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Source: THE MASTER HANDBOOK OF ACOUSTICS

CHAPTER

n the early days of sound recording, signal storage was a major problem. The final recording was laid down directly, without benefit of stop-and-go recording of portions that could be patched together later. This had the advantage of minimizing the number of recording generations, but the format was a stringent one with little latitude for artistic enhancement. With the introduction of high-quality signal storage (magnetic tape or digital memory) many creative decisions, normally reserved for the recording session, were moved to the mix-down session. This opens up the opportunity of making quality enhancements in the mix-down session, long after the recording session is finished and forgotten. These quality enhancements are often made with filters of one kind or another. Is there a short traffic rumble on take 6 of the recording? A high-pass filter might cure it. Did that narrators dentures contribute an occasional high-frequency hiss? Run it through the de-esser. How about that congenital 3 dB sag in the response at 4 kHz? Easy, add a 3 dB peak of appropriate width with the parametric equalizer, and so on. The introduction of integrated circuits in the 1960s made signal processing equipment lighter, more compact, and less expensive. The coming of the digital revolution made it possible to routinely accomplish sophisticated signal processing tasks, heretofore impractical or

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impossible. This chapter is a very brief overview of both analog and digital sound processing principles and practice, especially the principles.

Resonance
The man on the stage is doing the old trick of breaking the wine glass with sound. Instead of doing it with the divas voice, he holds the goblet in front of the loudspeaker, which shatters it as a high-intensity tone is emitted. The secret lies in the brief preparation made before the audience was assembled. At that time he placed a small coin in the goblet and held it in front of the loudspeaker as the frequency of the sine generator was varied at a low level. He carefully adjusted the generator until the frequency was found at which the coin danced wildly in the glass. During the demonstration no tuning was necessary, a blast of sound at this predetermined frequency easily shattered the glass. The wild dancing of the coin in the glass in the preliminary adjustment indicated that the excitation frequency from the loudspeaker was adjusted to the natural frequency or resonance of the goblet. At that frequency of resonance a modest excitation resulted in very great vibration of the glass, exceeding its breaking point. As shown in Fig. 6-1, the amplifo tude of the vibration of the glass changes as the frequency of excitation is varied, going through a peak response at the frequency of resonance, fo. Such resonance effects appear in a wide variety of systems: the interaction of mass and stiffness of a mechanical system, such as a tuning fork, or the acoustical resonance of the air in a bottle, as the mass of the air in the neck of the bottle reacts with the springiness of the air entrapped in the Frequency body of the bottle. See Helmholtz resFIGURE 6-1 onators, Chap. 9. The amplitude of vibration of any resonant system is Resonance effects are also dominant in maximum at the natural frequency or resonant freelectronic circuits as the inertia effect of quency (f) and is less at frequencies below and above that frequency. an inductance reacts with the storage
Amplitude

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effect of a capacitance. Figure 6-2 shows the symbols for inductance (L), commonly a coil of wire, and capacitance (C), commonly made of sheets of conducting material separated by nonconducting sheets. Energy can be stored in the magnetic field of an inductance as well as in the electrical charges on the plates of a capacitance. The interchange of energy between two such storage systems can result in a resonance effect. Perhaps, the simplest example of this is a weight on a spring. Figure 6-3 shows two forms in which an inductance and a capacitance can exhibit resonance. Let us assume that an alternating current of constant amplitude, but varying frequency is flowing in the parallel resonant circuit of Fig. 6-3A. As

FIGURE 6-2
Symbols for inductance (L) and capitance (C).

L L C

Generator C

Voltage

Freq.

Voltage

Freq.

B
FIGURE 6-3

A comparison of (A) parallel resonance and (B) series resonance. For a constant alternating current flowing, the voltage across the parallel resonant circuit peaks at the resonance frequency while that of the series resonant circuit is a minimum.

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the frequency is varied, the voltage at the terminals reaches a maximum at the natural frequency of the LC system, falling off at lower and higher frequencies. In this way the typical resonance curve shape is developed. Another way of saying this is that the parallel resonant circuit exhibits maximum impedance (opposition to the flow of current) at resonance. Figure 6-3B illustrates the series resonant arrangement of an inductance L and a capacitance C. As the alternating current of constant magnitude and varying frequency flows in the circuit, the voltage at the terminals describes an inverted resonance curve in which the voltage is minimum at the natural frequency and rising at both lower and higher frequencies. It can also be said that the series resonant circuit presents minimum impedance at the frequency of resonance.

Filters
The common forms of filters are the low-pass filter, the high-pass filter, the band-pass filter, and the band-reject filter as illustrated in Fig. 6-4. Figure 6-5 shows how inductors and capacitors may be arranged in numerous ways to form very simple high- and low-pass filters. Filters of Figure 6-5C will have much sharper cut-offs than the simpler ones in (A) and (B). There are many other highly specialized filters with specific and unusual features. With such filters, a wideband signal such as speech or music can be altered at will.

Low-pass filter

High-pass filter

Band-pass filter

Band-reject filter

Response

Frequency

FIGURE 6-4
Basic response shapes for the low-pass, high-pass, band-pass, and band-reject filters.

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High-pass filter

Low-pass filter

FIGURE 6-5
High-pass and low-pass filters of the simplest form. The filters in (C) will have sharper cut-off than the others.

Adjustable filters can be readily shifted to any frequency within their design band. One type is the constant bandwidth filter which offers the same bandwidth at any frequency. For example, a spectrum analyzer may have a 5-Hz bandwidth whether it is tuned to 100 Hz or 10,000 Hz, or any other frequency within its operating band. An even more widely used adjustable filter offers a pass band-width that is a constant percentage of the frequency to which it is tuned. The 13-octave filter is such a device. If it is tuned to 125 Hz the 13-octave bandwidth is 112 to 141 Hz. If it is tuned to 8,000 Hz the 13-octave bandwidth is 7,079 Hz to 8,913 Hz. The bandwidth is about 23% of the frequency to which it is tuned in either case.

Active Filters
Active filters depend on integrated circuits for their operation. An integrated circuit can have many hundreds of components in a small enclosure. Their fabrication depends on growing transistors and

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resistors on a semiconductor wafer and interconnecting the components by an evaporated metal pattern. Great circuit complexity can be compressed into unbelievably small space in this way. A low-pass filter assembled from inductors and capacitors in the old-fashioned way is shown in Fig. 6-6A. Another low-pass filter based in an integrated circuit is shown in Fig. 6-6B. The four small resistors and two small capacitors plus the integrated circuit illustrate the space-saving advantage of the active filter. It is interesting that the feedback capacitor C1 in Fig. 6-6B has the electrical effect of an inductance.

Analog vs. Digital Filters


Filters can be constructed in analog or digital form. All the filters discussed to this point have been of the analog type and applied widely in equalizers. By adjusting the values of the resistors, inductors, and capacitors any type of analog filter can be constructed to achieve almost any frequency and impedance matching characteristic desired.

C1

B
FIGURE 6-6
Two low-pass filters, (A) traditional analog type, (B) an active filter utilizing an integrated circuit.

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Digitization
A digital filter contains no such physical components as inductors and capacitors. It is basically a computer program that operates on a sample of the signal. This process is described in Fig. 6-7. The incoming analog signal is represented in Fig. 6-7A. Through a multiplication or modulation process, the analog signal of Fig. 6-7A is combined with the sam-

FIGURE 6-7
(A) the analog signal, (B) digitizing pulses, (C) the digitized analog signal resulting from the modulation of (A) with (B). Application of a sample-and-hold circuit to (D) which completes the quantization process reducing the analog samples to discrete values suitable for storage in memory.

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pling pulses of Fig. 6-7B. These pulses, in effect, break down the analog signal into a series of very brief samples having amplitudes equal to the instantaneous value of the signal amplitudes, as shown in Fig. 6-7C. This process is called digitization. With no loss of information, energy between the sampling points is discarded. The sampling rate must be at least twice the highest frequency of interest. If a sampling rate less than this amount is used, spurious signals are generated.

Quantization
It is now necessary to convert the samples of Fig. 6-7C to discrete values that can be stored in a computer. This is done by a sample-andhold circuit so that the amplitude of each digitized pulse is converted into discrete values suitable for computer storage. The sample-andhold circuit forces the amplitude of the sample to have a constant value throughout the sample period. This process is essentially a stepping backward through a digitized signal, sample by sample, subtracting from each sample some large proportion of the sample before it. The resulting samples are thus mainly changes in the signal sample. The closer the spacing of the digitized samples, the more accurately the analog signal is represented. However, a restraining influence on increasing the number of samples is that more computer memory is required to store the data. The calculations required are intensive in multiplication and accumulation operations.

Digital Filters
Digital filters can be made without benefit of inductors or capacitors. A typical digital filter of the so-called FIR type is shown in Fig. 6-8. The analog signal is applied to the input on the left. The analog-to-digital (A/D) converter digitizes and quantisizes the analog signal. An oscillator (clock) determines the number of digitizing pulses per second and controls all timing of the device. The type of filter is determined by the program in the read-only memory.

Application of Digital Signal Processing (DSP)


Digital signal processing has been successfully applied in various ways, including1:

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A/D converter

Read-write memory

Multiplier

Clock and register

Programmable read-only memory

Accumulator

FIGURE 6-8
Block diagram of a typical digital filter, in this case one of the finite impulse response (FIR) type. The type of filter is determined by the read-only program.

Mixing two signals (convolution) Comparing two signals (correlation) Changing ac signals to dc (rectification) Amplifying signals Acting as a transformer Spectral analysis Speech processing, recognition, etc. Noise cancellation Music synthesis and processing. Specific tasks ideally suited to digital signal processing include: Subsonic filters Ultrasonic filters Limiters Compressors Expanders Companders Noise gates Bass correction

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Noise reduction Image enhancement Stereo synthesizer This very brief treatment of digital signal processing has skipped over such vital aspects as aliasing, the multiplier/accumulator, sampling rates, quantization levels, etc.

Application of DSP to Room Equalization


The application of digital signal processing to the loudspeaker-roomlistener problem is currently being pursued vigorously. Huge errors are common in both the room and the loudspeakers. The problems of the listener are primarily the great differences in sensitivity between listeners. The basic idea is to measure the frequency and phase response of the speaker/room combination and apply an equalizer that perfectly compensates for the defects. This is a very complicated operation, but theoretically possible through the great potential of digital signal processing. One problem to be solved is how to generalize the solution based on data picked up at one point in the room. There are many other problems but the next decade should bring major advances in this field.

Endnote
1

Burkhard, Mahlon, Filters and Equalizers, Chapter 20, Handbook For Sound Engineers, Glen Ballou, Ed., Carmel, IN, Howard W. Sams Co., 2nd Edition (1987).

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