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Guide to Electronic Music Techniques

Phasing

Article from Electronics & Music Maker, November 1981


Many terms used in electronic music are surrounded by mystery. The musician is blinded by the technical jargon of the engineer and the engineer is often perplexed by the terms used in music. This month and in future issues we will be taking one or more of these terms and describing them with the aim of removing some of this mystique. Phasing is a technique very popular today, but was first used many years ago as far back as the Beatles era at the Abbey Road Studio. In simple terms, phasing as it is meant in the musical effects sense, involves delaying a signal for a small amount of time (by storing it) and then mixing the delayed signal with the undelayed.

Now, because any complex sound (and music of course is a very good example) consists of many frequencies, each component of the signal when delayed by a fixed amount and then added to the original will produce a new signal, whose individual frequency component levels will have changed according to the frequency of the component being considered and the length of the delay introduced. In simple terms, due to the wavelength of the individual frequency components, some frequencies will be completely cancelled, some will increase in amplitude and some will not change very much at all. It all depends upon the phase relationship of the signal that comes out of the delay unit with respect to the undelayed signal. For example, if a 1kHz signal is delayed by one half period (0.5 mS), then the output of the delay device will be exactly 180° out of phase with the original signal and therefore cancellation will occur at this frequency.

In fact, cancellation will occur at all frequencies at which the delay time is an odd number of half periods. This produces what is known as a comb filter because of the shape of its frequency response, as shown in Figure 1.

Figure 1. Typical comb filter response.

However, the story does not end here because the time delay introduced must be slowly changed from shorter to longer, longer to shorter and so on. The rate at which the delay is changed is made to be about 0.5Hz. As the delay time changes, the frequencies at which the notches and peaks occur in the comb filter response also change and this 'modulation' of the delay time has the effect of shifting the frequency response in Figure 1 from side to side.

At a modulation rate of 0.5Hz therefore, different frequencies will add and cancel in any two second period when added to the original signal, producing the characteristic phasing effect used so often today.

In simple terms then, the basic requirements for a Phaser are:

1. A variable delay system for audio frequencies.
2. A low frequency oscillator.
3. An audio mixer (adder).

Any phasing effects unit can be defined by the general system block diagram shown in Figure 2.

Figure 2. Block diagram of a modern phaser.


When phasing was first used many years ago it was achieved by recording the material twice on two tape recorders and then replaying the two recordings together; but with the speed of one of the machines being constantly increased and decreased at a rate of about 0.5Hz.

When the outputs of the two machines were added together they produced the same effect as that produced by modern systems. The 'delay line' in this case being a tape recorder.

It should be becoming clear now that a vital part of the phaser system is the delay line. Modern systems need not use tape and as 'electronic delay' is vital to many audio effects, some modern methods of realising this in practice will be described.

Let us first remember that the process of delaying a signal is achieved by storing it for a period of time. Furthermore, if it is required to achieve cancellation as with the Phaser, then the delay line must be able to store a signal for half of one cycle, i.e. 180°. The longest storage time must therefore be defined by the lowest frequency at which a 'notch' is required in our comb filter. So, the maximum delay time will define the lowest frequency of operation. For example, if it is required to produce phasing effects down to Middle C which is around 256Hz, then:

1 Period = 3.9mS
½ Period = 1.95mS

It can be seen then that our delay line must be able to store an audio signal for about 2mS.

There are several methods of producing 'live' delay (sometimes referred to as delay in 'Real Time'). Almost every electronic component introduces some delay, even a piece of wire. Some methods are of course impractical, for example taking the speed of signal propagation as 300000 metres per second in wire would require a length of approximately 363 miles for a 2mS delay! Electro-mechanical delay lines can be used (the spring line being a very common type) but they suffer from the disadvantage of poor frequency response, fixed delay and large physical size. Electronic phase shift networks can be used but it is difficult to produce large time delays over a wide range of frequencies.

More recently analogue shift registers have become available and these provide a much more attractive solution. The difference between this type of delay and all the others is that the signal is sampled at a certain frequency. Each sample (i.e. the instantaneous amplitude) is stored in a small capacitor. It can be shown that as long as this sampling process occurs at at least twice the highest input frequency, the original signal can be recovered. The individual samples are then shifted down the line using an electronic clock, the input signal finally appearing at the output N clock pulses later. For example, if the highest frequency to be used is 10kHz, the clock rate would have to be at 20kHz, i.e. a sample being taken every 50uS. Therefore if it were required to store the signal for 2mS, there would have to be 2mS/50uS = 40 sample stores.

Analogue shift registers are often referred to as 'Bucket Brigade' delay lines because the method of storing the signal and shifting it is analogous to a line of men passing buckets of water to each other, the amount of water in each 'bucket' representing the instantaneous amplitude of the signal when sampled. The detailed operation of these devices is of course more complex but this simplification illustrates the principles involved. The 'delay modulation' is now achieved simply by changing the clock frequency as it is this that determines how long it takes for a signal to get from input to output.

Figure 3. Digital phaser block diagram.


As a final comment, it is possible to convert the signal into digital numbers using an analogue to digital converter. If this is done, the signal is now in digital form and can be delayed using a digital shift register. These are much cheaper than their analogue counterparts but the cost is offset by the expensive input analogue to digital converter and digital to analogue converter required at the output in order to convert the digital signals back into analogue form. This is known as digital delay and, as the cost of the converters comes down, offers a very attractive alternative to the analogue system. A block diagram of a digital phaser is shown in Figure 3. The digital system has the advantage that it does not degrade the original signal, whereas analogue methods introduce noise and distortion which increase as more delay is used.


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Publisher: Electronics & Music Maker - Music Maker Publications (UK), Future Publishing.

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Electronics & Music Maker - Nov 1981

Feature by Paul Conway

Previous article in this issue:

> Working With Video

Next article in this issue:

> Organ Talk


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