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The great debate: digital vs. analog. I'm sure most of us can sit here all day and discuss why one is better than the other. While there are great benefits to both, most of today's top engineers such as Greg Wells, Tom Lord-Alge, Tony Maserati, and Michael Brauer, are starting to use a combination, or hybrid system, for mixing.

As technology advances, many would say the line between the two worlds is becoming very blurred. This article will show some different examples on how to set up a system that can work best for your workflow. Since a hybrid system is any combination of digital and analog gear, there is no right or wrong way to set up your system. It really depends on how you want to improve your workflow.

Why use a Hybrid Mixing set-up?
Mixing is all about feeling and a great mix has a lot of interaction between the mixer and the song. In the past, we had a large desk with EQs, aux sends, faders, and sometimes compression. This allowed mixing engineers to put their hands on everything and really "feel" the mix. In some cases, there would be three to four engineers on a desk moving faders and turning EQs as the mix is printed.

In the modern studio, most people are working on a laptop with a mouse, doing a lot of automation and clip gain. While great mixes are being done with this method, some engineers feel the mix lacks "the human touch." It's hard to make precise fader moves with a mouse or trackpad, or to close your eyes and cycle through an EQ band to find the sweet spot.

A hybrid system allows you to get the precision of the digital world with the use of automation, clip gain, plug-ins and editing functions, while also adding the analog warmth and human touch aspect of mixing that we all know and love.

As we mentioned earlier, an issue with in the box mixing is the inability to physically touch a fader or EQ. A control surface is a digital controller for your Control Surface DAW, which maps all your audio channels to a digital channel strip. This is done through Ethernet, USB, Firewire or Thunderbolt. To use a control surface, you don't necessarily need the I/O of a large interface. The controller does not pass any audio through it, so it won't "color" your sound like an analog desk, but will allow you to physically push the fader, turn the EQ and pan pots, and write automation on more than one source at a time.

For a smaller home studio, the For a larger pro studio setup, you can look into the Avid Artist Mix would be a great choice for $999, which has eight touch-sensitive 100mm long throw motorized faders, eight touch-sensitive rotary encoders, and eight high-resolution OLED displays. You can cycle "Banks" between the eight faders to allow access to your entire mix. This allows you to mix with a level of speed and control that is simply not possible when mixing with just a mouse. Avid S3 or S6. With the S6, you can customize the surface with only the modules you need for great cost efficiency and expand the fader bank to as many tracks as you see fit. You have touch-screen control to get deep DAW functions mapped out on the desk.

Slate Media Technology offers up another interesting option for control surfaces in their series of Raven products. These control surfaces, including the Raven MTI2 and MTX MK2, allow mix engineers to touch the production console's screen with multiple fingers at once to adjust faders, pans, sends, and plug-ins. While users are still working within the digital space, Slate's Raven gives them the option to get more hands-on with their mix.

Some people think a control surface is just a big mouse, but I think it is a great way to improve your workflow and give you more of a connection with the mix.

Computer and Console
This method is for the more advanced engineer, but it truly delivers the best of both worlds. This requires a hefty I/O setup coming from your DAW, but in the modern studio, we have many great AD/DA converters such as the Avid HD I/O, Antelope Orion 32HD, Apogee Symphony, Burl Mothership, and Universal Audio Apollo x16. It's easier than ever to get high-quality digital signal to your analog desk.

This method also allows us to load up our tracks with any desired plug-ins inside our DAW, then send that processed signal to the desk. Pro mixers such as Michael Brauer, Tom Lord-Alge, and Greg Wells use this method on every mix. They can use plug-ins like the Thermionic Culture Vulture or Waves Manny Distortion to achieve nice thick harmonic distortions before the desk, or pre-EQ the messy parts of a signal to free up the desk EQ for more shine and focus. Using plug-ins before the desk also helps with session recall.

In this scenario, most of the heavy mixing is done inside the box, and the desk is used for its analog warmth and summing. Since the DAW will save all the plug-in presets and automation, all that is left to recall is fader position, aux sends, and some EQ. Some great consoles for hybrid mixing setups include While plug-ins are becoming very close to the real deal, they are still just emulations of the gear we have all come to love. There is nothing like adding a API's The Box, API 1608, SSL XL Desk or Neve 1073LB, API 550A, 550B or 560, MAAG EQ4, or SSL 611EQ to a track. There is a magic that happens when signal is passed through the analog circuit, a beautiful imperfection that can't quite be matched by a plug-in. Toft ATB Series. This set-up allows for an easier use of analog bus compression, some great choices for that are the Manley Vari-Mu, Shadow Hills Mastering Compressor and the UnderTone Audio UnFairchild.
Why Still Use Analog EQ?A lot of pro mixers will say an analog EQ can handle more information than a plug-in, meaning you can send more signal to them and it will make the harmonics react in a musical way. Where a plug-in seems to "fold" when you send it too much information, you can use an analog EQ as a hardware insert in your DAW. All this requires is an open output from your interface to the input of the EQ, then back out of the EQ into your interface.

Now you may be asking, what if I only have one EQ? Waves makes a plug-in called QClone that can copy your analog EQ settings and store them within a plug-in. This allows you to save the EQ you have on track one, move the EQ to track two, then completely change your analog EQ and save that setting again as a new preset. This allows you to use one EQ as many times as you want, to get as many sounds as you want.

Just like the EQ, nothing beats the sound of the real deal. What About Analog Dynamic Processing? Universal Audio has done an amazing job in emulating their 1176 and LA-2A in plug-in form, but having even one real piece of outboard compression can make a huge difference in a recording or mix. Unlike the EQ, compression doesn't have a plug-in like QClone that can save the compression setting, so we have to use it sparingly. Typically, we would use our nicest compressor on the most important element of a song, such as the vocal. To achieve this in the box, we would set up a hardware insert the same as the EQ mentioned above. Now that the vocal has a present, warm, in your face feeling, we can use our DAW to fill in the rest of the compression. In the old days, we needed racks of compressors like Chris Lord-Alge has to achieve a big aggressive mix. With hybrid mixing, you can use the analog gear you have for the most important elements of the mix, and use high-end emulations for everything else. Some great compression plug-ins are UAD-2 Manley Vari-Mu, Fairchild, Tube Tech CL1B, 1176, and LA-2A or Waves CLA76 and CLA 2A.

Now we can achieve the sound of high-quality vintage microphones through modern day digital processing. The At a little higher price point of $1,499, the Hybrid Microphones Slate VMS utilizes state-of-the-art amplifiers to produce the purest and most linear signal possible, giving the VMS plugin module the ability to transform your recordings with reproductions of classic microphones and preamps. One microphone can be used to model eight classic microphones, and for only $999, you can have one mic for any application. Townsend Labs Sphere L22 condenser microphone provides exceptional attention to detail in the mic as well as the plug-in. This mic can be used on its own, or modeled as a variety of vintage microphones. Also offers "Re Mic" technology that allows the user to change the mic type and polar pattern even after tracking. Contact our expert audio consultants today via email or by calling 888.653.1184 to assess your hybrid mixing requirements and answer any technical questions that you may have about this style of workflow.

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Hybrid Mixing Techniques: Leveraging The Best Of In The Box And Out Of The Box Environments

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The great debate: digital vs. analog. I'm sure most of us can sit here all day and discuss why one is better than the other. While there are great benefits to both, most of today's top engineers such as Greg Wells, Tom Lord-Alge, Tony Maserati, and Michael Brauer, are starting to use a combination, or hybrid system, for mixing.

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As I'm sure you're well aware, the analogue vs. digital debate still rages on despite digital's advancements over the years. Whether you grew up working on analogue equipment or simply call yourself a "purist," some folks still consider "old-school" recording and mixing techniques to be superior. Among this debate is the idea of analogue summing, and whether it is in fact superior to its digital counterpart.

We'll discuss what analogue and digital summing are, and why it may or may not be important to your recording and mixing process.

In a nutshell, summing is actually a very basic concept. All of the tracks in a mixing session must be routed to a common stereo output to print a final mix. In Pro Tools or any other comparable DAW, this common output is the master fader.

Summing is simply the process of combining all of a session's tracks and outputting them to a single stereo source.

Before digital, summing was done inside of a console. And when digital recording was developed, many people complained of its inferior summing quality. The "warmth" of analogue summing was argued to have been superior to the "harsh" and "cold" summing that was happening digitally.

If digital summing is thought to be the problem, then an obvious solution is to A) continue working completely analogue (wildly impractical for those of us working at home-or anywhere, these days), or B) adopt a hybrid analogue-digital approach.

The hybrid method involves taking your tracks out of a DAW, summing them with a hardware analogue device, and bringing them back into the DAW as a final stereo mix.

This affords the benefit of working conveniently inside a DAW, with the added bonus of the supposed "warmth" and "width" of analogue summing.

With so many amazing mixers working entirely in-the-box- Mark Needham, Dave Pensado, and Mark Endert, to name a few-it's hard to believe analogue summing is vastly superior to digital.

There will always be those engineers who swear by analogue, which is great! But at the crux of the analogue vs. digital summing debate, there are incredibly skilled engineers producing remarkable music on both sides. This proves that analogue summing is not a common denominator amongst some of the industry's best engineers.

Skill, talent, and years of experience tend to outweigh any perceived benefit of analogue or digital equipment. Thus, the gear successful engineers use often stems from personal preference, habit, or perhaps even a little superstition!

Ultimately, one type of summing is not objectively better than the other. This is frustrating to some degree, in that it makes our quest for better sounding mixes that more difficult. But it's also liberating in the sense that we have the freedom to choose whatever works best for us.

Everyone wants a game-changing piece of equipment that takes good mixes to great, but the truth is, skill and experience are what make mixes fantastic.

This is good news! Those of us working in bedroom studios with smaller budgets still have all the resources we need to produce pro-quality mixes, 100% in-the-box-even with stock plugins, too.

If the opportunity to try analogue summing presents itself, though, it's absolutely worth a shot. As we've seen, there are guys on both sides doing brilliant work, proving that it isn't about one or the other.

If possible, try both, and see which is preferable given your taste and workflow.

Before troubling yourself with analogue summing-especially if you work out of a small project studio-it's important to first focus on the fundamental aspects of mixing music.

Getting to know the equipment that you do have will generally yield better results than acquiring new gear without really knowing how to use it. Likewise, an analogue summing device is only useful if your ears can perceive a difference-and whether you even like that difference in the first place.

Instead of worrying that your "okay" mixes would be "good" or "great" with analogue summing, make sure you know how to adequately utilize compression, EQ, time-based effects, and other signal processing first.

It sounds rudimentary, but you'll undoubtedly go further when you've learned a ton about your in-the-box plugins, how they operate, and how you can squeeze the most from them to suit your mixing and style.

At the end of the day, analogue summing is great. Digital summing is great, too. Figure out what suits your needs, and never stop learning!

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What You Should Know About Analogue Summing - Produce Like A Pro

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As I'm sure you're well aware, the analogue vs. digital debate still rages on despite digital's advancements over the years. Whether you grew up working on analogue equipment or simply call yourself a "purist," some folks still consider "old-school" recording and mixing techniques to be superior.

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A common question is whether or not to send tracks to an effects send using pre-fader or post-fader - so let's try and clear it up a little.

We're looking at Logic here, but most DAWs will let you route to an effects send using either pre or post. When an Aux send is configured post-fader, the effect's wet-to-dry ratio is preserved as you raise and lower the dry channel fader. So lowering a channel's fader simultaneously lowers the Aux send signal level. That keeps you from having to adjust effects return fader levels in the mixer or wet/dry ratios at the effects processor every time you make a level adjustment with a track's channel fader.

In most cases, you'll want to keep the balance between a dry track and its effects relatively constant throughout the mixdown process so post-fader effect sends are used more often than pre-fader effect sends.

However, there are some interesting uses for the pre-fader setting, for example creating an effect such as a sound dissapearing into exapansive reverb. A good example of this is when you hear someone in a film walking and the steps emerge out of a distant reverberant sound, come more close and present as they "approach" you, and then return to the distant echo as they walk away. This is done using a blend of the original and effects channels and changing their levels independant of eachother using the PRE setting.

As its name suggests, a pre-fader send is not influenced by dry channel-fader moves, because the signal is sent to the processor through the Aux send before it gets to the dry channel fader. Therefore, the processed signal level from a pre-fader Aux send remains constant, no matter how you move its corresponding dry channel fader.

So, in the case of the "footsteps" example mentioned above, this is acheived by keeping the effect send reverb level constant while you slowly increse its dry level (giving the sense that the person is "approaching") and then returning it to a low level (giving the impression that they are walking away).

Try it yourself with a sound: Select a sound and send it to an effects bus with a reverb on it. Select PRE for your send. Now, set the level of reverb you want the sound to have in the mix by turning up its pre-fader Aux send, which is routed to your reverb unit, until the effect sounds right.

Then, slowly lower the sounds' dry channel fader. As you lower the fader, the sound's dry level dips while the level of the processed signal remains constant. The result sounds as though the sound is moving away from you: the dry sound gets quieter, leaving just the reverberations of the room. Of course, you simply need to slowly raise the dry channel fader again to give the impression that the sound is "coming back"

Another fun and useful effect to play around with.

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A common question is whether or not to send tracks to an effects send using pre-fader or post-fader - so let's try and clear it up a little. We're looking at Logic here, but most DAWs will let you route to an effects send using either pre or post.

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Learn to Use the World's Most Famous Opto Compressor
It's ironic that the compression technology that's perhaps the most difficult to wrap your head around - the fabled T4 opto-electrical mechanism that drives the Teletronix LA-2A Leveling Amplifier - is also the easiest to operate. Virtually anyone with ears can manipulate the LA-2A's ultra-simple two main control knobs - Gain and Peak Reduction - to bring signals under control with consistency and punch. But the LA-2A Classic Leveler Plug-In Collection adds some unique "under the hood" features and variation between the LA-2A plug-in models that may not be immediately obvious at first listen.

The Teletronix LA-2A Classic Leveler Collection includes units from three different LA-2A eras - Gray, Silver, and the original LA‑2.

The LA-2A Collection - Three Different Flavors

The Teletronix LA-2A Classic Leveler Collection includes units from three different LA-2A eras - Gray, Silver, and the original LA-2. Here's a simple breakdown.

The UREI-built Silver LA-2A is generally appreciated for its faster time constant and its treatment of transients. This makes it a versatile performer that's suitable for the widest variety of applications - including drums, percussion, and bass guitar.

The Gray model, paying homage to Jim Lawrence's original mid-'60s Pasadena-built units, is typically used for material requiring a medium-speed compression - think lead and background vocals, keyboards, and (judiciously) acoustic and electric guitars.

The LA-2 model captures the mojo of one of the very earliest Teletronix units, offering the slowest response time and a distinctive "mellow" sound owing to its 50-year-old luminescent panel. It may be best used on legato vocals, strings, and horns. But you know the old maxim, "If it sounds good, it is good!"

Dialing in the Perfect Compression

The LA-2A's proprietary T4 photocell essentially works on a fairly simple principle. As the input signal increases, the light panel in the T4 cell gets brighter. As the light gets brighter, the photo resistor's impedance increases, which reduces the gain. As the input level decreases, the light dims and the resistor's impedance decreases. For the recording engineer, accessing this little trick of the light couldn't be easier. The LA-2A front panel boasts only two main knobs: Gain (this refers to its up to +40dB of output gain or "make-up gain") and Peak Reduction, which on other compressors might be called "Threshold."

Compression settings for the lead vocal, with the LA-2A Silver.

In a typical scenario, one simply inserts an instance of the LA-2A into a channel strip with the channel fader at unity gain, sets the Gain Reduction to tame the peaks and transients sufficiently, and then adjusts Gain to bring the new, more consistent signal level back to the desired place in the mix. I've done this in this example using an isolated vocal track and the Silver LA-2A. Keep your eyes on the gain reduction meter (which you can set to either display gain reduction, or output gain with a metering reference of -4dB or -10dB) to see how often the peak reduction kicks in, and use your ears to tell you if that makes for a natural sound or an overly squashed one. (Both can be effective; it's your call.)Here's the lead vocal track before treating it with LA-2A compression:

Using Ross Hogarth's "Lead Vocal" preset as my starting point, I hit the Silver LA-2A compressor pretty hard here, with a high Peak Reduction and Gain at 50%, as I felt my original vocal sounded somewhat harsh. I used the Gain to help it inch up in my mix. Using my channel Send, I also blended another instance of the LA-2A Silver to add just a touch of even more aggressive compression. To further shape the sound, I've added some EQ after the compressor (though you can certainly place it beforehand) to bring out some of the high-mids I felt I may have lost, and to roll off a bit of low end. The sound is anything but "pristine," but I like the body and warmth of it. Sounds like tubes to me...

For the Gray LA-2A, Bass guitar is a natural. Let's take a listen to this Fender jazz bassline before compression:

And here's the same Fender bassline processed with a low-medium Threshold, and Gain at about 40% to add some of that nice LA-2A distortion and warmth.

Though the simple controls of the LA-2A are great, you may be wondering why there aren't Ratio, Attack, or Release controls. Those values, which are so intrinsic to the sonic character of the unit, are actually fixed. The average compression ratio is always set at roughly 3:1, while the average Attack time is 10 milliseconds, and the Release time is about 60 milliseconds for 50% of the release, and anywhere from 1 to 15 seconds for the rest. Remember, this is a program and frequency dependent compressor, so it reacts a little differently depending on what goes into it. If you prefer a little unpredictability in your mixes, that can be a good thing. If you're an addictive knob twiddler, though, have no fear - the LA-2A has several other features that allow you to add expression to its ingeniously simple design.

Utilizing the Emphasis Control

The LA-2A's R37 Emphasis control - that "set-screw" knob on the lower left of the front panel - controls a shelf filter circuit in the compressor's sidechain input for frequency-dependent compression.

Emphasis is handy for emphasizing compression on higher frequencies.

When the control is fully clockwise in the default position, the sidechain signal is unfiltered and all frequencies in the source signal that exceed the compression threshold will trigger gain reduction equally.Rotating the Emphasis control counter-clockwise increases filtering of the sidechain signal. The Emphasis filter gradually reduces the lower frequency content of the sidechain signal, resulting in compression that is less sensitive to those frequencies, and more sensitive to high frequency content. As the sidechain filtering is increased, higher frequencies are compressed more.You can think of the sidechain as the compressor's scolding big brother - the sidechain tells the compressor how it should behave respective to the incoming material, and it helps shape that material for processing.Here's a drum track before treating it with the LA-2A Silver:

I've chosen to use the Emphasis control at its furthest counter-clockwise setting to help accentuate the "thwack" of a snare drum in a stereo drum mix. The snare wasn't quite sitting where I wanted it, while my kick drum seemed quite present, so a bit of Emphasis didn't lose me any significant lows, but it did help round out my mid-rangey snare drum with a little extra punch.

The New York Parallel Compression Technique

Nothing screams Big Apple like a little parallel compression, also known fondly as "New York" compression, thanks to its birth among a small coterie of Manhattan-based engineers. Typically, a compressor like the LA-2A is always going to be used as a channel insert - after all, compression is a dynamics processor, not an "effect" per se, and it has traditionally been used to allow maximum control over the individual volume levels of mono instruments.

The "New York" compression technique takes another approach, typically sending one or more stereo mixes of the drums to separate bus channels, compressing one bus channel conservatively, and the other very aggressively, then adjusting the levels of those channels to taste. This method retains the highly dynamic quality of the less compressed material while judiciously adding the color and dynamics of the heavily compressed signal, often at exaggerated levels that might sound inappropriate if applied directly to the source. .

Some engineers may send less overhead and cymbals to the more aggressively compressed bus channel in order to diminish excessive high-end noise.This will impact your drum balance, of course, so you may need to play with the respective channels' compressor settings until you find the right blend, as I've attempted to do in the following drum part. Note how much more aggressive this one is than the very same part in the previous drum example, with only a moderate Insert compressor with Emphasis.

A Few More Examples

Here are a few more examples of how the LA-2A sounds on instruments. It's legendary for use on pianos and helps bring out the tube warmth of many vintage guitar amps, too.Here's an example of a British Combo rhythm guitar line, first dry:

And here it is with moderate compression from the LA-2A Gray:

Here's an example of an acoustic piano, first without compression...

...then with the "JJ Bright Piano Warmth" present for the LA-2:

While most engineers will argue that the LA-2A is a distinctive flavor best used on highly dynamic "live-sounding" material, you may find uses for it that go well beyond its traditional settings. With three different flavors in the LA-2A Collection, the sky's the limit. Give it a try for yourself, and feel free share some of your own tips in the comments section below with other LA-2A afficionados.

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Tips & Tricks - Teletronix LA-2A Classic Leveler Plug-In Collection | Universal Audio

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Learn to Use the World's Most Famous Opto Compressor It's ironic that the compression technology that's perhaps the most difficult to wrap your head around - the fabled T4 opto-electrical mechanism that drives the Teletronix LA-2A Leveling Amplifier - is also the easiest to operate.

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The year is 1966. Simon and Garfunkel's album Sound of Silence is released. Herb Alpert and the Tijuana Brass is outselling the Beatles two to one, with four albums in the Billboard Top 10. On the radio, you will hear "Last Train to Clarksville" by the Monkees, The Supremes "You Can't Hurry Love," the Beatles "Nowhere Man," and the Beach Boys "Good Vibrations."

That same year, Bill Putnam is making the transition from the old standard vacuum tubes to new solid-state technology with his Universal Audio-brand preamps and compressors. The legendary engineer and equipment manufacturer had previously redesigned his original 108 tube microphone preamp (taken from his UA/United Recording consoles in Chicago and Hollywood), into the new 1108 - utilizing the recently invented Field Effect Transistor (FET). Subsequently, he redesigned his successful 175/176 tube compressor (based on the popular variable-Mu design) using FETs, and thus was born the 1176.

History

Billed in the 1968 release as a "true peak limiter with all transistor circuitry and superior performance on all types of program material," the 1176's major selling point was its ultra-fast attack time - a mere 20 µS (.00002 seconds) at its fastest setting. It also offered contemporary design, featuring knobs with clear surrounds, pushbuttons, and a brushed aluminum face panel with a blue stripe near the VU meter - none of those then typical big Bakelite knobs in sight. The initial price in 1967 was $489.00.

Similar to the 176, the 1176 featured no threshold control - just input and output controls - and included a continuously variable attack and release, which was a novelty at the time. The amount of compression was determined by setting the input level - the hotter the signal reaching the detection circuits, the greater the resulting compression. Another unique feature was the selectable ratios of 4:1 and 8:1 for compression, and 12:1 and 20:1 for limiting.

The Rev A Bluestripe 1176 compressor, known for its aggressive sound.

The 1176 became an instant favorite with producers and engineers because of its unique lightning-fast attack and release times (20 µS to 800 µS Attack, 50 mS to 1.1 seconds for Release), its Class A output stage, and its wide range of sounds, ranging from a very subtle, near-transparent compression at 4:1, to its most notable setting, the "All Buttons In" mode, where all the ratio buttons are depressed simultaneously. This allowed the 1176 to make a sound unlike any other processor ever heard before. Distortion increased, along with a plateaued slope and a lag time in response to initial transients, creating an explosive sound on drum room mics, making an incredible grungy bass or electric guitar sound, or squeezing a vocal so it sat right in your face.

Sound

So what does an 1176 "sound" like? The characteristic that most 1176 lovers agree on is that what comes out is better than what went in. There's a brightness and presence, an "energy" imparted, according to Mike Clink from Guns 'N' Roses. Bruce Swedien used his 1176 on all the Michael Jackson vocals he recorded. Some even use the 1176 with the compression turned off, just for the distinctive tone it imparts. The amplifiers and transformers all by themselves give a desirable "hot" quality to anything passing through them.

How It Works

The 1176 is a "feedback" compressor. The signal goes through the input stage, via the line input transformer, which has a big impact on the sound. From there, the signal is routed to the top of a FET, which is being used as a variable voltage resistor. Depending on the revision of the unit, the FET is either used to shunt some of the input signal to ground, or as the top leg of a voltage divider in parallel to the input of a pre-amplifier stage. The resistance between the Drain and Source leads of the FET will change in response to voltage applied to the Gate lead.

In the later revisions, the relationship between fixed resistances and the FET's dynamic resistance affects the gain of the circuit. As the signal flows to the output stage, it is tapped and also sent to a sidechain detector circuit. This sidechain develops a Gate control voltage for the FET based on the dynamics and level of the input signal, as well as the front panel control settings.

Universal Audio engineers bench testing the 1176.

For minimal distortion, careful circuit design was required to make the FET operate within a narrow linear range. The output stage is an all-Class A line level amplifier, designed to drive the 1960s standard load of 600 ohms. The output transformer is an integral part of the sound of the 1176, balancing the impedance matching for the output. By enclosing the negative feedback of the output amplifier, its non-linearities were compensated and corrected for, producing very low output distortion.

Because of the input and output transformers, and the Class A output circuitry, the 1176 has a very distinctive sound even when it is not compressing. By turning the Attack knob fully counterclockwise to the OFF position, the unit can be used as a straight line amp just for its unique sound. The 1176 also offers a whopping 45 dB of gain at .5% THD - as much gain as some mic preamps, and far more than most compressors.

The input knob controls the threshold, but the threshold is also dependent on the ratio selected - the higher the ratio, the higher the threshold. As the input level is turned up, a corresponding decrease in output level may be required. The front-panel meter will indicate GR (Gain Reduction) or output level at either +4 or +8 dBm. Output metering (as opposed to GR) will introduce additional distortion.

Brochure for the Brad Plunkett-designed 1176LN.

One of most popular features of the 1176 is its ability to vary its release time based on program material. After a transient, it releases quickly to avoid level drops, but if there is a continued state of heavy compression, the 1176 will exhibit a longer release to reduce pumping.

Revisions

One thing any 1176 devotee knows is that not all 1176s are created equal. There was a long trail of revisions as the unit progressed through the years. The initial units released in 1967 were A's, followed within months by AB revisions, and B revs in 1968. Rev C came in 1970, and sported the familiar all-black face panel. These were badged 1176LN, after a redesign by Brad Plunkett lowered the noise and increased linearity. The D and E revs from 1970 to 1973 are widely considered to be the best-sounding units. Additional gradual design changes were made to the audio circuitry from 1973 onward.

There is often a debate that rages about which revision sounds best, but aging components and the different calibrations of older units can drastically impact the sound regardless of which revision it is. Universal Audio actually has a full revision history of the 1176 and LA-2A, which you can find here.

In 2000, Universal Audio began reissuing the 1176LN. These units are still handmade today in Scotts Valley, California.

The Next-Generation 1176

In 1999, when Bill Putnam, Jr. relaunched Universal Audio, a reproduction of the Blackface 1176LN was the company's first product. Reissued in 2000, it was based on the circuit designs of the C, D, and E revs. One of the most critical aspects of recreating the sound of the original 1176 was faithfully reproducing the original output transformer, complete with the additional windings that provided the feedback signal to the final line output amplifier circuit. Bill Jr. found extensive design notes that enabled him to recreate and improve on the original design.

All these factors combine to make the 1176 one of the most unique and beloved compressors of all time with a huge devoted fan base. Still manufactured today, 45 years since its introduction, it is truly a studio legend.

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UA's Classic 1176 Compressor - A History | Universal Audio

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The year is 1966. Simon and Garfunkel's album Sound of Silence is released. Herb Alpert and the Tijuana Brass is outselling the Beatles two to one, with four albums in the Billboard Top 10. On the radio, you will hear "Last Train to Clarksville" by the Monkees, The Supremes "You Can't Hurry Love," the Beatles "Nowhere Man," and the Beach Boys "Good Vibrations."

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Article Content

The Neve 1073 is arguably one of the most popular preamps in the history of recorded music.

Originally introduced as part of the 80 Series consoles in the 1970s, these colorful preamps are known for their fat, warm sound and rich low end.

Vintage Neve 1073 preamps sound amazing, but their limited feature set and astronomical price tag push many engineers towards digital alternatives. In this roundup, we'll break down the top five plugins based on the Neve 1073 to help you figure out which one is right for you.

1073 Design

The original 1073 preamps utilize a very streamlined design. In addition to a single gain control knob, which can be used to drive signals until they saturate, the 1073 also features a three-band semi-parametric EQ and selectable high-pass filter.

The top EQ band controls a fixed shelf at 12 kHz. The midband controls a bell curve with selectable frequencies at .36, .7, 1.6, 3.2, 4.8, and 7.2 kHz. The low band controls a shelf with selectable frequencies at 35, 60, 110 and 220 Hz. The selectable high-pass filter can be set at 50, 80, 160 or 300 Hz. Each band offers +/- 16 dB of gain.

About the audio files ...

We put some of our favorite 1073-style plugins through the paces on drums, bass, guitars and vocals with the following settings:

Drums

Bass

Guitars

Vocals

Here are the unprocessed audio files ...
Drums
https://theproaudiofiles.com/wp-content/uploads/2019/02/Drums_1.wav
Bass
https://theproaudiofiles.com/wp-content/uploads/2019/02/Bass_1.wav
Guitar
https://theproaudiofiles.com/wp-content/uploads/2019/02/Guitar_1.wav
Vocals
https://theproaudiofiles.com/wp-content/uploads/2019/02/Vocals_1.wav

UAD offers "the only authentic end-to-end circuit emulation" of the Neve 1073. The UAD 1073 captures all of the original's unique circuit behaviors, including all 10 clipping points from the preamp and EQ stages for that classic gritty class‑A saturation.

The UAD 1073 also utilized UAD's Unison technology, which allows you to use the plugin while tracking.

In terms of features, the UAD 1073 sticks very closely to the original design. The only new features are an impedance switch and a switchable -20 dB pad.
Drums - UAD
https://theproaudiofiles.com/wp-content/uploads/2019/02/Drums_2.wav
Bass - UAD
https://theproaudiofiles.com/wp-content/uploads/2019/02/Bass_2.wav
Guitar - UAD
https://theproaudiofiles.com/wp-content/uploads/2019/02/Guitar_2.wav
Vocals - UAD
https://theproaudiofiles.com/wp-content/uploads/2019/02/Vocals_2.wav

Slate has a plugin for everything, and in this case, they actually have three plugins dedicated to emulating the legendary sound of the 1073 preamp.

First, the FG-73, available as part of the Virtual Console Collection, offers simple drive and trim controls for emulating the dual-stage red band preamp of the 1073.

The FG-N, available as part of the Virtual Mix Rack 2.0, is a modified version of the 1073 EQ with variable frequency controls and an additional midrange bell. It also features a variable gain control with an optional "drive" button for added grit - similar to the FG-73.

Finally, the Virtual Channel and Virtual Mixes plugins, both available in the Virtual Console Collection, feature Brit-N settings for emulating the channel circuitry and summing characteristics of using running a signal through a Neve console.

*For the purposes of this test, the VCC (in Brit-N mode) and FG-73 plugins were used with settings at default. Drive set to 6 (10 o'clock); Input and Out set to zero. The FG-73 is set to 20 (the minimum setting).
Drums - Slate
https://theproaudiofiles.com/wp-content/uploads/2019/02/Drums_3.wav
Bass - Slate
https://theproaudiofiles.com/wp-content/uploads/2019/02/Bass_3.wav
Guitar - Slate
https://theproaudiofiles.com/wp-content/uploads/2019/02/Guitar_3.wav
Vocals - Slate
https://theproaudiofiles.com/wp-content/uploads/2019/02/Vocals_3.wav

6020 Ultimate EQ is a collection of ten 500-series style equalizer models, including the British-E EQ, based on the 1073. Although the British-E is only a small portion of what this powerful plugin has to offer, it does an amazing job of emulating the sound of Neve EQs.

British-E features more versatile filter shapes than the original design with three fully variable bands: a high-shelf, a midrange bell, and a low shelf. It also includes a fully variable high-pass filter.

The most unique thing about this plugin is that it does not include markings for frequency or gain controls, forcing you to rely on your ears. For these examples, I attempted to match the settings as best as I could by ear.
Drums - McDSP
https://theproaudiofiles.com/wp-content/uploads/2019/02/Drums_4.wav
Bass - McDSP
https://theproaudiofiles.com/wp-content/uploads/2019/02/Bass_4.wav
Guitar - McDSP
https://theproaudiofiles.com/wp-content/uploads/2019/02/Guitar_4.wav
Vocals - McDSP
https://theproaudiofiles.com/wp-content/uploads/2019/02/Vocals_4.wav

The EQ 73 by IK Multimedia is a "true to the original" reproduction of the 1073, including all of the gain stages, circuit designs, EQ curves and preamp behaviors. IK have also emulated the preamp/EQ interaction, which "truly defines why this unit was so special," according to them.

Aside from the addition of stereo and mid-side processing, the EQ-73 also stays true to the original feature set.
Drums - IK Multimedia
https://theproaudiofiles.com/wp-content/uploads/2019/02/Drums_5.wav
Bass - IK Multimedia
https://theproaudiofiles.com/wp-content/uploads/2019/02/Bass_5.wav
Guitar - IK Multimedia
https://theproaudiofiles.com/wp-content/uploads/2019/02/Guitar_5.wav
Vocals - IK Multimedia
https://theproaudiofiles.com/wp-content/uploads/2019/02/Vocals_5.wav

GOLD2 is much more than a simple preamp emulation. It includes six different equalizers, two compressors, seven preamplifiers and advanced routing emulations.

The GOLD2 package includes individual plugins for the preamp, EQ and compressor modules, as well as a channel strip version. Each plugin also includes a zero latency or "ZL" version for use during tracking.

While GOLD2 doesn't introduce any new features to the 1073, it offers circuit emulations for several Neve consoles for added versatility.

*For the purposes of this test, only the H073 module and H073 summing were used.
Drums - Acustica
https://theproaudiofiles.com/wp-content/uploads/2019/02/Drums_6.wav
Bass - Acustica
https://theproaudiofiles.com/wp-content/uploads/2019/02/Bass_6.wav
Guitar - Acustica
https://theproaudiofiles.com/wp-content/uploads/2019/02/Guitar_6.wav
Vocals - Acustica
https://theproaudiofiles.com/wp-content/uploads/2019/02/Vocals_6.wav

Honorable Mentions

This is by no means an exhaustive list. There are dozens of emulations of 1073-style preamps and EQs.

Waves have an excellent version designed by the legendary Andrew Scheps called the . It offers the exact same controls as the original with the addition of "a little-known 10 kHz midrange band discovered in the original 1073 schematics but operable only on the rare 1078 channel strip."

The Stillwell 1973 Equalizer is a very affordable 1073 "tribute". While it doesn't model the 1073 exactly like many of the other plugins on this list, it "takes the best features and reduces them down to their essence."

Unique Recording Software's EQ does a great job of capturing the 1073 sound, but it also utilizes features from multiple Neve modules. The URS N4 offers four fully parametric EQ bands, as well as high-pass and low-pass controls, and a gain knob for dialing in distortion.

And that's just scratching the surface! What are some of your favorite Neve 1073 emulation plugins?

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The Neve 1073 is arguably one of the most popular preamps in the history of recorded music. Originally introduced as part of the 80 Series consoles in the 1970s, these colorful preamps are known for their fat, warm sound and rich low end.

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When you're mixing music there are two ways to affect the audio you are working with. Whatever your processor does there are two ways to make it work on a signal. Either by using it as a send effect, or as an insert.

But what's the difference between the two?

What's a send effect

Send effects are used when you have an original signal that you want to modulate, modify or process in any way without affecting the original signal. Your original signal sits there by itself unchanged but it has a separate copy being effected. You hear this with delays and reverbs; the original audio is there as well as the delayed signal working together to create a nice effect.

You use auxiliary sends to copy the original audio to a separate track where you further process it.

What's an Insert?

An insert is used when you want to process a signal directly. You insert an effect, such as a compressor or EQ directly onto the original audio. This is when you do not want to retain the original audio file. You want to change it completely(or subtly) and only hear the processed part.

So when is it smarter to use sends or inserts?

Well, you want to use sends when you are using time based effects. Time based effects, such as delays need an original sound source to work. How would you know a signal was delayed if it didn't have an original un-delayed track.

Inserts are used for processing directly. If you need to compress a signal, gate it to eliminate bleed or process it in any way that doesn't affect the time of the signal.

Compressors are not time based, they are dynamic processors that work directly on a signal. That's why you should usually use them as inserts. If you were to use a compressor as a send you wouldn't be compressing the original signal, you would have a copy that's compressed but it wouldn't matter because your original is still unchanged.

Breaking the "Rules"

Of course, there are instances where you can break the rules. Using compression as a send is a well known trick. It even has a swanky name, The New York Compression Trick.

Mo Volans' easy parallel compresison article on Audiotuts teaches you how to do a similar thing without using a send. You can just use the compressor as an insert as you normally would and then use the mix slider to apply the parallel compression.

Guitarist that use delays, reverbs and modulation effects in their signal chain are essentially using time based effects as inserts. But most guitar effects have a mix parameter to allow you to blend the two signals.

You can do the same thing with your DAW. For instance, if you have a guitar track you can usually insert modulation effects as an insert. Just be sure that you're blending the two signals. You don't want an insert to be 100% wet(or effected) since you won't have any of the original signal.

Summing up

It's smart to use send effects when you want to enhance a signal without eliminating the original. By adding a delay to an already nice sounding signal you just make it sound better.

The opposite goes for inserts. Use insert effects when you want to change the signal completely. Inserts change the signal with dynamic changes like compressors, or tonal changes like EQ.

Both methods have their uses, for certain situations.

How are you breaking the rules lately?

For more mixing tips like these, sign up for my free EQ course. You will get over 70 hacks to clean up your mixes and have all the track work together instead of fighting each other.
Sign up here.

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When is it Smarter to Send or Insert When Mixing Music?

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When you're mixing music there are two ways to affect the audio you are working with. Whatever your processor does there are two ways to make it work on a signal. Either by using it as a send effect, or as an insert. But what's the difference between the two?

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View of Manuel Haible, Heiko Wommelsdorf and Georg Werner's 2010 installation, 'Klanglumineszenz'. Photography: Roman März / Singuhr Hoergalerie 'The only technical things I know are treble, volume and reverb.' Johnny Thunders - New York Dolls What is Reverb? Reverberation , in psychoacoustics and acoustics , is a persistence of sound after the sound is produced. Valente, Michael; Holly Hosford-Dunn; Ross J. Roeser (2008). Audiology . Thieme. pp. 425-426. ISBN 978-1-58890-520-8 . All the time, in every space that we're in, reverb is always there. Reverb is the way sound interacts with the physical spaces that we're in. It affects a sound after it has stopped sounding. If you hit a drum in a room, what you hear is a mixture of the original sound and the reverb, which is reflected and diffused versions of the same sound, bouncing back to you from the surfaces in the room. Reverb is amazing and it's no wonder that it has become

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View of Manuel Haible, Heiko Wommelsdorf and Georg Werner's 2010 installation, 'Klanglumineszenz'. Photography: Roman März / Singuhr Hoergalerie 'The only technical things I know are treble, volume and reverb.' Johnny Thunders - New York Dolls What is Reverb? Reverberation , in psychoacoustics and acoustics , is a persistence of sound after the sound is produced.

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This month we're looking at two studio legends: the 1176LN Limiting Amplifier and Teletronix® LA-2A Leveling Amplifier compressors. Each one is quite different from the other. We've done articles and videos in the past showing how they work, but one little-known compression secret is that they can be used together on the same track. Why, you may well ask? Because it gives you more control over the compression and can create a very smooth effect, especially on a track that varies wildly in dynamics.

Sometimes you can't reign in the dynamics the way you need to with one compressor without causing audible compression artifacts. For example, if you like the sound of getting 5 to 7 dB of compression with the LA-2A, a large peak might suddenly cause 12 dB of compression and the release might not recover in time for the next audio transient. You'll hear the level slowly come back up: your basic "pumping and breathing" effect. So using a fast compressor right before the LA-2A does the trick. Some people use three or more compressors on a track! One basic rule is that each successive compressor has a lower compression ratio than the one before it, and the first one is typically just catching the peaks.

To review the two compressors, the 1176LN Classic Limiting Amplifier is a FET-based compressor. It has Input and Output gain, Attack and Release controls, and four compression Ratios. It doesn't have a threshold control, but turning up the input gain has the effect of turning down the threshold. Aside from that, it is very much like most utility compressors.

The LA-2A Classic Leveling Amplifier, on the other hand, is an opto-based compressor, which means it uses a electro-optical device to perform gain reduction. There are no attack, release, or compression ratio controls. There is Peak Reduction, which increases the amount of compression by lowering the threshold. The tube-driven output Gain control makes up for the gain lost by the peak reduction. Finally there is the Limit/Compress switch, which increases the compression ratio from around 3:1 (compress) to 10:1 (limit).

The most common combination is to put the 1176 first and set it up as a peak limiter. To do this, set a high compression ratio, like 12:1, and a fairly fast attack and release. The point here is to just flatten the higher peaks. This compressor should not be doing a whole lot, just knocking 2 to 3 dB off of the loudest peaks.

The input gain needs to be adjusted carefully to do this. Usually it is best to find a part of the track with the highest peak or peaks and cycle that, and adjust to get the effect I described. Then use the output gain to match the level. To do this, disable the compressor and listen to the average level of your track, then enable it and adjust the output gain until you can't really hear a difference in level.

The LA-2A should be set up just like you would if it were being used alone, with the Compress/Limit switch set to Compress, and Gain Reduction and Gain set to taste. But the 1176 knocked the big peaks down, which would have thrown the LA-2A into quite a bit of compression.

You can also reverse the order of the two, using the LA-2A in Limit mode to knock down the peaks, and the 1176 to smooth out the rest of the track. Just like all audio tools, there are no hard and fast rules. If something has the desired effect, go for it!

Check out the video to see both of these compressors being used together on a variety of audio tracks.

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Chaining the 1176LN and LA-2A Compressors for Maximum Control | Universal Audio

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This month we're looking at two studio legends: the 1176LN Limiting Amplifier and Teletronix® LA-2A Leveling Amplifier compressors. Each one is quite different from the other. We've done articles and videos in the past showing how they work, but one little-known compression secret is that they can be used together on the same track.

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The sounds of 1983 | Yamaha DX7 demo | Famous sounds [01]

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Review of the Prophet Rev2 by Dave Smith

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Review of the Prophet Rev2 by Dave Smith

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Review of the Prophet Rev2 by Dave Smith - is it the analog poly synth for you?

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Excuse the simplicity of the question, but I'm always coming across these terms in the magazine, and I don't know what they are: auxes, buses, sends and returns. Can you explain to me what are? Are they all part of the same thing or completely unrelated?

Tony Robbins via email

SOS contributor Mike Senior replies: All of these terms are related, in that they are all ways of talking about the routing and processing of audio signals. The word 'bus' is probably the best one to start with, because it's the most general: a bus is the term that describes any kind of audio conduit that allows a selection of different signals to be routed/processed together. You feed the desired signals to the bus, apply processing to the resulting mixed signal (if you want), and then feed the signal on to your choice of destination. If that description seems a bit vague, that's because buses are very general‑purpose.

For example, it's common in mixing situations to hear the term 'mix bus', which is usually applied to the DAW's output channel. In this case, all the sounds in your mix are feeding the bus, and it might then have some compression applied to it before the sound is routed to a master recorder or recorded directly to disk within the software. A 'drums bus', on the other hand, would tend to refer to a mixer channel that collects together all the drum‑mic signals for overall processing, routing them back to the mix bus alongside all the other instruments in the arrangement. Other buses are much simpler, such as those that can be found on a large‑scale recording mixer, feeding the inputs of the multitrack recorder, or those which carry audio to/from external processing equipment. Some don't even provide a level control.

An 'aux' is just a type of bus that you use to create 'auxiliary' mixes alongside that of the main mix bus: each mixer channel will have a level control that sets how much signal is fed to the aux bus in question. What you do with your aux buses is up to you: the most common uses are feeding a cue signal to speakers or headphones, so that performers can hear what they're doing on stage or during recording; and sending signals to effects processors during mixing. In the latter case, the aux bus that feeds the effects processor is usually referred to as a 'send', while the mixer channel that receives the effect processor's output will usually be called the 'return'. For more information, check out Paul White's 'Plug‑in Plumbing' feature back in SOS April 2002; you can find it at /sos/feb02/articles/plugins.asp.

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Excuse the simplicity of the question, but I'm always coming across these terms in the magazine, and I don't know what they are: auxes, buses, sends and returns. Can you explain to me what are? Are they all part of the same thing or completely unrelated?

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SOS Article - Studio Installation Workshop: Patchbays

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SOS Article - Studio Installation Workshop: Patchbays

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A technical overview of patchbays

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A patchbay can make your studio much quicker to use, but can also be a nightmare to set up, so here's advice on which to choose and how to connect it up for the best results. Plus, tips on building and fitting racks for your gear.

In my two previous studio installation articles (SOS September and November 2002), I've looked at soldering and wire preparation techniques, basic signal types and ground loops. In this part, I would like to build on these concepts by covering the theoretical and practical aspects of patchbays.

Patchbays, also known as patchfields or jackfields, hail back to the bygone days of manual telephone exchanges. In those days, to place a telephone call, the user called the operator who then physically connected the call by means of patch cables in a jackfield. In the UK, the connectors were designed by the General Post Office and termed GPO jacks. Variants of these original types later gained the additional designation of 'B type', 'GPO B' or 'BPO'. This type of connector is still used today in full-size professional audio jackfields and serves the same purpose as the telephone exchange: manually routing signals from one location to another.

It is important to note that, although a GPO jack has a similar quarter-inch sleeve diameter to the domestic (or A-type) jack found on instruments and guitar leads, the ring and tip are very different, and the two are physically incompatible. In addition to not forming a correct electrical connection, plugging a domestic jack into a GPO socket may well cause damage to the socket. The contacts become stretched and weak as a result. This also reduces the switch contact pressure and causes intermittent normalling, more of which in a moment. Conversely, because the ring and tip of B-type connectors are smaller than those of domestic types, plugging a GPO jack into a domestic jack socket will usually result in only the sleeve connecting.

In more recent years, as mixing consoles offered more and more inputs and outputs, the traditional GPO jacks started to consume too much panel space and a need arose for a smaller connector. This requirement gave birth to the bantam or TT ('tiny telephone') jack. These are physically much smaller than the GPO type, but are constructed in exactly the same manner - as the saying goes, if it ain't broke...

Although the bantam jack is long established, a core of audio professionals (myself included) still prefer the older GPO connectors. I personally dislike two aspects of bantam jacks. Due to their small size and resulting packing density, it is easy to skin your fingers when inserting bantam jacks, and I also feel that the larger contact material used in GPO types maintains a greater contact force over time, although I have not proven this. Perhaps my BBC background has biased me. One thing that definitely makes a difference to either type of connector is regular cleaning. The GPO connectors are often made of brass, which tarnishes quickly. A regular clean with a non-abrasive metal cleaner works wonders. As with all contacts in your signal path, the better the contact, the better the audio quality.

Choosing the most appropriate connector type for your installation is quite straightforward. For home studios, domestic A-type connectors are appropriate. These are much cheaper than the other types and are directly compatible with instrument leads and jacks found on much semi-professional equipment. This compatibility will allow you to mix and match patch leads with instrument cables for maximum flexibility. Even if you have to use an instrument cable in a balanced jackfield, with this type of connector the plug sleeve will correctly unbalance the signal path concerned by shorting the cold signal leg to ground via the sleeve. In semi-professional and professional installations, bantam or GPO connectors should be used. These offer much better long-term reliability and compatibility with hired professional cable looms, for example.

The term used to describe default signal paths through a jackfield is 'normalling' or 'normalising'. This concept is at the very heart of flexible jackfield design and will be examined in detail in a moment. Before we do that, it is important to discuss the physical layout of the jack sockets within a field. To determine the optimal layout, we first need to decide which equipment will normally be connected together, and also which will not usually be connected by default.

As an example, we will assume that we have the following elements to include on a jackstrip:

Mixing console stereo outputs.

DAT recorder inputs.

DAT recorder outputs.

Console stereo monitor returns.

Usually, you'd want to connect from the mixer outputs to the DAT inputs, and from the DAT outputs to the mixer monitor returns. On our jackstrip, it is convenient to arrange the connectors as in Figure 1. When designing strips, as a general rule I like to have outputs on upper rows and inputs on lower rows. This will not always be possible if you're going to use strips of patchbay sockets most efficiently, but using this layout rule where possible will give a very neat and logical layout and allows jacks to be normalled using very short wires. The normalling wires should be long enough to allow repairs and modifications to be carried out without having to replace the normalling wires, yet short enough to not pick up interference. They should also be held on to the sockets using hook-on joints as described in the first part of this series - other wires may then be added using lay-on joints without the normalling falling off.

If you need to normal jacks that are up to around 150mm apart, the normalling wires may be twisted using an electric drill. To do this, first cut the individual wires to the same length. Next, place one end of the wires in a vice, or hold with pliers. Finally, place the other ends in a drill chuck and then twist while keeping them taught. You will find that it is best to over-twist them slightly, as they will unravel a little when you release them from the chuck. With lengths beyond 150mm, when connecting between different strips for example, screened cable should be used for the normalling. In this situation, always remember to allow enough cable length to enable each strip to be withdrawn from the patchbay individually. Routing long normalling cables 'as the crow flies', will cause major problems later!

To keep the vast number of cables within a patchbay tidy, it is vital that the cables are run in an organised fashion, both within the rack and also along the jacks. It is important to avoid having the weight of the cable pulling on the jack sockets. To assist with both of these points, jackstrips usually include a metal bar that runs across the rear of the panel between the rows of jacks. This is known as a lacing bar, tie bar, or cable strain relief bar. The cables should be tied to this bar and neatly fanned out to each jack in turn, and always remember to mark the individual cables at the rear of the sockets to aid installation, modification and fault finding.

If you are constructing a simple unbalanced jackfield using two-pole A-type sockets for outboard equipment, the ground connections can also be made via the normals. Now when a piece of equipment is overpatched, the ground is overpatched with it, therefore avoiding unpredictable ground combinations. On the subject of earth loops, a word of caution. If you are using a multi-pair cable with foil screens but without individual jackets (most installation cable will be of this type), be very careful to dress the cables so that the foils do not unravel and short to one another. If they do, unavoidable ground loops will result.

Parallels are a collection of jacks connected in parallel with each other, as shown in Figure 3 - a good use for an odd collection of redundant jacks on a strip. They are used to feed multiple inputs from one output. This technique is useful, for example, when making copies using multiple machines.

Tie lines are simply a collection of cables, or a multicore cable, which run between two or more areas of the studio. One end of the cables is terminated onto jacks, whilst the other may be connected either to a piece of equipment, to a wall- or floor-mounted connector box, or to another jackfield. Tie lines are great for quickly routing signals. When planning your studio, always think about tie lines. Even a cable from one side of the room to another will help you avoid a mess of instrument cables strung across the room to the patchbay.

Finally, any good jackstrip will provide a way of identifying the individual jacks. On cheap strips this may simply consist of a white painted area in which to write. Although better than nothing, this method does not allow for easy updating or correction of errors. A far better solution is provided by paper strips in slide-in carriers. This type is found on all professional jack strips and is easy to change as required. A bonus of this type is that a computer can be used to neatly print the strips, which are then cut down to fit the carrier.

It is possible to construct racks in a number of different ways. Even a simple wooden cabinet, with wood screws and washers used for mounting equipment, is better than having expensive equipment in a disorganised heap on the floor. A very cheap, albeit time-consuming, method is to use proprietary square-tube systems which have knock-together corners. The two front uprights may then be drilled and tapped to take rack bolts directly. A simple MDF top may then be fitted to keep the dust out.

A faster method, again employing square-tube or wooden construction, is to use pre-punched metal rack strips. These are inexpensive and supplied in lengths of around 46U (a typical outboard effects unit is 1U). To use them, you should build a framework or open-backed cupboard to the dimensions shown in Figure 4, then cut the strips to length and fix them using wood screws or nuts and bolts.

Professional equipment racks are available in a range of sizes. They are normally very expensive, but can be found secondhand. These offer a huge range of options including:

locking removable side and front panels.

wheels.

rear support for heavy items such as power amplifiers.

captive power distribution.

fan-assisted cooling.

If using rack strips or ready-made racks, equipment is held into the rack by the use of rack nuts, also known as caged nuts, and rack bolts. The nuts offer great flexibility, as you only need to purchase enough for the job in hand, usually four per rack item, and fit them as required. If you later change the layout or make a mistake, the nuts may be sprung out and reinserted in a new location. Rack bolts can be fitted with flat or cupped plastic washers to protect the face of the gear. Although a dedicated cage nut tool is available, the nuts may be easily fitted and removed using a flat-blade screwdriver.

A number of suppliers offer security bolts for rackmounted equipment. These are a good investment and will foil casual theft during a house break-in, for example. The bolts use a modified Allen-type key and are all but impossible to remove without it. It is even extremely difficult to drill them out, as the bolts are made of stainless steel and have a centre 'pip'. Just remember where you hid the key - once again I have the T-shirt on that one...

When placing items other than jackstrips in a rack, thought should be given to spacing between equipment. Many items will require air flow to maintain a sensible operating temperature. Take care to mount these at least one unit (1U) height apart and avoid blocking any ventilation slots. The unused rack space may be filled with readily available blank panels which are made in a variety of colours and finishes. Perforated ventilation blanks are also available to further aid cooling of difficult units such as power amplifiers.

When fixing units to a rack, our old adversary the earth loop also rears it's ugly head once again. The act of screwing grounded metal-cased equipment to a common metal frame causes loops to be created via the power grounds. Although this will not normally cause problems in balanced systems using my suggested earthing strategies, unbalanced installations can suffer badly. If the equipment is fitted with a ground-lift switch, try using this first. The switch should disconnect the audio path of the equipment from its chassis and therefore cure the problem. If a ground lift is not available, it is possible use insulators and plastic washers to isolate the equipment casing from the rack. This approach should be viewed as a last-ditch fix, however, and will often indicate problems in the basic earthing scheme. If all other suggestions have been tried, the only cure will be to use transformers in the input and/or output wiring of the unit. As usual, under no circumstances should a power ground be disconnected.
Thanks to Bryant Broadcast & Data Communications (www.bryant-broadcast.co.uk) for supplying materials used in this article.
It's also worth mentioning those patch panels that consist of jacks on the front wired to jacks on the rear. These panels may be used with pre-wired looms to form simple jackfields without resorting to the soldering iron, and are therefore quite popular. The key to success with this type of panel is the word 'simple'. Where one machine or a bunch of tie lines need terminating, this type of pre-wired system can be very useful. In larger systems where much normalling and cross-wiring is required, pre-wired systems will not offer enough grounding flexibility to guarantee good results in all situations. I have come across systems of this type which work fine until external equipment is introduced, or an unusual patch configuration is required. The system then degrades into a mess of earth loops, and therefore hum and noise. For complex installations there is only one viable approach: a carefully planned and executed custom patchfield using the methods discussed.

Another alternative patchbay system is the Krone system. This consists of pre-wired patch components terminated to solderless IDC (insulation displacement connection) connection blocks. This type of system is usually only used in larger installations, but can offer all the flexibility of fully soldered systems, without the use of soldering. As such, a large system can be configured and indeed modified very quickly. The downsides are the use of special multicore cable for the IDC connections, and cost - although less labour intensive, the component costs are much higher than for 'raw' patch panels. For home or project studios, I would argue that the raw approach is much more cost effective; however, for extremely large installations Krone systems may be better due to the reduced labour costs involved.

Although the audio patchbay will be the most common in a small studio, patch panels are also available for other signal types. These will typically include various forms of video, digital audio and data. The video types and unbalanced digital audio will typically use MUSA or twist-and-pull BNC connectors, while balanced digital audio will mainly use XLR connectors. Data jackfields may use various types, such as D-Sub connectors or eight-pole RJ45, the latter being similar to the connectors found on telephone handsets. Due to the wide signal bandwidths used, MUSA jackfields are normalled using dedicated screened metal 'U' links. BNC fields frequently use very short BNC-to-BNC links

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A patchbay can make your studio much quicker to use, but can also be a nightmare to set up, so here's advice on which to choose and how to connect it up for the best results. Plus, tips on building and fitting racks for your gear.