Pad Culture: The Story Of The Launchpad Lightshow

How The YouTube Lightshow Phenomenon Evolved

When Launchpad first appeared in 2009, its creators at Novation and Ableton had no idea that it would become an Internet sensation. In fact, the humble pad kept a fairly low profile on the music equipment landscape for the first year of its life. Launchpad’s main usage during this time was by Ableton users who simply wanted to gain manual control over their Live Sets. Launchpad’s hands-on button grid and one-to-one control of clip launching in Live’s Session View made it the perfect interface for creators wanting an affordable, expandable and easy-to-use control surface.

Despite Launchpad’s immediate Ableton-focussed attraction, it always had a DIY element to it. But it took a while for users to see the real potential of the box of buttons sitting in their studio. Unlike other controllers on the market at the time — namely the Akai APC40 — Launchpad didn’t constrain its users to a pre-determined workflow. Instead, it provided a blank canvas in the form of User modes, which allowed creators to make unique and expressive hardware interfaces for practically any software using simple MIDI programming. Launchpad was not the first to provide this functionality: the familiar-looking Monome 64 was released more than a year before Launchpad, but it required a healthy amount of programming and experimentation to operate it. What Launchpad had, crucially, was the ability to plug-and-play, and for the user to quickly and intuitively design controller surfaces without needing to learn a new programming platform such as Open Sound Control (OSC) or Max/MSP.

The Madeon Effect

Launchpad’s potential as a performance instrument wasn’t fully appreciated until a young French artist known as Madeon posted a video on YouTube called ‘Pop Culture’. The date was July 11th 2011 and in this grainy performance debut, which was filmed in his parents’ basement, Madeon triggers impeccably arranged samples from “39 songs I like” in what will go down as one of the greatest live mashups in history. His instrument? The Novation Launchpad. In this gift from the social media gods, Launchpad’s potential was laid bare in a perfectly concise video that reached millions of people in a matter of weeks. It was a true viral sensation. Soon, the clip-launching paradigm was part of pop culture (pun intended), and Launchpad was the de facto hardware facilitator.

Madeon’s Pop Culture video captured the imagination of countless creators, but it took several more months for anyone else to step up to the plate. Australian native M4SONIC was one of the leaders of the second wave, first with an all-live mashup of Skrillex samples, then his own track Weapon, the video for which was posted to YouTube almost exactly a year after Pop Culture. M4SONIC’s sound was raw and untamed, and bore all the hallmarks of the EDM sound that was in its heyday. Other Launchpad creators, such as fellow Adelaide native Exige, were soon to join the fold, experimenting with multi-Launchpad setups and overdubbed, split-screen performances. Exige’s ‘Ultimate Launchpad Mashup’ was one of the earliest examples of this.

Raising The Bar

Before long, Launchpad Lightshows became more than just the artistic creations of a few headliners; each video set a benchmark for others to meet. Fans and fellow performers alike would watch in awe, often playing and replaying videos in stop-motion to deconstruct the performances and find inspiration in others’ work. Such was the open nature of the scene that it became popular for Lightshow artists to share their Ableton Session files, to show their working, encourage rookie creators, and prove that they weren’t faking it!

In parallel with the explosion of new, talented Lightshow artists, Novation were making improvements to the Launchpad hardware. First came the Launchpad S, which featured brighter LEDs and a faster processor, which allowed for much quicker refresh of the illuminated pads. Though it was still colour-limited to reproducing green, orange, yellow and red (like the Launchpad v1), the higher processor speed meant more resolution for flashing lights and LED effects. Lightshow pioneers like Nev took full advantage of this, as seen in his remix of Zedd’s ‘Spectrum’ from early 2014.

But what really accelerated the scene was the 2015 introduction of the Launchpad MkII and Launchpad Pro. Both new Launchpads featured RGB LEDs, capable of reproducing a huge variety of colours, from pure white to neon pinks and blues — and everything in between. As soon as this new technology hit the market, the complexity of Launchpad Lightshows escalated greatly. Suddenly, creators could draw pictures with their lights, and program mesmerising rippling light effects over the pads of multiple Launchpads. LA-based creator R!ot fused his pure songwriting talent with colorful light effects in his video to his song ‘Slowly’, while UK performer Kaskobi became well known for his covers of popular songs, coupled with intense video performances that featured as much emphasis on the LED spectacle as it did on the music.

Split Community

It’s at this point that many creators acknowledge that the scene began to split. At one end of the scale, there were the 100% ‘live’ artists, whose button pushes would directly correlate with a sound being triggered and an LED action. At the opposite end of the spectrum, a more coder-like performer would program complex light actions that would be triggered in sync with a musical section, but not necessarily as a result of a button push. In the space between these two poles, hundreds of performers experimented with new styles, sounds and shapes, to hone their skills to become Lightshow legends.

Despite these minor divisions, the overall trajectory of the scene is positive — new Lightshow artists seem to spring up every week from all corners of the planet. The community of Lightshow fans, who loyally support their favorite artists on web forums and via YouTube comments, push creators to find new ways to express themselves using Launchpad as their instrument. And despite some controversies surrounding the topics of ‘faking it’, Lightshow artists are all interested in one common goal: to move the scene further, and to keep pushing Launchpad’s technical capabilities. The story of the Launchpad Lightshow is one of phenomenal achievement, technical ability and personal triumph… And it’s only just begun.

If you’re new to the Launchpad Lightshow scene, its super easy to get involved. A good starting point is to follow our six-step guide to Launchpad success.

To discover the Launchpad range, head to the Launchpad home page.

Pad Culture: The Story Of The Launchpad Lightshow was originally published in Novation // Notes on Medium, where people are continuing the conversation by highlighting and responding to this story.

The Mathematical & Spiritual Virtues Of ‘The Pocket’

As a member of the funk powerhouse Parliament Funkadelic, Danny Bedrosian is no stranger to swung grooves, wonky rhythms and the other ingredients of break-neck beats. In this interview, he explains ‘the pocket’: one of the secrets to the P-Funk groove.

Like many musical stories, this one begins with James Brown, who Danny Bedrosian first saw at the age of eight or nine. “My Dad brought me to see James Brown, and I remember him saying to his band on stage, ‘keep it in the pocket, keep it tight, keep it in the pocket’ and it just drove me to figure out what that meant.”

Even at that tender age — long before he joined the hallowed P-Funk roster as keys player for George Clinton and Parliament Funkadelic — Bedrosian was no newcomer to musical theory. By his own admission, he was “reared” to be a concert pianist by his Armenian parents, so he had already learned some aspects of what the pocket was from playing orchestral music and hearing traditional Armenian folk. “A lot of that stuff came from me playing Beethoven as a kid, even though I couldn’t tell you what it was when I was younger. But I grew up with it and I could feel it, and I knew exactly what it was when I felt it.” recalls Danny.

Terminology

So, what is ‘the pocket’, and how is it defined? Danny describes it thus: “The pocket is the pulling back of the backbeat to create a more funky, kind of limping feel to the way the groove moves.” Using the example of a jazz bass-line played on an upright bass, he continues, “you can picture it as ‘walking’ bass, where all the notes align within the same rhythmic parameters. But when you have the pocket being applied, that downbeat has more of a stagger, or it lasts a little bit longer within the framework of what is considered one musical measure, whether it’s a quarter note, eighth note, half note, or whatever. Those notes end up being stretched as far as they possibly can without going into the next beat. At that very last moment, you create that pocket, that pull-back.”

“When you have the pocket being applied, that downbeat lasts a little bit longer within the framework of what is considered one musical measure.” — Danny Bedrosian

Other P-Funk alums have embraced the mathematical nature of the pocket, as Bedrosian explains, “Cordell ‘Boogie’ Mosson, who was a long-time bass player in Parliament Funkadelic, used to describe the pocket in terms of looking at the period of a musical measure on a graph, where there are a million places that you can land the beat within the parameters of that actual measure, without being too early or too late. The way that you can stretch the time between beats to the point where, with P-Funk music specifically, the beat comes at the last possible moment before it would be considered late, almost to the point where it is late to someone who is very metronomical. As long as you can maintain that within each beat, it can be a really beautiful pocket that transports people to places that they’ve never gone to before — both the audience and musicians. It really does have a mathematical and a spiritual side to it.”

An extension of the pocket is the concept of ‘the one’ — the first beat in the bar — and specifically when to play it, and with what emphasis. The track ‘Cold Sweat’ by James Brown is slated by Bedrosian and other musicologists as one the first modern funk songs, and one of the earliest examples of the pocket being used in modern music. “In ‘Cold Sweat’, the one is emphasised more than the rest of the groove. After that first ‘bam’ comes the backbeat, which is the second beat in the bar, which is also heavily emphasised. You’re hearing this emphasis on the one with the backbeat deep in the pocket, so the one and the two have serious prominence within the groove. Fast forward to P-Funk — our sub-genre of funk — and the one and the backbeat pocket are both emphasized to the highest regard.”

Feeling & Learning The Pocket

“The pocket is completely 100% feel oriented,” says Danny, and a lot of P-Funk’s trademark rhythms can be traced to the rhythmic feel of one of the band’s originators — and Danny’s personal mentor — Bernie Worrell. “Bernie was the original keyboard player for Parliament Funkadelic, and he is cited as the chief architect of the type of pocket that P-Funk works with. Cordell ‘Boogie’ Mosson pointed to Bernie as the one who really started it all for them in terms of how they looked at the pocket: where to put the one and where to put the back beat. With Bernie, it all literally came from his hips and his fingers — how he moved, and how he felt each note.”

And, though Bernie Worrell is sadly no longer alive, his influence lives on today through the material of George Clinton and Parliament Funkadelic, who continuously tour to packed houses around the world. “When we’re on stage, and everyone is hitting the same pocket,” says Danny, “it’s one of those things when you feel like ‘oh it’s all fitting right now’. When every single piece of the band is completely in a perfect cycle, everybody’s feeding everybody, and the rhythm is doing what it’s supposed to do naturally, it’s an incredible moment. That’s the part with the pocket that is less music theory, less maths, and more ‘spiritual’, or whatever you want to call it.”

While there is some difference of opinion among musical types as to whether the pocket can be learned, Danny’s viewpoint is optimistic. “A lot of people believe in the tenet that you can either hear and play the pocket, or you simply can’t. I agree with that for the most part, except that I think a person who has a basic idea of the concept can improve their ability. Basically, if you can figure out how to feel the pocket, then you can learn how to play it — it’s like dancing.” Perhaps the hardest part in personal pocket improvement is letting go, says Danny. “As long as the learner approaches the pocket with the same open mind that, say, a classical musician would approach learning a new piece, it’s all good. You have to completely surrender yourself to the music.”

“To feel the pocket, all that’s required is listening.” — Danny Bedrosian

The Pocket In Electronic Music

Speaking about the pocket in non-instrumental music, Bedrosian acknowledges the challenges in the early days of rhythmic machines. “I think the first incarnations of electronic music-making devices were mostly what would be considered ‘funk-less’ or ‘pocket-less’, although there were exceptions, like the Linn Drum and suchlike. The sampling era brought pocket back to electronically created music, allowing it to feel the same way.” Danny again uses another James Brown song as an example, “If you have a sample of ‘Funky Drummer’ in your song, you know that the pocket is there and it’s going to be there whether it’s a sample or the real thing. But, when making beats from scratch, the onus of making the machine have an organic feel is on the programmer. The operator has to have the wherewithal, the knowledge and the desire to inject that human feel into their music.”

There are of course countless examples of electronically made music that carries the hallmarks of the pocket — think J Dilla, Just Blaze and entire tomes of music such as Detroit Techno, where Motown roots rarely lay deep behind the electronic facade. Danny adds, “you heard it in some of the music erroneously called ‘neo soul’, where you had live drum beats mixed with electronic instrumentation. But in those cases, the programmer had a working knowledge of pocket on an electronic level, so they could create beats that are wholly electronic, but which had some, if not all, of the characteristics of good pocket in organic music.”

From Folk To Funk

As for how the pocket manifests itself in Danny’s keys playing, he stresses the importance of the ensemble playing as one and concentrating on every note, moment to moment. And it’s not just playing the P-Funk. “In Armenian, Arabic or Persian folk music,” remarks Danny, “a vocalist might sing a phrase that will pull the whole band back so far that they’d all have to be back on the one so deep in the pocket because of how that line was delivered.”

In the case of George Clinton — hands down one of the funkiest humans ever — the cues come from a variety of sources. “George will signal where in the pocket he wants you to play all the time, although it could be with a hand gesture, or a clap, or a walk, or with the back of his right foot, or with the microphone. He’ll do it with any number of different movements and motions, which you as the player — especially the rhythm section — has to follow. Period. If you don’t, then you’ll get caught up and have to remedy yourself really quickly — if you know how!”

“But when we’re all in the pocket, we definitely feel it when it’s good and when it’s right. And by extension, the audience feels it — either viscerally from the players if they’re sensitive to it, or from the eyes and the faces of the players — and that makes them happy. To feel the pocket, all that’s required is listening.”

Danny Bedrosian has has been a permanent fixture in the George Clinton & Parliament Funkadelic band since 2004, and takes his Novation Bass Station II with him everywhere he goes.

The latest album with his band Secret Army is EIGHTFINITY. Check it out on Spotify here.

To hear his experimentations with the pocket in electronic music, check out his 2015 album The Clock, where he deliberately programmed his beats with the pocket at the forefront.

P-Funk Rhythm Theory was originally published in Novation // Notes on Medium, where people are continuing the conversation by highlighting and responding to this story.

Focusrite Interfaces Help Create a Fast, Efficient and Cost-Effective Networked Environment For The Formosa Group

If you’ve ever stayed at the end of the movie to see the credits, you’ve seen their names: Supervising Sound Editors, Re-Recording Mixers and Sound Designers like Karen Baker Landers, Lon Bender, Doug Hemphill, Per Hallberg, Scott Hecker and Mark Mangini. These and other talented individuals now constitute The Formosa Group, a collective of professionals who are cornerstones of the sound teams for titles like Game of Thrones, Transparent, SWAT, Scorpion and other hit television and film properties. Brought together by the rapidly changing dynamics of Hollywood’s culture and economics, The Formosa Group brings massive technical and creative talent to bear on the content explosion taking place in recent years. To keep up with demand, The Formosa Group turned to RedNet Dante-networked audio interfaces from Focusrite to bring increased efficiency and effectiveness to their expanding campus of facilities.

In recent months, as The Formosa Group has expanded to seven locations around LA, they have invested deeply in RedNet, acquiring 38 RedNet HD32R 32-channel HD Dante network bridges, 10 RedNet D16R 16-channel AES3 I/O’s, a RedNet A16R 16-channel analogue I/O interface, and a pair of RedNet 6 MADI bridges, which complement the dozen-plus RedNet AM2 stereo audio monitoring units already in place and the existing RedNet 2 16-Channel A-D/D-A interfaces that have been working in facilities such as Formosa NoHo in North Hollywood and Formosa Santa Monica.

“Our mantra has always been ‘simplicity,’” says Bill Johnston, Vice President of Engineering at The Formosa Group. “That’s what the RedNet technology does for us — it makes things simple, transparent and very reliable.” Johnston points to a common signal path now used in several of The Formosa Group’s facilities, in which the output of all of the studios’ Avid Pro Tools HD systems are interfaced through RedNet HDR32 units, then into a BSS Soundweb London BLU-806 Dante-compatible processor and then into the Crown DCI Series amplifiers used to power the facilities’ monitoring systems. “We’ve been using the BSS processors as the Dante interface for the amplifiers, but now that the newer Crown amps are Dante compatible, we’ll be able to go directly into them in the future just using the AM2 units,” Johnston explains. “At that point, we have a digital audio signal all the way until it reaches the actual speakers.”

Johnston says putting the facilities on RedNet has streamlined their workflow significantly, in the process bringing costs down. “Once the audio is on a network, we can pull it from anywhere to anywhere just by bussing it, so we don’t have to send it through a router,” he says. “That eliminates the need for, and the expense of, MADI routers and MADI interfaces.” In fact, he says, selling the studios’ old MADI infrastructure has helped pay for more RedNet throughout the new facilities. And since RedNet has its own internal clocking — the RedNet A16R also outputs word clock — it can sync easily with the rest of the digital infrastructure in each studio, further expediting workflow. “We’re using multiple Pro Tools systems but through RedNet, we only need a single Sync I/O for everything. In fact, Avid told me, ‘You can’t do that — it won’t work.’ And I said, ‘But we’re doing it and it is working!’”

RedNet’s speed, accuracy and ease of use (Johnston says RedNet is “the very definition of plug and play”) make for an enhanced creative environment in which to work. “RedNet is fast, so it’s never in the way,” he says. “That’s always the biggest problem with any technology — it gets in the way of being creative because it takes some of your mental bandwidth to deal with the technical things. With RedNet, you don’t even know it’s there, and that’s the best thing you could ever say about any technology.”

RedNet will continue to be deployed throughout The Formosa Group’s facilities, including as the signal management for the Dolby Atmos speaker array for Formosa Santa Monica, the former Pacific Ocean Post facility. “It just makes things so much easier, by increasing out throughput and doing so totally transparently,” Johnston says. “And that lets us get down sooner to what we do best.”

Words: Clyne Media

Hollywood Post-Production Empire Expands With RedNet was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.

How Focussed Events Are Changing The Festival Landscape

The idea of packing your bags and traveling to a new place to experience a series of music and art performances is nothing new — in fact it’s been an important part of the modern cultural experience for many years. Woodstock in 1969 defined a generation, and events like Glastonbury and Lollapalooza, Coachella and Bonnaroo have come to epitomise the modern festival experience, attracting town-sized populations to remote locations on an annual basis. But in recent years, several small, specific and ultra-curated events have appeared on the scene, offering attendees personalised experiences that are often hard to find at the super-scale events. Ableton’s Loop festival is one such happening, now in its third year.

Held at Berlin’s spectacular Funkhaus venue, 2017’s Loop played host to around 2000 attendees, with programming over three days. The organisers describe Loop as “A summit for music makers… three days of discussions, performances, presentations, studio sessions, installations and interactive workshops aimed at exchanging ideas at the cutting edge of music, creativity, and technology.” Novation’s Marketing Manager (Germany & Austria) Alex Franz sums up Loop a being “very art-orientated.” He continues, “when Ableton does something, they always have the artist and the music at the center, and they do things in a very sensitive way. Loop is no different.”

This mindset manifests itself in a truly diverse lineup, with very little deliberate programming aimed at promoting the Ableton product itself. From UK beatboxer and Bjork collaborator SK Schlomo, Anticon founder DoseOne, and mastering engineer Mandy Parnell, to electro-acoustic pioneer Trevor Wishart, LA vocalist/producer Katie Gately and The Nile Project — a collective of multi-instrumentalists from the 11 countries that skirt the river Nile — Loop can hardly be categorised. But involved in every Loop artist’s journey — we assume through association — is Ableton.

Though the Loop experience is new to many, Ableton aren’t the first music company to start their own community gatherings. Synth originators Moog have been holding similar multi-venue, multi-day festivals under the ‘Moogfest’ banner, since 2010. (Moogfest-branded events actually started in 2004, but were in a conventional concert configuration until 2010.) Moogfest 2017 had a similar vibe to Loop — immersive workshops, presentations with contemporary talking points, and opportunities to meet artists face-to-face. Important aesthetic choices for both festivals include minimal branding, a lack of ‘VIP zones’ or exclusive areas for guests, and a general eye for detail and quality that is hard to achieve in, say, a field in Tennessee.

Loop and Moogfest are both a huge departure from the style of event that, for a while, looked to totally consume the festival circuit (and the disposable income of most attendees!). Human invasion-style migrations like Coachella, Bonnaroo and Reading/Leeds undoubtedly offer memorable experiences, but the artist is figuratively and literally as far away as they possibly can be from the attending public — behind a huge barrier, a security squad and a giant billboard in many cases.

Of course, peoples’ reasons for visiting festivals vary immensely. Many mainstream festival attendees go purely to have a good time with their friends; for some it’s just a rite of passage in their formative years. There’s nothing wrong with that. But for the more discerning customer — specifically those interested in making music, and the world that surrounds the DJ/artist/producer, Loop and other festivals offer an attractive set of opportunities, learning experiences and networking potential, which to some is a more rewarding way to spend a weekend and a few hundred dollars.

So, for those of you who want to splash out on a VIP ticket at Coachell-stonbury or Bona-palooza, and bask in the excess of the mainstream music industry, that’s cool. If you want to feel like part of a community, and grab a drink with the performers afterwards in a cool city like Berlin, maybe take a look at attending Loop 2018. We’ll see you there!

Ableton Loop & The Rise Of The Mini-fest was originally published in Novation // Notes on Medium, where people are continuing the conversation by highlighting and responding to this story.

Soulhack Shows Groovebox’s Individual Potential

Though Circuit makes a great companion product to other drum machines, samplers and sequencers, its solo potential is huge, as demonstrated in Soulhack’s Circuit performance video. We caught up with the London-based beatmaker to find out more about how he harnessed Circuit’s power to build a banger.

Soulhack’s performance starts with an identifiably Circuit-esque sound, which is sequenced from a pattern on Synth 1. But it’s not long before he gets stuck into some of his own samples. “This performance is very sample-heavy and there are several tricks I’ve used to make the most of Circuit’s potential,” he says.

To create this track, Soulhack used Circuit in conjunction with Ableton Live and the Components software application. He made several loops inside Ableton, then sent them to Circuit where he added synths and bass using the two onboard sound engines. The benefit to this approach, he says, is that he can use Live and various plugins to perfect his sound, while tailoring everything for performance on Circuit. “Circuit most importantly allows me to get away from a screen when performing music, which naturally makes me use my ears more.”

The track includes several samples: a garage-style shuffled hi-hat loop, a pitched-down vocal snippet, a kick drum, more hi-hats, a clap, a Rhodes loop and some additional effects. “I have impact/riser samples that give the track movement, but they are on one continuous eight-bar loop. I have to keep an eye on sample memory.”

“Circuit allows me to get away from a screen, which naturally makes me use my ears more.” — Soulhack

Circuit can hold up to 60 seconds of 44.1kHz WAV audio, and you can move samples between Circuit and a computer via USB, using the Components application. “At this tempo, an eight-bar loop is 15 seconds long, so I know I have 45 seconds of sample memory for everything else.” To conserve his sample time, Soulhack uses a trick that dates back to the earliest multitrack recording: bouncing, where two or more instruments share the same channel. “I’ve programmed a two-bar loop of the shuffle hats and clap together, to maximise the channel count. But I don’t just let it loop over and over; I will trigger it at different parts of the sequence to give it rhythmic variety, and I’ll experiment with the decay amount to shorten or lengthen the amount of time the sample plays for.”

“With such long loops,” says Soulhack, “it’s important to make sure the pattern length is long enough so that your samples play in full. For an eight-bar loop to play all the way through, for example, you need to program an eight-bar pattern by pressing and holding the first and eighth buttons in Circuit’s Pattern view.”

Soulhack experiments with pattern and step length while performing to add variation to his track, but he’s never more than a button push away from reverting to his original settings, thanks to his use of pre-programmed Sessions. “You can mess up the cycles and patches as much as you want on the fly. When you want to revert back to your original settings, just press the activated session button again.”

He explains more about his use of different Sessions to build the track. “Once you’re happy with the selection of sounds, patches and patterns, you can build different sections for your track and save them as individual Sessions. I usually have, for example, an intro, a drop, a breakdown and a second drop. It makes it super easy to arrange the track live, and also revert to your original parameters quickly.”

“Some of the best advice I can give you is to take your time. Although the Circuit workflow is very fast-paced, it pays off to spend time doing things like matching the shuffle of long, rhythmic samples to Circuit’s internal groove. Also, when you’re happy with all of your sounds in Ableton Live (or whatever DAW you prefer), EQ and compress them so they all sound as good as possible. Also, listen to how each of your sections will transition, and make sure the levels match. Doing this — rather than tweaking the sound on the Circuit — allows you to concentrate on the performance, not the sonics, when playing live.”

Find out more about Circuit here. Click here to download Components.

Going Solo With Novation Circuit was originally published in Novation // Notes on Medium, where people are continuing the conversation by highlighting and responding to this story.

Throwing Snow Puts Novation Sequence Synth Through Its Paces

There are two sides to Throwing Snow’s appreciation of Circuit Mono Station. The first is familiarity: as a user of the original Circuit, the London-based producer is already locked into the intuitive, screen-less workflow that Circuit Mono Station shares with its cousin. “Mono Station is really easy to get to grips with, because it’s the same workflow as the Circuit. But also, the top section is laid out like the Bass Station II. It’s very easy to swap between the two because it’s basically two of my favourite devices mashed together.”

This familiarity helps him when jamming and playing live, when he’s creatively zoned in to his machines. The movements to control Circuit and Circuit Mono Station are now programmed into Throwing Snow’s muscle memory; making adjustments to the instruments on the fly happens without conscious effort.

The second side to Throwing Snow’s admiration is almost contradictory to the first. An an artist, Throwing Snow thrives off Circuit Mono Station’s unpredictability in order to keep his creativity stimulated. Unlike a straightforward sequence synth, Circuit Mono Station has several functions that create creative variations in the behaviour of pattern playback, the arrangement of sequences, and the timbre of the instrument. Using these different pattern playback styles in combination with Circuit Mono Station’s paraphonic mode, he is capable of creating erratic, polyrythmical and trans-timbral sequences. “When the sequencer adds random notes together in paraphonic mode, it gives you something very interesting and unpredictable. Occasionally, I will sample one unique-sounding note because it sounds special. Other times, it might be that the entire pattern is really interesting, so I’ll keep the entire thing.”

Throwing Snow continues, “The paraphonic element of Circuit Mono Station has been really amazing for my creative process. It’s only now that I feel that the equipment has caught up to what I wanted to do!”

Find out more about Novation’s Circuit Mono Station here.

Circuit Mono Station: Unpredictability Where It Matters was originally published in Novation // Notes on Medium, where people are continuing the conversation by highlighting and responding to this story.

Answers To The Question, ‘How Many Ins & Outs Do You Need?’

When you’re deciding what audio interface to buy, it’s important that it can connect to your computer, and that its driver software will offer reliable, fast performance. But there’s more to the decision than that. Audio interfaces come in a bewildering variety of configurations, offering many different types of audio connector. You don’t want to waste money on features you’ll never use; and you don’t want to land yourself with an interface that can’t do what you need it to.

Let’s consider the other items you could connect to an audio interface. You might want to record audio from microphones, electronic instruments such as keyboards, samplers and synthesizers, guitars and basses, and perhaps turntables. At the other end of the chain, you may want to hear that audio over loudspeakers or headphones, or perhaps route it out into an outboard compressor, equaliser or effects unit.

What most of these items have in common is that they deal in analogue signals. To the physicist, sound is pressure waves in the air. An analogue signal represents these as a varying electrical voltage that corresponds (is “analogous” to) to the peaks and troughs in those waves. This stands in contrast to the digital information that passes between the audio interface and the computer, which consists of a series of numbers.

Parallel Lines

Electronic instruments and studio outboard typically pass these signals to each other at what is called ‘line level’, where the loudest peaks in a signal might reach a few tens of Volts. This is a convenient standard for conveying audio cleanly from source to destination, and it’s also roughly what the analogue-to-digital converters in our audio interfaces expect to work with. However, not all the devices that make and capture sound can generate such a wide voltage swing. Most electric guitar pickups and many microphones are passive devices that convert sound energy to electricity but don’t amplify that signal before passing it on; and since there isn’t actually very much energy in sound waves, the electrical output from a pickup or microphone might peak at a few tens of millivolts. A signal like this needs to be ‘pre-amplified’ before it’s passed to an analogue-to-digital converter or a line-level device.

At these very low levels, various factors come into play that are not usually an issue with line-level signals. One such is impedance, the tendency of a circuit to oppose the flow of an alternating current. Impedance is measured in Ohms and varies with frequency; where you see an impedance measurement given, the value quoted is usually for a 1kHz sine wave signal. Most guitar pickups are designed to be connected to devices presenting a very high input impedance of at least 1MΩ; connect them instead to a typical microphone input, which presents an impedance of perhaps 1.5kΩ, and the sound is likely to be compromised.

Another issue that can arise with any analogue signal, but especially with low-level signals, is interference. We’re surrounded by devices that generate electromagnetic fields, such as the ‘wall wart’ power supplies that come with many bits of studio equipment. If these are placed near cables carrying an audio signal, these fields will generate electrical energy in the cable, which will be audible as hum or buzz imposed on the wanted signal.

Sensible placement and routing of cables can help to keep interference to a minimum. So too can the use of ‘balanced’ cables and connectors, which are standard on a lot of audio equipment. The idea behind balancing is that, as well as a ground or screen connection, we run the wire carrying the signal alongside a second wire that will pick up the same interference. The destination device then ‘sees’ not the absolute voltage in the signal wire, but the difference between the voltages of these two wires, meaning the interference is cancelled out and we are left with just the wanted signal. Creating a balanced connection requires a connector with three electrical contacts rather than just two.

Jacks And XLRs

Two main types of connector are used to transfer analogue signals. One is the quarter-inch (6.35mm) ‘jack’. Cables of this form have identical plugs at either end, and are reversible; the devices they connect have identical sockets on input and output. Jack connectors and cables can be balanced or unbalanced, and you can tell which is which by inspecting the plug. The additional wire in a balanced connection is linked to an extra ring on the barrel of the plug. It is thus called a TRS connector, for ‘tip-ring-sleeve’, while the unbalanced version is just a TS. Depending on how balancing is implemented, using an unbalanced cable to join a balanced input and output may work, as will the opposite arrangement, but you’ll only get the benefits of balancing if all three are balanced.

The other type of connector in universal use is the XLR. Unlike the jack, this is a ‘gendered’ connector, and XLR cables typically have a ‘male’ connector at one end and a ‘female’ one at the other. The male connector has a circular barrel enclosing two or more pins, which mate with corresponding sockets in the female connector to make an electrical connection. XLRs are available in many different configurations, but the only one widely used in audio is the three-pin version, which is almost invariably used to create a balanced connection. The usual convention is that signals are output on male XLRs, with input sockets being female.

In practice, microphones almost always have XLR connectors, while guitars and basses use unbalanced quarter-inch jacks. Line-level connections are most often made using balanced quarter-inch connectors, though it’s also fairly common for studio gear to use XLRs and, occasionally, unbalanced connectors.

Three-way connectors don’t have to be used to transfer a single balanced audio signal. Alternatively, they can be made to carry two separate unbalanced signals. The quarter-inch TRS jacks used on headphones, for example, carry one stereo channel on the tip and the other on the ring, with the sleeve providing a common ground reference. Some audio interfaces also offer ‘insert points’ where a TRS jack combines an unbalanced output and input on the tip and ring.

You may also encounter a few other types of connector in use for analogue signals. RCA phonos are inherently unbalanced, and are mainly used on hi-fi equipment; a miniaturised 3.5mm version of the TRS jack connector is sometimes used, especially for headphones; and where many channels of audio need to be fitted into a small space, specialist multi-way connectors such as D-Subs and EDACs are employed. XLRs and RCA phonos are also sometimes used to transfer digital signals, as we’ll see in a future article.

Finally, analogue inputs on audio interfaces sometimes use the so-called ‘combi’ connector, which incorporates a TRS jack socket in the centre of a female three-pin XLR. These are usually configured so that the TRS socket accepts line-level signals while the XLR offers a connection point for mics; in some cases the TRS can also be switched into a high-impedance mode suitable for plugging guitars in.

Microphones, Gain & Phantom Power

As we’ve seen, the relatively small signals put out by microphones and guitar pickups require pre-amplifying to get them to the level where an analogue-to-digital converter can get to work on them. These signals also vary considerably in level in the first place: a capacitor mic used on a snare drum will put out a much ‘hotter’ signal than a ribbon microphone pointed at an acoustic guitar. For this reason, it’s essential to be able to vary the amount of preamplification that can be applied, so any XLR input designed for directly connecting a microphone will have a means of controlling the ‘input gain’.

The amount of preamplification is usually set using a physical control that operates in the analogue domain, but some interfaces such as Focusrite’s Red 4Pre and Red 8Pre feature digitally controlled preamps. These have the big advantage that input gain can be set and recalled precisely, which is very useful when you want to match the amount of preamplification you’re applying across two channels for stereo recording, or in case you need to reproduce a particular recording setup later on. Sometimes a variable gain control works in conjunction with a ‘pad’ switch, which applies an additional fixed degree of attenuation to help cope with very ‘hot’ signals.

Many interfaces that have microphone preamps also offer useful ways of processing the input signal. These can include high-pass filters, which remove unwanted rumbles and other low-frequency noise, and switches to reverse the polarity of the signal. Some interfaces also feature the ability to vary the input impedance of the preamps, which can introduce subtle changes to the tone with some types of microphone; this is part of what the Air controls found on Focusrite’s Red and Clarett interfaces do.

Finally, capacitor (‘condenser’) microphones need to draw power in order to work. This is usually supplied by means of a standard called ‘phantom power’, so called because it is carried on the same cable that connects the microphone without affecting the mic signal. Almost all mic preamps built into recording interfaces can supply +48V phantom power, but some allow it to be switched on and off for individual channels, while others make this a global setting. You are usually advised to leave phantom power switched off for mics that don’t need it, especially ribbon mics, though the risk of damage is not as great as it’s often made out to be.

Interfaces that draw their own power solely from the host computer over USB or Thunderbolt are often limited in their ability to supply phantom power to microphones. They can also struggle to drive some headphones as loud as you might wish.

Connector To Computer

Inside an audio interface, the analogue inputs feed analogue-to-digital converters, each of which generates a stream of data that can be recorded on the attached computer. At the other end, digital-to-analogue converters feed the analogue outputs, which can be connected to headphones, loudspeakers and other studio gear. However, there isn’t always a one-to-one correspondence between connector, converter and recording channel.

If an interface offers eight XLR inputs, it will certainly have eight A-D converters and recording channels, allowing the signals from up to eight microphones to be recorded simultaneously to separate tracks. However, if that same interface also offers eight jack inputs, these might share the same converters and recording channels, in which case you couldn’t also record eight line-level sources at the same time as recording the mics. On the output side, two headphone sockets might each have independent D-A converters, allowing them to be addressed separately in your recording software; but they might not!

What Do You Need?

To sum up, then, on any audio interface you’re likely to find some combination of the following analogue connections: balanced line inputs and unbalanced instrument inputs on quarter-inch jacks; microphone inputs on female XLRs; balanced line outputs on jacks or male XLRs; and headphone outputs on stereo, unbalanced jacks. So how many of each should you be looking for?

There are a lot of musicians and producers whose needs are fully met by quite a modest selection of I/O. For example, let’s take the case of someone who produces mainly electronic music, and whose sound sources are all software synths and samplers. Good low-latency performance and high-quality audio output are essential, but such a musician might have little need for audio input at all, and require only a stereo output to feed loudspeakers and a single headphone socket for mixing late at night. At the other end of the spectrum, someone who’s recording live bands using a large-format analogue console will need large numbers of line inputs and outputs to handle all the channels.

To help you evaluate your own needs, it’s worth asking yourself the following questions.

What other recording hardware do I want to integrate with my interface?

If your studio setup is going to consist only of a computer plus interface, with mics, instruments and loudspeakers attached, it’s vital that that interface can do everything you need of it. Unless you planned for the possibility when buying your interface, you’ll be stuck when you suddenly find yourself needing an extra mic or headphone feed in mid-session. On the other hand, if you’re going to be pairing your audio interface with an analogue mixer or with outboard mic preamps, you might actively prefer to not have preamps on the interface itself. Likewise, if you’re connecting multiple pairs of speakers to a single interface, you’ll need two line outputs for each pair of speakers you connect — but you might choose instead to use an outboard monitor controller, in which case you’ll only need to feed it a single pair of outputs.

On a related note, very few audio interfaces have more than two headphone outputs built in, so if several musicians are going to want to plug headphones in simultaneously, or in a different room, you’ll need to add an external headphone amp, which will require its own pair of line outputs from your interface. You may even want to set up different monitor mixes for different musicians, or to provide stems so that each musician can create his or her own cue mix, in which case further line-level outputs will be required.

As well as mixers and mic preamps, many people also have other hardware studio equipment that they want to integrate into their setup. For example, some people like to place compressors and equalisers in the input chain so that the sound they record is already processed in a way they find pleasing. Others want to incorporate such studio hardware at mixdown, so that tracks from their recording software can be sent out to rackmounting processors or even guitar pedals.

Some advocate the use of an external ‘summing mixer’, which takes the key elements of your mix on separate inputs and combines them in the analogue domain.

Studio processing and effects units almost invariably work with line-level signals. If your input signal is already at line level before it reaches your interface — as it will be if you’re using an external mic preamp, or if that signal is coming from a keyboard or sampler — you can easily run it through additional hardware before it meets the line inputs on your interface, in order to record it with compression or EQ. However, if you want to process the signal prior to recording using an external compressor or equaliser, but you don’t have a separate mic preamp, you’ll need an interface with insert points, so that these devices can be ‘patched’ into the line-level path between the interface’s mic preamp and its A-D converter. To incorporate external studio hardware at mixdown, you’ll need a spare line-level input and output for each channel of hardware you’re using — some hardware processors are mono only, others expect a stereo input, and some will deliver a stereo output from a mono input, while a summing mixer might take eight or sixteen line inputs and combine them to a single stereo output.

How many inputs will I need to record simultaneously?

Many musicians own multiple microphones, keyboards, guitars and other sound sources, but depending on how you work, you might need many fewer inputs to connect them to. For example, if you are a guitarist who is only ever likely to be working alone in the studio, you might never need more than two simultaneous inputs into your recording software — no matter how large your guitar collection, you can only play one of them at once! The same might be true of a keyboard player, too, but not necessarily: you might decide you want to use a sequencer to play multiple synths and sound modules at once. If that’s the case, you’ll need either sufficient line-level inputs to plug them all in at once, or an external mixer.

Most techniques for miking drums involve a stereo pair to capture the whole kit plus spot mics on bass drum, snare and toms, so if you’re going to be recording drum kits on a regular basis, eight XLR inputs is a sensible minimum.

For tracking an entire band live, you can probably double that figure at least, but there are few single audio interfaces that have more than twelve mic inputs built in. Recording on this sort of scale is more usually catered for either using modular systems such as Focusrite’s RedNet range, which allows additional mic preamp units to be added as needed, or by using an analogue mixer as a ‘front end’ for an audio interface with lots of line inputs.

Audio interfaces naturally get more expensive as you add more analogue inputs and outputs, so if you anticipate that you might want to do more ambitious projects further down the line, a sensible approach is to invest in an interface that meets your current needs but offers potential for expansion at a later date. Fully modular systems such as RedNet or Pro Tools HDX offer the greatest flexibilty in this department, but many other interfaces offer ways of adding further analogue inputs and outputs. For example, S/PDIF, ADAT and MADI digital sockets allow additional preamps and A-D or D-A converters to be attached, and some interfaces can be ‘aggregated’, meaning that you can connect multiple units together to create one ‘super interface’. We’ll be explaining these expansion options in more detail in the next part of this series.

Further Reading: Set To Surround

For many years, surround sound was only really needed for large-scale film and TV production, but there are signs that it is at last becoming more relevant to project studio owners. If you’re going to mix in surround, you’ll need an interface that has an analogue line output for each speaker you plan to connect — for a typical 5.1 setup, that means at least six analogue line outputs, along with another pair for a separate stereo reference if you like to work that way. It’s also worth making sure that any volume control on the interface can be configured to work across all of the analogue outputs at once, not just the first pair.

Further Reading: Connecting Turntables

Many people want to record from a record player, either for sampling or for converting their vinyl collection to digital form. However, if you simply connect the audio output of a turntable to a pair of line inputs on your audio interface, you’ll get very disappointing results: not only will the signal be too quiet, but it’ll sound very thin, and may be plagued with buzz and hum.

To overcome some of the limitations of vinyl as an audio format, treble frequencies are heavily boosted when a record is cut, and the correct frequency balance is restored on playback, when preamplification is also needed to bring the signal up to a healthy level. Many turntables also have a separate ground wire which needs to be connected to an earthing point to eliminate mains hum. If you have a hi-fi system that includes an amplifier with turntable inputs, all of these features will be built into that, and you should record the line output from the amplifier rather than connecting the turntable directly to your audio interface. If you don’t, you’ll need a separate phono preamp.

Words: Sam Pryor

This content is part of Focusrite’s System Science series. Read more here.

System Science #3: Analogue Connections was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.

There’s More To An Audio Interface Than The I/O

Modern computers are fantastic recording devices. They can work with more audio and MIDI tracks than we’re ever likely to need. They allow us to manipulate audio in ways the engineers of 30 years ago could only dream of. They let us apply EQ, compression and effects to more channels than would be possible in any analogue studio. Yet it’s important to remember that computers are not built specifically for recording. There are challenges that have to be overcome in order for all this to be possible, and issues arising that were never a problem when we recorded to tape. The biggest of these issues is latency: the delay between a sound being captured and its being heard through our headphones or monitors.

From The Beginning

Let’s consider what happens when we record sound to a computer. A microphone measures pressure changes in the air and outputs an electrical signal with corresponding voltage changes. This is called an ‘analogue’ signal, because the the variations in electrical potential are analogous to the pressure fluctuations that make up the sound. A device called an analogue-to-digital converter then measures or ’samples’ this fluctuating voltage at regular intervals — 44,100 times per second, in the case of CD-quality audio — and reports these measurements as a series of numbers.

This sequence of numbers is packaged in the appropriate format and sent over an electrical link to the computer. Recording software running on the computer then writes this data to memory and to disk, processes it, and eventually spits it out again so that it can be turned back into an analogue signal by, you guessed it, a digital-to-analogue converter. Only then, assuming we’re monitoring what we’re recording, do we get to hear it.

This is quite a complex sequence of events, and it suffers from a built-in tension between speed and reliability. In a perfect world, each sample that emerges from the analogue-to-digital converter would be sent to the computer, stored and passed back to the digital-to-analogue converter immediately. In practice, however, this makes the recording system too sensitive to interruptions. Even the slightest delay in sending just one out of the millions of samples in an audio recording would cause a dropout.

To make the system more robust, we don’t record and play back each sample as soon as it arrives. instead, the computer waits until a few tens or hundreds of samples have been received before starting to process them; and the same happens on the way out. This process is called buffering, and it makes the system more resilient in the face of unexpected interruptions. The buffer acts as a safety net: even if something momentarily breaks up the stream of data coming into the buffer, it’s still capable of outputting the continuous uninterrupted sequence of samples we need.

The larger we make these buffers, the better the system’s ability to deal with the unexpected, and the less of the computer’s processing time is spent making sure the flow of samples is uninterrupted. The down side is that the larger we make these buffers, the longer the whole process takes; and once we get beyond a certain point, the recorded sound emerging from the computer starts audibly to lag behind the source sound we’re recording. In some situations this isn’t a problem, but in many cases, it definitely is! Where musicians are hearing their own and each others’ performances through the recording system, it’s vital that the delay never becomes long enough to be audible.

A delay between sound being captured and its being heard again at the other end of the recording system is called latency, and it’s one of the most important issues in computer recording. The time lag between playing a note and hearing the resulting sound through headphones is highly off-putting to musicians if it’s long enough to become audible, so this needs to be kept as low as possible without using up too many of the computer’s processing cycles. There are several different factors that contribute to latency, but the buffer size is usually the most significant, and it’s often the only one that the user has any control over.

Numbers Game

Buffer sizes are usually configured as a number of samples, although a few interfaces instead offer time-based settings in milliseconds. The choices on offer are normally powers of two: a typical audio interface might offer settings of 32, 64, 128, 256, 512, 1024 and 2048 samples. You can calculate the theoretical latency that a particular buffer size setting will give you by doubling this number — to reflect the fact that audio is buffered both on the way in and the way out — and dividing the result by the sampling rate. So, for example, at a standard 44.1kHz sample rate, a buffer size of 32 samples should in theory result in a round-trip latency in seconds of (32 x 2) / 44100, which works out at 1.45 milliseconds.

A latency this low would be completely imperceptible in practice, but unfortunately, it can’t be realised. For one thing, there are other factors that contribute to latency apart from the buffer size, and some of these are unavoidable (see box). For another, some audio interfaces ‘cheat’ by employing additional hidden buffers that are outside the user’s control. This means that although they might report very low latency figures to the recording software, these figures are not actually being achieved. Most importantly, however, reducing the buffer size forces the computer to devote more of its processing power to managing the audio input and output, and if we go too far, we risk running out of processing resources. One of the key challenges of audio interface design is to ensure that it’s actually possible to use low buffer sizes in practice, and there’s a lot of variation in how well different interfaces meet this challenge.

Misreporting of latency also brings problems of its own, especially when we want to send recorded signals out of the computer to be processed by external hardware. If we want any dry signal mixed in, as might be the case with parallel compression, this will be out of time with the processed signal, resulting in audible phasing and comb filtering. In order to line up the wet and dry signals correctly, the recording software needs to know the exact latency of the recording system.

What Is A Driver?

In order for a meaningful transfer of data to take place between a computer and an attached interface, the computer’s operating system needs to know how to talk to it. This is made possible by software that interposes itself between the hardware and the operating system or recording software, and which includes a low-level program called a ‘driver’. (Technically, the driver is only a small part of the code that enables recording software to communicate with recording hardware. However, it’s common usage to refer to this code collectively as the ‘driver’.)

Computer operating systems usually come with a collection of drivers for commonly used hardware items such as popular printers, as well as generic ‘class’ drivers, which can control any device that is compliant with the rules that define a particular type of device. The USB specification, for instance, defines a class called ‘audio interface’. In theory, a hardware manufacturer could build a USB interface that met this class definition and not have to worry about writing drivers for it — although, as we shall see, there is more to it than this. Where no class driver is available, or where better performance is needed, a driver needs to be specially written and installed.

The driver and related software are critically important to achieving good low-latency performance. Well-written driver code manages the system’s resources more efficiently, allowing the buffer size to be kept low without imposing a heavy load on the computer’s central processing unit. The importance of drivers means it’s not possible to simply say that one type of computer connection is always better than another for attaching audio interfaces. Any technical advantage that, say, Thunderbolt has over USB is only meaningful in practice if the manufacturer can exploit it in their driver code.

Mac Vs Windows

In general, when software needs to communicate with external hardware, it does so through code built into the operating system, which in turn communicates with the driver for that particular device. One reason why Apple computers are popular for music recording is that Mac OS includes a system called Core Audio, which has been designed with this sort of need in mind. Core Audio provides an elegant and reasonably efficient intermediary between recording software and the audio interface driver. It supports essential features like multi-channel operation and does not add significant latency of its own. Mac OS even includes a built-in driver for class-compliant USB audio devices which offers fairly good performance, so many manufacturers of USB interfaces choose to use this rather than writing their own.

Historically, this stands in contrast with the audio handling protocols built into Windows, such as MME and DirectSound. These not only add to the latency, but lack features that are vital for music production. So, when Steinberg developed the first native Windows multitrack audio recording software, Cubase VST, they also created a protocol called Audio Streaming Input Output. ASIO connects recording software directly to the device driver, bypassing the various layers of code that Windows would otherwise interpose. At the time when ASIO was developed, there was no other way of conveying multiple audio streams to and from an audio interface at the same time. More recent versions of Windows have introduced newer driver models and protocols, but ASIO remains a near-universal standard in professional music software. Note that as it’s not a Microsoft standard, Windows doesn’t include any ASIO drivers at all, so even class-compliant devices must be supplied with an ASIO driver for use with music software that expects to see one.

Writing efficient low-level software such as drivers and ASIO code requires specialist skills and expertise, and once written, they need to be maintained to remain compatible with the latest version of each operating system. This is a significant burden on manufacturers of audio interfaces, and many of them choose to license third-party code instead of writing their own. In the case of USB devices under Mac OS, as we’ve seen, this code is already built into the operating system; in other cases, it’s usually developed by the manufacturers of the chipsets — the set of components on the audio interface that handles communication with the computer.

This has obvious advantages for the manufacturer, but it also creates a chain of dependence which can cause problems. For example, most FireWire audio interfaces used a chipset designed by TC Applied Technologies, and licensed driver code from the same manufacturer. Eventually, this code became highly optimised and offered very good low-latency performance; but it took many years to reach this point, and in the meantime, there was little manufacturers reliant on that code could do to improve things.

It might not be obvious whether your audio interface uses a custom driver or a generic one, because the driver code operates at a low level and the user does not interact with it directly. Almost all recording interfaces come with a separate program, sometimes called a ‘control panel’, to provide user control over the various features of the interface. These control panel programs are invariably written by the audio interface manufacturers, so the fact that two interfaces each have a unique control panel utility does not mean that they don’t share the same generic driver code.

Avoiding Latency

How much latency is acceptable? There’s no simple answer to this question. Sound travels about one foot per millisecond, so in theory, a latency of 10ms shouldn’t feel any worse than moving 10 feet away from the sound source — and guitarists on stage are often further than 10 feet from their amps. Nevertheless, many players complain that even this amount of latency is detectable; and there are situations where much smaller amounts of latency are audible. In any situation where a player or singer is hearing both the direct sound and the recorded sound, for example, any latency at all will cause comb filtering between the two.

When we’re using a MIDI controller to play a soft synth, the audio that’s generated inside the computer has only to pass through the output buffer, not the input buffer. In theory, this should mean the contribution of audio buffering to latency is halved, but in practice, the process of getting MIDI data into the computer also adds latency to the system. The amount of data involved is tiny compared with audio, but it still has to be generated at the source instrument, transmitted to the computer (usually, these days, over USB) and fed to the virtual instrument that is making the noise. All of these steps take a finite amount of time, and there is also the potential for ‘jitter’, whereby the latency is not constant but varies by a few milliseconds. MIDI latency is unlikely to be noticeable if you’re playing string pads from a keyboard, but it can be an issue where you’re triggering drum samples from a MIDI kit. Any system that employs pitch-to-MIDI detection, such as a MIDI guitar, is also prone to noticeable latency on low notes, as it needs to ‘see’ an entire waveform cycle in order to detect the pitch.

Avoiding The Issue

No digital recording system can be entirely free of latency. The only way to avoid latency altogether is to create a monitor path in the analogue domain, so that the signal being heard is auditioned before it reaches the A-D converter. This is common practice in large studios, where an analogue mixing console is often used as a ‘front end’ for a computer-based recording system. There are also small-format analogue mixers designed for the project studio that incorporate built-in audio interfaces. With this sort of setup, the mixer’s own faders and aux sends can then be used to generate cue mixes for the musicians which do not pass through the recording system at all, and thus are heard without any latency.

This type of arrangement has a lot to recommend it when you’re recording bands live. It makes it easy and quick to set up multiple different monitor mixes that can be routed to separate headphone amps, with no latency issues at all. However, not everyone has the space or budget for an analogue mixer and associated cables, patchbays and so forth. Routing signals through an analogue console can also affect sound quality, especially if it’s a budget model, and many people prefer the cleaner and simpler signal path you get by plugging mics and instruments directly into the audio interface.

With this in mind, most manufacturers build cue-mixing capabilities directly into their audio interfaces, recreating the same functionality but in the digital domain. Setting up these built-in digital mixers is usually the main function of the control panel utilities described earlier. This has the advantages of being much cheaper to implement, requiring no additional space or cabling, and not degrading the sound that’s being recorded. On the down side, although this approach reduces latency to levels that are usually imperceptible, it doesn’t eliminate it completely: the signal still passes through the A-D and D-A converters before it’s heard, and in a few cases, the digital cue mixer itself can introduce latency.

No Way Out

For the last fifteen years or so, almost all audio interfaces designed for multitrack recording have incorporated a digital mixer to handle low-latency input monitoring, as described above. However, the fact that it’s a widely used way of managing latency doesn’t mean that it’s the best way, and there are several problems with this approach.

The first issue is that it adds to the complexity of the recording system. Rather than working entirely within a single recording program with its own mixer, the user is forced to constantly switch back and forth between recording software and the interface’s control panel utility. As a result, sessions take longer to set up, troubleshooting is more difficult, and there’s no way to use the cue mixes configured in the audio interface mixer as a starting point for final mixes in the recording software. Tracks in your recording software have to be muted during recording, to avoid hearing the same signal twice, but unmuted when you want to play them back, and not all DAW software allows this to be done automatically. Moreover, many digital cue mixers and control panel utilities are poorly designed, inconsistent or difficult to use.

Attempts have been made to tackle this problem by allowing the recording software’s mixer window to control the low-latency mixer in the interface. Steinberg’s ASIO Direct Monitoring is probably the most widely supported of these, but it is far from being a universal standard, and other solutions require the user to choose both hardware and software from the same manufacturer. Moreover, none of these address the remaining issues with this approach to avoiding latency.

One of these is that in any setup where a separate mixer is being used to avoid latency, the signal is being monitored before it completes its journey into and through the recording system. This means that if any problem occurs further along in the recording chain, we won’t hear it until it’s too late. For instance, if we are monitoring input signals through an analogue console and the level is too hot for the audio interface it’s attached to, the recorded signal will be audibly and unpleasantly distorted even though what the artist hears in his or her headphones sounds fine.

Perhaps the biggest limitation with the workaround of using a mixer, though, is that it only works when the sound is being created entirely independently of the computer. An all-analogue monitoring path might be the best way for a singer to hear his or her own performance, but it’s of no use when we want to play a soft synth, or record electric guitar through a software amp simulator. It’s also no use when we want to give the singer a ‘larger than life’ version of his or her vocal sound through the use of plug-in effects. The only way to ensure that those sounds emerge promptly when we press a key or twang a string is to make the system latency as low as possible.

High Sample Rate, Low Latency?

As we’ve seen, the buffer size is usually set in samples. However, the duration of a sample depends on the sampling rate. So, if you’re recording at 88.2kHz, twice as many samples are measured and processed each second compared with standard 44.1kHz recording. In theory, then, doubling the sample rate should halve the system latency if you don’t change the buffer size, and this is sometimes recommended as a means of lowering latency.

In the real world, however, this is of limited use. The problem with most audio interfaces is not that low buffer settings aren’t available: it’s that they don’t perform as advertised, or that inefficient driver code maxes out the computer’s CPU resources at these settings. Doubling the sample rate also considerably increases the load on the computer’s resources, as well as generating twice as much data, so if a particular buffer size works for you at 44.1kHz, there’s no guarantee it will still work at 88.2 or 96 kHz. And in any case, we may want to choose a different sample rate for other reasons — most audio for video, for example, needs to be at 48kHz.

Ultimately, the only solution to the problem of latency that isn’t an undesirable compromise is to reduce it to the point where it’s no longer noticeable. This has been achieved in the live sound world, where major gigs and tours are invariably now run from digital consoles. The key to achieving unnoticeably low levels of latency in the studio is to choose the right audio interface: not only one that sounds good and has the features you need, but which will be capable of running at low buffer sizes without overwhelming your studio computer.

Modern computers are the most powerful recording devices that have ever existed. But if we can’t hear what we’re recording in real time, without cumbersome workarounds, we are not getting the full benefits of that power. We might even be going backwards compared with the tape-based, analogue studios of forty years ago.

Further Reading: Measuring Latency

Audio interfaces are supposed to report their latency to recording software, and you’ll usually find a readout of this reported value in a menu somewhere. However, it’s important not to take this value as gospel. Some interfaces do report the true latency, but many under-report the actual value.

There are various ways of obtaining a reliable measurement of system latency. The very best of these is to use an entirely separate recording system. Load up an audio file that contains easily identifiable transients — a click track is perfect — and feed this to two outputs on the measurement system. Connect one of these directly back to an input on the measurement system, and route the second through the system under test. When these two inputs are re-recorded, the latency will be visible as a time difference between them. This is the best way to be certain that all the possible factors contributing to system latency are taken into account.

If you don’t have a separate recording system handy, you can measure the round-trip latency by hooking up an output of your interface directly to an input (it’s a good idea to mute your monitors in case this creates a feedback loop). Again, you’ll need an audio file containing easily identified transients. Place this on a track in your DAW, route it to the output that is looped, and record the input that it’s looped to to an adjacent track. When you zoom in very closely, you’ll be able to see if the original and the re-recorded clicks line up. If they do, the latency that your DAW reports is accurate. If the re-recorded click is behind the original, then the true latency is equal to the reported latency plus the difference.

However, the latency alone isn’t the whole story. On a given computer, two interfaces might both achieve the same round-trip latency, but in doing so, one of them might leave you far more CPU resources available than the other. It is hard to find a completely objective way of measuring this trade-off between latency and CPU load, but by far the most thorough attempt is DAWBench. Created by Vin Curigliano, this assigns audio interfaces a score based on their performance on a fixed test system, evaluating not only the actual latency at different buffer sizes but also the amount of CPU resources available.

Further Reading: Other Causes Of Latency

As mentioned in the main text, buffer size is usually the most significant cause of latency, and it’s often the one that is most easily controlled by the user. However, it’s not the only factor that contributes to the latency of a computer-based recording system.

Some of these other factors are inevitable. The process of ‘sampling’ an incoming analogue signal and converting it to a stream of digital data takes time, and so does the digital-to-analogue conversion at the other end of the signal chain. These delays caused by sampling are very small — well under 1ms — and make little difference to the overall latency, but there are circumstances when they are relevant, particularly when you have two or more different sets of converters attached to the same interface. This is the case when, for instance, you connect a multi-channel preamp with an ADAT output to an interface that has its own preamps and converters. In this situation, converter latency can mean the two sets of signals are fractionally out of sync — not enough to be a problem if they are carrying different signals, but conceivably a problem if for instance a stereo recording was to be split between the two.

A less well-known fact is that recording software itself adds a small amount of latency. Focusrite’s measurements have shown that there is some variability here, with Pro Tools and Reaper being the most efficient of the major DAW programs, and Ableton Live introducing more latency than most. Again, though, the total extra latency is very small, and typically well under 2ms.

What’s better known is that audio processing plug-ins can introduce latency. Indeed, there is a common belief that they all do, but this is only true in products that use a hardware co-processor to handle plug-ins, such as the Universal Audio UAD2 and Pro Tools HDX systems. The vast majority of ‘native’ plug-ins — that is, plug-ins which run on the host computer — introduce no additional latency at all, because they only need to process individual samples as they arrive.

Processing plug-ins that add latency to the system typically fall into two groups: convolution plug-ins, including ‘linear phase’ equalisers, and dynamics plug-ins that need to use ‘lookahead’. In both cases, the plug-in depends on being able to inspect not just one sample at a time, but a whole series of samples. In order to do this, audio needs to be buffered into and out of the plug-in, adding further delay — and since most recording software applies ‘delay compensation’ to keep everything in sync, this delay is propagated to every track.

In general, it is therefore good practice not to introduce any plug-ins that cause delays until the mixing stage is reached, although not all recording programs make it easy to find out whether a particular plug-in adds extra latency. Some convolution plug-ins offer a ‘zero latency’ mode: this doesn’t actually eliminate the latency, but deliberately misreports it as zero to the host program, so that delay compensation doesn’t get applied.

Finally, although the digital mixers built into many audio interfaces typically operate at zero latency, there are a handful of (non-Focusrite) products where this isn’t the case — so it can turn out that a feature intended to compensate for latency actually makes it worse!

Words: Sam Pryor

This content is part of Focusrite’s System Science series. Read more here.

System Science #2: Drivers & Latency was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.

The Additional Factors That Contribute To A Great Interface

When you’re choosing an audio interface, core features such as sound quality, driver performance and expansion potential will be your biggest concerns. When you’ve figured out your needs in these areas, however, you might well find that there are quite a few products that seem to meet them — at least on paper. In this situation, it will often be other factors that guide your decision-making. In this article, we’ll look at some of the most important of these.

Form Factor

Broadly speaking, audio interfaces come in two forms. Most interfaces that offer more than a handful of inputs and outputs are packaged in 19-inch rackmount cases, a standard format that allows them to be placed neatly in a rack along with other studio equipment. Typically, all of the controls will be on the front, along with headphone sockets and perhaps a couple of audio inputs, while the majority of the connections will be on the reverse.

However, many home studio owners have no other rackmounting gear, and only need a couple of inputs and outputs, so smaller interfaces are normally designed to be placed conveniently at hand on a table top next to the computer keyboard. There’s less uniformity in the design of ‘desktop’ interfaces, and some manufacturers place controls on top of the unit as well as on the front panel. Desktop interfaces are smaller and more portable than their rackmount cousins, but this usually just reflects the fact that they have fewer inputs and outputs.

If your needs are fairly basic, or you’re setting up a bedroom studio, you might well find that a desktop interface is more convenient, but as your needs grow, there will come a tipping point where the advantages of the rackmounting format become compelling. It keeps all the messy cabling out the way, and when you do need to move your system around for location recording and so on, you can easily protect it in a robust flightcase.

Power Supply

Audio interfaces all need electrical power to work. There are three basic ways in which this can be supplied. Small interfaces with only a couple of inputs and outputs can be ‘bus powered’, drawing all the power they need from the USB or Thunderbolt connection to the computer. This has obvious advantages in terms of portability, and in particular, allows you to work with a laptop even where there is no mains power available. The flip side is that there simply isn’t enough power available over a typical USB connection to drive a more fully featured audio interface. Even basic interfaces often suffer from compromises such as lower headphone volume or the non-availability of phantom power for microphones when they are bus-powered.

Where bus powering isn’t an option, a power supply is required to bring the mains supply down to a voltage suitable for powering the interface’s circuitry. Many rackmounting interfaces have the power supply built in, meaning that all you need is a standard IEC mains cable (often wrongly called a ‘kettle lead’). Most desktop interfaces, however, use an external power supply resembling that used with a laptop or mobile phone charger. Many interfaces that are capable of being bus-powered come with such a power supply as an alternative, particularly where they can’t be operated at their full potential from bus power.

Using an external power supply allows the interface itself to be made smaller and to be put together more cheaply, but can be inconvenient and present long-term reliability issues. It also means that if the power supply is lost or broken, your interface becomes a doorstop until you can replace it — which is not always easy. Building the power supply into the case of the interface is generally a better solution, but it’s also more difficult and expensive from an engineering perspective, so internal power supplies tend to be a mark of premium interfaces.

Monitor Control

If you are planning on pairing your audio interface with an analogue mixing console, you probably won’t want lots of mic preamps built into it: you’d use the preamps on the console, and feed its channels or buses into line-level inputs on the interface. And at the other end of the signal chain, you’d expect to feed line-level outputs straight into the mixer and use its master section to handle volume control, speaker selection, mute, dim, talkback and so forth.

However, this sort of console-based setup is common only in reasonably large studios. Most audio interfaces intended for home and project studios are designed to be used without a mixer attached, so they have their own preamps built in. But what provision do they make for replacing the master section of an analogue console? The answer can vary quite a bit.

Nearly all audio interfaces that have analogue outputs have at least some way of controlling their level, almost always using rotary knobs. It’s important to note, though, that there are two types of knob in common use. One is the more traditional potentiometer (variable resistor), which has a fixed start and end point, and operates on the output signal after it’s been through the digital-to-analogue conversion stage. The other is the so-called ‘rotary encoder’. This is a digital control, and often has no end points to its travel, so it can be spun around and around until you get bored.

Unlike a potentiometer, a rotary encoder can be (and often is) assignable, meaning that the encoder has no one fixed function, but can be switched to change multiple parameters. In some cases it’s also possible to operate an encoder as a button to activate a secondary function. This means that a single rotary encoder could, for example, operate as a main volume control, a headphone volume control and a mute button.

Pots & Kettles

From both an ergonomic and a technical point of view, both designs have their pros and cons, although many of these are dependent on exactly how they’re used in a given product. Cheap potentiometers don’t always have a very linear response, which can make it hard to precisely set the level you want, and at low levels, level matching across channels can be imperfect. On the plus side, however, they always operate on the full-range digital signal, and if something goes badly wrong and your computer starts spitting out deafening noise, an analogue gain control will always give you a way of turning it down!

Digital level controls are more versatile: not only is a single control often assignable to multiple parameters, but you can control the level of multiple outputs at once, which is vital in a surround monitoring system. They can also be set more precisely, which is an advantage if you want to work in a calibrated monitoring environment. However, a volume control that works in the digital domain can compromise sound quality. By turning the output down before it reaches the D-A converter, you are reducing the word length of the digital signal, so you might find yourself listening to 12-bit sound! In other words, attenuating your signal in the digital domain reduces the dynamic range of that signal.

From an engineering and sound quality perspective, then, the best solution of all is a hybrid one, as employed in Focusrite’s Red interfaces. This uses a digital rotary encoder to control levels, but the control itself happens in the analogue domain; it’s of a similar nature to the digitally controlled preamps used in the Red interfaces.

If you’ve ever used a large-format analogue console, you’ll know there is a lot more to the master section than simple level control. By no means all master-section features are relevant to stand-alone audio interfaces, but some of them can be very useful, and it’s well worth considering which ones you might need. As well as mute, dim and mono fold-down, they include the ability to switch between two or more sets of monitor speakers, ideally with the option to apply a level offset to the second pair so as to match the apparent volume, and talkback, which allows the engineer to communicate with performers in another room. Relatively few audio interfaces implement all of these features, and those that do so often require them to be operated from the computer rather than the front panel, so if you have complex needs in this department, you might be better off buying a separate monitor controller to handle them.

Metering

Our most important tools for making judgements about recording quality are our ears, but we also rely heavily on our eyes, especially when it comes to tracking down faults and setting levels. For this reason, most audio interfaces offer at least some visual feedback on the signals that are going in and out of them, although there is wide variation in the form this takes.

Some interfaces offer very minimal metering, perhaps amounting just to a ’signal present’ LED on the analogue inputs. This makes them cheaper to build, and some would argue that extensive hardware metering is not really necessary in a computer-based recording context in any case. Our recording software, third-party plug-ins, and the control panel/mixer utilities provided with most audio interfaces offer much more detailed and comprehensive metering than could ever be provided on the front panel of a single-unit rack-mounting interface.

This is not an argument that convinces everyone, though, and there are situations where relying only on software metering can get you into trouble. It won’t help with troubleshooting in situations where no signal is appearing in your recording software, for example, and might mean you miss problems that occur in the analogue domain before the signal reaches the A-D converter. And many people simply prefer not to look at the computer screen all the time, or use it to view the recording software’s arrange window rather than its mixer.

The majority of multi-channel interfaces thus tend to feature ‘ladder’ metering displays with between five and ten LEDs for each channel. These are usually configured in a ‘traffic light’ colour scheme where the topmost LED lights up red to warn of clipping, meaning that you need to turn down the gain in order to make a clean recording. The next one or two will be amber, indicating that there is a risk of running out of headroom, and the remainder will be green. The exact levels at which each LED is triggered vary, but in general, you should aim to set gain controls such that signals only occasionally go into the amber and absolutely never into the red.

A typical multi-channel audio interface might have eight such ladder displays for its analogue inputs, plus an additional pair showing the levels at the main outputs. However, some also allow the meters to be switched so that they show the levels at the digital inputs, which can be useful.

In high-end interfaces such as Focusrite’s Red range, you’ll often find even more sophisticated and versatile metering options. Rather than using the simpler LED ladders, the Red 4Pre, Red 8Pre and Red 16 Line feature separate full-colour, high-resolution screens for metering input and output levels, along with a third that is dedicated to preamp and input settings. This means they come much closer than most interfaces to conveying all the detail that is on offer in software metering.

Less Is More

Many of the features we’ve looked at so far could be thought of as being primarily about ergonomics: their goal is to make the interface as fast and easy to work with as possible. If this is a consideration for you — and it should be — it’s vital to remember that the same considerations apply to any software that is needed to use the interface.

Many manufacturers make a big song and dance about the power and flexibility of the internal mixing features within their interfaces, and the control-panel utilities that are needed to configure them. Some of these are easily as deep and complicated as a stand-alone digital mixer — yet they completely lack the hands-on tactile control that a ‘real’ mixer provides. As a result, sophisticated control-panel utilities present an intimidating learning curve, and may well implement the same features as your recording software in confusingly different ways.

However, if your audio interface is capable of operating at very low buffer sizes, and hence without noticeable latency, there is absolutely no reason why you should need to learn how to use a complex mixing and routing utility. Focusrite believe all of that is better handled within your recording software, which is why the Focusrite Control utility that comes with all of Focusrite’s current products is intentionally simple and uncluttered.

Further Reading: Sweetening The Deal

Many audio interfaces come with a software bundle as well as the physical interface. This, naturally, includes any drivers and control-panel software needed to operate the interface, but it can go much further than this. It may, for instance, include recording software — usually a cut-down version of one of the well-known packages such as Steinberg’s Cubase — and/or effects and processing plug-ins. Most people already have recording software and don’t install the free, cut-down versions that are bundled with audio interfaces, but the plug-in packages can be a very valuable addition to your setup. Focusrite’s interfaces come with one or more of the Midnight, Scarlett and Red plug-in bundles, all of which offer premium-quality EQ and dynamics, while Focusrite’s Plug-in Collective programme offers an ever-changing range of excellent deals on third-party plug-ins.

Further Reading: Fan Friction

Almost all of the electrical power that is used by an audio interface ultimately ends up as heat, which needs to be dispersed in order to prevent damage to the components within the interface. Most interfaces manage to keep their internal temperature within safe limits through ‘passive cooling’, which means transferring the heat to the metal case that houses the interface, and thus into the air around it. However, this isn’t always sufficient, and in no-compromise designs such as the Focusrite Red range, more heat is sometimes generated than can be dispersed passively. For this reason, complex high-end interfaces often have an internal fan to drive cool air through the case. Care is usually taken to make these fans operate quietly, but they are rarely completely silent, and although this isn’t an issue when the interface is being used in a control room away from the main recording area, it can cause problems when you need to record quiet sources with the interface in the same room.

Words: Sam Pryor

This content is part of Focusrite’s System Science series. Read more here.

System Science #6: Physical Features was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.

Getting To Grips With Digital I/O, Expansion & Clocking

Audio interfaces don’t only have analogue connections. They also have digital inputs and outputs. With a few exceptions, digital inputs can’t be used to directly record musical instruments or microphones; their primary purpose is expansion. The role of digital I/O is to allow additional recording hardware to be attached, increasing the capabilities of the audio interface.

In other words, what digital inputs and outputs really do is make it possible to add extra analogue inputs and outputs. But why do this by a roundabout route? Why not just build more analogue I/O into the interface in the first place?

Every analogue input and output requires its own electrical circuitry and analogue-to-digital or digital-to-analogue converter, and takes up space on the front or back panel; so each one we add increases the cost of the interface and makes it harder to fit everything in. By contrast, digital expansion formats are incredibly compact. Depending on its format, a single digital input can accommodate two, eight, 56 or even 64 channels of audio — even more if we enter the world of audio over IP.

Digital expansion also leaves the user free to choose the additional input and output types that he or she needs. One person might find herself wanting an extra eight microphone inputs; another might need sixteen line-level inputs to connect a mixing console; someone else may wish to add more outputs in order to integrate studio hardware at mixdown. All of these possibilities can be catered for by a couple of tiny digital connectors.

Clock Wise

Digital audio consists of a series of ‘samples’, measurements of the level of a signal that are taken at regular intervals. In the case of standard CD-quality audio, there are 44,100 samples per second. To maintain this regularity, every digital audio device needs a timing reference or ‘clock’.

If we’re just using a single audio interface on its own, it will happily run from its own internal clock. Likewise, there are no clocking issues involved in connecting other hardware in the analogue domain. However, consider what happens when we digitally connect a second device, such as a mic preamp with a digital output, to our audio interface. In this case, the interface receives a stream of samples that are spaced not according to its own clock signal, but to that of the mic preamp. For a while, we might be lucky and find that the two clock signals run in sync, but as no two clocks are absolutely identical, one of them eventually will fall slightly behind the other, and the interface will no longer be able to make sense of the data it’s receiving.

The only way for digital audio to be transmitted successfully from one device to another is for the two devices to share the same clock signal. Or, to put it another way, in any studio where devices are connected digitally, one of them needs to be the clock ‘master’, and all the others must be clock ‘slaves’, accepting their timing reference from the master clock. The ways in which this is done vary depending on how many devices we’re connecting, and in what way.

Socket To Me

For reasons that are partly functional and partly historical, you can find several different types of digital input and output on typical audio interfaces. And just as in the world of USB cables, there is a confusing relationship between the type of signal that’s being carried, and the type of connector used to link the sending and receiving device.

One of the best-established standards in the studio world is AES3, sometimes known as AES/EBU. Each AES3 connection can carry a stereo, 24-bit digital audio signal at sample rates of up to 96kHz. The way in which this stream of data is formatted is always the same, and this formatting is the main feature that defines the standard; but the physical arrangements used to transmit it can vary. Older high-end studio gear, including audio interfaces, typically has AES3 inputs and outputs on three-pin XLR sockets. These are identical to the sockets used for connecting microphones to analogue inputs, and indeed, it’s possible to use standard microphone cables to send AES3 signals over short distances.

However, signals encoded in the AES3 data format can be sent over other forms of connection, too. Where long cable runs are needed for large broadcast installations and so on, a circular connector called a BNC socket is often used. At the other end of the scale, in home and project-studio gear, the humble RCA phono socket is commonly found. AES3 data sent over phono sockets is often called ‘S/PDIF’, reflecting its origins as the Sony and Philips Digital Interface Format.

All three variants mentioned above represent the AES3 data electrically, but there is also a variant that represents it optically. Sometimes known as TOSlink, this carries exactly the same data in the same format, but as pulses of red light rather than alternating high and low electrical voltages.

AES3 sockets on XLR or BNC are usually found only on high-end professional equipment, but the majority of audio interfaces aimed at home and project-studio owners feature a single stereo S/PDIF input and output on phonos and/or optical sockets.

Though the physical connection may vary, the sequence in which the samples are sent is always the same, and AES3 data always includes an ‘embedded’ clock signal. What this means in practice is that if you have a stereo mic preamp with an AES3 digital output, and you connect this to the corresponding input on your audio interface, the interface will receive not only the data but the timing reference needed to make sense of it. However, since the entire interface needs to operate from a single clock reference, you’ll need to switch it to use this clock source. This is usually done in the interface’s control panel software. Both devices must be operating at the same sample rate.

ADAT

AES3 and S/PDIF are ubiquitous and useful, but in most cases they offer only limited potential for expansion. Each input and output can only carry a stereo signal, and once we start adding more of them, we risk creating complicated clocking problems. Where greater numbers of additional channels are required, therefore, we often find another standard called ADAT or Lightpipe, originally developed by Alesis for their innovative eight-track digital recorders in the 1990s. ADAT carries digital information in a completely different format from AES3 — but, confusingly, it uses the same optical connectors and cables! In fact, many audio interfaces offer just a single optical input and output, which can be switched to accept either format.

Like AES3, an ADAT signal contains its own embedded clock. Unlike AES3, however, it can contain up to eight channels of 24-bit audio at standard sample rates. Sending eight channels of 88.2 or 96kHz audio over ADAT requires two sets of ports and support for an extension to the ADAT protocol called S/MUX. A single ADAT optical input is therefore a very convenient and cost-effective way to add eight analogue inputs to a recording system — a number which is often sufficient to transform its capabilities. Most audio interfaces that cater to the home and project-studio markets thus offer one or two optical inputs and outputs.

Word Clock

In theory, then, an audio interface with two sets of ADAT optical I/O allows us to add up to sixteen analogue inputs and outputs by connecting other devices that use the same format. But let’s suppose we want to increase the number of analogue inputs to our system by adding two eight-channel mic preamps. A mic preamp has no need to receive audio coming from the computer, so there’s no reason for it to have an ADAT input: to do its job it need only have an output. However, if a device doesn’t have an ADAT input, it can’t use the ADAT signal from another device as its clock reference. In other words, a preamp that only has a digital output can’t be a clock slave: it can only be a clock master, and each system can only contain one clock master.

Similar considerations apply when we want to use ADAT and AES3 connections at the same time, and they become even more of a problem when we have multiple pieces of digital gear linked together. We quickly reach a point where it’s not possible to rely on the embedded clock reference in ADAT or AES3 signals, because there’s no way to slave everything to a single master.

For this reason, lots of digital audio equipment includes a pair of BNC sockets for ‘word clock’ in and out. This carries no audio at all, but provides a stable timing reference for everything that’s connected. In the example described above, for example, you could connect a BNC cable from the word clock output on an audio interface to the clock input on the first mic preamp, and use a splitter or ’T’ connector to run another cable to the clock input on the second mic preamp. This allows the audio interface to be clock master, and everything else to be slaved to it.

Where possible, the device that is clock master should be the one carrying out the most important analogue-to-digital conversion; this might be the audio interface, or it might be a preamp or mixer that is feeding the interface. However, once you connect more than a handful of devices, you might find that you need a separate device is needed to supply a clock signal simultaneously to all of them. These devices are known as ‘master clocks’ and are possibly the most boring — but, in many cases, essential — pieces of studio equipment in existence!

MADI

As digital audio began to take off in earnest in the 1980s, the Audio Engineering Society quickly saw the need for a multi-channel standard for connecting digital devices. The result was MADI, which has actually been around since 1991, but which was only encountered in large-scale broadcast, live-sound and installation projects until recently. Although a single MADI connection can only convey audio in one direction, it can carry up to 64 channels of 24-bit audio at 44.1 or 48 kHz, so it becomes possible to add up to 64 inputs and the same number of outputs to an interface simply by including two small connectors. Adding multiple MADI connections would start to challenge the capabilities of the host computer!

Like AES3, with which it has some technical features in common, MADI can be carried either optically or electrically. The optical connector is different from that used by S/PDIF and ADAT, but electrical MADI connections use the same BNC connectors and cables as word clock and some AES3 connections. Both formats have the advantage of supporting very long cable runs, meaning that a MADI-equipped audio interface can be connected to a mixer or rack of preamps at the other end of a hall or studio complex.

Audio Over IP

A single MADI cable can thus carry more channels of audio than most of us are ever likely to need. However, just like ADAT and AES3, it’s a point-to-point connection that can only convey audio from a single source to a single destination. In typical situations where there are a lot of audio channels in play, that restriction becomes too limiting. Consider, for instance, a large studio complex where there are multiple live rooms and control rooms, along with a public performance space. Ideally, we’d want to be able to record audio in any of the control rooms from any of the live rooms and the stage, regardless of what anyone else is also recording, or whether the front-of-house engineer is also mixing for the audience. To enable all of the myriad configurations using any point-to-point standard would be almost impossible; complex cabling and digital patchbays would need to be set up, and adding any new devices into the system would prove a real headache.

A much more versatile solution is to set up not a series of simple one-way data links, but a network: a global pathway that connects all the devices in the studio and allows each of them to pass data to any of the others as needed. The value of this approach has been recognised for many years, and there have been many attempts to create audio networking systems suitable for studios, broadcast and live-sound use. Some of these required their own types of cable and connector, while others made non-standard use of the RJ45 connectors and Cat5 cable that are used for local area networks in computing. Some of these earlier systems are still in widespread use, especially in installation audio and broadcast, but if any sort of standardisation happens in the future, it seems most likely that it will focus on one of the newer standards that can carry audio over a standard network. These include Dante, developed by Audinate and featured in Focusrite’s RedNet and Red products, and Audio Video Bridging, which is an open standard defined by the Institute of Electrical and Electronics Engineers.

(If no clear winner emerges from all these competing Ethernet audio standards, it’s not necessarily a disaster, as a standard called AES67 has been developed to allow ‘layer 3’ protocols to talk to one another. Dante is already compliant with AES67, but AVB is a ‘layer 2’ technology and cannot be AES67-compliant without a hardware ‘bridge’.)

Networked audio systems generally need to be configured in software, which means they are not quite as simple as plugging an ADAT preamp into an audio interface. But their great advantage is scalability. More and more devices can be added to the network simply by plugging in a single cable. There is no need for digital patchbays or master clocks, and we have almost complete freedom as to where each device is positioned within the studio complex. What’s more, unlike any of the other standards for making digital audio connections, Ethernet cables can carry power as well as data, allowing devices such as headphone amps and monitor controllers to be freely moved around regardless of the availability of mains power.

Further Reading: Can An External Clock Improve Sound Quality?

External master clocks are often necessary in systems where multiple digital devices need to work together. However, it’s often claimed that there is another reason for using them: namely, that running a digital device from a high-quality master clock rather than its own clock can make it sound better.

It’s true that changing the clock source can make a difference to the subjective sound of a piece of digital equipment. In technical terms, however, doing so can only improve the performance of that equipment if its internal clock is faulty or sub-standard. This is more or less unheard-of in modern audio equipment. So, if you prefer the way your audio interface or analogue-to-digital converter sounds when clocked externally, you may be hearing a real difference, but this is almost certainly down to higher levels of noise and distortion.

This article from Sound On Sound provides a wealth of advice on the topic.

Further Reading: MIDI

As well as sockets that pass analogue and digital audio, some interfaces also feature 5-pin DIN sockets for MIDI In and Out. These don’t convey audio information at all, but a form of control data called Musical Instrument Digital Interface or MIDI. The MIDI standard was developed in the early 1980s to provide a way for keyboards, synths, sequencers and samplers from different manufacturers to interact. The MIDI data format is still in widespread use today, and although new instruments more often send it to host computers over a USB cable, it can be useful to have some hardware MIDI ports to control older equipment.

Further Reading: Aggregate To Accumulate?

In theory, the simplest way to expand a computer-based recording system would be to add more than one audio interface to it. In practice, however, this often isn’t an option, especially if you want to combine interfaces from different manufacturers. The ASIO driver protocol used by most Windows recording software can only interact with a single device: you can’t even use different interfaces for input and output, let alone combine the inputs from two or more separate interfaces. Apple’s Core Audio standard is more flexible in this department, and allows the inputs and outputs on multiple interfaces to be combined into a single ‘aggregate’ device. However, the aggregate devices are often unreliable and slow, and even if they work correctly, will leave you having to interact with two or more sets of driver and control panel software, as well as your recording software, so this approach is definitely not recommended.

More practical is the option of combining multiple interfaces of the same type. In this arrangement, the driver software is designed to present all the connected interfaces to the recording software and operating system as a single unit. It’s only feasible where the manufacturer’s driver and control panel software has been specially written to enable it. This is easier to achieve with PCIe and Thunderbolt audio interfaces than it is with USB, so there are very few USB interfaces that support this sort of ‘daisy-chaining’.

This sort of expansion is most useful with modular systems such as Pro Tools HDX, where adding additional PCIe cards increases the total number of inputs and outputs, but still leaves you free to choose what form those inputs and outputs should take. Ethernet audio protocols such as Dante and AVB take this flexibility much further, allowing any combination of the devices attached to the network to be viewed as a single multi-channel interface by your recording software.

Words: Sam Pryor

This content is part of Focusrite’s System Science series. Read more here.

System Science #4: Digital Considerations was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.