Getting Started with C++ Audio Programming for Game Development: Written specifically to help C++ developers add audio to their games from scratch, this book gives a clear introduction to the concepts and practical application of audio programming using the FMOD library and toolkit.

Sound is created from the vibrations of objects. These vibrations produce variations in the atmospheric pressure which propagate away from the objects in the form of sound waves. Our ears are capable of detecting incoming sound waves and converting them into nerve signals that our brain interprets as sound.

One way to visualize sound is to draw a graph of the variations in the atmospheric pressure at each moment in time. However, understanding how those graphs relate to what we hear can be extremely complex. For that reason, we usually start by studying the simplest type of wave, the sine wave.

The sine wave is interesting for educational purposes, because we can easily identify two of the main properties of sound from it: volume and pitch. Most audio libraries allow us to control both of these properties for any sounds that we play.

Now that we know what sound is, let us turn our thoughts towards recording the sound and storing it on a computer. The first step in this process is to convert the sound wave into an electrical signal. When we use a continuous signal to represent another signal of a different quantity, we call it an analog signal or in the case of a sound wave, an analog audio signal. You are probably already familiar with the devices that perform this conversion:

Analog signals have many uses, but most computers cannot work with them directly. Computers can only operate on sequences of discrete binary numbers, also known as digital signals. We need to convert the analog signal recorded by the microphone into a digital signal, that is, digital audio, before the computer can understand it.

The most common method used to represent analog signals digitally is pulse code modulation (PCM). The general idea of PCM is to sample (or measure) the amplitude of the analog signal at fixed time intervals, and store the results as an array of numbers (called samples). Since the original data is continuous, and numbers on a computer are discrete, samples need to be rounded to the nearest available number, in a process known as quantization. Samples are usually stored as integer numbers, but it is also possible to use floating-point numbers as shown in the following example:

There are two ways to control the quality of the sampled audio:

Another aspect that we should talk about is that many audio systems have more than one output. By sending different audio signals to separate outputs (called channels), it is possible to produce the illusion of directionality and space. The number of channels on these systems can vary from one (mono) or two (stereo), to several more on surround sound systems.

The PCM format described earlier can store audio for multiple channels at once, by interleaving one sample from each channel in the correct order. The following figure shows an example of this for a stereo system:

Besides volume and pitch, which we have examined earlier, there is another property that you will usually find in every audio library, called panning. Panning applies to stereo systems, and allows you to simulate the position of the sound, placing it anywhere between the left and the right channels. For positioning in configurations with more than two channels, you normally use other advanced features, such as 3D sound.

There are so many different file formats for storing audio on a computer that it is easy to feel overwhelmed at first. Thankfully, you will only use a couple of them in your games, most of the time. Audio file formats usually fall into one of the following categories:

Be aware that despite its popularity, the MP3 is a patented format, and you cannot use it commercially without paying royalties (refer to http://mp3licensing.com/ for more information). For this book, we will be using OGG files for long sounds, and WAV files for small sound effects.

In this chapter, we have seen that sound is a series of variations in atmospheric pressure, travelling in the form of sound waves. We also saw that sound waves have properties such as amplitude and frequency, which control how loud or high it is and that you can represent a sound wave using electrical signals (analog audio) and a series of numbers (digital audio). We learned that when converting an analog signal to a digital signal, you need to control the sampling rate and the bit depth. Finally, we saw that many audio systems have more than one output and that there are many different types of audio file formats.