MPEG-2-AAC标准文档：ISO/IEC13818-7:2003(E)

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"MPEG-2-AAC标准文档，ISO/IEC13818-7:2003(E)版本，由ISO/IEC JTC1/SC29/WG11 MPEG组织编订，是音频编码的第二版标准，由Ralph Sperschneider撰写。" MPEG-2 Advanced Audio Coding (AAC) 是一种高效的数字音频压缩标准，广泛应用于电视广播、数字音频广播、互联网流媒体和各种多媒体设备中。这个标准是基于MPEG-2的一部分，即ISO/IEC13818-7，该文档的2003年版本（第二版）详细阐述了MPEG-2 AAC的规范和技术细节。 MPEG-2 AAC的开发旨在提供比MPEG-1 Audio Layer III（通常称为MP3）更高的音质，同时保持相似的比特率。它采用了更先进的编码技术，包括多声道支持、感知编码、自适应频域量化以及噪声隐藏等，使得在有限的带宽下可以传输更高质量的音频。标准中的关键特性包括： 1. **感知编码**：MPEG-2 AAC利用人类听觉系统的特性，只保留人耳能察觉到的声音信息，忽略掉那些难以分辨的部分，从而实现高效压缩。 2. **自适应频域量化**：音频信号被转换到频域后，根据信号的特性进行量化，这种自适应性能够优化压缩效率。 3. **多声道支持**：支持立体声和环绕声等多种声道配置，如5.1声道和7.1声道，为用户提供沉浸式的音频体验。 4. **噪声隐藏**：在丢失数据或存在错误的情况下，通过分析邻近样本，隐藏或修复音频中的噪声或失真。 5. **位流优化**：允许在不同比特率之间平滑过渡，适应不同的网络条件和存储需求。 ISO/IEC13818-7:2003(E)文档详细定义了编码和解码过程，以及如何构建和解析MPEG-2 AAC的位流。此外，它还涵盖了错误检测和恢复机制，确保在网络不稳定时仍能播放音频。由于版权原因，完整的技术细节未在摘要中提供。但通过这份标准文档，开发者、工程师和研究人员可以深入了解MPEG-2 AAC的工作原理，从而在音频编码、解码和多媒体应用开发中实现这一标准。对于需要处理音频编码或与MPEG-2 AAC相关的任何工作的人来说，这份文档是至关重要的参考资料。

ISO/IEC 13818-7:2003(E)

3.20 constant bitrate: Operation where the bitrate is constant from start to finish of the coded bitstream.

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.

3.33 entropy coding: Variable length lossless coding of the digital representation of a signal to reduce

statistical redundancy.

3.34 FFT: Fast Fourier Transformation. A fast algorithm for performing a discrete Fourier transform (an

orthogonal transform).

3.35 filterbank: A set of band-pass filters covering the entire audio frequency range.

3.36 flag: A variable which can take one of only the two values defined in this specification.

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: A 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: A time function applied sample-by-sample to a block of audio samples before

Fourier transformation.

3.41 Huffman coding: A specific method for entropy coding.

3.42 hybrid filterbank: A 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: 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.

ISO/IEC 13818-7:2003(E)

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_elements (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.

3.57 non-tonal component: A noise-like component of an audio signal.

3.58NCC: 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.

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 8.5.1) or specifically by

a program_config_element() (see subclause 8.2.1). A given single_channel_element() (see

ISO/IEC 13818-7:2003(E)

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.

3.78spectral 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.

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. Their purpose is to increase the bitrate of the

stream which would otherwise be lower than the desired bitrate.

3.82 surround channel: An 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 subclase 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: A sinusoid-like component of an audio signal.

3.86 variable bitrate: Operation where the bitrate varies with time during the decoding of a coded

bitstream.

3.87 variable length coding: A 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): A code word assigned by variable length encoder (See variable length

coding)

3.89 variable length decoder: A procedure to obtain the symbols encoded with a variable length coding

technique.

3.90 variable length encoder: A 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.

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