All posts tagged “Discovering Reason”

Control Voltages and Gates

Posted by Mattias in Tutorials

This tutorial discusses the concepts of control voltages (CVs) and Gates in Propellerhead’s software. Of course, they’re not really voltages because everything is happening within the software running on your PC or Mac. But the concepts are the same so, before going on to discuss how to use them, let’s first take the time to understand where CVs and Gates came from, what they are, and what they do.

We’ll begin by considering one of the most fundamental concepts in synthesis: there is no sound that you can define purely in terms of its timbre. Even if it seems to exhibit a consistent tone and volume, there must have been a moment when it began and a moment when it will end. This means that its loudness is contoured in some fashion. Likewise, it’s probable that its tone is also evolving in some way. So let’s start by considering an unvarying tone generated by an oscillator and make its output audible by playing it through a signal modifier — in this case, an amplifier – and then onwards to a speaker of some sort. We can represent this setup as figure 1.

Figure 1: A simple sound generator

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Creative Sampling Tricks

Posted by Mattias in Tutorials

Yours truly is old enough to have been around when digital samplers first arrived. Admittedly I never touched a Fairlight or an Emulator back when they were fresh from the factory – those products were way out of a teenager’s league – but I distinctly remember the first time I laid hands on an S612, Akai’s first sampler. Its modest 128 kB RAM could hold a single 1-second sample at maximum quality (32 kHz) – but none the less it was pure magic to be able to record something with a mic and instantly trigger it from the keyboard. I spotted that particular Akai sampler hooked up in my local music store, and tried it out by sampling myself strumming a chord on a Spanish guitar. My first sample…!

As the years went by, I gradually became spoiled like everyone else; there were tons of high quality sample libraries available on floppies, and soon enough the market was swamped with dedicated sample playback instruments such as the S1000PB, the E-mu Proteus, the Korg M1 and the Roland U-series to name but a few. This trend carried over into software instruments; manufacturers and others kept sampling like crazy for us so it seemed more or less superfluous to do it yourself. Propellerhead was no exception – with no sampling facilities and no hard disk recording, Reason remained a playback device for canned samples for almost 10 years – but in Reason 5 and Record 1.5, they got around to adding a sampling feature. In typical Propellerhead fashion, don’t do it unless it’s done right. The trick to doing it right was to bring back the simplicity and instant gratification of those early samplers – just plug in a source, hit the sample button, perform a quick truncate-and-normalize in the editor, and start jamming away.

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Thor demystified 17: Filters pt 5: Formant Filters

Posted by Mattias in Tutorials

The final filter in Thor’s armoury is a rather special one named a Formant filter, so-called because it imposes formants on any signal passed through it. But what are formants, and why would you want to impose them on anything?

Let’s start to answer this by reminding ourselves of the four types of filters most commonly found in synthesizers. These are the low-pass filter (figure 1) the high-pass filter (figure 2) the band-reject or ‘notch’ filter (figure 3) and the band-pass filter (figure 4). Our journey into formant synthesis begins with the fourth of these.

Figures 1: low-pass filter

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Thor demystified 16: Filters pt 4: Comb Filters

Posted by Mattias in Tutorials

When we talk about an audio signal generated by an analogue (or virtual analogue) oscillator, we often describe it using three characteristics: its waveform, its frequency, and its amplitude. These, to a good approximation, determine its tone, its perceived pitch, and its volume, respectively. But there is a fourth characteristic that is less commonly discussed, and this is called the ‘phase’ of the signal.

Consider the humble 100Hz sine wave. You might think that this can be described completely by its frequency and its amplitude and, in practice, this is true provided that you hear it in isolation. But now consider two of these waves, each having the same frequency and amplitude. You can generate these by taking a single sine wave and splitting its output, passing one path through a delay unit as shown in figure 1. If no delay is applied, the two waves are said to be ‘in phase’ with one another (or, to express it another way, they have a phase difference of 0º) and, as you would imagine, you could mix them together to produce the same sound, but louder.

Figure 1: Adding in-phase sine waves

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Thor demystified 15: Filters pt 3: Resonance

Posted by Mattias in Tutorials

Most physical objects vibrate at frequencies determined by their size, shape, materials and construction, and the specific frequencies for each object are known as its resonant frequencies. Nonetheless, simply adding energy to an object doesn’t guarantee that you’ll obtain an output. Imagine an object having a single resonance of 400Hz placed in front of a speaker emitting a continuous note at, say, 217Hz. If you can picture it, the object tries to vibrate when the sound first hits it, but each subsequent pressure wave is received at the ‘wrong’ time so no sympathetic vibration is established. Conversely, image the situation in which the speaker emits a note at 400Hz. The object is now in a soundfield that it is pushing and pulling it at exactly the frequency at which it wants to vibrate, so it does so, with enthusiasm.

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