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s Quick filter settings can be realized by using standard ISO values for the allocation of frequency bands.
Subsequently, you can fine tune the frequency of your choice.
4.5 Digital audio processing
In order to convert an analog signal - e.g. music - into a series of digital words, a so-called Analogue 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 44.1kHz, 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 analogue form - a Digital
to Analogue 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 44.1kHz is slightly higher than twice the highest audible frequency of 20kHz. 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 6dB. 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 FEEDBACKDESTROYERPRO, 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 approxima-
tions, due to their being rounded off to a defined number of decimal places. This causes further noise. To
minimize these rounding off errors, the calculations must be carried out with a higher resolution than that of the
digital audio data being processed (as a comparison, an electronic calculator may operate internally with a
greater number of decimal places than can be shown on its display). The DSPs in the
FEEDBACKDESTROYERPRO operate with a 24 bit resolution. This is accurate enough to reduce quantizing
noise to levels which are usually below the audible threshold. However, when using extreme equalizer settings,
some quantizing side effects may be detected.
Digital sampling has one further, very disturbing effect: it is very sensitive to signal overload. Take the following
simple example using a sine wave. If an analog signal starts to overload, it results in the amplitude of the signal
reaching a maximum level, and the peaks of the wave starting to get compressed, or flattened. The greater the
proportion of the wave being flattened, the more harmonics, audible as distortion, will be heard. This is a
gradual process, the level of distortion as a percentage of the total signal rising with the increase of the input
signal level.
4. TECHNICAL BACKGROUND