ISO/IEC13818-7:2003(E) - MPEG-2音频视频压缩标准

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"ACC 2协议，全称为Advanced Audio Coding第二版，是音频编码标准，与MPEG-2音频标准相关。此协议是ISO/IEC13818-7:2003的一部分，由国际标准化组织ISO/IEC的JTC1/SC29/WG11分委员会制定，主要关注于音频和视频的压缩编码技术。" **ACC 2协议详解** Advanced Audio Coding (AAC) 是一种高效能的有损音频压缩格式，用于音频数据的压缩，尤其适用于数字音频广播、互联网流媒体和存储应用。在AAC的第二版中，它被纳入了MPEG-2标准体系，因此也被称为MPEG-2 AAC。这个标准的发布日期为2003年，由Ralph Sperschneider作为主要作者。 AAC的核心目标是在保持高质量音频的同时，实现比传统编码方式更高的压缩比率。这主要是通过使用更复杂的信号处理技术，如多频带激励编码（MBE）、感知量化、自适应预测和噪声掩蔽来实现的。这些技术使得音频信号中的冗余信息能够被有效去除，从而在有限的带宽内传输更多的信息。 **AAC的主要特点和优势** 1. **频谱效率**：AAC能够在较低的数据速率下提供与较高质量的MP3或其他音频格式相当的音质。 2. **多声道支持**：AAC支持多达5.1声道的环绕声，为用户提供沉浸式的音频体验。 3. **比特率自适应**：根据网络条件或存储空间，AAC可以动态调整编码质量，确保流畅的播放体验。 4. **感知编码**：利用人类听觉系统的特性，对人耳难以察觉的频率进行更深度的压缩。 5. **元数据支持**：AAC编码允许包含诸如艺术家信息、专辑名等元数据，方便用户管理和播放。 **与MPEG-2音频的关联** MPEG-2音频是MPEG-2标准的一部分，该标准主要用于视频编码。MPEG-2音频包含了多个层，其中AAC是MPEG-2音频标准的第7部分，提供了更先进的音频编码技术，相比早期的MPEG-1 Layer 3（即MP3）有显著的性能提升。 **应用场景** 1. **数字电视**：许多数字电视广播系统使用AAC进行音频编码，因为它能提供高质量的音频并节省带宽。 2. **移动设备**：由于其高效的压缩性能，AAC广泛应用于智能手机和平板电脑，例如在Apple的iOS设备中，iTunes Store提供的音乐文件大多采用AAC编码。 3. **流媒体服务**：Netflix、YouTube等在线视频平台也采用AAC进行音频编码，以保证在网络状况不佳时仍能提供良好的听觉体验。 ACC 2协议作为MPEG-2音频标准的一部分，为音频压缩领域带来了显著的技术进步，提高了音频质量和压缩效率，使其在各种应用场景中得到了广泛应用。

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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