ENSONIQ ASR-10, Musician'S Manual

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Section 7 — Sampling/Signal Source Concepts
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This section provides an overview of the sampling process. For a complete list of the sampling
parameters, refer to the following section.
What is Sampling?
Before getting into talking about samples and sampling, let’s begin with a simple explanation of
sound. A sound begins as a series of vibrations, or pressure waves, in the air. When these
vibrations reach the diaphragm of a microphone, they cause it to move back and forth. This
creates a fluctuating electrical signal that rises and falls around a center, or zero line.
A simple sound wave, once it has been converted into an electrical signal, might look like this:
0
+100%
-100%
A conventional analog tape recorder would record this signal by converting the electrical
fluctuations into magnetic fluctuations and then imprinting a continuous (or “analog”) copy onto
the magnetic surface of the tape.
A digital sampling system, such as the ASR-10 or a digital tape recorder, works a little differently.
When you digitally record the sound, the level of the signal is measured (or “sampled”)
thousands of times per second, and each number is recorded as a number in memory. If the same
signal that was shown in the analog recording illustration above was recorded digitally, it would
look like this:
Signal level measured
The digital sampler does not record the actual sound but rather encodes a series of discrete
numbers, each of which represents the level of the signal at a given instant in time.
On playback, the ASR-10 reconstructs the original signal by “connecting the dots,” producing an
output voltage that corresponds to the numbers in its memory (shown below). This voltage can
then be amplified and sent to speakers, which turn it back into pressure waves in the air so that
we can hear it.