Speaker power ratings are among the most confusing and misleading specifications in the car audio industry. Once you understand the process used to test speaker power handling, you’ll quickly realize that the information doesn’t always translate to information that’s helpful in choosing an amplifier. Let’s dive into the science behind choosing the best amplifier for your car audio speakers.

Speaker Power Ratings

In short, how much power a speaker can handle depends primarily on the voice coil’s diameter, length and number of windings. Secondary considerations include the gap size between the voice coil assembly, the T-yoke and the top plate of the speaker. Closer proximity helps to improve heat transfer away from the voice coil. A smaller gap also increases efficiency. However, if the gap is too tight, the voice coil or former might rub, which can cause damage and distortion. Cooling technologies like a vented pole piece, vents under the spider mounting plateau and vents in the former also help by allowing air to flow around the voice coil.

The type of enclosure used also plays a role in power handling. A sealed enclosure will trap heat around the motor assembly. A vented enclosure will allow heat to escape as air is exchanged through the vent resonance.

Audio Content and Power Handling

A concept that many audio enthusiasts don’t fully understand is how power is distributed based on audio frequencies. Looking at frequency response measurements on a real-time audio analyzer can exacerbate this misunderstanding.

When we look at acoustic audio measurements of pink noise on an RTA, we expect to see a flat line. This indicates that the amplitude of each frequency is equal. While we might want to bump up the bass to make the system fun or attenuate the high-frequency content by a few decibels, we don’t want peaks or dips anywhere in the graph.

Where the confusion lies is that the technician working on calibrating your car audio system is using pink noise with the RTA. Pink noise contains random frequencies with the same energy per octave. Put another way, dividing power by the range of frequencies in a given octave will give you the same energy per hertz.

Between 10 and 100 hertz, there are 90 1-hertz frequency bands. We have 900 bands between 100 and 1,000, and 9,000 between 1,000 and 10,000. Let’s say we have 10 watts of power to distribute among those frequencies. In our lowest octave, each one of the hertz bands gets an average of 111.11 milliwatts. In the band between 100 and 1000 hertz, each hertz band receives an average of 11.11 milliwatts. In the top octave between 1,000 and 10,000, each hertz band receives an average of 1.11 milliwatts. Once again, this power distribution produces a flat line on an RTA graph.

The bottom line is that pink noise matches how humans perceive sound. Our ears perceive pink as having the same volume level at all frequencies. Pink noise is occasionally called an equal intensity curve.

What Does Frequency Have To Do with Speaker Power Ratings?

Imagine, if you will, a typical mid-level car audio subwoofer. It might have a 2-inch diameter voice coil former with a winding that’s 1.5 inches tall with a two-layer winding. Rated power handling might be around 400 watts. Now, let’s consider the voice coil in a typical 6×9 speaker. The diameter might be 1 inch, and the winding might be 0.75 inch tall. The speaker might have a continuous power rating of 100 watts. All of this makes sense so far. Less mass in the winding means it can handle less heat.

Now, let’s think about a tweeter. It likely has a voice coil diameter of 1 inch, but the winding might be 0.2 inch tall, and it will surely have no more than one layer. Worse, the wire will be tiny in diameter. Even then, many stand-alone tweeters have a power rating of 100 watts. How can this small voice coil dissipate 100 watts of heat? What about P.A.-style speakers? They often have extremely short voice coils. Yet some claim to have 200-, 300- and even 500-watt power handling ratings. How is this possible?

If the power handling test uses filtered pink noise, then the speaker is tested with less power. A tweeter can’t reproduce bass or midrange frequencies. So, to test their power handling, the noise waveform would be filtered at something like 3,000 or 4,000 hertz. Filtering the bass information removes significant energy from the signal.

If we apply a 2-kHz high-pass filter to a 100-watt equivalent pink noise signal, the result would be only 1 watt of power going to the speaker. Midrange and high-frequency speaker power ratings are almost always quantified this way. So, your 100-watt tweeter can only handle 1 watt of power. Your P.A.-style midrange likely can’t play much below 300 Hz. It might only get 3 or 4 watts of power if appropriately filtered from our 100-watt example.

Matching Amplifiers to Speakers

Now that we’ve set the stage for understanding power ratings, we can finally talk about matching amplifiers to speakers. How powerful of an amplifier do you need for your speakers? The answer starts with the frequency range in which you’ll operate the speakers. With subwoofers, you’ll be playing bass frequencies, so almost all the energy in the music will arrive at the speaker. You’re in the same boat if you have a system with 6.5-inch or 6×9-inch speakers and no subwoofer. You’ll be sending bass information to the speakers. If your system has a subwoofer, you’ll likely only send frequencies at 80 hertz and above to the speakers. That’s about 1/10 of the maximum power compared with a full-range signal. In theory, you only need 1/10 the power to your mids as your subwoofers need. So, 500 watts to a sub and 50 watts to the mids. If you have actively filtered tweeters, they likely only need a few watts.

A lot of this is theoretical rather than practical. So, let’s look at midrange speaker power handling another way. Let’s say you have a set of mid-quality component 6.5-inch speakers. They have a 100-watt power rating, and the woofer might have an Xmax specification of 4 mm. Let’s examine how much the woofer cone moves with 100 watts of power.

The graph above shows us that the driver reaches its 4 mm excursion limit at 110 hertz. If you play music with content lower than 110 hertz, the driver might bottom out or, at the very least, sound terrible. If you want your audio system to sound terrible, driving midrange speakers beyond their excursion capabilities is a great way to do it. With an 80 hertz crossover point, 100 watts is too much power. As it turns out, 50 watts at 80 hertz results in a cone excursion of 4 mm.

What About Time?

If you look at speaker specifications, you’ll see both continuous and peak power ratings. Some companies incorrectly refer to the continuous rating as an RMS rating. Using the term RMS implies that the power measurement was done with an RMS current or voltage measurement, not the waveform’s peak values. RMS refers to the amplitude in an AC waveform that can do the same work as an equivalent DC voltage.

Companies with genuine engineering specifications for their speakers will test them at their continuously rated power level for eight to sometimes over 100 hours. The speaker needs to continue to function after the test, and the Thiele/Small parameters should typically remain within 10% of where they were when the trial started. In other words, the voice coil can’t overheat or fail, and the suspension can’t stretch significantly.

With all that said, speakers can handle momentary bursts of additional power beyond their ratings. The problem is that how long these bursts can last without causing damage is difficult to quantify. Let’s say you’re listening to a song with a vocalist and someone playing a guitar. In the middle, there’s a drum break like Phil Collins’ solo from “In the Air Tonight.” If you cranked up the volume during that solo, even at twice a speaker’s continuous rated power level, that’s not enough energy to overheat the voice coil. So as long as the speaker isn’t physically damaged, everything should be fine.

Picking the Best Amplifier for Your Car Audio Speakers

So, after all this science and confusion, how do you pick the best amplifier for your car audio speakers? You choose the amp that sounds the best. Whether an amp makes 45, 50 or 60 watts doesn’t matter, as that’s only a difference of 0.5 or 0.8 dB in maximum output. Is a 100-watt amplifier “better” than a 50-watt amplifier? It is if it adds less noise and distortion to the signal that passes through it.

Think about how reputable companies group the amplifier series they offer. ARC Audio has the ARC Series. Rockford Fosgate has the Power Series. Kicker has the IQ Series. Audison has the Thesis Series. Hertz has Mille. Sony has the Mobile ES line. Aside from some additional features, these higher-end amplifiers sound better than the lower models. They have better signal-to-noise ratio specifications and lower total harmonic distortion numbers. When playing the same music through the same speakers at the same volume, the sound produced by higher-quality amplifiers is more precise and accurate.

How Much Power Do My Speakers Need?

We can’t count the number of times we’ve seen posts on social media asking, “What amp is best for my speakers?” The poster will then list the speaker’s continuous power-handling capabilities, even if the number is irrelevant. You already know your tweeters will never need more than a few watts, so why would you use a 100-watt amp to drive them? Well, if the amp you have in mind has excellent distortion and noise measurements at those low levels, your tweeters will sound better.

What about everyday systems? How much power does a set of coaxial speakers need? If they are an entry-level speaker, 45 to 65 watts is likely more than enough power to drive them to their limits at lower frequencies. If you have a set of mid-priced component speakers, say in the $200 to $400 range, 75 to 100 watts is adequate. If you’ve purchased high-end speakers like the ARC Audio RS, Audison Thesis, Rockford T3 or T4, Hertz Mille, Kicker QS or Sony Mobile ES, an amplifier that produces 100 to 150 watts is a good power range.

Now, is buying an expensive, high-power amplifier a waste of money when using inexpensive speakers? Maybe. Speakers are almost always the weakest link in any audio system when it comes to how much distortion they add to the audio signal. Look at our articles on Understanding Speaker Quality, and you’ll see what we mean. You are better off buying speakers with distortion-reducing technologies like shorting rings, copper T-yoke caps or even more excursion capability and pairing them with a less exotic amplifier. The net result will be much better sound. If you have a great-sounding amplifier already, use it. Just ensure that the technician configuring and calibrating the system confirms that you can crank the volume without worrying about damaging anything. This doesn’t mean setting gains with a scope or distortion tester.

If you need help picking the best amplifier or speakers to upgrade your car audio system, drop by a local specialty mobile enhancement retailer. Please bring your favorite music and listen to it on their display at the same volume levels you would in your car or truck. This demonstration will give the product specialist an idea of the performance level you need regarding speakers and how powerful the amplifiers should be. Don’t hold back. If you like it loud, crank it. Hoping inadequate products will play louder only leads to disappointment, frustration and damaged equipment.

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.

Understanding how DC voltages combine is quite simple. They add linearly when we wire sources (like batteries) in series. But understanding how AC waveforms combine is complicated. AC waveforms don’t always sum intuitively unless the peaks and valleys align perfectly. We see many examples where people interchange the terms phase and polarity. In reality, those words refer to different phenomena. Let’s take a close look at audio signal phase and polarity.

What Is an Audio Signal?

We will use sinusoidal waveforms as our example audio signals for this discussion. The sounds we hear are combinations of infinite frequencies. If you look at music on an oscilloscope, it would look something like the image below.

Phase refers to the relative timing between two waveforms. A single waveform can’t have phase. We must compare it to something else. As such, we will need a reference waveform for this article. You can see that below.

In Phase

Let’s start with the basics. When we describe two waveforms as “in phase,” we infer that the peaks and valleys are aligned. The starting point of the waveform should happen at the same time. The image below shows two waveforms that are “in phase.”

Two waveforms can be in phase but have amplitude differences. The image below shows that condition.

Reverse Polarity

We often see amplifiers with a switch that inverts the polarity of the audio signal passing through the devices. These are frequently and mistakenly labeled as “Phase.” Let’s explain.

Phase is the relative difference in time between two signals. Polarity refers to a contrasting waveform direction. In DC voltage, we know that if we put a voltmeter’s red lead on the battery’s positive terminal and the black lead on the negative, we’ll see a positive voltage. If we reverse the leads and put the red on the negative and the black on the positive, we’ll see a negative voltage on a digital meter. We call this “having the polarity backward.”

In AC waveforms, the same thing happens. If two waveforms have the same starting point and frequency, but one goes positive while the other goes negative, we refer to one being in the opposite polarity.

Here are our two audio waveforms in the oscilloscope. The yellow waveform shows the opposite polarity to that of the pink waveform.

Let’s consider this in terms of audio signals going to a speaker. The speaker driven by the yellow signal would start from rest and then initially move forward. The speaker driven by the pink signal would begin moving toward the magnet. Activating the polarity switch has the same effect as swapping the positive and negative speaker wire connections.

The image above shows an amplifier with the polarity control switch incorrectly labeled as phase. As a bonus, the amplifier has the infrasonic filter incorrectly marked Subsonic. This labeling is a fairly common error.

If the above amplifier could adjust the phase of the waveform of one signal to be 180 degrees from the other, the waveforms would look like this:

Phase

When discussing phase shifts, we include the peaks and valleys of a waveform and the starting time. This start point or time is especially crucial to discussing audio signals that are never sinusoidal. The starting times move if there is a phase shift between two signals. A difference in starting times implies a delay on one of the waveforms. We can do this quickly and deliberately with a digital signal processor. Adding delay is common in “time aligning” the left and right speakers in a stereo sound system.

Electricians and engineers often discuss phase angles in AC waveforms when driving inductive loads like electric motors. In these cases, the voltage applied to the motor may be “out of phase” with the current flowing through the motor. The same thing happens with speakers and passive crossover networks. These components’ reactive elements (inductance and capacitance) can cause the current and voltage to be “out of phase.” These phase shifts are why we can’t accurately measure the power out of an amplifier without measuring the phase angle between current and voltage. If that was confusing, don’t fret. These are concepts that engineers and technicians learn about in college.

We often talk about degrees when discussing the phase of two waveforms. One complete cycle of a sine wave is 360 degrees. A half-cycle is 180 degrees. The angle between when the sine wave initially moves from the 0-volt level to the first peak is 90 degrees. Here is an example of a 90-degree phase shift between two waveforms:

Here’s what a 360-degree phase shift looks like:

Understanding that the starting points of the two waveforms differ is crucial to understanding how audio waveforms interact. If you were to “align” the peaks and valleys, we’d think these signals were “in phase.” This practice is a common issue for people who try to “time align” a subwoofer to a midbass speaker by reversing the polarity of one relative to the other, then adjusting a delay so there is a sizeable acoustic null shown on an RTA at the crossover frequency. This practice doesn’t consider the start time of the audio signals. Considering phase and start time is required to deliver a system where the sound from the subwoofer arrives at the listening position at the same time as the midbass or midrange drivers.

Why Are Signal Phase and Polarity Important in Car Audio Systems?

For people who’ve upgraded their vehicles with a radio and two pair of speakers, the primary concern is that the acoustic polarity of all the speakers in the system is the same. When a speaker produces a sound, it should combine acoustically with the output of other speakers in the system. You can test this quickly without any special tools.

Play music with a reasonable amount of bass. Use the balance and fader controls to listen to one speaker at a time. Start with the front left speaker. Now, use the balance to add the front right speaker so both play. The relative bass level should increase when you add the second speaker. If it doesn’t, the wiring likely has a polarity problem.

Repeat the test with the front left speaker playing, then use the fader control to add the left rear speaker. Once again, the bass should increase. If it does, fade rearward so only the left-rear speaker is playing. Now, use the balance control to add the right-rear speaker. Once again, we should get more bass. If the bass decreases when a second speaker plays, drop by a local specialty mobile enhancement retailer and ask them to check the wiring.

Signal Summing and Cancellation

Sometimes the distances between speakers cause incomplete summing of all frequencies. Let’s go back to looking at audio waveforms for a second. If two signals have the same amplitude, frequency content and starting times, how they add together (called summing) is predictable.

In the example above, the signals combine perfectly and the total amplitude doubles. An example would be listening to music at home on a set of stereo speakers and sitting equidistant from both speakers. When it’s perfect, or at least pretty good, signals that are equal in amplitude and in phase in both channels appear to come from a spot between the speakers. Some call this a “phantom center image.” It’s just how stereo recording works and doesn’t require a fancy name.

Let’s look at how signals with different phase relationships sum. First, let’s look at two waveforms, of which one is reversed in polarity.

If one speaker moves outward and the other moves inward, their output should cancel. If you have a pair of subwoofers and one is wired backward, the result is that the system produces no bass.

Now, what happens when one signal arrives before another? Here are the waveforms if one signal lagged the other by 90 degrees:

A few bad things happen when signals don’t start simultaneously. In terms of the combined waveforms, the amplitude only increases by 25% instead of 100%. This happens if you’re at home, sitting much closer to one stereo speaker than the other. The same happens in your car, where the left door speaker is maybe 30 inches away, and the right is 45 inches away.

Audio Content and Signal Summing

Remember, music is not a single frequency, as we’ve shown in the oscilloscope plots. It’s a combination of thousands of frequencies. We don’t talk about the phase between audio signals; we talk about delay times. Every frequency has a different wavelength. So if there is a timing difference between the signals, some frequencies combine, and others cancel. The result produces an uneven frequency response.

Let’s combine some signals in Adobe Audition to demonstrate this.

I created two sine sweeps. These are 20-second tracks that start at 20 Hz and increase in frequency by 20 kHz. The image below shows that the left and right channels are equal in amplitude and phase.

Let’s look at the frequency response of the two signals:

The frequency response graph isn’t perfectly flat because I don’t have control over the speed at which the sweep occurs. In short, the bass is present for longer than the highs. What you need to know is that the response is flat and smooth.

Let’s add about 1.7 milliseconds of delay to the right channel waveform. This is the time it takes sound to travel about 24 inches. That’s not an uncommon path length difference between a vehicle’s left and right speakers.

As you can see, the results are dramatic. The frequency response is full of dips. We call this comb filtering, which happens when two sound sources are at different distances from the listening or measuring position. This could be attributed to path length differences between speakers or to a speaker’s sound reflecting off a surface and recombining with the direct-path sound. If there are path length differences, your installer can use a digital signal processor to delay the signal to the closer speaker. There isn’t much you can do about reflections other than to relocate the speakers.

Audio Signal Phase and Polarity

We hope this serves as a primer to help you understand the concepts of audio signal phase and polarity. A solid understanding of these concepts is crucial to designing and calibrating car audio systems that sound amazing. If you want better sound from your car audio system, drop by a local specialty mobile enhancement retailer today. Ask to hear one of their demo vehicles and discuss the options for improving your vehicle’s stereo.

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.

There will always be debates about whether a Class D amplifier can sound as good as a Class AB. We’ve proven that it all depends on who designs the amplifier. We also see mumblings about single-ended amps versus full-bridge designs. STMicroelectronics created a different and unexpected way to deliver efficient power. These amplifiers are categorized Class SB and Class SBI. Let’s see if we can explain how they work.

Amplifier Output Device Topologies

Before we dig into the Class SB amplifier, we need to clarify a few things. There are two ways to configure the driver devices (transistors or MOSFETS) to provide large amounts of current to speakers. In a Class A configuration, the output device is halfway on when no audio is playing. The audio signal can modulate the output from its resting point at 50% up to nearly 100% and down to nearly 0%. While extremely linear, these designs are notoriously inefficient and waste large amounts of energy. A Class A amplifier circuit has a maximum efficiency of 25% and only gets worse as the output level decreases from full. Playing nothing at all, a true Class A amplifier wastes half the maximum current delivery capability of the power supply as heat.

In a Class B amplifier, we dedicate one transistor to the positive half of the audio waveform and a second to the negative half. When no music is playing, the output devices are functionally off.

We have to add a little voltage to the output devices so that the transition from one to the other is smooth. Many low-quality amplifiers don’t do this well. In those instances, we’ve seen a little step in the waveform through the transition. This step is called crossover distortion. When executed smoothly, there is no step. This mode of operation is commonly called a Class AB output device configuration. In reality, it’s just a voltage-biased Class B, but there’s clearly no going back on the name now.

In this context, Class A and Class B are called output device topologies. These are the only two ways output devices can be wired.

Class D Operation

When we talk about Class D amplifiers, we aren’t discussing how the output devices are configured. Class D is a description of the signal used to drive the output devices. Most Class D amplifiers use a Class AB output device configuration.

As you can see in the image below, the analog audio waveform is chopped into little pieces by the Class D driver IC. The width of the spikes relative to the switching frequency represents the output level. Very narrow spikes produce small amounts of output, and very wide spikes produce high output levels.

The benefit of Class D operation is that the output devices are switched fully on or off. They spend very little time part-way on. In essence, they block all current flow or allow it all through. Transistors and MOSFET devices are least efficient when they allow half the current through – as we saw in a Class A configuration. The result is a dramatic improvement in efficiency. Many well-engineered Class D amplifiers have 92% total efficiency.

Amplifier Integrated Circuits

Before we go off the rails with Class SB, let’s look at dedicated amplifier ICs. Almost every car radio for the last few decades uses a single chip as an amplifier. These chips typically have four channels of amplification and all the protection circuitry required to prevent DIYers from blowing up their radios. Most IC amplifiers provide 16 to 21 watts of power into 4-ohm loads from each channel.

Sony took things further with their High Power radios and used a Texas Instruments IC called the TAS5414C. In their head unit applications, I’ve measured over 42 watts of power from each channel. In what Sony calls their Subwoofer Direct mode, I’ve measured over 76 watts of power.

Bridge-Tied-Load Amplifiers

Most head units have only the vehicle’s battery voltage available to drive the speakers. They don’t typically have step-up power supplies like an amplifier. As such, we can only get about 13.5 or 14 volts across the speaker terminals. This means we are limited to a theoretical maximum of 24.5 watts. In reality, we see a few watts less as some voltage is wasted in the amplifier circuitry.

If your installer were to look at the output of a car radio at full power on an oscilloscope, you’d see the following waveforms on the speaker wires.

If we turn the volume down to almost nothing, we’d see the following:

As you can see, both speaker wires have a DC offset voltage. When no audio is playing, there’s about 6 to 6.5 volts present on the speaker wires. Because the voltage is common to both wires, the speaker doesn’t move. The speaker only responds to differences between the wires.

Now, let’s look at a single-ended amplifier. This would be an example of one channel of a typical car audio amplifier.

You can see that the probe with the green trace rests at the ground voltage. The probe with the purple trace shows the output voltage swinging from positive to negative and back. The speaker will move forward and rearward to follow the purple waveform.

STMicroelectronics’ Class SB and Class SBI

OK, now you should understand how output-switching devices can be configured and how different waveforms can be used to increase efficiency. STMicroelectronics combined things in the Class SB amplifiers to create something unique and, to put it mildly, creative. STMicroelectronics uses the SB abbreviation for “single-ended bridged” and SBI as “SB improved.” Their claim is Class AB sound quality with a 50% improvement in efficiency.

Class SB amplifiers function as single-ended amplifiers at low to moderate power levels. The waveform sweeps from negative to positive on one speaker lead while the other rests at ground.

If this were a typical Class AB amplifier, the output waveform would clip if we increased the signal to the amplifier such that the output tried to exceed the rail voltage limits. That would resemble the waveform below.

The Class SB amplifier gets creative when it runs out of rail voltage at high output levels. Check this out.

When the main output reaches clipping, the “normally at ground” output increases voltage in the opposite direction. The result is a net increase in amplitude. That’s very creative. In application, having full control over every component in the amplifier is crucial to this design functioning properly. With everything housed in a single IC, STM has that control.

Is a Class SB Amplifier Better?

Before claiming that STMicroelectronics has reinvented the car audio amplifier, consider that these ICs have specific applications. They are designed to be compact and efficient. As such, amplifiers created with them at the core can be compact with smaller heatsinks than an equivalent Class AB amplifier. Further, these amplifiers are available with digital inputs. This connectivity makes them ideal for integrating into a closed-network infotainment system, like MOST, AVB or A2B offer. The amplifiers produce about 45 watts of power. This might be good for the main channels of a factory-installed car audio system but won’t be adequate for high-power aftermarket solutions or subwoofers.

What Do Consumers Need to Know about Class SB Amplifiers?

So, what does the typical consumer need to know about Class SB amplifiers? The answer is not much. That said, knowing what new technologies are in use is always good. Class SB amplifiers are primarily a solution for vehicle manufacturers and low- to mid-power audio systems. Installers and technicians must understand how to recognize audio signals from a Class SB amplifier when integrating digital signal processors and new amplifiers into a vehicle. Many, but not all, aftermarket audio upgrades work with Class SB-powered factory-installed source units. If you want more performance from your car audio system, drop by a local specialty mobile enhancement retailer and ask them about the amplifiers, speakers, source units and subwoofers that will deliver the sound 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.