INTRODUCTION TO SYNTHESIZERS - hol.gr

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The circuit reading the samples from the memory and converting them to sounds is called the "digital-to-analog
converter", or D/A-converter. Just as the A/D converter, this circuit is also controlled by a clock. For each tick of
the clock, a new sample point is read out and added to the waveform. The speed of the clock controlling the
D/A-converter is called the "readout rate".
If the readout rate is set to the same rate as the sample was recorded, the result is an exact reproduction of the
original sound source - or as exact as the sampling frequency and sampling resolution used for the recording
process will allow. But we can also choose to change the readout rate to achieve a pitch change, or
transposition. If we increase the readout rate, it will be just like playing a tape with higher speed - the sound
will have a higher pitch. If we instead read the samples at a slower rate, the pitch will drop.
This method is often used to map the sampled sound over the keyboard and thus creating a playable sound. By
doubling the readout rate, the pitch of the sound will be exactly one octave above its original value.
Another transposition method is "skipping" some of the sample points while playing back a sampled sound, and
thereby speeding up the playback rate. By skipping every other sample value, we in fact raise the sound with
one octave. To lower a sample beyond its original pitch, new artificial values need to be added between two
sampled values. We can lower the sound with one octave by simply retrieving each sample point twice.
Multisampling
Unfortunately we can't just take one sampled sound and transpose it over the whole keyboard range. As we
increase the sample readout rate, we will notice that not only the pitch will rise, but the sound will also become
very weird and "chipmunk-like". And at the opposite side, an artificially lowered sound will start to break up and
distort just a few notes below the normal pitch.
To avoid this, we can use a technique called multisampling. Instead of using just one sample over the whole
keyboard range, we use several different samples with different pitches, and combine these to cover the
playable range. We could instance sample a C3 note, a E3 note and an A3 note to cover one octave. Now we
don't have to force one individual sample to stretch further than just a few notes.
If we wish, we can use a different sample for each key on the keyboard, but quite often we can get away with
as few as five or six different samples. To further enhance the playability of our sampled sounds, we can even
assign different sets of samples to be played depending on how hard or soft we strike a key on the keyboard.
Such a map of samples is usually called a program.
Editing
Once a sound has been digitized and resides inside our computer or sampler, we can manipulate and change it
in an almost infinite number of ways. By using a computer, we can visualize the sample and edit it almost any
way we like. Samples can be time-stretched, cut, shortened, reversed, flanged and twisted to create completely
new and astonishing sounds.
One of the most useful editing functions is the "loop" function. Looping a sound means that a small part of the
sampled sound is played repeatedly over and over again. By using a clever loop, a short sample of e.g. a violin
sound can be stretched out to last as long as we hold down the key. A benefit of the loop function is of course
the saving of precious sample memory.
But there is nothing to stop us from sampling whole bars of music. We can in fact rip off drum patterns, vocal
chords, guitar riffs or cool grooves from our favorite album and loop it to use them in our own music. Ethically
this might of course be highly doubtful and since we're most likely infringing on copyrighted material, it can
even involve legal aspects.
Playback modules
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