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Q. The literature included with my headphones states "the transducer is a microphone-derived device." How can a microphone element be used in a headphone?
-Timothy Balogh, Kingston, Pa.
A. A transducer is simply something which converts mechanical energy into electrical impulses (microphone) or converts electrical impulses into mechanical energy (loudspeaker) through movement of a coil of wire across a magnetic field. The headphone contains a miniature speaker, and an engineer for the firm that put out the literature found that a microphone assembly could be adapted for use in the headphones as a speaker element.
Q. Please explain the purpose of an analog or digital delay system in high fidelity or public address applications.
- James Mauro, New Brunswick, N.J.
A. A delay system can be used in quite a number of applications, depending upon the amount and type of delay it is capable of producing. In the most popular current applications, these delay systems are designed to delay sound a small fraction of second before feeding it into a pair of speakers located in the rear to create the reverberation heard in a concert hall.
Most such delay units produce this effect from the standard two-channel recordings, rather than a four-channel source.
Another application is for people seated in the rear of relatively large rooms who will hear the sound from the PA in the back of the room before they hear the "live" sound from the stage. This is often confusing and annoying. To overcome this problem, the PA's rear sound is delayed so that the sound from the speakers and the sound from the stage are heard at the same time. In such PA installations, speakers for listeners sitting close to the stage will not be fed the delayed signal.
There are several all-electronic means by which delay is introduced.
In some devices the signal is converted into digital information and shifted through a series of electronic circuits.
The time needed to shift from one circuit to the next is determined by a "clock," and the speed of this clock or oscillator is variable. Thus, the time used to shift the digitized signal from one circuit to the next can be varied to produce the variable delay times. Because the signal is shifted from one circuit to another, any one of these circuits can be used as the final signal point, rather than merely extracting the signal from the far end of the chain of many such circuits or "shift registers." By extracting the signal from a point other than the far end of the chain, we have another means of varying the time delay. The output signal is then reprocessed into analog information and processed, as with any other audio signal.
The other major means of making such delay lines uses "charge coupled" devices, which are strictly analog, to produce the delays. For a discussion at this level of sophistication, there is no essential difference in what the two basic circuit types do.
As described, these arrangements would produce a single delay. You could either listen to a delayed signal alone or you could listen to the main, or direct, signal mixed with this delay.
However, it is possible to introduce signals from several of the "part-way" points in the circuit, mixing all of them with the direct signal. This tends to create a more densely reverberant sound. The greater the number of such included delay points, the closer the sound will be to true reverberation, rather than distinct repeats of the original signal.
Another early and very basic delay system used two or more tape play heads, generally for PA systems where distinct repeats are called for. The space between the successive heads is often adjustable to create whatever delay times are required. Where the head spacing is fixed, the tape speed is varied to produce the same result.
Other delays can be produced by feeding the signal into something like a coiled garden hose and then extracting it at the far end. The delay is determined by the length of the tubing and the speed of sound in air. (See, "Construction of a Madsen-System Delay Tube," Audio, April and May, 1971.)
Portable Auto Equipment
Q. I am working on a portable music system using equipment designed for use in my car. For home use, what kind of d.c. adaptor would I need? How much power is needed to keep the system running efficiently?
-John Keller, Chicago, Ill.
A. Each separate component in your system requires a certain amount of power (of course, the speakers are not considered, for the power they require is supplied by either your amplifier or receiver, rather than from a special power supply). Because the various components are designed to operate on 12 volts d.c., we know that they must be supplied with that kind of d.c. power. The instruction manual which accompanies each piece of equipment will supply you with information as to current number of amps required to operate each piece of equipment.
Add together the number of amperes required by each component you plan to use. Actually, the power supply should be capable of handling a somewhat higher current than the combined drain of all your equipment-this is a safety margin designed to keep the power supply from overheating.
Power supplies of this kind are available from stores selling equipment to amateur radio operators. These power supplies are then plugged into the home a.c. socket, and each component will automatically draw the amount of current for which it is designed.
Batteries are capable of delivering current for a specified amount of time.
For example, if the automotive battery is capable of delivering 35 ampere-hours, and your equipment takes 10 amperes, then the battery will have to be recharged after 3' hours. The dealer from whom you purchase your battery should be able to assist you in obtaining the correct battery charger.
Q. I have noticed that on many of the tapes I have recorded in the past several years that the "s" sound is smeared. I record at 3 3/4 ips and clean and demagnetize my heads regularly. Do you suspect that there is anything wrong with my tape deck?
-Howard Eiserike, College Park, Md.
A. Your problem is probably due to recording at too high a level, resulting in tape saturation. Saturation is most likely to occur at the high frequencies where the "s" sound occurs, and because of the substantial amount of treble boost provided by the record electronics at 3 3/4 ips. Try reducing your recording level by about 3 to 6 dB.
(Source: Audio magazine, April. 1979; Joseph Giovanelli )
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