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AES / EBU ( Audio Engineering Society / European Broadcasting Union ) refers to the specification of a unidirectional , self-synchronizing and serial interface for the transmission of digital stereo, two-channel or mono audio signals between different devices according to the AES3 standard. It is mainly used in the professional recording studio environment. Not to be confused with the encryption algorithm Advanced Encryption Standard .

AES3 uses balanced cables with 110 ohms or unbalanced coaxial cables with a wave impedance of 75 ohms and twisted pair cables specified in the AES3 id . The connector used is XLR for symmetrical cables and BNC for asymmetrical coaxial cables . When using high quality coaxial cables (and connectors with 75 ohm wave impedance) the lengths can be over 300 meters.

Due to the increasing quality of LAN systems, high-quality twisted pair cables (category 5 or 6, colloquially also called LAN cables) are increasingly being used for AES / EBU cabling in the early 21st century . Their wave impedance lies within the values ​​specified for AES3.

Structure of the interface

Data format of the AES / EBU interface

The transmission format of the AES3 interface is divided into blocks, frames and subframes, as shown in the adjacent figure. A block contains 192 frames, each frame consists of 2 subframes. An audio sample with 16, 20 or a maximum of 24 bits dynamic and with 4 information bits is transmitted per subframe. Among other things, a so-called channel status is transmitted in the information bits , which includes information about the type of audio data. The remaining bits of a 32-bit long time slot are used to synchronize the frames and to protect against errors. One stereo channel can be transmitted per frame.

The usual sampling frequencies for the audio signal are 32 kHz, 44.1 kHz, 48 kHz and 96 kHz. In addition, rarely used sampling rates such as 88.2 kHz and 192 kHz are supported. The word width of the samples can be in the range from 16 bits to 24 bits.

The channel coding of the audio data is carried out using biphase mark coding . The signal to be transmitted does not have a DC component and can be transmitted in a galvanically isolated manner using pulse transformers to avoid hum loops . Electrically, the AES3 interface uses levels and drivers in accordance with the EIA-422 (RS422) standard . Due to the type of channel coding, the sampling rate clock can be recovered from the AES3 signal on the receiving side by means of a PLL .

Furthermore, not only PCM -coded audio signals are transmitted via this interface , but also, for example, Dolby-E or Dolby-Digital -coded multi-channel audio data - these definitions are no longer part of the AES3 specification and are specified in the SMPTE 337M standard.

The bit stream defined in AES3 can also be tunneled in various other interface protocols , that is, the AES3 signal is packaged again within the other protocol and, if necessary, also bundled several times. Examples are MADI , IEEE 1394 , AES50 .

The AES / EBU interface has been standardized by several organizations in approximately the same way:

  • from the AES as AES3
  • from the EBU as Tech-3250E
  • by the IEC as IEC 60958-4

The AES42 standard for digital microphones is based on the AES3 standard.

Signal quality

The digital signal transmission can lead to signal mutilation, especially with long and inferior cables: If the receiver can no longer reconstruct the bit stream 100% with the help of the checksum , the signal will deteriorate. It manifests itself in a loss of amplitude resolution (dynamics) and temporal desynchronization ( glitch ). The signal may then sound rough and be interrupted by crackling. In this case, a remedy is provided by higher-quality cables that are specified for digital transmission (wave impedance of 110 ohms) and avoiding adapters and extensions in the digital signal path.

As a rule, however, the quality of the digital transmission is lossless and therefore, under most circumstances, much better than with analog transmission.

Synchronization of the audio clock

If several digital signal sources are used in a production environment, the PCM currents must be clock-synchronized. In this case, professional PCM signal sources (A / D converters, digital tape machines, etc.) can be clocked externally by a so-called house clock generator to ensure synchronization of the PCM bit streams. Desynchronized digital signal sources manifest themselves through audible crackling and even noise when the signal is completely lost (see also jitter ).

See also

Individual evidence

  1. Michael Dickreiter, Volker Dittel, Wolfgang Hoeg, Martin Wöhr: Handbook of Tonstudiotechnik . 8th edition. de Gruyter , 2014, ISBN 978-3-11-028978-7 , p. 688 f . ( limited preview in Google Book search)
  2. ^ John Emmett: Engineering Guidelines - The EBU / AES Digital Audio Interface. (PDF; 300 kB) European Broadcasting Union, 1995, accessed on September 6, 2018 (English).

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