Expanding Your Audio Setup With ADAT

Despite being over 30 years old, ADAT Lightpipe remains the number one way to add more inputs and outputs to your DAW.

Audio interfaces fulfil multiple functions. They amplify signals from mics. They convert analogue signals to digital signals. They transfer these digital signals to the computer. And, on the way out, they accept digital audio signals from the computer, convert them back to analogue and perhaps amplify them for headphone listening.

Only one of these functions absolutely has to be done by the interface itself. That is the transfer of digital audio to and from the computer. The other functions can be delegated to other devices — and, since these secondary features cost money to implement, they often are.

Manufacturers want their products to be usable straight out of the box, so interfaces have mic preamps and A‑D conversion built in, along with line outputs and a headphone socket or two. But not everyone needs the same number of preamps and line‑level I/O. The solution is to provide extra digital paths to and from the computer, and allow users to add other devices later on to take advantage of these.

There are several different types of digital connection that can be used to expand an audio interface in this way. At present, Ethernet and MADI are mainly found on high‑end professional equipment, catering to situations where large numbers of inputs and outputs are required. On home and project‑studio gear, you’ll find ADAT, AES3 and two types of S/PDIF connection — but as AES3 and S/PDIF are only two‑channel systems, the focus of this article will be on ADAT Lightpipe.

Ups & Downs

A digital audio signal is encoded as a string of ones and zeroes. There are different conventions as to how this encoding should take place, and they are mostly incompatible with each other. There are also different ways in which a given string of ones and zeroes can be represented physically. In order for digital audio to be transferred between two devices, they need to support both the same physical connection, and the same encoding scheme.

Since it’s expensive to design and manufacture new connector and cable types, new formats often recycle existing physical infrastructure. This makes sense from an economic perspective, but can lead to confusion! For example, AES3 employs the 3‑pin XLR connector that is so widely used for analogue audio in studios, while the coaxial version of S/PDIF uses the RCA phono connector that is ubiquitous on hi‑fi equipment. This makes it possible to accidentally connect a digital output in either format to an analogue speaker input — a mistake most people are unlikely to repeat.

AES3 and S/PDIF actually use the same encoding scheme, with trivial differences. The same piece of audio will be represented using the same series of ones and zeroes in both cases. Moreover, this scheme is realised in the same way in both AES3 and coaxial S/PDIF. The ones and zeroes are defined as high and low voltages in an electrical conductor; a cable connecting two XLR or phono sockets can, in principle, carry either type of signal.

By contrast, the other optical version of S/PDIF encodes the audio into exactly the same pattern of ones and zeroes, but uses a very different physical representation of this data. In optical S/PDIF, they are not high and low electrical voltages, but pulses of red light directed along an optical fibre. This is terminated at either end by a connector called TOSlink, originally developed by Toshiba. There’s no viable way to terminate an optical cable in an XLR or phono connector, and even if you could, it would be spectacularly pointless.

In this case, though, there’s also potential for confusion, because when the engineers at Alesis developed the ADAT Lightpipe protocol (see box), they decided to reuse the same optical connectors and optical‑fibre ‘cables’. It’s thus physically possible to send S/PDIF data to a destination that’s expecting an ADAT signal, and vice versa. But doing this will be just as futile as trying to make an XLR input accept an optical signal, because S/PDIF and ADAT use different encoding schemes. It would be like expecting a German speaker to follow instructions issued in Mandarin: they would be heard, but not understood.

Streaming Services

From a technological point of view, AES3, S/PDIF and ADAT Lightpipe all have much in common. They are all synchronous formats, meaning that they transfer data in a continuous stream which keeps pace with the analogue signal it’s derived from or turned into. None of them has robust error correction, so if something goes wrong even momentarily, you’ll hear it! They are all fundamentally one‑way streets: a single cable can carry signals only in one direction. If you want to pass signal in both directions between two devices, you’ll need two connections. Likewise, the connectors on the units themselves are doomed to be either inputs or outputs forever, and can’t switch roles. This usually means that if, for example, your audio interface has only a single optical port, you’ll only be able to use it to add additional inputs; its quota of outputs is fixed.

All of these formats also have a fixed, and limited, data bandwidth, which translates into fairly strict limits on the amount of audio that can be piped down them. AES3 and S/PDIF support stereo (two‑channel) audio at sample rates up to 192kHz. ADAT Lightpipe supports eight channels of audio at sample rates of 44.1 or 48 kHz, though if you need to operate at higher sample rates, you can often do so using a kludge called S/MUX (see box), at the expense of reducing the channel count. In all of these formats, the audio is sent in a 24‑bit, fixed‑point format. This has a theoretical dynamic range of 141dB (accounting for dither), which is far greater than any real‑world signal, but is not as foolproof as the 32‑bit floating‑point format that is becoming popular in some circles.

Not just for the home studio: many flagship interfaces from premium manufacturers also offer ADAT I/O, either as standard or via expansion cards.

Ins & Outs

By far the most common use case for all these formats in the home studio is to add extra preamps or analogue inputs to an audio interface. Sometimes this is necessary in order to increase the total input count, for instance if you want to record a multi‑miked drum kit and don’t have enough inputs to plug all the necessary mics in, or you’re running out of inputs for your growing synth collection. In other circumstances, you might want to do this in order to add a better preamp, or one that has features not available on your audio interface. For example, some ‘voice channel’ products that incorporate analogue EQ and compression as well as preamplification have an S/PDIF output that allows them to connect digitally to your audio interface.

There are also good reasons why you might want to use digital expansion to add more analogue outputs to your system. Perhaps you want to integrate external effects units into your DAW mixes, or send stems out of your software to mix on a hardware console. Possibly you wish to send multiple CV and gate signals directly from your computer to control analogue synths. Or maybe you have invested in an immersive monitoring system and need extra outputs in order to feed all those loudspeakers! I’ll consider each of these use cases in a moment, and look at how ADAT expansion can help meet these needs. But before that, we have to look at one more topic that’s fundamental to digital audio. Clocking is one of the most important, yet most poorly understood aspects of digital audio.

Putting The Clocks Forward

In a digital audio signal, the ones and zeroes that represent the audio waveform itself are interspersed with additional ones and zeroes that fulfil what might be called administrative functions. In all of the formats we’re looking at here, for example, the actual audio data is packaged into a series of bundles or ‘frames’ of fixed size and duration, with extra data marking the beginning and end of each frame.

But how does the receiving device know whether a particular one or zero belongs to the audio data or the frame definition? It needs to be able to count the pulses and frames coming in with absolute precision, and this is only possible if both the sending and receiving device have exactly the same understanding of the duration of a frame. If there’s even the slightest discrepancy, the two units will drift apart over time. This will cause errors, which usually manifest themselves as clicks and pops in the audio.

The Focusrite Clarett+ OctoPre, ART TubeOpto 8 and Audient ASP880 feature, respectively, switchable insert points, valve preamps, and variable mic input impedances.

The solution to this problem is for both devices to adopt the same timing framework. In essence, this is done by putting one device in charge of defining time, and making others follow it. This device is known as the clock ‘master’ or leader.

The clock signal can be distributed on its own, using separate cables, most often in the ‘word clock’ format. This repurposes yet another connector from the analogue world, in this case the BNC type that is widely used in video and radio. However, the clock signal can also be embedded with the digital audio signal itself. This means that both audio and clock data can be transferred simultaneously using a single cable. Where this is practical, it has obvious advantages in terms of simplicity and cost, and so all of the formats described in this article include an embedded clock signal. (In essence, packaging the audio into frames is what achieves this: each time the receiving device detects a ‘new frame’ message, its internal clock is adjusted to that of the source.)

Can’t Clock, Won’t Clock?

It’s important to note that just because a clock signal is present on a digital audio connection, it doesn’t necessarily have to be used! The golden rule is that there can and should only be one clock master in any setup, no matter how complicated. For example, if your audio interface has ADAT Lightpipe in and out connectors, you might use two cables to connect these to the Lightpipe out and in on a second device. This will give you additional inputs and outputs — but only one of the two devices can be the clock master. If you choose to make your interface the clock master, the clock signal being sent from the other device to your interface is ignored. And in more complex setups, the embedded clock might be disregarded entirely. In systems involving more than three or four pieces of gear, it’s usually necessary to have a dedicated word clock signal distributed among them, especially if there are multiple digital formats in use simultaneously.

The ADAT format usefully has a clock signal embedded in it — but it’s up to the user to ensure that all your devices are configured to follow a single clock.

It’s also vital to realise that merely having a clock signal present on a digital audio connection doesn’t automatically mean the receiving device will follow it. It is up to the user to decide which of his or her devices should be the clock master, and to ensure that all the other devices are set up to follow an incoming clock signal. On an interface or other device that has several types of digital connection available, these will all be available as possible clock sources, and you’ll need to choose exactly the right one. You can’t be passive about this, and expect simply to plug in a cable and have everything work without user intervention.

The clock master determines the sample rate for the system. If that’s not your audio interface, then be aware that attempting to load DAW projects at different sample rates can cause much head‑scratching. For example, if your audio interface is being clocked at 44.1kHz from an external unit and you want to open a 48kHz project, you’ll need to change the sample rate on the external unit. In my experience, many devices are also lamentably poor at telling you whether or not they are receiving a valid clock signal. Frequently, you’ll have to delve into a setup page somewhere in the control panel software in order to find out, which isn’t terribly helpful when troubleshooting.

In practical terms, the decision as to which device should be clock master is usually a pragmatic one. You may well find that your particular combination of gear only permits one clocking configuration. If you do have a choice, it’s usually preferable on practical grounds to make the audio interface the clock master. This allows it to follow sample rate changes from the computer DAW more easily, and also means you don’t have to switch on the other gear when you don’t need to use it. Other considerations worth bearing in mind are:

• A system can combine AES3, S/PDIF and ADAT Lightpipe devices as long as they can all be clocked successfully.
• A device that has digital outputs but no digital or clock inputs can only and must be the clock master.
• A device that has digital inputs but no digital or clock outputs cannot be the clock master.
• A device that has both inputs and outputs can pass an incoming clock to another device, even if the input and output connections are of different types.
• A system can’t contain more than one device that has only digital outputs and no inputs, as there would be no way to clock it.

With these clocking caveats out of the way, let’s look in more detail at the use cases I described earlier, and see what’s out there to meet your needs in each case.

Many of us are perfectly happy with the kind of mic preamps and analogue inputs that are built into audio interfaces... The problem is that we just don’t have enough of them — and that’s where ADAT comes to the fore.

I Need More Inputs!

Many of us are perfectly happy with the kind of mic preamps and analogue inputs that are built into audio interfaces. They’re clean, they’re quiet and they have enough gain for our needs. The problem is that we just don’t have enough of them — and that’s where ADAT comes to the fore. As long as you’re happy to work at base sample rates, a single ADAT optical input on your interface allows you to add eight more analogue inputs, with an additional device connected using just a single cable. Many interfaces even have two sets of ADAT I/O, making it possible to add two expansion units for a total of 16 additional inputs. (Note that, as described above, at least one of these two units must have a digital or clock input, as they can’t both be clock master.)

Affordable ADAT preamp expanders include the Audient EVO SP8, Behringer ADA8200, and the Focusrite Scarlett OctoPre.

The entry point into the ‘just give me eight more inputs’ category is probably the Behringer ADA8200, which is surprisingly well featured for something so cheap, with eight mic preamps, eight line outputs, ADAT in and out and a word clock connector to boot. Unusually, though, it has all the inputs on the front panel, which isn’t always what you want. Also highly affordable are Audient’s new EVO SP8 and the Focusrite Scarlett OctoPre. Both offer eight well‑specified mic/line inputs, two of which can accommodate instrument signals, and both also feature eight line outputs. The SP8 also has digitally controlled preamps, with Audient’s clever Smartgain technology for setting preamp gain automatically.

Given the obvious usefulness of eight‑channel ADAT expanders, it’s perhaps surprising that the EVO SP8 and Scarlett OctoPre don’t face more direct competition at their price level. This perception is a bit deceptive, though, because there are in fact plenty of other alternatives, in the shape of audio interfaces capable of stand‑alone operation. Interfaces such as the Focusrite Scarlett 18i20, Steinberg UR816C and Arturia Audiofuse 8pre all have a secondary mode that allows them to operate purely as A‑D and D‑A converters connected over ADAT. Be aware that you may still need to install the interface drivers and control panel software in order to set up the internal routing before stand‑alone operation. Note, too, that this is not a universal feature on all audio interfaces, and it is explicitly not supported on, for example, the PreSonus Studio 1824c, so check the manufacture’s website before buying if you plan to go down this route.

Some audio interfaces also have the ability to act as stand‑alone ADAT converters, such as these models from Arturia, Focusrite and Steinberg. Note, though, that some of them may still need to be configured using a computer.

Moving slightly up the cost ladder can bring additional features or superior quality on your extra inputs. For example, Tascam’s Series 8p Dyna and the Focusrite Scarlett OctoPre Dynamic are both eight‑channel ADAT mic preamps with simple one‑knob compressors on every channel, while the Audient ASP800 and ASP880 feature proprietary console‑type preamp designs with tons of headroom for recording drums and other loud sources, and Focusrite’s Clarett+ OctoPre offers analogue insert points, and the company’s Air processing, on all its inputs. ART’s TubeOpto8 somehow crams eight channels of valve preamplification into a 1U ADAT expander, with pad, polarity and high‑pass filter switches on each input into the bargain. And, again, many audio interfaces in this price band can double as ADAT converters, including models from RME and MOTU.

The Tascam 8p Dyna and Focusrite Scarlett OctoPre Dynamic both feature analogue compressors on each of their inputs.

Some SOS readers will also have digital mixers and stage boxes, and many of these can be pressed into service as ADAT expanders. Manufacturers such as Soundcraft, Digico, Behringer, Allen & Heath and Yamaha all make optional ADAT cards that can be fitted to their mixers.

Many digital mixers can be optionally fitted with ADAT expansion cards.

Finally, a relatively costly but very cool expansion option for recording and mixing applications is a 500‑series chassis with ADAT connectivity, such as Cranborne Audio’s 500ADAT or Heritage Audio’s OST‑8 ADAT. These are modular units that allow you to mix and match your choice of preamps, processors and other modules from a vast array of manufacturers, and the ADAT connectivity makes it possible to access them all directly from your DAW.

Cranborne Audio and Heritage Audio both make 500‑series chassis that can connect directly to your audio interface via ADAT Lightpipe.

I Need More Outputs!

As well as offering eight mic/line inputs, almost all of the devices described above also offer eight line outputs. Usually, it’s possible to use these in one of two ways. They can either be addressed from your computer, as additional DAW outputs, or they can be set to mirror what’s coming into either the analogue or the digital inputs. The latter mode might be useful if, for example, you are recording to a computer but you want to use an analogue mixer to set up cue mixes for the performer, or if you want to run an old hard disk recorder as a safety backup.

The former mode is also useful if you’re working with an analogue mixer or other external hardware. Many people like the hands‑on immediacy of mixing on a console, but want to retain the processing power and editing capabilities of a DAW system. Stemming your DAW projects out to 16 or 24 outputs can be a way of enjoying the best of both worlds. Alternatively, you could use the additional outputs together with the inputs to incorporate outboard processors into your DAW mixes as ‘hardware inserts’. And, of course, if you want to set up an Atmos monitoring rig, you’ll need more than eight line‑level outputs.

If your main reason for expansion is to add extra outputs, though, pay attention to the alignment of the D‑A converters. Most serious outboard gear will be expecting ‘professional’ levels, which typically means that a full‑scale digital signal should generate an analogue level of between +18 and +24 dBu at the outputs. Many interfaces and ADAT expanders aimed at the semi‑pro and hobbyist market are set up to deliver lower levels; this isn’t disastrous in itself, but can complicate things, especially if your audio interface and your ADAT expander are aligned differently. In extremis, it may prove hard to drive outboard compressors properly if your converters put out levels much lower than the compressor is expecting to see. The same applies on the input side, too: if your hardware outboard is geared up to pump out levels up to +24dBu, your line inputs need to be able to accept this without clipping. Any decent manufacturer will make these specifications available as Maximum Output Level and Maximum Input Level respectively.

Another respect in which your interface and expanders can slip out of alignment concerns timing. The process of converting analogue signals to digital, and back again, takes time. These days, A‑D and D‑A conversion introduce only a tiny delay of less than one millisecond, and that’s rarely audible by itself. However, it’s very likely that this delay value will be slightly different for the inputs and outputs on your expander compared with those on your interface, or between expanders of different makes and types. This means that if, for example, you route the left side of a stereo signal through an output on your interface, and the right side through your expander, you may introduce phase issues. It also means that timing correction is going to be even more important than usual in an immersive monitoring setup if some speakers are fed from the interface and others from the expander.

If you don’t need extra preamps, just lots of line‑level I/O, a good option is the Ferrofish Pulse 16.

If you have no need for additional mic preamps and just want lots of line‑level I/O, there are fewer options available new than was the case a decade ago, but one good option might be Ferrofish’s Pulse 16. This is available in several configurations, one of which is equipped with ADAT connectors, and offers 16 high‑quality line‑level ins and outs in a compact 1U box. MOTU’s super‑flexible 8A is another very strong option, which has ADAT I/O built in alongside Ethernet and USB/Thunderbolt audio interface functionality. At the top end of this market, Antelope Audio’s Orion 32+ Gen 3 can also be used as a very high‑spec analogue‑to‑ADAT converter, though if you’re lucky enough to own one, you’ll get a higher channel count and more functionality using it as an audio interface or connecting over MADI.

MOTU’s 8A interface provides a convenient means of adding ADAT I/O to your computer.

No Ceiling

As I mentioned at the start of this article, the restrictions of the ADAT format begin to make themselves felt in serious professional applications, where it might be necessary to connect together many different digital devices, achieve huge I/O counts, operate at high sample rates, or simply to run digital cables over long distances. Even so, there are plenty of high‑end devices that offer ADAT expansion, both on the input and the output side. For example, ADAT I/O is available alongside other digital options even on top‑flight audio interfaces such as the Apogee Symphony I/O MkII, Focusrite Red series, Universal Audio Apollo range, numerous MOTU, RME and Antelope Audio models, the Avid Pro Tools | Carbon and the Lynx Aurora(n).

And at the other end of the cable you’ll find ADAT connectivity on plenty of high‑end preamps. Sometimes this is built‑in, as on the Heritage Audio Super 8, RME Micstacy and 12Mic‑D; other devices such as the Focusrite ISA 828 MkII make it available through an add‑on digital card. Either way, despite its limitations, ADAT has passed the test of time. Whatever you choose to invest in, as long as it has an ADAT port, you’ll be able to connect it and use it for many years to come. 

 

If preamp quality is your main concern, consider the RME 12Mic, Heritage Audio Super 8 or Focusrite ISA 828 MkII, all of which can output via ADAT (the latter via an optional expansion card).