+ SIGNAL PROCESSING

Powerful computers, DAW software and plug-ins have irrevocably altered the face of the audio landscape. Addicted to the convenience of automation, unlimited takes, performance restoration and infinite manipulation, few continue to splice tape. Yet curiously, in the wake of this revolution, analog processing enjoys a greater popularity than ever before. Analog EQs and compressors offer a multitude of functions, personality and flavors that plug-ins cannot deliver. Modern units are often built from the ‘bones’ of legacy products that were first designed in order solve specific problems, but go much farther by leveraging contemporary components and latter-day research.

+ EQUALIZATION

Equalization was developed as a solution for the telephone industry, who’s long carrier lines were unable to maintain high quality sound. In 1915, George Ashley Campbell working at AT&T patented the first Electric-Wave filter. By combining high-pass and low-pass filters he created a band-pass filter that could manipulate voices sent over telephone lines. These early equalizers were built directly into the circuits of the telephone transmission and receiving equipment. At the time, there was no forethought of ever building stand-alone equalizers.

In 1925 AT&T founded their research and development department, Bell Telephone Laboratories. That same year, Western Electric conducted the first electrical recording session using microphones to convert acoustic energy into voltage and feeding that voltage to the cutting lathe. Adding a circuit path on the way to the needle opened up the possibility of processing the audio signal. Soon, Bell Labs engineers Joseph P. Maxwell and Henry C. Harrison found a use for equalization in phonograph recordings. By 1926 they had already recognized that equalization would be a great way to counteract low-frequency over-modulation, which lead to overlapping grooves being cut into Western Electric discs. They devised a system where low frequencies below 250 Hz would be attenuated on their way to the recording head. A complimentary low frequency boost would be added to the circuits of a consumer’s player which would restore the low frequencies to their natural level during playback.

This concept is known as pre-emphasis and de-emphasis, and would eventually develop, seeing high frequencies also boosted during recording, and attenuated during playback. This became common in the record industry and before long, engineers were trying the same thing in sound movie theaters. In the 1930’s, John Volkmann, working for RCA, had created an independent equalization device which is recognized as the first operator-variable equalizer. His tool, featuring a set of selectable frequencies with boosts and cuts was designed to improve the sound of playback equipment in various movie theaters.

RCA, Bell Labs, Langevin, and Cinema Engineering would continue to develop standalone equalizers which would be used not only for reproduction in theaters, but throughout production and post production of films. These tools would become commonplace in music recording studios and radio broadcasting facilities as well, with notable contributions like the Pultec EQP-1 passive equalizer coming into the fold as early as 1951. The flavor of its active counterpart the EQP-1A is still one of the most popular program EQ’s today. In 1952 P.J. Baxandall published his paper “Negative-Feedback Tone Control” in which he introduced his circuit providing bass and treble increases and decreases using only potentiometers, and not switches, to shift from boosts to cuts. The smooth musical equalization curves Baxandall created would later be captured by the Dangerous Music BAX EQ.

This idea of different equalizers having their own unique flavors would define research and development in the late sixties and early seventies. In this era, recording consoles were custom-built for studios, each one catering to the desires of house engineers. Because of this, different studios each had their own unique sound. By 1967, Saul Walker introduced the API 550A equalizer, whose bandwidth is inherently altered relative to the amount of signal boosted. This EQ, like others of its time featured a fixed selection of frequencies, and variable boost or cut controls at those frequencies.

At the same time Bob Meushaw was developing an equalizer that was not limited to frequency presets, but instead offered infinitely sweepable frequencies within a certain range. His friend George Massenburg would further this idea, incorporating a fully “parametric” equalizer into a console designed for ITI Studios in Maryland. This design, though somewhat noisy, was the first to feature fully-sweepable frequency, gain, and dedicated bandwidth controls for equalization bands. Massenburg would continue to improve upon this design with his GML equalizers. Now, countless modular equalizers and console EQ’s feature different flavors and option based on these classic designs.

+ TIME-BASED PROCESSORS

By the late 50’s, and early 60’s, processors creating artificial delays, reverberation and other time-altering effects were making their way into recording studios. Many people are familiar with tape delays, in which sound would be recorded to a tape loop with one head, and played back with another shortly afterwards. Even before that, the first delay processors were being designed by Bell Labs in an attempt to recreate the echo incurred over long-distance telephone lines. Their defunct design, which saw audio vibrating through springs and picked up at the other end, caught the interest of tone-wheel organ manufacturer Laurens Hammond. In the late 1930‘s, he was trying to bring the sound of a cathedral pipe organ to the living rooms where his tone-wheel organs usually resided. Building off of the Bell Labs invention, he introduced the first spring reverb, a giant device, housed in the organ’s tone cabinet.

While spring reverbs became popular in organs and guitar amplifiers their sound failed to make an overwhelming impression on recording engineers. The more common practice in studios was to use echo chambers to add reverberation to a mix. Bill Putnam is known to be the proprietor of this technique, actually taking splits from his console to feed signal to speakers in echoey rooms, which would be mic’ed and returned to his console. Though this concept gained popularity in a good number of studios, but there were plenty that didn’t have the real estate to dedicate to those types of spaces.

A solution came in 1957, when the Germany’s EMT introduced an idea which played on the classic spring reverbs but achieved a much greater frequency response and superior overall sound. Instead of recording sound vibrations through springs, their EMT 140 reverb featured a thin sheet of metal suspended by springs inside of a wooden enclosure. Sound from a speaker would vibrate the metal plate, a microphone would record the vibrations, and a damping pad would allow control of the overall decay time. Plate reverbs were big, heavy, and required a quiet space in which to reside, but were still more practical than echo chambers in the eyes of many.

Like plate reverbs and tape delays, other effects have histories that predate familiar circuit-based versions, extending back to a time when they were purely mechanical processes. Flanging, for example is known to have been introduced by Phil Spector. Attempting to thicken a vocal by playing it from two tape recorders, he pressed his hand against the tape flange on the second machine to slightly offset the positions of the two decks. Due to the irregular pressure applied, the speed of the second machine wavered relative to the first creating a modulated phase shift. Studio tricks like these were common in the experimental recording era of the sixties, but eventually, cool, weird sounds became a regular expectation in mixes. The need for consistent access to these types of effects created a demand for electronic signal processors.

In 1969 Phillips engineers Sangster and Teer presented a solution to electronically delay audio signals by storing and passing the voltage through a series of capacitors. The timing of the voltage passage was electrically clocked at a speed which was operator-variable. This passage of voltage down the line conceptually mirrors a fire-fighting “bucket brigade” passing buckets of water from hand to hand, thus this was the name given to this type of circuit. The sound of the Bucket Brigade Device (BBD) was made famous by devices like the Electro-Harmonix Memory Man delay pedal, the Roland Dimension D stereo widener, and flangers like the MXR Flanger/Doubler and Mu-Tron Flanger. This first wave of purely circuit-based TBP’s were known for their organic character, which resulted from natural signal degradation throughout the BBD chain. In this same era, a new type of delay circuit that would be less colorful, and more accurate was already in the works.

When Dr. Francis Lee at MIT devised the first digital delay circuit in 1969, his plan was to use it for use in heart monitors and possibly even speech-learning tools. He formed Lexicon with engineer Chuck Bagnashi for that reason, but his teaching assistant, Barry Blesser, had the idea of running audio through the circuit. The successful experiment caught the attention of Gotham Audio who saw digital delay as a potential pre-delay circuit for the EMT plate reverbs that they were distributing in America. Gotham licensed the design, and in 1971, introduced the pro audio industry to the Delta T-101, the very first digital signal processor.

Lexicon would stay true to their original course offering a language-learning aide in the Varispeech Model 27Y, designed to correct pitch after a tape was slowed down, in order to study the words spoken. Used by itself, the device became popular in the music world providing digital pitch-shifting, a few years before the famous Eventide H910 harmonizer. Meanwhile, Barry Blesser left his MIT associates to work with EMT and created the sci-fi looking EMT250, the first production-model digital reverb, introduced in 1976. Lexicon would finally define their legacy as makers of some of the world’s finest digital reverbs with their 224 Reverb in 1978.

As more complicated processing algorithms were developed, and digital recording became available, the studio-in-a-computer days dawned in the early nineties. A program called Deck from OSC offered the first multi-track recording software, which was soon licensed to Digidesign, giving birth to Pro Tools. From there, the idea of “plugins” adding functionality to an existing program had already been happening in other places, like photo editing software. That said, when Waves introduced the Q10, the first audio DSP plugin for Pro Tools, the industry was forever changed. Eventually, any popular signal processing hardware device was rivaled by a software version. The whole thing spawned the endless debate: Do we still need hardware? Or can we mix entirely “in the box?”

+ MID/SIDE PROCESSING

An Introduction to Line-Level Mid/Side Processing

Dangerous Music’s chief circuit designer, Chris Muth, was the first person to design a device specifically for implementing Mid/Side processing for line-level audio signals, as well as the first to include insert loops for applying external signal processors like compressors and EQs. Those innovations were a vastly important step forward for mixing and especially mastering. The Muth legend continues in the Dangerous MASTER, a powerful and crystal clear mastering console complete with an all-analog Mid/Side Matrix, width control and insert loops.

In this tutorial, we’ll learn about Mid/Side processing techniques, using the MASTER’s Mid/Side configuration as our example.

Mid/Side processing, as the name suggests, allows the Middle and the Sides (L-R) of a stereo recording to be separated out from each other and manipulated independently. Sometimes Mid/Side processing is called “sum and difference” or “sum and minus,” but we’ll stick with the standard Mid/Side name.

We typically think of stereo as being made up of the left and the right – which is technically accurate – but it is equally valid to think of stereo audio as being made up of the Mid and the Sides that we perceive. It’s also worth keeping in mind that there is no exact line of demarcation between the Mid and the Sides – anyone who’s used a “pan” pot knows intuitively that the stereo field is a continuum between far left and far right. So, when we separate out the Mid and the Sides in Mid/Side processing, we end up with signals mostly in the middle and mostly on the sides. This fact doesn’t present any real-world issues when working in Mid/Side, but it may help you grasp better what’s happening as you experiment with it.

The creative possibilities that arise with Mid/Side processing are vast, and it’s considered one of the most powerful – and sometimes problematic – tools among audio engineers. The possibility of changing levels, tones and dynamics within a finished mix puts formidable power into the hands of the engineer, especially during mastering. We’re going to explore some of those possibilities, especially as it pertains to using the Dangerous MASTER, but before we do, let’s get some basics down.

Understanding the Mid/Side Matrix

Like most audio devices, you actually don’t need to understand how Mid/Side processing works in order to use it, but it’s helpful and interesting to have a basic sense of it, and of how the Dangerous MASTER’s circuits achieve the definitive Mid/Side processing in the analog domain. Many digital Mid/Side plugins are available, but they have yet to measure up sonically to the sound of Muth’s brilliant analog circuits and often introduce undesirable artifacts.

Mid/Side processing involves a relatively complicated manipulation of audio signals, and it can be used during tracking, mixing and mastering stereo recordings. We’ll leave tracking aside for this tutorial, as Mid/Side recording requires a specific microphone technique. (Note that many MASTER owners use its Matrix for Mid/Side recording). Instead, we will focus on how Mid/Side processing works in the manipulation of line-level audio during mixing and mastering.

A FEW QUICK DEFINITIONS

Mid – The middle of the stereo image. We perceive the Mid as a mono audio image in the middle of the stereo field, but is actually a “phantom image” created by the left and right speakers delivering the exact same source at the same volume and at the same time.

Sides – The Left and Right sides of the stereo image.

Matrix – Using a relatively simple formula, the Matrix is the processor that encodes the L/R signal into separate Mid and Side signals and then decodes the M/S signals back into L/R. (The MASTER uses an audiophile-grade, all-analog Matrix.)

Insert Loop – On the Dangerous MASTER, there are three insert loops, and one of them can utilize the Mid/Side matrix, meaning that you can use this insert to process the Mid and Sides of a stereo signal separately.

Width Control – Not found on all Mid/Side processors, this control is an essential part of the MASTER that allows the user to quickly dial in the relative level of the mid and sides, effectively changing the perceived width of the stereo image. Even when not using the Insert Loops, pressing the MASTER’s Mid/Side button engages the Matrix, allowing use of the Width Control.

This diagram shows the basic topology of the MASTER’s analog Mid/Side Matrix.

As you can see in the diagram above, engaging the Mid/Side Matrix gives you separate control of the middle image and the sides of a stereo signal. The Left mono channel of the outboard processor becomes the mono Mid channel and the Right mono channel becomes the Sides. Though a bit counterintuitive, it’s important to keep in mind that the Sides channel is a mono channel within the matrix. By using the external processing loops, you can enter into the powerful world of Mid/Side processing. Below we’ll share a number of examples. As with any audio processing, Mid/Side is only as limited as your imagination.

POSSIBILITIES FOR MASTERING WITH MID/SIDE

De-Ess the Vocal – By inserting a dedicated de-esser or other dynamic EQ unit on the Mid, one can control an ess-y vocal while leaving the sides alone. This is often desirable when there are problematic esses, but the sides require an open top-end that the de-esser might interfere with. Especially given the resurgence of vinyl, ess-y vocals can be a particularly nasty problem when cutting vinyl, while maintaining an open stereo image is essential.

Open Up The Wings – It’s not uncommon during mastering to use shelving EQs in the top end to “open up the mix.” (The Dangerous Music BAX EQ is a particularly musical tool for this.) By applying a high-shelf EQ to only the sides of the mix, elements that are commonly panned wide, such as cymbals and guitars, can pick pick up a great deal of high-end energy without causing excessive vocal sibilance, snare splashiness, or other undesirable results that the EQ might cause in the middle of the mix.

De-Ess and Open The Wings – It is not uncommon to use these two techniques in conjunction.

Rebuild The Middle – Another useful technique during mastering is simply turning the sides down on a mix where the engineer has “overpanned” and sucked the center out. This is why the Width Control on the MASTER goes in both directions, allowing you to narrow the mix and emphasize the Mid as needed.

Pump the Middle – Got a rhythm section down the center that’s begging for some hard compression, but you don’t want the sides to start breathing, or pumping, with the beat? Send the Mid into your favorite compressor and dial in edge and sustain up the middle without affecting the sonic elements on the sides. A powerful strategy for music with a heavy beat that needs to hit hard.

Match the Image – Sometimes there’s a track on an album that just doesn’t fit in due to its stereo image (compilation albums are notorious in this way). By using the width control on a unit like the Dangerous Music MASTER, you can dial in a stereo image that sits convincingly among the other tracks on the record.

Targeted EQ’ing – If you’ve got a lead vocal that needs some drastic EQ’ing, a kick drum that could use shaping, or a “snotty” snare that’s dominating the mix, simply use a parametric EQ to process those crucial elements in the middle while leaving the sides alone. Or, what if the middle sounds great, but there are some side elements that need individual attention – just patch an EQ channel to the Sides and go to work. Even more powerful, use individual channels of a stereo EQ to separately carve the Mid and Sides while maintaining consistent tonality from the EQ’s circuits across the stereo field.

Surreptitious Reverb Processing – Whether they openly admit it or not, many mastering engineers will use reverb to help get a record over the line. With Mid/Side processing, you can target the areas of the stereo image that you’d like to add reverb to, allowing you to use more in that specific area rather than bleeding it into the entire stereo image. Add a touch of extra space around a vocal or a solo horn in the center, for example, and let the sides remain unaffected. If you’re the type not to tell your clients you’ve added reverb, you’ll get away with even more!

Orchestral Sound Stage Manipulations – Many an audiophile will obsess over the realism of the soundstage in an orchestral recording, but what many of them don’t know is that those images are often derived quite artificially through the use of Mid/Side processing. Orchestral recording techniques often boil down to just a few simple microphones, and Mid/Side processing – even just control of the width – can add or subtract many yards/meters of perceived soundstage width and depth. Add in some EQ and a touch of additional reverb and you’ll feel like you’re controlling the architecture of the concert hall.

POSSIBILITIES FOR MIXING WITH MID/SIDE

Any of the above mastering techniques can be used during mixing. By running any stereo source through the Matrix, you’ve got control over the Mids and the Sides separately. Whether it’s a single stereo recording like a string quartet or a drum overhead pair or something more elaborate like a stereo subgroup of dense background a drum mix, Mid/Side opens up a whole world of possibilities for the creative mixer. Below are just a few possibilities to help get the creative ideas flowing.

Simple Width Control – With a device like the Dangerous Music MASTER’s straightforward width control on hand, adjusting the width of a stereo signal within a mix couldn’t be easier. It’s an effective way to help tracks either fit together, or distinguish themselves, within the bigger picture. Especially helpful when managing dense, layered mixes, where a bit of extra space can help.

Fake Stereo: Delay and Modulation – By “multing” (duplicating) a mono signal and routing it into both the left and right channels and sending it to the Mid/Side Matrix, one can do all kinds of interesting things. Put some short delay (start around 5 to 15ms) on one channel feeding the Matrix and you’ve suddenly got a whole world of slight differences between the Mid and Side content to play with. Modulate that delay with a chorus effect on the Sides, and you’ve got subtle movement in pitch and time with the center staying clear as a bell. This can create a “fake stereo” effect, adding depth and width to your mix. Very useful on pads and guitars.

Fake Stereo: Comb Filtering – Comb filtering occurs in real acoustic spaces when a signal and its reflection are arriving at the same point after having travelled different distances. The interaction of the two signals causes sharp peaks and dips across the frequency spectrum. This is called “comb filtering” because the peaks and valleys resemble the teeth on a comb. The human auditory processing systems use comb filtering to help “localize” a sound source in space. We can recreate comb filtering by choosing a predictable boost and cut pattern on the Mids and the exact opposite on the Sides, thus creating some very convincing artificial stereo effects. Add in a slight delay on one channel (start around 5ms and experiment from there), and the possibilities get quite interesting.

Rhythmic Delay Processing – In electronic music, many of the drum machines and older samples that have become so iconic are typically mono. It’s no secret that adding delay to your drum tracks has become a common production technique that wakes up those mono drums and converts them into wholly new beats. What’s less known is how to use Mid/Side processing to create these effects. For example, using the Dangerous Music MASTER’s Mid/Side send the sides of your drums out to a funky analog delay set to a subdivision of the song’s beat and the center will stay rock solid while the sides rock to an all new rhythm. Incredibly effective, and unique.

Drum Stereo Pair Work – Many simple stereo-pair drum recordings can leave you without options while mixing, and a common problem is splashy or harsh cymbals. EQ’ing the stereo-pair often leaves the center of the kit lackluster – the snare loses its cut, or the kick feels too muffled. By EQ’ing and compressing the Sides of the stereo pair, one can gain control over those cymbals while allowing the center to remain snappy and alive. Or, if you’ve got the opposite problem, EQ and/or compress the Mid while letting the Sides sizzle and crack. Mid/Side processing can make a stereo-pair function like a multi-mic recording.

Pitch-Shifting For Depth – Perfectly pitched recordings often lack character and depth (and are more common in today’s machine driven musical landscape). The magic of music is often found in the slight discrepancies of pitch and timbre. It’s been a long-known trick to shift the pitch between Left and Right order to create a sense of width – just tiny changes, a few cents to start and adjust to taste, sharp or flat, often some of each blended into each side. With Mid/Side you can apply this same technique and, rather than obtaining width, you can get front-to-back depth to emerge. Try this on “gang vocals” or a choir of background vocals, a horn section, or any string pads, and see if you can get a more 3D sonic image.

Reverb Sends – Experiment using the Sides as the stereo send to a reverb while leaving the center of the source dry. Subtle differences in how reverb is triggered by the Sides will often create a greater sense of space and interest around a vocal. Experiment with long pre-delay times in the reverb, and even try slightly modulating or pitch-shifting the send to the reverb for more miasmatic effects.

GO BE CREATIVE

These examples above are merely starting points for anyone to get a sense of what’s possible with Mid/Side processing. It’s an area of audio work that’s ripe for experimentation, and more often than not, Mid/Side processing leads to new discoveries. It’s an incredibly powerful tool, and when working in the analog domain, it unleashes all kinds of new possibilities for your investment pieces. Because of its audiophile-grade design and components, the Dangerous Music MASTER’s Matrix is one of the most sought-after Mid/Side processors available, and will deliver rock solid results, whether you’re simply trying to de-ess a vocal or creating a whole new sound.

+ THEN AND NOW

That conversation is still fresh today, and many would agree that whatever sounds the best is the right solution. That said, to one degree or another, most engineers are adopting a hybrid workflow, fusing the benefits of a DAW with the sound of their favorite hardware. Whether flangers, plate reverbs, limiters, analog EQ’s or full-on mixing consoles, the common belief is that passing voltage through electronics creates an organic sound that can never truly be expressed by a predictable mathematical formula. Dangerous Music is fully vested in creating solutions for this modern, hybrid studio.