Category: Basics of Mixing

Basics of Mixing (End) – 14.5 The Codecs of Music Files

Hello? This is Jooyoung Kim, a mixing engineer and music producer. Today, I’ll talk about the music file codecs, final article of basics of mixing series. Those posts are based on my book, Basics of Mixing, published in South Korea.

Let’s dive in!

Codec

The term codec stands for coder-decoder—a hardware or software that encodes and decodes digital signals. There are three main types of codecs:

  1. Non-compression: WAV, AIFF, PDM(DSD), PAM
  2. Lossless Compression: FLAC, ALAC, WMAL
  3. Lossy Compression: WMA, MP3, AAC

Non-compression codecs retain 100% of the original audio data with no compression applied.

Lossless compression codecs reduce file size while preserving all original data. This means they sound identical to uncompressed formats like WAV.

Lossy compression codecs remove some audio data to achieve a much smaller file size, which can affect sound quality depending on the compression level.

In the music industry, WAV, MP3, and FLAC are the most commonly used formats for mastering and distribution.

How is file size determined?

For WAV files, size is determined by sample rate and bit depth. How about mp3 and FLAC?

MP3 files use bitrate, rather than sample rate and bit depth. You’ve probably seen MP3 files labeled 256kbps or 320kbps. This means 256,000 bits or 320,000 bits of audio data are processed per second. Higher bitrates result in better sound quality but larger file sizes.

FLAC files use compression level to control file size. A higher compression level takes longer to encode but results in a smaller file. However, since FLAC is lossless, the sound quality remains unchanged regardless of the compression level.

If you want to compare how different codecs affect sound quality, you can use tools like Sonnox Codec Toolbox or Fraunhofer Pro-Codec.

This is the last article for the ‘Basics of Mixing’ series. Time is really quick..haha.

I hope these posts have helped expand your knowledge and improve your mixing skills.

Thanks for reading, and I’ll 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!

Basics of Mixing – 14.1 Pre-Mastering Audio

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

Before, we talked about hardware. Today, I’ll begin by introducing mastering. This article is based on my book, Basics of Mixing, published in South Korea.

Let’s dive in!

What is Mastering?

Why do we master audio?

  1. The loudness of mixed music is too low
  2. So gain more volume for the mixed music, maintaining the instruments’ balance

Of course, mastering involves many aspects. You can easily understand mastering with the two sentences above.

Simply put, mastering is the final step in music production. It can also be described as packaging the mixed music. The package does not change after mastering.

What a simple and beautiful setup! – https://gearspace.com/board/mastering-forum/1252793-mastering-studio-photo-thread.html

Room acoustics are the most important factor in building a mastering studio. Also, great quality ADC(Analog-Digital Converter)/DAC(Digital-Analog Converter), speakers and hardware are necessary too.

Building a mixing studio is also expensive. However, in mixing studio, audio hardwares take quite large volumes-they make room acoustics worse.

A desk with hardware may look good and fancy, but it takes up a lot of space. This can lead to comb-filtering effects. Removing the desk is a great solution, but as you know, it’s easier said than done..

One more thing, a mixing engineer listens to a song too many times, making it hard for them to recognize problems. Therefore, when you need to master a song or an album, I recommend hiring a mastering engineer.

Nevertheless a mixing engineer should provide a louder monitor file to clients than the mixed file. Due to budget constraints, mixing engineers sometimes have to master their own files.

In this chapter, for those situations, I’ll talk about the process and concepts about mastering.

The full details start next post! 🙂