AUDIOCLINIC (Jan. 1987)

Greetings Again

It seems like yesterday, or maybe it was the day before, that I wrote to thank all of you for your interest and wishing you a great 1986. Now I'm writing to express the same sentiments, but for 1987.

My special "thank you" goes out to those who took time from their busy schedules to shed additional light on various topics covered in this column. I have never pretended to have a perfect understanding of each and every detail of the audio field. This fact has shown up now and again, and you readers have been kind rather than destructive in your comments. You are a great bunch!

-J.G.

What's an Audio Signal?

Q. Is an audio signal simply a special sequence of electrical charges which is dependent on the music? If not, what exactly is it? How does an amplifier increase the amperage of an audio signal?

-Shane Voisard, Downers Grove, Ill.

A. Audio signals are a.c. waves whose amplitude and frequency vary with the amplitude and frequency of the sound waves that originally generated them. Watching such a signal on an oscilloscope, you'd see a wave whose height (amplitude) went up as the sound grew louder and whose wave peaks grew closer together, and became sharper, as the pitch of the sound, or the intensity of its high-frequency overtones, went up.

What a basic amplifier does is to increase the voltage and/or power of an audio signal. The changes in voltage which result in audible changes in musical dynamics are maintained.

They are simply scaled up in accordance with the amount of amplification provided by the amplifier.

Shortwave Tuners

Q. Why is it that the only bands on American tuners and receivers are AM and FM? A few "boom boxes" have appeared with shortwave bands, but they are quite uncommon. From what I read, all European tuners have at least one shortwave band and one medium wave band, in addition to AM and FM. There is much of interest to be heard on shortwave. Must the American enthusiast who desires such equipment buy it overseas, or are there ways of obtaining these devices in this country?

-David Breneman, Gig Harbor, Wash.

A. The shortwave or high-frequency band covers frequencies from 3 to 30 MHz, or wavelengths from 10 to 100 meters. It is mainly used for international broadcasting and for non-broadcast applications such as some amateur radio. Shortwave broadcasting and listening are not as popular in this country as they are abroad, so comparatively little shortwave receiving gear is available here.

The medium-wave or medium-frequency band covers wavelengths from 100 up to 1,000 meters, or frequencies from 300 kHz to 3 MHz. The normal AM broadcast band (about 550 to 1,600 kHz) falls within this range, and many foreign radio dials say "MW" (for medium wave) where ours would say "AM." (Our FM band falls in the VHF range, from 30 to 300 MHz.) The long-wave or low-frequency band covers wavelengths from 1,000 to 10,000 meters, or frequencies from 30 to 300 kHz. It is used for broadcasting in some countries.

Shortwave and long wave are sometimes included in home radios (and even in a few car stereos!), and shortwave was common in U.S. home radios before World War II. However, I suspect they'd be rare in component tuners and receivers, even overseas, because the wide frequency response of these components would create problems with shortwave and long-wave broadcasting and reception (static, for example), making them rather painful to listen to. And my experience with the shortwave portions of tuners and small radios has been that the shortwave bands are even more of an afterthought than the standard AM broadcast band usually is.

My recommendation is that you obtain whatever tuner you like, designed for normal AM/FM. Then obtain a good shortwave receiver. Some of these are portable units with superlative characteristics, including scanning, digital frequency entry, single sideband (SSB), and synchronous detectors. I definitely recommend a unit having a synchronous detector because it will reduce much of the effect of selective fading. Also, pick a shortwave receiver with good image rejection. The modern units accomplish this by using dual conversion circuits and low-pass filters. Thus, their front-ends can be broad-banded, allowing for simple digital frequency synthesis.

Because of the need for good i.f. selectivity, most shortwave radios do not produce good high-frequency response. But few shortwave broadcasters transmit high-fidelity sound, anyway; spectrum space won't permit that. Even where a station does broadcast a measure of high frequencies, i.f. selectivity often must be set to its narrowest position in order to remove beats produced by adjacent signals.

Microphonic Phono Cable

Q. My turntable's phono cable is microphonic. To illustrate: At full volume there is a rather loud "thump" from the speakers if I tap the cable with a pencil, and bending the cable sharply creates a sharp "pop." Why?

-O. O. Callaway, Carlsbad, Cal.

A. Shielded cables, like those used to connect a phonograph to other audio components, have capacitance between their shields and inner conductors. By squeezing or otherwise handling the cable, the capacitance changes in value-at least during the actual handling. These changes in cable capacitance are reproduced as noise if the cable is connected to the input of a high-gain audio circuit.

Note that the amount of this micro phonic action depends on the sensitivity of the audio circuits as well as upon the load impedance at the input of the cable. Thus, you will hear less noise if the free end of the cable is connected to a cartridge of low impedance than you will if it's connected to one of high impedance. The amount of noise produced during cable manipulation will be dramatically higher still if no load is connected to the free end of the cable.

In home audio installations, this microphonic problem is rarely encountered because the cables are usually not moved during record playing.

Where cables are subjected to shock (as in some recording situations), types designed to minimize these microphonic effects are available.

Biamping: Pros and Cons

Q. What is the meaning of the term "biamping"? How is it accomplished? What are its advantages and disadvantages?

-Edward Brown, Far Rockaway, N.Y.

A. Biamping is a system in which the woofer in a loudspeaker system is driven by one power amplifier and the tweeter (in that same loudspeaker system) is driven by another amplifier.

In any system with multiple drivers that cover different frequency ranges (such as woofers and tweeters), a crossover must be used to divide the frequency spectrum so that each driver gets only those frequencies it was designed to handle. In most home systems, this crossover is a passive network, built into the speaker cabinet, which divides the power amplifier's output signal. The crossover will have two or more outputs, each delivering a different frequency range to the appropriate drivers. The capacitors, inductors, resistors, and (perhaps) pots that make up such crossovers are usually rather large, because they are built to handle as much power as the amplifier is expected to deliver.

In biamplified systems, an "active" or "electronic" crossover is used instead. The active crossover is connected between the preamp and amplifier. Its active elements (transistors or integrated circuits) make up for signal losses which occur in the network, and may also perform other functions that we needn't get into here.

One advantage claimed for biamped systems is that, because neither amplifier carries the whole audio frequency spectrum, intermodulation effects are reduced. A further advantage is that an active crossover network does not share the passive network's tendency to "ring" at the crossover point. (This ringing is sometimes heard as a peak in frequency response.) Another advantage sometimes offered is that, because the woofer is no longer in series with an inductor, the power amplifier driving it can exercise better damping control over cone motion.

As for disadvantages, cost is certainly one, since a second stereo power amplifier is needed, as is a special crossover network. Another disadvantage is the amount of physical space required by the crossover network and the second power amplifier. Also, systems using receivers or integrated amplifiers may not have the separate preamp-out and amplifier-in jacks which electronic crossovers require.

Perils of Back-Cueing

In addressing the subject of back cueing (see "Audioclinic," March 1986), Mr. Giovanelli was correct in stating that high tracking force was the major cause of "cue burn." Unfortunately, at the radio station where I work, we are still plagued with cue burn, but it now has two different causes. The first of these results from the radical design of the modern stylus. The elliptical stylus has a very bad habit of digging up chunks and even ribbons of vinyl when a record is back cued. The quickest and easiest fix for this problem is to use a stylus that has a biradial or conical shape.

The second problem cannot be solved so easily. The quality of vinyl used in today's records is so poor that back-cueing only once with any shape of stylus tip can damage the lead-in track. Sorry, but the record companies will have to solve this problem.

-John M. Wiley, Chief Engineer, WSIC, Statesville, N.C.

Taking Up a Collection

Q. What do the letters "ASCAP" and "BMI" on record labels mean ?

-Tim Schindler, Mechanicsville, Md.

A. ASCAP stands for American Society of Composers, Authors, and Publishers; BMI stands for Broadcast Music, Inc.

Both of these organizations were created to help ensure that composers, lyricists, and music publishers are paid for the use of their material. They collect fees from the broadcast industry, from performance halls, and from places (such as skating rinks) where music is played as background or accompaniment to other activities. They then determine, by statistical sampling, which songs have been played most and least often, and apportion the fees they have collected according to how much each creator's or publisher's work has been used. These licensing organizations are listed on record jackets for the convenience of commercial users.

Channel Imbalance

Q. I have a problem with channel balance in my system, which includes a preamplifier and power amplifier.

The left channel is louder than the right channel, even when I set the mode selector in the "reverse" position. This occurs on all program sources. I know my speakers are not the cause of the problem because interchanging them does not alter the situation. I can eliminate the problem by setting the balance control to favor the right channel, but this is not a desirable solution. Any ideas? I've already returned the equipment to the manufacturer, who could find nothing wrong.

-Arthur L. Stoddard, Anchorage, Alaska

A. There's a logical process you can follow to find the source of any problem which occurs in only one stereo channel. Working back from the speakers or forward from the signal sources, swap cables between channels, one end at a time, and see what makes the problem move to the opposite channel. (Turn the system off whenever you connect or disconnect a cable to protect your components from accidental damage.) If the problem stays in the original channel when you swap cables, then it must be caused by something that follows the point at which you made the swap; if it switches channels, it is caused by something which precedes that point.

For example, if you had not already eliminated the speakers from consideration (as you have), you might disconnect the cable feeding the right speaker and connect it to the left one, and vice versa. If the left speaker is still louder, then the problem would have to lie either in that speaker or the room's acoustics. If the right speaker becomes louder, then the speaker and the room are not at fault-one speaker is simply getting a louder signal than the other.

Assuming the right speaker did get louder, we must trace further back. Swap the cables' other ends, so that the one which was formerly connected to the amplifier's left output terminal is now on the right terminal and the former right output cable is now on the left terminal. If the problem stays in the same channel, then the speaker cable is at fault; if it swaps channels, then the cable is okay.

Continue this process through the system until you find the trouble's cause. Your preamp's "reverse" switch has the same effect as switching the cables between its left and right output jacks. This absolves your signal sources--phono, tuner, etc. So the preamp, the power amp, and the cables between them are the most likely culprits.

If all the components are okay, then the problem may be in your room's acoustics. Try moving the right speaker to the left speaker's position, and vice versa, while leaving each speaker connected to its original amplifier channel. (If your speaker cables aren't long enough for this, substitute longer ones.) If the left channel still sounds louder, then the room is at fault.

Changing the room's acoustics could prove complex and expensive, but adjusting the balance control to favor the right channel (with the speakers returned to their normal locations) will solve the problem easily, and for free. That's what the control is for.

Correcting High-Frequency Hearing Loss

Q. Many people have hearing losses of 10 to 40 dB in the 4-kHz frequency region. This is usually the result of exposure to excessive sound levels. If a narrow-band parametric equalizer were used to boost this frequency region in proportion to the hearing loss, would the affected individual experience music from his audio system as it was recorded ?

-Britton K. Ruebush, Albuquerque, N.M.

A. It does seem to me that, if one accurately measures a hearing loss as being some given number of dB in some given frequency region, this loss could be corrected by producing the mirror-image curve with a parametric equalizer.

If the measurement is made at just one frequency in a given range, though, one wouldn't know how wide the "notch" is. Thus, if we measure a 4 kHz frequency and find it down by 10 dB, we still wouldn't know what the loss is at 3,900 Hz or 4,100 Hz; we also wouldn't know whether the 4-kHz point is the notch's center frequency or its deepest point. We must measure the whole notch, over a range of frequencies, to know just what equalizer curve would be its mirror image.

It's also possible that the increased sound level at the boosted frequencies might make the sound seem distorted to the listener; any damage the listener's hearing has already suffered from prolonged sound pressure might aggravate that effect. I also wonder if hearing those boosted frequencies for prolonged periods might not further aggravate the listener's hearing impairment.

(Source: Audio magazine, Jan. 1987, JOSEPH GIOVANELLI)

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