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4.1.4 Expanders/noise gates
Audio, in general, is only as good as the source from which it was derived. The dynamic range of signals will
often be restricted by noise. Synthesizers, effects devices, guitar pickups, amplifiers etc. generally produce a
high level of noise, hum or other ambient background hiss, which can disturb the quality of the program
material.
Normally these noises are inaudible if the level of the desired signal lies significantly above the level of the
noise. This perception by the ear is based on the masking effect: noise will be masked and thus becomes
inaudible as soon as considerably louder sound signals in the same frequency band are added. Nevertheless,
the further the level that the desired signal decreases, the more the noise floor becomes a disturbing factor.
Expanders or noise-gates offer a solution for this problem: these devices attenuate signals when their ampli-
tudes drop, thereby fading out the background noise. Relying on this method, gain controlling amplifiers, like
expanders, can extend the dynamic range of a signal and are therefore the opposite of a compressor.
In practice, it is shown that an expansion over the entire dynamic range is not desired. With an expansion ratio
of 5:1 and a processed dynamic range of 30 dB, an output dynamic range of 150 dB will be the result,
exceeding all subsequent signal processors, as well as human hearing. Therefore, the amplitude control is
restricted to signals whose levels are below a certain threshold. Signals above this threshold pass through the
unit unchanged. Due to the continuous attenuation of the signals below this threshold, this kind of expansion
is termed downward expansion.
The noise-gate is the simplest form of an expander: in contrast to the expander, which continuously attenuates
a signal below the threshold, the noise-gate cuts off the signal abruptly. In most applications this method is not
very useful, since the on/off transition is too drastic. The onset of a simple gate function appears very obvious
and unnatural. To achieve inaudible processing of the program material, it is necessary to be able to control the
signal's envelope parameters. This is part of the many features of the ULTRA-CURVE PRO.
4.2 Digital audio processing
In order to convert an analog signal - e.g. music - into a series of digital words, a so-called Analog to Digital
Converter or ADC is used. The converter functions by viewing the signal entering it a given number of times
over a period of time, e.g. 44,100 times per second, giving a rate of 44.1 kHz, and in each case measuring the
signal amplitude, and giving it a numerical value. This form of measuring the signal regularly over a period of
time is known as sampling, the conversion of the amplitude into a numerical value, quantizing. The two
actions together are referred to as digitizing.
In order to carry out the opposite - the conversion of a digitized signal into its original analog form - a Digital to
Analog Converter or DAC is used. In both cases the frequency at which the device operates is called the
sampling rate. The sampling rate determines the effective audio frequency range. The sampling rate must
always be more than twice the value of the highest frequency to be reproduced. Therefore, the well known CD
sampling rate of 44.1 kHz is slightly higher than twice the highest audible frequency of 20 kHz. The accuracy
at which quantization takes place is primarily dependent on the quality of the ADCs and DACs being used.
The resolution, or size of digital word used (expressed in bits), determines the theoretical Signal/Noise ratio
(S/N ratio) the audio system is capable of providing. The number of bits may be compared to the number of
decimal places used in a calculation - the greater the number of places, the more accurate the end result.
Theoretically, each extra bit of resolution should result in the S/N ratio increasing by 6 dB. Unfortunately, there
are a considerable number of other factors to be taken into account, which hinder the achievement of these
theoretical values.
If you picture an analog signal as a sinusoidal curve, then the sampling procedure may be thought of as a grid
superimposed on the curve. The higher the sampling rate (and the higher the number of bits), the finer the grid.
The analog signal traces a continuous curve, which very seldom coincides with the cross points of the grid. A
signal level at the sampling points will still be assigned a digital value, usually the one closest to the exact
representation. This limit to the resolution of the grid gives rise to errors, and these errors are the cause of
quantizing noise. Unfortunately, quantizing noise has the characteristic of being much more noticeable and
unpleasant to the ear than natural analog noise.
In a digital signal processor, such as the DSPs in the ULTRA-CURVE PRO, the data will be modified in a
number of ways, in other words, various calculations, or processes, will be done in order to achieve the desired
effect on the signal. This gives rise to further errors, as these calculations are approximations, due to their
being rounded off to a defined number of decimal places. This causes further noise. To minimize these round-
ing off errors, the calculations must be carried out with a higher resolution than that of the digital audio data
4. TECHNICAL BACKGROUND
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Summary of Contents for ULTRA-CURVE PRO DSP8024
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