AES3 is a
standard Standard may refer to:
Symbols
* Colours, standards and guidons, kinds of military signs
* Standard (emblem), a type of a large symbol or emblem used for identification
Norms, conventions or requirements
* Standard (metrology), an object th ...
for the exchange of
digital audio signals between
professional audio
Professional audio, abbreviated as pro audio, refers to both an activity and a category of high quality, studio-grade audio equipment. Typically it encompasses sound recording, sound reinforcement system setup and audio mixing, and studio mu ...
devices. An AES3 signal can carry two channels of
pulse-code-modulated digital audio over several
transmission media
A transmission medium is a system or substance that can mediate the propagation of signals for the purposes of telecommunication. Signals are typically imposed on a wave of some kind suitable for the chosen medium. For example, data can modulate ...
including
balanced lines,
unbalanced line
In telecommunications and electrical engineering in general, an unbalanced line is a pair of conductors intended to carry electrical signals, which have unequal impedances along their lengths and to ground and other circuits. Examples of unbalanc ...
s, and
optical fiber
An optical fiber, or optical fibre in Commonwealth English, is a flexible, transparent fiber made by drawing glass ( silica) or plastic to a diameter slightly thicker than that of a human hair. Optical fibers are used most often as a mea ...
.
AES3 was jointly developed by the
Audio Engineering Society (AES) and the
European Broadcasting Union (EBU) and so is also known as AES/EBU. The standard was first published in 1985 and was revised in 1992 and 2003. AES3 has been incorporated into the
International Electrotechnical Commission's standard IEC 60958, and is available in a consumer-grade variant known as
S/PDIF
S/PDIF (Sony/Philips Digital Interface) is a type of digital audio interface used in consumer audio equipment to output audio over relatively short distances. The signal is transmitted over either a coaxial cable (using RCA or BNC connectors ...
.
History and development
The development of standards for
digital audio interconnect for both professional and domestic audio equipment, began in the late 1970s
in a joint effort between the Audio Engineering Society and the European Broadcasting Union, and culminated in the publishing of AES3 in 1985. The AES3 standard has been revised in 1992 and 2003 and is published in AES and EBU versions.
Early on, the standard was frequently known as AES/EBU.
Variants using different physical connections are specified in IEC 60958. These are essentially consumer versions of AES3 for use within the domestic
high fidelity environment using connectors more commonly found in the consumer market. These variants are commonly known as S/PDIF.
Related standards and documents
IEC 60958
IEC 60958 (formerly IEC 958) is the
International Electrotechnical Commission's
standard Standard may refer to:
Symbols
* Colours, standards and guidons, kinds of military signs
* Standard (emblem), a type of a large symbol or emblem used for identification
Norms, conventions or requirements
* Standard (metrology), an object th ...
on
digital audio interface
Audio connectors and video connectors are electrical or optical connectors for carrying audio or video signals. Audio interfaces or video interfaces define physical parameters and interpretation of signals. For digital audio and digital video, ...
s. It reproduces the AES3 professional digital audio interconnect standard and the consumer version of the same,
S/PDIF
S/PDIF (Sony/Philips Digital Interface) is a type of digital audio interface used in consumer audio equipment to output audio over relatively short distances. The signal is transmitted over either a coaxial cable (using RCA or BNC connectors ...
.
The standard consists of several parts:
* IEC 60958-1: General
* IEC 60958-2: Software Information Delivery Mode
* IEC 60958-3: Consumer applications
* IEC 60958-4: Professional applications
* IEC 60958-5: Consumer application enhancement
AES-2id
AES-2id is an AES information document published by the
Audio Engineering Society for digital audio engineering—Guidelines for the use of the AES3 interface. This document provides guidelines for the use of AES3, AES Recommended Practice for Digital Audio Engineering, Serial transmission format for two-channel linearly represented digital audio data. This document also covers the description of related standards used in conjunction with AES3 such as
AES11 The AES11 standard published by the Audio Engineering Society provides a systematic approach to the synchronization of digital audio signals. AES11 recommends using an AES3 signal to distribute audio clocks within a facility. In this application, th ...
.
The full details of AES-2id can be studied in the standards section of the
Audio Engineering Society web site by downloading copies of the AES-2id document as a PDF file.
Hardware connections
The AES3 standard parallels part 4 of the international standard IEC 60958. Of the physical interconnection types defined by IEC 60958, two are in common use.
IEC 60958 type I
Type I connections use
balanced
In telecommunications and professional audio, a balanced line or balanced signal pair is a circuit consisting of two conductors of the same type, both of which have equal impedances along their lengths and equal impedances to ground and to other ci ...
, three-conductor, 110-ohm
twisted pair
Twisted pair cabling is a type of wiring used for communications in which two conductors of a single circuit are twisted together for the purposes of improving electromagnetic compatibility. Compared to a single conductor or an untwisted ba ...
cabling with
XLR connector
The XLR connector is a type of electrical connector primarily used in professional audio, video, and stage lighting equipment. XLR connectors are cylindical in design, and have three to seven connector pins, and are often employed for analog b ...
s. Type I connections are most often used in professional installations and are considered the standard connector for AES3. The hardware interface is usually implemented using
RS-422
RS-422, also known as TIA/EIA-422, is a technical standard originated by the Electronic Industries Alliance that specifies electrical characteristics of a digital signaling circuit. It was meant to be the foundation of a suite of standards that ...
line drivers and receivers.
IEC 60958 type II
IEC 60958 Type II defines an unbalanced electrical or optical interface for
consumer electronics
Consumer electronics or home electronics are electronic ( analog or digital) equipment intended for everyday use, typically in private homes. Consumer electronics include devices used for entertainment, communications and recreation. Usuall ...
applications. The precursor of the IEC 60958 Type II specification was the Sony/Philips Digital Interface, or
S/PDIF
S/PDIF (Sony/Philips Digital Interface) is a type of digital audio interface used in consumer audio equipment to output audio over relatively short distances. The signal is transmitted over either a coaxial cable (using RCA or BNC connectors ...
. Both were based on the original AES/EBU work. S/PDIF and AES3 are interchangeable at the protocol level, but at the physical level, they specify different electrical signalling levels and
impedances, which may be significant in some applications.
BNC Connector
![BNC connector 20050720 001](https://upload.wikimedia.org/wikipedia/commons/9/90/BNC_connector_20050720_001.jpg)
AES/EBU signals can also be run using unbalanced BNC connectors a with a 75-ohm coaxial cable. The unbalanced version has a very long transmission distance as opposed to the 150 meters maximum for the balanced version. The AES-3id standard defines a 75-ohm
BNC electrical variant of AES3. This uses the same cabling, patching and infrastructure as analogue or digital video, and is thus common in the broadcast industry.
Protocol
:''The low-level protocol for data transmission in AES3 and S/PDIF is largely identical, and the following discussion applies for S/PDIF, except as noted.''
AES3 was designed primarily to support stereo
PCM encoded audio in either
DAT format at 48 kHz or
CD format at 44.1 kHz. No attempt was made to use a carrier able to support both rates; instead, AES3 allows the data to be run at ''any'' rate, and encoding the clock and the data together using
biphase mark code
Differential Manchester encoding (DM) is a line code in digital frequency modulation in which data and clock signals are combined to form a single two-level self-synchronizing data stream. In various specific applications, this method is also call ...
(BMC).
Each bit occupies one ''time slot''. Each audio sample (of up to 24 bits) is combined with four flag bits and a synchronisation preamble which is four time slots long to make a ''subframe'' of 32 time slots. The 32 time slots of each subframe are assigned as follows:
Two subframes (A and B, normally used for left and right audio channels) make a frame. Frames contain 64 bit periods and are produced once per audio sample period. At the highest level, each 192 consecutive frames are grouped into an ''audio block''. While samples repeat each frame time, metadata is only transmitted once per audio block. At 48 kHz sample rate, there are 250 audio blocks per second, and 3,072,000 time slots per second supported by a 6.144 MHz biphase clock.
Synchronisation preamble
The synchronisation preamble is a specially coded ''preamble'' that identifies the subframe and its position within the audio block. Preambles are not normal BMC-encoded data bits, although they do still have zero
DC bias
In signal processing, when describing a periodic function in the time domain, the DC bias, DC component, DC offset, or DC coefficient is the mean amplitude of the waveform. If the mean amplitude is zero, there is no DC bias. A waveform with no DC ...
.
Three preambles are possible:
*X (or M) : 11100010 if previous time slot was ''0'', 00011101 if it was ''1''. (Equivalently, 10010011
NRZI encoded.) Marks a word for channel A (left), other than at the start of an audio block.
*Y (or W) : 11100100 if previous time slot was ''0'', 00011011 if it was ''1''. (Equivalently, 10010110
NRZI encoded.) Marks a word for channel B (right).
*Z (or B) : 11101000 if previous time slot was ''0'', 00010111 if it was ''1''. (Equivalently, 10011100
NRZI encoded.) Marks a word for channel A (left) at the start of an audio block.
The three preambles are called X, Y, Z in the AES3 standard; and M, W, B in IEC 958 (an AES extension).
The 8-bit preambles are transmitted in the time allocated to the first four time slots of each subframe (time slots 0 to 3). Any of the three marks the beginning of a subframe. X or Z marks the beginning of a frame, and Z marks the beginning of an audio block.
, 0 , 1 , 2 , 3 , , 0 , 1 , 2 , 3 , Time slots
_____ _ _____ _
/ \_____/ \_/ \_____/ \_/ \ Preamble X
_____ _ ___ ___
/ \___/ \___/ \_____/ \_/ \ Preamble Y
_____ _ _ _____
/ \_/ \_____/ \_____/ \_/ \ Preamble Z
___ ___ ___ ___
/ \___/ \___/ \___/ \___/ \ All 0 bits BMC encoded
_ _ _ _ _ _ _ _
/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ \ All 1 bits BMC encoded
, 0 , 1 , 2 , 3 , , 0 , 1 , 2 , 3 , Time slots
In two-channel AES3, the preambles form a pattern of ZYXYXYXY..., but it is straightforward to extend this structure to additional channels (more subframes per frame), each with a Y preamble, as is done in the
MADI Madi may refer to:
Places
* Madi, Chitwan, a municipality in Chitwan District in Nepal
* Madi Municipality, Sankhuwasabha, a municipality in Sankhuwasabha District in Nepal
* Madi Rural Municipality, Rolpa, a rural municipality in Rolpa Distr ...
protocol.
Channel status word
There is one channel status bit in each subframe, a total of 192 bits or 24 bytes for each channel in each block. Between the AES3 and S/PDIF standards, the contents of the 192-bit channel status word differ significantly, although they agree that the first channel status bit distinguishes between the two. In the case of AES3, the standard describes, in detail, the function of each bit.
*Byte 0: Basic control data: sample rate, compression, emphasis
**bit 0: A value of 1 indicates this is AES3 channel status data. 0 indicates this is S/PDIF data.
**bit 1: A value of 0 indicates this is linear audio PCM data. A value of 1 indicates other (usually non-audio) data.
**bits 2–4: Indicates the type of signal
preemphasis
Typically, prior to some process, such as transmission over cable, or recording to phonograph record or tape, the input frequency range most susceptible to noise is boosted. This is referred to as "pre-emphasis"before the process the signal will u ...
applied to the data. Generally set to 100 (none).
**bit 5: A value of 0 indicates that the source is locked to some (unspecified) external time sync. A value of 1 indicates an unlocked source.
**bits 6–7: Sample rate. These bits are redundant when real-time audio is transmitted (the receiver can observe the sample rate directly), but are useful if AES3 data is recorded or otherwise stored. Options are unspecified, 48 kHz (the default), 44.1 kHz, and 32 kHz. Additional sample rate options may be indicated in the ''extended sample rate'' field (see below).
*Byte 1: indicates if the audio stream is stereo, mono or some other combination.
**bits 0–3: Indicates the relationship of the two channels; they might be unrelated audio data, a stereo pair, duplicated mono data, music and voice commentary, a stereo sum/difference code.
**bits 4–7: Used to indicate the format of the user channel word
*Byte 2: Audio word length
**bits 0–2: Aux bits usage. This indicates how the aux bits (time slots 4–7) are used. Generally set to 000 (unused) or 001 (used for 24-bit audio data).
**bits 3–5: Word length. Specifies the sample size, relative to the 20- or 24-bit maximum. Can specify 0, 1, 2 or 4 missing bits. Unused bits are filled with 0, but audio processing functions such as mixing will generally fill them in with valid data without changing the effective word length.
**bits 6–7: Unused
*Byte 3: Used only for multichannel applications
*Byte 4: Additional sample rate information
**bits 0–1: Indicates the grade of the sample rate reference, per
AES11 The AES11 standard published by the Audio Engineering Society provides a systematic approach to the synchronization of digital audio signals. AES11 recommends using an AES3 signal to distribute audio clocks within a facility. In this application, th ...
**bit 2: Reserved
**bits 3–6: Extended sample rate. This indicates other sample rates, not representable in byte 0 bits 6–7. Values are assigned for 24, 96, and 192 kHz, as well as 22.05, 88.2, and 176.4 kHz.
**bit 7: Sampling frequency scaling flag. If set, indicates that the sample rate is multiplied by 1/1.001 to match
NTSC
The first American standard for analog television broadcast was developed by National Television System Committee (NTSC)National Television System Committee (1951–1953), Report and Reports of Panel No. 11, 11-A, 12–19, with Some supplement ...
video frame rates.
*Byte 5: Reserved
*Bytes 6–9: Four
ASCII
ASCII ( ), abbreviated from American Standard Code for Information Interchange, is a character encoding standard for electronic communication. ASCII codes represent text in computers, telecommunications equipment, and other devices. Because ...
characters for indicating channel origin. Widely used in large studios.
*Bytes 10–13: Four ASCII characters indicating channel destination, to control automatic switchers. Less often used.
*Bytes 14–17: 32-bit sample address, incrementing block-to-block by 192 (because there are 192 frames per block). At 48 kHz, this wraps approximately every day.
*Bytes 18–21: 32-bit sample address offset to indicate samples since midnight.
*Byte 22: Channel status word reliability indication
**bits 0–3: Reserved
**bit 4: If set, bytes 0–5 (signal format) are unreliable.
**bit 5: If set, bytes 6–13 (channel labels) are unreliable.
**bit 6: If set, bytes 14–17 (sample address) are unreliable.
**bit 7: If set, bytes 18–21 (timestamp) are unreliable.
*Byte 23:
CRC. This byte is used to detect corruption of the channel status word, as might be caused by switching mid-block.
Embedded timecode
SMPTE timecode
SMPTE timecode ( or ) is a set of cooperating standards to label individual frames of video or film with a timecode. The system is defined by the Society of Motion Picture and Television Engineers in the SMPTE 12M specification. SMPTE revised ...
data can be embedded within AES3 signals. It can be used for
synchronization and for logging and identifying audio content. It is embedded as a 32-bit binary word in bytes 18 to 21 of the channel status data.
The
AES11 The AES11 standard published by the Audio Engineering Society provides a systematic approach to the synchronization of digital audio signals. AES11 recommends using an AES3 signal to distribute audio clocks within a facility. In this application, th ...
standard provides information on the synchronization of digital audio structures.
the
AES52 standard describes how to insert unique identifiers into an AES3 bit stream.
SMPTE 2110
SMPTE 2110
SMPTE 2110 is a suite of standards from the Society of Motion Picture and Television Engineers (SMPTE) that describes how to send digital media over an IP network.
SMPTE 2110 is intended to be used within broadcast production and distribution fac ...
-31 defines how to encapsulate an AES3 data stream in
Real-time Transport Protocol
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applicati ...
packets for transmission over an IP network using the SMPTE 2110 IP based multicast framework.
SMPTE 302M
SMPTE 302M-2007 defines how to encapsulate an AES3 data stream in an
MPEG transport stream for television applications.
Other formats
AES3 digital audio format can also be carried over an
Asynchronous Transfer Mode
Asynchronous Transfer Mode (ATM) is a telecommunications standard defined by American National Standards Institute (ANSI) and ITU-T (formerly CCITT) for digital transmission of multiple types of traffic. ATM was developed to meet the needs of ...
network. The standard for packing AES3 frames into ATM cells is
AES47
AES47 is a standard which describes a method for transporting AES3 professional digital audio streams over Asynchronous Transfer Mode (ATM) networks.
The Audio Engineering Society (AES) published AES47 in 2002. The method described by AES47 is al ...
.
See also
*
ADAT Lightpipe
The ADAT Lightpipe, officially the ADAT Optical Interface, is a standard for the transfer of digital audio between equipment. It was originally developed by Alesis but has since become widely accepted, with many third party hardware manufacturers ...
Multichannel optical digital audio interface
*
Notes
References
Further reading
*
*
External links
Download page for AES3 standard*European Broadcasting Union
Specification of the Digital Audio Interface (The AES/EBU interface)Tech 3250-E third edition (2004)
*
*
*
IEC - Historical Collection IHS
{{European Broadcasting Union
Digital audio
Sound
Broadcast engineering
Wikipedia articles with ASCII art
IEC 60958
Audio Engineering Society standards
European Broadcasting Union