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.

Click Here to Read : Everything You’ve Wanted To Know About Audio Distortion – Part 1

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 we talk about any signal, be it audio, video or data, there is an accompanied reality for alterations and errors made to that signal as it passes through different electronic components, conductors or magnetic fields. While we get concerned when we hear that a component introduces distortion or when we read distortion specifications, distortion is part of nature and is simply unavoidable. Until any distortion reaches a significant level in an analog signal, it can’t be heard or seen.

Starting With A Foundation in Audio Distortion

With that in mind, let’s create a foundation for observing and understanding the properties of an audio signal in the electrical and frequency domains. This information will serve as the foundation for understanding distortion in part two of this article.

Any signal, be it Direct Current (DC) or Alternating Current (AC), can be analyzed in two ways – in its time domain or frequency domain. Understanding the difference between these two observation domains will dramatically simplify the life of anyone involved in the mobile electronics industry.

When we observe a signal in the time domain, we are looking at the amplitude of the signal relative to time. Normally, we would use a voltmeter or oscilloscope to look at signals in the time domain. When we consider a signal in the frequency domain, we are comparing the amplitude (or strength) of individual frequencies, or groups of frequencies within the signal. We use an RTA (real time analyzer) on a computer or handheld/benchtop devices to look at the frequency domain.

Direct Current

When analyzing the amplitude of an electrical signal, we compare the signal to a reference; in 99% of applications, the reference is known as ground. For a DC signal, the voltage level remains constant with respect to the ground reference and to time. Even if there are fluctuations, it is still a DC signal.

If you were to chart the frequency content of a DC signal, you would see it is all at 0 hertz (Hz). The amplitude does not change relative to time.

Let’s consider the DC battery voltage of your car or truck. It is a relatively constant value. Regarding amplitude versus time, it sits around a 12.7-12.9 volts on a fully charged battery with the vehicle off. When the vehicle is running and the alternator is charging, this voltage increases to around 13.5 to 14.3 volts. This increase is caused because the alternator is feeding current back into the battery to charge it. If the voltage produced by the alternator was not higher than the resting voltage of the battery, current would not flow and the battery would not be recharged.

Alternating Current

AC Signal – Time

If we look at an AC signal, such as a 1 kHz tone that we would use to set the sensitivity controls on an amplifier, we see something very different. In the case of a pure test tone like this, the waveform has a sinusoidal shape, called a sine wave. If we look at a sine wave on an oscilloscope, we see a smoothly rolling waveform that extends just as much above our reference voltage as it does below.

AC Signal – Frequency

It is now wise to look at this same signal from the perspective of the frequency domain. The frequency domain graph will, if there is no distortion, show a single frequency. In consideration of an audio signal, the amplitude (or height) of that frequency measurement depends on how loud that single frequency is relative to the limits of our recording technology or measurement device.

Audio

When we listen to someone speak or play a musical instrument, we hear many different frequencies at the same time. The human brain is capable of decoding the different frequencies and amplitudes. Based on our experiences, and the differences in frequency and time response between one ear and the other, we can determine what we are hearing, and the location of the sound relative to ourselves.

Analyzing the time domain content of an audio signal is relatively easy. We would use an oscilloscope to observe an audio waveform. The scope will show us the signal voltage versus time. This is a powerful tool in terms of understanding signal transmission between audio components.

A Piano Note

Middle C – Time

Let’s look at the amplitude and frequency content of a sound most of us know well. The following graph is the first 0.25 seconds of a recording of a piano’s middle C (C4) note in the time domain. This represents the initial hit of the hammer onto the string. If you look at the smaller graph above the larger one, you will see the note extends out much further than this initial .25 second segment.

Middle C – Frequency

We know that the fundamental frequency of this note is 261.6 Hz, but if you look at the frequency domain graphs, we can see that several additional and important frequencies are present. These frequencies are called harmonics. They are multiples of the fundamental frequency, and the amplitude of these harmonics is what makes a small upright piano sound different from a grand piano, and from a harp or a guitar. All of these instruments have the same fundamental middle C frequency of 261.6 Hz; their harmonic content makes them sound different. In the case of this piano note recording, we can see there is a large spike at 523 Hz, then increasingly smaller spikes at 790 Hz, 1055 Hz, 1320 Hz and so on.

Sine vs Square Waveforms

Every audio waveform is made up of a complex combination of fundamental and harmonic frequencies. The most basic, as we mentioned, is a pure sine wave. A sine wave has only a single frequency. At the other end of the spectrum is a square wave. A square wave is made up of a fundamental frequency, then an infinite combination of odd-ordered harmonics at exponentially decreasing levels. Keep this in mind, since it will become important later as we begin to discuss distortion.

Noise Signals

Noise is a term that describes a collection of random sounds or sine waves. However, we can group a large collection of these sine waves together and use them as a tool for testing audio systems. When we want to measure the frequency response of a component like a signal processor or an amplifier, we can feed a white noise signal through the device and observe the changes it makes to the amplitudes of different frequency ranges.

White Noise – Time

You may be asking, what exactly is white noise? It is a group of sine waves at different frequencies, arranged so the energy in each octave band is equal to the bands on either side. We can view white noise from a time domain as shown here.

White Noise – Frequency

We can also view it from the frequency domain, as displayed in this image.

Variations In Response

The slight undulations in the frequency graph are present because it takes a long time for all different frequencies to be played and produce a ruler-flat graph. On a 1/3-octave scope, the graph would be essentially flat.

Foundation For Time And Frequency Domains

There we have our basic foundation for understanding the observation of signals in the time domain and the frequency domain. We have also had our first glimpse into how harmonic content affects what we hear. Understanding these concepts is important for anyone who works with audio equipment, and even more important to the people who install and tune that equipment. Your local mobile electronics specialist should be very comfortable with these concepts, and can use them to maximize the performance of your mobile entertainment system.

If you’ve made it this far and want to learn even more about audio distortion,  click here for Part 2 of this article!

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.

 

For decades, there has been discussion after discussion about which of the different subwoofer enclosures are “the best” and why. Let’s take a look at why we need a subwoofer enclosure at all, and how the three popular styles – sealed, vented and bandpass – differ in their design and performance.

Back-Wave Management

If you were to hook any speaker up to an amplifier, hold it in your hand and play music into it, you would find that you don’t hear any bass. That is because the sound coming from the front of the speaker cancels out the sound coming from the back. We need a way to keep the sound coming from the back of the speaker cone from interfering with the sound coming from the front. If you were to cut a hole in the middle of a large, flat piece of wood and mount the speaker in it, you would hear a lot more bass. In fact, until the half-wavelength of the bass frequencies becomes longer than the dimensions of the piece of wood, you will get really good, solid bass. If we put a speaker in an airtight enclosure, none of the sound coming from the back interferes with the sound coming from the front.

Power Handling

The ability of a speaker to use the power produced by an amplifier is limited by two criteria – how far the speaker cone can move and how much heat the voice coil of the speaker can handle. Thermal power-handling limitations are based primarily on the design of a speaker – the size of the voice coil, how airflow is managed around the voice coil and the proximity of the stationary components of the motor assembly to the voice coil are the key contributing factors. The excursion-limited constraints are also part of the speaker’s design – how long the voice coil winding is, how tall the top plate is and how much suspension travel is available are the key factors.

Excursion

When it comes to reproducing bass, a speaker has to move four times as far each time the input frequency is halved. For example, a speaker moving 0.125 inches at 100 Hz has to move 0.5 inches to reproduce the same output level at 50 Hz and 2 inches at 20 Hz. You can see that, for the lowest of frequencies, cone excursion limitations are significant – very few speakers can move 2 inches without significant distortion.

When we put a speaker in an enclosure, the combination of the enclosure and the speaker create a high-pass filter. We are effectively decreasing the low-frequency output of the speaker. Why would we want to do this? The benefit of an enclosure is that we can control the motion of the speaker cone. Looking at a simple acoustic suspension (also known as a sealed) enclosure will be the simplest illustration of this explanation.

Compliance

Each and every speaker – from the biggest of subwoofers to the smallest of tweeters – has a springiness to the cone. We call this the compliance. We measure compliance by comparing it to a volume of air with the equivalent springiness. We call this characteristic of the speaker Vas. In general terms, a speaker with a very small Vas specification has a tight suspension, and a speaker with a large Vas has a softer suspension. There is a lot more to it than that, but for the discussion of enclosure features and benefits, that’s all we need to get into for now.

When we put a speaker in an enclosure, we stiffen the suspension. When you push in on the speaker cone, you are pushing against the speaker’s suspension (which wants to center the cone) and you are trying to pressurize the air in the enclosure. When the cone tries to move outward from rest, you are putting the air in the into a vacuum state – it wants to pull the cone back to its resting position. We do sacrifice low-frequency output, but we gain significant power handling and control over the motion of the speaker cone. For the latter, the combination of the air in the enclosure and the speaker suspension helps to stop the speaker cone from moving once an electrical signal starts it in motion.

Think of it like a shock absorber on a vehicle. You can see that having an enclosure is critical.

Acoustic Suspension Subwoofer Enclosures

The simplest of enclosures is called an acoustic suspension or sealed enclosure. In these enclosures, we are putting the speaker into an airtight box. When we put a speaker in an enclosure, the system resonates at a specific frequency that – we call this Fc. Below that frequency, the output is reduced at a rate of -12 dB per octave. If the system has a resonant frequency of 50 Hz, the output will be 12 dB quieter at 25 Hz.

Acoustic suspension enclosures are amongst the smallest of the different enclosures we will discuss. They are also the easiest to construct, and most forgiving regarding calculation error. If you combine the roll-off of the enclosure and speaker system with the increase in efficiency you get from the relatively small air volume of the vehicle interior (often called transfer function or cabin gain), you can get a very flat in-car response with good infrasonic output. Bass from an acoustic suspension enclosure is very tight and controlled, thanks to excellent transient response.

There is a down side. If you are looking for loud bass, then you need a driver that has a lot of excursion capability, and you need a reasonable amount of power to move the speaker cone back and forth to get the level of output you want. There is another drawback that isn’t talked about as much, and that is distortion. As a speaker increases in excursion, the amount of distortion it creates increases. Likewise, distortion increases near the resonant frequency of the speaker. So, what can you do?

Bass Reflex Subwoofer Enclosures

A bass reflex (also known as ported or vented) enclosure uses a vent to increase low-frequency output by making use of the speakers back-wave energy. The vent, often a round tube or sometimes a rectangular slot, has an area and a length. The specific area and length of the vent and their relationship to the total volume of the enclosure cause the column of air in the vent to resonate at a specific frequency when excited by the speaker. We typically tune bass reflex enclosures quite low to emphasize the bottom octave or so of the audible frequency range. They can be tuned higher to increase efficiency for high-SPL applications. There is always a sacrifice, though – when we tune the enclosure higher, we sacrifice low-frequency performance.

Bass reflex enclosures are typically larger than sealed enclosures. There is no hard-and-fast rule to associate with the size relationship, but 25–50% large is common. The trade-off for that extra volume is two-fold – more efficiency in the tuning frequency and more power handling, at some frequencies.

When the subwoofer used in a bass-reflex subwoofer enclosure produces frequencies around the resonant frequency of the vent/enclosure combination, the driver excursion is reduced to almost nothing and all the “work” is done by the vent. Put more succinctly, around the tuning frequency, most of the music is being produced by the vent. The benefit to this is that power-handling problems caused by cone excursion limitations are dramatically increased. Since the cone is barely moving, very high sound pressure levels can be achieved. Around the tuning frequency, power handling is limited by the thermal capabilities of the subwoofer.

As we mentioned earlier, one factor that contributes to loudspeaker distortion is cone excursion. With a bass reflex enclosure, the driver moves significantly less than with an acoustic suspension enclosure design. As long as the vent itself has enough area and a smooth transition at both openings, the distortion produced by a properly designed bass reflex enclosure can be impressively small.

Nothing is free, is it? A factor in deciding to use a bass reflex design is how fast the output decreases below the tuning frequency. Where an acoustic suspension enclosure rolls off at -12 dB per octave, a bass reflex enclosure rolls off at 24 dB per octave. Below the tuning frequency, the vent acts more and more like a hole in the enclosure, offering increasingly less back pressure as frequency decreases. Designing for, and managing, driver excursion is a fundamental part of bass reflex enclosure design.

Bandpass Subwoofer Enclosures

We will quickly touch on bandpass enclosures to wrap up this article. There are several different designs for bandpass enclosures. Some use a sealed enclosure, and some a vented one. Independent of whether the rear chamber is sealed or vented, the output of the subwoofer plays into a vented enclosure. This enclosure acts as a low-pass filter. Why would we want to design a bandpass enclosure?

First and foremost, all of the output of the enclosure is produced by the vent or vents. This allows a creative designer to build an enclosure in the trunk of a vehicle and have the vent opening play through the rear parcel shelf. There have been some amazing bandpass enclosures build in the front storage area of mid- or rear-engine vehicles. The vent allows the bass to enter the interior of the vehicle. Bandpass enclosures can also offer impressive gains in efficiency over acoustic suspension and bass reflex enclosures, but they do so at the sacrifice of bandwidth and enclosure volume.

A bandpass enclosure has two resonant frequencies – one for each of the enclosures. The resultant management of cone excursion can allow a great deal of bass to be produced from limited excursion drivers. While the speaker cone itself does not move a great deal, the amount of work done by the motor assembly is still significant. You are still putting power into the speaker, and work is being done. Because the front chamber of the enclosure acts as a filter, it can also be very difficult to hear when the speaker is distorting.

Regarding the complexity of design, and forgiveness of construction error, bandpass enclosures are the most complicated to execute perfectly. Unlike an acoustic suspension or bass-reflex design, bandpass enclosure designs must be tailored exactly to the speaker they are being used with. Never trust the concept of a “generic” bandpass enclosure.

Lastly, because a bandpass enclosure includes an acoustic low-pass filter, it has to be used with good-quality, appropriately sized midbass drivers. If not, the bass can sound lost or disconnected relative to the rest of the music.

For More Details On Subwoofer Enclosures, Visit Your Local Specialist

As you can see, there are many ways to install a subwoofer – or any speaker, for that matter. Navigating the available space in the vehicle, as well as different speaker sizes and designs, can be tricky. The design and construction of an enclosure can be complex, especially when complex shapes are involved. Visit your local car audio specialist retailer to explore different enclosure options 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.

When it comes to high-current wiring in a vehicle, there are two types of stranded power wire available: solid copper and copper-clad aluminum. This article looks at the differences between each kind of wire, and explain the challenges of ensuring your high-current device gets the power it needs to do the job you want done.

Car Audio Power Wire: Background

In mobile applications, or anywhere that a conductor may be exposed to movement or vibration, it is recommended to use only stranded conductors. Solid conductors (like single-strand house wiring) may offer slightly more conductor area for a given wire diameter, but over time, the solid wire can work-harden, become brittle and eventually break from repeated back-and-forth motion. Imagine using large-gauge solid copper wires in the wire boot in a door jamb or to your trunk or hatch lid. That is a recipe for disaster.

The term OFC (oxygen-free copper) has become abused and is used synonymously with solid or all-copper conductors. In actuality, OFC is a type of solid copper. When molten copper is cast and drawn into a conductor, the process to make an OFC conductor reduces the oxygen content of the wire. If all is done perfectly, the copper-oxygen content is around 42 parts per million (PPM) vs. a conventional copper with content that is roughly six times that amount.

In the mobile electronics industry, there is no way to know if the solid copper conductor you are purchasing is oxygen-free or not unless you can witness the casting process in person. Everyone in the industry uses “OFC” for a piece of wire that is not copper-clad aluminum (CCA).

Looking at the alternative, we have CCA conductors. In these conductors, the core of the wire is a cylinder of aluminum and around it is a layer of copper. From the side, it looks like copper, but if you cut off a piece and look at the end, you can see the gray aluminum content.

There are further variations. Some companies manufacture all-copper strand wire but coat the outside of each strand with a thin layer of tin to help prevent corrosion.

Car Audio Power Wire: Size

When it comes to flowing electricity, or, more specifically, flowing electrons, the most important thing to consider is wire size. In the mobile electronics industry, we use the American Wire Gauge (AWG) standard. This sets a specific diameter for a conductor. It’s not a debatable number – the conductor either meets the standard or it doesn’t.

Here is where the games begin. There is a second term used in our industry: gauge. In the steel sheet industry, gauge is an important tool for specifying material thickness. In car audio, it means nothing. If you have been around the industry for any amount of time, you will have seen wires that claim to be 0 gauge but have a conductor area equivalent to a 6 AWG. If a wire is labeled as 4 gauge, then sadly, you have no way of knowing how big it is, other than attempting to measure it.

Cutting a wire and looking at the area also doesn’t always tell the story. Some wires are wound quite loosely. This makes the wire very flexible, but does so because there is space around the strands. You sacrifice effective cross-sectional conductor area for flexibility.

Car Audio Power Wire: Materials

In solid copper stranded wire, we ideally want everything to be pure copper. That said, pure copper is quite expensive, even though the cost of pure copper has come down over the past few years; it currently sits at around $2.00–$2.25 a pound on the commodities market. When a manufacturer wants to purchase wire, there are many options: strand count, how the strands and bundles are woven, how tightly they are woven, and so on. Manufacturers also have a choice in the “kind” of copper they make the conductors with. It could be pure copper, recycled copper or a copper alloy. Again, you have no way of knowing unless you are witness to the process.

Don’t let the variations in copper scare you. A solid copper conductor always outperforms a CCA conductor. The biggest challenge with car audio CCA wiring is that it does not, and will not, specify the ratio of copper to aluminum. There are publically displayed measurements of different CCA wire samples where a smaller-diameter wire outperforms a slightly larger wire because it has less aluminum and more copper. Unless you measure it yourself, you just don’t know.

On its own, pound for pound, aluminum has about 60% more resistance to the flow of electricity. When we talk about CCA wire, there is some copper in there; in most cases, the difference diminishes to 30 to 40%.

Car Audio Power Wire: The Challenge

When you look at car audio wiring, there is no way to know what you are getting with a CCA amp kit or roll of wire. Some manufacturers make CCA wire that functions nearly as well as solid copper. In fact, one company makes an oversized CCA that has less resistance per foot than solid copper. The downside is that the wire doesn’t fit into a lot of connectors or terminal blocks. Overall, unless you want to take the time to measure the properties of the kit you are buying, it is better to stick to solid copper.

From the standpoint of long-term benefits, solid copper wire resists corrosion much better than CCA wiring. In climates where road salt or brine is used in the winter to keep surfaces clear of ice, we have seen instances where unprotected CCA power wires have failed completely in less than two years. Why risk the performance of your audio system, when you can simply choose the solid copper wire?

How do you know if you are getting something good? The Consumer Technology Association (formerly the Consumer Electronics Association) has developed a standard for wiring. It is called CTA-2015 (formerly CEA-2015) specification. It describes the minimum standards for wiring for use in mobile electronics applications. The standards include that the wire must be stranded solid copper, the minimum number of strands for a given AWG wire size, and the area of the wire and its maximum resistance. If you stick to the brands that support the CTA-2015 standard, you should never have any problems.

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.

For decades, car audio enthusiasts have been fiddling around with the gain control on their amplifiers in hopes of “getting more out of them.” Many professional installers have scientific, repeatable processes in place to ensure these controls are set to provide the maximum performance and reliability from your audio system. Let’s look at the most misunderstood, and most often adjusted, control on car audio amplifiers – the gain control.

What Is a Gain Control?

When a manufacturer decides to develop an amplifier, they need to decide how many channels it will have, how much power it will produce, what additional features it will include and what source units it will work with. Because modern source units have maximum preamp output voltages that range from 1.7 to 5 volts, amplifiers have to be adjustable to make their full rated power when driven with these signals.

Let’s make up an example: Imagine a 100 watt mono amplifier that was designed to produce full power (100 watts) when it receives 2 volts of audio signal. This is a reasonable amount of signal gain, but leaves us open to two significant problems. What if we want to use this amplifier with a source unit that can only produce 1.7 volts? We can’t get the amplifier to full power even with the volume control on our radio turned all the way up. In fact, we only get 72.25 watts out of our amplifier. On the flip side, if we have a source unit that can put out 4 volts of signal, then the amp would attempt to make 400 watts with our fixed gain setting. Since the power supply of the amp was only designed to provide enough voltage to produce 100 watts, the signal would be severely clipped and distorted, and there is a great chance that the amplifier and your speakers might be damaged.

The Solution

For a single amplifier to work with multiple sources, amplifier manufacturers have to make the input signal level adjustable. We call this the gain or sensitivity control. It doesn’t adjust how much power the amplifier will make, but it does adjust how much of the input signal the amp uses to make full power.

There is a secondary reason for adjustability: Not every speaker has the same sensitivity. This means that sometimes you have more power than you need. Let’s say your front speakers produce 90 dB of output from 1 watt of power, but your rear speakers are much larger and produce 93 dB of output from the same 1 watt of power. For them to appear to be of equal loudness at the listening position, we only need half the power to the rear speakers. We turn down the sensitivity of the rear channels of an amplifier to balance these out.

Making Gains (Using Your Gain Control!)

Your installer may use one of many different processes to adjust the gain controls of your amplifier. We want the gain controls to be as low as possible, but still allow you to get full power from the amplifier. Why do we want the gain low? That is, perhaps, the fundamental key to this article.

We want the amplifier to accept an input signal with as much voltage as possible for it to produce full power. Having more voltage on your interconnect cables helps drown out noise. Less amplifier sensitivity (lower gain setting) also helps to reduce noise. When the amplifier gains are set properly, you get full power from your amp without unnecessary hiss or background noise.

There are four common methods for adjusting gain controls: by ear, with a small amplified speaker, with an oscilloscope or with a distortion detection device. Setting by ear with music is very difficult and can lead to inconsistent settings. That being said, if your installer uses a test tone, the “by ear” process can work quite reliably. Using a small amplified speaker is similar to that process – there is a test tone, but the small speaker allows your installer to check the preamp signal from the source unit, and in and out of any signal processors.

Using an oscilloscope to set an amplifier’s gain control is one of the best ways to get an accurate reading. Oscilloscopes work for any frequency, so they are very flexible. Your installer can see exactly when the amp has reached its peak voltage.

Finally, companies like D’Amore Engineering and SMD have developed products designed specifically for mobile electronic installers to check for signal distortion on preamp or speaker signals. All you have to do is plug the device in and turn it up until the red Distortion LED comes on. Bam – done! A word of warning on these devices, though: They are very accurate and can detect distortions other than signal clipping. Many product design problems have been found when attempting to set gains with these.

How Can You Check Your Gains?

If the sensitivity controls on your amplifiers are set properly, you should be able to get your amplifiers to distort a little bit with the source unit at full volume. If you are wondering why a properly set amplifier will distort, that’s a great question. It’s called gain overlap. We want to have a little extra sensitivity in case we are playing a song that is recorded quietly. A great example of this is the well-known “Brothers In Arms” album by Dire Straits. It needs a good 5 extra dB of gain to get rocking. In fact, the original 1985 release from Warner Brothers Records had several songs where the loudest part of the song was below -5 dB. “Why Worry” has a peak level of -13.27 dB. A nightmare for an installer trying to set gains, but, luckily, that’s not a song most people rock out to.

If you can’t turn your volume control past halfway without your amplifiers running out of power (distorting), then it’s time to visit your local mobile electronics specialist. Likewise, if you hear a significant amount of hiss at low volume levels, then you likely need an adjustment.

Properly set gain controls won’t make your system quieter, and turning up the sensitivity doesn’t make your amplifier more power. Gain controls exist to ensure that your system is always working the best it can. Please leave them alone, or ask your installer about how they are set.

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