Lately, I’ve been mixing my new album (http://raskolnikovsdream.com). If you listen to my pieces you’ll see that I layer a lot of sound on top of one another. When you are mixing, the main focus is how to stack all these different sounds together. Sometimes, the sounds are all on the same frequency range, something that makes it quite difficult to mix them.

Enter panning. Panning is the location of the different elements on the stereo field. Back, in the days of yore, everything was mono. That means that there was only one channel for both left and right speakers. Stereo, seperates the signal into left and right. This creates a fuller sound, but the reason that stereo was created was not only to create a fuller sound, but to emulate a realistic live performance.

The next step after stereo is surround audio, but surround audio is something that is mostly reserved for audiophiles, since it is difficult to set up.

Surround panning in Cubase

Now, panning, as we said, concerns the placement of the audio elements in the stereo field from left to right. One of the biggest mistakes one can make is to place everything in the center. Let’s say for example that you are trying to mix a guitar with a vocal track and you place both in the center. The sound won’t be so clear. Now, try to move the guitar to the left. Suddenly, the vocals become clearer. This is what panning is for.

However, the stereo field has to be balanced. You should move audio tracks that share the same frequency spectrum equally to the left and to the right. So, for example, you should balance the guitar on the left, with a high hat on the right. Furthermore, you should know that everything placed in the center sounds more pronounced. So, you should leave the center for things like vocals, bass and kick drums. Vocals, because they are very important and bass and kick drum because low frequency sounds are not perceived very clearly by our ears and if you move them left or right they will mess the perception of the whole mix.

First of all, a compressor is a device that reduces the signal volume by a ration, once the signal exceeds a certain threshold. A typical compressor will have the following settings: Ratio, threshold, input gain, output gain, soft knee or hard knee, attack and release.

The threshold determines the Db above which the compressor takes action. The ratio is depicted in fractions such as 2:1 or 4:1. This means that for every 1 Db the signal surpasses the threshold, the compressor reduces it by 2 or 4 db respectively.

This diagram shows what happens. There is a certain threshold level. The signal enters and once it reaches the threshold, the compressor starts working. Α 1:1 ratio means that the signal enters and exits unaltered. An infinity:1 ratio means that the compressor works as a limiter. It doesn’t allow the signal to pass the threshold, no matter how loud it is. We’ll get to this in a while.

A compressor also has a knee setting. This determines whether the compressor starts working in an abrupt or soft way. When the knee is hard, the compression is more apparent in the sound.

The attack of a compressor determines how fast a compressor starts working. When this timing is very fast, for example 10 msec, then the compressor starts working immediately and the whole signal is compressed. However, when we choose a greater attack setting, such as 50 msec, we let the transient pass. That is the first part of the signal that hold no harmonic elements, but makes the sound more "punchy". Release, determines how quickly the compressor goes off.

A few lines above we mentioned the term "limiter". As we said, a limiter is a compressor with a ratio equal to infinity:1, which means that it doesn’t allow anything to pass the threshold. A compressor is useful at the mastering stage. It is actually the very last thing in the mastering chain. After all the process has taken place, we pass the signal through a good brickwall limiter, in order to make the signal louder, while ensuring that nothing exceed the threshold and causes distortion.

PSP Xenon is a very good software limiter

Of course, this doesn’t mean that we can go as loud as we want without distortion. A good brickwall limiter can add character to the sound and go louder than another one, but, after a point, everything distorts. Limiters are an entire different topic altogether, but know that they shouldn’t be used in mixing. If you are mixing and then you are passing your signal through limiters, then expect the quality of your sound to quickly deteriorate.

First of all, we must understand that equalizers are filters. Their job is to attenuate or amplify frequencies.

Note that in this tutorial we will deal with graphic equalizers, which are the most common VST equalizers. There are other types, especially in hardware forms, but we will not deal with them.

An example can be show in the parametric equalizer in the picture.

Hardware parametric equalizer

Graphic Equalizers allow you to shape the eq curve in a visual way. Most VST equalizers are graphic, even though there are analog simulations which try to emulate the "real" thing. Waves is known for its various emulations of analog gear.

The API 560

The most basic types of filters found in graphic EQs are the following:

High-pass and low-pass

Shelving Filters

Parametric

Notch 

A low-pass and a high pass filter look like that

Low-pass filter

High-pass filter

Low-pass and high-pass filter, are pretty obviously, named like that, because they allow only high or low frequencies in. There are many reasons you’d want that, and usually, when you use an equalizer for mixing purposes, you’ll have one of those in your setting. High-pass filters are used very commonly when you need to mix any other instrument besides a bass, in order to eliminate all the low frequencies, so that you can leave space for the bass. Furthermore, high-pass filters are used to attenuate all frequencies below 30-40 Hz, because they can be a source of clipping and noise in your mix, while they don’t contain any musical material (since so low frequencies can’t be heard very well by the human ear).

Pretty much the opposite applies for low pass filters, even though they are not as common as low-pass filters. Something we must mention for low-pass filters, is that they are used as anti-aliasing filters in the sampling procedure (something which will probably will never bother you ) 

High-shelving and low-shelving filters are called like that, because they create "shelves", as they can be shown in the pictures below. They’re role is to attenuate frequencies, without eliminating them completely.

High-Shelving Filter

High-Shelving Filter

Parametric filters are used to attenuate or amplify a set of frequencies. They are used to "color" instruments in mixes and masters. By attenuating or increasing certain frequecies, you can change the sound in subtle, yet productive, ways.

However, there are times when you’ll meet certain problems in your mix, like pops and glithces from a bad recording, that lie in a very specific range. In that case, in order to remove completely the problem frequencies, you must use a notch filter. These are filters with extremely narrow bandwidth, in order to minimize intereference with nearby frequencies.

Notch Filter

So, that was it! I hope that this little tutorial was some help to you!

First we will create a sine wave

Now, let’s take a look at its harmonic content

Now, let me help you understand what this thing shows. The yellow lines, above the middle, are the harmonics and their amplitude. Since all instruments use the same tuning (equal temperament tuning), which means that each note is a standard frequency, the harmonic series is always the same. If you remember from the previous article (A few things about audio waveforms and fourier-transform) harmonics are different from each organ. The harmonic series however, is always the same. If f is a frequency of a note in our tuning system, then the harmonic series is 2f, 3f, 4f, etc. What is different, is the amplitude that each harmonic has in each instrument.

In this scenario, we only have the first harmonic, the fundamental, which gives the basic note. The above picture from Absynth, tells us, that should we press a note, like for example C1, we will hear only the fundamental note, and in full amplitude.

The blue lines below the mid-line represent the phase of the harmonic and we will not occupy ourselves for now with this.

So, let’s create some other wave. We will create a square wave.

Now, let’s see its harmonic content.

Now, what we see here, is that in a square wave there are only odd numbered harmonics. That is, the fundamental, the 3rd, the 5th, the 7th, etc. The loudest is the fundamental.

Now let’s play around with the harmonics to see what comes out.

Let’s see what graphic wave function comes out of this

So, in this tutorial we saw, how we can use Absynth’s functions to draw waveforms to understand the relationship between harmonic content and wave. This is a powerful tool (especially for learning),but don’t think that you can really create the waveform you have in mind, in just 5 minutes. The key here is experimentation. You just have to play around to find something you like. The only rule that there is, is that odd harmonics sound harsh, even harmonics sound melodic. Other, than that, the only way to learn is to experiment and learn

Today we are going to talk about a topic that causes great confusion to beginning computer musicians: synthesizers and their components

When you start learning about this stuff you’ll eventually have to choose your first synthesizer, be it virtual or physical. Most of the music you’re going to write is based on a some type of synthesis, even if the instrument is not considered an actual synthesizer. Additionaly, it is a good idea to choose a synthesizer, even if you are planning to use only audio instruments. All synthesizers share some common traits which are fundamental in digital audio. By studying a synthesizer you actually get grip of what is going on with digital audio and you’ll gain knowledge that you’ll be able to transfer to other domains.

In this article, we will discuss about the basic types of synthesis. When I started learning about these things, and as I had to choose a synthesizer to start writing music, I was utterly confused. However, as my experience increased, I quickly understood that things are much simpler than it seems.

First of all, every kind of synthesis is based on an oscillator. An oscillator is anything that produces a repetitive signal, that is actually a wave.

The number of oscillators in a synthesizer can be an important factor in shaping the sound. Most synthesizers use 2 or more oscillators. The way that the oscillator produces sound depends on the type of synthesis it follows. There are many types of synthesis and we should cover them in another article, but here are some

• Subtractive Synthesis
• FM Synthesis
• Granular Synthesis
• Additive Synthesis
• Wavetable Synthesis
• Physical Modelling

Subtractive synthesis for example, which was one of the first kinds of synthesis, takes a certain waveform and subtracts harmonic content via filter, while additive synthesis adds waveforms together to create additional harmonic content.

Another component that you’ll meet in synthesizers are filters. Filters can have many forms and shapes (that will cover in another article). The purpose of every filter is to attenuate frequencies (well, actually filters can make some frequencies sound louder through resonance, but this is another subject). Filters are used to shape the sound in various way. A very common use, is to pass the signal through a low-pass or high-pass filter to give the sound a certain "frequency range". However, there are synthesizers like Native Instruments’ Massive, that uses some artistic filters, such as a scream filter, or an all-pass filter that reverses the phase of the signal.

A typical low-pass filter

Synthesizers also use to have effects. Built-in effects usually represent versions of effects you can find in many other VSTs, from various companies, like reverb and delay. However, some like Native Instruments’ Absynth, have effects with a certain artistic taste. Absynth 4 has only 4 effects, but each one is special. However, when you are choosing a synthesizer don’t get carried away with the preset sounds that are full of effects. Many VSTs for example can have in their demo versions, sounds that sound too impressive because of too much reverb. Always remember that you can add your own effects later. So, don’t choose a synthesizer based solely on effects (and demo presets ). On the other hand, some VSTs can have effects which can have their own sound, which is a good thing. Spectrasonic’s Omnisphere comes to mind and its various versions of distortion and compression.

Another very important component of synthesizers are modulation components. These are sources that are used to control other components in the synthesizer. The most common are LFOs and envelopes. LFO stands for Low Frequency Oscillator. It was once, when the circuits were still analog, an oscillator (that is a waveform) of low frequency, that was used as a modulator. Now, the oscillator doesn’t have to be of low frequency, but the basic principle is still the same. It is a waveform used to modulate components.

An ADSR envelope

Envelopes are modulators that define a certain progress of a value through time. The most common form is an ADSR envelope, which has the four basic envelope attributes: Attack, Decay, Sustain, Release. Envelopes are mostly used to control the amplitude of a note that is played. Attack defines the time it takes for the sound to reach its maximum amplitude value. Then decay determines the time it takes to reach the sustain. Sustain does not represent time, but an amplitude value. Release, determined the fade out of the note.

So for example, for piano, we’d have an attack of zero, a medium sustain and a somewhat big release time.

However, envelopes and LFOs are not the only modulation sources. Other synthesizers allow more modulator sources with various connections amon them.

So, these are the basic controls you’ll met in a synthesizer. Others might include an arpegiator, which is used excessively in trance to create arpegios, a trance gate, or other controls, like an analog knob, that adds an analog flavor to the signal.

In the last post, we talked about audio. Now, we’re going to talk about the brother of audio in writing music: MIDI

MIDI stands for Musical Instrument Digital interface. It is a protocol defined in 1983.

What it is very important to understand, is that MIDI is not sound. Midi represents events. All vst instruments use MIDI and many people are confused, believing that MIDI is actually some kind of a sound generator. MIDI simply transfers information about events like pitch, velocity and duration of a note.

So, for example, when you use a MIDI keyboard to play a vst instrument, and you press C1, the MIDI keyboard, just sends the information that this note has just been pressed.

What is so useful about MIDI is that it is an industry standard protocol that allows all sorts of devices to communicate with each other. Before the advent of MIDI, different devices followed different configurations. Now, it is possible to write something solely on sythesizer, and then transfer it into your DAW to use other sounds you like. Of course you can do the opposite. You can send MIDI messages from your DAW to another source. These messages don’t have to be sounds, they can be triggers for all sorts of events. MIDI cables are usually used in an in-out configuration, but there’s also the midi thru configuration, which tranfers directly the data from the MIDI in to another source, without a delay, something that is usually a problem in larger rigs. You’ll probably never have to deal with this issue as a computer musician .

MIDI was also, once, promoted as means to transfer songs, but with the coming of faster connections on the internet it has lost this function. However, if you search, you can find collections of MIDI songs on the internet, since they still have their uses. You can, for example, open your MIDI file in Cubase and read it like a score, with its score editor, if you are interested in analyzing songs.

In order to maximize compatibility, in 1991, GM (General Midi) was created. This was a protocol that defined, among other things, like polyphony, which program number was which sound. Before that, a synthesizer could use program 1, for example, for a strings sound. With GM, there was a certain spot for each instrument, defining, for example, that program 17 is a drawbar organ. Thus, you could be sure, that a midi song, was played in another synthesizer, with the sounds it was meant to be played.

Now, all the vst instruments use MIDI and they incorporate it in various ways. Drum simulators for example, have "velocity layers", which means, that they read the velocity of the note played and they play a different sample every time. Thus for example, instead of just sampling a snare drum and playing it louder if the velocity is louder, these VSTs trigger a different sample, that was recorded was the snare hit harder, thus producing a more natural sound.

A typical MIDI connection amongst various synthesizers

The most important MIDI parameters you’re going to occupy yourself when you write your music, is velocity (which we refered to above), which defines how loud a note is played. There is also aftertouch, which is extra pressure applied after you press a key and sustain and can be useful is some vst instruments such as Akoustik Piano. Pitch bend can also be used, when you’re trying to emulate a guitar, or play in exotic tunings. Most of the other MIDI commands are used rarely.

When you enter the world of computer music, you have to choose one MIDI-keyboard. Something that you will see concerning MIDI keyboards, is that they have many knobs and faders. One more thing you can use MIDI for, is to "map" (as we say), that is to connect, these faders and knobs, to functions within your DAW. You can connect your faders to the volume faders in your DAW mixer, and your knobs to knobs of your VST instruments, thus controlling every parameter of your DAW from your keyboard. Ableton’s Live, has, probably, the finest MIDI mapping functions. Since it’s role is to be used in live situations, you can map almost every parameter of the program to your MIDI keyboard.

A MIDI keyboard

When you choose a MIDI keyboard, do yourself a favor and buy one with 5 octaves. Don’t be cheap and buy one with two, because you’re going to regret it later, unless, you are TRULY limited in space and money. Furthermore, some keyboards offer weighted keys, which means that the feel of the play resembles that of a piano. They are easier to play if you’re a pianist, but they’re not such a great deal, even if they are generally easier to play than the rest.

When you write using MIDI, you’ll have to be accustomed to the idea of a piano roll. Even if you record everything via a MIDI keyboard, there will be times when you’ll need to edit your work. Piano rolls can be a little tiresome, because you see the notes as in a piano, but in a vertical manner, but this is a discomfort you’ll just have to handle.

A typical piano roll

So, that was it for today! MIDI is very simple, and there are really not many things to know. You just plug a cable and then play However, it won’t hurt you to know a few things about its history and its basic ideas behind its creation. Until next time, just keep playing!

In this post we will talk about some of the first things one should know before he starts making music on a computer: what is audio? We will try to present the fundamentals of audio, without delving into much detail, so that we don’t cause too much confusion.

Sound is nothing more than a wave transmitted through the air. There are two ways to capture this and reproduce it: the analog way and the digital way.

Analog encoding works by using a property as an "analog" of the properties of sound. We can use for example the physical texture of a phonograph or the magnetic fluctuations of a tape. Analog recording are continuous.

Digital recording are seperated in discrete steps. These steps are codified by other measures when we capture the waveform, and then they are decoded when we play it back.

In digital audio we have two properties: Sample rate and Bit Depth.

Sample rate determines the highest frequency that can be reproduced.

Bit depth determines dynamic range.

A digitally sampled waveform

The sample rate we use for CDs is 44100 Hz (see Herz on wikipedia). What this means is that when we record something, like for example an electric guitar plugged directly into our audio interface, or an acoustic guitar through a microphone, we take 44100 "pieces" of the sound per second. Sampling is not a notion that is limited to audio. Sampling also occurs, for example, in statistics, and digital audio theory has a lot to do with statistics . When a statistician chooses from a given population of 10.000 a sample of 1.000 to give them a questionnaire, he is trying to make viable claims for the whole population (of 10.000 people) from these 1.000.

The same happens with audio capturing. Sound is continuous. We can’t seperate it in discrete steps. However, when we sample a sound source, we take discrete steps, that we use to recreate the sound we heard.

Digital sampling occurs in discrete, quantized, steps as shown above

The human ear can hear sounds ranging from 20 hz to 20.000Hz. According to the Nyquist-Shannon sampling theorem, in order to reproduce a waveform, we need at least twice the sampling rate of the highest frequency of the waveform. This means, that since we need to have frequencies up to 20.000 that is the threshold of hearing, then, we need at least a sampling rate of 40.000 Hz.

However, there is an issue called aliasing, which causes frequencies that exceed the Nyquist frequency to appear as "aliases" of the original sound, causing inharmonic digital distortion. The subject of aliasing is something that will be covered in another article and is not so important for now, but I just mention it to get to my next point.

CDs use 44100 and not 40.000 sampling rate. The reason for this, is that in order to avoid aliasing, we use filters that attenuate the higher frequencies, above the threshold of human hearing. Due to the nature of filters, it is impossible to create a filter with a slope of 90 degrees (like a fall) that starts at 40.000 and stops at 40.001. Instead, filters are like slopes. The sampling rate of 44.100 was also used because of technical limitations of the time that the CD medium was first created.

 A typical low-pass filter with a typical slope

So, let’s get now to bit depth.

Bit depth determines dynamic range. This means three things. First, we can capture a higher range of harmonics, or other sounds, coming from the sound source, that would, otherwise, be too quiet to capture. Secondly, our signal becomes louder. 1 bit approximates to 6 extra db of dynamic range. Thirdly, all things in nature (audio included) have a certani degree of noise. Higher dynamic range means that the noise floor, can be much lower than the sound source. For example, if you have a dynamic range of 6 db, then, if you capture a guitar but you have a noise floor of 3 db, you lose much of its clarity. However, if you have 96 db of dynamic range, it becomes much easier to get the guitar much louder than the noise floor.

A better example to demonstrate this is through digital pictures.

In the picture tou see the immense difference between, 1 and 24 bits. As you add more bits, the picture gets clearer, but only in the 24-bit case, where we have 16 million colors we have a clear representation of the picture. The same thing goes with audio.

So, in this post we explained the two most fundamental properties of digital audio: sample rate and bit depth. To recapitulate what we just said, sample rate determines the highest frequency that can be reproduced, while bit depth determined dynamic range which translates into clarity and loudness.

CD format uses 44100 sample rate, which can surpasses the human threshold of hearing and a bit depth of 16 bits that provides 96 db of dynamic range that translates to 96 db of dynamic range.

You may have found some notions you didn’t understand, such as aliasing, loudness or decibel. We will cover these on next posts so stay tuned!

Further Reading: