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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 (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. 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 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 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 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 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 2110-31 defines how to encapsulate an AES3 data stream in
Real-time Transport Protocol 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 network. The standard for packing AES3 frames into ATM cells is
AES47. ==See also==