ISO/IEC 13818-7: MPEG-2 Audio Advanced Audio Coding标准详解

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"ISO/IEC 13818-7是MPEG-2音频AAC标准，用于高级音频编码（AAC）" ISO/IEC 13818-7，通常称为MPEG-2 Audio Advanced Audio Coding (AAC)，是国际标准化组织（ISO）和国际电工委员会（IEC）联合制定的一项音频编码标准。该标准是MPEG-2的一部分，旨在提供高效、高质量的音频压缩技术，适用于数字电视、DVD、流媒体服务、移动通信以及各种其他音频应用。 AAC是一种有损音频压缩格式，其目标是在保持音质的同时，尽可能地减小音频文件的大小。与早期的音频压缩格式如MPEG-1 Audio Layer 3（通常称为MP3）相比，AAC在相同的数据率下可以提供更高质量的音频体验，这主要得益于其更复杂的编码技术。它采用了更多的频带划分、多声道支持、感知量化和熵编码等高级特性。该标准的第三版发布于2004年10月15日，即ISO/IEC 13818-7:2004(E)，其中包含了一系列的技术改进和优化。AAC编码技术主要包括以下核心组件： 1. **频带划分和窗口函数**：AAC使用了多种窗口函数（如长窗和短窗）对音频信号进行分段处理，以便更有效地分析和编码音频的瞬时和持续特性。 2. **感知量化**：根据人耳的听觉特性进行量化，忽略人耳不易察觉的音频频率成分，进一步节省存储空间。 3. **自适应预测**：通过对未来的音频样本进行预测，减少需要编码的信息量。 4. **多声道编码**：支持立体声、环绕声和更多声道的编码，为用户提供丰富的立体声效果和多维度的听觉体验。 5. **感知熵编码**：使用如上下文自适应二进制算术编码（CABAC）或上下文自适应变长编码（CAVLC）等高效的熵编码方法，将量化后的音频数据编码为更紧凑的形式。 6. **噪声隐藏**：对于编码过程中丢失的部分，使用噪声隐藏技术来减少失真感。 7. **扩展功能**：AAC还支持如音频对象类型、采样率转换、元数据嵌入等扩展特性，允许在不同的应用场景中灵活使用。 MPEG-2 Audio AAC标准的广泛采用，尤其是在移动设备和流媒体服务中，证明了其在音频压缩领域的先进性和效率。随着技术的不断发展，AAC标准也经历了多次更新，例如AAC+和HE-AAC等更高效的变种，它们在低数据速率环境下依然能提供高质量的音频体验。

ISO/IEC 13818-7:2004(E)

3.21

CRC

the Cyclic Redundancy Check to verify the correctness of data.

3.22

critical band

this unit of bandwidth represents the standard unit of bandwidth expressed in human

auditory terms, corresponding to a fixed length on the human cochlea. It is approximately

equal to 100 Hz at low frequencies and 1/3 octave at higher frequencies, above

approximately 700 Hz.

3.23

data element

an item of data as represented before encoding and after decoding.

3.24

decoded stream

the decoded reconstruction of a compressed bitstream.

3.25

decoder

an embodiment of a decoding process.

3.26

decoding (process)

the 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

a 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. The

DCT is an invertible, discrete orthogonal transformation.

3.29

downmix

a matrixing of n channels to obtain less than n channels.

3.30

editing

the process by which one or more coded bitstreams are manipulated to produce a new

coded bitstream. Conforming edited bitstreams must meet the requirements defined in this

part of ISO/IEC 13818.

3.31

encoder

an embodiment of an encoding process.

3.32

encoding (process)

a 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.

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ISO/IEC 13818-7:2004(E)

3.45

intensity stereo

a 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

a mode of the audio coding algorithm using joint stereo coding.

3.48

low frequency enhancement (LFE) channel

a 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 subclause 8.2.1) or

channel_pair_element()´s (see subclause 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

a 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

a 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)

a transform which has the property of time domain aliasing cancellation. An analytical

espression for the MDCT can be found in subclause C.3.1.2.

3.54

M/S stereo

a 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

a combination of audio channels used to create a spatial sound field.

3.56

Multilingual

a presentation of dialogue in more than one language.

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ISO/IEC 13818-7:2004(E)

3.57

non-tonal component

a noise-like component of an audio signal.

3.58

NCC

Number of Considered Channels. The 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. This

number is used to derive the required decoder input buffer size (see subclause 8.2.2).

3.59

Nyquist sampling

sampling at or above twice the maximum bandwidth of a signal.

3.60

Padding

a 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

a 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

a set of equal bandwidth filters with special phase interrelationships, allowing for an efficient

implementation of the filterbank.

3.64

prediction error

the difference between the actual value of a sample or data element and its predictor.

3.65

Prediction

the use of a predictor to provide an estimate of the sample value or data element currently

being decoded.

3.66

predictor

a linear combination of previously decoded sample values or data elements.

3.67

presentation channel

an audio channel at the output of the decoder.

3.68

presentation unit

In the case of compressed audio a decoded audio access unit.

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ISO/IEC 13818-7:2004(E)

3.69

Program

a set of main audio channels, coupling_channel_element()'s (see subclause 8.2.1),

lfe_channel_element()'s (see subclause 8.2.1), and associated data streams intended to be

decoded and played back simultaneously. A program may be defined by default (see

subclause 8.5.2.1 and subclause 8.5.2.3) or specifically by a program_config_element()

(see subclauses 8.5.2.2). A given single_channel_element() (see subclause 8.2.1),

channel_pair_element() (see subclause 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

a mathematical model of the masking behaviour of the human auditory system.

3.71

random access

the process of beginning to read and decode the coded bitstream at an arbitrary point.

3.72

Reserved

the term "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)

defines the rate in Hertz which is used to digitise 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

a set of spectral coefficients which are scaled by one scalefactor.

3.76

scalefactor index

a 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

a function that describes the frequency spread of masking effects.

3.80

stereo-irrelevant

a portion of a stereophonic audio signal which does not contribute to spatial perception.

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