In this module, you will take your first steps in programming, make your first plugins, and learn how signals are processed in digital systems, while gaining an introduction to the tools used to develop synthesisers and effects processors.

LESSON PLAN

• Discover how digital audio works.
• Explore coding practices and tools (using Klang Studio).
• Learn about basic coding concepts (expressions, functions).
• Develop knowledge of basic digital signal processing (DSP).
• Build your first plugin, to control gain.

INSTRUCTIONS

Use the < > arrow(s) above to move through the topic. Each page explains a key concept using text and graphics, leading up to a practical task. You can return to any page at any time, using the arrows or ▼ drop-down menu, and also restart animations by simply clicking them.

Digital audio is the representation of sound using numbers.

The sound is measured at regular intervals, dictated by the sample rate.

Each time, the sound's amplitude is measured, in the range -1.0 to 1.0.

An analogue-to-digital converter (ADC) digitises sound by sampling its amplitude
at a high frequency (the sample rate, e.g. 44100 Hz) to produce a series of numbers:

Digital signal processing (DSP) is the process of manipulating sound as numbers.
In many cases the maths is very simple - a combination of multiplying (*) and adding (+).

In a realtime system (e.g. a plugin), we process each sample in turn.
We'll use a variable, called 'in', to refer to the current input sample.

To change the level of the signal, we multiply the sample by a scaling factor.
This is called applying gain - and uses the * operator.

This process is applied to each sample in the signal.

To output the result, the samples are converted back to analogue
sound by the system's digital-to-analogue convertor (DAC).

The gain can be controlled live by varying the scaling factor —
which could be a user parameter, set by the UI or automation.

Values less than 1.0 attenuate the signal.

Scaling by 0.0 mutes the signal; 1.0 leaves it unchanged.

Values over 1.0 amplify the signal...

We represent audio processes using block diagrams,
which describe how a signal is created and/or manipulated.

Block diagrams are a universal way of communicating concepts
designs, and techniques, widely used in textbooks, websites, papers.

The block diagram below simply copies the input to the output:

Block diagrams are directed graphs; they show signal flow using → arrows.

Signals generally flow left-to-right, top-to-bottom; but can go in any direction.

Labels are used to show important signal paths or stages
(inputs, outputs, channels, buses, mixes, sidechains, ...)

These will often mirror variable names in code, such as:

in	the input signal / current input sample
out	the output signal / current output sample

Outputs are represented with a 🔊 speaker symbol.

Outputs are represented with a 🔊 speaker symbol.

Processes may have multiple input or output channels (e.g. stereo).

To split (i.e. duplicate or copy) a signal, a node is used.

To split (i.e. duplicate or copy) a signal, a node is used.

Nodes make it easy to see when signals meet versus when they cross.

To avoid confusion; always show the direction of signal flow.

Scaling or multipling signals uses the ⊗ multiply symbol.

It multiplies two inputs to produce a new signal:

out = in × 0.5

Inputs can be any value, from any source - constants, variables, other signals.

The dial symbol represents a control parameter, provided by live input
(the UI) or automation (the DAW) - typically in the range of 0.0 to 1.0.