Journal of Sound and Vibration (1972) 21(3),
AN ELECTRONIC
365-368
AUDIO
DELAY SYSTEM
J. D. E. BEYNONANDP. G. JENNINGS Department of Electronics, University of Southampton, Southampton SO9 5NH, England (Received 9 December 197 1)
It is frequently desirable to be able to delay audio frequencies by times ranging from several milliseconds to hundreds of milliseconds. Although several methods have been developed to achieve this, most of them employ electromechanical delay systems. The article describes an all-electronic technique of delaying acoustic signals; experimental work on the system has established its feasibility. 1. INTRODUCTION It is frequently desirable to be able to delay audio frequencies by times ranging from several milliseconds to hundreds of milliseconds. Such delays are frequently used in systems for artificially increasing the reverberation time of auditoria and for superimposing “echo” effects on musical instruments which already employ audio amplifiers; they can also be used to advantage in conjunction with voice-switched amplifiers, such as are used in “loudspeaking telephones”, to prevent loss of the first part of the first word, particularly when this is a sibilant. The conceptually-simplest way of achieving such a time delay is by means of an acoustic delay-line. However, the speed of sound in solids is generally so high (w IO4 m/s) that 10 metres of delay line would be necessary to produce a delay of only 1 millisecond! Time delays of the order of tens of milliseconds or more can be obtained by using a modified tape recorder incorporating separate recording and pick-up heads; the delay depends on the distance between the heads and the tape speed. However, the relatively complex mechanical apparatus requires regular attention involving replacement of the tape loop, cleaning of the tape heads and guides, and servicing of the motor which must be run all the time the system is in use. Such a system would be quite unacceptable in applications such as the loud-speaking telephone. We have investigated the possibility of making an all-electronic audio delay system which is capable of introducing delays of from less than a millisecond up to tens or hundreds of milliseconds, according to the number of delay circuits employed. 2. THE ELECTRONIC DELAY 2.1. PRINCIPLE In the electronic system the audio signal is first converted into a pulse-width modulated (PWM) wave; a series of monostable multivibrators is then used to delay the signal by the required amount and the delayed signal is subsequently reconverted to its original form by means of a simple demodulator. Monostables have been used in preference to other standard digital delays such as shift registers, since the latter can be used only in conjunction with relatively complex modulation systems such as pulse code- or delta-modulation. 365
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J.D. E. BEYNON AND P. G. JENNINGS
2.2.MODULATOR The method used to achieve a PWM wave is illustrated in Figure 1. Let us denote byf,,, the highest frequency component in the audio signal to be delayed. The audio signal [Figure 1, curve (a)] is then superimposed upon a sawtooth waveform [curve (b)] whose frequency is at
(b)
+-Threshold of tmjqer Clrcult
(d)
Figure 1. Stages in producing a pulse-width modulated and sawtooth; (d) output of trigger circuit (PWM) signal.
signal. (a) Audio signal; (b) sawtooth;
(c) audio
least 2fmax (this lower limit on the frequency of the sawtooth waveform follows from Shannon’s Sampling Theorem). The resultant signal [curve (c)] is fed to a trigger circuit set to switch about the mid-point of the waveform. The output of the trigger circuit [curve (d)] is then the required PWM signal. 2.3. DELAY CIRCUITRY The principle of the delay is illustrated in Figure 2. The negative-going edges of the PWM signal coincide with the trailing edges of the sawtooth, and are therefore spaced evenly in time. They therefore carry no audio information, and are removed by means of a differentiator and diode network. The positive pulses, which contain the necessary information, form a pulseposition modulated (PPM) waveform [Figure 2, curve (c)l. (a)
e//
(b)
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(c)
I
I
(d) 1
n
(e)
I
(f)
A
I
1 ‘1
I
1
‘1
‘\
1 \
\
\ \
(g)
(h) (i) Figure 2. Stages in producing a delay in a PWM signal. (a) Sawtooth; (b) PWM; (c) PWM differentiated and rectitied; (d) output from first monostable; (e) output differentiated and rectified; (f) output from second monostable differentiated and rectified; (g) output from last monostable differentiated and rectitkd; (h) sawtooth differentiated and rectitied; (i) output from bistable.
AN ELECTRONIC
AUDIO DELAY SYSTEM
367
A monostable can now be used to delay these pulses by nearly one period, (If the monostable were to delay any pulse by more than one period, then the subsequent input pulse would have no effect on the output, and the information associated with it would be lost.) The output waveform of the monostable [curve (d)] is similarly differentiated and the negative-going pulses removed; the resulting waveform [curve (e)] is identical in form to the input to the monostable [curve (c)] but has been delayed by nearly one period. As long a delay as is required can be achieved simply by cascading a sufficient number of identical monostables. The PWM signal is recovered by passing the output of the final monostable to the “set” input of a bistable multivibrator. The “reset” input to the bistable comes from the negativegoing edges of the original sawtooth waveform. If we assume that the b&able is initially in State 1, the first pulse to arrive comes from the monostable, and switches the bistable to State 2. The next pulse comes from the sawtooth generator, and causes the bistable to swtich back to State 1. This is repeated continuously, the bistable being switched to State 2 at times dependent on the audio input, and back again at the regular sampling interval. The output from the bistable [curve (i)] is thus of the same form as the output from the modulator [curve (b)], but delayed by the required time. 2.4.
DEMODULATOR
To convert the PWM signal back to analogue form, the output from the bistable multivibrator is fed to a low-pass filter, whose cut-off frequency is less than half the sampling rate. (This, again, is to satisfy Shannon’s Theorem.)
3. FEASIBILITY STUDY The maximum delay that can be obtained from each monostable is dependent on the sawtooth frequencyf,, which, as has already been explained, must be at least 2f,,,; the lower f,,,, the greater the delay that can be obtained from each monostable. To carry out a feasibility study of the system described above we limited fmsx to 1 kHz. If an ideal low-pass filter were available, the delayed audio signal could be reconstituted at the output without any problem of suppressing the 2 kHz sawtooth repetition frequency. However, because of the finite rate of attenuation of practical filters, the sawtooth frequency was chosen to be 4 kHz in order to minimize the problem of suppressing this unwanted frequency in the demodulated audio output; a 4 kHz repetition frequency corresponds to a sampling period of O-25 ms. In the PPM signal of Figure 2, curve (c) [which is derived from waveform (d) of Figure 11, the shortest time between two adjacent pulses is about 75% of their mean period. This is therefore the longest possible time by which one monostable can delay the signal without the risk of missing the subsequent incoming pulse. Thus, in our case the delay per monostable must be less than 75 ‘A of O-25ms, i.e. 0.19 ms; in practice this time was reduced to 0.18 ms, since the stages were found to take 0.01 ms to switch back to their stable state. Experiments have been carried out in which 10 monostables were used in the delay line. A total delay of 1.8 ms has been successfully achieved, thus demonstrating the feasibility of the scheme. 4. DISCUSSION
The only difficulty which has arisen has been in connection with the suppression of the 4 kHz sawtooth repetition frequency. The low pass filter used in our experiments (a 6th order Chebychev filter) gave a “signal-to-whistle” ratio of 50 dB, but subjectively this was not considered adequate. Limited time has prevented the construction of a better filter but,
368
J. D.
E. BEYNON
AND P. G. JENNINGS
without doubt, increased suppression of the sawtooth frequency could be obtained using, for exampIe, a notch filter in conjunction with the low pass Chebychev filter employed above. However, in a system designed for the whole of the audio frequency range (up to, say, 15 kHz) a sawtooth repetition frequency of at least 30 kHz would be necessary. This is well above the response limit of the human ear and the presence of such a component, suppressed by several tens of decibels with respect to the signal, should pose no problems. Of course, to delay signals of 15 kHz by a given time requires many more monostables than would be required to delay a 1 kHz signal by the same amount, since the delay available per stage would be much smaller. Thus, to delay a 15 kHz signal by a few milliseconds would require of the order of 100 monostables. However, the basic delay element is very simple and the wiring-up of 100 or more monostables presents no particular problem, especially as such circuits are now readily available in integrated-circuit form. For voice-switched amplifiers a fixed overall delay is all that is required. In other applications, such as the simulation of “echo “effects, however, advantage could be taken of the fact that, by using the above system, samples of the PPM signal at several stages along the delay circuitry could be taken. By reconstituting the original waveform from each delayed PPM signal and adding these together, a very effective “echo” effect could be produced; indeed, since the magnitude of each of the delayed audio components could be varied at will, a large variety of effects could be obtained.