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Digital Audio Connections And Synchronization

A brief tutorial on standardized digital audio inputs and outputs along with connections for digital-audio synchronization signals.

By PSW Staff October 2, 2017

Many products are equipped with standardized digital audio inputs and outputs, enabling them to transfer digital audio to and from an assortment of devices without conversion to and from the analog domain.

Some products also have dedicated, standardized connections for digital-audio synchronization signals. To help you understand these technologies, PreSonus has prepared a brief tutorial on digital connections and synchronization.

ADAT Optical

The Alesis ADAT modular digital multitrack tape recorder allowed users to record eight tracks of digital audio simultaneously. The ADAT Optical interface protocol, commonly referred to as “ADAT Lightpipe,” was developed to stream eight channels of 16-, 20-, or 24-bit digital audio at 44.1 kHz or 48 kHz, allowing multichannel digital transfers between ADAT digital recorders and other digital audio devices over a single fiber-optic cable. The ADAT Lightpipe format has been adopted by many audio manufacturers because it’s a compact way to transfer multichannel digital audio data between devices. 

ADAT Lightpipe uses the same type of optical cables and Toslink connectors as the S/PDIF two-channel optical digital audio protocol (discussed shortly). These cables can be purchased at your local recording-equipment store. Toslink is an optical-fiber connection system developed by Toshiba that uses a JIS F05 connector. The generic name for this standard is “EIAJ optical.”

Today, many audio interface manufacturers use ADAT optical and its cousin S/MUX (more on that in a minute) to provide an easy and affordable way to add inputs and outputs to your recording rig when needed. PreSonus uses ADAT Lightpipe I/O on both the Studio 192-series and the Quantum-series interfaces for exactly this purpose. 

For example, the Studio 192 Mobile is only equipped with four analog inputs, making it compact enough to fit in your laptop bag. But because of its ADAT I/O, it can be expanded to record up to 22 inputs simultaneously, simply by adding an ADAT-equipped converter, like the DigiMax-series preamps.

The DigiMax D8 is a companion 8-channel preamp with ADAT output to expand the inputs of any audio interface with an ADAT input. The DigiMax DP88 takes this a step further, by offering both ADAT input and ADAT output, allowing you to expand both the inputs and the outputs of your ADAT-equipped audio interface.

It should be noted that ADAT is a universal protocol. So, even if your audio interface was built by another manufacturer, if it has ADAT inputs and/or outputs, you can connect it to other ADAT-equipped devices. Just be sure to read the section on digital synchronization!

S/MUX

“Sample Multiplexing” or S/MUX is used to transmit high-bandwidth digital audio using lower-bandwidth technology, such as ADAT Lightpipe. S/MUX works by joining two or more digital audio channels to represent a single higher-bandwidth channel. By using S/MUX technology, you can stream 8 channels of digital audio at 88.2 kHz or 96 kHz over the same Lightpipe connection originally designed to stream 16 channels of 44.1 kHz or 48 kHz audio.

The Studio 192-series and Quantum-series interfaces support S/MUX, allowing you to recording and playback 16 channels at 88.2 kHz or 96 kHz.

The DigiMax DP88 also supports S/MUX, allowing you to add eight channel of analog I/O to any S/MUX-equipped device.

AES/EBU

Developed by the Audio Engineering Society and the European Broadcasting Union, AES/EBU (officially known as AES3) is a 2-channel format that can carry audio signals at up to 192 kHz. 

AES/EBU employs a 3-pin XLR connector, which is the same connector used for most professional microphones. A single cable carries both channels of audio data. The StudioLive Series III console mixers are each equipped with an AES/EBU output.

S/PDIF

S/PDIF (Sony/Philips Digital Interface) was co-developed by Sony and Philips to transfer stereo digital audio. It is essentially a consumer version of AES/EBU, and as with AES/EBU, a single cable carries both channels of the stereo audio signal. Digital Audio Tape (DAT) machines were among the first devices to be equipped with this protocol but is has since become popular in consumer audio products such as DVD players, as well as in semi-pro and professional audio products. 

The most common connector used for S/PDIF is an RCA coaxial connector. While S/PDIF RCA coaxial uses the same connector as the analog RCA connection on consumer audio products, the cables are not the same, and these connections should not be confused.

PreSonus products that offer S/PDIF RCA coaxial inputs and outputs include the Studio 68, Studio 192-series, and Quantum-series interfaces. The StudioLive RML-series digital mixers are equipped with a S/PDIF RCA output and the Central Station PLUS and Monitor Station V2 have a S/PDIF RCA input. 

S/PDIF also can be sent over Toslink optical connections, and these have become fairly common in semi-pro and professional audio gear. This is the same connector used for ADAT Lightpipe but the digital protocol is different. The Central Station PLUS offers a S/PDIF Toslink optical input, in addition to its S/PDIF RCA input; it is the only current PreSonus product that offers S/PDIF on Toslink.

Digital Clocking, Word Clock, and BNC

Digital clocking signals are used to synchronize digital audio signals flowing between devices to avoid data errors. Why is that necessary? Read on!

Analog audio is transferred through a cable as a continuous electrical waveform–it’s not divided into discrete steps—and electricity travels through a straight wire at almost the speed of light. So, when you route audio between analog devices, the signals arrive instantaneously, for practical purposes. Therefore, you don’t have to synchronize analog audio when routing between devices.

Transferring digital audio is a very different matter. Computers and other digital devices operate one step at a time, which happens very quickly but it’s not instantaneous, and digital signals are not inherently in perfect time. While uncompressed digital audio plays at a fixed rate (i.e. the sampling frequency), digital clocks are not perfect; their frequency can drift, and they almost always have at least some irregular errors, known as jitter. Therefore, two devices, each following its own clock, are highly unlikely to stay in agreement about precisely when a sample starts and ends. The result is usually an artifact: a pop or glitch in the audio.

To avoid this problem, all digital devices in communication with one another need to follow a single master clock. That means the master clock has to send a signal that essentially says, “everyone start at this moment and follow me!” Even if the master clock’s timing is imperfect, all the slave devices will follow the timing errors exactly and will stay in sync with each other. Therefore, you won’t get timing-related artifacts. In general, the better the master clock, the better the resulting audio will sound, so whenever possible, use the best clock you have or experiment with your rig to find the best result.


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