Society of Motion Picture and Television

Engineers
The Society of Motion Picture and Television Engineers or SMPTE, founded in 1916 as
the Society of Motion Picture Engineers or SMPE, is an international professional
association, based in the United States of America, of engineers working in the motion
imaging industries. An internationally-recognized standards developing organization,
SMPTE has over 400 standards, Recommended Practices and Engineering Guidelines for
television, motion pictures, digital cinema, audio and medical imaging. In addition to
development and publication of standards documents, SMPTE publishes a journal, provides
networking opportunities for its members, produces conferences and exhibitions, and
performs other industry-related functions.

Membership is open to any individual or organization with interest in the subject matter.

Medical Diagnostic Imaging Test Pattern

SMPTE standards documents are copyrighted and may be purchased from the SMPTE
website, or other distributors of technical standards. Standard documents may be purchased
by the general public. Significant standards promulgated by SMPTE include:

 All film and television transmission formats and media, including digital.
 Physical interfaces for transmission of television signals and related data (such as
SMPTE time code and the Serial Digital Interface)
 The SMPTE color bar test pattern and other diagnostic tools
 The Material eXchange Format, or MXF

SMPTE's educational and professional development activities include technical

presentations at regular meetings of its local Sections, annual and biennial conferences in
the US and Australia and the SMPTE Journal. SMPTE also has a number of student
Sections and sponsors scholarships for college students in the motion imaging disciplines.

Related organizations include

  National Television System Committee
 Moving Picture Experts Group
 Pro-MPEG
 The ITU Radiocommunication Sector (formerly known as the CCIR)
 BBC Research Department
 European Broadcasting Union

Algunos de las principales normas aplicadas en televisión.

 SMPTE 259M: the Serial Digital Interface (SD-SDI)

 SMPTE 344M: Enhanced Serial Digital Interface (ED-SDI)
 SMPTE 292M: High Definition Serial Digital Interface (HD-SDI)
 SMPTE 372M: High Definition Serial Digital Interface HD-SDI (DL HD-SDI)
 SMPTE 421M: VC-1 video codec
 SMPTE 424M: 3 Gbit/s Serial Digital Interface (3G-SDI)
 SMPTE 291M: Ancillary Data Packet and Space Formatting

AES/EBU
The digital audio standard frequently called AES/EBU, officially known as AES3, is used
for carrying digital audio signals between various devices. It was developed by the Audio
Engineering Society (AES) and the European Broadcasting Union (EBU) and first
published in 1985, later revised in 1992 and 2003. Both AES and EBU versions of the
standard exist. Several different physical connectors are also defined as part of the overall
group of standards. A related system, S/PDIF, was developed essentially as a consumer
version of AES/EBU, using connectors more commonly found in the consumer market.

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

The AES-3id standard defines a 75-ohm BNC electrical variant of AES3. More recently,
professional equipment (notably by Sony) has used this physical interconnection type. This
uses the same cabling, patching and infrastructure as analogue or digital video, and is thus
common in the broadcast industry.

F05 connectors, 5 mm 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.
The ability to insert unique identifiers into an AES3 bit stream is covered by the AES52
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. This requires a CAT5 or CAT6 type of network infrastructure to
support this.

The Protocol
Simple representation of the protocol for both AES/EBU and S/PDIF

The low-level protocol for data transmission in AES/EBU and S/PDIF is largely identical,
and the following discussion applies for S/PDIF as well unless otherwise noted.

AES/EBU was designed primarily to support 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, AES/EBU allows the data to be run at any rate, and recovers the clock
rate by encoding the data using biphase mark code (BMC).

The bit stream consists of 64-bit frames transmitted once per sample time. This is divided
into two 32-bit subframes (or channels): A (left) and B (right). Each subframe consists of
32 time slots used to transmit individual data bits or synchronization information. 24 bits
are available for audio data, of which 20 bits are normally used.

192 consecutive frames are grouped into an audio block. Certain status information is
transmitted once per audio block. At the default 48 kHz sample rate, there are 250 audio
blocks per second.

The 32 time slots of each subframe are used as following:

Time slots 0 to 3

These slots contain a specially coded preamble that identify the subframe and its position
within the audio block. They do not obey normal BMC encoding rules, 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) that isn't at the start of the
data block.
 Y (or W) : 11100100 if previous time slot was "0", 00011011 if it was "1". (Equivalently,
10010110 NRZI encoded.) Marks a word that isn't for channel A (e.g. a word for channel B
(right) in a stereo signal).
 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 the data block.

They are called X, Y, Z from AES standard; M, W, B from the IEC 958 (an AES
extension).

The 8-bit preambles are transmitted in 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 Y
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
___ ___ ___ ___
/ \___/ \___/ \___/ \___/ \ Normal 0 bits
_ _ _ _ _ _ _ _
/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ \ Normal 1 bits

| 0 | 1 | 2 | 3 | | 0 | 1 | 2 | 3 | Time slots

It is straightforward to extend this structure to additional channels (more subframes per

frame), as is done in the MADI protocol.

Time slots 4 to 7

If the audio word length is more than 20 bits, these slots carry the least significant bits of
the audio sample data.

If the audio word length is 20 bits (the default) or less, these time slots can carry auxiliary
information such as a low-quality auxiliary audio channel for producer talkback or studio-
to-studio communication.

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).

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 may be instructed to mute its
output during the presence of defective samples. It is used by most CD players to indicate
that concealment rather than error correction is taking place.
 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 has a single 192 bit
control word.
 C (30) Channel status bit: like the user bit, the bits from each frame of an audio block are
grouped to make a 192-bit channel status word. 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 used, it is set to
provide even parity over bits 4–31
The Channel Status Word in AES/EBU

As stated before there is one channel status bit in each subframe, making one 192 bit word
for each channel in each block. This 192 bit word is usually presented as 192/8 = 24 bytes.
The contents of the channel status word are completely different between the AES3 and
S/PDIF standards, although they agree that the first channel status bit (byte 0 bit 0)
distinguishes between the two. In the case of AES3, the standard describes in detail how the
bits have to be used. Here is a summary of the channel status word:

 byte 0: basic control data: sample rate, compression, emphasis

o bit 0: A value of 1 indicates this is AES/EBU channel status data. 0 indicates this is
S/PDIF data.
o bit 1: A value of 0 indicates this is linear audio PCM data. A value of 1 indicates
other (usually non-audio) data.
o bits 2–4: Indicates the type of signal preemphasis applied to the data. Generally
set to 100 (none).
o 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.
o 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
AES/EBU data is recorded or otherwise stored. Options are unspecified, 48 kHz
(the default), 44.1 kHz, and 32 kHz.
 byte 1: indicates if the audio stream is stereo, mono or some other combination.
o 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.
o bits 4–7: Used to indicate the format of the user channel word.
 byte 2: audio word length
o 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).
o 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.
o bits 6–7: Unused
 byte 3: used only for multichannel applications
 byte 4: Additional sample rate information.
o bits 0–1: indicate the grade of the sample rate reference, per AES11.
o bit 2: reserved
o 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.
o bit 7: This "sampling frequency scaling flag", if set, indicates that the sample rate is
multiplied by 1/1.001 to match NTSC frame rates.
 byte 5: reserved
 bytes 6–9: Four ASCII 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 by 192 every frame. At 48 kHz, this
wraps every 24h51m18.485333s.
 bytes 18–21: as above, but offset to indicate samples since midnight.
 byte 22: contains information about the reliability of the channel status word.
o bits 0–3: reserved
o bit 4: if set, bytes 0–5 (signal format) are unreliable.
o bit 5: if set, bytes 6–13 (channel labels) are unreliable.
o bit 6: if set, bytes 14–17 (sample address) are unreliable.
o 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 cause by switching mid-block. (Generator polynomial is x8+x4+x3+x2+1, preset to 1.)

 S/PDIF
 ADAT
 AES11
 AES52
 AES-EBU embedded timecode
 MADI
 AES-2id