Dec 7, 2022 - Michael Hahn

EQ is a type of audio processor that changes the tonal balance of a signal by boosting or attenuating the relative energy of different ranges of frequencies.

If that sounds complicated, don’t worry. If you’ve ever adjusted the treble, middle and bass knobs on your amp, you already have an intuitive feel for EQ.

Equalizers can come in the form of DAW plugins, studio hardware or guitar pedals like the Empress ParaEq MKII

EQ processors rely on audio filters with different shapes and behaviors. For example, a low-pass filter (LPF) removes high frequencies from the signal while letting low frequencies pass un-altered, and a high-pass filter (HPF) does the opposite.

A typical EQ device uses multiple filters to allow better control of the resulting sound.

In addition to the new variable LPF and HPF controls, the ParaEq MKII features three bell filter bands for comprehensive tone shaping.

There are different styles of EQ for different applications in audio.

For example, you might have seen the kind with banks of multiple slider controls in live sound systems or guitar pedals. This type of EQ is called a graphic equalizer. This is the same EQ type found in the popular Boss GE-7 or MXR 10 Band EQ pedals.

In these designs, each slider controls a narrow range around one specific frequency.

These narrow bands are used to notch out frequencies that contribute to feedback from the performers’ microphones on stage.

Guitar pedal designers adopted this style of EQ since it’s more immediate if you’re just getting started with equalizers.

However, graphic EQ comes with a significant drawback. To boost or cut a wider range of frequencies, you’ll need to adjust several sliders at once.

Since each slider controls a separate EQ band, adjusting several at a time creates peaks and valleys where the frequencies overlap. There’s also more active electronics involved in graphic EQ circuits which results in more coloration of the original guitar tone.

This is the type of EQ you’ll find in the Empress ParaEq MKII. It works by adjusting three key parameters for each band—frequency, Q and gain.

The frequency control selects the area where the boost or cut will take place, and the gain control determines how strong its effect will be. Increasing the gain of an EQ band will create a boost while attenuating it will result in a cut.

Q stands for “quality factor,” but you can think of it like the bandwidth of the filter. A wider Q will affect a broader range of frequencies around the target, while a narrower Q will perform more like a slider on a graphic EQ.

The Empress ParaEq MKII gives you three Q options to choose from for broad sculpting or surgical EQing, while the Deluxe allows for variable Q adjustment with a potentiometer.

With the basics out of the way, I’ll break down every important frequency range guitarists need to know to use EQ.

In the end, every guitar signal is different. These recommendations are just general guidelines to get you started with EQ.

Even so, knowing them will help you develop your ear and get used to balancing your sound with parametric EQ.

For guitars tuned in standard tuning, the fundamental frequency of the E string sits at 83 Hz.

Although the fundamental frequency is 83Hz, the guitar signal for the low E is made up of many harmonics above it that create the sound we know and love.

So for a low E note, you’d have all the multiples of the 83Hz which combine at different levels to create the tone of the guitar (83Hz, 166Hz, 249Hz, 332Hz, etc.)

That may make it seem like this low range is essential to make sure your instrument is heard.

Surprisingly, too much deep bass can cause issues in a full band mix. In a typical ensemble with bass and drums, the kick drum and bass guitar are especially active in this area. Adding low frequencies from your guitar can cause muddiness and lack of clarity.

Try cutting low bass with a shelving filter or high-pass filter to make sure it doesn’t interfere with the bass guitar or kick drum.

The low midrange is where the guitar starts to come into its own in a mix.

As you move up from the low bass, there will be less competition with the rhythm section, allowing you to take up more space with your guitar.

Most mix engineers associate this area with the weight and body of rhythm guitar tracks.

Try boosting this range to emphasize percussive techniques like palm muting or power chords.

Most players agree that the midrange is the most important frequency zone for guitar.

But this area is a broad range that begins at 300Hz and extends up to 5kHz. It’s worth dividing into a few specific areas since they have distinct effects.

For example, boosting midrange frequencies around 300-800Hz adds a sense of fatness that many associate with vintage guitar tones.

Conversely, some modern heavy genres call for a scooped midrange sound.

This is the classic “smiley face” EQ that you might have seen used on amps with graphic EQ like the Mesa Mark series.

You can accomplish the same effect by cutting this frequency range with a broad Q filter on the ParaEq MKII.

Scooped midrange tones are known for their tightness and bite in the high gain world, but this type of EQ is equally useful for clean playing.

Typical black panel Fender-style amps have a naturally scooped midrange that many players associate with “glassy” clean tones.

If your amp leans more mid-forward, you can use an EQ pedal to restore some of this glassiness by attenuating this range.

When it comes to standing out in the mix, the upper midrange frequencies play an important role.

For example, TS-style overdrives have a pronounced midrange boost at around 800Hz and Klon-style pedals have a noticeable bump at 1kHz.

It’s this quality that makes these pedal types so popular for punching through the mix.

You can achieve the same effect much more transparently by using an EQ pedal to naturally emphasize this area in your guitar signal.

Try boosts in the area between 800Hz and 2.5kHz to push your guitar forward in the mix for lead playing and solos.

This area is especially critical since it’s where the majority of detail occurs in human speech.

Since your hearing is naturally sensitive here, boosting these frequencies can cause harshness and fatigue if you go overboard.

However it’s a key frequency range for articulation and note definition when it comes to guitar EQ.

Try boosts in this area to increase definition or cuts if the sound feels too harsh and fatiguing.

While all instruments contain sound energy across the frequency spectrum, most have a range where the majority is concentrated.

As I mentioned above, this is usually the midrange when it comes to electric guitar.

Even so, the upper end of the treble range contains frequencies that do contribute to the sound of the guitar. However, these aren’t nearly as critical as they are for instruments like vocals or cymbals.

In fact, many mix engineers prefer to use a low-pass filter to make sure guitar tracks stay well contained in the midrange.

You may also find this helpful for removing high frequency noise and hiss that creeps in above the guitar signal with some signal chains.

Try setting a low-pass filter at 10kHz to see if this approach works for you.

Nov 30th, 2022 - Michael Hahn

Graphic EQ and Parametric EQ are two equalizer layouts that offer different methods of frequency control. Graphic EQ uses many narrow adjacent EQ bands to target specific frequencies, while parametric EQ allows the user to change the width of each band for broader adjustment.

Graphic EQ is typically used to notch out specific frequencies that are contributing to feedback in a live performance sound reinforcement system. Each band is represented by a slider that can be pushed up or down to boost or cut that frequency.

Parametric EQ is widely used in recording and mixing to change the character of a sound or emphasize certain qualities. Instead of sliders, parametric EQ uses traditional rotary knobs and switches to control several parameters that shape the EQ band. As I mentioned above, the most common ones are:

• Filter type (HPF, LPF, BPF, shelf, bell, notch)
• Q (bandwidth or ‘quality factor’)
• Frequency
• Gain

Some users find graphic EQ easier to learn at first since adjusting multiple bands creates a visual representation of the overall curve. This is the reason for the name ‘Graphic EQ.’

However, parametric EQ is much more flexible once you get used to it. On top of that, it provides specific advantages for guitar players.

Parametric EQ gives you access to a much wider range of shapes and curves using far fewer bands than graphic EQ.

For example, consider a shelving filter like the type you’ll find on the Empress ParaEQ MK II Deluxe.

Shelving filters are a basic EQ type for boosting or attenuating material above or below a corner frequency. They’re typically used to broadly adjust the highs or lows to taste. You can think of them like the treble and bass controls on a car stereo.

To approximate a shelving filter with a graphic EQ, you’d have to adjust every single band on the other side of the corner frequency.

That means that on a typical 31-band graphic EQ, you’d need to use five or more bands to approximate a shelving boost at 5 kHz!

Each slider has it’s own narrow range of action, so instead of a gentle sloping curve to a flat shelf, the overlapping edges of each band create noticeable peaks and troughs.

That said, graphic EQ can still be useful for many applications.

The narrow Q of each individual band makes them extremely effective for removing specific problematic frequencies.

Some players are known for using a graphic EQ pedal in their amp’s effects loop to tailor its preamp section to the power amp and speakers.

It’s effective since some speaker types have pronounced peaks in the midrange that stick out uncomfortably with certain rigs.

A good graphic EQ can reduce individual peaks for a smoother sound and a better match between guitar, amp and speaker.

There are also some sought-after graphic EQs with desirable character for guitar tone.

For example, the Mesa Mark series is known for its powerful, post-preamp graphic EQ that helped create the biting, mid-scooped tones of early metal.

For most other situations, parametric EQ is the better approach.

Gentle tone shaping, corrective sculpting and selective boosting are all best achieved with a good, transparent parametric EQ.

Here’s a few guitar tone situations where parametric EQ shines.

Too much bass is a recipe for muddiness, even if your amp isn’t running on the edge of overdrive.

Depending on your amp, guitar, speakers and pickups, your tone stack alone may not be enough to clear up the low end mud.

A precise low cut using a a high pass filter or a broad reduction with a low shelving filter can clear things up.

On the opposite end of the spectrum, some gear combinations can be painfully shrill and trebly.

The culprit is often a peaky range of frequencies in the upper mids where the human ear is most sensitive.

A good parametric EQ will allow you to find the offending range and attenuate it without impacting the rest of your sound.

Here's Aaron Marshall from Intervals to teach you about using the High Pass Filter and Low Pass Filter on the ParaEq MKII to remove the lows and highs that are muddying up your tone.

Plenty of players love the mid-forward character of classic TS and K-style overdrives.

But when used to push an amp into saturation, it’s common to turn their gain control fully counterclockwise so the amp’s own overdrive takes center stage.

With a parametric EQ you can dial in the ideal mid-boost frequencies without any solid state clipping from the overdrive pedal.

The Para EQ MKII’s boost feature has considerably more level on tap than a typical overdrive, making it the perfect tool for hitting your preamp tubes hard.

Here Aaron shows how the ParaEq MKII can be used to mimick an overdrive pedal.

5th August, 2022 - John Peippo

How many times have you done something and needed to reread the manual months later? Why not copy those instructions right onto said thing-a-majig? Check out our customer service Echosystem where we've stuck the looper manual onto the front just so we always know how to use it.

How do you not lose your much needed tools? Build them a home! Empress is no stranger to tool gremlins. That's why we love using colourful tape, foam, and other organizational devices. And for routinely used items, make sure to build their home in open sight and not in a hidden location. Below are the foam tool caddies we made for the customer service and repair office and for the mini pedals production cell.

We’re pretty techy here, and I’m sure you are too! So for Empress, using keyboard shortcuts is probably one of our number one time savers. Our design team uses them constantly to blast through computer work, and on the production floor, each builder uses them to prompt all of the build commands. We have a walkin' bass button and a smashin' drums button, do you?

How do you get work done without losing your mind? Use a checklist, of course! Breaking down an entire task into segments can ease your mind. Plus no need to remember things when everything is all laid out for you. As examples, here's our Buffer testing steps which go along with a helper ZOIA to ensure each pedal meets the same exact standard. Next is the start of day and end of day lists builders follow to make sure the cell is prepared for pedal building or is ready for the next builder. And the last image, is of our very flashy enclosure box reminder; turn on the light for which pedal needs more enclosures, and then you can replenish with ease!

Not everyone has access to 3D printers yet, but we can't talk about life hacks without bringing them up. Do you need a specific tool? Well you can make it! At Empress you can’t walk a foot without spotting something that was 3D printed. From tools, to holders, to even full-on pedal prototypes, it can all be printed. We’ll even print specific production parts such as the ZOIA’s SD card retainer or the pressfits for the Reverb’s lightpipes. Below are our 3D printers that are running almost constantly (hidden to the right is the back of the ZOIA production cell in case you wanted to know where ZOIAs are born!). Next is our Euroburo calibrator/tester, which is probably our largest chunk of 3D plastic. You can see it in action in our Euroburo Build Video. And lastly is one of the 3D printed jigs we use to hold the PCBs nice and secure during assembly. This one is for our Heavy pedal, but almost all of our pedals have their own specific jig made just for them.

That’s all for now!

If you enjoyed the little tour, please let us know at media@empresseffects.com and we would be delighted to make more content like this to show you our little home up in Canada!

June 17, 2022 - Michael Hahn

Steve: "Here's a common disheartening situation that I think most musicians relate to. At the jam space and onstage, your rig sounds excellent. However, moving into the studio - where every nuance can be heard - it's a different story. Instead of hearing the beautiful nuance of your playing, your bandmate's ears are flooded with hisses and hums. There are two reasons why this happens.

Number one: there are a lot of noisy pedals out there! Many of them are classics that you can't live without. Why are they so noisy? The plain answer is because it's expensive and tiresome to reduce hiss, hum, crackle - and whatever else - to the threshold of human hearing. Low-noise design frequently requires costly parts. We often spend more time working on reducing the noise in a circuit than working on the fun parts of the circuit that make the crazy sounds.

Number two: a pedal chain can only be as quiet as its most noisy pedal. You can have ten noiseless pedals in a row, but if you just put one noisy pedal in the chain and it's engaged, the whole pedal chain will be noisy. It's sometimes possible to hide the noise with a noise gate or reduce it with a better power supply. But these are both uphill battles. One pedal can ruin your pedal chain's picnic. Pedal chain picnics are a thing.

So, why is it essential for our pedals to be low-noise? We don't want your Empress pedals to be the ones ruining the chain for all the other pedals. "

Steve: "We wanted to create groundbreaking effects from the beginning, but we didn't want our customers to choose between interesting sounds and high-fidelity.

Here's an example: The Superdelay was our first experience designing a pedal around a powerful DSP chip. It took years of work and 11 board revisions before we got the noise floor low enough to meet our standards.

There were times we didn't think the Superdelay was going to happen. Empress was still getting off the ground at the time, and we were carrying a lot of risk.

But the Superdelay took off and became our best-selling pedal for years. I still see it on many professional boards even though it's been discontinued for over five years.

We've continued to adhere to our commitment to noise performance. At this point, it's an obsession."

Jay: "Each pedal requires a different approach, but I'll give an example. We took extreme measures to make the Echosystem and Reverb as quiet as possible.

I'll go through the three most important measures.

The power of VCAs—more headroom, less noise

Both pedals rely on twelve VCAs to mix the wet and dry signals for the left and right sides (we use the amazing V2164 by SSI).

VCAs don't have discrete steps like a digital pot—the on and off is nice and smooth, and there's no digital zipper noise as you adjust the volume.

But even with this approach, we had to triple the signal path to get a noise level that was acceptable to us.

The additional VCAs produce a signal that's three times as loud. Luckily, the noise doesn't triple since it's random and partially canceled out between the two sides.

In the end, if the signal gets louder and the noise stays the same, you get a higher signal-to-noise ratio. That's the main goal at the end of the day.

Of course, performance like this comes at a high cost. Triple the VCAs means triple the supporting circuitry. Going above and beyond like this is a significant factor in the final cost of a pedal.

Input pads and digital gain

But that's not the only way we can increase the signal-to-noise ratio. Our selectable input pad feature also plays a role. Since the pedal's internal noise is fixed, a higher output guitar will naturally give you better noise performance.

With the selectable input pad, users with quieter guitars can tweak this setting to squeeze more performance out of their rig.

Additionally, we use some digital tricks to get another boost in signal strength during the AD/DA process.

Some might consider this controversial, but we increase the gain in the digital domain since DACs are slightly noisier than ADCs.

Most guitar signals don't approach the limit of digital headroom, but for those that do, we use a limiter to prevent distortion. It's incredibly transparent and only acts on peaks that are just a few samples long. In eight years of production, no one has ever noticed that it's running.

The extra gain allows the DAC to output a hotter signal, further increasing the SNR.

Power supplies, transformers, and proprietary secrets

On the analog side, our power supplies have an extra passive filter at the output, and we use the quietest regulators we can find. As I mentioned before, no expense was spared in designing these pedals."

Steve: "Stuff like this is why they're not cheap, and most folks probably have no idea what goes into it!

There are some proprietary techniques that we can't fully reveal. For example, we managed to run our VCAs outside the manufacturer's spec to achieve lower noise. It's a bit complicated, so I won't get into it here!"

Last but not least, the transformer-isolated output on the Reverb and Echosystem breaks ground loops when using two amps in stereo. We treat it like a throwaway feature, but a dedicated box would cost over a hundred ClamCoins, a cryptocurrency we know nothing about."

The Blackfin board used on the Echosystem and Reverb. The big transformer on the top right is for breaking ground loops.

Jay: "You'll need to get serious about the rest of your chain to keep noise low.

Just turning on a Reverb or Echosystem isn't going to decrease the noise that comes from elsewhere on your pedalboard.

The standard good practices apply—we covered most of them in a recent post on the Empress Blog.

In short, though, my advice is to use quality gear. A good power supply and a buffer first in the chain go a long way.

Noise has the potential to increase with every device you add between the guitar and the amp, so be realistic about the kind of noise floor you'll get with a massive rig.

That said, Empress' effects are over-engineered to perform in this area."

Steve: "In short, the gain-staging and treatment of your guitar signal have been carefully considered all along the signal path in every Empress design.

We're literally spending triple the money in spots to squeeze another 3dB-4dB of noisefloor.

We recognize that noise performance is an essential feature of professional quality audio tools. Your pedalboard is no different.

That's the essence of our commitment to good design."

June 10, 2022 - Michael Hahn

Using a good buffer at the start of your signal chain is standard advice for any pedalboard these days.

The reason has to do with the electrical qualities of the signal produced by your guitar’s pickups. Since a traditional pickup is a passive device, its output signal has a naturally high output impedance.

This type of signal can easily lose top end and sparkle as it passes through a long cable or a chain of connected devices. The problem is especially bad with an entire chain of true bypass pedals since they behave like a long length of cable when the effect is turned off.

Buffers fix it by transparently changing the high impedance pickup output to a nice, strong low impedance signal. Of course, you can do this with any always-on pedal in your rig, but a well-designed buffer is purpose-built for the job and can provide better results.

Some buffers like the Empress Buffer+ include additional noise filtering to help reduce hiss from your guitar’s pickup for the rest of the pedals in line.  

Simply converting to a high impedance signal isn’t a silver bullet for noise issues, but it will ensure that your tone stays intact throughout your signal chain.

Powering your pedals correctly is essential for the best tone and noise performance.

For starters, you’ll need to match each pedal’s voltage and current requirements with the correct power supply to get them up and running.

If you’re just getting started with guitar pedals, you might be using a “daisy chain” to power multiple pedals from the same outlet.

In theory, this works fine if all your pedals require the same voltage and their total current doesn’t exceed the maximum available from your wall wart.

However, not all pedals play well together when they share the same power supply in the real world. You may find digital pedals like reverbs, delays, and DSP amps interfere with old-school analog designs and cause distracting noise.

You might have pedals with different requirements that can’t easily be powered with a one-size-fits-all solution.

Modern isolated power supplies deliver clean, accurate power to your effects. They’re the key ingredient that turns a collection of pedals into a professional sound design tool.

If one pedal in your chain is wreaking havoc on the rest, you may have to isolate it completely.

The easiest way to do it is with a true bypass looper.

In the past, pedal designers didn’t always account for how their stompboxes would perform on a board full of effects. Quality power filtering and transparent bypass weren’t key concerns the way they are today.

In many cases, vintage pedals can color your tone when bypassed and even introduce noise or bleed from their effect.

True bypass loopers are a cheap and effective fix for this issue. They’re simple passive devices that use proper 3PDT switching to completely remove any pedals inserted in the send/return loop from the signal path.

Place any offending pedals inside a true bypass loop to fully isolate their electronics when not in use.

If the noise in your signal chain comes from high gain distortion pedals, you’ll need to take a more active approach to noise reduction.

In these cases, the noise is a side effect of the amplification process that adds saturation to your signal. Since it’s baked in, you won’t be able to reduce it much with any of the mentioned methods.

Instead, many heavy players use a noise gate to mute their signal when they stop playing so that the background hiss doesn’t come through.

These pedals work by setting a threshold for when the gate will clamp down on the noise. The trick is to set the sensitivity so that your playing opens and closes the gate without chopping off notes or articulations.

The Empress Heavy has a built-in noise gate to keep your sound tight and noise-free even with aggressive high gain.

A ground loop is an electrical issue that causes a specific noise problem in your signal. For example, a ground loop produces a nasty hum that are far louder than the typical hiss and buzz of a basic guitar chain.

Ground loops can occur whenever your rig is plugged into multiple electrical outlets. For example, you might encounter them if you’re using a multi-amp setup or if your amp is located far away from your pedals on stage.

You’ll need a device that electrically isolates the devices from each other to fix it. Typically, this is done with the same transformer-based circuitry you’d find in a DI box.

The Empress Reverb and Echosystem both include transformer isolation on their stereo outputs to let you lift ground loops without needing an additional line isolator. The silver box in the top right corner of the image below is the output tranformer from the Reverb and Echosystem.

Finally, if nothing else seems to work, you might consider addressing the noise at the source—your pickups.

After all, magnetic guitar pickups are a 1940s-era technology. However, if you’re open to trying a modern design, plenty of options are built specifically to address noise.

Remember that the single-coil pickups found in most Fender-style guitars have a naturally higher noise floor than dual-coil designs like the humbucker or Filtertron.

The dual coil humbucker was the original noise-suppressing pickup made literally to “buck” the hum.

That said, if you need the authentic sound of a single-coil, you can consider any of the noiseless designs that have been introduced over the years to combat the problem.

If you’re not concerned with recreating the sounds of old-school passive pickups, you could also consider an active design for an even better signal-to-noise ratio. But, again, there are plenty of options if you’re willing to experiment!

19th May, 2022 - John Peippo

Just What is a Sample and Hold?

It may not be evident at first, but what a sample and hold really does is suspend time. It's a time capsule, holding onto the past while the world moves on around it. And it can seem alien at first because we often think of time as fluid and continuous. So to understand sample and holds, I want to use the analogy of a camera -- another type of time capsule.
If we take pictures fast, like thirty frames per second, we have a movie. If we slow down how often the pictures are taken, what looks like continuous motion begins to feel more herky-jerky. Once-fluid actions become periodic motion; a person starts getting up from a chair, and the next thing you know, they are halfway out the door. A sample and hold is like a camera, too: its input is the lens, taking in whatever it is pointed out; its trigger is the shutter, determining when to snap a shot of that view; and the output is the photograph, our moment frozen in time -- until the next photo.

Getting started using a sample and hold to make a Staircase waveform

We can take our "photos" (sample our inputs) really fast, or really slow, or really herky-jerky, too. To illustrate this, let's patch up that staircase waveform.

To get started, we'll need three things:

1. A sawtooth LFO.

2. A square LFO.

3. A sample and hold.

Since we want to freeze our sawtooth -- take its photograph at various times -- its output will be connected to the sample and hold's input. We're going to use the square to work as an automatic shutter, timing when those pictures are taken at regular periods, so we will connect its output to the sample and hold's trigger input. Each time the square rises -- sample and hold trigger (sometimes called clock) inputs look for the rising edge of a pulse or other CV signal -- it will trigger the sample and hold, and the current position of the sawtooth (the sample) will be frozen (the hold) at the sample and hold's output -- our collection of pictures that will add up to a staircase LFO.

If you just loaded the modules in their default condition and connected them, you're probably looking at that output and thinking: something's wrong, I don't see a staircase at all, just a straight line.

That is because using a sample and hold is all about timing. Since the sawtooth LFO and the square LFO share the same default rate, we're taking our pictures of the sawtooth LFO at the same position each time.

If we speed up our square LFO, we start to take pictures at a faster rate than the sawtooth LFO cycles, so the sample and hold's output begins to change. At first, though, it may not be very clear what shape we're sampling -- if the square wave LFO is only a little faster than the sawtooth, the samples may seem to jump around as they catch the sawtooth in a pattern that isn't immediately obvious. That's jbecause one way to think about the speed of our square is as a fidelity control. Right now, things are pretty lo-fi, but as we increase the speed of the square, the sawtooth begins to take form. When it is about four times as fast as the sawtooth, we should really begin to see the staircase form. As we increase the speed more, our staircase has even more steps to descend. And then, a funny thing begins to happen. As our fidelity increases, there are more and more steps, and the steps are smaller and smaller. Our sample and hold's output is looking a lot like a sawtooth wave again, maybe a little bumpier than usual, but definitely more sawtooth than staircase.

Of course, you can apply this technique to more than just sawtooth waves. Make your envelopes chunky! Turn an envelope follower into staircase follower! You can even modulate the sampling rate -- in our example, the rate of the square LFO -- to switch between smoother and more steppy forms, or you can use an irregular timing source to produce more unexpected results.

If this sort of waveform manipulation interests you, you might also look at the steps module in ZOIA, which takes continuous CV sources and breaks them up into discrete, even steps. A very similar process, but achieved by slightly different means.

5 More Create Ways to Use Sample and Holds

There are countless applications for sample and holds, but let's talk about a few more before I leave you with some resources.

Sci-fi Effects

Probably one of the most well-known uses of sample and holds is to create the beeps and the boops of classic sci-fi robots and alien technology. In this case, the sample input is usually white noise because it provides a source of random frequencies. When you clock the sample and hold, the output is a sequence of steadily changing voltages.

Hooked up to an oscillator's frequency input or a filter's cutoff frequency (or used with a ring modulator, or a delay line, or... the list goes on), you can quickly discover a whole world of classic sound effects and some new ones, too (like in the staircase LFO example, try modulating the clock speed!).

You can do the same thing in ZOIA by connecting a random module to the input of a sample and hold. But the random module itself has an option to produce a new random value only when it's triggered -- the same effect with fewer modules!

Generative Melodies

It's not such a great leap to get from a string of random frequencies to a generated melody. Really, we just take the example above, constrain its range with some attenuation (you might want unexpected notes, but you don't want them all over the place), and then quantize the results.

You can get more elaborate, too, using shift registers (which can be constructed in ZOIA using sample and holds strung together) or CV loopers or logic to make the randomly generated melody feel more purposeful and focused.

Delayed Trigger

Because the output of a sample and hold won't change until it receives a trigger, they can be a great way to get your patches' ducks in a row, so to speak. For instance, you could use the clock from a sequencer to change several other parameters at the beginning of each step; just use it to clock the sample and holds placed between a modulation source and a destination.

Or you could use the output of a clock divider to have those changes happen in some ratio of the sequencer's clock, like adding a "strobing" effect to filter frequency changes by using a sine wave as the sample and hold's input and multiplication of the clock to trigger the sample and hold!

Sample Rate Reduction

You might remember that I mentioned earlier how the clock rate of a sample and hold acted as a sort of fidelity control. We can take this even further in the audio realm by using an audio rate clock -- like a square wave VCO -- and sending audio to the sample and hold's input. The frequency of the clock will determine the sampling rate of the audio, so you can really use this to add some dirt and crud to a single. It's loads of fun! Your sample and hold module needs to be AC-coupled to accept audio signals, but a lot are.

ZOIA's sample and hold doesn't accept audio signals, but that's what the aliaser module is for! It can perform the same kinds of sample rate reduction. You can even get sample rate reduction by modulating a VCA with a super-fast LFO (try using a clock divider to push the LFO speeds into low audio ranges).

Freeze Modulation in Its Tracks

A close cousin of the sample and hold is the track and hold. Unlike the sample and hold, which won't change its output until it receives a new trigger, the track and hold will allow the input of the sample and hold to pass to its output until it receives a gate -- a sustained voltage -- at its trigger input. Then, it will hold onto the voltage that was at its input when it received the gate, just like a sample and hold, until the gate ends, at which point the input will be passed to the output again.

You can use track and holds -- and the sample and hold module in ZOIA has a track and hold option -- to stop LFOs and other modulation sources in mid-cycle, essentially freezing them in place until you release the gate. It's a lot of fun and an interesting alternative to the staircase example we started with!

If you want to learn even more about all the things you can do with sample and holds, I really recommend DivKid's video:

Before we go any further, let's define these terms a little. They're more slippery than they first appear. (Ask Marc Doty, whose epic series on polyphony is linked at the bottom of this article.) We can get a basic understanding if we put on our philologist (word-knower) hats. From the Greek, mono means one; poly means many. Phony means voice. So, polyphony is many voices, while monophony is just one voice. A choir or a soloist.

But what is a "voice" when discussing synthesizers? Attempting to define a voice is where things get slippery. Some people describe the difference between monophonic and polyphonic synths as the ability to play chords. Except, many monophonic synths have multiple oscillators, which can be tuned to the intervals of a chord. Then, they might clarify that it's about being able to play different notes on a keyboard simultaneously.

However, if we look at a eurorack module like Mutable Instruments' Rings, it can play multiple notes at once, just not at the same time. So is it not polyphonic? You can see how this seemingly simple question has a lot of complicated answers.

While debates about those complicated answers will continue to rage on despite our best efforts, I'd like to present the difference between monophonic and polyphonic in the way that I have come to think of it. And that is in terms of expression. To make this even more precise, let's do away with notes and chords for a moment and speak only of sound (after all, some synthesizers strive for the atonal).

A monophonic synth allows you to express one sound at a time, while a polyphonic synth can will enable you to express more than one sound simultaneously.

Suppose we return to our example of the mighty multi-oscillator synth, one not so different in my imagination from the vaunted Moog Model D. In that case, we see that while it can form chords from its arrangement of oscillators, it can only express one chord at a time. Resoundingly monophonic. Meanwhile, the Mutable Instruments' Rings is restored to an untroubled position in the pantheon of polyphonic instruments. While only one of its voices can be triggered at a time, those voices can overlap and express different sounds simultaneously.

We return to our conundrum with definitions in place: do we choose a monophonic synth or a polyphonic one? At first glance, it might appear the choice is simple. More is always better, right?

Context is critical here. Polyphonic synths and monophonic synths can serve different purposes, and it is often the case that the limitations of one are advantages of the other. I used the example of a choir and a soloist before. The choir can make beautiful harmonies, but it can't take the lead like a soloist can. So, while polyphonic synths excel at pads that fill the sonic spectrum, they can be less suited to leads that cut through that space.

I used to play bass in an R&B band, and figuring out where I sat in the mix with our keyboardist was a constant struggle. He liked to play a lot of chords with an anchor in the low end, and the bass would get lost at times. I would counter by adding fuzz or a synth pedal, but the added harmonics could crowd out our guitarist. (Our singer, meanwhile, continued to do what he did, blissfully unaware of the ongoing war for the mid-low frequencies. Ah, the life of a singer.) So my point is that there are times when filling space is useful. But there are also times when it is decidedly not.

The style of play also matters. The limitation of one voice becomes an advantage when you consider things like legato passages, where you want to smoothly glide from note to note. The same can be true of staccato passages, where too many voices at once can sound busy and confused.

More is often better, but there's a reason I'm perfectly happy with two legs instead of three. So, it really does come down to context. Every lock has a key, and sometimes a polyphonic key is the wrong fit for that particular musical lock.

Music has a lot of locks, so to speak, so sometimes the answer is a skeleton key, like ZOIA. A ZOIA patch can be a monophonic synth, a polyphonic synth, or even a combination.

Over MIDI, ZOIA can be a versatile polysynth. For example, the MIDI note in module supports up to eight notes at a time, each with its own pitch, gate, and, optionally, velocity output. However, it should be noted that while it is possible to make a patch that utilizes all eight of these outputs, there is often a compromise between the number of voices and their complexity.

If you want to use FM, for instance, then the number of oscillators you need doubles (or more), or if you're going to employ ZOIA's effects, each of these additions may mean compromising on the number of voices at your disposal. On the other hand, if you are designing a monophonic patch, you may run out of ideas long before you run out of CPU.

When used as a guitar synth or employing the CV inputs of Euroburo to control a synth, it may seem like you are limited to monophony. After all, the pitch detection on ZOIA is monophonic; there are only so many CV inputs to utilize. But suppose you use triggers, rather than gates, to control the envelopes of your voices. In that case, you can trigger overlapping voices to play simultaneously, much like how Mutable Instruments' Rings employs polyphony.

Of course, ZOIA has many internal methods of producing sound, from sequencers to generative approaches, and these can also have monophonic or polyphonic applications. Moreover, these can be combined with the previous methods to create patches that are more than just monophonic or polyphonic; they are multi-timbral. (Multi-timbral is just synth-speak for producing two different types of sound at once, like clapping while you play the kazoo. Or, arguably, something more aurally pleasing than that.)

This isn't to say ZOIA is the best answer to our initial question. There are many options for synthesizers and synth pedals out there, and plenty of them do things that ZOIA can't. (I've opened up the enclosure before, and there is no fold-out keyboard tucked inside.) Variety is the spice of life. But it may be that ZOIA is an answer that you've overlooked and is more versatile than you first imagined.

7th April, 2022 - Michael Hahn

Mechanically, an expression pedal is a simple device. In most cases, expression pedals consist of a single passive potentiometer with a TRS output connection. Some designs offer multiple outputs, configurable ranges, and other features, but they all perform the same essential function.

In a typical setup, you connect the pedal's output to an expression control input on your effect. Different pedals offer various configurations for controlling their parameters, but I'll get into that below.

Why use an expression pedal?

Changing effect controls on the fly adds a new dimension to your playing.

After all, sound texture is a vital part of any musical arrangement. Emphasizing a passage with effects is a powerful trick for building intensity and atmosphere.

But controlling specific parameters can take it even further. For example, there are endless ways to create drama with a well-timed adjustment to your sound's timbre.

Not only that, some effects depend on manual control to create their signature sound.

For example, you're probably familiar with the most common expression-controlled pedal—wah. However, the vocal quality of the effect relies on manual sweeps of the filter that match your playing.

If you've ever wished you had a third hand to tweak your pedal settings as you play, expression pedals can give you the flexibility to play your effects like an instrument.

Five creative ways to use your expression pedal

Expression pedals can be set up to control just about any parameter you want. Depending on the capabilities of the pedal you connect them to, you may be able to choose from many different options.

I’ll break down five common ways to use expression pedals and explain the results they produce to give you some ideas to get started.

Wet/dry blend

The blend of wet and dry signal determines the overall strength of ambience effects like delay and reverb.

Varying the wet/dry balance with an expression pedal can create contrast between sections, or provide a textural effect at extreme settings.

Controlling wet/dry mix is as simple as moving the expression pedal to the heel position and setting the mix control fully counterclockwise. Hold the save button to assign the expression value to the heel and repeat the process with the mix control clockwise and the pedal in the toe position.

Now you can gradually fade in washes of ambience that obscure your dry signal completely at the highest setting.

Delay oscillation

Who doesn’t love the classic runaway delay oscillation effect? When it’s a great vintage-style delay, the unique distortion and wailing trails are always fun to play with.

When you adjust the delay time, the oscillation pitch changes as the signal feeds back into itself.

The Echosystem’s tape mode can be configured for expression control of the delay time parameter. With the feedback set high enough you can use the expression pedal to “play” the pitched oscillation of the delay, even as you play new notes into the feedback loop.

It’s a great way to get musical wails and unpredictable pitched feedback as you play.

Modulation speed and depth

Modulation is a broad category of effects pedals.

Despite the different styles, this effect type almost always relies on an LFO to control the speed and depth of its action.

LFO stands for low-frequency oscillator. This waveform is so slow that you don't perceive it as a musical tone. Instead, the LFOs frequency, depth, and waveshape define the character of a modulation effect's sweep.

Changing the rate or depth of the LFO sweep with an expression pedal has unique effects for different modulation types.

For example, setting an expression pedal to control the Empress Tremolo 2's rate lets you ramp up and down from a shaky stuttering effect to a slow pulsing tone.

Johnny Greenwood famously used this technique in the rhythm guitar track on "Bones" from The Bends.

Pitch shift

Another classic rocker-controlled effect is the dive-bomb pitch shift made famous by the Digitech Whammy.

The original pedal comes with a rocker treadle built-in, but it's large, heavy, and requires a high current power supply.

If you don't want to devote all that pedalboard real estate to a full-size whammy pedal, you can accomplish the same effect using an expression pedal and a compact pitch shifter like the EHX Pitch Fork.

Multiple parameters

Several of the best expression pedals feature additional outputs to control two or more parameters at once.

Sweeping the range of multiple controls creates a morphing effect that blurs the parameters together.

Also many digitally controlled pedals allow for control of multiple paramaters at once. The Empress Reverb and Echosystem offer comprehensive parameter control via expression pedal. In fact, you can blend between two positions on every knob as you rock the expression pedal from heel to toe.

The ZOIA is inspired by the flexibility of modular synthesis. It offers expression control of nearly every parameter in any combination and direction!

If you're looking to create evolving textures with hands-on control of a whole chain of processors, there's no better way than with ZOIA.

The control port on the Reverb, Echosystem and ZOIA can be used for

expression pedals, MIDI, an external switch, tap tempo, or CV.

 If you're in the market for an expression pedal, there are plenty of great options to suit any budget.

Here are six top picks to get you started.

Moog EP-3

The Moog EP-3 is a cost-effective and well-built expression pedal. It features a single output and comes with a 6" TRS cable if you don't have one on hand.

The EP-3 is among the most affordable choices on this list, but its construction is mostly plastic. Keep that in mind if you need a more roadworthy option but I've had one for years that has taken a beating and works flawlessly.

Mission Engineering EP-1

Mission Engineering takes expression pedals seriously.

They have a vast line of options that cater to the specific expression needs of popular modern gear.

The standard EP-1 model is a solid all-around choice in a familiar full-size wah style enclosure.

Boss FV-30

Boss's latest take on rocker-style pedals is a robust all-metal design that feels great underfoot without taking up too much space.

It features dual outputs for controlling two effects at once and range controls to set the boundaries of the sweep.

Source Audio SA161

Source Audio's SA161 is another great dual output expression pedal with solid build quality and multiple outputs for controlling two different parameters.

Dunlop DVP3

If you're looking for a pedal to work as a volume pedal and an expression pedal in a compact package, the Dunlop DVP3 is one to consider.

It offers a durable design with adjustable rocker tension for the perfect feel.

Any Volume Pedal

Speaking of double duty, any volume pedal can act as an expression pedal if it’s connected with the right type of cable.

The one you’ll need is a ¼” TRS to dual ¼” TS cable. These are sometimes called insert cables because they’re used to add pro audio gear like compressors to the signal flow of a mixer.

With the tip TS jack plugged into the volume pedal’s input and the ring TS jack in the output, your volume pedal will behave just like a TRS expression pedal.