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!

What is reverb and what is the difference between reverb and delay?

Well, to over simplify it (almost to the extent of being close to lie), you could see it as reverb being room reflections and delay being echoes.

Before we dive into the components. Let’s imagine a large hall. There is nothing in it, just you. Clap your hands, a single hard clap. The most likely thing that happens is that the sound from your clap bounces off the walls and comes back to you. Echo. The closer you stand to a wall, the faster the sound bounces back to you. The further away, the longer it takes. And the lounder your clap is, the sound is able to travel longer through the air. Because just like when you move around, the sound looses energy when moving through the air. The more energy (louder) the longer it travels. Which can easily be tested. Same hall, your clap echoes back to you, but if you whisper the word “Hello”, you will most likely not receive any echo back. But yell all you can, and the hall will reward you with a reply. The mechanics behind an echo is that the sound moves in a direction, hits a hard surface and bounces. Depending on the direction of the sound compared to the surface, it might bounce back to where it originated from, or in a completely different direction, kind of like a ball on a snooker-table. One surface gives one echo, several surfaces may generate several echoes. To get this effect, delay can be used, and depending on the settings of the delay, you can get the sound to bounce one or more times, with various timesettings for the echo to return. As the name suggests, the delay is moving the sound backwards in the timeline, delaying the sound. If you take a signal direct to the delay, and then into your mix, you just put the sound backwards in time, delaying it, but if you play the source in your mix, and simultaneously send it to a delay, it will be played in the mix first, and then return via the delay according to the settings on the delay.

Now, let’s make the same fictional experiment, but with reverb. Reverb is in a way just like delay, only that it is not only one reflection, it is multiple reflections from multiple directions. Imagine yourself in a church instead of a large hall. Plenty of hard flat surfaces close to you, all around you, even in the high roof. Now imaginary clap again. This time the sound travels from your hands in all directions and are getting bounced back towards you from many directions at the same time, both direct bounces and bounces of bounces where the sound has bounced of one surface, to the next, and perhaps a few more, before getting back to you. Now, you do not hear each individual bounce returning to you, but a collective bounce reaching your ears. The energy in the sound is also important here. A whisper might bounce back to you, but will not “stay in the air” for as long as the clap does. You will get a (non-mathematical) formula like this:

• Low sound close by – strong direct sound, some or no reflections
• Low sound further away – low direct sound, almost no or no reflections
• Strong sound close by – strong direct sound, strong reflections
• Strong sound further away – low direct sound, strong reflections

Please note that above statements are also oversimplifying hugely. Point being that a sound is a mix between its direct trajectory and the reflections. The closer you are to the source, the stronger the direct sound is. The stronger the sound is, the more reflections will hit you, and the lower the sound is, fewer reflections will hit you.

Now, why do I spend time writing about this?

Well, in my opinion, knowing this will help you place your sound in the mix. Moving it back and forth as you wish, create a large room or a small room, taking sounds to the same room or separating them into different rooms. Forging sounds together or dividing them. Like panning, but with several more dimensions.

All rooms have a ”reverb profile” with sound reflections. Unless it is designed specifically to not have any reflection at all. And most rooms are filled with stuff that effects the “reverb profile” of the room. The more stuff, the less reflections. Then there is designing a “reverb profile” (like cinemas, offices, shopping malls etc) where you actively put up different kinds of dampeners in well chosen places to reduce unwanted reflections.

This is something you should be observant about when you record something with a microphone. The closer the mic is to the sound source, the less of the room acoustics you get in your recorded sound signal, the further away, the more of the room acoustics you get. Here is nothing right or wrong, just be aware so you can make a conscious decision on how you want to use the signal you record. In some studios, the room acoustics are so good that you really want to have it with your recorded sound signal, and sometimes, it is highly unwanted to have extra room acoustics along with your sound signal. It all depends on how you want to use it and what you are after.

As always, stay curious and play around!

Music creation is very individual, as are the type of music that appeals to you and what music you find yourself disliking.

As for my view of music creation, it is a process that I need to go through. But it always contains some or all of the categories below:

• Rhythm – preferably complementing
• Tension between sounds
• Movement and direction – preferably opposite
• The baseline (not to be mistaken for the bassline)
• Harmonies
• Dynamics – intensifying and decreasing the energy
• Flow of energy and intentional focus
• Add or subtract? The perfect balance (not as in L/R)

Sounds easy and crystal clear?

Before you continue, let’s just introduce a term that I will use frequently in my explanations below. Element.

An element can be a sound, a melody, a rhythm, basically any component in the song or track that is a unique entity. For example, a melody played on a piano can be one element, and even if using the same piano sound playing chords, that could be another element. Drums could be seen as one element, or various elements, depending on how you want to use them. In my mind, there are no restrictions or strict definition as to what an element is, even so, I break down everything in my songs are divided into elements, one way or another.

Now, let me break it down.

Rhythm – preferably complementing 

Rhythm, what is it about? To me it is the position of the note or sound, in comparison to the other surrounding notes or sounds. If the beat goes 1 – 2 – 3 – 4, and each sound or note is put on either 1, 2, 3 or 4, it will both get crowded on those positions, and probably feel a little stiff. To me, it is important that each part of each element (see above explanation if you missed it) gets its own space within the rhythm, to separate the elements from each other and make use each element through out the track, it needs to (in my opinion) gets its own space in time. And rhythm is just placing things (elements) in (or perhaps on) a timeline. Now, naturally, there are multiple times in each track that elements will occupy the same space, it is inevitable (yes, you may take it as a challenge!) to do so, but not all the time, and to me it is important to intentionally create differences between elements rhythmically, to enjoy the benefits of each element and create a tension between them (read more about tension below).

Tension between sounds

To me, it’s important and essential to create tension in a track. Another way to put it is to create friction. One sound or element meeting and interacting with another sound or element. Creating magic between the two, this can be done in several ways, the two most obvious is of course by using harmony or rhythm. But it can also be by using filters (equalizer) making sure that the elements meet in the same frequency space, or meet with a deliberate distance between them. It can also be created by various FX’s, like putting two elements in the same room (by using reverb) or let them be in different rooms. Or let them bounce off of each other with delays perhaps?

Movement and direction – preferably opposite

An element probably has a direction, upwards, downwards, forward, backwards, inwards, towards the background our out in your face. If all elements move the same way, I find it to be hard to control the force of combined elements, they tend to go their own way and leave me a bystander, often feeling overrun by the force. So I’d like to take time to design the movements in opposite directions to get the track to move the way I want it to move (yes, I confess, I am a little control freak in this way). Occasionally it can be fun to let an element or two create unintended forces in directions I did not foresee, but if that happens, I tend to balance it with some other element, to keep it not spinning out of control.

The baseline (not to be mistaken for the bassline)

The baseline. As stated above, not to be confused with the bassline, or the drums, both which are often used as baseline. To me, the baseline of a track is the thing every other element uses to bounce off of, to get a point of reference so you can feel the movement and/or create tension. This can of course be done without a baseline, but to me it is much easier to have a baseline though out the track. Now, in all honesty, the baseline can change, but I tend to use the same element or combination of elements throughout a track.

Harmonies

Harmony. I mean this from two different perspectives. Harmony as in creating chords, happy or sad, and harmony as in coexist in harmony. To create harmony (chords) it is common to do it with one instrument, like piano or guitar, but it can also be created between different sounds playing one note each, building a harmony. Which ever way you want to create harmony (both perspectives) each individual note needs to coexist with the other individual notes, to form the unity of the harmony. This is true in the case where the harmony is created with a single instrument or with several instruments or sounds. And a harmony does not necessarily need to create a chord that is either happy or sad, it can create a balance between two notes, and a third element, let’s say a melody, can decide if the harmony is neutral, sad or happy. Or it can alternate and in one passage of the track create one harmony, and in another passage create a different harmony, with a little change in the melody. To put it blunt, you can use both C major and C minor in the same song, depending on what you want to communicate.

Dynamics – intensifying and decreasing the energy

Talking dynamics in music, most people tend to think about compression. But this is a completely different dynamic I am after. To create more energy in a track, it is easy to add element after element, to intensify the melody, or increase the number of notes played. It is as easy to decrease the energy by removing elements. My goal, as I often work with few elements is to get the increase and decrease within the elements and using the various elements to complement each other. Just adding a or subtracting one or a few elements to get the desired dynamic changes I want to communicate throughout the track.

Flow of energy and intentional focus

The entire dynamics of a track, that I addressed above, is to get a flow of energy through the track. And I also like to use this as an excuse to put intentional focus on a specific element. If I add a melody, the mind and ears are easily focused on the newly introduced element, moving the other already ongoing elements a bit to the back of the attention span. Not necessarily in the sound landscape, just the attention span. While one element is in focus, it is easy to remove one of the elements in “the back” to decrease the energy flow keeping focus on just the elements you want to use to build the particular part of the track.

Add or subtract? The perfect balance (not as in L/R)

This raises the question. Add or subtract? To me, just adding elements and stack ‘em on top of each other easily clutters the track. And I do not like clutter. I want to keep each track as clean as possible, giving space to each element and to keep it balanced. Each element I introduce in a track needs to have a purpose, needs to have a dedicated space and needs to add value. Both by introducing it in the track, and by removing it. Giving elements place in the sound landscape can be done in various ways, rhythmical, harmony wise, panning to left or right, filtering it (using equalizer, cutting frequencies out or emphasize a particular frequency). But it can also be accomplished by removing elements that occupy the same space on the timeline. Or, of course, by adjusting the volume with automations to the faders. Or in a million other ways, limited only by our imagination.

My point to all of the above is to intentionally introduce sounds and elements, and allow each element to have its own purpose and space in the track. Good luck in your music creation and remember to have fun and stay curious!