Sample Mode; Sampling - Recording Into The S2000; What Is Sampling - Akai s2000 Owner's Manual

SAMPLE MODE

SAMPLING - RECORDING INTO THE S2000

So far, we have seen how to load sounds into the S2000, place them in a MULTI for multi-
timbral sequencing, layering, etc., and how to edit the programs you have loaded. Being a
sampler, of course, you may also make your own recordings and create programs out of those.

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
In the ADC, the sound is 'sampled' at a rate of 44,100 times per second. You can liken this to
movies. 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 S2000 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 re-arrange 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 (DAC) and output to your mixer or amplifier.
One of the inherent problems with sampling is the RAM and it is not possible to have an
endless supply of it installed in the sampler. As a result, our recordings (or samples as they are
more commonly known in the music industry) have to be kept fairly short. In order to make them
last longer so that long notes can be sustained, we need to loop them. This involves selecting
a portion of the sound that will repeat over and over again when we hold our finger(s) on the
keyboard.
Page 130
VOLTAGE
MIC
WAVEFORM
ADC
S2000 Operator's Manual - Version 1.30
DIGITISED
WAVEFORM