Make: Electronics

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Make: Electronics Page 34

by Charles Platt


  Plastic shoebox. Quantity: 1.

  Procedure

  The purpose of the audio amplifier chip is to provide enough power to get a decent amount of sound out of your loudspeaker. The purpose of using a 5-inch speaker is to enable you to hear lower-frequency sounds than the baby speakers that we have used previously. Bass notes have long wavelengths that small speakers are not able to generate effectively.

  Maybe you remember from building the intrusion alarm that a speaker makes much more noise if you prevent the sound waves from the back of the cone from cancelling the sound waves from the front of the cone. The obvious way to achieve this is by enclosing the speaker in a box. I suggest a plastic box, because they’re cheap, and we don’t care too much about sound quality as long as we can hear at least some of the low frequencies. Figure 5-39 shows the speaker bolted into the bottom of a plastic box, and Figure 5-40 shows the box turned upside-down after snapping its lid into place.

  Figure 5-39. A resonant enclosure is necessary if you want to hear some bass (lower frequencies) from your speaker. A cheap plastic shoebox is sufficient for demo purposes.

  Figure 5-40. Drill some half-inch holes in the bottom of the box, then bolt the speaker in place, running a wire out through a hole in one end. Snap on the lid, and you’re ready for not-quite-high-fidelity audio.

  Normally, a speaker should be mounted in a cabinet of heavy, thick material that has a very low resonant frequency—below the limits of human hearing. To minimize the resonance of the shoebox, you can put some soft, heavy fabric inside it before you snap the lid on. A hand towel or some socks should be sufficient to absorb some of the vibration.

  Adding an Amplifier

  Back in the 1950s, you needed vacuum tubes, transformers, and other power-hungry heavyweight components to build an audio amplifier. Today, you can buy a chip for about $1 that will do the job, if you add a few capacitors around it, and a volume control. The TEA2025B that I’m recommending is intended for use in cheap portable cassette players and CD players, and can work in stereo or mono mode, from a power supply ranging from 3 to 9 volts. With 9 volts and the two sides of the chip bridged together to drive one 8Ω speaker, it can generate 5 watts of audio power. That doesn’t sound much compared with a typical home theater system rated at 100 watts per channel, but because loudness is a logarithmic scale, 5 watts will be quite enough to irritate any family members in the same room—and possibly even in other rooms.

  If you can’t find the TEA2025B chip, you can use any alternative listed as an audio amplifier. Try to find one that is designed to drive an 8Ω speaker with up to 5 watts in mono mode. Check the manufacturer’s data sheet to see where you attach capacitors around it. Note carefully whether some of the capacitors have no polarity marked, even though they have fairly high values, such as 100 μF. These capacitors must function regardless of which way the alternating current is flowing, and I’ve marked them “NP” in my schematic in Figure 5-41, meaning “nonpolarized.” (You may find them identified as “bipolar” or “BP” in parts catalogs.) As noted in the shopping list, you can put two 220 μF capacitors in series, negative-to-negative, to get the same effect as a single 100 μF nonpolarized capacitor.

  For this project, it’s essential to include the regular 100 μF electrolytic smoothing capacitor across the power supply. Otherwise, the amplifier will pick up and—yes, amplify—small voltage spikes in the circuit.

  Figure 5-41. The audio amplifier chip should be wired with capacitors around it as shown, “NP” denoting the ones that are not polarized. The acronym“BP,” meaning bipolar, is also often used to mean the same thing. The output from pins 2 and 15 of the chip can be passed through a coil or a 10 μF capacitor to demonstrate audio filtering.

  The input shown in the schematic can receive a signal from a typical media player, such as a portable MP3 player, CD player, or cassette player. To connect its headphone jack to the breadboard, you can use an adapter that converts it to a pair of RCA-type audio jacks, and then stick a wire into one of them as shown in Figure 5-42. The wire will connect to the 33K resistor on the breadboard circuit. The chromed neck of the RCA jack (which is sometimes gold-plated, or at least gold-colored) must be connected with the negative side of your power supply on the breadboard; otherwise, you won’t hear anything. You can ignore the second output on the adapter, because we’re working in mono, here, not stereo.

  Figure 5-42. To sample the output from the headphone socket of a music player, you can use this adapter and push the stripped end of a piece of hookup wire into one of the sockets. Then use alligator clips on a jumper wire to connect the audio to your breadboarded circuit. Don’t forget to use an additional jumper wire to connect the outside of the socket to the negative side of the power supply on the breadboard. Because we’re only using one speaker, the amplifier is connected to only one of the stereo outputs from your player. The other is ignored.

  The 33K resistor is necessary to protect the amplifier from being overdriven. If you don’t get enough volume using your music player, decrease the 33K value. If the music is too loud and distorted, increase the value. You can also try omitting or increasing the 10K resistor next to it, which is included in an effort to reduce background hum noise.

  I’ve shown two switches at the top of the schematic: one to bypass a coil, the other to bypass a capacitor. You can use alligator clips instead, as long as you can easily compare the sound when each of the components is inserted into the circuit.

  Figure 5-43 shows a coil consisting of a spool of hookup wire being used. The red and black alligator clips resting loose on top of the shoebox will go to the output from the chip (on pins 2 and 15). There is no polarity; it doesn’t matter which clip goes to which pin.

  Figure 5-43. The red and black alligator clips, lying on top of the shoebox, should connect with the output from your amplifier chip. The red jumper wire passes the signal through a coil of hookup wire on its way to the speaker. Note the change in sound when you short out the coil.

  Begin by turning the volume control on your music source all the way down before you apply power. Don’t be surprised if you hear humming or crackling noises when you activate the amplifier; it will pick up any stray voltages, because in this simple experiment, I haven’t suggested that you should shield the input, and the amplifier circuit can pick up noise, as its wires can act like antennas.

  Note that you may also get additional unwanted sound if you use the amplifier on a conductive desktop surface. Remove any aluminum foil or conductive foam for this project.

  Make sure that your player is playing music, and slowly turn up its volume control until you hear it. If you don’t hear anything, you’ll have to check for circuit errors.

  Now comes the interesting part. Insert the 100-foot spool of hookup wire between one output from the amplifier, and one input of the speaker (it doesn’t matter which one), or if you used switches, open the switch that bypasses the coil. You should find that the music loses all its high-end response. By comparison, if you disconnect the coil and substitute a 10 μF capacitor, you should find that the music sounds “tinny,” meaning that it loses all its low range, leaving only the high frequencies.

  You’ve just tested two very simple filters. Here’s what they are doing:

  The coil is a low-pass filter. It passes low frequencies but blocks high frequencies, because brief audio cycles don’t have time to overcome the coil’s self-inductance. A bigger coil eliminates a wider range of frequencies.

  The capacitor is a high-pass filter. It passes high frequencies and blocks low frequencies because longer audio cycles can fill the capacitance, at which point the capacitor stops passing current. A smaller capacitor eliminates a wider ranger of frequencies.

  You can go a lot farther into filter design, using complex combinations of coils and capacitors to block frequencies at any point in the audible spectrum. Search online f
or audio filter schematics—you’ll find hundreds of them.

  Crossover Networks

  In a traditional audio system, each speaker cabinet contains two drivers—one of them a small speaker called a tweeter, which reproduces high frequencies, the other a large speaker known as a woofer, which reproduces low frequencies. (Modern systems often remove the woofer and place it in a separate box of its own that can be positioned almost anywhere, because the human ear has difficulty sensing the direction of low-frequency sounds.)

  The schematic that you just looked at and may have constructed is known as a “crossover network,” and truly hardcore audiophiles have been known make their own (especially for use in car systems) to go with speakers of their choice in cabinets that they design and build themselves.

  If you want to make a crossover network, you should use high-quality polyester capacitors (which have no polarity, last longer than electrolytics, and are better made) and a coil that has the right number of turns of wire and is the right size to cut high frequencies at the appropriate point. Figure 5-44 shows a polyester capacitor.

  Figure 5-45 shows an audio crossover coil that I bought on eBay for $6. I was curious to find out what was inside it, so I bought two of them, and took one apart.

  First I peeled away the black vinyl tape that enclosed the coil. Inside was some typical magnet wire—copper wire thinly coated with shellac or semitransparent plastic, as shown in Figure 5-46. I unwound the wire and counted the number of turns. Then I measured the length of the wire, and finally used a micrometer to measure the diameter of the wire, after which I checked online to find a conversion from the diameter in mils (1/1,000 of an inch) to American wire gauge.

  As for the spool, it was plain plastic with an air core—no iron or ferrite rod in the center. Figure 5-47 shows the spool and the wire.

  Figure 5-44. Some nonelectrolytic capacitors have no polarity, such as this high-quality polyester film capacitor. However, they tend to be much more expensive, and are hard to find in values higher than 10 μF.

  Figure 5-45. What exotic components may we find inside this high-end audio component that’s used with a subwoofer to block high frequencies?

  Figure 5-46. The black tape is removed, revealing a coil of magnet wire.

  Figure 5-47. The audio crossover coil consists of a plastic spool and some wire. Nothing more.

  So here’s the specification for this particular coil in an audio crossover network. Forty feet of 20-gauge copper magnet wire, wrapped in 200 turns around a spool of 1/16–inch-thick plastic with a hub measuring 7/8 inch in length between the flanges and 1/2-inch external diameter. Total retail cost of materials if purchased separately: probably about $1, assuming you can find or make a spool of the appropriate size.

  Conclusion: there’s a lot of mystique attached to audio components. They are frequently overpriced, and you can make your own coil if you start with these parameters and adjust them to suit yourself.

  Suppose you want to put some thumping bass speakers into your car. Could you build your own filter so that they only reproduce the low frequencies? Absolutely—you just need to wind a coil, adding more turns until it cuts as much of the high frequencies as you choose. Just make sure the wire is heavy enough so that it won’t overheat when you push 100 or more audio watts through it.

  Here’s another project to think about: a color organ. You can tap into the output from your stereo and use filters to divide audio frequencies into three sections, each of which drives a separate set of colored LEDs. The red LEDs will flash in response to bass tones, yellow LEDs in response to the mid-range, and green LEDs in response to high frequencies (or whatever colors you prefer). You can put signal diodes in series with the LEDs to rectify the alternating current, and series resistors to limit the voltage across the LEDs to, say, 2.5 volts (when the music volume is turned all the way up). You’ll use your meter to check the current passing through each resistor, and multiply that number by the voltage drop across the resistor, to find the wattage that it’s handling, to make sure the resistor is capable of dissipating that much power without burning out.

  Audio is a field offering all kinds of possibilities if you enjoy designing and building your own electronics.

  Theory

  Waveforms

  If you blow across the top of a bottle, the mellow sound that you hear is caused by the air vibrating inside the bottle, and if you could see the pressure waves, they would have a distinctive profile.

  If you could slow down time and draw a graph of the alternating voltage in any power outlet in your house, it would have the same profile.

  If you could measure the speed of a pendulum swinging slowly to and fro in a vacuum, and draw a graph of the speed relative to time, once again it would have the same profile.

  That profile is a sine wave, so called because you can derive it from basic trigonometry. In a right-angled triangle, the sine of an angle is found by dividing the length of the side opposite the angle by the length of the hypoteneuse (the sloping side of the triangle).

  To make this simpler, imagine a ball on a string rotating around a center point, as shown in Figure 5-48. Ignore the force of gravity, the resistance of air, and other annoying variables. Just measure the vertical height of the ball and divide it by the length of the string, at regular instants of time, as the ball moves around the circular path at a constant speed. Plot the result as a graph, and there’s your sine wave, shown in Figure 5-49. Note that when the ball circles below its horizontal starting line, we consider its distance negative, so the sine wave becomes negative, too.

  Why should this particular curve turn up in so many places and so many ways in nature? There are reasons for this rooted in physics, but I’ll leave you to dig into that topic if it interests you. Getting back to the subject of audio reproduction, what matters is this:

  Any sound can be broken down into a mixture of sine waves of varying frequency and amplitude.

  Or, conversely:

  If you put together the right mix of audio sine waves, you can create any sound at all.

  Suppose that there are two sounds playing simultaneously. Figure 5-50 shows one sound as a red curve, and the other as pale blue. When the two sounds travel either as pressure waves through air or as alternating electric currents through a wire, the amplitudes of the waves are added together to make the more complex curve, which is shown in black. Now try to imagine dozens or even hundreds of different frequencies being added together, and you have an idea of the complex waveform of a piece of music.

  Figure 5-48. If a weight on the end of a string (length b, in the diagram) follows a circular path at a steady speed, the distance of the weight from a horizontal center line (length a, in the diagram) can be plotted as a graph relative to time. The graph will be a sine wave, so called because in basic trigonometry, the ratio of a/b is the sine of the angle between line b and the horizontal baseline, measured at the center of rotation. Sinewaves occur naturally in the world around us, especially in audio reproduction and alternating current.

  Figure 5-49. This is what a “pure” sinewave looks like.

  Figure 5-50. When two sinewaves are generated at the same time (for instance, by two musicians, each playing a flute), the combined sound creates a compound curve. The blue sinewave is twice the frequency of the red sinewave. The compound curve (black line) is the sum of the distances of the sinewaves from the baseline of the graph.

  Theory

  Waveforms (continued)

  You can create your own waveform as an input for your audio amplifier with the basic astable 555 timer circuit shown in Figure 5-51. You have to be careful, though, not to overload the amplifier input. Note the 680K series resistor on the output pin of the timer. Also note the 500Ω potentiometer.

  Figure 5-51. A 555 timer is
wired in astable mode using the component values shown here to generate a wide range of audible frequencies when the 100K potentiometer is adjusted. After the output is reduced in power, it can feed into the amplifier chip that was used previously.

  Disconnect your music player and connect the output from the 555 circuit to the input point (the 33K resistor) in the amplifier circuit shown earlier in Figure 5-41. You don’t have to worry about a separate connection on the negative side as long as the 555 timer shares the same breadboard and the negative side of its power supply.

  Make sure that the 500Ω potentiometer is turned all the way to short the output from the timer to the negative side of the power supply. This functions as your volume control. Also make sure the 100K potentiometer is in the middle of its range. Switch on the power and slowly turn up the 500Ω potentiometer until you hear a tone.

  Now adjust the 100K potentiometer to create a low-pitched note. You’ll find that it doesn’t have a “pure” sound. There are some buzzing overtones. This is because the 555 timer is generating square waves such as those shown in Figure 5-52, not sine waves, and a square wave is actually a sum of many different sine waves, some of which have a high frequency. Your ear hears these harmonics, even though they are not obvious when you look at a square-shaped waveform.

  Route one of the connections to your loudspeaker through your spool of hookup wire, and now you should hear a much purer tone, as the buzzing high frequencies are blocked by the self-inductance of the coil. Remove the coil and substitute the 10 μF capacitor, and now you hear more buzzing and less bass.

 

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