All posts tagged “Synthesis”
Propellerhead’s product specialist Mattias Häggström Gerdt shows you how to use the Combinator to access more modulation options in Reason’s synths and how you can control this with CV. More possibilities for making interesting, evolving sounds is just inside the Combinator!
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.
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.
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.
There’s one thing you’ll never hear when synthesiser enthusiasts wax lyrical about their instruments: an argument about which has the sweetest or fattest high-pass filter. They’ll debate endlessly the benefits of Moog’s discrete component low-pass filters, argue about the pros and cons of CEM and SSM low-pass filter chips, and possibly come to blows over whether the 12dB/octave low-pass filter in the early ARP Odysseys is better or worse (whatever that means) than the 24dB/octave low-pass filter in the later models. But nobody ever got punched because they insulted someone’s high-pass filter.
What’s more, there was a time when you had to work quite hard to find a high-pass filter on an integrated (i.e. not a modular) synth. The groundbreaking instruments of the late ’60s and early ’70s – Minimoogs, ARP2600s and EMS VCS3s – didn’t have them and, by and large, it was left to emerging manufacturers such as Korg, Yamaha and Roland to bring them to the public’s attention.
So why is the high-pass filter such a poor relation when compared with its twin, the low-pass filter? To understand this, we again have to consider the nature of natural sounds.