We’re back to bust another car audio myth wide open. This article will discuss the myth that DVC subwoofers can handle more power than a single voice coil driver. After some research, it’s clear that consumers seem to think dual voice coil (DVC) subwoofers have the equivalent of two separate voice coil assemblies, allowing them to handle twice as much power as an SVC design. That’s not the case, so let’s explain how it all works.

Subwoofers and Voice Coils

Almost all car audio speakers use what’s known as a moving coil design. These speakers include everything from a 0.5-inch tweeter to a 3-inch midrange, a 6-inch woofer or an 18-inch subwoofer. In a moving coil speaker, the current passing through a voice coil winding causes the cone assembly to push away from or pull toward a fixed magnet.

The amount of current that passes through the voice coil determines the strength of the magnetic field around the voice coil winding. A stronger magnetic field moves the voice coil farther. In most cases, the limit regarding speaker cone travel depends on the selection of suspension components. The spider and cone surround should prevent the voice coil former from smashing into the back of the motor assembly. There’s only a surround in a tweeter, but excursion requirements are minimal.

All moving coil speakers are surprisingly inefficient. On the high side, maybe 2% of the energy from the amplifier is converted into sound, though it’s usually well under 1%. The rest of the energy heats the voice coil. So, if we send 100 watts of power to a subwoofer, the voice coil becomes a 99-watt heater. This wasted heat energy is why coils rated for more power are larger in diameter, taller or have more layers. The added mass allows the coil to absorb more heat energy.

Speaker Voice Coil Construction

The image below will serve as our basis for understanding how a voice coil works. Every voice coil starts with a former. The former material might be anodized aluminum, Kapton, fiberglass, stainless steel or paper. These tubes aren’t much different from what you’d find at the core of a roll of paper towels. The technician building the voice coil will wind a conductor made of copper or copper-clad aluminum around the former. It’s worth noting that more resistance can be a good thing in this instance, so copper isn’t always the best material choice. The technician winds the conductor onto the former to create a voice coil winding of a specific length and number of layers.

The image shown above represents a voice coil with a single winding. You’d find this in most moving coil speakers like tweeters, midrange drivers, woofers and many subwoofers. As there is a single conductor, the exposed voice coil wires connect to a pair of tinsel leads, which connect to the terminals on the speaker chassis or frame. These terminals have positive and negative labels applied. If a technician applies a DC voltage to the corresponding terminals in the same polarity as the labels, the speaker cone will move outward, away from the motor.

What Are DVC Subwoofers?

The technician will start winding two conductors instead of one to create a voice coil assembly with dual windings. As the former spins, the windings lay side by side. The total length of the winding and the number of layers are the same as if there were only one conductor. Put another way, DVC subwoofers have the same amount of copper as if it were an SVC design.

A Bit of Simple Math

Let’s assume that in the SVC coil assembly, the former has a diameter of 2 inches, and the winding is 3 inches tall. The circumference of the winding is 6.28 inches. If wound with 20 AWG wire with a diameter of 0.0254 inch, there would be a total of 118 wraps for a length of about 742 inches. If we calculate the total resistance of the wire, it comes out to 0.997 ohm.

So now we have a voice coil assembly with a single winding with a nominal resistance of 1 ohm. What if we wanted to make this a DVC subwoofer assembly?

As we mentioned, the technician would wind two conductors of the same size side-by-side around the former. Given that the total area to cover is the same, we have the same amount of wire on the voice coil. However, the length of each conductor would be half as long. So each voice coil would have a nominal resistance of 0.5 ohm. Once again, there is the same amount of copper on the winding, so the power handling is the same as if it had a single conductor. A DVC subwoofer offers no power handling benefit over an SVC driver.

Benefits of Dual Voice Coil Subwoofers

So what are the benefits of dual voice coil subwoofers? The answer is flexibility and nothing more. Your installer can wire the two voice coils in series or parallel or power each individually from different amplifier channels. In our example, let’s assume the woofer with a single winding can handle 200 watts of power continuously. Therefore, the dual voice coil subwoofer can handle the same 200 watts of power, given that the coil assembly has identical dimensions.

The impedance the DVC subwoofer presents to an amplifier can change, though. It can be a 1-ohm subwoofer if we wire the coils in series. It can be a 0.25-ohm subwoofer if we wire the coils in parallel. Alternatively, each voice coil could be connected to two separate amplifier channels, presenting each with a nominal impedance of 0.5 ohm. Each amplifier channel can provide up to 100 watts of power for 200 watts in total.

Multiple Voice Coil Configurations

Many car audio companies simplify the wiring options for their DVC subwoofers by including a switch or jumper block. For example, the Rockford Fosgate T1-Series subwoofers have a Selective Woofer Impedance Fuse Termination that uses a jumper block to let your installer select between series and parallel voice coil connections. The actual jumpers inside the removable block are fuses, hence their mention in the acronym.

While most car audio subwoofers are available in single- or dual-coil designs, other options exist. For example, Harman International has a patent on Selectable Smart Impedance technology that uses three 6-ohm voice coil windings on a single former along with a switch on the basket to select between a 2- or 4-ohm impedance. In the 2-ohm configuration, the three 6-ohm coils are wired in parallel. Two coils are wired in series and connected in parallel with the remaining 6-ohm winding in the 4-ohm configuration. The result is 4 ohms. Yes, more current flows through the single 6-ohm coil, but the total power handling remains consistent because they are all wrapped together in a single assembly.

Many subwoofers in factory-installed sound systems might use triple or quad voice coil configurations. The benefit is that several low-cost, low-power amplifiers can drive the subwoofer. For example, four 50-watt amplifiers will still provide 200 watts to a subwoofer but may not need significant power supply components that would be part of a single-channel 200-watt amp.

Less Conventional Subwoofer Voice Coil Designs.

A similar application is in high-power subwoofers used in SPL competitions. A competitor might be trying to provide 8,000 watts of power to the subwoofer using four separate 2,000-watt amplifiers. A separate amplifier can feed each coil. An alternative would be to wire two pairs of coils together and connect those to a pair of 4,000-watt amplifiers. Finally, all the coils could be wired in series (or parallel) and connected to a single amp.

Many years ago, Clarion introduced a speaker system called Full Digital Sound that featured midrange drivers with six voice coils. A technology for computer speakers limited to the 5 volts of power available from a USB port was the basis for the FDS design. The multiple voice coils allowed six amplifiers to drive the speaker to reach moderate volume levels.

Myths about DVC Subwoofers

Now you know how speaker manufacturers create subwoofer voice coils, and you understand why DVC subwoofers are no better or worse than a single voice coil design. The dual voice coil design might have more installation flexibility or options, but in terms of performance, there is no benefit. If you aren’t sure which subwoofer design will work best with your car audio amplifier, drop by a local specialty mobile enhancement retailer and talk with a product specialist. They can help you choose a solution that will optimize the power production capabilities of your amplifier while offering exceptional sound quality.

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.

At least once a week, someone comments on social media or in a forum that you can’t hear low-frequency distortion from subwoofers or amplifiers. We have no idea where this myth came from. However, if you understand what distortion is and how it works, you’ll know this myth is utter and complete nonsense. Yes, we know we’ve challenged the status quo. Don’t worry; we can always back up everything we say with science.

What Is Distortion?

In car audio systems, two types of distortion affect the accuracy of the sound reproduced by the audio equipment in our vehicles. Harmonic distortion creates audio information in multiples of the sounds the system produces. The second distortion is intermodulation distortion, or IMD, which adds unwanted audio content at frequencies that are the product of two different sounds. Yep, a mouthful. Let’s explain with examples.

Example of Harmonic Distortion

Let’s start by discussing harmonic distortion. Let’s say a performer plays a B1 note on a bass guitar. With conventional tuning, that would produce a sound with a fundamental frequency of 61.74 hertz. That’s well into the range in which most car audio subwoofers play. As with all sounds, natural harmonics are present to give the instrument its tone or character. For now, we will ignore those.

In a theoretically perfect audio system, the preamp, amplifier and speaker (or subwoofer) would reproduce this 61.74-hertz tone with no additional harmonic content. If we looked at this theoretically ideal system, we’d see a single spike on a spectrum analyzer at 61.74 hertz, as shown below.

In the context of this discussion about distortion, nothing else shows in the spectral domain above an amplitude of -95 dB. We can say that the 61.74-hertz note has at least a harmonic distortion level that’s better than -92 dB. Why not -95? Well, the signal itself, in this example, was created at a level of -3 dB FS. We chose this level to leave room to add some harmonic content without inducing clipping. So, 95 minus three is 92. This -92 dB level equates to <0.002512% THD.

Let’s introduce some harmonic distortion. Harmonic distortion implies that the additional unwanted content is an even multiple of the fundamental frequency. We’ll throw in some 123.48- and 185.22-hertz information as an example.

Amplifiers and Speakers Add Harmonic Distortion

The above is typical behavior for a car audio source unit and amplifier. We can see a second-order harmonic of 61.75 hertz at 123.48 hertz at a -75 dB FS level. A third-order harmonic at 185.22 is now present at -85 dB. If we combine the amplitude of those harmonics (with a bit of fancy math), we get a level of -74.5861. To compare that to our fundamental, we would subtract three (to compare to our fundamental frequency) for a distortion level of -71.5861 dB or 0.0263%.

Consider this for a second: We fed the amp with a signal with no more than 0.0025% distortion. That level equates to harmonic content below -95 dB FS. The amplifier has added content at 123 and 185 hertz because of distortion. This distortion is information above the low-pass crossover point. It’s information the subwoofer will try to reproduce.

Example of Intermodulation Distortion

OK, we now should understand how harmonic distortion works. We get unwanted multiples of any frequency that passes through the amplifier. What about intermodulation distortion? Let’s add a G1 note from our bass guitar to our musical experience. The G1 has a frequency of 49 hertz. This frequency is also well into the subwoofer region of a car audio system. Let’s look at G1 in the spectral domain.

Nothing stands out as abnormal so far. We have our fundamental 49 hertz at a level of -3 dB FS, and nothing else. Once again, this means harmonic distortion is better than ~0.0025%. We won’t be talking about harmonic distortion, so we need to add that B1 note at 61.74 hertz to explain intermodulation distortion.

So far, everything looks logical and makes sense. We have two notes or sounds played simultaneously. Having two frequencies playing is a requirement for explaining how intermodulation distortion works. First, we need to do a little math. The difference between 49 and 61.74 hertz is 12.74, which is called the f2-f1 frequency.

Showing what IMD Looks Like

The first thing that happens when an amplifier adds IMD is the addition of audio information at this 12.74-hertz frequency. Let’s add it to our spectral frequency analysis graph.

As you can see, the amplifier has added information that wasn’t in the original recording at this 12.74-hertz frequency. The f2-f1 frequency is only the first issue related to IMD. The second issue is sidebands, which are additional distortion frequencies spaced at 12.74 hertz (in this example) on either side of the fundamental frequencies. Here’s what one set of sidebands looks like on our graph.

It’s easy to see that an amplifier or speaker that adds significant harmonic and intermodulation distortion would change how our music sounds. Remember that this is an example with only two frequencies playing, and we’ve excluded the harmonic content that the instruments would add naturally.

Every frequency from every instrument or performer is subjected to some harmonic and intermodulation distortion. Simultaneously, intermodulation distortion adds unwanted content between and on either side of every frequency.

Subwoofers and Hearing Low-Frequency Distortion

A while back, we published a short series of comparisons of subwoofers to analyze their distortion characteristics. A good-quality 10-inch subwoofer with robust excursion capabilities adds 2% to 3% total harmonic distortion between 40 and 100 hertz when playing at 90 dB SPL measured at 1 meter. Increase the output to 100 dB SPL, and you are in the 5% THD range. Yep, compared to electronics, speakers add a LOT of distortion.

Let’s reverse what that 5% distortion means if we play a 61.74-hertz note. Converting the percentage value back to a decibel number, we get -26.02 dB. To simplify the explanation, if the subwoofer created a single second-order harmonic (at 123.48 hertz), it would be 26.02 dB below the fundamental frequency. That would be very audible.

Let’s look at that in the spectral domain, shall we?

Here’s what you need to remember when looking at this chart: The information at 61.74 is the audio signal from the amplifier. As shown, it doesn’t contain any distortion. The nonlinearities of the subwoofer itself add the harmonics at 123.48 and 185.22 hertz. You will hear these sounds that were not in the original recording.

Subwoofer Crossover Points and Hearing Low-Frequency Distortion

Harmonic and intermodulation distortion add frequency content to an audio signal because of nonlinearities in a source unit, digital signal processor, amplifier, speaker or subwoofer. By a long way, speakers are the worst in the amount of distortion they add to audio signals. Choosing good speakers is crucial. Every component in the audio playback chain adds a bit of distortion. Well-engineered audio equipment adds less distortion. The result is that your music sounds more precise, more detailed and more accurate.

Let’s tie all this talk about distortion back into the context of low-frequency audio playback. First, we know that the harmonic distortion characteristics of an amplifier and the subwoofer itself will add audio information to what we hear. This harmonic content will primarily focus on the 60- to 250-hertz range. Above those frequencies, harmonic levels drop off to below audible levels as other audio information will mask them. There will also be intermodulation distortion content that’s mixed in with the original frequencies.

So what does this sound like? The higher frequencies cause the subwoofer itself to be much easier to locate in an audio system. We typically choose a steep crossover point around 80 hertz for the top of the sub. However, harmonics at relatively high levels, one or two octaves above that crossover point, can trick us into hearing the sub-bass from behind the listening position. Of course, this assumes your subwoofers are behind you in the vehicle. If the amplifier and sub were perfect and did not play any audio much above 80 hertz, it would be much harder to pinpoint the subwoofer in an audio system. Aside from time-alignment phase issues, if you can pick out the location of a subwoofer easily, it’s probably adding a lot of unwanted distortion.

Hearing Sounds Not in Your Music

From a tonal standpoint, a subwoofer system with moderate to severe distortion of harmonics usually sounds boomy or tubby rather than tight and dynamic. The addition of unwanted midbass harmonics changes the sound of the instrument. In our theoretical perfect recording, our bassist playing only that B1 note creates only audio content at 61.74 hertz. Once distortion has affected the signal, we hear more like a good pluck of the B1 and a little bit of B2 and B3. It’s just not the same thing. Now multiply that by every note they play. What you hear is a slightly different instrument. You’ll still know it’s a five-string bass, but it won’t sound the same.

A kick drum generates another good low-frequency sound that can be affected by unwanted distortion. The sound of the beater hitting a kick drum’s skin is unique. When analyzed critically, it’s easy to pick apart. Depending on the drum and its tuning, you might have a fundamental around 40 or 50 hertz for an 18- by 24-inch kick drum. In a good recording, you can pick out the sound of the beater hitting the drum head. You can also hear the resonance of the sound bouncing back and forth inside the body. Hearing this requires that the audio system be balanced correctly in the spectral domain and have the lowest distortion possible.

Is Bass Distortion as Easy to Pick Out as Midrange Distortion?

So, why does the myth that we can’t hear low-frequency distortion exist? Almost all of us are used to hearing voices. No, the voices with our ears, not the “in our heads” kind. Many of us talk with people almost nonstop every day. When there’s something wrong with a voice in a recording, it’s very easy to detect. However, let’s say you are someone like Mark Petrocelli. Mark is the drum technician for Styx drummer Todd Sucherman, considered one of the top drummers in the world.

As with many experts in the field of drumming, Mark and Todd can tell you when the head of a drum is too tight or loose just by listening; they can probably tell you whether Todd is using a different stick or whether a skin needs replacing. Musicians simply become attuned to the instruments they play. The same goes for listening to audio equipment. If you focus on the music – the lyrics and the sound — then most gear might sound good. If you focus on the sound of each instrument, picking them out from the mix and analyzing their timbre and timing, you become very good at picking out good quality audio gear from the mediocre.

Hearing Low-Frequency Distortion is Easy

So, is it easy to hear distortion at low frequencies? Of course it is. Are most people good at detecting this distortion compared with midrange frequencies? No, not at all. However, that doesn’t mean they don’t exist.

Let’s wrap this up with a quick story. Last year, we attended an industry trade show in Canada, where we had the opportunity to audition a half-dozen demo vehicles. The one that stood out was the one that used subwoofers with significant distortion-reducing technologies. The bass was several orders of magnitude more accurate than all the other vehicles’. There was more definition to each note and better separation. Adding a shorting ring and copper inductance-reducing pole piece cap to the subwoofer design made a huge difference.

So the next time someone says that you can’t hear low-frequency distortion, you’ll know that comment makes no sense. You can hear distortion; however, it reveals itself as unwanted midbass information. No, it doesn’t stand out like garbled midrange frequencies, but it’s there. If you want a car audio system with clear, detailed, accurate bass reproduction, drop by a local specialty mobile enhancement retailer and audition their subwoofer solutions. Ask about the technologies included in the subwoofer motors that make them more accurate.

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 looking at the options for frequency response correction for audio systems, your installer has two choices: graphic or parametric equalizers. Both types of equalizers perform the same task in helping to smooth the peaks and dips in what you hear from your audio system. However, the two equalizer types deliver the same results in very different ways. Let’s take a deep look into parametric equalizers and explain how they work.

What Is a Car Audio Equalizer?

As we covered in a previous article, an equalizer is a device that allows your installer to focus on a specific frequency range and either increase or decrease the level relative to the rest of the music. Equalizers must be used with a real-time audio analyzer to correct for anomalies in the frequency response of an audio system. Everything from your smartphone and smart speaker on your bedside table to movie theaters and concert venues uses equalizers to improve an audio system’s frequency response.

What Is a Graphic Equalizer?

Before diving into the operation of a parametric equalizer, we should review how a graphic equalizer works. The number of bands included in a graphic equalizer indicates their suitability and capabilities. Each of these adjustments is called a band. Each band is assigned a specific frequency in a graphic equalizer (or EQ for short), and the number of bands defines the range of frequencies each covers. For example, if you have a 31-band graphic equalizer, each band will cover about 1/3 of an octave. Conversely, each band covers an entire octave if you have a 10-band graphic equalizer.

The technician can boost or cut these bands based on acoustic measurements made with a calibrated real-time audio analyzer. For example, if there is a 4-dB dip in frequency response at 1 kHz, the technician can boost the 1-kHz EQ band by 4 kHz. Graphic equalizers are very common in higher-end aftermarket car radios.

What Is a Parametric Equalizer?

Though the purpose is the same, parametric equalizers can be more flexible than their graphic cousins. In a parametric equalizer, your technician can select a specific center frequency and bandwidth for each equalizer band. For example, suppose a peak in a system’s frequency response is at 1.1 kHz. In that case, the technician can specify 1.1 kHz as the center frequency of the EQ and then apply whatever amount of attenuation is required to flatten that peak.

A parametric equalizer has a bandwidth adjustment labeled as Q. The concept of Q, or more specifically, Q-factor, is initially unintuitive. The Q-factor represents the ratio of the center frequency to the bandwidth the adjustment covers. A high-Q filter is very narrow, and the low-Q filter affects a broader band of frequencies.

How Is the Q-Factor Calculated?

The calculation to determine the Q-factor of an equalizer adjustment is relatively simple. That said, it’s not arithmetic we typically need to perform when calibrating an audio system. If you look at the image below, you can see the center frequency of the adjustment represented by the uppercase letter F. This would be something like 1.1 kHz, as mentioned in the previous paragraph. The following required information is the -3 dB bandwidth of the range over which the filter adjusts. Let’s use the example of the low-side to high-side -3 dB frequencies being 3,300 hertz apart. The lowercase letter f defines this bandwidth. To calculate the Q-factor, you would divide F by f, 1,100 ÷ 3,300, which equals 0.33. As you can imagine, it would take some measurement to determine the bandwidth of the -3 dB frequencies, so we never do the math to make adjustments.

A higher Q filter might be required if the technician needs to address a very narrow spike or dip. Let’s say that only 1,100 hertz of bandwidth requires boosting or attenuating. Our equation now becomes 1,100 ÷ 1,100 for a value of 1.

Common Q-Factor Knowledge

There are a few pretty common Q-factor values. If the technician wants a parametric equalizer to act like a 1/3-octave graphic equalizer, then a Q-factor of 4.318 will work. If you don’t have many equalizer bands, common in several car audio signal processors, and they want each band to cover an entire octave, then a Q-factor of 1.414 is ideal. These are often the default settings in many systems, though they are rarely suitable for any particular system.

The two images above show how a higher Q-factor affects a narrower range of frequencies. The image below is the ARC DNA software used with the Blackbird amplifier. It allows for Q-factors as low as 0.1 up to 20. The image below shows four equalizer bands, each set to add 10 dB of signal boost at a center frequency of 500 hertz. The white trace has a Q-factor of 10. The gray trace has a Q-factor of two. The green trace has a Q-factor of 0.5. Finally, the yellow trace has a Q-factor of 0.1. It’s unlikely you’d need the widest bandwidth adjustments, but they can come in handy on some processors that don’t have shelving filters.

Benefits and Drawbacks of Parametric Equalizers

The most significant benefit of a parametric equalizer is that it can zero in on specific frequency response issues quite easily. As mentioned, if a peak or valley is at 900 hertz or 1,100 hertz, then a parametric EQ is a better tool than a graphic EQ band. Many technicians and amateur enthusiasts get hung up on over-equalizing audio systems. Understanding how to interpret measurements from an RTA is crucial to making an audio system sound accurate.

A drawback of many parametric equalizers is that they are often limited in the number of adjustment bands they offer. Some processors have eight, 10 or 15 bands of parametric equalization. This is typically adequate to adjust a single speaker in a fully active three-way system, but fewer than 10 bands might be somewhat limited when used in a two-way application. We’ve seen several digital signal processors with more than 30 bands of equalization with parametric modes. In short, those will have enough adjustability for any system configuration and likely enough to get an amateur into lots of trouble.

What Is a Paragraphic Equalizer?

Just when you think you have a handle on graphic and parametric equalizers, we’ll throw in a third option: the paragraphic equalizer. As you can imagine, this is a hybrid of the graphic and parametric types. A paragraphic equalizer typically allows you to adjust the center frequency of each equalization band but not the Q-factor. These aren’t very common in car audio applications.

Is A Parametric Equalizer Necessary?

The type of equalizer used to calibrate an audio system is much less important than the accuracy and relevance of the system’s frequency response measurement. A technician’s experience in understanding what the RTA shows. It’s not uncommon for someone to adjust a system to deliver what looks like a smooth response on the computer screen, only to have it sound like it still needs work in certain frequencies. Parametric equalizers are powerful tools and can be an ideal solution to calibrate any audio system, but as with any tool, the craftsman’s skill matters the most.

When it’s time to take your car audio system to the next level, drop by a local specialty mobile enhancement retailer and ask about adding a digital signal processor to your car audio system with an adequately powerful graphic or parametric equalizer. Make sure they have extensive experience in proper audio system design and calibration, which is the key to reproducing music with realism and accuracy.

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.

The single most crucial upgrade a person can make to their car audio system is to incorporate a properly configured equalizer. Whether you have a radio and two speakers or a high-end four-way system with active crossovers, ensuring that the system’s frequency response is accurate remains the most important consideration concerning quality. Let’s examine how a car audio equalizer works and why they’re crucial.

What Is an Equalizer?

There are several ways to describe an equalizer. We’ll offer several definitions, as all do an excellent job of explaining their functionality. You can consider an equalizer to be a frequency-specific volume control. Unlike the main volume control, your installer would set this once rather than have it used frequently. Another description is that an equalizer is an audio filter that boosts or attenuates specific frequencies. Ultimately, an equalizer tool allows for the amplitude adjustment of particular frequencies in an audio signal.

In the image above, we see the response curve of an equalizer in a digital signal processor that has added 6 dB of boost at 1 kHz. Any audio signal that passes through this equalizer channel will have the frequencies at and around 1 kHz boosted by this amount.

This second image shows the equalizer set to cut or attenuate frequencies around 1 kHz by 6 dB.

An equalizer can cut or boost the amplitude of different frequencies by specific amounts. While seemingly simple, it’s a powerful and essential tool in creating great-sounding car audio systems.

Why Is a Car Audio Equalizer Important?

Imagine the perfect speaker. Let’s say it’s a 6.5-inch driver that delivers ruler-flat response from 80 hertz to 20 kHz. Magically, it has ideal directivity at most frequencies, producing even amounts of sound in every direction. It has distortion-reducing features like an aluminum shorting ring and copper T-yoke cap. It has a flat-wound voice coil for maximum efficiency. It also has a rigid cone that doesn’t resonate uncontrollably. The suspension design allows the speaker to move linearly through a wide excursion range, so it continues to sound good, even at higher volume levels. The engineers who designed this speaker would have spent significant time measuring its performance in an anechoic chamber to ensure accuracy.

Now, what happens when we put that speaker in the door of a car? It still functions the same, producing audio frequencies at the same amplitude with magically perfect dispersion. Would it sound great? Probably not. Why? Because the listening environment dramatically affects what we hear. This change in frequency response isn’t the speaker’s fault. Notably, a speaker should never be designed for a specific listening environment. Speakers should be as accurate and faithful to the source signal as possible.

Why does the environment affect what we hear? The answer is reflections and resonances. When we listen to a speaker in a car or truck, we hear the sound directly from that speaker. We also hear sounds that arrive a moment later that have bounced off the roof, windshield, side window, seats, floor, dash and center console. These reflections change our perception of the system’s sound reproduction. Depending on the distance from the speaker to the reflecting surface and from the surface to the listening position, the reflections might add emphasis at specific frequencies or reduce the amplitude at others.

Another issue is resonance. Because of the distances between certain surfaces, we can get additional peaks in the system response. For example, a distance of 5 feet between the left and right windows might cause a peak in the system response at 225 hertz and a smaller one at 450 hertz. If the distance from the roof to the floor is 3.5 feet, that might cause a resonance at 321 hertz. The materials on the surfaces affect how much resonance occurs. If your vehicle has a deep plush carpet and cloth roof, the resonance might not be as significant as if it had a vinyl floor and roof. The listening environment wreaks havoc with what we hear from our perfect speaker. The result is peaks and dips in the frequency response, and the system doesn’t sound natural.

How an Equalizer Improves Audio System Performance

Before we can adjust an equalizer, a technician must accurately measure what might need fixing. We use a real-time audio analyzer to make these measurements. A real-time analyzer is essentially a calibrated microphone and display that shows the frequency response of the measurement. Most car audio technicians use computer-based audio analyzers with a single microphone or a multi-microphone array.

Audio analyzers need to be very accurate. If they show a dip at a specific frequency, we should be able to apply a measurable amount of boost to that frequency with an equalizer to produce a flat response curve. This requirement means the microphone must be calibrated accurately. A microphone used for music recording or voice communication likely isn’t ideal for audio measurements as it will impose its frequency response variations.

In the case of all audio analyzers, knowing how to interpret the data they offer is crucial to making audio system adjustments. Understanding measurements and knowing what to adjust in an equalizer takes extensive training to deliver great-sounding audio systems. With that said, the basic concept is that the RTA graph will show peaks and valleys in the acoustic response of an audio system. The technician can then cut (attenuate) peaks or boost dips to deliver a system with a smooth response.

Types of Car Audio Equalizers

There are two ways to discuss types of car audio equalizers. We can look at the different physical solutions or how the equalizers adjust signals. We will discuss the available solutions in this article and save the operational differences for another time.

The simplest of equalizers would be the bass and treble controls in a car radio. Turning either up or down affects a relatively wide range of frequencies, and these are better suited to changing the overall tonal balance of a car audio system than correcting response issues. Many late-model vehicles include bass, midrange and treble control adjustments in their infotainment systems.

Regarding system frequency response correction and calibration, you need fine control over different frequency ranges. For the context of this article, this will mean you need a lot of bands of equalization. A band is a single EQ adjustment that can boost or cut a narrow range of frequencies. Companies like AudioControl, Phoenix Gold and PPI had analog 30- or 31-band equalizers that were popular with autosound competitors years ago.

These days, digital signal processing has made equalization easy. From source units like the Sony XAV-9000ES and XAV-9500ES to stand-alone processors from companies like Audison, Rockford Fosgate and ARC Audio, access to the tools required to calibrate modern car audio systems is easy.

These processors use software on a computer or iPhone/iPad to allow the technician to calibrate your audio system to adjust the equalizer bands on each output channel. As mentioned, we aren’t going to get into the details of the types of equalizer modes (graphic or parametric) in this article.

How Do I Know My Car Audio System Needs Equalization?

While all this information is essential, knowing whether your car audio system requires equalization is equally important. Unless it’s a factory-installed system in a relatively new model vehicle, the answer is yes, it needs equalization. Why don’t those systems need equalization? The answer is simple – they’re already set up for the speakers that came with the vehicle. Even modest, unbranded systems often include equalization in the radio that helps to deliver smooth frequency response.

If you’ve ever heard someone say they upgraded their radio or speakers to an aftermarket unit, but the system doesn’t sound as good, this is why. The equalization would be removed if it were built into the factory-installed radio. It might still exist if the new radio were feeding a factory-installed amp. At the other end of the spectrum, new speakers likely have different frequency response curves than what came from the factory. Changing speakers essentially invalidates the equalization.

If you’re building a new system from scratch with a premium radio and great amplifiers, speakers and subwoofers, part of the system design should include a way to equalize the system. You could opt for a radio like the aforementioned Sony units with an equalizer on each channel. You could also use a stand-alone digital signal processor between the source unit and the amplifiers. Another option is to choose amplifiers that have built-in digital signal processing.

If you’re serious about optimizing your audio system, you’ll need an equalizer for each speaker. The corrections required for a woofer in the front door will differ significantly from that of a speaker in the rear door or on the parcel shelf. More importantly, the equalization will be very different on the left side of the vehicle compared to the right. Having a single EQ for the entire vehicle is better than nothing, but if you want rock-solid imaging and fantastic realism, you need equalization for each channel.

You need someone with experience to set up the EQ. As we mentioned earlier, knowing how to interpret the measurements from the RTA is crucial. Adding equalization that won’t positively affect the system’s performance is easy. There might be a peak or dip on the response graph of the RTA, but it might be something that can’t or shouldn’t be fixed with an EQ. Having a thorough understanding of the laws of physics and extensive experience are crucial to creating a system that sounds genuinely lifelike.

Upgrade Your Audio with a Car Audio Equalizer Today!

If you want to elevate the performance of your stereo system, then you need a properly calibrated car audio equalizer. Start by visiting some local specialty mobile enhancement retailers in your area. Audition their demo vehicles or work they’ve done for other clients. If those systems sound good, they can likely deliver similar results in your vehicles. Don’t be afraid to shop around. You may have to travel to another city to find a shop that can deliver what you want.

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.

Here are three sentences that make car audio experts cringe: “I set the gains halfway” is definitely at the top of the list. “Can I use a 60-watt amp with my 80-watt speakers?” demonstrates a complete misunderstanding of the irrelevance of power ratings. Something like “I tried the remote bass control from my Sony amp in my Rockford Fosgate amp” couldn’t be scarier. Can you imagine the potential for damage? Oh, maybe you can’t. OK, let’s talk about car audio amplifier remote level control compatibility.

What Is a Car Audio Amp Remote Level Control?

Many modern amplifiers, particularly monoblock subwoofer or multi-channel amplifiers with a dedicated high-power subwoofer channel, have an option for or include a remote level control. A level control or bass control is typically a tiny metal box or plastic enclosure with a knob on the front. Your installer can mount the control in the dash or center console so it’s easily accessible. Depending on the amplifier design, the level control might adjust the output of the amplifier, or it may alter a bass boost circuit. Some are fancy and include multiple knobs and a button. Some even include voltage displays and clipping indicators. Ultimately, these controls provide an easy way to adjust the bass level for different music or listening preferences.

The simplest of these controls use a potentiometer. Turning the knob on the potentiometer changes the resistance. In most amplifiers, the remote level control attenuates the signal from the source unit by creating a voltage divider. If you turn the knob up, you get all the signal from the amp. If you turn it down, you get either no signal or a specific percentage of the signal. How the level control works depends on the associated circuitry in the amplifier.

Most potentiometers have three terminals. The outer terminals connect to either end of a resistive element or path. The middle terminal connects to a wiper arm that’s connected to the adjustment knob. When you turn the knob, it moves the arm along the path. The resistance between one end terminal and the wiper connection changes as you turn it. If it’s turned close to one terminal, the resistance will be minimal. The resistance will be large if it’s turned far from one terminal. Potentiometers are sold based on the total resistance of the path or element.

Car audio amplifiers often use multi-ganged potentiometers for circuits like crossovers or level controls. These are multiple potentiometers connected to the same shaft. For example, if a stereo amplifier has a single sensitivity or gain control, separate potentiometers for the left and right channels would be needed. These potentiometers are mounted on a single shaft for convenience.

In applications where the remote control adjusts a bass boost, the minimum setting on the control usually applies no boost. Turning the control to its maximum applies 12 or even 18 dB of boost around a specific frequency. Once again, the amount of boost and center frequency depend on the amplifier’s design.

Level Controls for Digital Signal Processors

There’s a second type of level control dedicated to digital signal processors. Some companies have simple potentiometer-based controls that adjust an analog signal that feeds back to the microprocessor in a digital signal processor. The processor interprets the signal’s amplitude and then applies that to something in the processing path. The DSP software might allow the installer to configure the remote level control as a bass boost control, a subwoofer level control, a master volume control or even a center-channel level control. In these cases, the control itself is still a simple potentiometer. It’s the software in the processor that adds the flexibility.

Many processors also have compatible computerized level controls. Rather than analog signals or voltages, there’s digital communication between the remote and the computer processor in the DSP. These typically use communications with data lines, power, ground and possibly illumination connections.

Remote Level Control Connections and Wiring

Here’s where things get scary for those who understand basic electronic circuit design. Let’s start by discussing the different connections on modern amplifiers for remote bass or level controls. Typically, you’ll see an RJ45, RJ12 or RJ11 jack, or maybe a 3.5-mm or ¼-inch headphone-style jack.

Simple Level Controls

The level control might require no more than two electrical connections if it’s a simple design. One wire would go to the potentiometer wiper and the other to a terminal at the end of the path or element. As the wiper turns, the resistance between the connections increases or decreases, increasing or decreasing the signal amplitude sent back to the amp.

Here’s the first opportunity for a remote level control from one amplifier to be incompatible with the circuitry from another amp. Let’s say you have a Rockford Fosgate remote, and its potentiometer has a 1,000-ohm rating. For some reason, you want to use it with a Sony amplifier. The Sony remote might use a 10,000-ohm potentiometer. It might work, but the effective adjustment range might be wrong. Worse, there could be a change in the circuitry function, which might boost the signal. This increase in amplitude could overdrive portions of the circuitry and cause massive amounts of distortion.

Wire Connections

The second instance where something might go wrong is that the remote level control and amplifier use a connector with more than two pins. It might have three, four, six or eight. Even if it only uses two connections, the pins must be in the proper position on the connector. The two connections could be backward and still work, but they must match the traces on the amplifier circuit board. In this case, you would likely get no output from the amplifier. If all three wires connect to the potentiometer, you may get all the signal or none of it.

What about a fancier remote that includes something like an LED that illuminates when the amplifier turns on? That LED will need power from the amplifier. It might be a low voltage in the 2.5-to-3-volt range or as high as 12 volts with circuitry on the circuit board in the remote. What if the pinout from a remote didn’t match that of another amplifier, and you feed 12 volts DC into an audio signal path? You could easily damage the amplifier. If the LED is expecting a low voltage, and you send it something substantial, you’ll burn it out instantly.

Completely Incompatible Remote Technologies

Below, you’ll see an image of the JL Audio DRC-205. This remote is designed for their FiX integration processors, TwK calibration processors or VXi-Series DSP-equipped amplifiers. It’s easily one of the best-looking remotes on the market. The remote includes two rotary controls, a multi-color LED and a pushbutton. The color of the LED can indicate which processor preset is loaded or what mode the device is in. The button can change the presets on a DSP. The rotary controls can serve as master volume control and subwoofer level control. Best of all, it’s easy to take the knobs off the remote and mount it neatly on the dash or in the center console of a vehicle. It can be upgraded with the optional VXi-BTC Bluetooth communicator to function wirelessly.

The DRC-205 uses a standard eight-position RJ-45 connector and CAT-5e network cables to connect to supported processors. Can you imagine the havoc that would ensue if you connected a conventional analog remote to the JLid port on a VXi amp or a DRC-205 to the remote level control port on a different brand of amplifier? The results would be instantly catastrophic.

Don’t Experiment with Car Audio Amplifier Remote Level Controls

We’ll wrap this up with an unambiguous statement: Never connect the remote level control from one amplifier brand to another. In addition, unless you’re sure of the compatibility, there may be issues with the compatibility of level controls from one series of amplifiers to another within the same brand. If you’ve lost the level control that came with an amp or need to purchase an optional one, get the specific unit designed for the amplifier you have. Don’t risk guessing. Drop by a local authorized retailer for the brand of amplifier you need the remote for. They can help you get the current part to ensure that everything functions properly.

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.