Musikality Net

In this article, we will use Absynth to understand what harmonics and waves are all about. Absynth allows you to see both the waveform and its harmonic content. So let’s take a look.

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, I’ll talk about a subject that can be a little be confusing to beginners in digital music: Fourier transform and harmonics. A harmonic is defined by wikipedia as follows

In acoustics and telecommunication, the harmonic of a wave is a component frequency of the signal that is an integer multiple of the fundamental frequency. For example, if the fundamental frequency is f, the harmonics have frequencies f, 2f, 3f, 4f, etc. The harmonics have the property that they are all periodic at the fundamental frequency, therefore the sum of harmonics is also periodic at that frequency. Harmonic frequencies are equally spaced by the width of the fundamental frequency and can be found by repeatedly adding that frequency. For example, if the fundamental frequency is 25 Hz, the frequencies of the harmonics are: 25 Hz, 50 Hz, 75 Hz, 100 Hz, etc.

Harmonics are what give to an organ its dinstict timbre. If you strip an instrument of all of its harmonics and leave only the fundamental frequency, it will sound the same on all instruments.

Various factors contribute to the different harmonics produced by each instrument. The end result is that two notes never sound the same on two different instruments.

A piano waveform (let) and a violin waveform (right)

But, how can we know what harmonics are included in an instrument?

To achieve this we use Fourier Trasnform.

Fourier transform is defined by wikipedia as follows

In mathematics, the Fourier transform is an operation that transforms one complex-valued function of a real variable into another. The new function, often called the frequency domain representation of the original function, describes which frequencies are present in the original function. This is in a similar spirit to the way that a chord of music can be described by notes that are being played. In effect, the Fourier transform decomposes a function into oscillatory functions. The Fourier transform is similar to many other operations in mathematics which make up the subject of Fourier analysis. In this specific case, both the domains of the original function and its frequency domain representation are continuous and unbounded. The term Fourier transform can refer to both the frequency domain representation of a function or to the process/formula that "transforms" one function into the other.

Alright, what all this means is the following.

A waveform contains many harmonics. Each harmonic can be represented by a mathematical function. The simplest wave function that can be is the sine wave. A sine wave contains only one frequency.

A sine wave along with some mathematical properties

What we do with fourier transform, is to take the function of a waveform and make it simpler, by finding the correspondent sine waves. So, we can derive the harmonic content of a waveform, and then represent it on a spectrum analyser. In the spectrum analyser we actually have each frequency that is present in the waveform and its corresponding amplitude, like shown in the picture..

Not difficult, ah?

In future articles we will write more things about spectrum analysers and fourier transform. I am also planning to include an article on Native Instruments’ Absynth, which allows you to control each individual harmonic of a waveform.

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:

Since this is the first official post, we’ll try to cover the very first and most fundamental question anyone has to face if he wants to write music on a computer: Which software do I use to do that?

This questions breaks down to this one: Which DAW do I choose?

However, before we answer this question, we have to ask: What is a DAW?

Beginners are absolutely lost about what each software is supposed to do, so let’s clarify this out first

DAW means Digital Audio Workstation. DAWs offer a coherent way to record, edit and play back audio. They can range from simple one-track audio editors, to full-blown multitrack editors that can help you record, mix and master a complete record.

The DAW is the first piece of software you’ll have to choose. In the early days DAWs were much simpler than now, and most people were still using analog gear. However, as time passes, more people are throwing away their analog gear in favor of software. The DAW is the first and most important step. It will determine which plugins you can use, compatibility with your system (PC or Mac) and hardware (like audio interfaces) and workflow.

Furthermore, DAWs are professional programs, which have a long learning curve. They are not the average computer program you can learn in one week, unless, of course, you are already an experienced computer musician/producer. This means that you must choose your DAW carefully, since it is highly likely that you will stick with it for the rest of your career.

However, rest assured that technology has advanced so much, that the major DAWs don’t have any huge differences. You can make a complete record with any of the big players in the industry. Most people consider their first DAW to be the best, since, this is the one they know the best how to use. In the end, what is most important, is to choose one and learn it inside-out. Only then will you be able to see what you really need and you will be able to find the one that truly suits you, if you feel the need to change. Furthermore, don’t forget that workflow in this work is extremely important, so, if you learn how to be fast on a program, then, you will not really feel the need to change, since you will be able to record and mix a song so easily, as if it is like second nature to you.

Now, let’s present the main DAWs in existence

Undeniably, the most popular ones are Apple’s Logic (Mac only – also take a look at http://en.wikipedia.org/wiki/Logic_Pro) and Steinberg’s Cubase (both Mac and PC, but mainly PC- also take a look at http://en.wikipedia.org/wiki/Steinberg_Cubase)

Digidesign’s Pro Tools (Mac only, once it also supported PC – also take a look at http://en.wikipedia.org/wiki/Pro_Tools) was one the industry standard (and is still used), but now it is not used as much.

Then we have Cakewalk’s Sonar (http://en.wikipedia.org/wiki/Cakewalk_Sonar), which gets better every year.

There is also the Samplitude (and its big brother Sequoia), as well as Nuendo, Cubase’s big (and expensive) brother.

We also have to mention Digital Performer (for Mac), which is a favorite among some known artists such as Matmos and Autechre

Finally, we’ll mention what I like to call the "special" players. These are DAWs, with very specific purposes. These are  Image Line’s Fruity Loops (http://en.wikipedia.org/wiki/FL_Studio), Propellerhead’s Reason (http://en.wikipedia.org/wiki/Reason_(software))and Ableton’s Live (http://en.wikipedia.org/wiki/Ableton_Live). Fruity Loops is an excellent DAW for beginners, which comes with many good virtual instruments and is very simple in its usage. Reason, does not account as a true DAW. It is rather an emulation of a studio, where you have a mixer and various instruments. Reason doesn’t support audio, only midi. However, it is VERY simple in its use and is, probably, one of the best choices for beginners. Live is what its name states, the best software for live situations. Its workflow is extremely efficient for DJs that like to mix stuff on-the-fly, allowing you to throw loops as you like, while fitting them in the tempo. However, it is not the best choice if you want a DAW for a full production.

Of course, there some other minor players like Sony’s ACID pro, Cocko’s Reaper and Orion, but here we’ll stay just with the more mainstream products.

So, which DAW should you choose?

First of all, ask yourself these questions:

1)What is my system? A Mac or a PC? If you don’t want to change the system, then stick with what is compatible with your system, in case the program does not work on both environments.

2)Do I want a complete package or something for beginners? If you want a complete package, then your best bet is Logic or Cubase. They are the most expensive DAWs, but they are also, probably, the best ones in existence. They have a long learning curve, but if you are dedicated, you can be sure that you will have in your possession tools that have almost no restriction. On the other hand, software like Reason, Fruity Loops and Live, might help you in your first steps, since trying to learn a program that has a manual of a 1000 pages, not counting the fact that you are supposed to know how digital audio works, can be discouraging (of course, that’s why Musikality Net exists, to help you learn this stuff ).

3)How much money am I willing to give? If money is not an issue, then go for the big players. If it is an issue, then consider the cheaper alternatives, like Sonar and Samplitude. These DAWs, might not be the most popular, however, they constantly get better, as they are trying to get their own slice from the pie. After all, in their effort to constantly get better and take the throne from the most popular DAWs, their companies constantly add features, that might lack from Cubase and Logic. Fruity Loops might be a good solution, since it offers lifetime free upgrades.

Cubase 4 screenshot

Finally, you have to ask yourself how much effort are you willing to put into this, and what are your goals. If for example, you’re planning to have a rock band, then using Reason is out of the question, since Reason supports only midi and is only for electronic musicians. If you’re doing this as a hobby, then maybe Logic and Cubase are not for you and you should stay with Fruity Loops and Live. It all comes down to your goals, resources and effort.

Finally, there are many people who download programs illegaly and prefer to use what is available Availability can be a big factor, since most hobbyists use PCs and, thus, it is easier to find pirated plugins for PC than for Mac. On the other hand, professionals usually use Macs, because of their great stability.

Screenshot from Thor, Reason’s 4 new synthesizer

Of course, we also have to make a special mention to open source software. At the time, there are not any open source DAWs that can pose a challenge to the professional ones. However, they are supported by large communities and they can be a good choice for those who do this as a hobby. Audacity is a good choice for those making their first steps in audio, as it supports audio-only functions and has a very simple interface. Ardour is probably the biggest open source DAW. Finally, there is also Ubuntu Studio, a special edition of the most popular Linux distribution dedicated to media production.

Well, I hope that this article covered the most basic concepts on the subject. For any questions, please leave a comment!

Hello world!

This is the first post in Musikality Net, where I’ll explain what this site is all about.

Musikality Net was created with one purpose in mind: to help independent musicians.

The dawn on the digital era and the internet revolution has caused great changes in music as we know it. Today’s musicians and music doesn’t have many things in common with the music of two centuries ago, in the way it is written, played and promoted. However, many changes occur silently through the years. Musicians find themselves unaware of them and in confusion as to what is going on and how they should adapt to this new environment of the digital era. Furthermore, many people, without any formal music education, try to write music themselves, but they are overwhelmed by the abundance of information.

I myself have struggled through all this since my childhood. What I’ll try to do in this blog, is to gather all the things I have learned in order to provide a coherent and reliable source of knowledge for all those who have tried to learn, but didn’t have any chance.

I hope I’ll make my best

So let’s get it on…