ISO/IEC 13818-7:2006 情報技術—動画および関連するオーディオ情報の一般的なコーディング—パート7：Advanced Audio Coding（AAC） | ページ 5

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 1 to entry: 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

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

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 1 to entry: 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

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

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 1 to entry: 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 1 to entry: 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

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 1 to entry: 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

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

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