- Email: [email protected]
Test Procedures
CHAPTER 22
Test Procedures Chapter 2 explained how audio devices are measured by professionals using modern test gear. Sadly, many people can’t afford $10,000 or more for a precision analyzer, or even $400 for a basic oscilloscope. But there are many tests you can do yourself without measuring equipment. For example, the Wave files included with Chapter 3 let you assess the audibility of low-level artifacts just by listening through your own speakers or headphones. That chapter also explained how the FFT display available in many audio editor programs shows harmonics and noise at levels too soft to hear. As explained in Chapter 8, to measure the distortion of 32-bit floating point “summing” math used by most DAW software, I applied 30 sequential gain changes to a pure sine wave, then analyzed the added distortion using Sound Forge’s FFT feature. With a very good multimeter you can measure the frequency response of outboard audio gear, though oscilloscope software that uses your sound card for its input is also available. Even better is getting a used “real” oscilloscope, which will cost much less than a new one. Chapter 20 showed how an inexpensive small diaphragm omni condenser mic coupled with room measuring software can measure the frequency response of loudspeakers and microphones. So it’s possible to do your own equipment testing using the tools you already own or can obtain inexpensively. There are two very different types of audio tests: measuring tests and listening tests. For assessing raw fidelity, measuring is the better choice because it’s 100 percent repeatable, and the results are highly accurate, assuming it’s done properly. But there’s nothing wrong with listening tests when done correctly and you account for the limitations of hearing. The gold standard for all subjective comparisons—not just audio gear—is the double-blind test. In a sighted test, the person listening knows what source is playing and can be influenced by expectation. A single-blind test is much better, where someone else switches the A and B playback without winking or otherwise letting on. But with a double-blind test, even the person running the test doesn’t know what’s playing at the moment, which avoids any chance of accidentally giving a clue to the test subject. I’m satisfied using single-blind tests, and I’ve done that with friends many times. In my home studio, the person listening is behind me, unable to see my hands or facial expression. You can even test yourself blind, either using ABX software or by closing your eyes while clicking linked solo buttons, as described in Chapter 3. 607
608 Chapter 22
Figure 22.1: A loop-back uses DAW software to play test signals from a sound card’s output through the device to be measured. The result is then recorded back to the DAW for later analysis.
Whether your intent is to measure with a meter or just listen with your ears, the first thing needed is a signal source. Obviously, music is a fine source if the intent is to identify quality differences between two devices such as sound cards, though you must match both volume levels to within 0.1 dB. For accurate and repeatable measuring, using test signals makes more sense. I create sine wave test tones in Sound Forge, and most other audio editor programs can do this. The “pink_noise.wav” file from Chapter 20 can be used when a noise source is more appropriate. Sound Forge can also create a sine wave that sweeps the full audio range, or just a portion. The basic setup is called a loop-back; the output of a sound card or external converter is sent to the hardware device being tested, then looped back to its own input or the input of another sound card, as shown in Figure 22.1. “A person with one voltmeter knows what the voltage is. A person with two voltmeters never really knows for sure.”
Although you might think that testing audio gear requires a meter or oscilloscope, using a decent-quality sound card and audio software is easier and more accurate than inexpensive meters. Most budget meters don’t even state a frequency response, and when they do, it’s often limited to 10 KHz or even lower. A competent sound card—even an inexpensive consumer model—will be reasonably clean and flat from 20 Hz to 20 KHz. Then you can record the test signal(s) once and analyze the recording in different ways later. However, when using a sound card for testing, I suggest you first assess its quality. Connect the output of the sound card directly back to its own input, then record your test signals to learn the sound card’s own frequency response and how much noise and distortion it adds.
Test Procedures 609
Frequency Response The best signal source for measuring frequency response is a series of sine waves at low, mid, and high frequencies, or a slowly swept sine wave to see even more detail. Depending on what you’re testing, you may not need to measure at many frequencies. Often, 20 Hz, 1 KHz, and 20 KHz are sufficient for devices expected to be flat. If a device rolls off at the frequency extremes and you want to see the trend, you could also measure at 50 Hz and 10 or 15 KHz. When measuring gear that contains a transformer, or an analog tape recorder, using a sweep ensures you won’t miss any irregularities. Either way, I generally create test files with a peak level of 26 dB. Sine waves should sustain each frequency for 5 to 10 seconds, giving you time to read the level meter later. Then load the file to a track in your DAW software and record the output of the device being tested to a second track. Another approach uses a stand-alone signal generator program that plays tones or noise through your sound card, which you route to the device being tested. Then you can record the signal from the gear being tested using any basic audio editor. I use Vincent Burel’s excellent and free LF Generator, which can output various wave shapes at any audio frequency, including sweeps, as well as pink noise. You can also test your hearing with sine waves, but you should first verify how high a frequency your loudspeakers can produce using a decent small diaphragm omni condenser mic and room measuring software as described in Chapter 20. When playing very highfrequency tones through a loudspeaker, move your head slightly while listening to be sure you’re not in a null spot. High frequencies are surprisingly directional due to loudspeaker beaming and also comb filtering that occurs naturally in rooms. Of course, using good headphones avoids these problems as long as you’re certain they can reproduce frequencies higher than you can hear. Another method plays a 96 KHz recording of a tambourine or jangling keys, or other source known to contain frequencies beyond 20 KHz, as verified with an FFT. The tambourine.wav file from Chapter 3 is ideal for this. Sweep the frequency of a low-pass filter plug-in downward starting from above 20 KHz until you can just barely hear a difference, then note the frequency on the plug-in.
Ringing A 20 Hz square wave will tell you more about an audio device in two seconds than all the fancy test gear in the world. Square waves are the most challenging signal for any device to pass accurately, and they reveal not only response errors but also ringing that can indicate circuit instability. Figure 22.2 at the top shows what ringing looks like, using a 20 Hz square wave before and after applying a high-Q boost at 1 KHz. The ringing is evident as each rapid wave transition creates a small sine wave that decays over time. The middle figure shows the original square wave after adding a 6 dB per octave low-pass filter set to
610 Chapter 22
Figure 22.2: A low-frequency square wave is an excellent test signal because it readily reveals ringing as well as frequency response errors. From top to bottom: ringing, high-frequency roll-off, low-frequency roll-off.
filter content above 1 KHz. Here you can see the rounding of the edges at each transition, exactly as happens with a passive resistor-capacitor filter as shown earlier in Figure 8.2. Applying a high-pass filter, at the bottom, causes the flat top and bottom portions of each cycle to droop as the sustained “DC” component of each cycle falls off.
Harmonic Distortion Distortion is not difficult to measure, requiring only a high-quality sine wave source and an FFT analyzer, which is included in many audio editor programs. Note that an FFT display doesn’t tell you a single distortion amount in percent or dB. Rather, you see all the individual components that, if added together, would give the total amount of distortion. But an FFT shows the nature of the distortion, which is more useful for assessing audibility than a single number. Depending on what type of device you’re testing, you might measure at 1 KHz only or over the full audio range. Transformers tend to have higher distortion at low frequencies, so you could test at 50 Hz or even 20 Hz. Absolute signal level is also a factor; some gear has more distortion at higher levels, but some devices are worse at soft levels. It’s important to
Test Procedures 611 start with a pure sine wave having known low distortion, though most software creates reasonably pure tones. Again, you should verify the source purity, as I did for the summing math test in Chapter 8. I examined the original sine wave with an FFT and confirmed that all the residual artifacts were safely below 2100 dB. This equates to 0.001 percent distortion plus noise, which is lower than most outboard gear you’re likely to test. If all of the individual artifacts are below 100 dB, it’s unlikely you’ll ever hear them. And that’s the real point of such testing.
IM Distortion IM distortion is more complicated to measure than simple harmonic distortion, but it’s still possible using the same methods. IM distortion occurs when two or more frequencies play together and create new sum and difference frequencies. For example, if you play music or test tones containing an A note at 440 Hz and also the C# note above at 554 Hz, IM distortion adds new tones at 554 1 440 5 994 Hz and 554 2 440 5 114 Hz. Note that neither 994 Hz nor 114 Hz is a standard musical note pitch. This is one reason IM distortion generally sounds more obnoxious than harmonic distortion whose components are musically related to the source frequency. To test IM distortion in audio gear, you’ll create sine waves at two different frequencies, then mix them together and save the result in a file to play through the device under test (DUT). Standard IMD tests play 19 KHz and 20 KHz at equal volumes, then measure the level of the resulting 1 KHz difference frequency. (The 39 KHz sum is considered irrelevant.) But IMD tests can use other frequencies, which is needed when measuring loudspeakers and microphones. Since two frequencies result from IM distortion—the sum and the difference—it’s not difficult to determine the approximate distortion percentage. However, large amounts of IM distortion also create additional components that are harmonics of the sum and difference frequencies. Again, this is less important than seeing the big picture to verify that whatever distortion you measure is too soft to be objectionable. When measuring the IM distortion of loudspeakers, you should choose frequencies that are both played by the woofer or by the tweeter at the same time. If a speaker crosses over from woofer to tweeter at 2 KHz, using 500 Hz and 3 KHz will not tell you the amount of IM distortion from either driver. To test the woofer, you might use 100 Hz and 180 Hz, which results in 280 Hz and 80 Hz for the sum and difference frequencies, letting you ignore the inevitable harmonic distortion at multiples of 100 Hz and 180 Hz. If you played 100 Hz and 300 Hz, the resulting 200 Hz and 500 Hz IM components will include regular harmonic distortion of the 100 Hz source at those same frequencies. Likewise, 5 KHz and 8 KHz are suitable for testing tweeters, but not 5 KHz and 10 KHz. You should also choose frequencies that are reasonable for the drivers to play at high volumes, where their
612 Chapter 22 distortion is greatest. A sustained loud 19 KHz tone is very demanding of a tweeter, and isn’t representative of normal music. Playing such a high frequency for more than a few seconds at high volume can damage the driver, and maybe your hearing, too. When measuring microphone IM distortion, you’ll again choose frequencies that are reasonable for the speaker and mic to deal with at high volume levels. The best way to measure microphone IMD is with two speakers, with one tone played through each speaker. This avoids contaminating the resulting sum and difference frequencies with the loudspeaker’s own IM distortion, which is typically higher than most microphones at low frequencies. You’ll create separate files for each test frequency, panning them hard left and right for playback. The microphone should be halfway between each speaker and also onaxis to capture the flattest response from both speakers. Assuming your monitoring is set up correctly, with each speaker angled toward your ears, just put the microphone where you listen at the same height as the speakers. Figure 22.3 shows the IM distortion I measured for my audio-technica 4033 and DPA 4090 microphones using this method, by playing 300 Hz and 500 Hz through the JBL 4430 speakers in my home studio. As you can see, the AT 4033 has half the IM distortion of the DPA 4090. Adding the sum and difference signal levels, the AT is about 0.2 percent IMD versus about 0.4 percent for the DPA. You can also see harmonic distortion at various multiples of the 300 Hz and 500 Hz tones. The blip to the left of the 276 dB label is the 600 Hz second harmonic of the 300 Hz tone, and the 1 KHz second harmonic of the 500 Hz source tone is just to the right of the same label at an even higher level. It’s not visible here, but when you hover the mouse cursor over the graph, a pop-up balloon shows the frequency and level at that location. This is how I identified the exact frequency and level of the various peaks. It’s difficult to measure harmonic distortion of speakers or microphones unless the distortion for one of them is known at a given frequency and SPL, or is at least known to be lower than the other device being tested. However, THD and IMD are not unrelated, so excessive IMD implies that THD will also be high, and likewise for low amounts. The 600 Hz and 1 KHz second harmonics in Figure 22.3 could be due to either the microphone or the speakers, but the 200 Hz and 800 Hz components can only be due to the microphone. Since the second harmonic from the speakers is louder than the sum and difference frequencies from the mic, it’s reasonable to conclude that the speaker’s distortion is greater. However, frequencies recorded by a microphone can align with a peak or null in the room’s response and affect the results, possibly by a large amount. I suggest measuring the raw response first to get a baseline for all four frequencies, then do the IMD tests without moving the microphone. Finally, I’ll point out that home loudspeaker and microphone tests as described here will never be as accurate as measurements made with real test equipment in an anechoic
Test Procedures 613
Figure 22.3: Microphone IM distortion was measured by playing 300 Hz through one speaker and 500 Hz through another, while recording the output of each microphone. An FFT display shows the levels of the resulting sum and difference frequencies.
chamber. But you can definitely get useful data. So trust, but verify. If your results don’t agree with what’s expected, question the results until you’re confident they’re correct. Indeed, question everything, including measurements that seem too good to be true.
Null Tests As explained in Chapter 1, a null test works by subtracting two audio signals to see what remains. If the result is total silence, then the signals are by definition identical. And if
614 Chapter 22 there is a residual, its audibility can be assessed based on the level and frequency distribution, or simply by listening. The beauty of a null test is that it reveals all differences between two audio signals, including differences you might not even be looking for. If someone claims playing Wave files from one hard drive sounds different from playing them from another hard drive, a null test will tell you for certain whether or not that’s true. You simply copy a file to both hard drives and see if they null when played in a DAW program. To subtract audio signals, you reverse the polarity of one, then mix it with the other at the same volume. Having both sources match exactly in level and phase is the key to a successful null test, and it can be more difficult than you might imagine. By watching the residual output on a wide-range VU meter that displays down to the noise floor, you’ll tweak the level of one signal to get the best null. Then, if you’re comparing recordings, you can slide one in time relative to the other to avoid time and phase differences. If the result is total silence, or at least well below 280 dB, you can be confident that both sources are audibly identical. Most null tests are done in a DAW program, placing the Before and After versions on separate tracks, with both files aligned to start at the same time and with equal volumes. If the tracks are off in time by even one sample, or their levels differ by even 0.1 dB, identical files that would have nulled to silence will yield some amount of residual. I used null tests in my AES Audio Myths video on YouTube to disprove two common myths: One myth is that audio plug-ins have a “sweet spot” signal level, which if exceeded harms sound quality. The other is that digital EQ cannot be countered exactly. This is a great application for null tests because both tracks contain the same source file, avoiding the need for time alignment and level matching. Figure 22.4 shows the setup that sends a mix through an EQ plug-in after raising the volume by 18 dB. The mix was normalized to peak at 21 dB, so the audio through the plug-in reaches 17 dB above digital zero. Track 1 contains the mix file, and Track 2 plays the same file with boost applied by the Track Trim to be before the EQ. The Sonalksis FreeG freeware volume plug-in restores the level after the EQ. This plug-in is great for null tests because it lets you adjust the volume in 0.01 dB steps, though only a whole number dB amount was needed here. Track 2 also has its polarity switch engaged, needed to create a null. Otherwise, when both tracks play, the output would be 6 dB louder than one track, rather than silence. Figure 22.5 shows the setup for the second null test to confirm that digital EQ can be countered with equal but opposite settings. Here again, the same music file is placed on both tracks, aligned exactly to single sample timing. In this test, Track 1 contains two instances of the same EQ plug-ins, with three bands set to opposite amounts of boost and cut, with the same bandwidth (Q). Track 2 uses no plug-ins, but its polarity is reversed to cancel Track 1.
Test Procedures 615
Figure 22.4: This null test setup confirms that modern 32-bit plug-ins are immune to overload, even when sent signals that greatly exceed the digital zero clipping point.
Figure 22.5: This null test proves that EQ can be reversed exactly using a second plug-in having equal but opposite settings.
616 Chapter 22 The first test proves that 32-bit floating point plug-ins are not overloaded by signals that exceed digital zero. As long as the volume is reduced somewhere later in the chain, before the mix is rendered to a file or sent out the sound card for monitoring, clipping distortion will not occur. The second test proves that EQ can be reversed exactly as long as both plug-ins are set precisely opposite. When playing either of these DAW projects, the result is total silence down to the 290 dB floor of the output bus level meter. Rendering the mix to a Wave file and examining that in an audio editor further confirm that the file contains only silence. Null tests can also be done in real time using power amplifiers and other audio gear by subtracting the device’s input and output signals. However, this is more complicated to set up. When testing something that amplifies the signal, such as a power amplifier, you need a way to make the input and output levels exactly equal. This can be done with variable resistors that are passive and thus won’t add distortion, but you also need a way to reverse the polarity of one signal. That requires an additional circuit whose own distortion or nonflat frequency response can skew the test. A transformer can reverse polarity passively, but most transformers have more distortion than the solid-state audio circuits you’re likely to test. There are also situations where a null test isn’t feasible at all. When someone in a hi-fi forum insisted he heard a change after demagnetizing an LP, I asked him to record an LP before and after demagnetizing. Amazingly, he did that, and he mailed me a CD with both Wave files. Alas, playback on even the best turntables varies constantly. A null test requires two signals whose samples are in exact lockstep. However, I was able to get the files to null anyway, albeit briefly. By placing the faster file a few samples ahead of the slower version in my DAW, when played the sound got softer, passed through near-silence as the samples aligned briefly, then became louder again as the tracks drifted apart. The same happens when recording from analog tape recorders or even sound cards. The clock that sets the sample rate for a sound card is highly accurate and stable, but it still drifts a little. So you can often obtain a total null, if only for a second or two. Null tests also fail if one of the sources has been phase shifted relative to the other. Even if the amount of phase shift is small, and inaudible, it still changes the waveform, preventing complete cancellation.
Disproving Common Beliefs Audio software can also be used to disprove other common audio myths, even without a null test. Some people wrongly believe that mixing two sine waves in a console or DAW creates sum and difference frequencies. Chapter 13 explained that amplitude and frequency modulation are linear processes that create new frequencies, but that’s not the same as simple summing. To demonstrate this, I created a Wave file in Sound Forge having equal
Test Procedures 617 amounts of 50 Hz and 200 Hz, then used the FFT analyzer to display the content. The FFT showed only two blips: one at 50 Hz and another at 200 Hz, with nothing at 150 or 250 Hz. Another simple home test assesses degradation from a seemingly transparent sound card or other device by listening only. Most modern gear is very clean, so auditioning music recorded through a device using a single loop-back may not show an audible difference. The solution is to record repeatedly through the device to accumulate the degradation. When I wanted to assess the degradation from an inexpensive SoundBlaster sound card, I recorded the same piece of music through it 20 times in succession. Each recording became the new playback source for the next, so in the end I had 20 separate files, each progressively worse than the previous one. This test is also shown in my AES Audio Myths video, and the result files after 1, 5, 10, and 20 passes can be downloaded from my personal website www.ethanwiner.com.
Oscilloscopes As you’ve seen, many simple but useful tests can be performed using only basic audio software and a sound card. But for more serious testing, it’s useful to have an oscilloscope. A oscilloscope lets you see what’s really happening in an audio device at frequencies much higher than a sound card can handle, and decent models can be bought new for less than $400 or used for as little as $25. You don’t need a sophisticated oscilloscope for audio testing, and even an inexpensive used model is better than a computer simulation using a sound card. Even though nobody can hear 80 KHz, power amplifiers and other circuits can oscillate or ring at those high frequencies. An oscilloscope lets you see the actual waveforms and gives an insight not attainable any other way. I used a VST oscilloscope plug-in for the Analog Synthesizers video from Chapter 14 to show what happens when sweeping a resonant low-pass filter. It’s not possible to explain oscilloscopes in depth here, but I’ll cover the basics. The main purpose of an oscilloscope is to display signals as they change over time. A voltmeter is fine for measuring steady levels like test tones or for checking flashlight batteries, but it’s impossible to observe a signal’s instantaneous value—or even tell if you have a square or sine wave for that matter—because the meter’s pointer can’t move fast enough. An oscilloscope uses an electron beam that creates a dot of light when the beam strikes the phosphor coating inside a cathode ray tube (CRT). The beam then sweeps repeatedly across the screen quickly enough to track the input waveform. Modern high-end oscilloscopes use digital displays, but budget models still use CRTs. Every oscilloscope has both a vertical and horizontal amplifier, as well as a built-in variable frequency sawtooth oscillator to generate the recurring sweep. A CRT uses a single dot of
618 Chapter 22
Figure 22.6: Oscilloscopes have both vertical and horizontal inputs to move a single dot quickly enough to display a waveform.
light, so the dot must be swept constantly from left to right to create the illusion of a continuous solid line. This is how CRT televisions work, too. The signal you’re watching goes to the vertical amplifier’s input, which shifts the dot up or down, as shown in Figure 22.6. Without the horizontal sweep, there’s only a single dot in the center of the screen, and sending a signal to the vertical input simply moves the dot up and down. By using both the vertical and horizontal inputs, the signal’s voltage can be determined by the amount of vertical deflection and its frequency determined by the horizontal position. The sweep rate is variable, letting you view any number of cycles of the input wave over a wide range of input frequencies. Both the vertical amplitude and horizontal time can be read directly from the calibrated lines, called a graticule, drawn on the face of the CRT as shown in Figure 22.7. Switches set the gain of the vertical amplifier, which is calibrated in volts per division, where each division is one line on the screen. The frequency of the horizontal sweep oscillator is controlled in a similar fashion to vary how long it takes the dot to move one division to the right. When the dot reaches the right edge of the screen, the sawtooth ramp resets, quickly sending the light beam to the left edge to start a new trace. In all but the least expensive models, the dot is turned off during the retrace. Otherwise, a confusing double trace results. One of the most important features of an oscilloscope is triggered sweep, which synchronizes the horizontal sweep to the input waveform. Without this feature the displayed waveform would flicker and wander around, because the input waveform is likely at a different part of its cycle when each new sweep begins. Instead of immediately beginning
Test Procedures 619
Figure 22.7: A graticule is a group of lines that identify voltage level vertically and time span horizontally.
a new sweep as soon as the beam is reset to the left edge of the screen, the trigger circuit delays the sweep until the input signal returns to the same voltage as when the previous sweep began. Triggering usually occurs on the rising edge of the input waveform, though better oscilloscopes have a switch that lets you trigger on either the rising or falling edge. Another standard feature is a continuously variable-level control on the vertical input for making relative, rather than absolute, voltage measurements. If you’re measuring the response of a filter, you’d set the unfiltered wave to exactly fill the screen. Then, it’s easy to see when the signal is attenuated by one-half, one-quarter, or whatever after passing through the filter. In fact, many oscilloscopes have an additional dB scale printed on the graticule to allow reading dB changes directly. Likewise, a calibrated fine-tuner called a vernier is provided for the horizontal sweep speed as well to simplify relative frequency measurements. Every oscilloscope also includes an AC/DC switch, which inserts a capacitor into the signal path to observe only the AC component of a signal, ignoring any DC offset present. This is useful for inspecting small levels of high-frequency noise on a DC power supply’s output, letting you raise the vertical gain without having the much larger DC voltage push the trace off the top or bottom of the screen. Most oscilloscopes let you disable the automatic sweep, and most also let you feed a signal directly into the horizontal amplifier to measure the phase difference between two sources. Figure 22.8 shows that identical in-phase signals applied to both inputs creates a diagonal line. Contrast that to the other patterns you get when different amounts of phase shift are introduced. This setup is similar to the software phase correlation meter in Figure 5.6 from Chapter 5
620 Chapter 22
Figure 22.8: Sending audio signals separately to the vertical and horizontal inputs lets you assess the relative phase difference between them.
showing how to verify a stereo mix for mono compatibility. Indeed, this was an important use of oscilloscopes in recording studios before digital metering plug-ins were available. Another useful feature found on medium- and high-priced oscilloscopes is dual-channel capability. This is essential if you need to view two different signals at once, such as an input and output, or both channels of a stereo device. Dual-channel oscilloscopes offer two different modes: alternate and chop. In alternate mode, the first signal sweeps across the top half of the screen, and then the second signal sweeps across the lower half. The chop mode is created by a single sweep, with the channels switching back and forth very rapidly. Depending on the frequency of the signal and the chopping rate, one mode or the other will provide a stable pair of traces. Most dual-trace oscilloscopes also offer a differential mode. Here, different signals are sent to each vertical input, and a switch selects whether the inputs are added or subtracted as in a null test. Once you get to the most expensive oscilloscope models, you’ll find storage capability, which is a method to freeze the display even after the input has been removed. This is needed for viewing events that occur quickly only once. Early storage oscilloscopes used a charged grid inside the CRT face to retain the beam pattern, though modern units use digital memory to store the data.
Summary This chapter explains several simple but useful tests anyone can perform using only audio software and a sound card. The two basic test types are measuring and listening, and
Test Procedures 621 listening tests can be either sighted or blind. Blind tests are needed when differences are subtle to be sure there really is a difference. For listening tests, music is a fine source, but for measuring, sine and square waves are necessary because they’re precise and repeatable. The basic test setup uses a loop-back to play and record through a sound card or external converter. This is more accurate than using a budget multimeter that can’t measure past 10 KHz. Further, seeing the recorded waves, or an FFT, reveals much more than a single voltage number on a meter. Frequency response is one of the easiest tests to perform and is done by sending sine waves or a sweep through the device being measured. Low-frequency square waves are also useful because they reveal ringing that might be hidden when playing static tones. Analyzing the distortion of an audio device is equally simple, using an FFT display to see the individual distortion components. You can also measure the distortion of microphones and loudspeakers, though the distortion of one of them must be known, and preferably lower, in order to assess the other. We also covered null tests, using two DAW projects I created to disprove the myth that 32-bit plug-ins have a sweet spot signal level above which sound quality is harmed and that EQ cannot be countered exactly. Finally, the basic principles of oscilloscopes are shown, including using the vertical and horizontal inputs to identify phase differences between two audio channels.
Test Procedures Chapter 2 explained how audio devices are measured by professionals using modern test gear. Sadly, many people can’t afford $10,000 or more for a precision analyzer, or even $400 for a basic oscilloscope. But there are many tests you can do yourself without measuring equipment. For example, the Wave files included with Chapter 3 let you assess the audibility of low-level artifacts just by listening through your own speakers or headphones. That chapter also explained how the FFT display available in many audio editor programs shows harmonics and noise at levels too soft to hear. As explained in Chapter 8, to measure the distortion of 32-bit floating point “summing” math used by most DAW software, I applied 30 sequential gain changes to a pure sine wave, then analyzed the added distortion using Sound Forge’s FFT feature. With a very good multimeter you can measure the frequency response of outboard audio gear, though oscilloscope software that uses your sound card for its input is also available. Even better is getting a used “real” oscilloscope, which will cost much less than a new one. Chapter 20 showed how an inexpensive small diaphragm omni condenser mic coupled with room measuring software can measure the frequency response of loudspeakers and microphones. So it’s possible to do your own equipment testing using the tools you already own or can obtain inexpensively. There are two very different types of audio tests: measuring tests and listening tests. For assessing raw fidelity, measuring is the better choice because it’s 100 percent repeatable, and the results are highly accurate, assuming it’s done properly. But there’s nothing wrong with listening tests when done correctly and you account for the limitations of hearing. The gold standard for all subjective comparisons—not just audio gear—is the double-blind test. In a sighted test, the person listening knows what source is playing and can be influenced by expectation. A single-blind test is much better, where someone else switches the A and B playback without winking or otherwise letting on. But with a double-blind test, even the person running the test doesn’t know what’s playing at the moment, which avoids any chance of accidentally giving a clue to the test subject. I’m satisfied using single-blind tests, and I’ve done that with friends many times. In my home studio, the person listening is behind me, unable to see my hands or facial expression. You can even test yourself blind, either using ABX software or by closing your eyes while clicking linked solo buttons, as described in Chapter 3. 607
608 Chapter 22
Figure 22.1: A loop-back uses DAW software to play test signals from a sound card’s output through the device to be measured. The result is then recorded back to the DAW for later analysis.
Whether your intent is to measure with a meter or just listen with your ears, the first thing needed is a signal source. Obviously, music is a fine source if the intent is to identify quality differences between two devices such as sound cards, though you must match both volume levels to within 0.1 dB. For accurate and repeatable measuring, using test signals makes more sense. I create sine wave test tones in Sound Forge, and most other audio editor programs can do this. The “pink_noise.wav” file from Chapter 20 can be used when a noise source is more appropriate. Sound Forge can also create a sine wave that sweeps the full audio range, or just a portion. The basic setup is called a loop-back; the output of a sound card or external converter is sent to the hardware device being tested, then looped back to its own input or the input of another sound card, as shown in Figure 22.1. “A person with one voltmeter knows what the voltage is. A person with two voltmeters never really knows for sure.”
Although you might think that testing audio gear requires a meter or oscilloscope, using a decent-quality sound card and audio software is easier and more accurate than inexpensive meters. Most budget meters don’t even state a frequency response, and when they do, it’s often limited to 10 KHz or even lower. A competent sound card—even an inexpensive consumer model—will be reasonably clean and flat from 20 Hz to 20 KHz. Then you can record the test signal(s) once and analyze the recording in different ways later. However, when using a sound card for testing, I suggest you first assess its quality. Connect the output of the sound card directly back to its own input, then record your test signals to learn the sound card’s own frequency response and how much noise and distortion it adds.
Test Procedures 609
Frequency Response The best signal source for measuring frequency response is a series of sine waves at low, mid, and high frequencies, or a slowly swept sine wave to see even more detail. Depending on what you’re testing, you may not need to measure at many frequencies. Often, 20 Hz, 1 KHz, and 20 KHz are sufficient for devices expected to be flat. If a device rolls off at the frequency extremes and you want to see the trend, you could also measure at 50 Hz and 10 or 15 KHz. When measuring gear that contains a transformer, or an analog tape recorder, using a sweep ensures you won’t miss any irregularities. Either way, I generally create test files with a peak level of 26 dB. Sine waves should sustain each frequency for 5 to 10 seconds, giving you time to read the level meter later. Then load the file to a track in your DAW software and record the output of the device being tested to a second track. Another approach uses a stand-alone signal generator program that plays tones or noise through your sound card, which you route to the device being tested. Then you can record the signal from the gear being tested using any basic audio editor. I use Vincent Burel’s excellent and free LF Generator, which can output various wave shapes at any audio frequency, including sweeps, as well as pink noise. You can also test your hearing with sine waves, but you should first verify how high a frequency your loudspeakers can produce using a decent small diaphragm omni condenser mic and room measuring software as described in Chapter 20. When playing very highfrequency tones through a loudspeaker, move your head slightly while listening to be sure you’re not in a null spot. High frequencies are surprisingly directional due to loudspeaker beaming and also comb filtering that occurs naturally in rooms. Of course, using good headphones avoids these problems as long as you’re certain they can reproduce frequencies higher than you can hear. Another method plays a 96 KHz recording of a tambourine or jangling keys, or other source known to contain frequencies beyond 20 KHz, as verified with an FFT. The tambourine.wav file from Chapter 3 is ideal for this. Sweep the frequency of a low-pass filter plug-in downward starting from above 20 KHz until you can just barely hear a difference, then note the frequency on the plug-in.
Ringing A 20 Hz square wave will tell you more about an audio device in two seconds than all the fancy test gear in the world. Square waves are the most challenging signal for any device to pass accurately, and they reveal not only response errors but also ringing that can indicate circuit instability. Figure 22.2 at the top shows what ringing looks like, using a 20 Hz square wave before and after applying a high-Q boost at 1 KHz. The ringing is evident as each rapid wave transition creates a small sine wave that decays over time. The middle figure shows the original square wave after adding a 6 dB per octave low-pass filter set to
610 Chapter 22
Figure 22.2: A low-frequency square wave is an excellent test signal because it readily reveals ringing as well as frequency response errors. From top to bottom: ringing, high-frequency roll-off, low-frequency roll-off.
filter content above 1 KHz. Here you can see the rounding of the edges at each transition, exactly as happens with a passive resistor-capacitor filter as shown earlier in Figure 8.2. Applying a high-pass filter, at the bottom, causes the flat top and bottom portions of each cycle to droop as the sustained “DC” component of each cycle falls off.
Harmonic Distortion Distortion is not difficult to measure, requiring only a high-quality sine wave source and an FFT analyzer, which is included in many audio editor programs. Note that an FFT display doesn’t tell you a single distortion amount in percent or dB. Rather, you see all the individual components that, if added together, would give the total amount of distortion. But an FFT shows the nature of the distortion, which is more useful for assessing audibility than a single number. Depending on what type of device you’re testing, you might measure at 1 KHz only or over the full audio range. Transformers tend to have higher distortion at low frequencies, so you could test at 50 Hz or even 20 Hz. Absolute signal level is also a factor; some gear has more distortion at higher levels, but some devices are worse at soft levels. It’s important to
Test Procedures 611 start with a pure sine wave having known low distortion, though most software creates reasonably pure tones. Again, you should verify the source purity, as I did for the summing math test in Chapter 8. I examined the original sine wave with an FFT and confirmed that all the residual artifacts were safely below 2100 dB. This equates to 0.001 percent distortion plus noise, which is lower than most outboard gear you’re likely to test. If all of the individual artifacts are below 100 dB, it’s unlikely you’ll ever hear them. And that’s the real point of such testing.
IM Distortion IM distortion is more complicated to measure than simple harmonic distortion, but it’s still possible using the same methods. IM distortion occurs when two or more frequencies play together and create new sum and difference frequencies. For example, if you play music or test tones containing an A note at 440 Hz and also the C# note above at 554 Hz, IM distortion adds new tones at 554 1 440 5 994 Hz and 554 2 440 5 114 Hz. Note that neither 994 Hz nor 114 Hz is a standard musical note pitch. This is one reason IM distortion generally sounds more obnoxious than harmonic distortion whose components are musically related to the source frequency. To test IM distortion in audio gear, you’ll create sine waves at two different frequencies, then mix them together and save the result in a file to play through the device under test (DUT). Standard IMD tests play 19 KHz and 20 KHz at equal volumes, then measure the level of the resulting 1 KHz difference frequency. (The 39 KHz sum is considered irrelevant.) But IMD tests can use other frequencies, which is needed when measuring loudspeakers and microphones. Since two frequencies result from IM distortion—the sum and the difference—it’s not difficult to determine the approximate distortion percentage. However, large amounts of IM distortion also create additional components that are harmonics of the sum and difference frequencies. Again, this is less important than seeing the big picture to verify that whatever distortion you measure is too soft to be objectionable. When measuring the IM distortion of loudspeakers, you should choose frequencies that are both played by the woofer or by the tweeter at the same time. If a speaker crosses over from woofer to tweeter at 2 KHz, using 500 Hz and 3 KHz will not tell you the amount of IM distortion from either driver. To test the woofer, you might use 100 Hz and 180 Hz, which results in 280 Hz and 80 Hz for the sum and difference frequencies, letting you ignore the inevitable harmonic distortion at multiples of 100 Hz and 180 Hz. If you played 100 Hz and 300 Hz, the resulting 200 Hz and 500 Hz IM components will include regular harmonic distortion of the 100 Hz source at those same frequencies. Likewise, 5 KHz and 8 KHz are suitable for testing tweeters, but not 5 KHz and 10 KHz. You should also choose frequencies that are reasonable for the drivers to play at high volumes, where their
612 Chapter 22 distortion is greatest. A sustained loud 19 KHz tone is very demanding of a tweeter, and isn’t representative of normal music. Playing such a high frequency for more than a few seconds at high volume can damage the driver, and maybe your hearing, too. When measuring microphone IM distortion, you’ll again choose frequencies that are reasonable for the speaker and mic to deal with at high volume levels. The best way to measure microphone IMD is with two speakers, with one tone played through each speaker. This avoids contaminating the resulting sum and difference frequencies with the loudspeaker’s own IM distortion, which is typically higher than most microphones at low frequencies. You’ll create separate files for each test frequency, panning them hard left and right for playback. The microphone should be halfway between each speaker and also onaxis to capture the flattest response from both speakers. Assuming your monitoring is set up correctly, with each speaker angled toward your ears, just put the microphone where you listen at the same height as the speakers. Figure 22.3 shows the IM distortion I measured for my audio-technica 4033 and DPA 4090 microphones using this method, by playing 300 Hz and 500 Hz through the JBL 4430 speakers in my home studio. As you can see, the AT 4033 has half the IM distortion of the DPA 4090. Adding the sum and difference signal levels, the AT is about 0.2 percent IMD versus about 0.4 percent for the DPA. You can also see harmonic distortion at various multiples of the 300 Hz and 500 Hz tones. The blip to the left of the 276 dB label is the 600 Hz second harmonic of the 300 Hz tone, and the 1 KHz second harmonic of the 500 Hz source tone is just to the right of the same label at an even higher level. It’s not visible here, but when you hover the mouse cursor over the graph, a pop-up balloon shows the frequency and level at that location. This is how I identified the exact frequency and level of the various peaks. It’s difficult to measure harmonic distortion of speakers or microphones unless the distortion for one of them is known at a given frequency and SPL, or is at least known to be lower than the other device being tested. However, THD and IMD are not unrelated, so excessive IMD implies that THD will also be high, and likewise for low amounts. The 600 Hz and 1 KHz second harmonics in Figure 22.3 could be due to either the microphone or the speakers, but the 200 Hz and 800 Hz components can only be due to the microphone. Since the second harmonic from the speakers is louder than the sum and difference frequencies from the mic, it’s reasonable to conclude that the speaker’s distortion is greater. However, frequencies recorded by a microphone can align with a peak or null in the room’s response and affect the results, possibly by a large amount. I suggest measuring the raw response first to get a baseline for all four frequencies, then do the IMD tests without moving the microphone. Finally, I’ll point out that home loudspeaker and microphone tests as described here will never be as accurate as measurements made with real test equipment in an anechoic
Test Procedures 613
Figure 22.3: Microphone IM distortion was measured by playing 300 Hz through one speaker and 500 Hz through another, while recording the output of each microphone. An FFT display shows the levels of the resulting sum and difference frequencies.
chamber. But you can definitely get useful data. So trust, but verify. If your results don’t agree with what’s expected, question the results until you’re confident they’re correct. Indeed, question everything, including measurements that seem too good to be true.
Null Tests As explained in Chapter 1, a null test works by subtracting two audio signals to see what remains. If the result is total silence, then the signals are by definition identical. And if
614 Chapter 22 there is a residual, its audibility can be assessed based on the level and frequency distribution, or simply by listening. The beauty of a null test is that it reveals all differences between two audio signals, including differences you might not even be looking for. If someone claims playing Wave files from one hard drive sounds different from playing them from another hard drive, a null test will tell you for certain whether or not that’s true. You simply copy a file to both hard drives and see if they null when played in a DAW program. To subtract audio signals, you reverse the polarity of one, then mix it with the other at the same volume. Having both sources match exactly in level and phase is the key to a successful null test, and it can be more difficult than you might imagine. By watching the residual output on a wide-range VU meter that displays down to the noise floor, you’ll tweak the level of one signal to get the best null. Then, if you’re comparing recordings, you can slide one in time relative to the other to avoid time and phase differences. If the result is total silence, or at least well below 280 dB, you can be confident that both sources are audibly identical. Most null tests are done in a DAW program, placing the Before and After versions on separate tracks, with both files aligned to start at the same time and with equal volumes. If the tracks are off in time by even one sample, or their levels differ by even 0.1 dB, identical files that would have nulled to silence will yield some amount of residual. I used null tests in my AES Audio Myths video on YouTube to disprove two common myths: One myth is that audio plug-ins have a “sweet spot” signal level, which if exceeded harms sound quality. The other is that digital EQ cannot be countered exactly. This is a great application for null tests because both tracks contain the same source file, avoiding the need for time alignment and level matching. Figure 22.4 shows the setup that sends a mix through an EQ plug-in after raising the volume by 18 dB. The mix was normalized to peak at 21 dB, so the audio through the plug-in reaches 17 dB above digital zero. Track 1 contains the mix file, and Track 2 plays the same file with boost applied by the Track Trim to be before the EQ. The Sonalksis FreeG freeware volume plug-in restores the level after the EQ. This plug-in is great for null tests because it lets you adjust the volume in 0.01 dB steps, though only a whole number dB amount was needed here. Track 2 also has its polarity switch engaged, needed to create a null. Otherwise, when both tracks play, the output would be 6 dB louder than one track, rather than silence. Figure 22.5 shows the setup for the second null test to confirm that digital EQ can be countered with equal but opposite settings. Here again, the same music file is placed on both tracks, aligned exactly to single sample timing. In this test, Track 1 contains two instances of the same EQ plug-ins, with three bands set to opposite amounts of boost and cut, with the same bandwidth (Q). Track 2 uses no plug-ins, but its polarity is reversed to cancel Track 1.
Test Procedures 615
Figure 22.4: This null test setup confirms that modern 32-bit plug-ins are immune to overload, even when sent signals that greatly exceed the digital zero clipping point.
Figure 22.5: This null test proves that EQ can be reversed exactly using a second plug-in having equal but opposite settings.
616 Chapter 22 The first test proves that 32-bit floating point plug-ins are not overloaded by signals that exceed digital zero. As long as the volume is reduced somewhere later in the chain, before the mix is rendered to a file or sent out the sound card for monitoring, clipping distortion will not occur. The second test proves that EQ can be reversed exactly as long as both plug-ins are set precisely opposite. When playing either of these DAW projects, the result is total silence down to the 290 dB floor of the output bus level meter. Rendering the mix to a Wave file and examining that in an audio editor further confirm that the file contains only silence. Null tests can also be done in real time using power amplifiers and other audio gear by subtracting the device’s input and output signals. However, this is more complicated to set up. When testing something that amplifies the signal, such as a power amplifier, you need a way to make the input and output levels exactly equal. This can be done with variable resistors that are passive and thus won’t add distortion, but you also need a way to reverse the polarity of one signal. That requires an additional circuit whose own distortion or nonflat frequency response can skew the test. A transformer can reverse polarity passively, but most transformers have more distortion than the solid-state audio circuits you’re likely to test. There are also situations where a null test isn’t feasible at all. When someone in a hi-fi forum insisted he heard a change after demagnetizing an LP, I asked him to record an LP before and after demagnetizing. Amazingly, he did that, and he mailed me a CD with both Wave files. Alas, playback on even the best turntables varies constantly. A null test requires two signals whose samples are in exact lockstep. However, I was able to get the files to null anyway, albeit briefly. By placing the faster file a few samples ahead of the slower version in my DAW, when played the sound got softer, passed through near-silence as the samples aligned briefly, then became louder again as the tracks drifted apart. The same happens when recording from analog tape recorders or even sound cards. The clock that sets the sample rate for a sound card is highly accurate and stable, but it still drifts a little. So you can often obtain a total null, if only for a second or two. Null tests also fail if one of the sources has been phase shifted relative to the other. Even if the amount of phase shift is small, and inaudible, it still changes the waveform, preventing complete cancellation.
Disproving Common Beliefs Audio software can also be used to disprove other common audio myths, even without a null test. Some people wrongly believe that mixing two sine waves in a console or DAW creates sum and difference frequencies. Chapter 13 explained that amplitude and frequency modulation are linear processes that create new frequencies, but that’s not the same as simple summing. To demonstrate this, I created a Wave file in Sound Forge having equal
Test Procedures 617 amounts of 50 Hz and 200 Hz, then used the FFT analyzer to display the content. The FFT showed only two blips: one at 50 Hz and another at 200 Hz, with nothing at 150 or 250 Hz. Another simple home test assesses degradation from a seemingly transparent sound card or other device by listening only. Most modern gear is very clean, so auditioning music recorded through a device using a single loop-back may not show an audible difference. The solution is to record repeatedly through the device to accumulate the degradation. When I wanted to assess the degradation from an inexpensive SoundBlaster sound card, I recorded the same piece of music through it 20 times in succession. Each recording became the new playback source for the next, so in the end I had 20 separate files, each progressively worse than the previous one. This test is also shown in my AES Audio Myths video, and the result files after 1, 5, 10, and 20 passes can be downloaded from my personal website www.ethanwiner.com.
Oscilloscopes As you’ve seen, many simple but useful tests can be performed using only basic audio software and a sound card. But for more serious testing, it’s useful to have an oscilloscope. A oscilloscope lets you see what’s really happening in an audio device at frequencies much higher than a sound card can handle, and decent models can be bought new for less than $400 or used for as little as $25. You don’t need a sophisticated oscilloscope for audio testing, and even an inexpensive used model is better than a computer simulation using a sound card. Even though nobody can hear 80 KHz, power amplifiers and other circuits can oscillate or ring at those high frequencies. An oscilloscope lets you see the actual waveforms and gives an insight not attainable any other way. I used a VST oscilloscope plug-in for the Analog Synthesizers video from Chapter 14 to show what happens when sweeping a resonant low-pass filter. It’s not possible to explain oscilloscopes in depth here, but I’ll cover the basics. The main purpose of an oscilloscope is to display signals as they change over time. A voltmeter is fine for measuring steady levels like test tones or for checking flashlight batteries, but it’s impossible to observe a signal’s instantaneous value—or even tell if you have a square or sine wave for that matter—because the meter’s pointer can’t move fast enough. An oscilloscope uses an electron beam that creates a dot of light when the beam strikes the phosphor coating inside a cathode ray tube (CRT). The beam then sweeps repeatedly across the screen quickly enough to track the input waveform. Modern high-end oscilloscopes use digital displays, but budget models still use CRTs. Every oscilloscope has both a vertical and horizontal amplifier, as well as a built-in variable frequency sawtooth oscillator to generate the recurring sweep. A CRT uses a single dot of
618 Chapter 22
Figure 22.6: Oscilloscopes have both vertical and horizontal inputs to move a single dot quickly enough to display a waveform.
light, so the dot must be swept constantly from left to right to create the illusion of a continuous solid line. This is how CRT televisions work, too. The signal you’re watching goes to the vertical amplifier’s input, which shifts the dot up or down, as shown in Figure 22.6. Without the horizontal sweep, there’s only a single dot in the center of the screen, and sending a signal to the vertical input simply moves the dot up and down. By using both the vertical and horizontal inputs, the signal’s voltage can be determined by the amount of vertical deflection and its frequency determined by the horizontal position. The sweep rate is variable, letting you view any number of cycles of the input wave over a wide range of input frequencies. Both the vertical amplitude and horizontal time can be read directly from the calibrated lines, called a graticule, drawn on the face of the CRT as shown in Figure 22.7. Switches set the gain of the vertical amplifier, which is calibrated in volts per division, where each division is one line on the screen. The frequency of the horizontal sweep oscillator is controlled in a similar fashion to vary how long it takes the dot to move one division to the right. When the dot reaches the right edge of the screen, the sawtooth ramp resets, quickly sending the light beam to the left edge to start a new trace. In all but the least expensive models, the dot is turned off during the retrace. Otherwise, a confusing double trace results. One of the most important features of an oscilloscope is triggered sweep, which synchronizes the horizontal sweep to the input waveform. Without this feature the displayed waveform would flicker and wander around, because the input waveform is likely at a different part of its cycle when each new sweep begins. Instead of immediately beginning
Test Procedures 619
Figure 22.7: A graticule is a group of lines that identify voltage level vertically and time span horizontally.
a new sweep as soon as the beam is reset to the left edge of the screen, the trigger circuit delays the sweep until the input signal returns to the same voltage as when the previous sweep began. Triggering usually occurs on the rising edge of the input waveform, though better oscilloscopes have a switch that lets you trigger on either the rising or falling edge. Another standard feature is a continuously variable-level control on the vertical input for making relative, rather than absolute, voltage measurements. If you’re measuring the response of a filter, you’d set the unfiltered wave to exactly fill the screen. Then, it’s easy to see when the signal is attenuated by one-half, one-quarter, or whatever after passing through the filter. In fact, many oscilloscopes have an additional dB scale printed on the graticule to allow reading dB changes directly. Likewise, a calibrated fine-tuner called a vernier is provided for the horizontal sweep speed as well to simplify relative frequency measurements. Every oscilloscope also includes an AC/DC switch, which inserts a capacitor into the signal path to observe only the AC component of a signal, ignoring any DC offset present. This is useful for inspecting small levels of high-frequency noise on a DC power supply’s output, letting you raise the vertical gain without having the much larger DC voltage push the trace off the top or bottom of the screen. Most oscilloscopes let you disable the automatic sweep, and most also let you feed a signal directly into the horizontal amplifier to measure the phase difference between two sources. Figure 22.8 shows that identical in-phase signals applied to both inputs creates a diagonal line. Contrast that to the other patterns you get when different amounts of phase shift are introduced. This setup is similar to the software phase correlation meter in Figure 5.6 from Chapter 5
620 Chapter 22
Figure 22.8: Sending audio signals separately to the vertical and horizontal inputs lets you assess the relative phase difference between them.
showing how to verify a stereo mix for mono compatibility. Indeed, this was an important use of oscilloscopes in recording studios before digital metering plug-ins were available. Another useful feature found on medium- and high-priced oscilloscopes is dual-channel capability. This is essential if you need to view two different signals at once, such as an input and output, or both channels of a stereo device. Dual-channel oscilloscopes offer two different modes: alternate and chop. In alternate mode, the first signal sweeps across the top half of the screen, and then the second signal sweeps across the lower half. The chop mode is created by a single sweep, with the channels switching back and forth very rapidly. Depending on the frequency of the signal and the chopping rate, one mode or the other will provide a stable pair of traces. Most dual-trace oscilloscopes also offer a differential mode. Here, different signals are sent to each vertical input, and a switch selects whether the inputs are added or subtracted as in a null test. Once you get to the most expensive oscilloscope models, you’ll find storage capability, which is a method to freeze the display even after the input has been removed. This is needed for viewing events that occur quickly only once. Early storage oscilloscopes used a charged grid inside the CRT face to retain the beam pattern, though modern units use digital memory to store the data.
Summary This chapter explains several simple but useful tests anyone can perform using only audio software and a sound card. The two basic test types are measuring and listening, and
Test Procedures 621 listening tests can be either sighted or blind. Blind tests are needed when differences are subtle to be sure there really is a difference. For listening tests, music is a fine source, but for measuring, sine and square waves are necessary because they’re precise and repeatable. The basic test setup uses a loop-back to play and record through a sound card or external converter. This is more accurate than using a budget multimeter that can’t measure past 10 KHz. Further, seeing the recorded waves, or an FFT, reveals much more than a single voltage number on a meter. Frequency response is one of the easiest tests to perform and is done by sending sine waves or a sweep through the device being measured. Low-frequency square waves are also useful because they reveal ringing that might be hidden when playing static tones. Analyzing the distortion of an audio device is equally simple, using an FFT display to see the individual distortion components. You can also measure the distortion of microphones and loudspeakers, though the distortion of one of them must be known, and preferably lower, in order to assess the other. We also covered null tests, using two DAW projects I created to disprove the myth that 32-bit plug-ins have a sweet spot signal level above which sound quality is harmed and that EQ cannot be countered exactly. Finally, the basic principles of oscilloscopes are shown, including using the vertical and horizontal inputs to identify phase differences between two audio channels.