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Intro to Sampling and Samplers

Computer music production, and by extension this course, would not be possible if not for sampling. In this introductory presentation I discuss the two ways we think about sampling as it relates to computer music.

The first way is purely technical. Inside your audio interface exists an analog to digital (A to D) converter. That converter takes analog signal (from a microphone or line input) and converts it into a digital file. That file is then read and played back via the digital to analog converter. The crux of the matter is that all sounds that have been recorded have been sampled. That would be the majority of sounds on your hard drive. This is something we often take for granted, and in all honesty, will mostly take for granted in this course.

Our focus in this course is sampling as a creative process. We “define” sampling in music production as the appropriation and/or reappropriation of a sound, typically a digital audio file. For example when I preview through my kick drum samples and eventually settle on one for my song I am “sampling” that sound for my own work.

Taking things one step further we can load samples into dedicated instruments. These instruments are appropriately called samplers. We’ll spend the majority of this week working with these instruments. Their function is intuitive enough but don’t let the relative simplicity of the method fool you. Whereas normally on a synth you might find an oscillator with a variety of wave shapes, an empty sampler isn’t capable of generating any sound. It’s up to the user to choose the samples (sound generation source). Therefore the sampler is literally capable of an infinite number of sounds.