Basics of Mixing – 14.4 Quantization Error & Dither

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

Today, I’d like to introduce dither/dithering, the final step in mastering. This article is based on my book, Basics of Mixing, published in South Korea.

Let’s start!

Before we talk about dithering, we must know “quantization error”. Then, what is quantization?

You may have heard the word “quantum” like quantum physics or quantum electrodynamics. Quantum means ‘discreteness’ or ‘lack of continuity’. In microscopic world, particles like electrons, protons, and neutrons have discrete physical properties. So, the word ‘quantum’ often used in particle/nuclear/quantum physics.

https://www.masteringthemix.com/blogs/learn/what-is-dither-and-does-it-matter

By the way, when we record sounds, they are saved as individual samples by sample rate and bit depth format.

In this process, continuous analog audio signals are converted into discrete digital signals. We can call this is a ‘quantization’ too.

When we play audio, digital signals are converted to analog signals in this kind of staircase format. The differences in magnitude value of original analog signals between converted signals are “quantization error”.

How to solve this problem?

The solution is really simple. Just mix noise with converted signals! The noise may sound slightly annoying, but it makes the audio sound more natural than before.

We call this noise, the solution, “dither”. Dither is more effective on digital audio files with lower bit depth.

As the bit depth increases, these quantization errors become less audible, but since quantization errors can be noticeable in very small sounds, this noise is added at the very last stage of mastering.

In the past, dither was simply used as overall noise from low to high frequencies.

When using the Dither function of Fabfilter Pro-L2

However, since you can use dither as long as there is irregular noise, these days, there are many cases where engineers use dither for ultra-high frequency noise that is low in real sound. Since this is noise, it is common to apply it only once, at the very last stage when all volume adjustments are complete.

Next time, I will explain the codec of music files and finish the basics of mixing. Then, I will see you again in the next article!

5 Brainworx Bettermaker Plugins Review (Bus Compressor, C502V, EQ232D, Passive Equalizer, BM60)

Hello! This is Jooyoung Kim, a mixing engineer & music producer.

Today, I’d like to talk about some plugins from Brainworx’s Bettermaker lineup:

  1. Bus Compressor DSP – Compressor
  2. C502V – Compressor
  3. EQ232D – EQ
  4. Passive Equalizer – EQ
  5. BM60 – Reverb

I originally planned to introduce these plugins last month when they were on sale, but things got a bit hectic, and now I’m sharing them when they’re not discounted… oops! 😅 Still, after testing them out, I can confidently say they’re really well-crafted plugins. If you’re interested, it’s worth keeping an eye out for a sale.

These plugins were provided to me by Brainworx via Plugin Boutique in NFR (Not For Resale) format. If you purchase through the links I’ll include later, I’ll earn a small commission, which genuinely helps me keep doing what I do. 😊

Ready? Let’s dive in!

1) Bettermaker Bus Compressor DSP

Click image to purchase Brainworx Bettermaker bus compressor dsp

The Bettermaker Bus Compressor is a plugin recreation of Bettermaker’s hardware Bus Compressor. The original hardware allows digital recall and adjustments, and the plugin’s interface mirrors that design pretty closely.

The plugin’s parameters are similar to most compressors, but a few features stand out:

  1. You can listen to the sidechain.
  2. You can choose between Peak Level or RMS detection.
  3. You can dial in the amount of THD (Total Harmonic Distortion).

These three features are the heart of the Bettermaker Bus Compressor DSP. The sidechain splits the incoming signal, applies a filter, and uses the filtered signal as the key—pretty standard for bus compressors, but super useful.

The VCA THD feature adds harmonic distortion.

On the left, you’ve got the clean signal; on the right, it’s with THD applied. Looking at an oscilloscope, you’ll notice more changes—sonically, it starts to feel very mechanical. A subtle touch adds an edgy vibe, but crank it too much, and it feels overly distorted.

The frequency response is flat except in the ultra-low end, and the compression curve is smooth as butter. After messing around with it, I think it’s an incredibly well-made bus compressor. SSL-style compressors sometimes smear the low end or feel a bit hollow, but this one keeps the lows tight and solid without eating them up.

This one’s a winner. Out of the five plugins I’m covering today, it’s my favorite. It’s usually a steal during sales (probably around $29, so I’d recommend testing it out and grabbing it when it’s discounted.

2) Bettermaker C502V

Click image to purchase Brainworx Bettermaker C502V

This is another plugin modeled after Bettermaker’s C502V hardware. It offers three modes:

  • BM – Bettermaker Compressor
  • SG – SSL G-Compressor
  • DX – dbx 160 Compressor

Switching modes even changes the UI, which is a nice touch.

After testing, I could hear what each mode was going for. The SSL mode feels clean and polished, while the dbx mode has that signature gritty snap. The Bettermaker mode, though, sounds a bit different from the Bus Compressor DSP—it’s more forward and punchy, at least to my ears.

Here’s the harmonic distortion in BM mode with THD maxed out.

First is BM mode, second is SG, and third is DX. The differences in harmonic distortion and compression curves explain why this plugin sounds distinct from the Bus Compressor DSP. The C502V has a steeper knee, and it varies by mode.

I’d recommend this one too—it’s like having three compressor flavors in one. I especially liked the BM and DX modes. If you don’t already own a dbx-style compressor, this could be a solid solution.

3) Bettermaker EQ232D

Click image to purchase Brainworx Bettermaker EQ232D

This plugin recreates Bettermaker’s EQ232P MKII hardware—a super clean EQ with no THD, just pure frequency shaping.

It’s split into sections:

  1. A high-pass filter (HPF) for cutting lows.
  2. EQ 1, handling lows to mid-highs.
  3. EQ 2, covering mid-highs to ultra-highs.
  4. A P EQ section with Pultec-style frequency response.

Think of it as a mastering EQ (sections 1–3) plus a Pultec-inspired EQ combined into one.

On the left, I activated just the P EQ section and tweaked it a bit. The curve is as complex as you’d expect from a Pultec-style EQ. Harmonic distortion is pristine, and there’s no compression curve to speak of—it’s just an EQ, plain and simple.

It’s perfect for mastering when you need something ultra-clean. Definitely worth checking out.

4) Bettermaker Passive Equalizer

Click image to purchase Brainworx Bettermaker passive equalizer

This one looks like a Pultec clone at first glance, but it’s actually Bettermaker’s unique take on a valve-based passive EQ, turned into a plugin. The hardware allowed computer-based recall, and the plugin follows suit with a similar UI.

The frequency response is close to the Bus Compressor DSP.

But as a valve design, the saturation is intense! Left is the default state; right is with Heat engaged.

Looking at the compression curve and oscilloscope, it’s far from flat.

Honestly, it feels more like a saturation box with EQ tacked on than a pure EQ. And as a saturation tool, it’s got charm—the Heat setting is pretty tasty.

5) Bettermaker BM60

Click image to purchase Brainworx Bettermaker BM60

The BM60 is a reverb plugin based on the Lexicon PCM 60. The original hardware had intuitive controls, and this plugin adds extras like Predelay, Width, Monofilter, and a Ducker, making it really versatile.

It offers two reverb types—Room and Plate—with Size and Reverb Time split into four steps. The parameters are straightforward, and the sound feels just as intuitive.

It’s great for light, easy reverb duties. Reverbs and delays are tough to describe with numbers, but I’d recommend this one as much as the Bus Compressor DSP.

Purchase Brainworx’s plugins at Plugin Boutique

They’re not on sale right now, but I’ve left a link to Brainworx’s plugins for future reference. Take a look when you get a chance!

Bonus: This month, if you buy anything from Plugin Boutique, you can snag either Audiomodern’s Freezr (a freeze sequencer plugin) or Heavyocity’s MicroFX Refiner (a bus processor plugin) for free. Don’t miss out when you shop!

That’s it for now—see you in the next post! 😊

Basics of Mixing – 14.3 Oversampling and Upsampling

Hi! This is Jooyoung Kim, a mixing engineer and music producer.

During my undergraduate studies in physics, I often used my extra credits to take music courses. Looking back, I regret not taking any Python classes—especially now that I’m studying plugin development, data processing, and methodology. Without AI tools, I wouldn’t have been able to start coding at all.

Currently, I’m in the final semester of my master’s program in the Department of New Media Music. I’m not too worried about my thesis, so this semester, I’m focusing on a personal project: developing a saturation plugin in my own way. I know it will be challenging, but I also aim to write a paper introducing a new methodology for building audio plugins.

Now, let’s talk about oversampling and upsampling. This article is based on my book, ‘Basics of Mixing‘, released in South Korea.

What are oversampling and upsampling?

Many audio plugins offer oversampling, but what exactly is it, and how does it work?

  • Oversampling: Increasing the existing sample rate by an integer multiple (e.g., ×2, ×4, ×8, etc.).
  • Upsampling: Increasing the existing sample rate, but not necessarily by an integer multiple.

In oversampling, the process inserts zeros into the empty values and interpolates them using a low-pass FIR (Finite Impulse Response) filter..

It’s me! The right one is the photo on the left doubled horizontally.

Think of it like stretching an image by an integer factor: the blank spaces are filled in through interpolation. After processing the audio, the plugin then downsamples the result back to the original rate.

On the other hand, upsampling converts the sample rate to another rate through interpolation. This is a different process from oversampling.

Why do we use oversampling and upsampling?

Oversampling helps reduce aliasing and creates a more natural-sounding result, while upsampling is primarily used for changing the sample rate.

However, does an oversampling function always make a plugin sound better? That’s a different discussion.

For example, the SSL Bus Compressor 2 plugin provides an oversampling option. Below, you can see the frequency response of the plugin:

The left graph shows the response without oversampling. The right graph shows the response with 2× oversampling.

Both graphs exhibit harmonic distortion at the same positions. While other measurements appear similar, even small differences between the two can result in noticeable changes in sound. In my experience, non-oversampled processing often sounds better.

That was a brief explanation of oversampling and upsampling! See you in the next post!

Basics of Mixing – 14.2 Aliasing and Nyquist-Shannon Sampling Theorem

Hi! This is Jooyoung Kim, mixing engineer and music producer.

The paper I submitted to JASA is currently under review.

The assigned editor is a renowned scholar in the field of acoustic signal processing at Tsinghua University. I’m more nervous than I was during my college exams..

By the way, today we will talk about the aliasing and sampling theorem – the basic knowledge for mastering audio.

Let’s start!!

Aliasing

Aliasing is a phenomenon caused by converting analog data to digital data. You can see the same picture; the left one looks normal, but the right one appears distorted. The right image was made by resizing left image in low resolution.

Why does it happen?

Nyquist-Shannon Sampling Theorem

A study on factors affecting transmission published by Nyquist in 1924. Reading the paper, it seems that there were studies at the time that showed that waveforms such as sine waves, triangle waves, and square waves had a significant impact on transmission.

But the courageous Mr. Nyquist says in a strong tone that this is not the case…chill guy..

Nyquist laid the foundation for sampling theory through his research on telegraphy in 1924 and 1928.

Claude E. Shannon expanded on this in his 1949 paper, ‘Communication in the Presence of Noise,’ which solidified the modern sampling theory. If you are curious about what kind of paper was actually written, I have included a link so please take a look..

For those working in music and audio, the finer details aren’t crucial, so I’ll focus on the key points.

1) Only frequencies up to half the sampling rate can be accurately represented (this is known as the Nyquist frequency).

2) The original source with a value higher than the Nyquist frequency is expressed as a value below the Nyquist frequency when sampled (this is the aliasing phenomenon).

3) Therefore, if you filter it in the high-pass to filter out this noise, it becomes clean (this is called the anti-aliasing filter, The original Nyquist paper made telegraph transmission easier through this filtering).

The blocky artifact shown in the first image is also a form of aliasing caused by sampling errors. Since it samples space, it is a bit different from the sound of sampling time.

An ADC (Analog-to-Digital Converter) chip performs these tasks!

High-end ADCs and DACs improve the precision of this process.. Still, Lavry Gold converters are extremely expensive… Maybe due to low demand?

The first part of this video shows how aliasing sounds when you lower the sample rate. Using a filter before this plugin to remove frequencies near the Nyquist limit will significantly reduce aliasing noise.

That’s all for today.

In the next article, I will talk about oversampling and upsampling.

See you in the next post!