Basic Electronics for Recording Engineers - Part 3

This is the third installment in a series on Basic Electronics for Recording Engineers. This episode focuses on equalizers and amplifiers. Why do we call them equalizers? And amplifiers are at the heart of every piece of audio gear, even if we don’t usually recognize them as such.

I discuss different approaches to designing an equalizer. And cover the concepts of amplification from the original vacuum tube circuits, through transistors, and then to opamps. Opamps make large format consoles possible, and they are at the heart of almost every audio device in your studio.

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Transcript

115        Basic Electronics for Recording Engineers – 3  

Equalizers and Amplifiers                                                           February 9, 2026

I’m Doug Fearn and this is My Take on Music Recording

 This is part three of a series on basic electronics for recording engineers. We work in a technological endeavor, and the more we understand the tools we use every day, the better we can record.

In the first episode of this series, I explained some fundamental things like voltage, current, and wattage. And in the second part, the difference between alternating current and direct current, and the reasons why both are needed in our studio equipment.

We ended with resonance, which is a concept that is fundamental to some electronic circuits, like equalizers; as well as how musical instruments work; and it applies to acoustics.

In the 1920s, radio broadcasting was as big as the internet was when it came on the scene in the 1990s. Radio networks had programs that were distributed to hundreds of affiliated local stations around the country, by way of dedicated telephone lines.

Telephone calls only needed to convey voice audio frequencies between 300Hz and 3kHz. But radio demanded a wider frequency range, especially for music. Federal Communications Commission rules required that radio stations had to prove that their facilities could transmit audio from 50Hz to 5kHz. And most stations did far better than that. Only AM radio existed back then.

In order to get the programs to the affiliated stations, the networks used dedicated telephone lines, leased from the Bell Telephone Company. The telephone company – there was only one back then – had to somehow extend the limited bandwidth of their wires to provide the expanded audio bandwidth required for radio. And resistance in the long wires reduced the level, so amplifiers had to be inserted every mile or so in order to deliver the +8dBM level the broadcast stations needed.

Bell Labs, the R&D arm of the telephone system, figured out how to do those things. It was a cutting-edge technology.

The terminology used by the telephone system was purposely kept simple and as non-technical as possible, since most of the technicians doing the actual installation work were not electronically-educated. So the term equalizer was used to describe the equipment that modified the frequency balance so that it yielded the flat response that was needed. All the frequencies were made “equal,” as read on the VU meter of the telephone company test equipment.

That’s how we ended up with the term “equalizer” to describe our frequency-modifying device we use every day.

But in the studio, we rarely use equalizers to make a system have flat frequency response, at least not in our creative use of equalization. We use equalizers to shape the frequency response as necessary to achieve our artistic goal of providing the best experience for the listener.

There are hundreds of equalizer designs out there, as you know. But they all have one thing in common, and that is that they are considered to be resonant circuits.

Capacitors are central to an equalizer. They can be combined with an inductor, or with a resistor. Or maybe both an inductor and resistor, depending on the design.

As I explained earlier, a capacitor and an inductor are sort of mirror-images of each other in frequency realm.

For a non-audio example of a resonant circuit, most radio receivers up until recently utilized a fixed-value inductor and a variable capacitor to tune in a particular radio station and exclude all others. That circuit used the property of resonance to emphasize one radio frequency and minimize all others.

In the audio world, a modified version of the tuned circuit used in a radio receiver could be designed to null-out one problem frequency without affecting other frequencies – a notch filter. That filter could be useful to eliminate a source of 60Hz hum that got into a recording, for example.

Such a sharply-tuned circuit is said to have a high Q. Q stands for quality, but that applies to radio circuits, not audio. We use the term Q to indicate how sharp or broad a peak or null in the audio actually is. High Q means that only a very narrow range of frequencies are boosted or cut, centered on the design frequency. Low Q simply broadens out the peak to cover a wider frequency range.

Some equalizers have a variable Q, which can be useful to match the need.

But how does this circuit actually do what we want?

I am speaking here of conventional analog equalizers. The same effect can be mostly duplicated in a digital circuit, but the concept is still based on the real-world of analog signals. Our hearing is totally analog, as is the sound we perceive.

In a circuit, both capacitors and inductors are frequency-selective. A capacitor, for example, rolls off frequencies below a point that can be calculated, depending on the value of the capacitor.

A large-value capacitor, measured in several microfarads, might pass audio down to 20Hz, while a smaller-value capacitor, measured in fractions of a microfarad, might only pass frequencies that are very high, depending on the capacitance. Those capacitors are at the heart of high-pass filters, and low-pass filters.

It’s really more complex than that, but that will give you a sense how such filters operate.

If we want to boost or cut frequencies in the middle of our hearing range, we have to create a different type of filter, and that requires the addition of an inductor or resistor to our capacitor.

An equalizer that uses inductors is sometimes called an LC equalizer. The L is an abbreviation for inductance, named after the scientist who first defined inductance.

Passive equalizers do not incorporate any active amplification in the frequency-modifying circuit. Passive equalizers can only attenuate a range of frequencies. They cannot boost them. So how do we get a boost eq? Well, by attenuating all the frequencies above and below the frequency range we want to boost.

As a general rule, the amount of boost an equalizer provides, say 12dB, means that the circuit will have 12dB of loss, which must be made up by an amplifier. Multi-band equalizers will have additional loss, some from each section, and more make-up gain will be needed.

Passive equalizers are popular, but there are other ways to achieve the same basic frequency modification. And many of those circuits utilize the resonant elements within the amplifier circuit. Active equalizers have several advantages, mainly that they cost less to build, and can provide a wide range of parameters that can be adjusted.

That’s the basic concept. There is more practical detail on equalizers in the episode, “The VT-5 Equalizer: Design and Use” from September 2020.

I have mentioned amplifiers several times in this series of episodes. I think most people know what an amplifier is, but let’s take a closer look.

Amplitude is the technical term for describing what we call the level of an audio signal. Or the loudness of a sound, for that matter.

Most audio starts from a microphone, and the output of a microphone is miniscule, measured in millivolts, thousandths of a volt. A typical microphone output might be -50dBm, which is about 3 millivolts. Three thousandths of a volt. That’s pretty low voltage. It’s not high enough to be used in any practical way. A passive speaker might require 10 watts to operate. That’s about 10 billion times higher than our -50 mic level. Decibels are logarithmic units.

To boost the microphone output to speaker level we need about 90dB of amplification. And that amplification is done in several stages.

The first is the microphone preamplifier, which boosts the mic signal to around +4dBm. Then the power amplifier driving the speaker boosts that up to about +40dBm.

Of course, equalizers and other circuits, like a compressor, actually have a loss. So the actual amount of gain is even higher, when you consider the path from mic to speaker.

There are several ways to build an amplifier, depending on the requirements. For a microphone preamplifier, we are dealing with a minute input signal. Mic preamps typically have between 30 and 70dB of gain, and that is accomplished through multiple amplifier stages. Each stage boosts the gain by a certain amount, to give us the full amplification we need.

This can be done a few different ways. The first audio amplifiers used vacuum tubes, a technology still used today in studio equipment for its highly-musical sound. Solid-state amplifiers are more common, and they can use either discrete transistors or integrated circuits. An audio integrated circuit, an IC, combine dozens or even hundreds of individual transistors in one small package. Note that the majority of ICs in the world today are digital devices, not analog. Digital ICs can have millions of individual transistors inside them.

Whatever the amplifying device, they all work on the same basic principle: a small signal, like from our microphone, controls a much larger voltage, from the power supply in the equipment. In the UK, they call vacuum tubes valves, and that is a great way to understand how they work. In a valve, turning a small control like a wheel or lever, regulates the flow of a source of water or other fluid. In an electronic amplifier, the knob is small signal, and it controls the flow of a much larger source of power.

The same principle applies in both a vacuum tube amplifier stage and a solid-state one. The devices are different but the result is that same.

There is one difference, however: tubes are voltage amplifiers and transistors are current amplifiers. That would seem to be a minor difference, and it is, except for some practical circuit considerations. I have talked about that in the episode, “Vacuum Tubes: Why They Sound Better for Audio,” from July 2020.

Electronic engineers divide the individual components used in any electronic circuit into two classes: active devices and passive devices. In an amplifier, an active device is one that does the actual work of increasing the level of a signal, while passive devices fill a supporting role to make the circuit work.

Typical active devices are transistors, integrated circuits, and vacuum tubes.

Passive devices include resistors, capacitors, inductors, and transformers.

All active amplifying devices are designed to have particular properties, including the amount of gain they provide. The gain can be specified in several ways, but the best term for our use in this discussion is amplification factor. That is simply a number indicating how much the signal is multiplied. Typical values range from 10 times to 100 times.

If you need a lot of amplification, that is, a lot of gain, it would seem that using an active device that matches the gain required would work. And it probably would. But that is not the best way to design any amplifier.

But with higher gain comes increased noise. Other undesirable things can happen to the audio as well in a high-gain amplifier stage. From a quality point of view, it is usually better to use multiple stages of lower amplification rather than one high-gain stage.

Equipment designers have to satisfy many conflicting requirements in order to make a viable product. And for many equipment manufacturers, one of the biggest factors is cost. Multiple gain stages means that there will be increased component costs.

A good example of this in the tube world is the 12AX7. It is a dual triode from the 1940s that was a miniaturized version of a tube designed in the 1930s for consumer electronics products, like radio receivers or guitar amplifiers. It has an amplification factor of 100.

The 12AX7 provides a huge amount of gain in a single stage, which greatly reduces the cost to make a product. The downside is that a tube with that much gain produces a lot of internal noise. It also has significantly more distortion than a tube with less gain. And the tube is microphonic, meaning that it acts like a microphone, picking up noise and vibration in its vicinity. The acoustic energy rattles the internal structures of the tube, resulting in audible sound in the tube output. I discovered those deficiencies when I was around ten years old, experimenting with various amplifier circuits.

Tubes like the 12AX7 made products like radios and guitar amplifiers cheaper. But in a guitar amplifier, noise is not a big consideration. And the distortion adds to the electric guitar sound. And the microphonic characteristic provides lots of additional distortion and odd sounds when the amplifier is driving the internal speaker to very high levels. The tube is rattled by the speaker vibration.

Classic sound for a guitar amp. Not a good idea for studio-quality equipment.

All transistors also have an intrinsic amplification factor. And since an audio integrated circuit is made with transistors, ICs also have an intrinsic amplification factor.

Audio ICs are almost all operational amplifiers, usually just called opamps. They are the building blocks of amplification central to all modern solid-state studio gear.

Opamps are designed to operate with negative feedback. Basically, a portion of the output of the IC, is fed back into the amplifier input. That would be positive feedback, like you get with a live mic picking up its output from a speaker in a PA system. But in this application, the phase of the feedback signal is inverted, so no howl is heard.

Negative feedback improves the sound by making the output closer to flat frequency response. And it reduces distortion considerably. Negative feedback is used in almost every amplifier, tube or solid-state  – except maybe a guitar amplifier.

The downside of negative feedback is that it reduces the gain of the amplifier. And too much negative feedback can impact the sonic character of the audio. So the least amount of negative feedback is used. It’s a compromise.

In an IC op amp amplifier circuit, the gain of the amplifier is set by the amount of negative feedback. Op amps without feedback would have extraordinarily high gain and they would quickly be destroyed. The gain of that op amp is adjusted by changing the negative feedback.

Op amps make our solid-state audio equipment possible, since they are tiny devices and very versatile. A console or mixer will have hundreds, or even thousands of op amps.

Negative feedback is used in almost every audio amplifier circuit. It improves performance. The feedback path might go from the amplifier output back to its input, or the feedback loop might just be around one stage of a multi-stage amplifier circuit. These are design considerations that affect the sound of a studio device.

A challenge in the design of any electronic gear for music recording is accommodating a huge range of signal levels. An extreme example would be recording a symphony orchestra, where the sound level produced can easily have a 100dB range.

Making a recording system work over a 100dB range is challenging, if you want to have low noise and low distortion.

Designers have to be very careful to make sure that there is not a single amplifier stage that overloads and distorts at a lower level than any other amplifier stage. That challenge is often called gain staging, and that topic will start the fourth installment of this series.

Thanks for listening, commenting, and subscribing.

 You can reach me at dwfearn@dwfearn.com

 This is My Take on Music Recording. I’m Doug Fearn. See you next time.