Which is better, analog or digital audio? By the end of this video,
you'll understand the most important differences so that you can decide for
yourself. But if this is our first time meeting, my name is Kyle. Welcome to Audio University.
Let's start by taking a look at the most common analog audio formats - tape
and vinyl. It'll help if we have a basic understanding of how these formats work.
Analog tape is the highest quality option for analog recording and playback. A tape
machine allows us to take waves of alternating electrical current from a microphone and store
them as charges along magnetic tape. When you play that magnetized tape back through a tape machine,
it's converted back into electrical currents that can then be played through a speaker.
In an effort to distribute audio on a larger scale, recordings can be copied from analog
tape onto vinyl records. While the very first recordings were recorded directly to wax disks
or cylinders, vinyl is only a method of playback and distribution. Physical grooves are impressed
upon the discs, creating a track for a turntable stylus to run through. The tiny movements of
the stylus as the disc turns are converted into electrical signals that are then sent to speakers.
Although the sound quality of vinyl isn't as good as the original tape,
vinyl is easier to mass produce, it takes up less space, and it's more durable.
Of course, most modern music is recorded using digital technology. Let's look at the basic
principles of digital audio so that we can better understand how it compares to analog.
Almost all digital audio files operate on pulse code modulation or PCM. There are several
parameters of PCM that can affect sound quality, such as bit depth and sample rate. However,
the general idea is the same across the board. With digital audio, instead of storing the signals
from a microphone onto magnetic tape, we store them onto a hard drive. Using a series of values,
we can encode a model of the signal that can be played back through speakers later.
In the early days of the Internet and portable music players like the iPod, the full uncompressed
PCM data was just too large to be streamed over the Internet or stored onto small hard drives.
In an attempt to make file sizes smaller, we used data compression. You can learn all about
data compression in another Audio University post. I'll leave a link to that post in the description
below. Currently, we're starting to see more and more streaming platforms offer lossless audio
streaming, which means that there's no quality loss compared to the original uncompressed
version. However, over the last few decades, most platforms have only offered lossy audio, which can
drastically reduce sound quality. Unfortunately, this has given digital audio a bad reputation.
But stay with me and you'll see the digital audio isn't as bad as you might believe.
As you can see, even though analog and digital might use different methods,
the goal is exactly the same. We're trying to preserve the electrical signals from a microphone
and distribute them so that they can be recreated later through speakers.
And honestly, both formats accomplish this goal incredibly well. But there
are some details that we need to consider when deciding which is better for the task at hand.
One of the biggest differences between analog and digital audio is the noise level. You
probably are already familiar with the concept of a noise floor and a signal-to-noise ratio.
Beneath the audio signal, whether it be music, sound effects, or spoken word,
there's noise. Even if you eliminate all of the noise in the room,
you'll still be left with the inherent noise of your recording medium and this is where analog
audio really falls short of digital audio. The inherent noise of both analog tape and vinyl
is significantly greater than the inherent noise of a typical digital audio system. I'm sure you've
heard the sound of tape hiss on an old recording or the clicks and pops of an old vinyl record.
Don't get me wrong - this can add a nostalgic element to the listening experience. But let's
be honest. Wouldn't it be better if there were no noise at all? Digital audio has a noise floor, but
it's extremely low. The theoretical noise floor of a 24-bit digital recording is -144 dBFS, which
is practically non-existent. When it comes to noise, there is no question. Digital audio wins.
Secondly let's compare the signal fidelity between analog and digital audio. We first
need to define fidelity. Fidelity can be defined as "the degree of exactness with which something
is copied or reproduced". So the question here is, "Which format can capture, store,
and recreate the most accurate representation of the original input signal?". I've heard many
justifications as to why analog audio is better than digital audio and over time I've learned
that most of these justifications are grounded in a misunderstanding of how digital audio works.
Okay, this is where I might upset some people, but remember - I'm just the messenger
attempting to lay out the facts to the best of my ability. One of the most common misconceptions has
to do with the resolution of a digital audio waveform. In fact, I used to believe this one,
myself. The misconception is that an analog to digital converter is incapable of creating a
perfectly accurate waveform and the logic behind this belief seems valid at first glance. If a
digital audio system only takes a finite number of snapshots or samples every second and each sample
can only represent a finite number of values, how could it possibly recreate the infinitely smooth
waveform of the original sound wave? To understand the answer, we can look to the Nyquist theorem.
The Nyquist theorem states that a waveform can be captured and reproduced with no loss, so long as
the waveform is sampled at a rate beyond twice per cycle. Therefore, the sample rate of an audio file
determines the highest frequency that can be sampled with no loss. In theory,
an audio file with a 48 kHz sample rate is capable of perfectly recording and reproducing frequencies
up to just below 24 kHz - half of the sample rate. Given that the typical range of human hearing
and the low pass filters within microphones and most other audio equipment generally top out
at 20 kHz, we can safely say that a sample rate of 48 kHz provides adequate frequency
range. Taking a close look at the samples within the DAW, we can see that they are represented as
a series of dots connected by straight lines. This representation can be a bit misleading.
While it's true that the samples are just a series of dots that represent the position of
the waveform at each passing moment in time, the resulting waveform wouldn't look jagged like this
one. The truth is that the waveforms playing back through the digital-to-analog converter
will be identical to the original waveforms recorded by the analog-to-digital converter. So
long as the highest frequency doesn't surpass the Nyquist frequency, there's only one possible path
that the waveform can take. Now, I'll be honest here. I still have trouble fully wrapping my mind
around this part. That's why I want to recommend a video by Monty Montgomery on xiph.org. It's free
to watch and it's widely regarded as one of the best videos for debunking the myth that analog
audio is somehow smoother or higher resolution than digital audio. I find myself re-watching
this video every year and each time I watch it, I catch on to things that didn't make sense to
me before. So don't feel bad if you don't fully understand everything your first time watching.
There is another drawback to digital audio that's also important to mention, called aliasing. It
also has to do with the Nyquist theorem. If the analog-to-digital converter attempts to sample a
frequency that exceeds the Nyquist frequency, it'll result in aliasing. On a basic level,
aliasing describes a situation where the A-to-D converter confuses a high frequency for a much
lower frequency due to sample rate restrictions. In an effort to mitigate this problem,
engineers will apply low-pass filters to the signal chain which prevent frequencies that exceed
the Nyquist frequency from being quantized by the system. For this reason, we don't use a sample
rate of 40 kHz to capture a 20 kHz signal, but instead use sample rates like 48 kHz or beyond.
This is another area that can become very complicated the deeper you dive into it,
so I'll point you toward another video that explains the concept extremely well.
The video is by Dan Worrall and it can be found on the FabFilter YouTube channel. If you've ever
wondered how high you should set your sample rate, you're going to find that video very enlightening.
Okay. Aside from noise, the signal fidelity of analog audio versus digital audio
seems pretty close. But what is it about analog audio that makes it sound warmer
or better to many people? It can probably be boiled down to the distortion and noise that
comes along with using analog equipment. As we already discussed, each time you pass a signal
from one device to another the inherent noise from each device becomes a part of the signal.
Not only that, but analog circuitry will also add harmonic distortion to the signal which
isn't always a bad thing. In terms of fidelity, this is another win for digital audio because
the signal can be processed with digital signal processing, which doesn't really suffer from the
noise and distortion that you get through analog processing. So if you're going for transparency,
digital audio is superior to analog audio. But, like I said before, the harmonic distortion
that's associated with analog audio isn't necessarily a bad thing. Sometimes you're not
going for a transparent sound and the way analog equipment changes the signal might be desirable.
As you approach the limits of analog circuitry or tape, the quality of the signal will start
to change. That non-linear response to signals is something that can be modeled by digital systems,
but it's built right into analog systems. Digital systems will behave the same no
matter where the signal level is in relation to the limitations, until the signal actually
exceeds those limitations. When the signal level in a digital system exceeds 0 dBFS,
it immediately causes clipping, which sounds terrible 99% of the time. I'm going to classify
this detail as a win for both analog and digital. On one hand, it's nice to work within the linear
and transparent digital environment for achieving the best possible fidelity, but on the other hand
the tones you can achieve with skillful gain structure in the analog realm can also be great.
While we're on the subject of the differences in the production process between analog
and digital audio, there are some things worth considering. If you've been working
in the audio production industry for over 25 years, you know where I'm going with this.
But if you're relatively new to the industry, this might not be so obvious.
We already discussed the general process of analog and digital recording. In analog,
the signals are stored to magnetic tape and played back using a tape machine
and in digital, the signals are stored to a hard drive and played back using a computer.
Aside from the basic recording and playback process, there are a lot of differences between
analog and digital music production. One of the biggest differences is the editing process. Listen
closely here because the benefit of digital audio cannot be overstated when it comes to editing.
In a digital audio workstation, or a DAW, we can make cuts and adjustments to the clip without
any worry of making an irreversible mistake. In the non-destructive digital editing environment,
we can always just press the 'Undo' button to get back to where we left off. Analog edits
must be performed with a razor blade and adhesive tape to physically cut the tape and reassemble it.
If you mess up in analog editing, there's a chance that there won't be a way to salvage your work.
Not only that, but there are virtually no limitations to the number of tracks that can be
stacked in digital production. These advancements in technology have completely revolutionized
not only music production but music itself. Think about it - the ability to make precise
edits to multitrack recordings and record as many tracks as you want with no negative consequences.
Once the production process is complete, we still need a way of distributing that production to
millions of listeners. This is easily the biggest difference between analog and digital audio. The
best method for distributing analog audio is vinyl. Tape is bulky, in addition to being more
difficult to reproduce and maintain. However, vinyl has its share of disadvantages, too.
First, vinyl is a physical medium, which means that each copy is costly to produce
and needs to be physically shipped to listeners. Secondly, vinyl changes over time.
Not only is there a quality loss when the master tape is printed to vinyl,
but every time you listen to a vinyl record the quality degrades even further. Today, digital
audio can be streamed over the Internet with no loss. You can instantly distribute infinite copies
across the world with no quality loss! Plus, when you compare the amount of space required
for a thousand songs on vinyl versus a thousand songs on a hard drive, there's just no contest.
Many of these details are relatively insignificant when it comes to sound quality. If you really want
the best sound quality possible, I'd recommend focusing on more important things like setting
up your speaker placement properly. This will make a huge difference and it's completely free.
In fact, I want to give you a free gift for making it this far in the video.
In the description below this video, you'll find a link to a free speaker
placement guide that will help you ensure that your system is set up for optimal performance.
If you found the video helpful,
hit the "Like" button and share it with your friends. I'll see you in the next one.