Why Do Equalizers Sound Different? Part 2: Implementation

After last week’s toe-dipping into the theoretical design of equalizers, let’s get serious and look at how decisions regarding their real-life implementation affect the outcome.

Last week I was talking a bit about how conversations about specific qualities and non-qualities of equipment all too often end up to be a mess of abstract buzzwords. The problem with that is that these buzzwords like “open”, “musical” or “tight” have absolutely no properly agreed-upon meaning. Somehow it seems like talking about audio is quite hard.

On the other hand, there’s a widespread skepticism against the use of more technical, clearly defined properties such as frequency response, harmonic distortion and so on. The common belief is that these technical measurements cannot fully describe the sound of a piece of equipment.

That’s true. But isn’t it better than nothing? I leave that thought with you for a moment.

Anyway let’s not get derailed too much right in the beginning. Where were we?

We were looking at some of the decisions that a developer faces when building even a simple equalizer. Or more precisely, even before he or she can start to actually build it.

These basic decisions about the general concept and user interface matter infinitely more than the choice of exotic premium capacitor or the wordlength used in a DSP algorithm. They often make a world of a difference that is immediately noticed in practical use. And it’s easily measurable as well.

But the choice of technology to implement these concepts also has a huge impact on the result. The reason is that each technology brings its own limitations and constraints to the table. So let’s look at what these are for electronics and digital signal processing.

Analog Equalizers

Let’s presume we’ve put together a nice concept for an equalizer and choose to implement it using electronic components. As we learned last week, to create an analog filter we need components that have a “memory”, a knowledge about the recent past of the signal we are processing.

In electronics, we can get such behavior using capacitors and/or inductors. A capacitor stores energy in an electrical field while an inductor stores it in a magnetic field. And of course, they release this energy back into the wild at some point in time. The third basic component we need is a resistor, which cannot store, but dissipate energy and convert it into heat. We can use these energy-storing and energy-dissipating properties to implement the previously defined mathematical description of a filter by connecting them in certain ways.

I’ll leave out the boring details of different circuit structures for now. The important thing to understand is that by connecting some basic elements in certain ways, we are actually implementing a set of mathematical equations. In essence, we let these components compute stuff for us.

Passive Filters

If we only use these three basic components, we are a bit limited in what we can do, because we don’t have a component yet that is able to boost energy. We can only redistribute it in time or get rid of it in the form of heat. That’s called a passive filter.

In the early days of audio electronics, this was the common way to build an equalizer, since active components were still quite expensive and impractical to use in larger numbers. The good old Pultec equalizer is a standard example. And also the tonestacks you’ll find in guitar amplifiers are almost exclusively passive.

The fact that passive electronics can only take energy away from the signal results in an inevitable level loss when going through such a circuit. Usually, an amplifier stage after the passive filter circuit makes up for this level loss. But since the filtering itself happens without any active component, we still call it passive.

Due to all these limitations, the possibilities with passive filters are really quite limited. The biggest issue is that without active components right inside the filter circuit, it’s impossible to decouple different parts of the circuit from each other. Consequently, changing a component value (for example using a potentiometer) will have more effect than we’d like.

For example, adjusting the level of the treble band might not only change the level of high frequencies, but also the corner frequency of the filter. Or the mid band can behave differently depending on the setting of the low band.

This rather touchy behavior of passive equalizers is indeed quite common and makes much of the character of these old devices. They never quite behave as the labeling of the knobs suggests. For example, on most guitar amplifier tonestacks, you get the flattest possible response if you set the mid band to 12 o’clock and both bass and treble bands to their lowest setting.

Active Filters

With the advent of transistors and especially integrated circuits, it became much easier and cheaper to build filter circuits closer to spec. Almost all the hardware equalizers you find anywhere are active. With these, you don’t have these nasty interactions and side effects, and the treble band is really independent from the mid band and so on. You can expect the actual behavior of the unit to be much closer to the labeling printed on the frontpanel.

Apart from that, however, an important difference between passive and active technology is their distortion behavior. While the makeup amplifier in a passive equalizer applies its distortion characteristics to the output signal as a whole, the case is much more complicated for active circuits. As the active components sit right inside the actual filter implementation, the distortion might be applied to only part of the signal. Sometimes even exactly the part of the signal that is filtered out. As a result, the THD of – say – a lowpass filter can be much higher above the cutoff frequency while it’s rather clean below the cutoff.

Other Issues

Generally, electronic circuits often require compromises due to limited capabilities of components. Especially variation in component values is a big issue that can make it hard to exactly match a specification. Temperature variation and aging are issues as well.

As a result, it’s well possible that one unit of the same equalizer model sounds dramatically different from another one at similar settings. However, this can often be alleviated by readjusting the settings to better match.

However, what these component variations do not change is the general behavior of the box. That means, how it reacts to knob adjustments and how it treats certain signals. And that’s what – for me – constitutes the real character of an audio device.

Digital Equalizers

With computers, we don’t have any component variations. Everything is nice and predictable. And after all, computers are for computing. And what we want to do is to have something compute some equations for us. Perfect, right?

Yes and no. It is true that we can use digital technology to implement filters much more precisely. And much more predictably. But we have different limitations here that lead to totally different problems and solutions.

First of all, there’s one thing we can do with electronics but we can’t with computers, and that’s doing math with continuous signals. We need a set of individual numbers to type into the pocket calculator. Obviously, the solution is sampling.

The bad news is: there’s no way to exactly transfer an analog filter design into digital domain without introducing problems.

There are mainly two issues with digital filters in comparison to analog ones: frequency warping and rounding errors. The former leads to the frequency response being squeezed towards higher frequencies. You already encountered the phenomenon a couple of weeks ago. The latter leads to low-level noisy artifacts especially at low frequencies.

A while ago, I wrote down the typical issues with digital filters in two articles for RedShark Sound. So head over there for some more details.


As you can see, with two comparably long blog posts, we’re still only barely scratching the surface. I hope I could give you a good overview on what to look for, and of course what to listen for. To get deeper into the rabbit hole with me, subscribe to my weekly newsletter so you don’t miss when we’re taking even closer looks at your favorite audio tools!

As always, for any detailed questions and remarks, have fun in the comments section below!