So, perhaps you’ve seen export settings for wave-files, or upload specifications on various distribution sites, the file needs to be 44100 Hz (or 44,1 kHz) and 16-bit.

What’s that all about?

Let me first state two things before you continue reading.

1. This is absolutely nothing that you need to understand, it is just surplus knowledge if you are interested.
2. As usual, I tend to over simplify, almost to the extent that it is on the limit on actually being true. In this case it is the concept I aim to explain, not the exact technical specification. I am sure you can find more exact specifications by using a little google-fu!

Let’s start with bit depth.

As you probably already know, sound is not actually a “wave” passing through the air, it is rather an energy movement through the air that pushes the atoms against each other in a chain reaction. This leads to parts in the air that are more dense followed by parts that are more thin.

In order to transform this movement through the air, we have two things at our disposal. One being our ears. The second being a microphone. The mechanics of the two are similar (but not identical). Both have a membrane to catch the movement in the air and transform it. Let’s focus on the mic.

The membrane is designed to physically move with the air, or rather, by the air. As the membrane is unaffected, it is still in its middle position (which we call the 0-position), and it can be moved (pushed) in and it can be moved (pulled) out. In the oversimplification spirit, let’s use the number I started with 16 (from 16-bit). This would translate into something like 16 possible positions to capture the current position of the membrane. Imagine 16 cameras. 8 mounted on each side of the membrane with the sole purpose of capturing an image of where the membrane physically are, in each given moment. Now, the cameras are not overlapping, which means that as the 16 cameras are triggered to capture an image, the membrane will only be seen in one of them, leaving the other 15 cameras with blank pictures.

The number 16 in the above example, is the depth of the ability to capture the movement of the membrane. Of course it is not “only” 16 “frames”, but the mechanics behind it. Should you decrease it to 8-bit, it would be half the numbers of cameras (fictive) with 4 cameras on each side of the membrane, while increasing it to 32 would (fictively) give 16 cameras on each side. Naturally, the larger the number, the more precision in capturing the exact position you will get. But. (Yes, there always seems to be a but!) You will also get more data generating larger files. And there is a sweet spot where the common man can’t hear the difference when increasing the bit depth. (However, it is debated if it is 16-bit like the “CD-standard” or if it is 32-bit. I’ll leave it up to you to decide your favourite.)

The other number above is 44100 Hz. This is literally 44100 movements during one seconds, where Hz stands for Hertz and is the unit used to measure cycles (or events occurring) per second. Let’s stick with events occurring per second. In our above example, this means that the cameras are triggered to capture images 44100 times each second. Another often used sampling frequency is 48 kHz or 48000 Hz. Which would translate to triggering the cameras to take pictures 48000 times each second. Again, it is debated which is “best” the “CD-standard” of 44100 Hz or the other commonly used 48000 Hz. Again I leave it up to you. But let’s ask the question, why 44100 or 48000? Wouldn’t 96000 times be better? Or could it be enough with 10000? Again, as for the higher number, the more times we “snap images” with all the cameras, the more data is gathered and the larger the file gets. And it is a sweet spot somewhere around 44100 or 48000 where the common man can no longer hear any difference to the captured sound.

But of course there is a reason for being above 40000. The human hearing range. The human ear can (at best) hear 20 000 Hz. That would be the equivalent of a very high pitched tone. To be able to capture 20000 Hz by using our cameras in above example, we need to take pictures at least 20000 times per second. But if we would (in theory) capture a “wave” passing the membrane 20000 times in one second, and we take pictures 20000 times during that second, we would see the membrane at the exact same position on each picture-stack. Which would register as no-movement at all. Doubling it to 40000 pictures per second would also (in theory) give a series of images showing the membrane in two positions only. Thus, by increasing the frequency of the image capturing, we can build a more precise “wave” from the captured pictures where we can actually see the movement as it is.

Rendering the “sound wave” (the graphical representation we can see on our screen) we take the number of cameras we have decided (the depth) and put a “dot” in the dedicated frame for the camera that captured the membrane, leaving all the other frames empty, and we repeat that for as many times (the sample frequency) we have decided, getting a graphical representation of where the membrane were at each given timeframe, and stacked after each other. If our sound is 1 second we would use 44100 sets of pictures from our 16 cameras. If our sound is one minute it is 60 times more information.

As I started with, this is both surplus knowledge, and oversimplified. But the basic principal is about as stated above. I hope it gives you something. If no, do like Elsa in Frost and “Let is go!”. 

I will write about three kinds of distortion.

• Analog distortion
• Digital distortion
• Intentional distortion

I want to kick this off by saying that distortion is nothing natural, it is manmade, and I believe it to be unintentional at first, but then put to use to sound signals to create a distinct sound. What distortion is, is simply put, a sound wave, that is naturally curved, but cut off and flattened in the tops. A sound signal that is captured and hits the limits of the technology used to capture it when the signal wants to go a little further, but the components have already hit its maximum limit. Which is the perfect cue to the first topic:

Analog distortion

With analog distortion I mean every distortion that is not specifically digital, which I will cover below. Analog distortion can occur as a physical limitation in microphones, headphones or speakers, but also as an electrical limitation in one or more components in the electrical chain that the sound signal passes.

Physical distortion appears when the sound or electrical signal wants to push the membrane further than physically possible. In a microphone this happens when the sound, typically a transient (a strong peak of energy in the sound) pushes the membrane of the mic to its maximum capacity, either inward or outward, and wants to keep pushing it. Like a rubber band, you can only stretch it so far before it snaps. The membrane of the microphone does not snap, but is not physically able to move further. This caused a “flat” top on the electrical signal as the membrane cannot provide any further electrical signal than its maximum. This is when recording the sound signal. When playing it back, the same physical limitations can appear in headphones or loud speakers. The electrical signal wanting to move the element past its physical limitation. The difference between the two states (recording or playing back) is that when the distortion happens when recording, it cannot be undone. When it appears when being played back, all you have to do is lower the volume (decrease the electrical signal) and the distortion will go away.

The same goes for electrical distortion, but instead of a physical limitation it is a limitation on what voltage the electrical signal can be, passing through various components along its path. Various components produce different kinds of distortion, like a tube, which provides a warm kind of distortion which is often desirable (when looking for distortion). Again, here is a difference if it happens when recording or when playing it back. Taking a typical path from a microphone, through a mixer, into a recording device, there are some steps where the signal might get distorted.

1. In the input gain of the channel where you connect the mic.
2. On the fade that controls the mic.
3. The output of the mixer.
4. The input (gain/level) on the recording device.

Now, it would be safe to assume that keeping the signal as low as possible on all these points would be wise to avoid distortion. And it is a correct assumption, but of course there is a ‘but’. That ‘but’ being the electric S/N ratio (Signal to noise ratio) of the components. All electric components have a kind of background noise that you cannot avoid, so you want to separate your desired signal as much as possible from this noise, keeping it as strong and loud as possible, of course without triggering distortion.

Digital distortion

As where analog distortion (where ever it may appear) can add a certain character to the sound, the digital distortion, however, does not. As where analog distortion may be wanted, the digital distortion is always unwanted (when recording or creating music). What it does is it clips the sound, which becomes very unpleasant to listen to when it comes out through the headphones or speakers. The phenomenan occurs when the audio signal (that is assigned a digital value) reaches the highest available value, and cannot get any higher. As when the analog signal is distorted it is still “soft around the edge” whereas the digital is absolute and sharp.

Intentional distortion

Last, but certainly not least, intentional distortion. As I have mentioned a few times above, distortion may not always be a bad thing, it can add character to the sound signal when used right. (And not only on screaming rock guitars!) The distortion can be used on everything, and there are a lot of FX that emulates distortion in various ways, if you work in a DAW (Digital Audio Workstation – or Sequencer as some people say). It is also widely implemented in various synths, weather hardware synth or software synth. (It is called “drive” sometimes!). If you have trouble getting a sound to “cut through” the other elements in your mix, distortion may be one way to get it to “pop out” of the mix, without necessary boosting the volume of it.

As always! Stay playful, creative and curious!

Wait, what? Is it different steps?

Yes, I would like to say that it is. And again, in below reasoning, I tend (as always) to over simplify things, just to make things really clear and distinct. And on top of that, the process look different to different people. So this is my view and my opinion that I share with you.

The production

To me, this step is including writing the song, recording individual elements and put them all together in the same audio project. Regardless if it is all software synths, with or without song or recorded instruments. It can be within a DAW (Digital Audio Workstation) like Cubase, Logic, FL Studio or reason, or it can be audio tracks in a porta studio or in a massive 48-channel recording device. My view on this step is to gather all the elements of the track, lay them as the should be played back, trim, tune, chop, add effects, compression, cut frequencies out (apply EQ or hi/lo-cut-filters), boost frequencies (apply EQ). The goal of this entire step is to have gathered all elements of the track in one place, and making sure that each element is as good as possible and is ready for use in the final track. This, to me, is all done in this step and while I tend to apply some mixing thoughts in this process, this is not where I do the actual mix, because that it the next step.

The mix

To me, the purpose of mixing is to place each element of the track where it is supposed to be (front, back, left right) by adjusting balance (L/R) and adjusting the faders (typically both faders and pan can be automated in a DAW, but it is not necessary – but perhaps desirable). When all elements are in their right place, it is time to “squeeze” them to fit together by starting to adjust parameters, this can be everything from faders and pan-knob to EQ, FX or compression. It is almost always a renegotiation of things and details you have already put your heart into for each individual element. But now your focus shifts from getting the best out of each of the individual elements to make them work together as a united sound mass, where it is no longer the individual parts that is in the spotlight, but rather the entire track and is combined overall expression. Like a detail in a painting can be allowed to be just a detail, while the over all focus is on something completely different, the detail is still a part of the painting, but the painting itself expresses far more than the detail, as it is the combined experience of everything in it. As we “see” the track with our ears, we are “seeing” the entire track, and the sum of all its parts. And it is this sum that we create in the mix.

Mastering

Is mastering necessary? And is it more to it than just putting a multiband compressor with a fancy present on the stereo output and be done with it? Let me first state that I am in no way a master on mastering. It is an artform that I have yet to explore and learn the craftmanship of. I have full respect for all those of you who are skilled in this art.

To me, as a mere beginner in this subject, mastering is a way to create a unity of several tracks, much in the same way mixing is to create a unity of individual sound elements. To get an album, EP, or two songs on a single to create a unity, they need to be brought together. While mixing one track is just focused on bringing the best out of that track, mastering tracks to bring out the best of the collected songs in a unity. Of course, it is easy to claim that each song is an individual creation and should be presented as such. I cannot disagree with that. But at the same time, I can also not look at mastering without seeing the whole picture of the tracks that are supposed to be presented together.

Perhaps, a little history lesson can put things in perspective.

Back in the old days, when music was released on vinyl (which is still popular today, in some genres) it was important to master songs to fit within the physical limits of the vinyl playback. A bass that was too loud could actually create so much vibrations so that the needle would jump to a different part of the track, which would disrupt the playback of the track. A very undesirable feature. The same could be said for CD’s even if it did not cause actual vibrations when the laser read the one’s and zero’s off of the CD itself. But still limited in a way.

Then there is the broadcast perspective… often “kinder” versions of songs are created, to adapt for the broadcasters conditions. Both radio and video broadcast through ether waves have their limitations and is handled differently by each station, but in general, the final touch of sound processing is put closest to the physical transmitter, leaving a smoother signal to send long distances from the studio to the location of the antenna. However, this is not something I want to go in depth with.

Nowadays, when a lot of music is produced in a computer and released to various streaming services and consumed through a computer or smartphone or any other similar device, it opens up for other challenges. For example, consumption patterns today are very different from putting the needle to a vinyl, go over and sit in a chair and listen to the entire first side, go back, flip the vinyl around and repeat on the other side. Today it is more likely that you find a song, add it to a playlist full of other songs. If one track is mixed and mastered very loud, and the next very quiet, the listening experience may not be pleasant and it is likely that you would want to adjust the volume between the two tracks. Various streaming services handle this in various way, but in my opinion, the “loudness war” from the early 2000 is not quite yet over. (Me being one of the one’s guilty of keeping it going!)

So, to put the stakes at the table, moderate and modesty could (or should) be a leading star when it comes to mastering. And this where I only theorize and present my thought, and act completely different. I use mastering as an extended step of mixing, to get the most out of each individual track. But I am still learning and trying to avoid the “loudness war”. (But it is oh, so tempting, and I constantly fall into the temptation of pushing the sound just a little further in the mastering process….)

And as always, stay playful and curious!