AES/EBU
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Audio Engineering Society (AES) and the European Broadcasting Union (EBU) work on standards for digitizing analogue audio, as used to interconnect both professional and domestic equipment. Work was started in the mid 1980’s. This culminated in the publishing of the AES3 standard (frequently also known as AES/EBU) for professional use as well as, using different physical connections as specified in IEC 60958, within the domestic “Hi-Fi” environment. This work has provided the most commonly used method for digitally interconnecting audio equipment worldwide using physically separate cables for each stereo audio connection.
The digital audio standard frequently called AES/EBU that is officially known as AES3, was first published in 1992 as a standard (and revised in 1995, 1998, and 2003) for carrying digital audio signals between various devices.
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[edit] Hardware Connections
The AES3 standard parallels part 4 of the international standard IEC 60958. Of the physical interconnection types defined by IEC 60958, three are in common use:
- IEC 60958 Type I Balanced - 3-conductor, 110-ohm twisted pair cabling with an XLR connector, used in professional installations (AES3 standard)
- IEC 60958 Type II Unbalanced - 2-conductor, 75-ohm coaxial cable with an RCA connector, used in consumer audio
- IEC 60958 Type II Optical - optical fiber, usually plastic but occasionally glass, with an F05 connector, also used in consumer audio
More recently, professional equipment (notably by Sony) has been fitted with BNC connectors for use with 2-conductor, 75-ohm coaxial cable. This uses the same cabling, patching and infrastructure as analogue or digital video.
F05 connectors, 5mm connectors for plastic optical fiber, are more commonly known by their Toshiba brand name, TOSLINK. The precursor of the IEC 60958 Type II specification was the Sony/Philips Digital Interface, or S/PDIF. For details on the format of AES/EBU data, see the article on S/PDIF. Note that the electrical levels differ between AES/EBU and S/PDIF.
For information on the synchronization of digital audio structures, see the AES11 standard.
Other AES3 transport structures.
AES3 digital audio format can also be carried over an Asynchronous Transfer Mode network. The standard for packing AES3 frames into ATM cells is AES47, and is also published as IEC 62365.
[edit] The Protocol
The low-level protocol for data transmission in AES/EBU and S/PDIF is the same but a few smaller differences. What is said in this paragraph is valid for both the formats, unless where stated.
The bit stream is sent in a structure summarized in the figure on the right. The audio is sent in audio blocks. Each block is made of 192 frames numbered 0 to 191. Each frame is divided in 2 subframes (or channels): A (left) and B (right). Each subframe contains the information for one single sample of the PCM audio. As the subframe is split in 32 time blocks numbered 0 to 31, each sample is described with 32 bits, and this is independent from the quality of the audio stream (it can be 16 bit or 24). Each time block roughly equates to one bit. The subframes combines sample data from the audio source, auxiliary data, etc. The bit stream is encoded with BMC. The 32 bits of the subframes are used as following:
[edit] Time slots 0 to 3
They do not actually carry any data but they facilitate clock recovery and subframe identification. They are not BMC encoded so they are unique in the data stream and they are easier to recognize, but they don't represent real bits. Their structure minimizes the DC component on the transmission line. Three preambles are possible :
- X (or M) : 11100010 if previous state was "0", 00011101 if it was "1"
- Y (or W) : 11100100 if previous state was "0", 00011011 if it was "1"
- Z (or B) : 11101000 if previous state was "0", 00010111 if it was "1"
They are called X, Y,Z from AES standard; M, W,B from the IEC 958 (an AES extension).
[edit] Time slots 4 to 7
These time slots can carry auxiliary information such as a low-quality auxiliary audio channel for producer talkback or studio-to-studio communication. Alternately, they can be used to enlarge the audio word-length to 24 bits.
[edit] Time slots 8 to 27
These time slots carry 20 bits of audio information starting with LSB and ending with MSB. If the source provides fewer than 20 bits, the unused LSBs will be set to the logical “0” (for example, for the 16-bit audio read from CDs bits 8-11 are set to 0).
[edit] Time slots 28 to 31
These time slots carry associated bits as follows:
- V (28) Validity bit: it is set to zero if the audio sample word data are correct and suitable for D/A conversion. Otherwise, the receiving equipment is instructed to mute its output during the presence of defective samples. It is used by players when they have problem reading a sample.
- U (29) User bit: any kind of data such as running time song, track number, etc. One bit per audio channel per frame form a serial data stream. Each audio block have a single 192 bit control word.
- C (30) Channel status bit: its structure depends on whether AES/EBU or S/PDIF is used.
- P (31) Parity bit: for error detection. A parity bit is provided to permit the detection of an odd number of errors resulting from malfunctions in the interface. If set, it indicates an even parity.
[edit] The Channel Status Bit in AES/EBU
As stated before there is one channel status bit in each subframe, making a 192 bits word each audio block. This means that there are 192/8 = 24 byte available each audio block. The contents of the channel status bit are completely different between the AES / EBU and the SPDIF. For the AES / EBU, the standard describes in detail how most part of the bits have to be used. Here is only an overview from a higher point of view, as only the aim of the 24 bytes are described:
- byte 0: basic control data: it says if the audio is compressed or not, if there is any kind of emphasis, which is the sample rate.
- byte 1: it says whether the audio stream is stereo, mono or other their combinations.
- byte 2: it indicates the audio word length.
- byte 3: it is used only for multichannel applications.
- byte 4: it indicates the suitability of the signal as a sampling rate reference.
- byte 5: reserved.
- bytes 6 - 9 and 10 - 13: these two slots of four bytes each are used to transmit ASCII characters.
- bytes 14 - 17: these 4 bytes provide a slot of 32 bit in which there is a sample address incrementing every frame. It is used to number the frames
- bytes 18 - 21: as above, but in a different format, as it is relative to the actual time (it starts from zero at midnight)
- byte 22: it contains information about the reliability of the audio block.
- byte 23: the final byte is a CRC. If it is not received, it means that the transmission was interrupted before the end of the audio block, and so the audio block is ignored.
[edit] See also
[edit] References
- J. Watkinson, The Art of Digital Audio Third Edition, Focal Press, 2001
- THE EBU/AES DIGITAL AUDIO INTERFACE, technical document from the EBU
