain control
output
time
signal
independently
switched
coupling
Figure 2 — MPEG-2 AAC Decoder Block Diagram

6 © ISO/IEC 2006 – All rights reserved

2 Normative References
The following referenced documents are indispensable for the application of this document. For dated
references, only the edition cited applies. For undated references, the latest edition of the referenced
document (including any amendments) applies.
ISO/IEC 11172-3: Information technology — Coding of moving pictures and associated audio for digital
storage media at up to about 1,5 Mbit/s — Part 3: Audio
ISO/IEC 13818-1: Information technology — Generic coding of moving pictures and associated audio
information — Part 1: Systems
ISO/IEC 13818-3: Information technology — Generic coding of moving pictures and associated audio
information — Part 3: Audio
ISO/IEC 14496-3: Information technology — Coding of audio-visual objects — Part 3: Audio
3 Terms and Definitions
For the purposes of this part of ISO/IEC 13818, the following definitions apply.
3.1
access unit
in the case of compressed audio, an audio access unit
3.2
alias
mirrored signal component resulting from sampling
3.3
analysis filterbank
filterbank in the encoder that transforms a broadband PCM audio signal into a set of spectral coefficients
3.4
ancillary data
part of the bitstream that might be used for transmission of ancillary data
3.5
audio access unit
for AAC, the smallest part of the encoded bitstream which can be decoded by itself, where decoded means
"fully reconstructed sound"
NOTE Typically, this is a segment of the encoded bitstream starting after the end of the byte containing the last bit of
one ID_END id_syn_ele() through the end of the byte containing the last bit of the next ID_END id_syn_ele.
3.6
audio buffer
buffer in the system target decoder (see ISO/IEC 13818-1) for storage of compressed audio data
3.7
bark
standard unit corresponding to one critical band width of human hearing
3.8
backward compatibility
newer coding standard is backward compatible with an older coding standard if decoders designed to operate
with the older coding standard are able to continue to operate by decoding all or part of a bitstream produced
according to the newer coding standard
© ISO/IEC 2006 – All rights reserved 7

3.9
bitrate
rate at which the compressed bitstream is delivered to the input of a decoder
3.10
bitstream
stream
ordered series of bits that forms the coded representation of the data
3.11
bitstream verifier
process by which it is possible to test and verify that all the requirements specified in this part of
ISO/IEC 13818 are met by the bitstream
3.12
block companding
normalizing of the digital representation of an audio signal within a certain time period
3.13
byte aligned
bit in a coded bitstream is byte-aligned if its position is a multiple of 8-bits from either the first bit in the stream
for the Audio Data Interchange Format (see 6.1) or the first bit in the syncword for the Audio Data Transport
Stream Format (see 6.2)
3.14
byte
sequence of 8 bits
3.15
centre channel
audio presentation channel used to stabilize the central component of the frontal stereo image
3.16
channel
sequence of data representing an audio signal intended to be reproduced at one listening position
3.17
coded audio bitstream
coded representation of an audio signal
3.18
coded representation
data element as represented in its encoded form
3.19
compression
reduction in the number of bits used to represent an item of data
3.20
constant bitrate
operation in which the bitrate is constant from start to finish of the coded bitstream
3.21
CRC
Cyclic Redundancy Check to verify the correctness of data
8 © ISO/IEC 2006 – All rights reserved

3.22
critical band
unit of bandwidth which represents the standard unit of bandwidth expressed in human auditory terms,
corresponding to a fixed length on the human cochlea, approximately equal to 100 Hz at low frequencies and
1/3 octave at higher frequencies, above approximately 700 Hz
3.23
data element
item of data as represented before encoding and after decoding
3.24
decoded stream
decoded reconstruction of a compressed bitstream
3.25
decoder
embodiment of a decoding process
3.26
decoding (process)
process defined in this part of ISO/IEC 13818 that reads an input coded bitstream and outputs decoded audio
samples
3.27
digital storage media
DSM
digital storage or transmission device or system
3.28
discrete cosine transform
DCT
either the forward discrete cosine transform or the inverse discrete cosine transform, an invertible, discrete
orthogonal transformation
3.29
downmix
matrixing of n channels to obtain less than n channels
3.30
editing
process by which one or more coded bitstreams are manipulated to produce a new coded bitstream
NOTE Conforming edited bitstreams are defined in this part of ISO/IEC 13818.
3.31
encoder
embodiment of an encoding process
3.32
encoding (process)
process, not specified in ISO/IEC 13818, that reads a stream of input audio samples and produces a valid
coded bitstream as defined in this part of ISO/IEC 13818
3.33
entropy coding
variable length lossless coding of the digital representation of a signal to reduce statistical redundancy
© ISO/IEC 2006 – All rights reserved 9

3.34
Fast Fourier Transformation
FFT
fast algorithm for performing a discrete Fourier transform (an orthogonal transform)
3.35
filterbank
set of band-pass filters covering the entire audio frequency range
3.36
flag
variable which can take one of only the two values defined in this part of ISO/IEC 13818
3.37
forward compatibility
a newer coding standard is forward compatible with an older coding standard if decoders designed to operate
with the newer coding standard are able to decode bitstreams of the older coding standard
3.38
frame
part of the audio signal that corresponds to audio PCM samples from an audio access unit
3.39
Fs
sampling frequency
3.40
Hann window
time function applied sample-by-sample to a block of audio samples before Fourier transformation
3.41
Huffman coding
specific method for entropy coding
3.42
hybrid filterbank
serial combination of subband filterbank and MDCT
3.43
IDCT
Inverse Discrete Cosine Transform
3.44
IMDCT
Inverse Modified Discrete Cosine Transform
3.45
intensity stereo
method of exploiting stereo irrelevance or redundancy in stereophonic audio programmes based on retaining
at high frequencies only the energy envelope of the right and left channels
3.46
joint stereo coding
any method that exploits stereophonic irrelevance or stereophonic redundancy
3.47
joint stereo mode
mode of the audio coding algorithm using joint stereo coding
10 © ISO/IEC 2006 – All rights reserved

3.48
low frequency enhancement (LFE) channel
limited bandwidth channel for low frequency audio effects in a multichannel system
3.49
main audio channels
all channels represented by either single_channel_element()'s (see 8.2.1) or channel_pair_element()´s
(see 8.2.1)
3.50
mapping
conversion of an audio signal from time to frequency domain by subband filtering and/or by MDCT
3.51
masking
property of the human auditory system by which an audio signal cannot be perceived in the presence of
another audio signal
3.52
masking threshold
function in frequency and time below which an audio signal cannot be perceived by the human auditory
system
3.53
modified discrete cosine transform
MDCT
transform which has the property of time domain aliasing cancellation
NOTE An analytical espression for the MDCT can be found in C.3.1.2.
3.54
M/S stereo
method of removing imaging artefacts as well as exploiting stereo irrelevance or redundancy in stereophonic
audio programmes based on coding the sum and difference signal instead of the left and right channels
3.55
multichannel
combination of audio channels used to create a spatial sound field
3.56
multilingual
presentation of dialogue in more than one language
3.57
non-tonal component
noise-like component of an audio signal
3.58
Number of Considered Channels
NCC
number of channels represented by the elements SCE, independently switched CCE and CPE, i.e. once the
number of SCEs plus once the number of independently switched CCEs plus twice the number of CPEs, with
respect to the naming conventions of the MPEG-AAC decoders and bitstreams, NCC=A+I
NOTE This number is used to derive the required decoder input buffer size (see 8.2.3).
3.59
Nyquist sampling
sampling at or above twice the maximum bandwidth of a signal
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3.60
padding
method to adjust the average length of an audio frame in time to the duration of the corresponding PCM
samples, by conditionally adding a slot to the audio frame
3.61
parameter
variable within the syntax of this specification which may take one of a range of values. A variable which can
take one of only two values is a flag or indicator and not a parameter
3.62
parser
functional stage of a decoder which extracts from a coded bitstream a series of bits representing coded
elements
3.63
polyphase filterbank
set of equal bandwidth filters with special phase interrelationships, allowing for an efficient implementation of
the filterbank
3.64
prediction error
difference between the actual value of a sample or data element and its predictor
3.65
prediction
use of a predictor to provide an estimate of the sample value or data element currently being decoded
3.66
predictor
linear combination of previously decoded sample values or data elements
3.67
presentation channel
audio channel at the output of the decoder
3.68
presentation unit
in the case of compressed audio, a decoded audio access unit
3.69
program
set of main audio channels, coupling_channel_element()'s (see 8.2.1), lfe_channel_element()'s (see 8.2.1),
and associated data streams intended to be decoded and played back simultaneously
NOTE A program may be defined by default (see 8.5.3.1 and 8.5.3.3) or specifically by a program_config_element()
(see 8.5.3.2). A given single_channel_element() (see 8.2.1), channel_pair_element() (see 8.2.1),
coupling_channel_element(), lfe_channel_element() or data channel may accompany one or more programs in any given
bitstream.
3.70
psychoacoustic model
mathematical model of the masking behaviour of the human auditory system
3.71
random access
process of beginning to read and decode the coded bitstream at an arbitrary point
12 © ISO/IEC 2006 – All rights reserved

3.72
reserved
when used in the clauses defining the coded bitstream, indicates that the value may be used in the future for
ISO/IEC defined extensions
3.73
sampling frequency
Fs
rate in Hertz which is used to digitize an audio signal during the sampling process
3.74
scalefactor
factor by which a set of values is scaled before quantization
3.75
scalefactor band
set of spectral coefficients which are scaled by one scalefactor
3.76
scalefactor index
numerical code for a scalefactor
3.77
side information
information in the bitstream necessary for controlling the decoder
3.78
spectral coefficients
discrete frequency domain data output from the analysis filterbank
3.79
spreading function
function that describes the frequency spread of masking effects
3.80
stereo-irrelevant
portion of a stereophonic audio signal which does not contribute to spatial perception
3.81
stuffing (bits)
stuffing (bytes)
code words that may be inserted at particular locations in the coded bitstream that are discarded in the
decoding process whose purpose is to increase the bitrate of the stream which would otherwise be lower than
the desired bitrate
3.82
surround channel
audio presentation channel added to the front channels (L and R or L, R, and C) to enhance the spatial
perception
3.83
Syncword
a 12-bit code embedded in the audio bitstream that identifies the start of a adts_frame() (see 6.2, Table 5)
3.84
synthesis filterbank
filterbank in the decoder that reconstructs a PCM audio signal from subband samples
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3.85
tonal component
sinusoid-like component of an audio signal
3.86
variable bitrate
operation in which the bitrate varies with time during the decoding of a coded bitstream
3.87
variable length coding
reversible procedure for coding that assigns shorter code words to frequent symbols and longer code words to
less frequent symbols
3.88
variable length code
VLC
code word assigned by variable length encoder (see variable length coding)
3.89
variable length decoder
procedure to obtain the symbols encoded with a variable length coding technique
3.90
variable length encoder
procedure to assign variable length codewords to symbols
4 Symbols and Abbreviations
The mathematical operators used to describe this International Standard are similar to those used in the C
programming language. However, integer division with truncation and rounding are specifically defined. The
bitwise operators are defined assuming twos-complement representation of integers. Numbering and counting
loops generally begin from zero.
4.1 Arithmetic Operators
+ Addition.
− Subtraction (as a binary operator) or negation (as a unary operator).
++ Increment.
− − Decrement.
* Multiplication.
^ Power.
/ Integer division with truncation of the result toward zero. For example, 7/4 and −7/−4 are truncated
to 1 and −7/4 and 7/−4 are truncated to −1.
// Integer division with rounding to the nearest integer. Half-integer values are rounded away from zero
unless otherwise specified. For example 3//2 is rounded to 2, and −3//2 is rounded to −2.
DIV Integer division with truncation of the result towards −∞.
14 © ISO/IEC 2006 – All rights reserved

| | Absolute value. | x | = x when x > 0
| x | = 0 when x == 0
| x | = −x when x < 0
% Modulus operator. Defined only for positive numbers.
Sign( ) Sign.
Sign(x) = 1 when x > 0
Sign(x) = 0 when x == 0
Sign(x) = −1 when x < 0
INT ( ) Truncation to integer operator. Returns the integer part of the real-valued argument.
NINT ( ) Nearest integer operator. Returns the nearest integer value to the real-valued argument. Half-integer
values are rounded away from zero.
sin Sine.
cos Cosine.
exp Exponential.
√ Square root.
log Logarithm to base ten.
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