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Next: FM synthesis on SND
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Previous: Tutorials-and-More about SND
Signal theory tells us that by means of the Fourier Transform, any
periodic waveform can be represented as a sum of harmonically related
sinusoids, each one with its own particular amplitude and phase. The
timbre or the spectra of a sound can be viewed from two different
perspectives known as the ``time domain'' and the ``frequency
domain''. Depending on conditions and how a sound is being analyzed
one view might be more useful than the other. For example in the case
of additive synthesis, the frequency domain is more useful while if we
were to edit a complete chunk of sound the time domain will certainly
be more useful.
In theory the Fourier Transform (FFT) generates most of the
information necessary to reconstruct a signal from a complex
sound. Additive synthesis can be used for reconstructing a signal by
synthesizing each partial as given on the FFT of the original signal
but more generally speaking additive synthesis is used to sum and mix
sinusoids to produce more complex sounds. This allows for control over
the individual simple components by means of individual envelopes for
amplitude and frequency.
Therefore one should be able to add up a bunch of sine waves and get
any complex arbitrary signal. The simplest case is when all overtones
are integer multiples of the fundamental frequency. In this simple
case the waveform is periodic. As the periodic waveform repeats over
time we can implement additive synthesis by using a table to store the
values of one cycle instead of adding the output of all the equivalent
sine oscillators (it is a lot more efficient). Here's a simple Snd
instrument inspired on Fernando Lopez-Lezcano's (circa 1996)
that implements additive synthesis by using the table-lookup
unit generator. This is called harmonic synthesis by means of
wave-table synthesis.
(define dowave
(lambda* (start-time dur frequency amplitude harmonics)
(let* ((start (floor (* start-time (srate))))
(len (floor (* dur (srate))))
;;
;; create a table called "waveform" with the harmonics
;;
(waveform (partials->wave harmonics ))
;;
;; create the table lookup unit generator
;;
(s (make-table-lookup :frequency frequency :wave waveform))
;;
;; add a simple amplitude envelope...
;;
(amp-env (make-env :envelope '(0 0 0.5 1 1 0)
:duration dur
:scaler amplitude))
(output (make-vct len)))
(do ((i 0 (1+ i)))
((= i len))
(vct-set! output i (* (env amp-env) (table-lookup s)) ))
(vct->channel output start len))))
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You can try the following function calls to listen to the tones.
- Open a new sound:
(new-sound "/zap/test.snd")
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- A simple function call to produce a single tone sound.
(dowave 0 1 400 .5 '(1 .5 ))
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- A function call to produce a ``square-wave'' sound.
(dowave 0 1 400 .5 '(1 1 3 .33 5 0.2 7 0.14286 ))
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Open the frequencies view or F-view on Snd's graphic interface to get
a good look at the spectra of the sound. Move the sliders to see how
the sound evolves.
- A simple function call to do a ``Sawtooth'' sound.
(dowave 0 1 400 .5 '(1 1 2 .5 3 .333 4 .25))
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- A call to have a ``triangular-wave'' sound.
(dowave 0 1 400 .5 '(1 1 3 0.1111 5 0.04 7 0.20408))
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- Type the (play) function call to listen the sound (you can
always type ' C-x u ' to undo or erase the active sound you just
created on Snd's editing window).
In Partial Synthesis the overtones are not integer multiples of a
fundamental frequency and thus we do not resort to the previous
shortcut of a table. Therefore, in this case each component (partial)
needs to be implemented independently and separately and furthermore
sum each signal component to get the more complex sound. This means a
lot of computing resources which translate in more computations and
time.
Here is a very simple instrument that implements additive synthesis
with three partials.
(define doadd
(lambda* (start-time duration frequency amplitude
#:key
(partial1 1.0)(amp1 0.3)
(partial2 2.0)(amp2 0.3)
(partial3 3.0)(amp3 0.3)
(ampenv '(0 0 0.5 1 1 0)))
;;
(let* ((start (floor (* start-time (srate))))
(len (floor (* duration (srate))))
(sine1 (make-oscil :frequency (* partial1 frequency)))
(sine2 (make-oscil :frequency (* partial2 frequency)))
(sine3 (make-oscil :frequency (* partial3 frequency)))
(amp-env (make-env :envelope ampenv
:scaler amplitude
:start start
:end len))
(output (make-vct len)))
;;
(vct-map! output (lambda ()
(* (env amp-env)
(+ (* amp1 (oscil sine1))
(* amp2 (oscil sine2))
(* amp3 (oscil sine3))))))
(vct->channel output start len ))
))
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The function calls for this instruments are as follows:
- Open a new sound:
(new-sound "/zap/test.snd")
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- A simple function call to the instrument with it default values:
In the most general case all parameters of each sine wave are also a
function of time (that is, they are controlled by envelopes). This is
the most interesting case but also the most expensive computationally
and the most difficult to control. But the problem is: "how do we create
or generate the enormous amount of data that we need to accurately
represent hundreds of points in the envelopes or all partials?"
- Import from somewhere else ?
- Program generated ?
- High level musical concepts ?
- Synth parameters ?
- Manually ?
- Analysis -> Re-synthesis ?
Next: FM synthesis on SND
Up: SND
Previous: Tutorials-and-More about SND
© Copyright 2001-2006 CCRMA, Stanford University. All rights reserved.
Created and Mantained by Juan Reyes
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