AES数字音频接口标准简介
AES67标准
AES67标准“AES67是一个开放的网络数字音频标准，它是基于IP网络架构，采用现有的IT网络协议，实现低时延、高性能的专业音频传输的互用性指导方针。
它的开放协议与可兼容性，极大地推动了数字音频技术基于IP网络架构之上的发展。
AES67标准产生的背景从广播电视音频系统的发展进程来看：从使用诸多线缆的模拟音频系统，到引入了“同步”概念的数字音频系统，再到TDM音频矩阵系统，进而到基于以太网实现多个工作间网络互联音频系统，接下来到基于IP架构下实现远程互通互联（工作间之间、不同地域之间）的网络音频技术，如今已形成基于IP架构下的多地点集中式分布系统。
咅频技术能力的演进，IT技术的发展，让整个音视频技术行业搭上了IT 技术的顺风车，音频行业也面临AoIP( Audio over Inter- net Protocol，互联网协议架构下的音频）时代的到来。
目前媒体网络联盟MNA (Media Networking Alliance)通过的网络协议有四个，Livewire、Ravenna、QLAN、Dante。
这四个协议分别对应着业内用的最好的四个厂家，The Telos代表Livewire，Lawo 代表Ravenna，QSC代表QLAN，雅马哈代表Dante。
但是，不同网络协议间是互不兼容的，这对于用户来说非常麻烦。
因为用户大部分选择的不是某一个品牌，而是一个系统，这个系统里可能有很多不同设备，有的设备用这个协议，有的用那个，这些设备往往不能互通。
现在就是有这么一套互通的机制——AES67，能够把不同的协议联通在一起。
当前，Dante、Livewire、Ravenna、QLAN四个协议所覆盖的厂家已经达到90%以上，这意味着AES67标准能打通市场上90%的设备，并解决了用户最棘手的问题。
AES67标准的关键技术1、同步机制网络上任何地点的接收端通过一个公共时钟，可以与其他接收端同步回放，公共时钟可以保证所有流均被以相同的速率采样和还原，同一速率的多个流可以被轻易合成。
AES67标准
AES67标准“AES67是一个开放的网络数字音频标准，它是基于IP网络架构，采用现有的IT网络协议，实现低时延、高性能的专业音频传输的互用性指导方针。
它的开放协议与可兼容性，极大地推动了数字音频技术基于IP网络架构之上的发展。
AES67标准产生的背景从广播电视音频系统的发展进程来看：从使用诸多线缆的模拟音频系统，到引入了“同步”概念的数字音频系统，再到TDM音频矩阵系统，进而到基于以太网实现多个工作间网络互联音频系统，接下来到基于IP架构下实现远程互通互联（工作间之间、不同地域之间）的网络音频技术，如今已形成基于IP架构下的多地点集中式分布系统。
咅频技术能力的演进，IT技术的发展，让整个音视频技术行业搭上了IT 技术的顺风车，音频行业也面临AoIP( Audio over Inter- net Protocol，互联网协议架构下的音频）时代的到来。
目前媒体网络联盟MNA (Media Networking Alliance)通过的网络协议有四个，Livewire、Ravenna、QLAN、Dante。
这四个协议分别对应着业内用的最好的四个厂家，The Telos代表Livewire，Lawo 代表Ravenna，QSC代表QLAN，雅马哈代表Dante。
但是，不同网络协议间是互不兼容的，这对于用户来说非常麻烦。
因为用户大部分选择的不是某一个品牌，而是一个系统，这个系统里可能有很多不同设备，有的设备用这个协议，有的用那个，这些设备往往不能互通。
现在就是有这么一套互通的机制——AES67，能够把不同的协议联通在一起。
当前，Dante、Livewire、Ravenna、QLAN四个协议所覆盖的厂家已经达到90%以上，这意味着AES67标准能打通市场上90%的设备，并解决了用户最棘手的问题。
AES67标准的关键技术1、同步机制网络上任何地点的接收端通过一个公共时钟，可以与其他接收端同步回放，公共时钟可以保证所有流均被以相同的速率采样和还原，同一速率的多个流可以被轻易合成。
AES信号时钟失锁检测的初探及应用
AES信号时钟失锁检测的初探及应用萨日娜;赵岩【摘要】本文对AES信号的数据结构、编码方式、自同步原理及时钟失锁监测原理进行了剖析，并对其在内蒙古广播电视台广播系统中的应用进行了阐述。
【期刊名称】《数字传媒研究》【年(卷),期】2015(032)003【总页数】4页(P29-32)【关键词】AES数据结构;编码方式;校验位;音频四选一【作者】萨日娜;赵岩【作者单位】内蒙古广播电视台,内蒙古呼和浩特市010058【正文语种】中文【中图分类】G220数字化广播时代，对于种类繁多的音源设备和各种传输链路来说数字音频信号的标准在播出系统中显得更为重要。
在常见的数字音频标准SPDIF、AES/EBU、MADI 中，AES/EBU更为主要和常用。
AES即AES/EBU标准,是由音频工程师协会/欧洲广播联盟（Audio Engineering Society/European Broadcast Union）制定的一种通过基于单根数字双绞线来传输数字音频数据的串行位传输协议。
其中AES是指AES3-2003标准，EBU是指EBU发布的数字音频接口标准EBU-3250，两者内容在实质上是相同的，统称为AES/EBU数字音频接口。
因此，AES/EBU标准常简称为AES、AES3标注，同时其对应的ANSI（美国国家标准学会 AMERICAN NATIONAL STANDARDS INSTITUTE）标准为ANSIS4.40-1992， IEC（国际电工委员会International Electro technical Commission，简称IEC）标准为IEC-958。
当采用非平衡传输时该标准遵循AES-3id-2001，此时称为AES3id。
2.1 平衡模式AES3-2003标准中建议使用数字音频双绞线作为标准AES信号的传输方式，该方式为平衡传输模式，详细电气特性见表1。
2.2 非平衡模式随着系统的复杂性和多样性的日益加剧，对AES信号的传输距离有了更高的要求。
数字音频接口技术
2.数字音频接口数据结构
数字音频接口是一种单线单向串行传输数字音频数据的接口。它能够传输两个声道的数字音频数据、相关的控制信息，有一些检测错误能力。一个声音样本附带有一位控制信息，这些控制信息汇集成一个控制块。信号是双相调制的，可以从信号中恢复时钟。引导符用来区分块边界和每一个声音样本。
a）帧、子帧和块的结构音频数据放在一个被称为子帧（Sub-frames）有结构里。包含一个4位的引导符，4位的辅助数据，20位的音频数据。三位分别称为合法标记、用户位、通道状态。另外还有一个校验位。引导符标识子帧的开始。按照子帧的数据容量，音频样本最大可达到24位，这点非常令人激动，只要把时钟频率加倍，我们就能够通过数字音频接口传输24位96kHz的数字音频信号。在音频样本大于20位时，数据同时占据辅助和音频数据域。在音频样本小于等于20位时，数据存放在音频数据域中，辅导数据可用于其它应用方面，例如语音数据。音频数据在子帧中是最低有效位在先，最高有效位在后。合法标记位，表示此音频样本是正确的，没有包含错误，适合作数模转换。如果音频样本有错，例如CD唱片上的不可恢复的缺陷，CD-DSP无法进行纠错，它将把此音频样本标记为含有错误，留待以后的电路部分作适当的处理。用户位没有定义，可由用户随意使用。通道状态位包含有重要信息。通道状态块由192位组成，每个音频样本附带一位通道状态位，左右声道的通道状态块是独立的。每192帧通道状态块更新一次。校验位为偶校验位，可检出子帧中奇数个错。
c）专业用途、民用用途
3、媒介接口（Media Interface）
数字音频信号是通过特定的传输媒介传送的，常见的有同轴电缆、Toslink光纤、AT&T光纤、平衡线等。
数字音频协议介绍(DOC)
数字音频协议介绍目录AES/EBU (1)ADAT (6)I2S (7)时分复用（TDM） (9)MIDI (11)AES/EBU简介：AES3: Audio Engineering Society Standard #3EBU: European Broadcasting UnionAES3 接口在1985 年已经被指定并在1992 年正式成为标准接口。
自从定为标准后，AES3 反复更新和调整以适应先进设备的要求，其应用非常普遍。
但另一方面来说这使得它有点复杂。
规格：•2 通道•平衡传输信号•XLR连接头•音频数据达24Bit / 192kHz•缆线长：100m 或更多•阻抗：110Ohm (±20%)•负载电平：输出端2 - 7 Vpp（110 Ohm ，缆线不能长）•大量的通道状态信息AES3 和AES/EBU 比较AES3 数字音频接口和AES/EBU 数字接口只在一个细节上有区别：EBU 标准规定在接口的发送端和接收端强制安装有耦合变压器，而这在AES3 标准中只是可选功能。
功能：发展AES3 标准的目的是为了使数字音频数据可以重复利用模拟音频信号传输网络，要构成一个传输网络需几万米的线来连接设备比如广播电台等。
这些都是平衡缆线，传输信号的频率可达10MHz，若进行适当的信号均衡的话缆线长度可达300m。
若需通过这些模拟信号音频线来传输数字信号的话，需满足以下几个条件，这些条件很容易就可以达到：•由于传输链可能有变压器，因此信号必须是不含直流分量。
•由于没有额外的缆线来传输位时钟（bit clock ）和采样时钟（sample clock ），因此信号自身需携带有时钟信号。
•极性逆转对重拾音频信息无影响。
这些条件可以通过双向标记编码方案（bi-phase-mark coding scheme）来满足。
概述：通过双相标记编码，每个比特的边界都以切换信号极性的方法标记出来。
为了区分信号“1”与信号“0”，需在“1”位插入一个额外的过渡标记代码（如图所示）。
数字音频知识
数字音频知识AES/EBU：实时立体声数字音频信号格式。
在相应设备之间进行传送。
这种格式是AudioEngineeringSociety/EuropeanBroadcastUnion(录音师协会/欧洲广播系统联盟)的缩写。
这种数字格式亦由这两个组织联合制定的。
AES/EBU是由平衡XLR口输出，其他方面同S/PDIF格式相似。
automatedmixing：自动混音。
将各轨的音量、立体声声像位置、或各轨的其它参数如均衡(EQ)值等同乐曲信息放置在一起。
播放时这些信息将控制各轨完成自动混音过程。
一些录音程序可通过屏幕上一些可编辑的多段音量/声像包络来实现自动混音。
另外一种方法是用鼠标拖动显示屏上的推子或旋钮并进行录音，播放时音量/声像会随着推子或旋钮的变化而变化。
另外音量和声像的变化也可以通过将其所对应的控制器信息录入音序器中来实现自动混音。
backup：备份。
虽然硬盘存储被认为是非常可靠的存储方式，但是存于硬盘上的数据很可能会在不经意间毁于一旦。
在以PC为基础的录音系统中，将文件从一个硬盘备份到另一个硬盘就象用WINDOWS 的drag-copy（拖动复制）一样简单。
另外一些录音机可将数据备份到DAT的两个立体声轨上。
需要时，可将所备份的声音数据从DAT 带上恢复回来。
crossfade：淡入/淡出技术。
特别用在前期制作中的一种技术。
这种技术可使一个声音片段平缓地过渡到另一个声音片段。
有些录音机需要两轨来完成这一过程，一轨将声音进行淡出处理，同时另一轨将声音进行淡入处理。
有些则只需要一轨来完成一个声音片段淡出的同时另一个声音片段淡入的过程。
这时控制程序将产生一个新的文件，包含了两个声音片段的混合过渡情况。
很多控制程序还允许用户选择选择第一个声音片段淡出及第二个声音片段淡入的曲线类型。
当选择的曲线为等幂指数曲线时，可保证整体音量在淡入/淡出的过程中没有明显的变化，即声音过渡在听觉上比较自然一些。
DSP：数字信号处理，即一个对音频信号进行处理并使音频信号产生变化的过程。
aes3-2003
AES3-2003Revision of AES3-1992REVISEDAES standard for digital audio —Digital input-output interfacing —Serial transmission format for two-channel linearly represented digitalaudio dataPublished byAudio Engineering Society, Inc.Copyright © 2003 by the Audio Engineering SocietyAbstractThe format provides for the serial digital transmission of two channels of periodically sampled and uniformly quantized audio signals on a single shielded twisted wire pair. The transmission rate is such that samples of audio data, one from each channel, are transmitted in time division multiplex in one sample period. Provision is made for the transmission of both user and interface related data as well as of timing related data, which may be used for editing and other purposes. It is expected that the format will be used to convey audio data that have been sampled at any of the sampling frequencies recognized by the AES5, Recommended Practice for Professional Digital Audio Applications Employing Pulse-Code Modulation — Preferred Sampling Frequencies.An AES standard implies a consensus of those directly and materially affected by its scope and provisions and is intended as a guide to aid the manufacturer, the consumer, and the general public. The existence of an AES standard does not in any respect preclude anyone, whether or not he or she has approved the document, from manufacturing, marketing, purchasing, or using products, processes, or procedures not in agreement with the standard. Prior to approval, all parties were provided opportunities to comment or object to any provision. Attention is drawn to the possibility that some of the elements of this AES standard or information document may be the subject of patent rights. AES shall not be held responsible for identifying any or all such patents. Approval does not assume any liability to any patent owner, nor does it assume any obligation whatever to parties adopting the standards document. This document is subject to periodic review and users are cautioned to obtain the latest printing. Recipients of this document are invited to submit, with their comments, notification of any relevant patent rights of which they are aware and to provide supporting documentation.Contents Foreword (3)Foreword to second edition (3)Foreword to third edition (4)1 Scope (5)2 Normative references (5)3 Definitions and abbreviations (6)4 Interface format (7)4.1 Structure of format (7)4.2 Channel coding (10)4.3 Preambles (10)4.4 Validity bit (11)5 User data format (11)6 Channel status format (11)7 Interface format implementation (19)7.1 General (19)7.2 Transmitter (19)7.3 Receivers (19)8 Electrical requirements (20)8.1 General characteristics (20)8.2 Line driver characteristics (20)8.3 Line receiver characteristics (22)8.4 Connectors (24)Annex A (25)Annex B (26)Annex C (28)Annex D (29)Foreword[This foreword is not a part of the PROPOSED DRAFT REVISED AES standard on digital audio — Digital input-output interfacing — Serial transmission format for two-channel linearly represented digital audio data, DRAFT AES3-xxxx.]Foreword to second editionThis document discusses the format and line protocols for a revision of the AES recommendation, originally published in 1985, for the serial transmission format for linearly represented digital audio data over conventional shielded twisted-pair conductors, of up to at least 100 m in length, without equalization. The organization and style of the revised document are patterned after portions of International Electrotechnical Commission (IEC) Publication 958 and International Radio Consultative Committee (CCIR) Recommendation 647, which are well known in the international technical community.It has been six years since AES3-1985 was adopted as a standard, and much experience with equipment, installations, and applications in professional audio and broadcasting has been accumulated. AES3 has been widely accepted as the primary means of transmitting digital audio for two-channel and multichannel (by combinations of connections) professional and broadcast studio use. Another standard, AES10, has been recently adopted for multichannel use and will in the future provide a more efficient means of transmission for a large number of channels (56). AES10 is based on and designed to be compatible with AES3 for transmitted data, the terminology associated with the data, and the intended use of the data. Also AES11, Synchronization of digital audio equipment in studio operations, refers to a signal conforming to this revised form of AES3. Applications and uses of one or two channels of digital audio as supported by this revised version of AES3 will remain important and numerous.The revision is intended to simplify and clarify language, improve electrical performance, minimize confusion with the IEC Publication 958 "consumer use" specification, allocate certain previously reserved bits to new applications, and improve compatibility by improving uniformity of transmitter implementation in regard to validity, user, channel status, and parity bits. To further facilitate adoption of this standard for the diverse applications and conditions for which it is intended, a separate engineering guideline document — not part of this standard — is in preparation.AES3 has been under constant review since the standard was issued, and the present document reflects the collective experience and opinions of many users, manufacturers, and organizations familiar with equipment or systems employing AES3. Experience includes operation in locations such as large broadcast centers, small recording studios, and field operations. This revision was written in close cooperation with the European Broadcasting Union (EBU). At the time it was written, the AES Working Group on Digital Input-Output Interfacing included the following individuals who contributed to this standard: T. Attenborough, B. Bluethgen, S. Busby, D. Bush, R. Cabot, R. Caine, C. Cellier, S. Culnane, A. Fasbender, R. Finger, B. Fletcher, B. Foster, N. Gilchrist, T. Griffiths, R. Hankinson, S. Herla, R. Hoffner, B. Hogan, T. Holman, Y. Ishida, C. Jenkins, T. Jensen, A. Jubb, A. Komly, R. Lagadec, P. Lidbetter, B. Locanthi, S. Lyman, L. Moller, A. Mornington-West, C. Musialik, J. Nunn, D. Queen, C. Sanchez, J. Schuster, T. Setogawa, T. Shelton, S. Shibata, A. Swanson, A. Viallevieille, D. Walstra, J. Wilkinson, and P. Wilton.R. A. Finger, chairSC-02-02 Working Group on Digital Input-Output Interfacing1991 MarchForeword to third editionThis revision of AES3 was prepared under project AES3-R by a writing group of the SC-02-02 Working Group on Input-Output Interfacing of the SC-02 Subcommittee on Digital Audio.This revision is intended to consolidate the four amendments to the second edition into a single text with minimum impact on the normative sense of the original. Clause and figure numbering has been updated accordingly. Minor normative changes have been made to define a User Bit Management state for metadata and to update "Electrical requirements, General characteristics" to follow a more generalised model adopted by IEC 60958-4 in 2002. Minor editorial changes have also been made, including the tabulation of the Channel Status Data specification.At the time it was written, the AES Working Group on Digital Input-Output Interfacing included the following individuals who contributed to this standard: J. Dunn, J. Brown, R. Cabot, R. Caine, C. Chambers, R. Chinn, I. Dennis, C. Dinneen, A. Eckhart, R. Finger, J. Fujimori, M. Furukawa, C. Gaunt, J. Grant, S. Harris, S. Herla, W. Krafft, S. Lyman, A. Mason, H. Nakashima, J. Nunn, W. Oxford, J. Paul, D. Queen, J Schmidt, S. Scott, P. Skirrow, Y. Sohma, C. Travis, M. Yonge, J. Yoshio.NOTE In AES standards documents, sentences containing the verb "shall" are requirements for compliance with the standard. Sentences containing the verb "should" are strong suggestions (recommendations). Sentences giving permission use the verb "may." Sentences expressing a possibility use the verb "can."REVISEDAES standard for digital audio —Digital input-output interfacing —Serial transmission format for two-channel linearly represented digital audio data1 ScopeThis document specifies a recommended interface for the serial digital transmission of two channels of periodically sampled and linearly represented digital audio data from one transmitter to one receiver.It is expected that the format will be used to convey audio data that have been sampled at any of the sampling frequencies recognized by the AES5 Recommended Practice for Professional Digital Audio Applications Employing Pulse-Code Modulation — Preferred Sampling Frequencies. Note that conformance with this interface specification does not require equipment to utilise these rates. The capability of the interface to indicate other sample rates does not imply that it is recommended that equipment supports these rates. To eliminate doubt, equipment specifications should define supported sampling frequencies.The format is intended for use with shielded twisted-pair cable of conventional design over distances of up to 100 m without transmission equalization or any special equalization at the receiver and at frame rates of up to 50 kHz. Longer cable lengths and higher frame rates may be used, but with a rapidly increasing requirement for care in cable selection and possible receiver equalization or the use of active repeaters, or both.The document does not cover connection to any common carrier equipment, nor does it specifically address any questions about the synchronizing of large systems, although by its nature the format permits easy synchronization of receiving devices to the transmitting device.Specific synchronization issues are covered in AES11 AES recommended practice for digital audio engineering -- Synchronization of digital audio equipment in studio operations. An engineering guideline document to accompany this interface specification has been published as AES-2id AES information document for digital audio engineering -- Guidelines for the use of the AES3 interface.In this interface specification, mention is made of an interface for consumer use. The two interfaces are not identical.2 Normative referencesThe following standards contain provisions which, through reference in this text, constitute provisions of this document. At the time of publication, the editions indicated were valid. All standards are subject to revision, and parties to agreements based on this document are encouraged to investigate the possibility of applying the most recent editions of the indicated standards.AES11, AES recommended practice for digital audio engineering—Synchronization of digital audio equipment in studio operations, Audio Engineering Society, New York, NY, USA .AES18, AES recommended practice for digital audio engineering—Format for the user data channel of the AES digital audio interface, Audio Engineering Society, New York, NY, USA.ITU-T Recommendation V.11: Electrical characteristics for balanced double-current interchange circuits operating at data signalling rates up to 10 Mbit/s, International Telecommunication Union, Geneva, Switzerland..IEC 60268-12, Sound system equipment, part 12: Application of connectors for broadcast and similar use, International Electrotechnical Commission, Geneva, Switzerland.IEC 60958-3, Digital audio interface - Part 3: Consumer applications, International Electrotechnical Commission, Geneva, Switzerland.ISO 646, Information processing—ISO 7-bit coded character set for information interchange, International Organization for Standardization, Geneva, Switzerland.3 Definitions and abbreviations3.1sampling frequencyfrequency of the samples representing an audio signalNOTE When more than one audio signal is transmitted through the same interface, the sampling frequencies are identical.3.2audio sample wordamplitude of a digital audio sampleNOTE Representation is linear in 2’s complement binary form. Positive numbers correspond to positive analog voltages at the input of the analog-to-digital converter (ADC). The number of bits per word can be specified from 16 to 24 in two coding ranges, less than or equal to 20 bits and less than or equal to 24 bits.3.3auxiliary sample bitsfour least significant bits (LSBs) which can be assigned as auxiliary sample bits and used for auxiliary information when the number of audio sample bits is less than or equal to 203.4validity bitbit indicating whether the audio sample bits in the same subframe are suitable for conversion to an analog audio signal3.5channel statusbits carrying, in a fixed format derived from the block (see 3.11), information associated with each audio channel which is decodable by any interface user3.6user datachannel provided to carry any other information3.7parity bitbit provided to permit the detection of an odd number of errors resulting from malfunctions in the interface3.8preamblesspecific patterns used for synchronization. See 4.3.3.9subframefixed structure used to carry the information described in 4.1.1 and 4.1.23.10framesequence of two successive and associated subframes, see 4.1.23.11blockgroup of 192 consecutive framesNOTE The start of a block is designated by a special subframe preamble. See 4.3.3.12channel codingcoding describing the method by which the binary digits are represented for transmission through the interface 3.13unit intervalUIshortest nominal time interval in the coding schemeNOTE There are 128 UI in a sample frame.3.14interface jitterdeviation in timing of interface data transitions (zero crossings) when measured with respect to an ideal clock 3.15intrinsic jitteroutput interface jitter of a device that is either free-running or is synchronized to a jitter-free reference3.16jitter gainratio, expressed in decibels, of the amplitude of jitter at the synchronization input of a device to the resultant jitter at the output of the deviceNOTE This definition excludes the effect of intrinsic jitter.3.17frame raterate of transmission of frames4 Interface format4.1 Structure of format4.1.1 Subframe formatEach subframe is divided into 32 time slots, numbered from 0 to 31. See figure 1.Time slots 0 to 3, the preambles, carry one of the three permitted preambles. See 4.1.2 and 4.3; also see figure 2.Time slots 4 to 27, the audio sample word, carry the audio sample word in linear 2’s complement representation. The most significant bit (MSB) is carried by time slot 27.When a 24-bit coding range is used, the LSB is in time slot 4. See figure 1(a).When a 20-bit coding range is sufficient, time slots 8 to 27 carry the audio sample word with the LSB in time slot 8. Time slots 4 to 7 may be used for other applications. Under these circumstances, the bits in time slots 4 to 7 are designated auxiliary sample bits. See figure 1(b).If the source provides fewer bits than the interface allows, either 20 or 24, the unused LSBs are set to logic 0. Time slot 28, the validity bit, carries the validity bit associated with the audio sample word. See 4.4.Time slot 29, the user data bit, carries 1 bit of the user data channel associated with the audio channel transmitted in the same subframe. See clause 5.Time slot 30, the channel status bit, carries 1 bit of the channel status information associated with the audio channel transmitted in the same subframe. See clause 6.Time slot 31, the parity bit, carries a parity bit such that time slots 4 to 31 inclusive will carry an even number of ones and an even number of zeros (even parity).NOTE The preambles have even parity as an explicit property.Figure 1 — Subframe format4.1.2 Frame formatA frame is uniquely composed of two subframes. See figure 2. Except where otherwise specified the rate of transmission of frames corresponds exactly to the source sampling frequency.The first subframe normally starts with preamble X . However, the preamble changes to preamble Z once every 192 frames. This defines the block structure used to organize the channel status information. The second subframe always starts with preamble Y .The modes of transmission are signaled by setting bits 0 to 3 of byte 1 of channel status. Examples include:Two-channel mode: In two-channel mode, the samples from both channels are transmitted in consecutive subframes. Channel 1 is in subframe 1, and channel 2 is in subframe 2.Stereophonic mode: In stereophonic mode, the interface is used to transmit stereophonic audio in which the two channels are presumed to have been simultaneously sampled. The left, or A , channel is in subframe 1, and the right, or B , channel is in subframe 2.Single-channel mode (monophonic): In monophonic mode, the transmitted bit rate remains at the normal two-channel rate and the audio sample word is placed in subframe 1. Time slots 4 to 31 of subframe 2 either carry the bits identical to subframe 1 or are set to logic 0. A receiver normally defaults to channel 1 unless manual override is provided.Primary-secondary mode: In some applications requiring two channels where one of the channels is the main or primary channel while the other is a secondary channel, the primary channel is in subframe 1, and the secondary channel is in subframe 2.(a)(b)034Preamble LSB24-bit audio sample wordMSB V U C P 31282703478272831P V C U 20-bit audio sample wordAUX LSB PreambleV Validity bitU User data bitC Channel status bitP Parity bitAUX Auxiliary sample bits MSBFigure 2 — Frame format4.2 Channel codingTo minimize the direct-current (d.c.) component on the transmission line, to facilitate clock recovery from the data stream, and to make the interface insensitive to the polarity of connections, time slots 4 to 31 are encoded in biphase-mark.Each bit to be transmitted is represented by a symbol comprising two consecutive binary states. The first state of a symbol is always different from the second state of the previous symbol. The second state of the symbol is identical to the first if the bit to be transmitted is logic 0. However, it is different if the bit is logic 1. See figure 3.Figure 3 — Channel coding4.3 PreamblesPreambles are specific patterns providing synchronization and identification of the subframes and blocks.To achieve synchronization within one sampling period and to make this process completely reliable, these patterns violate the biphase-mark code rules, thereby avoiding the possibility of data imitating the preambles.A set of three preambles is used. These preambles are transmitted in the time allocated to four time slots at the start of each subframe, time slots 0 to 3, and are represented by eight successive states. The first state of the preamble is always different from the second state of the previous symbol, representing the parity bit.Depending on this state the preambles are:Channel Coding Preceding state01Preamble X1110001000011101Subframe 1 Y1110010000011011Subframe 2 Z 1110100000010111Subframe 1 and block startX XChannel 1Y Channel 2Z Channel 1Y Channel 2X Channel 1Y Channel 211Clock(2 times bit rate)Source coding Channel coding (biphase mark)Like biphase code, these preambles are d.c. free and provide clock recovery. They differ in at least two states from any valid biphase sequence.Figure 4 represents preamble X.NOTE Owing to the even-parity bit in time slot 31, all preambles will start with a transition in the same direction. See 4.1.1. Thus only one of these sets of preambles will, in practice, be transmitted through the interface. However, it is necessary for either set to be decodable because a polarity reversal might occur in the connection.ClockFigure 4 — Preamble X (11100010)4.4 Validity bitThe validity bit is logic 0 if the audio sample word is suitable for conversion to an analog audio signal, and it is logic 1 if it is not.There is no default state for the validity bit.5 User data formatUser data bits may be used in any way desired by the user.Possible formats for the user data channel are indicated by the channel status byte 1, bits 4 to 7.The default value of the user data bit is logic 0.6 Channel status formatThe channel status for each audio signal carries information associated with that audio signal, and thus it is possible for different channel status data to be carried in the two subframes of the digital audio signal. Examples of information to be carried in the channel status are: length of audio sample words, number of audio channels, sampling frequency, sample address code, alphanumeric source and destination codes, and emphasis.Channel status information is organized in 192-bit blocks, subdivided into 24 bytes. See figure 5. The first bit of each block is carried in the frame with preamble Z.The specific organization follows, wherein the suffix 0 designates the first byte or bit. Where multiple bit states represent a counting number, tables are arranged with most significant bit (MSB) first, except where noted as LSB first.Byte01234567Key:a use of channel status block jindication of alignment levelb linear PCM identification k channel numberc audio signal pre-emphasis l channel numberd lock indication m multichannel mode numbere sampling frequency n multichanel modef channel mode o digital audio reference signalg user bits management p reserved but undefinedh use of auxiliary sample bits q sampling frequencyi source word length r sampling frequency scaling flags reserved but undefined Figure 5 - Channel status data formatBitbit0Use of channel status block0Consumer use of channel status block (see note).state1Professional use of channel status block.bit1Linear PCM identification0Audio sample word represents linear PCM samples.state1Audio sample word used for purposes other than linear PCMsamples.bits 2 3 4Audio signal emphasis0 0 0Emphasis not indicated. Receiver defaults to no emphasis withmanual override enabled.1 0 0No emphasis. Receiver manual override is disabled.1 1 050 µs + 15 µs emphasis. Receiver manual override is disabled.states1 1 1International Telecommunication Union (ITU-T) J.17 emphasis(with 6,5-dB insertion loss at 800 Hz). Receiver manualoverride is disabledAll other states of bits 2 to 4 are reserved and are not to be used untilfurther defined.bit5Lock indication0Default. Lock condition not indicated.state1Source sampling frequency unlocked..bits 6 7Sampling frequency0 0Sampling frequency not indicated. Receiver default to interfaceframe rate and manual override or auto set is enabled.0 148-kHz sampling frequency. Manual override or auto set isdisabled.states1 044,1-kHz sampling frequency. Manual override or auto set isdisabled.1 132-kHz sampling frequency. Manual override or auto set isdisabled.NOTE 1 The significance of byte 0, bit 0 is such that a transmission from an interface conforming to IEC 60958-3 consumer use can be identified, and a receiver conforming only to IEC 60958-3 consumer use will correctly identify a transmission from a professional-use interface as defined in this standard. Connection of a professional-use transmitter with a consumer-use receiver or vice versa might result in unpredictable operation. Thus the following byte definitions only apply when bit 0 = logic 1 (professional use of the channel status block).NOTE 2 The indication of sampling frequency, or the use of one of the sampling frequencies that can be indicated in this byte, is not a requirement for operation of the interface. The 00 state of bits 6 to 7 may be used if the transmitter does not support the indication of sampling frequency, the sampling frequency is unknown, or the sample frequency is not one of those that can be indicated in this byte.In the latter case for some sampling frequencies byte 4 may be used to indicate the correct value.NOTE 3 When byte 1, bits 1 to 3 indicate single channel double sampling frequency mode then the sampling frequency of the audio signal is twice that indicated by bits 6 to 7 of byte 0.bits0 1 2 3Channel mode0 0 0 0Mode not indicated. Receiver default to two-channel mode.Manual override is enabled.0 0 0 1Two-channel mode. Manual override is disabled.0 0 1 0Single-channel mode (monophonic). Manual override is disabled.0 0 1 1Primary-secondary mode, subframe 1 is primary. Manualoverride is disabled.0 1 0 0Stereophonic mode, channel 1 is left channel. Manual override isdisabled0 1 0 1Reserved for user-defined applications.0 1 1 0Reserved for user-defined applications.0 1 1 1Single channel double sampling frequency mode. Sub-frames 1and 2 carry successive samples of the same signal. The samplingfrequency of the signal is double the frame rate, and is double thesampling frequency indicated in byte 0, but not double the rateindicated in byte 4, if that is used. Manual override is disabled. statesVector to byte 3 for channel identification.1 0 0 0Single channel double sampling frequency mode – stereo modeleft. Sub-frames 1 and 2 carry successive samples of the samesignal. The sampling frequency of the signal is double the framerate, and is double the sampling frequency indicated in byte 0,but not double the rate indicated in byte 4, if that is used. Manualoverride is disabled.1 0 0 1Single channel double sampling frequency mode – stereo moderight. Sub-frames 1 and 2 carry successive samples of the samesignal. The sampling frequency of the signal is double the framerate, and is double the sampling frequency indicated in byte 0,but not double the rate indicated in byte 4, if that is used. Manualoverride is disabled.1 1 1 1Multichannel mode. Vector to byte 3 for channel identification.All other states of bits 0 to 3 are reserved and are not to be used until furtherdefined.bits 4 5 6 7User bits management0 0 0 0Default, no user information is indicated.0 0 0 1192-bit block structure. Preamble Z indicates the start of block.0 0 1 0Reserved for the AES18 standard.0 0 1 1User defined.states0 1 0 0User data conforms to the general user data format defined inIEC 60958-3.0 1 0 1Reserved for metadataAll other states of bits 4 to 7 are reserved and are not to be used until furtherdefined.bits0 1 2Use of auxiliary sample bit0 0 0Maximum audio sample word length is 20 bits (default). Use ofauxiliary sample bits not defined0 0 1Maximum audio sample word length is 24 bits. Auxiliarysample bits are used for main audio sample data0 1 0Maximum audio sample word length is 20 bits. Auxiliarysample bits in this channel are used to carry a singlecoordination signal. See note 10 1 1Reserved for user defined applications.statesAll other states of bits 0 to 2 are reserved and are not to be used untilfurther definedNOTE 1 The signal coding used for the coordination channel is described in Annex A.Encoded audio sample word length of transmitted signal.See notes 2, 3, and 4bits 3 4 5Audio sample word length ifmaximum length is 24 bits asindicated by bits 0 to 2 above.Audio sample word length if maximum length is 20 bits as indicated by bits 0 to 2 above.0 0 0Word length not indicated(default).Word length not indicated (default).0 0 123 bits 19 bits0 1 022 bits 18 bits0 1 121 bits 17 bits1 0 020 bits 16 bits1 0 124 bits 20 bitsstatesAll other states of bits 3 to 5 are reserved and are not to be used until furtherdefined.bits 6 7Indication of alignment level0 0Alignment level not indicated0 1Alignment to SMPTE RP155, alignment level is 20 dB belowmaximum code.1 0Alignment to EBU R68, alignment level is 18.06 dB belowmaximum code.states1 1Reserved for future use.NOTE 2 The default state of bits 3 to 5 indicates that the number of active bits within the 20-bit or 24-bit coding range is not specified by the transmitter. The receiver should default to the maximum number of bits specified by the coding range and enable manual override or automatic set.NOTE 3 The nondefault states of bits 3 to 5 indicate the number of bits within the 20-bit or 24-bit coding range which might be active. This is also an indirect expression of the number of LSBs that are certain to be inactive, which is equal to 20 or 24 minus the number corresponding to the bit state. The receiver should disable manual override and auto set for these bit states.NOTE 4 Irrespective of the audio sample word length as indicated by any of the states of bits 3 to 5, the MSB is in time slot 27 of the transmitted subframe as specified in 4.1.1.。
数字音频协议介绍
AES/EBU (2)ADAT (6)I2S (7)时分复用(TDM) (10)MIDI (12)AES3 接口在1985 年已经被指定并在1992 年正式成为标准接口。
自从定为标准后，AES3 反复更新和调整以适应先进设备的要求，其应用非常普遍。
但另一方面来说这使得它有点复杂。
•2 通道• 平衡传输信号• XLR 连接头• 音频数据达24Bit / 192kHz•缆线长：100m 或者更多•阻抗：110Ohm (± 20%)• 负载电平：输出端 2 - 7 Vpp(110 Ohm ，缆线不能长)• 大量的通道状态信息AES3 数字音频接口和AES/EBU 数字接口只在一个细节上有区别：EBU 标准规定在接口的发送端和接收端强制安装有耦合变压器，而这在AES3 标准中只是可选功能。
发展AES3 标准的目的是为了使数字音频数据可以重复利用摹拟音频信号传输网络，要构成一个传输网络需几万米的线来连接设备比如广播电台等。
这些都是平衡缆线，传输信号的频率可达10MHz，若进行适当的信号均衡的话缆线长度可达300m。
若需通过这些摹拟信号音频线来传输数字信号的话，需满足以下几个条件，这些条件很容易就可以达到：• 由于传输链可能有变压器，因此信号必须是不含直流分量。
•由于没有额外的缆线来传输位时钟(bit clock )和采样时钟(sample clock )，因此信号自身需携带有时钟信号。
• 极性逆转对重拾音频信息无影响。
这些条件可以通过双向标记编码方案(bi-phase-mark coding scheme)来满足。
通过双相标记编码，每个比特的边界都以切换信号极性的方法标记出来。
为了区分信号“1”与信号“0”，需在“1”位插入一个额外的过渡标记代码 (如图所示)。
这个代码是对极性反转的证明，其不含直流分量。
因此其可以通过变压器。
即使比特流中含有很长的“0”或者“1”的序列，但其信号状态还是持续改变的。
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万方数据
万方数据
AES/EBU数字音频接口标准简介
作者：王戎
作者单位：福州广播电视集团技术中心
刊名：
东南传播
英文刊名：SOUTHEAST COMMUNICATION
年，卷(期)：2007(10)
1.彭泽安;郭育扬浅谈数字音频接口技术和D/A转换器
2.李清斌浅谈数字音频接口技术及标准 2001(05)
1.汪波.许卫行.WANG Bo.X(U) Wei-hang AES/EBU数字音频的参数及测量技术[期刊论文]-电声技术2005(4)
2.泰克科技(中国)有限公司数字音频和嵌入音频[期刊论文]-现代电视技术2004(10)
3.甄占京数字音频AES/EBU通道状态介绍[期刊论文]-现代电视技术2005(3)
4.陈浩.CHEN Hao AES/EBU数字音频格式在现场扩声系统中的应用[期刊论文]-电声技术2007,31(10)
5.姜路.李臻AES/EBU数字音频传输的测量[期刊论文]-广播电视信息2010(10)
6.张抒数字音频接口及连接[期刊论文]-有线电视技术2001,8(23)
7.汤伟AEs数字音频接口标准及其在音频系统应用的特点[期刊论文]-音响技术2008(3)
8.刘红数字音频信号的传输与测量[期刊论文]-广播电视信息2010(5)
9.谢科.钱泓毅数字音频接口标准简介[期刊论文]-音响技术2002(4)
10.牛睿数字音频技术的兼容与发展[期刊论文]-科技信息2010(16)
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