quiz 1 review. analog synthesis overview sound is created by controlling electrical current within...

Quiz 1 Review

Analog Synthesis Overview Sound is created by controlling

electrical current within synthesizer, and amplifying result.

Basic components: Oscillators Filters Envelope generators Noise generators

Voltage control

Basic Analog Components Oscillators: Create periodic fluctuations in

current, usually with selectable waveform. Envelope Generators: Generate a control

function that can be applied to various synthesis parameters, including amplitude, pitch, and filter controls.

Filters: Given an input signal, attenuate or boost a frequency range to produce an output signal

Noise Generators: Generate a random, or semi-random fluctuation in current that produces a signal with all frequencies present. (Know difference between white and pink noise.

Digital Synthesis Overview Sound is created by manipulating

numbers, converting those numbers to an electrical current, and amplifying result.

Numerical manipulations are the same whether they are done with software or hardware.

Digital Waveforms (two ways) Compute the sine in real time, every time it is

needed. (advantages and disadvantages) Compute once, store in table, look up when

needed. (advantages and disadvantages)

Digital Oscillator Basics Phase Increment Algorithm (from Roads, p.93) Interpolation and Quantization (errors

result in noise)

Additive Synthesis Basic: The addition of elementary waveforms to

create a new waveform. Disadvantages (Roads, pp. 107-108)

Subractive Synthesis Start with a “rich” sound source

typically noise, sawtooth, square, or pulse wave(s)

multiple sources possible (additive) Attenuate or boost frequencies, or

frequency ranges. filter

Filter Definition (Roads p. 185) A filter is any operation on a signal

(From Rabiner et al, “Terminology in Digital Signal

Processing.”

Commonly, we limit the term filter to devices (hardware or software) that were designed specifically to boost or attenuate regions of a sound spectrum.

Filter Basics (p. 185) Filters work by using one or both of the

following methods: Delay a copy of the input signal (by x number of

samples), and combine the delayed input signal with the new input signal.

(Finite Impulse Response, FIR, or feedforward filter)

Delay a copy of the output signal (by x number of samples), and combine it with the new input signal.

(Infinite Impulse Response, IIR, feedback filter)

Filter Terms (pp. 186-193) Typical types: lowpass, highpass, bandpass,

and bandreject (notch) Terms: cutoff frequency, center frequency,

bandwidth, Q, slope, gain, resonance, order, stopband and passband.

Relationship of slope to phase integrity. A resonance control means that you have

what type of filter? Relationship of order to number of samples

used in delay. (p. 412)

FIR Math [0] Simple Lowpass Filter (averaging):

output = half_of_current_input + half_of_previous_input

y[n] = (0.5 x x[n]) + (0.5 x x[n - 1])

Simple Highpass Filter (difference): output = half_of_current input -

half_of_previous_input y[n] = (0.5 x x[n]) — (0.5 x x[n - 1])

IIR Math[0] The feedback loop introduced creates

the possibility of an infinite impulse (delayed sample).

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y[n] = (0.5 × x[n]) + (0.5 × y[n −1])