Adjusting or modifying audio signals is nothing new. Analog signal processors have been around recording studios and live performances for decades. Everything from equalizers to crossovers and compressors were conceived back when vacuum tubes were popular. As technology advanced, the size, cost and complexity of signal processors decreased. Now, many car audio source units contain more processing power than early recording studios. This article looks at digital signal processors (DSPs), what they do and why you need them.

A Hostile Environment

If we were to take a full-range home speaker into an open field and measure the frequency response, we’d see a fairly flat and smooth response curve. If you take that same speaker into a small room and measure the response again, you will see peaks and dips at various frequencies. This change in frequency response is not caused by the speaker, but by the room itself. Reflections cause nodes and anti-nodes (peaks and valleys) that dramatically affect the perceived frequency response of the speaker system. To maximize our enjoyment of that speaker, we need to apply signal correction to the speaker so what we hear is similar to what we would have experienced in that field.

In a car, we are very rarely able to sit directly in the middle of the left and right speakers. The driver is usually twice as far from the right speaker as from the left. We hear the output of the left speaker first and it seems as if that speaker appears to be playing louder – because it is closer. Keep this in mind as we discuss digital signal processors (DSPs).

Speaker Limitations

No single speaker can reproduce the entire audio spectrum from 20 Hz to 20 kHz with accuracy, detail and even dispersion of sound. Even if there were one that could do this, the distortion levels in the midrange and high-frequency sounds would still be high because of the excursion requirements of the speaker at low frequencies. Because of this, we make use of several different speakers to cover the audio band. Woofers or subwoofers cover the bass, and typically play up to 80 or 100 hertz. Midrange drivers cover the range from 100 Hz to around 4,000 Hz. Finally, we use tweeters to cover the remainder of the frequencies above 4,000 Hz. While these are approximations, they are common crossover points for these speakers.

A crossover is a device that limits the passing of audio signals. There are two common types used in car audio: high-pass and low-pass. Their name describes their function. A high-pass crossover allows frequencies higher than the crossover point to pass through, and a low-pass allows frequencies below the crossover point to pass. A high-pass crossover would be used to keep the deep bass out of a small door or dash speaker, while a low-pass crossover is used to keep midrange and high-frequency information out of a subwoofer. We can combine both kinds of crossovers to produce what is known as a bandpass crossover – we limited the low- and high-frequency information. We would use this on a midrange speaker when combining it with a woofer and a tweeter. (We will discuss crossovers in detail in another article.)

In car audio, we use both active and passive crossovers. Passive crossovers are a combination of capacitors, resistors and inductors that we connect to the speaker wires between the amp and the speaker. The behavior of the components, and how they are configured, limits what frequencies are allowed to pass through to the speaker.

An active crossover is an electronic device that affects the frequency response of the signal before the amplifier. The benefit of active crossovers is that it is easy to adjust them to different frequencies. Most, if not all, crossover components have to be replaced to adjust the crossover frequency of a passive network.

This information gives us a basic understanding of why we need signal processing. For decades, the mobile electronics industry survived and thrived using analog processing. Companies like AudioControl, Phoenix Gold, Rockford Fosgate and Zapco made equalizers and crossovers, and enthusiasts flocked to them like moths to a flame.

As computing power advanced, we saw products like the Rockford Symmetry appear. The Symmetry was an electronically controlled analog processor – a fantastic creation that allowed users to make many adjustments from a single computerized control panel.

The next evolution in signal processing was to do everything in the digital domain, instead of analog. How does that work?

Building Blocks

A DSP is a powerful audio signal processor with hardware and software that is optimized to perform high-speed processing in real time. Some of the less-expensive processors include the analog-to-digital and digital-to-analog converters within the chip itself. On the higher-end units, the analog converters are external components. Better D/A converters offer increased resolution and improved signal-to-noise ratio performance. Once the audio signal is in the digital domain, one DSP doesn’t vary much from another. Algorithms are written in a similar fashion for filtering, equalization and time alignment.

Why would we want a DSP and not an analog processor? In a DSP, there are no associated concerns about component tolerances or temperature variations that will affect the response of the processing. With the right interface, users can access different system presets quickly and store an unlimited number of configurations on their computers. Most DSP units don’t include any analog adjustments, like potentiometers or switches, which can get dirty or wear out over time. Vibrations that could lead to component failure in an analog system rarely affect DSPs.

Features of Digital Signal Processors

Once an analog signal is converted to digital, the available signal processing is limited only by the software that is written for the chosen unit. The limit on the features of the software is typically determined by the available memory of the processor itself. It takes space to store the program, and additional space to store the converted analog information as the processor works with the information. When you see one processor with more features than another, the difference is usually a memory limitation.

Inputs And Signal Summing

Most DSP units on the market can combine and adjust the level of audio signals on the input to the DSP. If you have a radio with front, rear and subwoofer outputs, you may want to maintain all of these channels discretely as you process the audio signal.

What about when you are trying to integrate with a factory amplifier? Perhaps you have a front door midrange and tweeter output from an amplifier that you need to use for your new front speakers. Most digital signal processors will allow you to combine signals from multiple inputs to facilitate applications like this.

Since different sources have different peak voltage levels, the inputs to your DSP have adjustable sensitivities. Just like the gain control on an amplifier, we want to set the input gains on our DSP to maximize the signal-to-noise ratio of the processor.

Crossovers And Filtering

As we mentioned, different size speakers are designed to focus their performance within different audio ranges. A 3-inch midrange will not play the same frequency range as a 1-inch tweeter or a 6.5-inch woofer. We use the crossovers in the DSP to divide up the frequencies sent to each output and speaker.

A benefit of doing all the crossover processing in the digital domain is that many digital signal processors offer different crossover filter alignments and roll-off slopes. The alignment describes the shape of the roll-off around the -3 dB point. This shape also affects how signals sum back together acoustically. Options are Butterworth, Linkwitz-Riley, Chebychev, Bessel and more. It’s not that one is better than another, but that each is distinct and different. We could write an entire article about crossover alignments.

The crossover slope describes how fast the audio stops playing as a signal moves away from the crossover point. Because it’s all digital, most digital signal processors offer slopes from -6 dB to -48 dB per octave, in steps of 6 dB or 12 dB, depending on the chosen alignment. In most cases with DSPs, 24 dB/Octave Linkwitz-Riley filtering works quite well, but there are dozens of different tuning approaches, so use what works well for you.

Time Alignment And Signal Delay

One of the coolest features of a digital signal processor is its ability to store the audio signal for a variable amount of time before sending it to the speaker. This storage ability allows a properly trained installer to delay the signal going to the speakers closest to the listener so the sound from created by them arrives at the listening position at the same time as the rest of the speakers. For four-way systems (subwoofer, midbass, midrange and tweeter), this setup and fine-tuning can take a little time.

Equalization

The ability to fine-tune the frequency response of each speaker in an audio system is a huge key to making that system sound amazing. We have to measure the response of each speaker at the listening position, then adjust the equalizer so each speaker produces a smooth response. There are many ways to achieve this.

Graphic equalizers typically offer 31 bands of equalization per channel and are spaced 1/3 of an octave apart. This spacing usually provides enough frequency resolution to resolve response issues. Graphic equalizers are easy to understand: You pick the desired frequency band, then boost or cut the signal by the amount of your choice.

Parametric equalizers are much more powerful, but can be a little more difficult to configure. In a parametric equalizer, the user can choose the frequency, bandwidth and amount of signal boost or reduction. Understanding the selection of frequency is simple, but understanding filter Q factor is more difficult. When it comes to Q, the basic concept is that a higher number means that the band adjustment affects a narrower range of frequencies. A low number, like 0.7 or 1, covers a wider range of frequencies. Setting up a parametric equalizer accurately takes some practice. That said, some software applications will provide setting information automatically after you measure the frequency response of the speaker or system.

Output Level And Remote Controls

Having the ability to tune the output level of each speaker finely is critical to the performance of an audio system. To achieve an accurate and balanced soundstage, the amplitude (level) of each speaker in the system must be adjusted very accurately. Output level control is also quite important to matching the efficiency of the different speakers.

Many DSP units have the option of a remote control. These controls can be used to adjust the overall system volume and adjust the subwoofer output level, and can typically load presets for the processor. More advanced controllers give you access to some of the system tuning features, allowing you to make adjustments without the need for a laptop computer. Displays on these remote controls vary from simple single-color dot-matrix LCD panels to full-color OEL displays that are easy to see in bright sunlight.

Digital Signal Processor Tuning – Art Or A Process?

There are many schools of thought about how to configure a DSP. Whether you do it using instrumented measurements or different acoustic techniques, we want to achieve proper protection for the speakers, smooth frequency response from both channels of the audio system and aligned arrival times from each speaker.

Many car audio manufacturers train their dealers in different methods of achieving a great “tune” on their customer vehicles. If you are looking to improve the sound of your mobile entertainment system and already have great speakers and amplifiers, visit your local car audio professional. They would be happy to demonstrate the benefits of DSPs, and provide you with the information you need to make an educated decision about buying one.

This article is written and produced by the team at www.BestCarAudio.com. Reproduction or use of any kind is prohibited without the express written permission of 1sixty8 media.

When people are looking at purchasing a car audio amplifier, the specification they check most often is how much power it can produce. Power is rated in watts – a universal unit of measurement of power. In this article, we explain what a watt is, and how it is measured – both the correct and incorrect way.

Dictionary Time!

Let’s get the formal definition of a watt out of the way first. A watt is an SI (Systéme International) unit of the measurement of power. The power does not have to be electrical. In fact, the unit watt was named after James Watt and created to quantify the work a steam engine could do. In that kinetic application, a watt was the work done when the velocity of an object was moving steadily at 1 meter per second with a force of 1 newton opposing it. When referring to an electrical motor, 1 horsepower equals 746 watts.

As much fun as talking about horsepower is, we are car audio enthusiasts, so let’s get back on track with an explanation of the electrical watt.

In electrical terms, a watt is a transfer of 1 joule of energy over a period of 1 second. The next logical question is what is a joule? A joule is yet another SI unit of measurement, and it defines the amount of work required to move a charge of 1 coulomb through an electrical potential of 1 volt. Yes, the question now moves to the coulomb – what in the world is that? A coulomb is a unit of electrical charge – and is equal to -6.242 x 10^18 electrons.

Lost yet? Don’t fret; we are just appeasing the math and measurement nerds among us. Let’s break this down to what matters.

When we want to use electricity to do work, we have to flow electrons through a device like a filament, motor or voice coil. The result will be, in the case of a speaker, that the magnetic field created by the flow of electrons will cause the voice coil to be attracted to or repelled from the fixed magnet in our speaker. When we flow more electrons, more work is done, and the speaker moves farther toward or away from the magnet.

Power Math

Here is where we start to talk about power equations. There are three common methods of calculating the power in a circuit – but we need to know the values of other variables such as voltage, resistance or amperage. Any two of these variables can be used to calculate the power done in a circuit. Here are the equations:
If we have a circuit with a resistance of 4 ohms and we apply a voltage to it with a potential of 10 volts, then we have 25 watts of power. Increasing that voltage to 20 volts means the power available is now 100 watts. We can substitute and rearrange the variables in the equations above to figure out any other variable – it’s simple algebra.

Measuring Power

When a technician has an amplifier on a test bench and wants to measure power, the technician typically connects the amp to a bank of high-power load resistors, then measures the output of the amplifier when the signal has reached a distortion level of 1%. The measurement taken is voltage. Most often, we assume the load is not variable. Let’s say we measure 44 Volts RMS out of an amplifier and we have the amp connected to a 2 ohm load. That works out to 968 watts. It’s very simple and very repeatable – but it doesn’t work in the real world. Let’s look at why.

Resistance versus Reactance

This is going to get a bit technical. Audio signals are alternating current (AC) signals. AC signals are required to make the speaker cone move back and forth from its rest position, but they make power measurement much more complicated. The way conductors and loads react to AC signals is different from direct current (DC) signals.

Because AC signals change direction, the polarity of the magnetic fields they create also changes direction. Trying to change the polarity of magnetic fields wreaks havoc with the behavior of current flow. Once current gets flowing and sets up a magnetic field, it doesn’t like to stop. Imagine a DC voltage – all the electrons are moving in the same direction all the time. They are happy and have no complaints. When it comes to AC signals, though, that flow of electrons has to change directions. With a 20 k Hz signal, the change of directions happens 20,000 times a second. Electrons are lazy – they like to keep doing what they were doing. Because of this, they oppose a change of direction.

An inductor is truly nothing more than a coil of wire. We see inductors in passive crossover networks and the filter stages of Class D amplifiers. When electrons are flowing through an inductor, they set up a strong magnetic field. When you take away the voltage source, the electrons try to keep flowing. In fact, if you have seen a relay with a diode connected to it, that diode is there to give that flow of electrons somewhere to go, other than back into the circuit that was controlling the function of the relay.

When we apply an AC signal to an inductor, the higher the frequency, the harder it is to change the direction of the flow of electrons. The resistance to the flow of alternating current is called inductive reactance. Think of it as resistance, but only applicable to AC signals. Inductors oppose a change in current flow. If we disconnect our alternating current source and measure the DC resistance of an inductor with a multimeter, the number we see on the screen is the resistance. To measure the reactance of an inductor, we need a device that can apply an AC signal and measure the effective voltage drop across the inductor.

The formula to calculate inductive reactance is Xl = 2 x pi x F x L, where F is the frequency of the applied AC signal, L is the inductance value of the inductor measured in henries and Xl is the inductive reactance in ohms. You can see that inductance increases with frequency, as we mentioned earlier.

The voice coil of a speaker is and acts as an inductor.

Current and Voltage

We have more bad news for you. Because an inductor opposes the change in current flow, a timing error arises. Timing of what, you ask? The relative time between the AC voltage across the inductor and the AC value of the current flowing in the inductor. In a perfect inductor (one with no DC resistance), the current through the inductor lags the voltage across the inductor by 90 degrees or ¼ of the frequency of the signal being passed through.

Let that sink in for a second, then think back to our equations for power. Power is voltage times current. But what if the current peak isn’t happening at the same time as the voltage peak? We can’t simply multiply the two numbers together to get the power in the circuit. Worse, the amount of time that the current lags voltage depends on the DC resistance of the inductor and the inductive reactance – for most car audio speakers, the DC resistance is usually somewhere between 2 and 8 ohms. The inductance is in between 0.04 mH for a high-quality tweeter to more than 5 mH for a big subwoofer.

There’s one more challenge: The inductance changes depending on the drive level of the speaker and the position of the speaker cone.

We’re sure you agree – It’s all very complicated, but don’t give up just yet.

How do we measure the real power in an AC circuit? There are a couple of ways. We can measure instantaneous current and voltage at a very high sampling rate and multiply them together. The sampling rate would have to be 20 or 30 times the frequency we measure to be reasonably accurate. We can also use conventional meters to measure the amount of current and voltage in the circuit, then use a Phase Angle Meter to find the relative relationship between the two. Pretty much none of us have a standalone phase angle meter in our toolboxes. What we can’t do is just multiply voltage and current times each other.

Those SPL Guys And Watts

If you are reading this, then you likely roam the Internet with some frequency. You have undoubtedly seen SPL enthusiasts attempt to measure the power produced by their amplifiers by “clamping”’ it. They connect a current clamp to one of the speaker wires coming out of the amp and put a voltmeter across the terminals of the amplifier.

This creates three problems:

1. They should connect the voltmeter to the speaker terminals. Because of the high current flow, the resistance in speaker wire can waste a measurable amount of power.
2. With a voltmeter and current clamp, we don’t know the phase relationship between the current flowing through the voice coils and the voltage across the voice coil.
3. They typically perform these tests at extremely high power levels. The massive amounts of power heat up the voice coils quickly. This heat also increases their resistance quickly. This increase in resistance will cause the current flowing through the speaker to decrease. If the connected current clamp is in “peak hold” mode, it will store a peak reading of the initial current flowing through the voice coil. The reduction in current flow eases the load on the amplifier power supply and allows it to produce more voltage. As current decreases, the voltage out of the amplifier may increase, giving a false reading to the voltmeter in peak hold mode. This heating and resistance increase can happen in a matter of seconds.

If you thought our definition of the watt was complicated, then explaining how to calculate power in a reactive load would push you over the edge, so we won’t explain it all. That’s a topic saved for college or university courses on AC power. What we will do is provide a solution for making complicated power measurements.

The reality is when it comes to measuring power out of an amplifier while connected to a speaker, getting accurate results is very difficult. A few companies produce car audio power meters. The most popular unit is the D’Amore Engineering AMM-1. The AMM-1 is a handheld meter that simultaneously measures current and voltage, and calculates the phase angle between them to provide an accurate power measurement. The AMM-1 will show you how much real-world power your amplifier is making. (Please don’t cry if it’s less than you thought.)

The AMM-1 can also show volt-amps. Volt-amps are calculated by multiplying current times the voltage. You can also see the phase angle of the load on yet another screen. If you are serious about measuring power when an amplifier is driving a reactive load like a speaker, then this is the tool you need.

What You Need to Know

When you are shopping for an amplifier, the numbers you usually see quoted are measured into resistive loads. Most amplifiers have no problem with driving reactive loads, so you can trust the published numbers, as long as the distortion specification is clearly defined.

The CEA-2006A (now called CTA-2006A) specification for power measurement defines the maximum signal distortion during measurement as being 1%, and no more than 14.4 volts can be supplying the amp. Comparing power specs using this standard has leveled the playing field in the car audio industry.

We will look at some other very important amplifier specifications in another article. These other specifications may, in fact, be more important to choosing the right amp for your system than how much power the amp makes. Until then, drop into your local car audio specialist retailer to find out about the latest amplifiers available for your system. There are some amazing new amps on the market with a lot of cool features.

Happy listening!

This article is written and produced by the team at www.BestCarAudio.com. Reproduction or use of any kind is prohibited without the express written permission of 1sixty8 media.

If you have purchased a set of premium car audio speakers from a respected mobile electronics retailer in the past few years, then you should be familiar with the concept of sound deadening. If you aren’t familiar with this, or want to know more, then read on! We think you will find sound deadening is an often-overlooked upgrade that has more benefits than most people are aware of.

What Is Sound Deadening?

Automobile manufacturers apply small sheets of dense asphalt or butyl-based material to the floor, firewall or door panels of their vehicles. This damping material adds mass to the panel, making it more difficult for sound and vibration to move the panel and transfer sound into the interior of the vehicle. Automakers walk a fine line between adding weight to a vehicle to reduce noise versus losing fuel economy and handling characteristics due to this added mass. For this reason, most don’t go overboard with sound deadening. They are missing out on a great opportunity.

In spite of what they say in their marketing materials, manufacturers don’t really put that much emphasis on their audio systems. Even when vehicles include multichannel systems with well-recognised namebrands like Bose, Lexicon or JBL, little effort is put into maximizing the performance of the speakers. Proper application of sound deadening can have a dramatic effect on the performance of an audio system.

Aftermarket Deadening Materials

One of the first companies to actively promote sound deadening was Dynamat. Dozens have since followed suit with different approaches to controlling noise inside the vehicle. All of them work on the same principle of absorbing sound energy in one fashion or another and preventing it from being transferred to the interior of the vehicle. Sound deadening has two main benefits when it comes to car audio – exterior noise blocking and audio system performance improvement by preventing backwave cancellation.

When you look at the inside metal skin of a car or truck door, you can see that there are openings to allow access to power window motors, door handles and other components in the door cavity. These openings are typically covered with a thin sheet of plastic. The purpose of the plastic is to keep water away from the interior door panel. That’s important, of course, but these openings work against your efforts to get good sound from your new speakers. There is just as much sound energy being produced from the rear of the speaker as there is from the front. If this rearward-facing sound is allowed to mix with the sound coming from the front, they cancel each other. The result is poor bass and midbass response. Sealing up these openings with a layer of sound deadening means the energy being produced by the rear of the speaker cannot mix with the frontal energy.

Just how dramatic can this cancellation affect be? We have seen instrumented measurements of a factory 6×9” speaker where the difference between having sound deadening or not produced an increase in output of up to 8 dB at several frequencies between 100 and 500 Hz. If you think about how much additional amplifier power it would take to produce the same increase in output, that’s more than six times are much. To be clearer, if you put 10 watts of power into the speaker and measured the response, you would need 63 watts of power into the same speaker to get the same output without the sound deadening. As you can see, that’s a significant difference, and the benefit is not just in efficiency, but in improved low frequency output. The speaker doesn’t have to work as hard, and that alone will improve the overall sound of your system.

It is well worth noting that an upgrade in speaker quality will not produce the same improvement in performance. With a properly sealed and damped door, an inexpensive speaker can easily outperform speakers costing five to 10 times as much money. Sound deadening is critical to the performance of an audio system.

Signal To Noise

The second benefit of sound deadening is in keeping the interior of the vehicle quiet. When you make the interior quieter, the benefit is two-fold. Driving is more comfortable, since you hear less road, wind and tire noise. This reduction in noise also makes it easier to hear your audio system. You don’t have to turn it up quite as loud to drown out the remaining noise. You can hear the quiet parts of your music more easily. Your Bluetooth hands-free system will also sound better. In the same way that controlling backwave cancellation reduces the need for a speaker to work hard, having a quieter interior does the same.

Kinds Of Deadening

There are many different kinds of sound deadening. The most popular are butyl sheets bonded to a thin aluminum layer. The combination works well to span large openings, but is thin and flexible enough to adhere to complex shapes. Other materials are made of vinyl and asphalt-based.

There are three key considerations when looking at different sound deadening products: How flexible is it? How thick is it? How well does it stay adhered once installed? On the engineering and development side, testing the damping characteristics at different temperatures can show quite varied results. Some materials don’t work as well in high or low temperatures. We have seen many people attempt to use materials not specifically designed for automotive applications. When the material melts and ends up as a gooey, black mess at the bottom of your door or leaks onto your carpet, the cost to repair the damage can be significant.

There are also several products on the market that add a layer of foam to the top of the aluminum layer. This foam is great when used between the inside door skin and the metal door because it eliminates buzzes and rattles.

See Your Specialist Car Audio Retailer To Learn More

The next time you are driving by a specialist car audio retailer, drop in and ask about sound deadening. Many people have chosen to apply sound deadening to otherwise stock vehicles. We guarantee the difference in performance from the audio system, combined with the increased comfort while driving, will be well worth the investment.

This article is written and produced by the team at www.BestCarAudio.com. Reproduction or use of any kind is prohibited without the express written permission of 1sixty8 media.

If you were able to grasp the concepts outlined in the first article about audio distortion, then this one will be a piece of cake. If not, head back and have another read. It can be a bit complicated the first time around.

Undistorted Audio Analysis

When looking at the specifications for an audio component like an amplifier or processor, you should see a specification called THD+N. THD+N stands for Total Harmonic Distortion plus Noise. Based on this description, it is reasonable to think that distortion changes of the shape of the waveform that is being passed through the device.

The two graphs below show a relatively pure 1kHz tone in the frequency and time domains:

A Look At Harmonic Distortion

If we record a pure 1 kHz sine wave as an audio track and look at it from the frequency domain, we should see a single spike at the fundamental frequency of 1 kHz. What happens when a process distorts this signal? Does it become 1.2 or 1.4 kHz? No. Conventional distortions won’t eliminate or move the fundamental frequency. But, it will add additional frequencies. We may have a little bit of 2 kHz or 3 kHz, a tiny but of 5 kHz and a smidge of 7 kHz. The more harmonics there are, the more “harmonic distortion” there is.

You can see that there are some small changes to the waveform after being played back and recorded through some relatively low-quality equipment. Both low- and high-frequency oscillations are added to the fundamental 1 kHz tone.

Signal Clipping

In our last article, we mentioned that the frequency content of a square wave included infinite odd-ordered harmonics. Why is it important to understand the frequency content of a square wave when we talk about audio? The answer lies in an understanding of signal clipping.

When we reach the AC voltage limit of our audio equipment, bad things happen. The waveform may attempt to increase, but we get a flat spot on the top and bottom of the waveform. If we think back to how a square wave is produced, it takes infinite harmonics of the fundamental frequency to combine to create the flat top and bottom of the square wave. This time-domain graph shows a signal with severe clipping.

When you clip an audio signal, you introduce square-wave-like behaviour to the audio signal. You are adding more and more high-frequency content to fill in the gaps above the fundamental frequency. Clipping can occur on a recording, inside a source unit, on the outputs of the source unit, on the inputs of a processor, inside a processor, on the outputs of a processor, on the inputs of an amplifier or on the outputs of an amplifier. The chances of getting settings wrong are real, which is one of the many reasons why we recommend having your audio system installed and tuned by a professional.

Frequency Content

Let’s start to analyze the frequency content of a clipped 1 kHz waveform. We will look at a gentle clip from the frequency and time domains, and a hard clip from the same perspective. For this example, we will provde the digital interface that we use for OEM audio system frequency response testing.

Here are the frequency and time domain graphs of our original 1 kHz audio signal once again. The single tone shows up as the expected single spike on the frequency graph, and the waveform is smooth in the time domain graph:

Low Distortion Analysis

The graphs below show distortion in the audio signal due to clipping in the input stage of our digital interface. In the time domain, you can see some small flat spots at the top of the waveform. In the frequency domain, you can see the additional content at 2, 3, 4, 5, 6 kHz and beyond. This level of clipping or distortion would easily exceed the standard that the CEA-2006A specification allows for power amplifier measurement. You can hear the change in the 1 kHz tone when additional harmonics are added because of the clipping. The sound changes from a pure tone to one that is sour. It’s a great experiment to perform.

High Distortion Analysis

The graphs below show the upper limit of how hard we can clip the input to our test device. You can see that 1 kHz sine wave then looks much more like a square wave. There is no smooth, rolling waveform, just a voltage that jumps from one extreme to the other at the same frequency as our fundamental signal – 1 kHz. From a frequency domain perspective, there are significant harmonics now present in the audio signal. It won’t sound very good and, depending on where this occurs in the audio signal, can lead to equipment damage. Keep an eye on that little spike at 2 kHz, 4 kHz and so on. We will explain those momentarily.

Equipment Damage From Audio Distortion

Now, here is where all this physics and electrical theory start to pay off. If we are listening to music, we know that the audio signal is composed of a nearly infinite number of different frequencies. Different instruments have different harmonic frequency content and, of course, each can play many different notes, sometimes many at a time. When we analyze it, we see just how much is going on.

What happens when we start to clip our music signal? We get harmonics of all the audio signals that are distorted. Imagine that you are clipping 1.0 kHz, 1.1, 1.2, 1.3, 1.4 and 1.5 kHz sine waves, all at the same time, in different amounts. Each one adds harmonic content to the signal. We very quickly add a lot more high-frequency energy to the signal than was in the original recording.

If we think about our speakers, we typically divided their duties into two or three frequency ranges – bass, midrange and highs. For the sake of this example, let’s assume we are using a coaxial speaker with our high-pass crossover set at 100 Hz. The tweeters – the most fragile of our audio system speakers – are reproducing a given amount of audio content above 4 kHz, based on the value of the passive crossover network. The amount of power the tweeters get is proportional to the music and the power we are sending to the midrange speaker.

If we start to distort the audio signal at any point, we start to add harmonics, which means more work for the tweeters. Suddenly, we have this harsh, shrill, distorted sound and a lot more energy being sent to the tweeters. If we exceed their thermal power handling limits, they will fail. In fact, blown tweeters seem as though they are a fact of life in the mobile electronics industry. But they shouldn’t be.

More Distortion

Below is frequency domain graph of three sine waves being played at the same time. The sine waves are at 750 Hz, 1000 Hz and 1250 Hz. This is the original playback file that we created for this test:

After we played the three sine wave track through our computer and recorded it again via our digital interface, here is what we saw. Let’s be clear: This signal was not clipping:

You can see that it’s quite a mess. What you are seeing is called intermodulation distortion. Two things are happening. We are getting harmonics of the original three frequencies. These are represented by the spikes at 1500, 2000 and 2500 Hz. We are also getting noise based on the difference between the frequencies. In this case, we see 250 Hz multiples – so 250 Hz, 500 Hz, 1500 Hz and so on. Ever wonder why some pieces of audio equipment sound better than others? Bingo!

As we increase the recording level, we start to clip the input circuitry to our digital interface and create even more high-frequency harmonics. You can see the results of that here:

Now, to show what happens when you clip a complex audio signal, and why people keep blowing up tweeters, here is the same three-sine wave signal, clipped as hard as we can into our digital interface:

You can see extensive high-frequency content above 5 kHz. Don’t forget – we never had any information above 1250 Hz in the original recording. Imagine a modern compressed music track with nearly full-spectrum audio, played back with clipping. The high-frequency content would be crazy. It’s truly no wonder so many amazing little tweeters have given their lives due to improperly configured systems.

A Few Last Thoughts about Audio Distortion

There has been a myth that clipping an audio signal produces DC voltage, and that this DC voltage was heating up speaker voice coils and causing them to fail. Given what we have examined in the frequency domain graphs of this article, you can now see that it is quite far from a DC signal. In fact, it’s simply just a great deal of high-frequency audio content.

Intermodulation distortion is a sensitive subject. Very few manufacturers even test their equipment for high levels of intermodulation distortion. If a component like a speaker or an amplifier that you are using produces intermodulation distortion, there is no way to get rid of it. Your only choice is to replace it with a higher-quality, better-designed product. Every product has some amount of distortion. How much you can live with is up to you.

Distortion caused by clipping an audio signal is very easily avoided. Once your installer has completed the final tuning of your system, he or she can look at the signal between each component in your system on an oscilloscope with the system at its maximum playback level. Knowing what the upper limits are for voltage (be it into the following device in the audio chain or into a speaker regarding its maximum thermal power handling capabilities), your installer can adjust the system gain structure to eliminate the chances of clipping the signal or overheating the speaker. The result is a system that sounds great and will last for years and years, and won’t sacrifice tweeters to the car audio gods.

This was the second article of the two part Everything You’ve Wanted To Know About Audio Distortion – Click Here for Part 1.

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