Years ago, I wrote a piece called "The Binary System, Audio, and Passionate Industry Arguments" for the PMFC community website. It tackled digital audio, common misunderstandings, and what the industry agrees on (and what it definitely doesn't). The article was incredibly well-received, sharing the space with publications from industry giants like Bob Katz and Bob St John.

It was originally published on June 21, 2023 - so it definitely deserved a fresh look. Since the PMFC website has vanished from the internet, I’m publishing this fully updated 2026 version here on the Mix Artist Academy.

Off we go:

Digital Audio Demystified

by Jan ‘Yarn’ Muths [Originally published on the PMFC website blog 19/06/2023, updated and re-written for Mix Artist Academy 10/07/2026 ]
"There are 10 types of people: those who understand the binary system and those who don’t."
Few topics are discussed online as passionately as digital audio and its underlying theory. I’ve followed countless pro audio discussions across forums and social media for over two decades. One thing remains clear: digital audio has the power to ignite fierce arguments. In this blog, I aim to provide a comprehensive summary of what we have learned, what the industry collectively agrees upon, and the aspects that still fuel disagreement and intense emotions. Digital audio keeps on evolving and changing rapidly, and there is no end in sight to this trajectory. It is an amazing and exciting technology to be involved in. It requires keeping an open mind and re-evaluate old view-points every once in a while to check if they still hold true as technology progresses.

Let’s get right into it and look at some of the hot-topic discussions.

Sample Rate Demystified

Fast Facts & Definitions:

Sample rate (a.k.a. sampling frequency) is the number of times per second the amplitude of a continuous analog sound wave is measured, or "sampled." Each amplitude measurement is then converted into digital data by a converter chip. The timing of each sample point is triggered by a periodic pulse from an extremely accurate word-clock generator. Since there is a consistent number of sample points per second, the rate can be measured as a frequency in kilohertz (kHz).

The Nyquist-Shannon Theorem: To capture a sound frequency accurately, your sample rate must be at least double the highest frequency you want to record:

Nyquist frequency = Maximum Frequency Captured

Sample Rate = 2 x Nyquist frequency

Industry Consensus:

  1. Thanks to the work of Shannon and Nyquist, we understand that the sample rate determines the frequency bandwidth of the recorded signal - particularly the upper frequency limit, known as the Nyquist frequency. The math is simple: the Nyquist frequency is half the sample rate. For example, with a sample rate of 48kHz, signal frequencies up to 24kHz are recorded.
  2. Doubling the sample rate also doubles the file size for a given bit depth and recording time.
  3. Certain industries have their preferred sample rates (such as 48KHz in broadcast and video streaming), and it's advisable to adhere to those standards.
  4. Unnecessary sample rate conversions should be avoided. 
  5. To my knowledge, the world-population of consumers who demanded a refund because a sound was recorded at the wrong sampling rate is: ZERO. Correct me if I’m wrong.
Sample Rate and Nyquist Frequency
This sums up the most important points. Now let’s get to the juicy stuff. 
Empty space, drag to resize

Areas of Disagreement:

"Higher sample rate means higher audio resolution."
Sorry to split terminology hairs here, but the parameter that best describes audio resolution is bit depth - not sample rate. The sample rate really only affects the recording bandwidth. It does not affect the quality of signals in the audible bandwidth.
Imagine the sample rate as an extremely steep high-cut filter across every thing in your production.
Empty space, drag to resize
"What is the best sample rate?"
Discussions that start like this turn into a heated bunfight in no time. And don’t expect a conclusive agreement at the end of it. More interesting is why individuals arrive at vastly different conclusions. When highly competent professionals conduct sample rate comparison tests individually, there might not be an agreement at the end. And that’s ok, because sample rate is merely a small component in the extensive chain of devices and acoustics that come into play during critical listening. A garage studio specialising in recording teenage punk bands will undoubtedly arrive at different conclusions compared to a high-end studio built to capture orchestral music.
Similarly, engineers who utilise their digital audio workstations only as tape machines may have different perspectives than those heavily reliant on virtual instruments and digital plug-in processing.

In short: the industry doesn’t agree on the best sample rate.
Empty space, drag to resize
“Frequencies above 20KHz are irrelevant anyway”.
I know for a fact that I cannot hear sine waves above 20KHz. Actually, as a middle aged human who recently turned fifty, I'm relieved that my hearing still extends towards 16kHz. However, dismissing the ultrasonic range as irrelevant oversimplifies the matter. If we followed that logic, we should only ever use 40KHz sample-rates, and those aren’t available for very good reasons. 
Converters aim to capture all frequencies up to the Nyquist limit as accurately as possible. To prevent aliasing, they must strictly block any frequencies above that threshold. To achieve that, some kind of brick wall filter with extremely steep filter slopes is needed. Different converters use a different mix of technologies, such as analog LPFs, oversampling and digital brick wall filters. The process of removing high frequencies with a brick wall style slope can have an effect on nearby frequencies, sending phase ripples downwards into the audible range. While this was a significant concern in the early days of digital audio, modern converters have made remarkable progress in keeping those side-effects at a minimum. 
A SR of 48KHz requires a filter with a slope of appox. >120 dB per minor third.
Still, a healthy distance between the filter’s processing area and the human hearing range is advisable, justifying the need for our gear to operate beyond the upper hearing limits.
To illustrate this, let's draw a parallel from the world of analogue audio. 
During my time as manager of the Customseries 75 factory, I learned that a very wide frequency response beyond the human hearing range is a common, undisputed goal in high-end analog audio design. For instance, an SSL XL9000K console I looked after and maintained for many years has an open frequency response up to about 100 KHz. Similarly, ELI's renowned Distressor extends to 160 kHz, and Millenia's HV-3C preamp goes beyond 300 kHz.
So, does high end sound require a wide bandwidth extending into the ultra-sonic range? Or is a wide bandwidth a by-product of high-end audio design and the desire for a superior transient response?
It's a bit of a chicken and egg scenario - both affect one another.
Engineering genius: The Millenia HV-3C with an incredible 300KHz response
Let’s bring it back to digital audio, and let me risk copping some stick for dishing out my experience and personal workflows. I record at 96KHz these days, and the main reason is that my computer can handle the extra workload with ease while hard-drive space is cheaper than ever. Also, my recording system has a slightly lower round-trip latency at higher sample rates. These are the factual reasons behind my choice.
However, I also have subjective reasons that are rooted in countless hours of listening and my personal taste (not in scientific proof). At higher sample rates, I’ve noticed that vocals processed with Melodyne appear to handle more manipulation before artefacts begin to grind my gears. And my favourite reverb processor tends to sound more realistic, spacious and 3D at 96K, and a little more 2D at 44.1KHz.
Well, there I said it - I’m wondering how much flak I’m going to receive for putting myself out here.

If you’re boiling up with a cranky reply in all-caps, please remember my point above, the one about the world population being zero.

My Sample Rate Recommendations - Short and Punchy:

1. Before starting a new project, think about the final product: Digital streaming, sound for picture, vinyl records or even an Audio CD?  

2. Before your first recording, set your sample rate to match the final product format - then stick with that through recording, editing mixing and mastering.

3. If you need to serve two final products, eg Audio CD (44.1KHz) and also Broadcast (48KHz), then choose the higher of the two.
Empty space, drag to resize

Coming soon:
The next chapter on Bit Depth, word length and the numbers 16, 24 and 32.
Leave your email below to get notified.

Bit Depth Demystified

Fast Facts & Definitions:

Bit depth (a.k.a. word length) is the number of bits assigned to each sample. In analogue-to-digital conversion, the continuous voltage amplitude is quantised (rounded) to the nearest discrete resolution step. The number of available steps is defined by the bit depth.

For every bit added to the word length, the available resolution steps double. This reduces quantisation error (rounding differences) and directly increases the dynamic range by 6.02 dB per bit. Because computers process data in bytes (1 byte = 8 bits), audio bit depth typically uses multiples of 8, such as 16-bit, 24-bit, or 32-bit.

Industry Consensus:

The bits available per sample are used to express the dynamic range (loud and quiet), and the greater the number of bits, the wider a dynamic range can be captured.
In a linear PCM system, each bit represents approximately 6 dBs of dynamic range. (fellow nerds, you may enjoy the equation in the side-box for exact figures). 

Let’s briefly define ‘dynamic range’. 
Signal-to-Error Ratio:

S/E = 6.02 * n + 1.76dB

n = 12 bit ➔ 74dB
n = 16 bit ➔ 98dB
n = 24 bit ➔ 146dB
In the analogue domain, the upper limit of the dynamic range is restricted by distortion, which can manifest abruptly (e.g., transistor technology) or more gradually (e.g., transformer or tube technology). Distortion of around 0.5% total harmonic distortion (THD) typically quantifies the upper limit of the dynamic range.
On the bottom end of the scale, the dynamic range is limited by the noise floor of a device.

In the digital domain, the top of the dynamic range is the clipping point, at which the binary number range has reached maximum.
AD-converter clipping is known to be unpleasant to the ear and it is good practice to avoid it, although I’ve seen it done for creative reasons.
At the bottom of the scale, we encounter quantisation error, which you can envision as the digital equivalent to analog self-noise.
Pretty much every post-2000 converter is capable of recording in 16 bit or 24 bit resolution, with a theoretical dynamic range of approximately 96dB or 144dB respectively. This is quite impressive, considering that the human hearing range spans about 120 dB, from the threshold of hearing to the threshold of pain.
However, we need to point out that those are theoretical maximum figures.

The actual dynamic range should be stated in your converter’s operating manual, and it is always less. Many quality ADCs have a practical dynamic range of 120dB to 128dB and it is my understanding that the analogue components inside the converter (such as line input stages, resistors or balanced line drivers) are the “bottle-neck” here. Either way, modern converters match or exceed the human hearing range, therefore we don’t have to worry about quantisation error degrading our signal, if one gain-stages with care.
Empty space, drag to resize

16bit or 24bit?

This one is simple. Workflows in which 16bit is the best recording resolution are scarce as Yowie’s footprints. The vast majority of the industry appears to agree that capturing recordings in 24 bit is the way to go, even if the final product is going to be a 16 bit file. Digital processors, such as faders, pan-pots and our beloved plugins, benefit from greater input values when applying their math. So, 24-bit should be all we'd ever need. Its dynamic range far exceeds the dynamic range of all analog sound sources I ever record in my studio (and my studio’s live rooms are very, very quiet!).
But maybe it’s not as simple after all. Recently we've witnessed the emergence of 32-bit ADCs, starting with location recorders by Sound Devices, Zoom, and others. 32-bit-float has an unbelievable theoretical dynamic range of 1500dB (no, that's not a typo!), and I’ve heard stories where clipped recordings of a jet-fighter fly-over could be restored by simply reducing clip-gain, and all the above-zero content reappeared undistorted.
32 bit converters don't use just one ADC stage per input; they use dual (or triple) stacked ADCs running in parallel. One ADC handles the lower gain structure (quiet sounds), while a second, padded ADC handles the high-amplitude signals (loud sounds). The onboard hardware seamlessly stitches their digital outputs together into a single 32-bit floating-point file. The analog preamp itself is designed with massive headroom, but the magic is that even if one converter stage clips, the secondary stage doesn't.

It's worth noting that 32-bit converters have also made their way into studio devices like AVID's Carbon interface, although the absence of "float" in the name suggests it may be a linear PCM converter. 

The acclaimed Sound Devices MixPre-3 32-bit-float recorder

In any case, I predict that we'll see a growing presence of 32-bit converters in the future, which might necessitate a complete reevaluation of traditional gain-staging workflows. This said, 32bit-float and 64 bit-float resolution has been around for a long time, but typically only inside a DAW or digital processor unit - sandwiched between 24 bit ADCs and DACs.
Certain DAWs allow for 32bit-float recordings, which might be useful if the subsequent workflow benefits from a higher resolution, such as some time-stretching or pitch shifting algorithms. However, it's crucial to point out that a 32-bit-float recording made from a 24-bit converter will ultimately result in a 24-bit resolution file, with 8 additional empty bits added (or an exponent of 1). A larger file, yet the same 24 bit resolution.

Now, let’s get into some bit resolution topics that get audio peep’s knickers in a knot. Here are a couple of examples:
Empty space, drag to resize

Areas of Disagreement:

From 2 to 16 Million: Debunking Volume Steps Misconception

I’m glad the following misunderstanding is no longer as common as it was 20 years ago, but let’s clear it up just for fun:
“24 bits means 12 volume steps up, and 12 down (for positive and negative half-wave)” 
This bunkum is as far-fetched as a drop bear riding a kangaroo, and I’m happy to explain. In a binary system, each bit has 2 possible states, either 0 or 1.
While this is adequate for the number range from 0 to 1, to express the decimal number 2 in a binary system, a second bit is required. So, the decimal number 2 equals the binary number 10 (pronounced ‘one zero’ to avoid confusion).

I’m sure you get the silly joke in the title now :-)
  • Decimal
  • Binary
  • 0

  • 00

  • 1

  • 01

  • 2

  • 10

  • 3
  • 11
With 24 bits assigned to each sample, we have a total of 224 possible values. That’s over 16 million different resolution steps. Let’s take a moment to take this in.
If you’re recording at 48K and 24-bit, your converter accurately finds the closest amplitude value among >16 million choices, and it does so 48,000 times every second. And now consider higher sampling rates, and did you know your converter is probably oversampling its input 256 or even 1024 times?
Truely mind boggling technolgy!

possible combinations = 2n,

where n = number of bits
  • word length
  • resolution steps
  • 8 bits
  • 256
  • 12 bits
  • 4,096
  • 16 bits
  • 65,536
  • 24 bits
  • 16,777,216
Empty space, drag to resize

Double Trouble! 32 vs. 64: The Battle of the Bits

In recent years, the numbers 32-bit and 64-bit have also been thrown around in computer technology. Some may recall Apple's transition to 64-bit with the Catalina OS, which discontinued support for 32-bit applications. 
However, it's crucial to note that we’re talking about the word-length of software code, not the resolution of audio processing. Nevertheless, audio professionals still benefit from 64-bit applications, particularly for resource-intensive workflows that demand substantial RAM.
  • Bit Depth
  • Bit Rate
  • The quantity of bits used to measure the amplitude, typically 16-bit, 24-bit or 32-bit float.
  • The amount of data transmitted per second, typically measured in kilobits per second (kbps)
Empty space, drag to resize

Myth: busted!

While we are busting myths, let’s look at DAW audio engines. Many DAWs boast a 64-bit float summing engine. Does it sound 'wider' or 'more analog' than 32-bit float?

In short: no!

32-bit float already provides a noise floor so low it could accurately measure the diameter of a proton. 64-bit float summing simply prevents mathematical rounding errors when stack-loads of tracks (several hundreds), heavy automation, and endless plugins are summed together. It’s about mathematical precision during complex calculations, not an audible 'audio quality' upgrade.

My Bit Depth recommendations, short and punchy:

1. Record in the highest resolution your converter hardware supports. For the vast majority of us that's 24bit, rarely 32bit. Avoid 16bit recordings where possible.

2. Some DAWs allow you to record in 32-bit float even if the converter is 24bit. I don't see any advantage here.

3. When bouncing a mix in the DAW, or when exchanging processed, files use 32-bit-float (not 24 bit). Every mastering engineer will happily receive a mix in 32-bit float.
Empty space, drag to resize

Coming soon:
The next chapter on Master Faders workflows, myths & analogue / digital facts.
Leave your email below to get notified.

Master Faders Demystified

An oldie, but a goodie: 
“Master faders sound bad in Pro Tools!”
or
“Your DAW master faders must remain at unity gain”
This old myth has been busted many times, but it keeps coming round like a boomerang. 
This is a prime example where old workflows need to be questioned and re-learned when technology moves forward. The origins of this myth lie in the analog mixing era, where it held true that the console's master fader should remain at unity (or as close to unity gain as reasonably possible). Only then would you get the best signal-to-noise ratio out of Neve consoles from the 70s. Or Harrison consoles from the 80s. Or SSL consoles from the 90s. 

Analogue mixes have the best signal-to-noise ratio with the master fader at 0dB. [Customseries 75 console, powered by Neve]

When the first digital mixers appeared, the summing stage had the same bit resolution as the ADCs. Consequently, the more signals were mixed together, the more one had to lower the channel faders to prevent overloading the summing bus. Music creators complained, technology companies took action.
Next, summing stages began working with a few more bits internally, allowing adjustments on the master fader instead of multiple channel faders. This approach preserved the mix balance but impacted the sum's resolution due to accumulating rounding errors and poor (or total lack of) dithering. Back then, I want to say in the 90s, leaving the digital master fader at unity gain actually resulted in better sound.
Once more, music creators grumbled, and technology companies pushed technology ahead.

Let’s jump forward to the mid-late nineties when Digidesign’s TDM mixer took over the world of professional audio, which it dominated for almost 20 years. It utilised 48bit linear summing (+8 bit overflow). That’s when I first witnessed a demonstration in which the “master faders sound bad” myth was officially busted without any reasonable doubt. 

Since then, we've witnessed the evolution to 64-bit-float summing with the introduction of HDX for big studios with the budget in 2011, and Pro Tools 11 for everyone else in 2013. Similar summing technology is found in most DAWs today. Null-tests have repeatedly demonstrated that the master fader has no negative impact on the sound, not even a single bit (pun intended!).

Deliberate attempts were made to overload the summing bus (a near-impossibility in 64-bit-float summing). To compensate, the master fader was set to an extremely low level, yet it did not affect the mix resolution. The output perfectly nulls against the source. 

Old cobwebs cleared, myth busted - fair and square.

While Pro Tools 11+ and some DAWs use 64-bit float summing, others use 32-bit float summing. For the master fader, it doesn't matter: even a 32-bit float summing engine has over 750dB of dynamic range and handles above-zero signals flawlessly. Whether your DAW sums at 32-bit or 64-bit float, the master fader acts as a transparent volume control. The output perfectly nulls against the source.

My master fader recommendations, short and punchy:

Every production must have a master fader. If you work with an analogue console, it'll have one built-in so you don't have to create one in your DAW. 
But if you sum signals in your DAW, a master fader is needed. Use it to gains-stage levels for your DAC.
Not every master fader works the same - experiment with master-inserts and work out if those are pre or post master-fader. Either way is fine, but you need to know to gain stage properly.
Eg, in Pro Tools, the master inserts are post master fader, so take this under account when moving the master fader as it can have unwanted effects on processors.
Empty space, drag to resize

Coming soon:
The next chapter on Recording Levels will be published soon.
Leave your email below to get notified.

Digital Recording Levels Demystified

I’m sure you’ve heard of 
“Always record as hot as possible without clipping”
If you asked why that is, chances are the explanation was that you use the full bit resolution only when maxing out the signal to 0dBFS. This is generally true, but I’d like to clarify that the 24th bit is in action when the amplitude reaches between -6dBFS and 0dBFS on the digital peak meter.
Personally, I find the word “Always” in the statement problematic. Today, this workflow is more of a personal choice, rather than a requirement - and blindly following this workflow can have negative side-effects in certain situations.  

This practice originated in the mid-80s, in the early days of digital recorders. Those devices were jittery and poorly dithered by today’s standards (if dithered at all). With these grainy-sounding 12-bit or 16-bit converters, preserving every bit of resolution was paramount.
Fast forward some 30 years, and today’s converters don’t suffer from the same troubles anymore. The technology has long gone through its teething stages, and early hiccups have been addressed. All modern converters I've had the pleasure of working with deliver the same clean and transparent results at all reasonable recording levels - hot, and also a little cooler. 
However, what still makes a difference with hotter or cooler gain staging is how analog front-end components behave, such as consoles, external preamps, and analog compressors - particularly vintage gear! 
These devices typically operate at +4dBU line level (aka. 1.228VRMS or 0VU). Also known as Studio Operating Level (S.O.L.), it’s the amount of voltage manufacturers designed line-level gear towards. Some call it the voltage sweet-spot, and for good reason.

Studio Operating Levels

  • 0VU
  • =  1.228V(RMS)   =
  • +4dBU

In 1942, the Acoustical Society of America officially defined the Studio Operating Level standard we still use today

Above that point, analog gear operates in its headroom, a few additional dBs above +4dBU. The amount of headroom available depends on the specific gear you’re using. Some cheaper devices offer only 8-10dBs of clean headroom, while others provide considerably more, varying from device to device. 
So, what dBU input value causes 0dBFS in the digital domain? Well, it depends. For instance, the Apogee Symphony II and UAD Apollo X8P all reach their maximum with an input of +24dBU, while other converters may do so at +22dBU or even as low as +16dBU. Some converters are user adjustable. Let’s stick with the first 2 for a minute.

ADC max line level input specifications in comparison

If you aim to peak at 0dBFS, your analogue front-end needs to generate a peak of +24dBU at its outputs, which is 20dB above the intended operating level. Some gear can handle this with ease (modern high-end gear, also most preamps built into converters,) while others may start to distort (vintage valve gear, I’m looking at you!).
It's worth noting that we're discussing peak levels here, and the signal's RMS will be lower. So, the signal’s crest factor also plays a role here, too! 
The key takeaway is that not all preamps and analog compressors deliver their best performance when driven high up in the headroom. Is the extra bit resolution worth some headroom distortion? Make an informed choice, know your gear, find out where your front-end’s strengths and weaknesses are. Record hot where it’s creatively appropriate. 
But please don’t “always” do so because of some dusty old 90s workflow. 

Klanghelm's superb MJUC: Analogue emulation plugins have SOL built-in

And if you think this only applies to analogue hardware, think again. Most high-end analog emulation plugins (like Universal Audio, Waves, or Soundtoys) are modelled to expect that same traditional -18 or -20 dBFS average input level, and levels above that can lead to in-built distortion. Slamming them with a hot digital signal close to 0 dBFS can cause them to clip internally. Creatively, that can be a desired choice to make - but it's probably not a good idea every time. Analogue modelled plugins can reduce the gigantic transparent dynamic range of your DAW by reintroducing analogue headroom characteristics. 
And therefore reintroducing an analogue dynamic range "problem" that was solved by digital audio technology a long time ago. Talking about coming 'round in circles.
But so what, if it sounds good it is good. Don’t overthink it when you’re mixing.
But if you have spare time, test your plugins to figure out how they behave when driven with normal signal levels (eg near -20dBFS RMS), as well as hot input levels (eg 10 or 15 dBs louder). Don't forget compensating at your DAW's master fader when doing so to avoid DAC clipping. You probably find some of your plugins are "transparent" and behave equally when driven with normal levels, or driven with hot input signals. While others may sound transparent when driven low, and add crunchy "character" when driven hot.
Knowing which of your plugins belong into the "transparent" category, and which ones belong into the "character" bucket can be a valuable asset when mixing in a DAW.

My recording level recommendations, short and punchy:

In Pro Tools, I set my meter ballistics to RMS + peak, then aim for approximately -20dB RMS. This allows 20 dB headroom for the peaks to fall where they fall - which is usually more than enough.
From there, I selectively give important signals a small nudge up, and less important signals a nudge down.
At the end of the recording, I have a well-balanced rough mix with all DAW faders at unity.
Empty space, drag to resize

Coming soon:
The next chapter on Converter Specifications will be published soon.
Leave your email below to get notified.

Converter Specs Demystified 

Sample rates, bit depth, signal to noise ratio… all these are common specifications and different converters can look pretty similar on paper. Some manufacturers even advertise their entry level products with catchy phrases such as 
“We use the same ADC chips as [insert expensive product here]”
But marketing like this might just be a bunch of hot air. I’ve listened to old 16 bit mastering converters that clearly sounded better than some other 24 bit converters - although the technical spec-sheets wouldn’t have suggested so. Good sound comes from a well tuned overall converter design, not from an individual chip or component.
Some of the often overlooked factors include the power-supply, which should provide the required current and stable DC, without fluctuations, ripples or noise. And most-importantly the accuracy of the converter’s word clock. The clock should be as sharp and stable as possible, and we’re talking fractions of nano-seconds here. If the word clock fluctuates even the slightest from the ideal interval, jitter occurs. Jitter can manifest itself in various forms, including random fluctuations, periodic deviations, or even sporadic spikes in timing accuracy.
These temporal jitter variations introduce irregularities in the sampling process, when analogue voltage values are captured slightly earlier or later. And since analogue voltages constantly rise and fall, the converter will capture a voltage value that’s no longer the true essence of the audio signal, thus compromising the accuracy with which the bit resolution captures the signal.

And while we are clearing out old cobwebs, remember that modern internal clocks are incredibly stable. The old 2000s workflow of buying an expensive external master clock to 'upgrade' a single interface's sound is outdated - modern converters always perform at their lowest jitter specs when running on their own internal clock. 

Empty space, drag to resize

Final Thoughts

If you made it to the end of this article, then I’d like to congratulate you and welcome you to the small and select circle of audio enthusiasts who take the time to understand their gear on a deep level. The world of digital audio is a complex and ever-evolving landscape that continues to fuel intense debates and discussions among professionals and enthusiasts alike. I encourage you to always keep searching for more information, and welcome the changes as this fast evolving technology will continue to rattle what we define as the best practice again and again. 
Originally written for PMFC website blog 19/06/2023
Updated and partially re-written 10/07/2026 

by Jan ‘Yarn’ Muths

Mastermind of mixartist.academy

Recording and mix engineer at mixartist.studio

previously:
Manager of Studio301Manufacturing,
Host of the Production Talk Podcast
Lecturer at SCU Lismore
Lecturer at SAE Byron Bay, Australia
Created with