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.

London Creative Hub Expands Into Education With Focusrite & Novation Hardware Packages

Tileyard is the largest professional music and entertainment community in Europe, providing a home to more than 80 state-of-the-art music studios, as well as 120 creative industry businesses, in the exciting King’s Cross area of central London. The creative hub accommodates many different facilities — including homes to artists, composers, songwriters, producers, labels, publishers and even Ampify, a division of Novation.

The newest commitment Tileyard has made is in education. Recently, students have been supplied with hardware from Focusrite and Novation at Tileyard Education, a new postgraduate and professional training centre based at Tileyard Studios. In addition, Tileyard Education has partnered with the Institute of Contemporary Music Performance (ICMP) to provide ICMP final-year students with access to the new education facility. On-demand music education platform MusicGurus delivers the facility’s online learning, and a suite of DAW short courses has also been devised for launch in 2018.

Tileyard Education has been developed by Managing Director Harry Leckstein, previously Head of Music Business at BIMM London where he launched an undergraduate degree. “I’d written a BA (Hons) in Music Business, and while I was there I wrote an outline for an MBA Music and Entertainment Industry,” said Leckstein. “I wanted to teach it where the industry was, rather than in a traditional higher education institution, and a friend of mine said I should check out Tileyard.”

What originally started as one course is now scheduled to be six MA programmes. Tileyard Education is launching MA Commercial Songwriting and Production together with MA Music Industry Entrepreneurship in 2017, and planning MA Commercial Producer, MA Digital Film and TV, MA Creative Digital Design, and MA Games Developer and Entrepreneur over the next few years with partners Nottingham Trent University and University of Wales Trinity Saint David.

Only 20 students are being taken per programme per year, so it is a very intense experience that allows the small cohort — taught and mentored by the community — to get really embedded. Other unique components include one-to-one industry mentoring and the ability for graduates to present their work to the Tileyard investment fund on graduation.

There are a number of spaces in the facility — a main recording studio and live room, a tech lab, four writing rooms and a classroom. The tech lab is equipped with Novation Launchkey Mini keyboard controllers for Ableton Live software, and Focusrite Scarlett Solo interfaces for recording guitar and vocals simultaneously, onto separate tracks. For the writing rooms, Tileyard Education chose Launchkey 49 controllers and Scarlett 2i2 interfaces for recording two vocals at once, or miking up a vocal and guitar cab separately.

The facility installed the Launchkeys and Scarletts for their high quality, reliability and easy of use. “I think from a student point of view it’s about comfort,” said Leckstein. “Some students, certainly at an MA level, have got enough to contend with. And if they’re familiar with certain types of machinery it makes that role easier for them. We wanted to ensure that they had spaces and facilities that they were familiar and comfortable with, so that they could focus on the main job at hand which was the songwriting, creating and production.”

Find out more about Tileyard Education here.

Tileyard Education Tools Up 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.

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.

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.

USB, PCIe, Thunderbolt or Ethernet?

The audio interface might be the most important piece of kit in your entire studio. But with so many on the market, how do you know which one is right for you? This series of Web articles explains some of the key aspects of interface design, giving you the information you need to make an informed choice. First of all, we look at the different ways interfaces can connect to computers. What’s the difference between USB 3, USB-C, Thunderbolt and other standards, and which one is best for you? Here’s our guide through the maze of protocols and connectors.

Out In The Open

Computers are designed to be expandable. Manufacturers can’t predict how large a screen you might want, or what sort of keyboard you like, or whether you’ll want to add a graphics tablet, a scanner, or a high-performance audio interface. So, where possible, they leave these choices up to the user.

If all of these different peripherals had their own different ways of connecting, no computer could work with all of them. The solution is to come up with universal connection systems. Rather than develop a socket specially for printers, we create a standard socket that anything can connect to, as long as it speaks the right language. We can use it to attach a printer, but we can also connect a graphics tablet, a soundcard, a games controller, an external hard drive or a novelty photo frame.

In an ideal world, there would perhaps be just one universal system for connecting all hardware. We don’t live in an ideal world, so computers sport a wide range of different connectors. In more or less common use today, you’ll find USB 2.0, 3.0 and 3.1; Thunderbolt 1, 2 and 3; Gigabit Ethernet; PCIe… So what’s the difference, and which ones are best for audio recording?

Bits And Bandwidth

At the most basic level, what all these connections do is to make an electrical link between the computer and the other device. This link is then used to transfer digital data: high and low voltages that represent binary digits or ‘bits’. A microchip called a controller at either end of the link takes care of sending and receiving these ones and zeroes, but as far as the controller is concerned, all that matters is that a string of ones and zeroes has successfully been transmitted. It doesn’t know or care whether those bits add up to a photo of your kids, a chapter of your latest novel, or the best vocal take of your life.

The number of bits that can be sent over a connection every second is known as its bandwidth, and historically, not all types of connection had a bandwidth that was adequate for audio recording. The original USB 1.1 specification, for instance, offered just enough bandwidth to send stereo 24-bit, 44.1kHz audio in both directions at once. That was all right if you just wanted to record the odd vocal or guitar part, but not much use if you were tracking an entire band live on the studio floor.

Thankfully, those days are behind us, and all of the connectors you’re likely to find on a modern computer offer enough bandwidth for big multitrack recording projects. However, this bandwidth can still be eroded if that connection is shared by multiple devices.

In Real Time

With some types of computer peripheral, it’s more important to transfer data reliably than to transfer it continuously. When we’re backing up to a hard drive, it’s not disastrous if things occasionally get held up: what really matters is that the data gets there intact in the end. For sound recording, though, the mere ability to transfer large amounts of data isn’t enough. We need to maintain a continuous stream of data, and if that stream is interrupted, we can’t afford to wait around while the missing bits are tracked down.

Most connection types offer different transfer modes to reflect these different priorities. Data sent to a USB hard drive, for instance, would be transmitted as a ‘bulk transfer’. This offers no guarantees as to how long the process will take, but incorporates robust error correction that ensures any lost or corrupted data gets re-sent. A USB audio interface, by contrast, would transmit its data as an ‘isochronous transfer’. In this mode, the data is transferred in real time, but if any of it gets lost or damaged, the transfer simply ploughs on regardless.

Even in an isochronous transfer, however, the devices at either end of the link have to package up the data, transmit it and then unpack it again. This takes time, and in a typical audio recording scenario, it happens twice: once to data on its way into the computer, and again on the way out. If this process takes more than a few milliseconds, it can play havoc with a recording session. Performers relying on hearing themselves through headphones can be completely thrown if their voice or instrument comes back to them audibly delayed.

This delay is called latency, and keeping it to an acceptably low level is one of the biggest challenges in designing an audio interface. Some connection standards make this challenge easier to meet than others, but good low-latency performance is ultimately down to the design of the interface and its associated software drivers. A well-designed interface that uses one standard will out-perform a poorly designed interface that uses another, whatever the theoretical advantages of the latter.

Making Connections

In the case of PCIe, the electrical connection between computer and expansion devices is achieved by plugging the expansion card directly into a slot on the computer’s motherboard. Other types of connection use cables to join the computer and the expansion device, and in general, they all require their own individual connectors and cables. However, it’s not as simple as every way of connecting having one single connector type. For example, the original FireWire 400 specification supported two different types of connector: a larger six-pin one, and a smaller four-pin one for laptops and mobile devices. When the specification was upgraded as FireWire 800, a third connector type appeared.

USB 2 and 3.0 are even more complicated. Most of the other standards that are current today are ‘peer to peer’, meaning that they treat all the connected devices as equals. USB uses a different model where the computer is a ‘master’ and all other devices are ‘slaves’, and to emphasise that distinction, USB 2 and 3.0 cables have different plugs at either end. Not only that, but at the ‘slave’ end, there are numerous different ‘Type B’ connectors available to suit different devices.

Type C For Confusion

The potential for confusion grows ever greater when we consider Thunderbolt and USB 3.1. Instead of having a single standard with several different connector types, we now have single connectors that can support several different standards.

When Apple and Intel first cooked up Thunderbolt, the original idea was that it would use optical rather than electrical cables. By the time it was actually implemented on real computers, that plan had changed — but instead of introducing a new connector, the first and second generations of Thunderbolt used the existing ‘mini DisplayPort’ plug and socket. This made sense, because Thunderbolt was in some ways not really a new standard at all, but a container for two existing standards, being intended as a way of creating an external PCI Express connection while incorporating DisplayPort for connecting screens.

In the meantime, the consortium behind USB was working on a new standard called USB 3.1, which differs from its predecessors in several ways. One of the most obvious of these is that whereas a USB 2 or 3.0 cable had to have a Type A plug at the computer end and a Type B plug at the other end, USB 3.1 cables are all reversible, and have the same plug at either end. This is yet another new connector and is known, predictably enough, as a Type C connector. (There are currently few, if any, audio interfaces with Type C sockets.)

Intel’s response was to extend the remit of Thunderbolt even further, so that the new Thunderbolt 3 standard actually incorporates USB 3.1 and HDMI alongside DisplayPort and PCI Express. As part of this move, Intel switched from the mini DisplayPort connector to the USB Type C connector. This did away with one of the major disadvantages of Thunderbolt 1 and 2, which is that the cables themselves have to incorporate dedicated microchips to boost and detect the signal, making them much more expensive than typical USB, FireWire or Ethernet cables.

USB 3.1 and Thunderbolt 3 connections thus use the same cable and plug types, but although you can connect a USB 3.1 device to a Thunderbolt 3 socket on your computer and expect it to work, the reverse is not true. So it’s vital to realise that merely because your computer has a Type C socket does not mean it will necessarily be able to connect a Thunderbolt audio interface. Thunderbolt-compatible sockets should bear the Thunderbolt logo; if yours doesn’t, assume it’s ‘just’ a USB 3.1 socket. At the time of writing, all the Thunderbolt interfaces on the market still use the original mini DisplayPort connectors, so will need an adaptor if they are to be connected to a Thunderbolt 3 Type C port.

PCI Express

As we mentioned earlier, PCIe devices actually live inside the computer, on cards inserted into slots in its motherboard. Compared with some external connection protocols such as USB or FireWire, PCIe cards integrate on a more basic level into the architecture of the computer. In fact, most other connection types can ‘piggyback’ on PCIe. If, for instance, we wanted to increase the number of USB ports on our computer, we could install a PCIe card to add those connections.

The privileged status of the PCI Express card means that from a purely technical point of view, it can support excellent performance for audio interfaces, offering huge channel counts and the potential for very low latency. Yet dedicated PCIe audio interfaces are relatively thin on the ground, and tend to be confined to the higher end of the market.

One reason for this is simply that many computers don’t have the slots to put them in. Desktop and tower PCs with PCIe slots make up less than half of total computer sales, and PCIe is not available on any laptop or tablet computer, nor on any current Apple machine.

Another reason is that most of us don’t actually want our audio interface to be inside our computer. Musicians need easily accessible sockets for connecting microphones, loudspeakers, headphones and so on. The backplane of a PCIe card isn’t easy to get at, and doesn’t have enough space for them all. Most PCIe audio interfaces thus don’t place audio connectors directly on the card; instead, they come with external boxes that handle things like analogue-to-digital conversion, metering, and connecting audio gear. This is the basis of several high-end professional solutions, including the popular Pro Tools HDX system from Avid, but it’s relatively complex and expensive to implement.

A third reason is that many of the advantages of PCIe can now be had in another way. As we’ve already seen, the idea behind the Thunderbolt protocol is to ‘externalise’ PCIe, allowing devices outside the computer to hook into its internal architecture in the same privileged way. It is not yet clear whether Thunderbolt audio interfaces can quite match the low-latency performance of the very best PCIe cards, but any remaining advantage to the latter would seem to be outweighed by the greater convenience of the Thunderbolt format.

USB

Despite having been designed precisely with applications such as audio recording in mind, the Universal Serial Bus protocol has had a sketchy history in this department. From USB 2.0 onwards, the standard has offered enough theoretical bandwidth for serious multitrack recording, but the real-world low-latency performance of USB interfaces has been variable. The best of them, including Focusrite’s second-generation Scarlett range, come pretty close to matching what’s possible with Thunderbolt or PCIe. Others struggle to deliver latency figures low enough for real-time monitoring, or do so only with very high CPU loads. This is a crucial point to research when you’re choosing a USB interface.

Although there are now some audio interfaces that are marketed as ‘USB 3’ devices, this is sometimes just a branding exercise designed to stop them appearing behind the times. Most of these are 100 percent compatible with USB 2 and don’t necessarily use either the additional bandwidth or the new ‘Super Speed’ transmission type specified in USB 3. It remains to be seen whether genuine USB 3 interfaces will improve on the low-latency performance of USB 2 devices, but the early signs are that they won’t.

Even USB 2, however, has plenty to offer when manufacturers get their driver software right. For one thing, it is genuinely universal: you’d have to look pretty hard to find a desktop, tower or laptop computer that doesn’t have USB ports. It’s also backwards-compatible — USB 2 devices should work as advertised on USB 3 ports, and on USB-C ports with an adaptor — and apparently future-proof, especially now that Thunderbolt has become a container for USB 3.1. If you buy a USB interface now, it will be a long time before changing fashions in computer design leave you unable to connect it to anything. What’s more, many USB interfaces are now ‘class compliant’. This means they will work with the generic USB audio drivers built into Mac OS and iOS, so you can use these interfaces not only on Macs but also on iPads.

Perhaps the most compelling argument for USB is cost. Because it’s such a universal standard, USB controller chips, sockets and other components are very cheap, so in general, an audio interface that connects over USB will typically cost less than a Thunderbolt interface with otherwise identical features.

At the same time, however, USB does have some inherent disadvantages. One is that, compared with other connection protocols, it is difficult to implement ‘cascading’ or ‘daisy-chaining’ of peripherals. This means that you’ll need either a USB hub or additional USB ports on your computer if you want to run many USB devices at once. It also means that, with a few exceptions, USB audio interfaces generally can’t be run in multiples to add extra inputs and outputs.

Thunderbolt

Although multitrack audio recording on a computer can be challenging, the amount of data involved is relatively modest compared with, say, the capture of high-definition video footage. It was these very intensive tasks that prompted the development of an externalised version of PCIe, and the latest version of this new standard, Thunderbolt 3, can shift a mind-boggling 40 Gigabits per second.

In principle, Thunderbolt combines the advantages of PCIe — very high data bandwidth and privileged low-level access to the internal architecture of the computer — with the benefits of a well thought-out cabled protocol. Like USB interfaces, Thunderbolt devices can be ‘hot plugged’, meaning that you don’t need to shut the computer down before attaching or disconnecting them. Like USB, the Thunderbolt cable carries power as well as data. And unlike USB or PCIe, Thunderbolt permits multiple devices to be connected in a ‘daisy chain’.

In short, Thunderbolt has the capacity to allow us to record as many tracks as we’re ever likely to need, with latency figures that are potentially as low as are achievable with PCIe cards. And, in stark contrast to early USB or FireWire interfaces, Thunderbolt interfaces have delivered on this potential right from the start. So why aren’t we all using them?

One reason is cost. USB is not only a relatively mature technology, but also one with a huge mass-market take-up. This means that USB components are available very cheaply, and that there are off-the-shelf drivers and other software that can be licensed. Making the switch to a new technology such as Thunderbolt commits manufacturers to significant investment in development, and to paying a premium for any hardware or software they license. With Thunderbolt 1 and 2, there was also that pesky active cable to consider; understandably, consumers baulked at paying £50 or so for what looked like a short piece of plastic-coated wire.

Another is availability. Apple’s enthusiastic commitment to Thunderbolt has not been mirrored by PC motherboard manufacturers; and whereas it’s possible to retrofit USB or Ethernet ports to a computer by installing the appropriate PCIe card, this is not true of Thunderbolt, which can only be added if the motherboard has built-in support for it. For quite a while, it looked as though Thunderbolt would remain limited to Macs and a few specialist Windows machines. Although it is too soon to be certain, however, the signs are that Thunderbolt 3 is at last making serious inroads into the PC market.

Gigabit Ethernet

Computers don’t only need to talk to peripherals such as printers and audio interfaces. They also need to talk to other computers, and whereas other standards such as USB and Thunderbolt are primarily designed for adding additional hardware to a single computer, Ethernet has been developed mainly for communication between computers. So, although it offers plenty of bandwidth for audio recording, it wasn’t originally designed for that purpose, and there are many technical challenges that need to be overcome in making ‘audio over IP’ work properly. As a result, although Ethernet is the oldest connection protocol in common use today, and one of the most ubiquitous, it’s only in recent years that it has become a viable option for studio recording.

The great strength of Ethernet is also, as far as recording is concerned, its Achilles heel. An Ethernet network can allow tens or even hundreds of computers to communicate, but it can’t anticipate when each one of them will need to send data. The upshot is that it’s hard to guarantee an arrival time for any given packet of data, because the network might turn out to be busy when we want to send it. In other words, it’s difficult to establish the kind of real-time data transfer we need for audio streaming.

Audio equipment manufacturers have come up with numerous ways around this problem. Many of them use the Ethernet medium in non-standard ways, and require a dedicated network that is not shared with non-audio traffic. By contrast, Dante and AVB exploit developments within the Ethernet standard itself to send audio as normal network traffic. It is likely that the next few years will see one of these two become the industry standard for Ethernet audio, though it’s not yet clear which one will win out: Dante has a clear commercial lead, but AVB has the potential plus of being an open and non-proprietary system.

One of the big advantages Ethernet has over USB or Thunderbolt is that it can connect over much longer distances. USB Type C cables are restricted to 2 metres for USB 3.1 operation, and Thunderbolt 3 can only achieve its full potential over cables of less than half a metre in length. Ethernet cables, by contrast, can be up to 100 metres long. Ethernet can also supply power: Focusrite’s RedNet AM2 Dante headphone amp is an example of an audio device that can be powered in this way.

The biggest unique advantage of Ethernet audio, however, is its scalability. In practical terms, the other standards described above are only useful for simple setups where a single audio interface is exchanging data with a single computer. In an Ethernet network, however, there is almost no limit on how many devices can be connected. Multiple computers on a network can access the data from a single audio interface, or a single computer can simultaneously record audio from multiple interfaces in different live rooms. If we need more headphone outputs in the middle of a session, we can simply plug additional Dante or AVB-enabled headphone amps into the network.

Hybrid Systems

Although the network sockets built into typical computers are usable for audio over Ethernet, it’s usually necessary to install a purpose-designed Ethernet card for best results. A Dante system using Focusrite’s RedNet PCIe card, for example, can achieve very good low-latency performance. However, fast Ethernet connectivity can also be added in other ways, and this fact has been exploited to create interfaces that combine both USB or Thunderbolt audio and audio over Ethernet. In effect, a single unit acts both as a Thunderbolt or USB audio interface, and as an Ethernet switch or hub for connecting Dante or AVB audio devices. This concept is the basis of Focusrite’s Red 4Pre, Red 8Pre and Red 16 Line, and is one of the key developments that is bringing Ethernet audio to a wider market.

Most of the many audio-over-Ethernet protocols in existence were designed with the needs of major broadcasters, live-sound venues and installation clients in mind, because the key benefits of scalability and long cable runs are most relevant in those scenarios. As a result, Ethernet audio setups have tended to be expensive, and they have a reputation for being complex to set up. However, there is now an increasingly wide range of Dante and AVB products suitable for studio use, and hybrid systems such as the Red 4Pre offer an affordable yet powerful point of entry into this world.

Bus Power

One of the main reasons why Intel eventually chose to use electrical rather than optical cables for Thunderbolt was that they can deliver power as well as data. This is an important factor for many add-on devices, including audio interfaces, because it can potentially eliminate the need for those devices to have mains power supplies of their own. So, for instance, USB 2 devices are permitted to draw up to 500mA at 5V from the host computer, while USB 3 raises this to 900mA, and Thunderbolt theoretically supports up to 550mA at 18V. However, it’s important to note that the specifications usually refer to the maximum power that an attached device is allowed to demand: they don’t guarantee that the computer will be able to meet that demand.

It’s therefore possible to design bus-powered USB, Thunderbolt and Ethernet audio interfaces, but manufacturers can’t rely on there being as much power available as they’d like. This restriction can compromise the audio performance of an interface, and for this reason, bus powering is typically used only for small desktop interfaces, where the convenience outweighs potential disadvantages.

Serial Vs Parallel

Digital data is made up of binary digits or bits. These can be represented any way you like — holes in a punch card, pulses of light in an optical cable — but in most computer connections they are encoded as voltages. For instance, USB2 operating in High Speed mode represents zeroes by a voltage of 0V and ones by 400mV.

Since any given piece of wire can only carry one voltage at a time, the only way to send many bits down one piece of wire is to do so one after the other. If we want to send a larger amount of data, it needs to be broken down into individual bits, which are then sent in order and reconstructed by the receiving device. A connection of this type is known as a ‘serial’ connection.

However, the internal architecture of a computer is based around groups of bits. Eight bits make up a byte, and groups of bytes are sometimes called ‘words’. The same goes for many forms of data, including audio. Each sample in a stream of digital audio is normally represented using words of either 16 or 24 bits. So, in principle, if we connected 16 or 24 wires instead of just one, we could sent an entire sample in one go, instead of having to break it down into its constituent bits and send these one after the other. Given the same clock speed, in other words, we can double the rate at which data is transmitted by doubling the number of wires.

This type of connection is called a ‘parallel’ connection, and on the face of it, you might think that parallel connections are obviously better suited than serial connections for transferring lots of data in a short space of time. In fact, however, all of the connection protocols common on modern computers are serial standards. To achieve a higher bandwidth, it is easier, cheaper and more reliable to simply increase the clock speed of a serial connection — thus sending more bits per millisecond — than to accurately synchronise the sending of data across a parallel link.

Most computer peripherals, and certainly audio interfaces, require a bi-directional connection to the computer. In other words, data needs to travel in both directions between the interface and the computer. But, of course, if we have only a single piece of wire to make that connection, then data can only travel down it in one direction at a time. To achieve bi-directional transfer over a single connection, we need to continually switch the direction of travel, so that ‘packets’ of data travelling in one direction alternate with packets going the other way. This is how USB 1.1 and USB 2 work, and it’s called a ‘half duplex’ connection. By contrast, FireWire, USB 3, Thunderbolt and Gigabit Ethernet are all ‘full duplex’ connections, meaning that they provide separate paths for data travelling in each direction.

Words: Sam Pryor

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

System Science #1: Connection Methods was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.

How To Make Sense Of Audio Interface Specifications

Your audio interface is potentially the most important piece of hardware in your studio, so it’s vital to choose one that is right for you. That means choosing an interface that will connect to your computer and work well with it. It means opting for a selection of analogue inputs and outputs that meets your current needs. It also means anticipating your future needs, and making sure you have the right digital inputs and outputs on hand to expand your system when the time comes.

However, once you’ve looked around at what’s available, you’ll probably find that multiple interfaces from different manufacturers offer very similar features. For example, there are many USB 2 audio interface that feature eight analogue inputs, a similar number of outputs, and one or two ADAT optical ports for expansion. What makes one of these better than another?

One factor that comes up again and again in Internet reviews and discussions is sound quality. You’ll find endless forum posts saying that this or that interface offers superior audio quality, or advising you to choose one interface over another because its mic preamps sound better. Unfortunately, what’s all too rare in these discussions is consistency and common sense, while misinformation and received wisdom is rife. This is not helped by some manufacturers who publish misleading or partial specifications to present their products in the best possible light.

What We Can Measure vs What We Can Hear

There are differences in the sound quality of audio interfaces; and for the most part, these are down to factors that can be measured. The more you pay for your audio interface, the better it’s likely to perform on the test bench. However, it’s important not to overstate the extent to which these differences are likely to be audible. In other words, there might be objective ways in which one interface outperforms another, but these may well be difficult or impossible to hear in normal use.

For example, you can expect any audio interface on the market now to offer a frequency response that is flat to within half a decibel or better across the entire range of human hearing, on both inputs and outputs. What this means is that any frequency imbalances introduced by today’s audio interfaces will be as nothing compared with the failings of microphones, loudspeakers and acoustic treatment. In fact, in any normal listening environment, simply moving our head a few inches will introduce a much bigger changes in the frequency balance we hear, as the way in which direct sound from the loudspeakers combines with reflected sound changes from point to point.

Setting up a fair comparison of subjective sound quality is notoriously difficult, and requires extremely rigorous preparation, level matching to 0.1dB or better, and a very good listening environment. It’s the sort of thing that audio interface manufacturers might do when fine-tuning the design of their latest and greatest product, but it’s very unlikely that comparing two interfaces in a shop or dealer showroom will reveal real differences between them. For this reason, we often need to resort to published specifications in order to try to compare the audio performance of two different interfaces. But which specifications are important, and how should we interpret them?

Design And Execution

Roughly speaking, we can divide specifications up into two categories: specifications that describe what something is designed to do, and measurements of how well it performs. In the first category, we could include factors such as the gain range of a mic preamp or the maximum output level of a line output. In the latter category, we find measurements of frequency response, noise and distortion.

The first kind of specification can help us decide whether an interface actually meets our needs or not. Let’s have a look at some of the most important of them.

Analogue inputs, and especially microphone preamps, have to be able to cope with a wide range of different types of input. A sensitive microphone placed next to a snare drum will put out a very ‘hot’ signal, while a ribbon mic on an acoustic guitar will require far more preamplification before the signal can be recorded at an appropriate level. For this reason, any input designed for connecting a microphone will have a gain control that governs the amount of preamplification being applied. The wider the range of this gain control, the more versatile the mic preamp.

Some high-end stand-alone mic preamps, such as Focusrite’s ISA units, offer a total gain range of 80dB or so. This allows them to cope with the full range of microphone and source combinations you’re likely to encounter in any real-world recording situation. For the majority of recording applications, however, such a wide gain range isn’t really necessary, and most audio interfaces offer somewhere between 55 and 65 dB, which is sufficient for nearly all needs. This is sometimes extended by means of a switchable ‘pad’, which typically attenuates the signal by a fixed 15 or 20 dB, allowing very high-output sources to be connected without risk of overload.

Note that two interfaces with the same gain range don’t necessarily behave identically: the preamps in interface A might span -5 to +55 dB while those in interface B might run from +5 to +65 dB, in which case both would have a 60dB gain range, but interface A would be better able to cope with very loud close-miked signals, while interface B would be a better choice for recording very quiet sources.

Making Gains

One factor that can make a big difference in practical terms is the way in which preamp gain control is implemented. On more affordable audio equipment, gain is usually adjusted using a simple potentiometer. This is quick and effective, and cheap to implement, but it tends to be imprecise and makes settings hard to recall exactly. Increasingly, therefore, interfaces targeting professional users, such as Focusrite’s Red 4Pre and 8Pre, use digitally controlled preamplifiers that allow gain to be set precisely in steps of 3dB or less. This is invaluable when you need to match gain across two or more channels, as is the case for stereo miking techniques.

Another specification you’ll often find listed for inputs is impedance. In general, microphone inputs typically present an input impedance somewhere between 1 and 7 kΩ, while line inputs are in the order of 10kΩ. If your interface has instrument inputs designed for directly plugging in an electric guitar or bass, these should present a much higher impedance, ideally at least 1MΩ. An interesting feature that’s available on some microphone preamps is the ability to vary the input impedance. This is part of what the Air circuit on Focusrite’s Clarett and Red products does, and it can make a difference to the sound of some microphones, though others are not sensitive to input impedance.

On many interfaces, especially those targeted at home and project studios, the line inputs are routed through the preamp stage, meaning that their gain is variable in the same way as that of the mic inputs. This gives them more flexibility to cope with lower levels put out by non-standard devices such as vintage synthesizers, but does not offer such a pure signal path. More costly interfaces usually give the line inputs their own direct path to the analogue-to-digital converter, for maximum fidelity. It’s often still possible to change the sensitivity of these inputs up to a point, though, either by means of a variable ‘trim’ control or by switching between the +4dBu ‘professional’ and -10dBV ‘consumer’ reference levels.

As well as a gain range, a maximum input level in dBu is also usually specified for mic and line inputs. Similarly, a maximum output level will be quoted for the line outputs. This represents the voltage that will be delivered by a full-scale digital signal, and is relevant if you want to feed audio from the line outputs of your interface into other studio gear such as outboard processors, mixers or monitor controllers. For example, if the maximum output level of your interface’s line outs is much lower than the input level a compressor is designed to accept, you might struggle to drive the compressor hard enough to achieve heavy compression.

Peak Performance

So much for specifications that tell us how an interface is designed to behave. When we come to the ones that measure how well it performs, it’s often necessary to inspect published figures with a sceptical eye. As someone might have said, there are lies, damned lies, and specifications!

First of all, it’s vital to know exactly which aspect of the interface a given figure refers to, and this isn’t always made clear in published specifications. For instance, some unscrupulous manufacturers have been known to publish figures that only refer to the analogue-to-digital converter chips they’re using, rather than the entire input stage of the interface. These figures might look impressive, but they are largely meaningless: in the real world, it’s the performance of the input stage as a whole that matters.

For line-level inputs and outputs, there are three key measurements. One is frequency response, which measures our interface’s ability to reproduce sound accurately without introducing unwanted changes to its timbre. Almost all interfaces on the market today perform very well in this respect, and you should certainly expect a quoted frequency response specification to be flat to within ±1dB between 20Hz and 20kHz, the conventionally accepted limits of human perception. Focusrite’s Clarett interfaces are an order of magnitude better, measuring flat within ±0.1dB in this range. Some manufacturers quote a much higher upper limit to the frequency response, such as 40kHz: this sounds impressive, but has little bearing on what we can hear, and is only relevant when recording at elevated sample rates.

Range Rovers

Almost all audio interfaces available today have a published frequency response that is pretty much perfect, so this doesn’t provide any grounds for comparing them. More telling are the figures for dynamic range: the ratio between, at one end of the spectrum, the loudest signal the interface can accommodate and, at the other, the quietest signal that is detectable above the noise floor.

The signals we record have a dynamic range of their own. For instance, a classical symphony might peak with 100 musicians playing triple-forte, and end with a solo violin playing softly. Its dynamic range would be the ratio between the loudest point of the loudest section, and the point where the reverberation of that solo violin fades out of hearing below the ambient noise in the hall.

We can think of the dynamic range of our recording system’s inputs and outputs as a ‘window’ through which the music we’re recording needs to fit. The wider the system’s dynamic range, the taller the window, and the more leeway we have in setting microphone gain levels so as to fit everything in.

Dynamic range is measured in decibels, and an ‘A-weighted’ figure is usually quoted. What this means is that greater emphasis (weight) is placed on that part of the frequency spectrum to which the human ear is most sensitive — roughly speaking, from 1–10 kHz. Unweighted measurements for the same piece of equipment will typically be several dB worse than the A-weighted figures.

In practice, the best-performing premium audio interfaces on the market today, such as Focusrite’s Red 4Pre, Red 8Pre and Red 16 Line achieve an A-weighted dynamic range of 120dB or better on line outputs and approaching 120dB on their line inputs. The Red interfaces’ class-leading output performance is made possible by the use of ‘parallel path’ conversion, a no-compromise design approach which splits the signal between two digital-to-analogue converters and adds together their outputs — because the wanted audio from both is correlated while the noise is not, this achieves a 3dB increase in the ratio of wanted signal to noise. More affordable interfaces that target the home and project-studio owner fall slightly short of this, but still achieve performance that would have been unheard-of ten years ago: Focusrite’s Clarett interfaces, for example, feature measured dynamic ranges of 116dB on the line inputs and 119dB on the line outputs.

It should be pointed out, first, that these figures are only relevant to 24-bit recording, as a 16-bit recording has a theoretical maximum dynamic range of 96dB, and second, that you are very unlikely to ever find yourself having to record a real-world source with a dynamic range anything like this large. Having more dynamic range available on our recording system merely gives us a little more ‘wriggle room’ in recording typical real-world signals. Nevertheless, a good dynamic range specification across all of an interface’s inputs and outputs is indicative of attention to detail and good design by the manufacturer.

Noise

The third measure of performance that is usually quoted for analogue inputs and outputs evaluates how successfully they avoid adding unwanted additional contributions to the signals passing through them. These contributions fall into two categories: distortion and noise.

In the case of line inputs and outputs, these are usually lumped together in a single measurement called Total Harmonic Distortion plus Noise, or THD+N for short. The idea behind this measurement is fairly simple: we feed in a pure 1kHz sine wave at a known level, and we inspect what comes out the other end for traces of anything that’s not a pure 1kHz sine wave. These will typically be a mixture of distortion products at related frequencies such as 3kHz or 5kHz, and noise, which is anything added that is unrelated to the input signal.

THD+N is sometimes expressed in decibels and sometimes as a percentage, but always measures the ratio of unwanted to wanted material in the signal. Once again, Focusrite’s Red 8Pre represents the state of the art, with THD+N measuring as little as -104dB (0.0006%) on line inputs 9–16 and -102dB (0.0008%) on its line outputs.

For mic preamps, a separate noise measurement is usually given, which is called Equivalent Input Noise or EIN. An ideal mic preamp would perfectly amplify the signal that’s fed to it, but any real-world circuit will inevitably add some noise in doing so. To measure EIN, we connect something that behaves like a microphone in terms of impedance, turn the preamp gain up to maximum, and measure the level of the noise. We then subtract the amount of gain from the measured noise level to return an EIN figure, which will be a negative number in decibels — the more negative the better!

Of all these specifications, equivalent input noise is perhaps the one that’s likely to prove most telling in real-world situations: an interface with low EIN figures and a high gain range will do a better job of recording quiet sources than a less well-performing one. Unfortunately, it’s also the measurement that is most liable to be abused in published specifications. It’s only possible to make a fair comparison of EIN figures between different products if identical test conditions are quoted. By convention, for example, EIN used to be measured without any weighting, but many manufacturers now quote A-weighted figures, which are typically 3 to 5 dB better than the unweighted measurements. It’s also conventional to use a 150Ω source impedance, which is representative of typical microphones, but some manufacturers massage their EIN figures by using a lower source impedance. This can make the figures appear several dB better.

Assuming an unweighted response and a 150Ω source impedance, it’s impossible for EIN to measure lower than -131dBu. In practice, -124dBu is a good real-world figure under these test conditions, while -128dBu would be a good A-weighted measurement. Too often, however, the test conditions are not made public. In this case it is simply not possible to compare EIN specifications and they should be ignored.

Specs Appeal

As far as specifications go, even the least good audio interfaces on the market today perform orders of magnitude better than the recording systems of forty years ago. To actually hear differences in most of the measurements described above, you’d need trained hearing, a superb monitoring environment and a very carefully calibrated test procedure. There are few, if any, real-world recording situations where a difference of 3dB on input dynamic range, or ±0.5dB on frequency response, will be audible.

But does this mean that specifications don’t matter? Not at all. Published specifications are a very good measure both of a manufacturer’s commitment to quality, and of their openness. Where a manufacturer lists full specifications, with all the relevant measurement conditions clearly stated, you can be confident that they have nothing to hide, and that they are proud of the performance of their product. On the other side of the coin, where manufacturers miss out vital information, or arbitrarily change measurement conditions to massage the results, what else might they be hiding?

Further Reading: Understanding Decibels

Many of the measurements you’ll find listed in manufacturer specifications and elsewhere are given in decibels, abbreviated to dB. It’s crucial to understand that a figure expressed only in dB doesn’t represent an absolute level, but the ratio between two levels. So, for example, if the dynamic range of a line output is given as 120dB, this tells us nothing about the peak signal level that can be obtained from this output. What it tells us is that the peak signal level is a certain (large!) number of times higher than level of the the noise floor.

To obtain absolute level measurements, we compare readings with fixed reference levels. A positive measurement indicates the signal is larger than the reference, while a negative measurement indicates that it is smaller. Different references are used in different contexts. For instance, we measure the loudness of sound in air by reference to the quietest sound that is audible by human hearing, and we give the resulting measurements the suffix dBSPL, for ‘sound pressure level’. Any audible sound would by definition register a positive value in dBSPL.

In digital audio, we take as the reference the largest value that the system can represent. This is known as 0dBFS for ‘full scale’. Since nothing within the system can exceed this value, levels in dBFS are always negative, because they always compare a smaller with a larger value.

In analogue circuits, where sound is represented as a changing voltage, the reference value we use is a sine wave at a fixed voltage. The most common standard used in professional audio is dBu, which takes a signal of just under 0.8V as the reference, but you may also find measurements expressed in dBV, which takes 1V as its reference. Note that in both cases the ‘root mean square’ value rather than the peak-to-peak measurement is usually used.

dB dBu dBFS dBV to volts audio conversion digital - calculator volt to dBu and dBV dB mW SPL dB decibels 0 dBFS - convert dB volt normal decibels relatioship relation explanation analog audio absolute level true rms convertor converter decibel to dbfs converter calculation online attenuation loss gain ratio reference audio engineering sound recording dBFS dBVU 0 dB audio logarithm level converter peak to peak p-p impedance voltage pro consumer audio digital analog recording level - sengpielaudio Sengpiel Berlin

Further Reading: Headphone Loudness

One of the most common complaints that is raised against some audio interfaces, particularly bus-powered ones, is that their built-in headphone outputs don’t go loud enough. Is there a way to tell from the manufacturer’s specifications how loudly an interface will be able to drive a pair of headphones? There should be, but many manufacturers either don’t provide the relevant information at all, or do so in ways that make it hard to compare.

As with equivalent input noise, if you are able to compare specifications for headphone outputs, it’s important to be certain that they were made under the same measurement conditions. The standard procedure, as followed by Focusrite, is to feed a full-scale digital signal to the headphone output, turn the volume all the way up and measure the power delivered into a 600Ω load. This will give a reading in dBm (which is a measurement of power, not voltage). So, for example, Focusrite’s figures show that the headphone amplifier in the Clarett 8PreX can deliver 16dBm into a 600Ω load, compared with the 10dBm of the bus-powered Scarlett Solo.

The sound pressure level generated by headphones from this input will then depend on the impedance of the headphones, and their efficiency; in general, low-impedance phones with a high efficiency rating will deliver the loudest sound for a given input. Note though that efficiency or sensitivity is sometimes quoted in dB/mW and sometimes in dB/V, which are not the same!

Be aware that it is not good practice to listen loudly on headphones for long periods, and doing so can cause serious hearing damage.

Words: Sam Pryor

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

System Science #5: Sound Quality & Specs was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.

Stories From The Touring Women Who Keep The Music Sounding Sweet

“‘Listen, I’m not going to treat you like a girl.’” Dana Wachs is recounting her interaction with a male staff member at a show she was working as front of house tech. Wachs is a self-professed gear nerd and a self-taught jack-of-all-trades who works as an audio engineer, tour manager, and monitor engineer. “But in effect, by saying that, what he did was to treat me like a ‘girl,’” said Wachs.

The gendering of roles within the audio and technological side of the music industry is one of the more complex parts of the problem found in dealing with misogyny and sexism in music. Is this discrimination founded on ignorance or are people really that condescending and threatened by women in the field?

“I’ve definitely had band members or audience members or other techs, who are like, ‘Do you find it hard being a female tech?’ As if it’s never crossed their minds that I could do this,” said Rachel Ryan, over the phone from her home in Los Angeles between tours. “It’s just that they’ve never seen a female tech, so they’re genuinely confused. That’s coming from a very different place than just ‘let’s talk about your experience as a female, as an engineer, as both’.”

Sharon Levinson — another FOH engineer who doubles as tour manager for acts like Amon Tobin, Swans, and Kid Koala — approaches it from a slightly different perspective. “When I was 20, when I first started, I looked like I was 15, so it was probably less about the fact that I was a girl and more about the fact that I looked 15 and probably didn’t know what I was doing,” said Levinson. “[Men not taking me seriously] doesn’t happen as much now as it used to. Or if it does, it’s usually only the first five minutes and then they realize I know what I’m doing and it’s not an issue.”

“When I was younger, I did experience a lot of condescension,” said Wachs. “But that may have been because of the fact that I was just young and nothing to do with being a woman. Other times, it seemed more obvious that I was being targeted because I was a woman.”

If there is any difference between the way men and women techs work with each other within the industry then Ryan, Levinson, and Wachs all seem to share a similar view: That it is more about individual styles and personalities than boys versus girls.

“It always depends on the individuals,” said Levinson. “I’ve been hired by bands that include girls specifically because I’m female…being the only girl on a tour can be isolating, but if there are two of us (or more) it kind of cancels out the gender issue.”

Tour life is gruelling. Twelve hour days, hundreds of miles of driving, organizing a crew of 40, weeks away from friends and family, solving problems on the fly.

“On the recent Deerhunter/Kings Of Leon tour, the fiber communication between the FOH console and the split somehow died just two minutes before their set,” Wachs recalls. “I ended up having to mix FOH from the monitor console, using the monitor engineer’s show file with no effects, low-pass engaged in every channel, on headphones, probably about 50 feet behind the mains. That definitely brought my heart rate up a bit!”

These problems can sometimes be magnified or diminished, depending on the context and the country. “In Japan, many of the crew were women.” said Wachs. “They worked hard. Everything was efficient and ran smoothly.”

Ryan has had similar experiences working abroad during her time out on the road. “I’ve definitely been to other countries where I’ve been the only female tech and they’ve never seen one before,” said Ryan, who has been behind the board since she was a teen. Working with bands like Silversun Pickups, School of Seven Bells, Phosphorescent, and Little Boots, she has an enormous amount of experience. “I happen to work with a lot of female bands who are specifically looking for female crew or have female band members,” she said. “I also work, within the industry, with a lot of men, who are engineers or tour managers or even in bands. This last tour, out of 45 people, four of us were female.”

“I hate to separate female engineers, but because there are issues, usually from males in the industry, we need to talk about it or else it will never change.”

— Dana Wachs, Touring Engineer

Ryan has a great fondness for some of the women she’s had the pleasure to work, with behind the board and on stage, over the years. “I love all the girls that I’ve met,” she said. “We run into each other on tour. It’s good to have that community of people — male or female, it’s good to have a support system. Touring is hard, whether you’re male or female. It’s important to have your people to keep you grounded.”

Ryan is part of a few different female-focused groups, including The Knife Set (a group of women all working in tour management, audio engineering, and tech). She is also a member of Soundgirls, an online resource center and community for audio engineers, of which Wachs is also a member.

“Soundgirls not only shares plenty of job opportunities, but is a resource for technological troubleshooting, as well as supporting women in a vastly male field,” said Wachs. “I hate to separate female engineers, but because there are issues, usually from males in the industry, we need to talk about it or else it will never change.”

The importance of these discussions and facts becomes clear for both the recognition of issues and the successive dismantling of the currently male-dominated workforce.

“There are not a lot of women in this industry.” said Ryan. “If we don’t talk about that, no-one knows about it and then it never changes. Young women don’t realize that they have these opportunities; representation matters.”

But what about particular aspects of gender that are, at times, impossible to escape? For instance, what happens when a touring FOH engineer or TM gets pregnant and then has to face additional discrimination because of biology?

Levinson, a new mother, worked right up until and immediately following the birth of her 14 month old son, Eli. She dealt with the challenges of pregnancy while out on the road with the band Operators, though they’re not the challenges one might expect.

“The funniest thing about being pregnant happened in London, Ontario, “ Levinson said. “I was about six or seven months pregnant. It was after the show and I was coiling cables when these two drunk guys came up and went, ‘Uh, can we help you?’ And I said, ‘Um, no, I’m good. Why?’ ‘Well, we just feel bad for you.’ It was pretty funny. Like, dude, I can definitely do this way better than you can. I got this.”

A big part of working toward positive change comes from calling out inappropriate behavior in the workplace. In this case, that’s in the club or on the bus, and making sure people are held accountable for their actions.

“It needs to be known that certain behavior is not acceptable or professional,” Wachs said. “The leaders and employers in any industry need to hold their employees to task, and I would hope they would not want their company or brand (or band) to be represented by antiquated and offensive ideals.”

Like with any job, the longer you work in your field, the more experience you have to draw from and form your own opinion. Your own network of trusted allies grows, and you develop your own way of dealing with misogyny.

“I find that as I get older, I have better ways of coping, both internally as well as how to shut it down when it’s happening,” said Ryan. “I hope that more or less I’ve found a way to shut it down politely. There’s definitely things you let go and things you call out. But I have less of a problem calling people out in such a way that stops them without then causing more problems. And that’s taken a very long time to figure out how to do that in a way that works best for me.”

While antiquated attitudes are frustrating and sexism within this industry is a serious issue, the main point all these women have made is simple: Be good at what you do.

“In many respects, I’ve been very lucky with the engineers and the crew that I have met over the years,” said Ryan. “It’s almost always been about whether or not you’re good at your job and less about gender.”

“It’s more about figuring out all the pieces that are involved in any given tour,” Levinson said. “Sometimes, it’s not even the audio stuff that’s tricky but the visuals. With the Amon [Tobin] tour, it was a logistical nightmare. It was seven huge, heavy road cases in a semi [transport truck]. [Amon] has the best ears of anybody out there in the music world, so the sound had to be perfect,” she continues. “But I also had to figure out if there were forklifts available or like, if the cube was going to fit down a hallway. I spent hours trying to get dimensions of venues just to make sure we could load in.”

“I’m a big proponent of therapy. Everyone should go to therapy!” Ryan said, jokingly. “Tour can be such a bubble, and it’s very easy to get sucked in and to lose yourself a little bit. I have found that having an outside source, whether it’s your friends, your parents, therapist helps you cope with the differences that touring life presents.”

And perhaps, most perfectly, Wachs sums up why gender simply isn’t important as an audio tech. “An engineer once asked me how I could mix while wearing heels. I pointed out to him that I was not wearing them on my ears.”

This content is an initiative of Focusrite/Novation, whose ‘Women In Music Technology’ series spotlights women working in behind-the-scenes role. The aim is to inspire the next generation of women in music technology to reduce the gender imbalance in the music industry.

Road Warriors was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.

Simon Hanhart Brings Red & RedNet To Focusrite’s Spiritual Home: AIR Studios

Producer, mixer and engineer Simon Hanhart recently implemented a Focusrite Red 8Pre and two RedNet MP8R audio interfaces on a session at AIR Studios, the famous London complex home to Focusrite’s original Rupert Neve-designed recording console modules. Hanhart’s 24-input interconnected system took advantage of the MP8R and 8Pre’s Dante audio-over-IP technology, plus Red 8Pre’s DigiLink connectivity while in the fabled Studio 1 with a 12-piece string section, vocalist and concert pianist.

Hanhart’s impressive credits include Asia, Marillion, McFly, David Bowie, SikTh, London Symphony Orchestra, Yngwie Malmsteen, Cerys Matthews and Iron Maiden’s Adrian Smith. On this occasion, he was working with vocalist, songwriter and producer John Mitchell, pianist Dominic Ferris and Richard Cottle, who arranged the piece and conducted the London String Group. The song, ‘Made Of Paper’ was written by Tim Harbour.

“In a situation like this, you’re working with natural instruments that need to sound like they would sound if you were standing eight feet away from them,” Hanhart said. “You’ve got to capture that, so you don’t want anything in the way of that signal path.”

During the day, Hanhart spoke highly of the Red 8Pre’s sound quality. “It just works. It sounds great. You’re not even aware of it because it’s just translating what’s happening in the room straight into the machine,” he said.

The Red 8Pre was racked in the machine room, connected to a Mac-based Pro Tools | HDX rig via DigiLink. Out in the live room were the RedNet MP8R interfaces, which were controlled by software in the control room. All the microphones for the 12-piece string section, piano and vocal went into the Focusrite mic pres, and then Hanhart monitored back through the vintage Neve console in the control room.

“It’s the next generation: one great setup that can be accessed from a number of different production rooms.”

— Simon Hanhart, Producer/Engineer

Having Dante in the recording facility was particularly useful to Hanhart. “It makes complete sense that you can network a number of different units together, particularly in a multi-studio facility like AIR,” he said. “You want to make sure that you have one great setup that can be accessed from a number of different production rooms or studio areas. It’s the next generation really.”

Hanhart owns a Red 8Pre, so everything was compatible when mixing the track at home. “I’m always aware that it would be nice to have a little mobile setup, and of course I could take the Red 8Pre with me, and that would be a Thunderbolt situation,” he said. “Having the options on connectivity is absolutely fantastic. It’s a unit that covers so many bases. It’s absolutely brilliant.”

Check out the Focusrite website for more information on the Red 8Pre and RedNet MP8R. Also, check out this video from the session.

Focusrite Interfaces At AIR Studios was originally published in Focusrite // Spectrum on Medium, where people are continuing the conversation by highlighting and responding to this story.