DIY Audio Project #1 | Tube Saturator with Baxandall EQ (Part 4): Fix, Tune, and Refine

Hello, this is Jooyoung Kim, a mixing engineer and music producer.

Lately, I’ve been struggling with a few things like studying English for the TOEFL, preparing class materials, and conducting research. Among all these things, troubleshooting my tube saturator is the most convoluted work.

I wanted to post my article when I finished my build. However, since this continuous refinement makes it hard for me to write it, I’ve decided to document my progress step by step from now on.

Let’s start!

Main Problems

While fixing this machine, too many problems kept popping up. I’ve listed them below in the order they occurred:

  1. The unit wouldn’t power on, even though I flipped the power switch. -> resolved
  2. As soon as I powered it on, the toroidal transformer got so hot that it could fry an egg. -> resolved
  3. No output signals. -> resolved
  4. I found some issues with the meters. -> resolved
  5. One channel was muted, and a persistent white noise appeared. -> unresolved
  6. To fix the problem 5, adding capacitors made things worse. -> unresolved

So far, those are the steps I’ve taken. Therefore, I’ll explain each step chronologically.

The unit wouldn’t power on

It turned out to be just a cold solder joint. However, it was a real struggle to track down. While re-soldering everything, the problem still wasn’t solved. Then, I realized I hadn’t checked the solder joints on the power cord! Once I reflowed them, the power finally came on!

The toroidal transformer got so hot

This problem stemmed from a design mistake. Specifically, the connection between the 0V line and GND caused the transformer to overheat. To fix this, I disconnected the line and insulated it with heat shrink tubing.

No output signals

I figured that if I mounted the THAT1646 (SOP-16) onto an SOP-16 to DIP-8 breakout board, it would work. However, it turned out to be a completely wrong assumption. So, I ended up replacing all of them with DIP-8 standard devices.

Meter problems

Once I finished soldering the entire circuit, I ran into three major issues with the meters:

  • The right meter channel wasn’t working at all.
  • Even though the meter circuit followed VU RMS standards, the ballistic response was way too fast.
  • The audio signals were too hot, causing the needles to peg.

I found that the first problem was just a wiring mistake on my part.

With a little help from Gemini and ChatGPT, I sorted out the second and third issues by feeding them the circuit text. It turns out the meter circuit was designed for -10 dBV (consumer audio level). To fix this, I swapped out a couple of resistors to pad down the input levels for both channels. As for the meter ballistics being too fast, I replaced the damping capacitors with higher-value ones to slow the needles down.

One channel was muted, and a persistent white noise appeared

Yes… and this is the exact problem I’m still trying to sort out. Following ChatGPT’s advice, I added several decoupling capacitors to stabilize the power supply for the THAT1646 and OPA1612 chips.

At first, I misunderstood the advice and connected the capacitors in series with the supply voltage line and the chip pins. However, drawing on my background in physics, I realized that since it’s a DC power rail, connecting a capacitor in series didn’t make any sense. So, I double-checked the circuit configuration and finally realized it was a mistake.

To make matters worse, I totally messed up.

Honestly, I just needed to solder a few capacitors in parallel to the power rails. But instead, cluelessly overcomplicating things, I ended up desoldering all the chips and trying to install sockets.

Since cleanly desoldering an 8-pin chip is an absolute pain, I just lost my patience and clipped the pins with nippers. Then I tried using square-pin headers, which didn’t fit into the sockets at all. So I had to order brand-new OPA chips, re-solder the pins, and wire up the capacitors in series again.

After going through all that self-inflicted hell, I finally tracked down some round-pin headers, re-soldered the chips properly, and got them plugged into the sockets.

But guess what? Now, instead of music with some background noise like before, I’m getting pure, signal-driven noise blasting through. Ugh… hahahahahaha…haha…

hahahahahahahahahahahahahahahah……ahaha..aaaaaa!!!!!!!

So yeah, this is the story of my troubleshooting journey so far.

While writing this post, I tried fixing a few more things last night. However, it turned out that my efforts were completely ineffective.

Juggling multiple things at once makes this feel like it’s dragging on forever, and I just want to wrap it up already. If all else fails, my last resort will be to rip everything out and wire it up on a fresh PCB… ugh. Trying to finish this all during break was definitely too ambitious.

See you in the next post!

Just Found Out: Yeonjoon Yoon’s Etudes: Triptych – Live Was Out + Life Update (26.8.3)

Hello, this is Jooyoung Kim, an audio engineer and music producer.

Recently, I’ve been busy conducting research related to Korean traditional musical instruments, which requires several listening tests. Therefore, I met Yeonjoon Yoon, a composer and pianist who has worked with some Korean traditional musicians.

After the test, I asked him when the album that we worked on together last year would be released. He surprisingly said it had already been released!

So I decided to introduce the album.

I used Peluso P-87 microphones for stereo miking on the grand piano, and Rode NT-55 microphones for the drum overheads and ambience positions. I also positioned two Shure SM57s on the top and bottom of the snare. I can’t quite remember, but I believe there was a rack for the kick microphone, so I likely used a Shure SM57 for that role as well.

There is also a peculiar musical instrument called the Saenghwang (a mouth organ). I remember that I installed a Peluso CEMC 6 over the head of the musician.

I used a Midas m32 console for both live mixing and recording simultaneously.

Yes, that’s all—maybe. It’s been about 9 months, so I can’t remember all the details accurately.

I only remember that the venue’s interior was made of concrete and brick, so it had a pretty long reverb time. Because of that, I struggled with the acoustics while mixing the live sound. However, in the recorded tracks, the piano actually sounded a bit too dry, so the artist wanted to add a huge amount of ambience—around -10 dB (whereas in popular music, ambience effects like reverbs are usually added around -30 to -20 dB).

These are unique and interesting tracks, so check them out..!

Meter part

As you know if you read my DIY hardware post series, about a month ago, I finished wiring and assembling all the components for my DIY audio hardware. When I powered it on, I immediately ran into two major problems. Only the left channel’s meter responded while the right channel’s meter remained completely dead, and there was no audio output at all.

I wasn’t even sure where to begin troubleshooting. After wiring up all the components for my DIY audio hardware, I ran into two major problems right away. First, only the left channel’s meter responded, while the right channel’s meter remained completely dead. Second, there was no audio output at all.

After spending so much time on the project without making any progress, I became so frustrated that I stopped working on it for a while. It honestly felt like burnout.

When I came back to it yesterday, I started thinking more logically. The meter was connected just before the THAT1646 output driver stage. If the left meter was responding correctly, that meant the entire signal path leading up to that point had to be functioning. In other words, the problem couldn’t be somewhere earlier in the circuit.

Following that line of reasoning, I first discovered that I had simply wired the right meter incorrectly. That explained why only the left meter had been moving.

With the meter issue solved, the only remaining suspect for the missing audio output was the THAT1646 itself. So I started digging into its documentation and pinout.

OPA 1612…!!

However, after replacing the devices, I discovered another problem. A significant amount of broadband noise was present, measuring higher than -75 dBFS across nearly the entire frequency spectrum.

I went back through the schematics and the datasheets, comparing my design with the recommended application circuits. That’s when I realized I had made a fundamental mistake: I hadn’t included the power-supply bypass capacitors for either the OPA1612 op-amps or the THAT1646 output drivers.

Looking back, it seems obvious. High-performance analog ICs like these rely on proper local decoupling to keep their supply rails stable and free from high-frequency noise. Without those capacitors placed close to the power pins, the circuit is far more susceptible to noise and instability.

So, despite thinking I was finally done troubleshooting, I ended up ordering yet another batch of components. Now I’m waiting for the capacitors to arrive before I can continue.

At this point, the project is slowly turning into Frankenstein’s monster. Every problem I solve seems to require another small modification to the hardware. Hopefully, these will be the last stitches before it finally comes to life.

That’s it for today.

Lately, I’ve barely had enough time to breathe between studying English and working on my research, so progress on this project has been a little slower than I’d like.

On the bright side, I recently picked up an SPL Vitalizer MK3, so you can look forward to a review of that in an upcoming post.

Thanks for reading, and I’ll see you in the next one!

Advanced Synthesizers and Gear – (4) Modulations and Vocoder

Hello! This is Jooyoung Kim, an audio engineer and music producer.

Last post, we figured out the fundamental components of the modular synthesizer: VCO, VCF, and VCA. Today, I’ll talk about modulation , which is essential in various systems, including vocoder.

I have already written a few posts on modulation principles and related synthesizers—especially frequency modulation. Please read them first if you are not yet familiar with the basics of FM (Frequency Modulation), AM (Amplitude Modulation), PM (Phase Modulation), and RM (Ring Modulation).

Basics of Synthesizers (4) – Modulation Synthesis and FM (Frequency Modulation)

Modulation

Although the PM graph in the AI-generated image contains some visual inaccuracies, the overall illustration still serves as a helpful conceptual guide when paired with the article.


So, what is modulation?

Essentially, it is the process of altering a parameter of one audio signal (Carrier) using another (Modulator). This creates a temporal cycle, which produces more complex and evolving timbres.

For instance, we can modulate a signal’s amplitude, frequency, or phase; these methods are known as AM, FM, and PM, respectively.

How about ring modulation?

Since Gemini kept glitching, I just had it generate the Python code and tweaked it myself to make this chart..:)

At first glance, Ring Modulation’s (RM) waveform appears highly similar to that of Standard AM. However, a closer look reveals distinct differences, particularly at the null points (zero-crossing points). While Standard AM exhibits no phase inversion, Ring Modulation instantly flips the phase of the carrier signal by 180° every time the modulator crosses the zero axis.

In Standard AM, the unmodulated carrier component is always present because the modulator operates with a positive DC offset. As a result, the waveform’s envelope simply scales down to a null point at its lowest peak and expands back up, maintaining a perfectly continuous phase alignment.

Ring Modulation, conversely, is a form of suppressed-carrier transmission. Without a DC offset, the carrier is completely wiped out whenever the modulator hits zero. The moment the modulating signal goes negative, the carrier wave inside the envelope mirrors itself upside down.

This constant, rapid flipping completely destroys the original harmonic structure of the sound. Instead of creating musical, harmonic overtones, it introduces harsh, non-harmonic, and metallic sidebands.

Buchla & Tiptop Audio Model 258t

Additionally, you can easily find these modulation functions built directly into various VCO modules as well. For instance, the Buchla & Tiptop Audio Model 258t panel shown in the figure above features an FM function with a 3.5mm input jack.

Vocoder

Arturia Minifreak Vocoder

Even though both the Vocoder and the Talkbox (Vox Box) use the human voice to shape sound, their underlying principles are entirely different. In a vocoder, the human voice functions as a modulator that controls the amplitude of a carrier signal—essentially acting as a multi-band AM system.

MXR Talk Box

On the other hand, a Talkbox uses the physical structure of the human mouth to filter the original instrument sound. It behaves like a dynamic formant filter—a concept I will dive into deeper in a future post—emphasizing specific frequency bands as the performer shapes their mouth to speak.

n recent years, vocoders have transitioned into convenient virtual instruments that are incredibly easy to integrate into any modern DAW. To help you get started, here are some of the most acclaimed vocoder plugins on the market today.

(The links below are affiliated with Plugin Boutique, meaning I may earn a small commission at no extra cost to you if you decide to make a purchase—your support keeps this blog running!)

Zynaptiq Orange Vocoder
Softube Vocoder
Arturia Vocoder V
Behringer Vocoder VC16

What’s more, vocoder modules exist in modular synthesizer systems as well like Behringer VC16!

To be honest, I don’t usually attach sound examples in my posts—embedding audio into a blog takes a serious amount of time and effort. That said, every modulation type has a completely different vibe. If you truly want to design your own sounds, you just have to dive in and get familiar with the unique sonic characteristics of each method.

That’s all for today. See you in the next post!

Advanced Synthesizers and Gear – (3) VCO, VCF, and VCA

Hello! I’m Jooyoung Kim, an audio engineer and music producer.

Today, I’d like to talk about the essential components of sound generation in a synthesizer. Every synthesizer or modular system utilizes these three core functions: the VCO, VCF, and VCA.

Their functions are explained below:

  • VCO (Voltage Controlled Oscillator): Generates basic waveforms such as sine, square, sawtooth, and triangle waves.
  • VCF (Voltage Controlled Filter): Processes the signal from the VCO by attenuating specific frequencies. Common types include low-pass, high-pass, and band-pass filters, which are essential for sound design.
  • VCA (Voltage Controlled Amplifier): Acts as a gate or gain stage that shapes the volume of the audio signal over time, typically controlled by an envelope generator (which defines the attack, decay, sustain, and release of the sound).

VCO – Basic Waveforms

Doepfer A-110-1 Standard VCO

You can easily find these waveforms on a VCO module. The image above is an example from Doepfer A-100 module system; the waveforms are depicted on the panel, located below the output connectors.

As you can see, there are four types of basic waveforms: sine, square (pulse), sawtooth, and triangle. They are created by combining various harmonics, which can be expressed using mathematical equations.

Therefore, I will explain all types of waveforms with graphs and equations.

Sine Wave

Sine wave is the most fundamental wave. It is defined by the function:

f(t)=Asin⁡(ωt+ϕ)f(t) = A \sin(\omega t + \phi)
  • A: Amplitude (peak value)
  • ω: Angular frequency
  • ϕ: Phase shift
  • t: time

Square Wave

Square wave is a periodic wave that switches between two levels. It can be represented as an infinite summation of odd harmonics of sine waves:

f(t)=4Aπ∑n=1,3,5,...∞1nsin⁡(nωt)f(t) = \frac{4A}{\pi} \sum_{n=1,3,5,…}^{\infty} \frac{1}{n} \sin(n \omega t)
  • A: Amplitude (peak value)
  • ω: Angular frequency
  • t: time

Sawtooth Wave

This waveform increases linearly and drops instantly. It contains all integer harmonics:

f(t)=2Aπ∑n=1∞(−1)n+1nsin⁡(nωt)f(t) = \frac{2A}{\pi} \sum_{n=1}^{\infty} \frac{(-1)^{n+1}}{n}{\sin(n \omega t)}
  • A: Amplitude (peak value)
  • ω: Angular frequency
  • t: time

Triangle Wave

It is similar to a sawtooth, but symmetric. It consists only of odd harmonics, but with a faster decay of amplitude:

f(t)=8Aπ2∑n=1,3,5,...∞(−1)(n−1)/2n2sin⁡(nωt)f(t) = \frac{8A}{\pi^2} \sum_{n=1,3,5,…}^{\infty} \frac{(-1)^{(n-1)/2}}{n^2} \sin(n \omega t)
  • A: Amplitude (peak value)
  • ω: Angular frequency
  • t: time

Each waveform has distinctive sound characteristics, and these basic waves are the foundation of synthesizers.

As a side note, because the sawtooth wave contains the richest harmonics among these basic forms, it is frequently used to create pad sounds to fill up the background.

VCF – Basic Filters

Behringer 121 Dual VCF

Basic filters are typically categorized into three types: low-pass (high-cut), high-pass (low-cut), and band-pass. Filters have various parameters, but I will introduce only the basics today.

  • Cutoff Frequency: Due to the characteristics of the Butterworth filter (the standard filter used in audio), 3 dB of attenuation occurs at the cutoff frequency.
  • Slope: 6dB/oct is the fundamental slope. This represents one ‘pole’ of the filter. Since filter order (the number of poles) determines the steepness, you will commonly see slopes of 12, 18, 24 dB/oct, etc.

Many filter modules include a resonance function that emphasizes frequencies near the cutoff frequency. When modulated, the fluctuating cutoff frequency also causes the resonance peak to shift. This shifting creates the perception of pitch fluctuation, even though the actual fundamental pitch remains unchanged.

VCA – ADSR Envelope Generator

WYWS MS507

Although I have covered the ADSR concept previously, I will break it down again for those who are new to it. While a VCO and VCF are essential for creating and shaping a sound, they do not have a built-in gate mechanism to trigger or shape the signal’s output. This is where the ADSR envelope comes in.

ADSR stands for Attack, Decay, Sustain, and Release. These four parameters define how the amplitude of a signal changes over time as it passes through a VCA.

  • Attack (time): The time it takes for the sound to reach its peak level from the moment a key is pressed.
  • Decay (time): The time it takes for the sound to drop from its peak level to the designated sustain level.
  • Sustain (level): The volume level that the sound maintains while the key is held down.
  • Release (time): The time it takes for the sound to fade away completely after the key is released.
Behringer 182 sequencer

In modular synthesizers, sequencer modules are often used instead of keyboards.

So far, we have explored the essential building blocks of a synthesizer. These modules serve as the ‘alphabet’ of sound design; therefore, becoming familiar with them is crucial for mastering these complex musical instruments.

That’s all for today. I hope this overview helps, and I look forward to seeing you in the next post!