Social media conversations are often a great source of content ideas here at BestCarAudio.com. We were recently talking to someone about why loading down an amplifier isn’t an ideal solution. As we’ve demonstrated in our Test Drive Reviews, lower impedances result in more distortion in most cases. In all cases, lower impedance loads reduce amplifier efficiency. Let’s discuss why amp efficiency is crucial to audio system performance and reliability.

What Is Amplifier Efficiency?

Every electronic device consumes more power than it can put out. It might take 50 watt hours to charge a power bank for your cell phone, but you may only get 48 watt hours out of it. That’s an efficiency of 96%. The same applies to mechanical systems. Friction and heat losses mean you must put more energy into a device than you get out. Comparing the power capabilities of an engine on an engine dynamometer to the power at the wheels on a chassis dynamometer is an excellent example of a system with substantial mechanical losses.

The circuitry in car audio amplifiers consumes some of the energy provided by the battery. Most good amplifiers we’ve tested draw about 1 to 2 amps of current while idling and not playing music. At the other end of the spectrum, a great amplifier might be about 83% efficient when driving a 2-ohm load at its maximum output capability. This efficiency specification means that the amp would consume 100 watts for every 83 watts it fed to a speaker. These efficiency numbers only apply at full power, as efficiency drops quickly at lower output levels.

Where Does the Wasted Energy Go?

What happens to the extra energy that an amplifier consumes if it isn’t sent to a speaker? Well, the circuitry that processes the audio signal requires a little bit. If there is a digital signal processor in the amp, that will consume a little power as well. For the most part, however, the energy is wasted as heat. Heat is caused when current flows through a resistance. The formula to calculate the power a resistor dissipates is I^2 x R, where I is the current in amps and R is the resistance in ohms.

Even small increases in current result in significantly more energy being wasted. This is obvious because the current value is squared in the equation. For example, if we have 5 amps of current flowing through a 1-ohm resistor, that resistor will dissipate 25 watts of heat. You need a reasonably large resistor to dissipate that much energy. If we increase the current to 8 amps, the resistor dissipates 64 watts of heat. A 75-watt resistor is quite large.

Whenever you see a tiny car audio amplifier, ask yourself, how efficient is the design? Small heatsinks have a tough time dissipating large amounts of thermal energy.

Amplifier Efficiency Comparison

The good amplifier for this article will be the Rockford Fosgate T500-1bdCP that we subjected to a full test drive review in early 2024. This is the amp with the larger footprint mentioned above. The amplifier is rated to produce 500 watts into 2- and 1-ohm loads, but it actually delivered 554 and 697 watts, respectively. Those power production numbers were at 83 and 68% efficiency.

The other amp is rated to produce 700 watts to a 2-ohm load and just over 1,000 watts to a 1-ohm load. On our test bench, it could only muster 338 and 664 watts using the CTA-2006-D standards for power testing. For argument’s sake, we’ll call that half of what it’s rated for. We did push the amp harder to see if it had any more output, and it managed 660 watts and 935 watts when our D’Amore Engineering AMM-1 indicated clipping.

Understand Tool Limitations

This is a crucial reminder that the clipping light on the AMM-1 is NOT a 1% THD+N indicator. As such, power measurements taken with the AMM-1 are not comparable with CTA-2006-D-compliant manufacturer specifications. You’ll need another tool to measure distortion to determine when to stop increasing power. Of course, we suggest the QuantAsylum QA403 for this task, if you can manage to get your hands on one. Nevertheless, the guest amp was 69 and 58% efficient in delivering these higher numbers, which is abysmal. No, this isn’t some no-name flea market or internet brand. It’s something that many “high-end” shops sell every day.

Let’s crunch some numbers to determine how much current each amplifier draws to produce the measured power. Then, let’s add a column that looks at how much power these amplifiers can produce per amp of current they consume. For fun, we’ll add another column to show how much power they waste as heat.

It’s easy to see that the Rockford Fosgate amp is significantly more efficient when you break down the numbers this way. It produces almost 10.8 watts per amp of current compared with just under 9 watts per amp for the other unit. That’s 20% better efficiency than the guest amp. When driving the 1-ohm load, the RF is 17.2% more efficient.

Reason One Why Car Audio Amp Efficiency Matters

Every part of the power supply chain in a vehicle has some efficiency losses, from the battery and alternator to the power and speaker wire, amplifier and speakers or subwoofers. Let’s use our efficiency numbers above to compare a pair of hypothetical 1,000-watt amplifiers. We’ll call these the RF1000 and the G1000. The chart below shows how much current each draws to produce 1,000 watts of power to a 1-ohm load and 750 watts into a 2-ohm load based on the above measurements.

Now, let’s do some math on how much power is wasted in a 16-foot run of 4 AWG power wire and on the return path of a vehicle chassis with the same resistance. For the math, we’ll use the ANSI/CTA-2015 standard for 4 AWG power cable resistance.

The difference in voltage drops between the two amplifiers isn’t massive, at 0.12 and 0.167 volt, respectively, in favor of the theoretical Rockford Fosgate amplifier. However, if the amplifier has less voltage, it will reach its maximum output at a lower level. Neither amp would likely produce 1,000 watts of power as they would only see about 12 of the 13 volts provided by the electrical system.

What’s more of a concern is the heat wasted in the power connection. As we mentioned, the power dissipated in a resistor is based on the square of the current flowing through the resistor. As such, small changes in current flow produce moderately significant changes in how much heat is produced. Looking at the Power Wasted column in this chart, the less efficient amplifier results in a 50% increase in wasted energy in the power cable and ground return path. This increased heat will increase resistance, further increasing the amount of heat wasted, and so on. It can turn into a runaway condition if you’re trying to deliver maximum power for an extended period.

Amplifier Cooling Capacity

The amount of power an amplifier wastes as heat, combined with the size and efficiency of the heat sink, determines how long the amplifier can play before it overheats. In the case of the T500-1bdCP, we played it at full rated power for more than an hour without the amplifier overheating and going into protection. On the other hand, the guest amplifier lasted less than two minutes, playing at full power.

The guest subwoofer amplifier measures roughly 12.5 by 6 inches and is 2 inches tall. That’s a volume of 150 cubic inches. The second amp is much larger at 11.5 by 9 by 2.25 inches, which is just under 233 cubic inches. While this isn’t how heatsink capacity is calculated, it does give you a rough approximation.

The size and design of an amplifier heatsink matter significantly. Yes, the Rockford Fosgate amplifier is much larger. This is a conscious design decision that the Rockford Fosgate engineering team made to ensure that it would continue playing under the most extreme conditions. You should be wary when you see high-power compact amplifiers. It’s unlikely that these companies have developed a magical solution to improve heatsink or amplifier efficiency. As such, you sacrifice thermal stability and amplifier longevity for size. We know we’d rather have a larger amp that will last for decades than a small one that might dry out the thermal compound and capacitors in a few years.

Amplifier Quality Means Many Things

While we focus a lot on content that explains the sound quality and performance of car audio amplifiers, understanding their efficiency characteristics is crucial to choosing a solution that will perform reliably and stand the test of time. Not all manufacturers publish accurate amp efficiency data, making it harder to purchase the best solution. One tip is to look at some brand-specific Facebook groups to see who still uses equipment from decades ago. Those companies likely understand how to deliver a total package that checks all the boxes for performance. Drop by a local specialty mobile electronics retailer today to discuss your high-performance car audio amplifier needs. Hopefully, they will have details on the most efficient products they offer.

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.

Listening to a genuinely high-end home or car audio system can be amazing. For those who care about sound quality, several technical considerations separate a very good audio system from one that’s truly magnificent. These might be beyond what most strive for in their car audio upgrade. With that said, if your goal is a genuinely realistic music experience, these are essentially the criteria you seek. Let’s start by talking about car audio system frequency response.

What Is Car Audio System Frequency Response?

While we could dedicate a dozen articles to the topic, the concept of frequency response describes the relationship between the amplitude of different frequencies in a car audio system. A generalization would be that we want the bass, midbass, midrange and high frequencies to be balanced so that none stands out.

Car audio system design and calibration are more complicated and involved than in a home audio system. Sitting in your driveway listening to music differs significantly from driving down the road. While in motion, the vehicle generates noise from the tires, the drivetrain and the exhaust system. There’s wind noise and noise from other vehicles. Primarily, this noise resides in the lower frequency range. If we want to hear the bass information in our music over this noise, the music needs to be louder in this range.

We’ll make a clear distinction: A car audio system calibration designed to sound good while driving will not sound identical to what the recording engineer heard in the studio. There will be more bass. This doesn’t ruin the experience, as bass can be a lot of fun. However, we must remember this as we move forward with this discussion about frequency response.

Target Response Curves

Depending on the retailer you’re working with and the brands they offer, they might have several target response curves to calibrate an audio system. These curves differ primarily in their midbass output levels, though some have different high-frequency characteristics.

As you can see from the image above, there have been many different targets for audio systems over the years. These days, companies like Audiofrog, Harman (JBL/Infinity), JL Audio, ARC Audio and Hertz/Audison have specific response targets that are either built into their processors as a target or have been provided to their dealers as a suggested reference for system calibration.

Target Curve Discrepancies

While all the curves are similar in overall shape, the response between 100 and 500 hertz remains controversial. Let’s use the example of a well-known performer with a deep voice: Johnny Cash. While best known for his country music, his version of “Hurt” by Nine Inch Nails is popular among many music enthusiasts. His voice has no trouble reaching frequencies below 100 Hz. His voice won’t sound natural if there’s 6 dB of boost at 100 hertz.

So why do some curves have this bump in midbass response? For percussion to sound more fun. A drum like a 16-inch floor tom might be tuned to have a fundamental frequency of 80 or 120 hertz. Boosting the amplitude of those frequencies adds a lot of impact to the music. However, it does so at the detriment of vocal accuracy. This is where the discussion of personal preference comes into play. Do you want a car audio system that’s fun or one that’s accurate?

Reference-Quality Car Audio

Our team members have spent decades auditioning car audio components and audio systems of all genres – home, car, studio, theaters and live performances. We’ve heard instruments played live with no amplification and thousands of recordings with every imaginable level of processing. We each have our personal preferences for system calibration. There’s one common element: smooth midrange.

If you’re listening to Adele, Ed Sheeran, Billie Eilish, Lorde, Ozzy Osborne or Bruce Dickinson, their voices should sound real. That’s the bottom line. Billie shouldn’t sound screechy. Lorde shouldn’t sound nasally. Dickinson shouldn’t sound boomy. We listen to people speak most of the day, every day. We can tell our friends apart by the sound of their voices. It’s literally their signature. So if your goal is a high-end car audio system, getting the voices right is the number one goal. Yes, you can have more bass or slightly laid-back highs. That’s the personal preference aspect of system calibration. However, the system needs more work if voices don’t sound incredibly realistic.

How To Recreate Great Vocals in a Car Audio System

There are two primary requirements to making a performer’s vocal sound incredibly realistic: high-quality speakers and proper system calibration. As we’ve said, speakers are the most essential part of any car audio system. If you choose a low-quality speaker that doesn’t offer a smooth frequency response, the chances of unwanted distortion being added to the playback are very high. We aren’t talking about clipping but harmonic and intermodulation distortion artifacts. This information can’t be removed from the system with processing. As such, it colors or adds to the sound, making it sound less than real.

If you have a speaker with a ripple in the frequency response at 700 hertz, that’s likely a resonance in the cone, surround or dust cap. The ripple, usually represented by a small peak in the output, adds unwanted energy around that frequency. That bump is audio content that is not in the recording. It is crucial to choose speakers with ruler-flat frequency response curves and use them in the correct frequency range.

Next, your installer must ensure that your audio system is designed and configured correctly. If you have a 6.5-inch midrange in the door and it’s paired with a tweeter that can only play down to 3 kHz, the directivity characteristics of the woofer will result in the system’s frequency response being sensitive to the listening position. However, if a midrange driver that can play 300 to 3,000 hertz is added, the system will deliver smoother sound everywhere in the vehicle. The technician calibrating the system can set the two-way system for the vehicle driver’s seating position. Technically, the system must be recalibrated if the driver sits more upright or farther back. You can learn more about speaker directivity in this article.

Time Matters

The other considerations in system configuration are crossover and delay settings. We’ll spend an entire article in this series talking about soundstage position and the source of different sounds. Suffice it to say that all the music in a high-end car audio system should sound like it’s coming from a single point.

There’s a caveat to the above statement: Ninety-nine percent of the car audio industry thinks a car audio system should sound as though you’re sitting in the front row at a concert with the musicians spread out in front of you as if the dash were a miniature stage. Some consumers don’t like their audio systems to sound that way. They want to be immersed in the listening experience. This is similar to what you’d hear at a dance club or when wearing headphones.

There should be well-defined left and right channels, but the forward-to-back placement is very different. Ultimately, the music should be coherent. You want the midbass to sound connected to the vocals and the bass to connect to the midbass. The source of each frequency range should be transparent. For example, if your system has a soundstage on the dash, the midbass and bass should also sound like they’re on the dash. This is achieved using high-quality audio equipment and proper system configuration and calibration.

Why System Calibration and Frequency Response Matters

In the past, equalizers were monaural devices installed above or below a radio. Even with a dozen adjustment bands (or more), these units were more of a tone control than a correction device. Modern car audio systems with a digital signal processor can be made to sound stunningly realistic if the technician has the tools and training to complete the process properly.

In concept, all a technician must do is adjust the acoustic output of a system to match the curve. However, many techs go down the path of making unnecessary adjustments to fix peaks and valleys on the RTA that are attributable to microphone location rather than actual acoustic anomalies. A single microphone can only provide so much information, so microphone arrays are becoming more popular. Companies like Audison, with its bit Tune system, offer multi-mic solutions that give an averaged measurement of the listening environment. This can help eliminate some of those peaks and valleys and speed up the process of getting the equalizer adjustments correct.

Experts Can Improve Your Car Audio System

There have been endless discussions about “the last 5%” of the calibration process regarding eking out the most realism from an audio system. Can a microphone tell you everything? No, it can’t easily measure the source of a sound. For that, human hearing is a better tool. Can a technician make a system sound amazing without any final tweaks made by ear? It’s likely, but it depends on their process. The best-sounding systems we’ve ever heard had final adjustments based on decades of experience in calibrating audio systems. These car audio systems transcend “good,” rendering voices and instruments with amazing detail, clarity and tonal accuracy. That accuracy is the result of proper system calibration concerning frequency response.

It’s amazing to hear a vocalist sound like they’re out on the hood of a vehicle with the performers spread out on either side. Achieving this takes planning, thoughtful execution and great quality products. It’s very achievable and not as expensive as many would think. Drop by several of the local specialty mobile enhancement retailers in your area and ask for a demonstration of the high-end audio systems they’ve created. Approach each experience with an open mind and ignore the make and model of the vehicle, the brands used in the installation and where components are installed. Play your favorite music, close your eyes and just listen. If the shop can deliver a more accurate presentation than others, you can move to the system design and produce selection phase for your vehicle.

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.

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.