SAMPLE
Page 118 S3200XL Operator’s Manual - Version 1.00
SAMPLE MODE
In SAMPLE mode, you may record sounds and in EDIT SAMPLE mode you may, amongst
other things, trim, loop, join, merge, reverse, re-sample and timestretch recordings. But before
we can do any of that, we first need to make some recordings.
WHAT IS SAMPLING?
Sampling is a process where we record sound digitally. All natural sound comes in the form of
variations in sound pressure. Using a microphone, we can convert those changes in air
pressure into rising and falling voltages. Once they exist in that format, we can process them
through ANALOGUE TO DIGITAL CONVERTERS (ADC) to turn those voltages into streams of
digital data. Once they exist as digital data, we can edit them with alarming precision.
SOUND MIC
VOLTAGE
WAVEFORM
ADC
DIGITISED
WAVEFORM
In the ADC, the sound is ‘sampled’ at a rate of 44,100 times per second. You can liken this to
film. If we take a lot of photographs in very quick succession, when we play it back, we have the
illusion of movement. The same is true of sampling. If we take enough samples, we get an
accurate reproduction of the sound. To carry this analogy even further, if you think of the very
early days of film where they didn’t take so many frames in a second, the results were jerky and
distorted. The same could be said about old samplers - because they sampled less (that is, the
sampling rate was lower), the sound quality was not so good. In order to reproduce sound
accurately, you need to sample at a frequency that is at least twice the upper reaches of the
sounds frequency range. In other words, if a sound contains frequencies that extend to, say,
15kHz, you need to sample at 30kHz at least. Instruments such as cymbals which are very
bright and contain many overtones need to be sampled at 40kHz. A bass drum, however,
which has very few upper harmonics, could feasibly be sampled at 20kHz.
The S3200XL samples at 44.1kHz, the same as compact disc so you can be sure you are
getting CD quality sound from your sampler.
The digitised waveform is loaded into RANDOM ACCESS MEMORY (RAM) where it exists as
numbers. As you know, computers and microprocessors are very good at dealing with
numbers and so we are able to rearrange those numbers and so alter the sound.
At the end of the process, we need to be able to convert those numbers back into an electrical
analogue waveform and so the numbers are reconstituted into analogue via DIGITAL TO
ANALOGUE CONVERTERS and output to your mixer or amplifier.
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