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c  

The digital audio standard frequently called c  , officially known as c  and also published as
part of IEC 60958, is used for carrying digital audio signals between devices. It was developed by the
Audio Engineering Society (AES) and the European Broadcasting Union (EBU) and first published in 1985
and later revised in 1992 and 2003. Several different physical connectors are defined as part of the overall
group of standards. A consumer variant of the standard, S/PDIF is also available.

 

The development of standards for digitising analogue audio, as used to interconnect both professional and
domestic equipment was started in the mid 1980s a joint effort between the Audio Engineering Society and
the European Broadcasting Union. This culminated in the publishing of the AES3 standard in 1985. Early
on, the standard was frequently known as AES/EBU. Revisions were issued 1992 and 2003. Both AES and
EBU versions of the standard exist. Variants using different physical connections, essentially a consumer
version of AES3 for use within the domestic ³Hi-Fi´ environment using connectors more commonly found
in the consumer market are specified in IEC 60958. These variants are commonly known as S/PDIF. 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 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:
áY IEC 60958 Type I Balanced ± 3-conductor, 110-ohm twisted pair cabling with an XLR connector,
used in professional installations (AES3 standard)
áY IEC 60958 Type II Unbalanced ± 2-conductor, 75-ohm coaxial cable with an RCA connector, used
in consumer audio
áY 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. 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.
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 (also published as IEC 62365). This requires a CAT5
or CAT6 type of network infrastructure to support this.
? 
 
Simple representation of the protocol for both AES/EBU and S/PDIF
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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  rate, and recovers the clock rate by encoding the data using biphase mark
code (BMC).
The bit stream consists of 64-bit `  transmitted once per sample time. This is divided into two 32-bit
`  (or channels): A (left) and B (right). Each subframe consists of 32
 
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   . 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:
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These slots contain a specially coded  " 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 :
áY 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.
áY ´ (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).
áY 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, ´, 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 ´ marks the beginning of a frame, and Z
marks the beginning of an audio block.
V  V  V  V  V V  V  V  V  V 

         
    
       
      
     
      
      
          
          
  V  V  V  V  V V  V  V  V  V 

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It is straightforward to extend this structure to additional channels (more subframes per frame), as is done
in the MADI protocol.
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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.
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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).
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!
These time slots carry associated bits as follows:
áY 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.
áY 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 channel of each audio block has a single 192 bit
control word.
áY 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.
áY 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
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 # c  
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:
áY byte 0: basic control data: sample rate, compression, emphasis
Y bit 0: A value of 1 indicates this is AES/EBU channel status data. 0 indicates this is
S/PDIF data.
Y bit 1: A value of 0 indicates this is linear audio PCM data. A value of 1 indicates other
(usually non-audio) data.
Y bits 2±4: Indicates the type of signal preemphasis applied to the data. Generally set to 100
(none).
Y 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.
Y 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.
áY byte 1: indicates if the audio stream is stereo, mono or some other combination.
Y 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.
Y bits 4±7: Used to indicate the format of the user channel word.
áY byte 2: audio word length
Y 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).
Y 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.
Y bits 6±7: Unused
áY byte 3: used only for multichannel applications
áY byte 4: Additional sample rate information.
Y bits 0±1: indicate the grade of the sample rate reference, per AES11.
Y bit 2: reserved
Y 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.
Y 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.
áY byte 5: reserved
áY bytes 6±9: Four ASCII characters for indicating channel origin. Widely used in large studios.
áY bytes 10±13: Four ASCII characters indicating channel destination, to control automatic switchers.
Less often used.
áY bytes 14±17: 32-bit sample address, incrementing by 192 every frame. At 48 kHz, this wraps
every 24h51m18.485333s.
áY bytes 18±21: as above, but offset to indicate samples since midnight.[ " #  ]
áY byte 22: contains information about the reliability of the channel status word.
Y bits 0±3: reserved
Y bit 4: if set, bytes 0±5 (signal format) are unreliable.
Y bit 5: if set, bytes 6±13 (channel labels) are unreliable.
Y bit 6: if set, bytes 14±17 (sample address) are unreliable.
Y bit 7: if set, bytes 18±21 (timestamp) are unreliable.
áY byte 23: CRC. This byte is used to detect corruption of the channel status word, as might be
caused by switching mid-block. (Generator polynomial is V8+V4+V3+V2+1, preset to 1.) 
 
áY S/PDIF
áY ADAT
áY AES11
áY AES52
áY AES-EBU embedded timecode
áY MADI
áY AES-2id
(`  
áY European Broadcasting Union, Specification of the Digital Audio Interface (The AES/EBU
interface) Tech 3250-E third edition (2004)
áY Watkinson, John (2001). ? 
 
 ?   
. Focal Press. ISBN 0240515870.
áY Watkinson, John (August 1989). "The AES/EBU Digital Audio Interface". $%   &

  . EBU-02. http://tech.ebu.ch/publications/aes-ebu-guide.
áY Emmett, John (1995). "Engineering Guidelines: The EBU/AES Digital Audio Interface" (PDF).
EBU. http://tech.ebu.ch/docs/other/aes-ebu-eg.pdf.
áY AES-2id-2006: AES information document for digital audio engineering ² Guidelines for the use
of the AES3 interface. (Downloaded from the AES standards web site; see external links)
áY Mark ´onge (June±July 2005). "AES3 Channel Status Revisited". · (101): 20±22.
http://www.ibs.org.uk/public/lineuparchive/2005/101_Jun-
Jul/04_AES3_Channel_Status_Revisited.pdf. Retrieved 2009-03-23.
áY Bernd Noack. "AES3 / AES-EBU channel status byte settings".
http://www.bnoack.com/data/AES_channelstatus.html.
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áY Download page for AES standards
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