Posted on

Why You Shouldn’t Use Delay to Adjust the Phase of Crossovers – FB Post Archive

Written by Andy Wehmeyer – President, Audiofrog, Inc.

Originally posted to Facebook on Andy’s feed here

Andy’s tech [tip] for the day. This is an explanation of why you shouldn’t use delay to adjust the phase of crossovers in a car audio system that uses delays to fix distance for an offset listener.

The phase at the crossover is important for only one reason–so that the acoustic sum doesn’t include a big null in the frequency response. We don’t hear absolute phase, we hear relative phase. A crossover is designed to separate the total band of reproduced frequencies into sub bands so that the drivers for those bands can be optimized for use in that band. The region where the two drivers interact is the only place that the relative phase of the drivers matters because AC signals sum according to magnitude AND phase.

The HUGE misunderstanding EVERYWHERE in car audio is about what matters in designing a crossover. There are two things that matter most and the rest are all small considerations. One, in a passive network is impedance. We have to keep the system impedance above the minimum limit the amp can drive. Second, we listen to speakers, not to electrical filters. The electrical filters simply change the amount of power that is delivered to each of the speakers. A high pass filter (whether passive or active) reduces the power delivered to the speaker at low frequencies. A low pass filter reduces the amount of power delivered at high frequencies. The slope of the crossover determines the rate at which power is attenuated.

Crossovers do this by changing the phase of either the current or the voltage in the AC signal by a time constant. A capacitor stores energy as voltage, so the rise in voltage available at the cap is delayed compared to the current that’s applied. When current flows to the cap, it takes a little while for the cap to charge to the same voltage that’s applied. That lag changes the phase by 45 degrees and the resulting power over the frequencies where the capacitor “works” (low frequencies) is reduced because power is voltage times current times the cosine of the phase angle of 45 degrees.

An inductor (low pass filter) is similar but opposite. When AC voltage is applied to an inductor, the current lags by 45 degrees. Same thing, but high frequencies are reduced.

So, a capacitor “resists” changes in voltage and an inductor “resists” changes in current.

Active filters do the same thing. Digital filters don’t have to be designed this way, but many are. Why? Because it works just fine.

The low frequency roll off of a speaker is a crossover that’s built into the speaker. The spider and surround are a mechanical analog of a capacitor. The high frequency roll off of a speaker is also a crossover built into the speaker. That rolloff is controlled by two things– the mass of the moving assembly (mechanical analog of an inductor) and the inductance of the voice coil at high frequencies.

So, the electrical phase of the speaker changes due to these two built-in crossovers. Anytime and anywhere the frequency response changes direction, the phase also changes direction. This applies to the acoustic output of the speaker, too. When you put a speaker in a room and measure the response, frequency response changes as a result of reflections also changes phase.

When we talk about minimum phase systems, we are talking about a system in which the phase is directly related to the frequency response. That system only exists as a single driver in a nonreflective environment. A second driver, which has a phase response of its own, combined with the first driver is NOT a minimum phase system. A single speaker in a room is not a minimum phase system because each reflection changes the measured acoustic phase independent of the response of the speaker by itself.

Despite this condition, there are regions in a non-minimum phase system that behave like a minimum phase system. What defines that is if inverting the frequency response of the electrical signal sent to the speaker by the same magnitude as the dip or peak fixes the problem. In these regions, equalization is effective. In regions that don’t act like minimum phase, equalization won’t fix the problem. An acoustic null is a region in which boosting with an EQ doesn’t affect the problem by the same magnitude as the boost. We’ve all experienced that in trying to boost a deep and narrow dip in the response. When the relative ACOUSTIC phase between two drivers is 180 degrees, boosting doesn’t help. However, changing the polarity of one of the drivers changes the relative phase by 180 degrees and now we don’t have that dip anymore.

When we design a crossover, we try to remove power at high frequencies at the same rate as we remove power at low frequencies from the other driver. reducing the voltage by 6dB cuts power in half. If we have two speakers both playing the same frequency with both having their output level cut in half, then the resulting response is flat. Two halves make a whole.

OK…Now remember that we don’t hear the electrical filters. They exist only to modify the acoustic response, which is what we hear. What we hear is result of the application of the electrical filters. It’s just like EQ. We use an EQ to modify the acoustic response. This seems obvious to everyone. Crossovers are EXACTLY THE SAME. It doesn’t matter what the electrical response (magnitude or phase) is EXCEPT that is should combine with the speaker to produce a measured ACOUSTIC response that matches the Butterworth, Linkwitz-Riley, 2nd order, third order, or what have you. In many cases, additional EQ is required to make the acoustic output match one of those classical alignments. In a passive network, we can change the Q of the filters to boost or cut at the knee. In an active system, we can do the same thing by choosing a different filter shape or Q. Or we can use an equalizer to tweak the response so it matches.

All of these classical alignments assume (and require) that the sound from one speaker arrives at the same time as the sound of the other speaker because the summation of the low pass and high pass acoustic responses depend on magnitude and phase. Phase at the listening or measuring position is ALSO determined by distance. So, we can use delay to align the arrival of the sound from the low pass and the high pass output of the two speakers.

When we design a home audio speaker, we design ONE speaker in an anechoic room so its response is correct. Then, we REPLICATE it for the other speaker in a stereo pair. Then, we sit in between. We are the same distance from the speakers and the speakers are the same. If we sit a little closer to one than the other, then we can delay the ENTIRE speaker to correct for the distance. The responses of the speakers and all of the correction filters that are contained within each one are still the same.

In a car, this is not the case because we use asymmetrical filters to correct the responses of the left and right channels independently. If we set the delays of all the speakers individually so that the sound arrives at our ear at the same time, we do two things. We align the arrivals so we can design crossovers according to the classical alignments AND we correct for the offset listener.

In order to hear a correct image of a stage, the left and right acoustic signal has to arrive at our ears at the same time, in phase and precisely matched in level at all frequencies. So, let’s say that we delay the right mid in a three way to fix the crossover, what have we done to the relative phase between that mid and the one on the other side that has to be in phase with the first one in order to provide a correct image? We’ve changed it. Now, the image will shift over those frequencies toward the speaker from which the sound arrives first.

This is why using delay to fix phase at the crossover is fine for home audio speakers, but isn’t OK for car audio systems that use delay to optimize for a single offset listener. Get the crossover right using EQ and set the delays based on measured distance. It’s the only correct way.

In systems (like those with upmixers and center speakers), the use of delay to create an image is less important because sound is steered to a real center speaker and we don’t rely on the left and right being precisely matched in frequency and phase to create that center image.

So, when you look at the phase curve in the analyzer, try adjusting the EQ or the crossover to achieve flat frequency response and flat phase and leave the delay setting alone.

The reason that i recommend 4th order slopes in cars is because the rate of attenuation is so steep that the acoustic output of the speaker is much more likely to track the electrical response. And the electrical response of a 4th order LR filter sums flat in phase and magnitude. This minimizes the need for EQ. It’s much more predictable and much more likely to just work without a bunch of futzing around.

July 22, 2015 at 8:07 AM

Posted on Leave a comment

Case Study: Setting Crossovers

You’ll be reading about another example of client support. Shayne is a customer of mine from back when I was working at Easy Way Electronics of Langdon, ND. This is also a case study of how to use my forthcoming guide to setting crossovers on aftermarket head units and amplifiers.

Hello Barry,

I bought these from you years ago at Easy Way.  I ask cdt for help and they sent me to you. I’m hoping you can help me out.

I bought a CDT component system that I need some support in setting it up in a 2011 Tundra Crewmax.  Following is a list of my components:

  • MX-69S – Front doors
  • MX-2 Hybrid – Dash
  • MX-1i – Windshield corner (unless sail panels would be better)
  • MX-6X – Rear doors
  • MX-1000SX – Crossovers
  • Kenwood DDX-9705 – Head unit  (Has extra crossovers built in)  
  • Phoenix Gold Ti2 1600.5 – amp ( I have attached data below)
  • PG 10″ subs

Hopefully you can help,

You’ll be copying the installation I completed on this Tundra.

CDT Audio MX-Designs MX-3692 3-way 6x9 components
CDT Audio MX-Designs MX-3692 3-Way 6×9 Component Speakers
click or touch to view a Toyota Tundra installation

The rear doors were already upgraded and each of the 4 “corners” was powered by one amplifier channel.
The passive crossovers were placed under the driver’s seat.

I am wondering what settings to use on the crossovers, as well as the crossovers on the actual speakers themselves?

See the attached images for reference of MX-1000SX switch positions. You probably have the first generation of these crossovers which have some components moved. The switch functions didn’t change.

CDT Audio MX-1000SX 3-Way Flexible Passive Crossover

You will be bypassing the amplifier crossover settings. I’ll explain.

Kenwood DDX9705S Aftermarket Head Unit

The head unit will be set so the system type is 2-way. You need to see choices for Front, Rear, and Subwoofer when in the X’over menu.

Kenwood DDX9705S Xover speaker setup
Kenwood DDX9705S Xover speaker setup

What head unit and amp crossover settings should I use?

The head unit:

Kenwood DDX9705S DSP section setup
Kenwood DDX9705S DSP section setup

If you find the speakers are distorting as the volume is raised, or the doors will not stop vibrating, I recommend adjusting both the subwoofer LPF and FRONT speaker HPF crossover frequencies up by 10 Hz. Keep moving up until it sounds better. Better in my mind means the speakers are not popping when you have a loud song playing and you’ve got to shout for a passenger to understand you. These Tundras are known for noisy doors once good speakers are installed on high power, which you probably already know. I can help with door treatments if you need it.

If your doors don’t rattle or resonate you will then have the option of setting the Front HPF lower than the rear doors. As you have 6×9 midbass woofers that’s to be expected. We’re working with compromises and balancing the system.

The amplifier:

Phoenix Gold Ti2 1600.5 5-Channel Amplifier Input Controls

I’m not going to review everything on this panel. The highlighted controls are going to be discussed in the order of Front, Rear, then Sub.

Front HPF: This needs to be set lower than the frequency you choose in the Kenwood HU. There’s no need to use any special tools. As this knob has a range of 10 Hz to 4,000 Hz it will be difficult to be precise. Be sure the knob is turned fully to the left(counterclockwise) and then move it a small amount clockwise. If you do this with music playing, after setting the crossovers on the HU, you should just begin to notice low frequencies getting reduced as the front HPF knob is slowly turned clockwise. Back it off counterclockwise. 

The point of that is to have a backup HPF should your HU lose memory or be reset to factory defaults.

Rear:X-OVER switch is set to HP (high pass)This should disable the Rear LPF knob so disregard that. See Front HPF instructions above.As you do not need the same low frequency range out of the rear door speakers as the front doors you will not match the front HPF frequency. I see no need to drop below 80 Hz for the rear doors. As mentioned earlier, if you hear the speakers pop you need to move the rear HPF frequency higher. Do that on the HU.

Sub:L to R
Phase: Unlike the usual phase reversal setting, such as what your HU offers, this is fully adjustable between 0 and 180 degrees. To begin you will leave this knob turned fully counterclockwise to 0 degrees.

LPF: This is controlled mainly by the HU. To effectively bypass this setting turn the knob fully clockwise.

Subsonic: Two approaches.
1) If your subwoofers are sealed this will be turned fully counterclockwise to 10 Hz. 
2) If your subwoofers are ported you will multiply the port tuning frequency by 0.75. 

Example: 35 Hz port tuning frequency
35 x 0.75 = 26.25
Estimate the knob position to roughly 26 Hz. If you find the knob’s center position that is just to the left of center.

Can I tap into the factory speaker wire, or should I run new wire?  What gauge?

It’s so easy to run new wire all the way into the doors that you should do it. 16 AWG is sufficient. If you have 14 AWG that’s fine as well. No need to go larger, and don’t use 18 AWG or smaller.

Do I want to bridge the amp channels to the front stage and just use the rears as fill?  Or is that too much power?

No, do not bridge the amplifier. That is too much power.

If you made it to the end I applaud you. This is an intermediate or advanced topic of which most people will not even attempt. Leave your questions in the comments below.

Barry Schanz
Barry Schanz Enterprises, LLC
dba Rubyserv

Mastering Mobile Audio Systems Begins With a Plan

Posted on Leave a comment

Aiming Component Speakers

Aiming Component Speakers

What is a real example of consulting with me about mobile audio systems? I spoke with a gentleman today about his sound system plans regarding a set of Focal Utopia No.7 3-way components. He had questions.

Where to install and aim the mid-range speakers and tweeters?

Is my head unit, a Sony RSX-GS9 with time alignment, as good as a separate DSP?

My follow-up e-mail, with some emphasis for SEO and clarity in your browser:

If you can’t play it loudly, well-balanced, for a long time reliably then what’s the point?

Aiming Component Speakers

Mid and tweeter placement and aiming in a 3-way active configuration:
– When we have full control over the High Pass and Low Pass filters, meaning both the crossing frequency and the slope, there is no need to aim the midbass woofer or the mid-range speaker. Our guide is the following chart:

Loudspeaker Dispersion Chart

Looking at the half-circle diagrams from left to right, this is showing the relationship between the wavelength of sound at a particular frequency in relation to the diameter of the speaker cone.

Green = sound spreading as widely as possible away from the front and back of the cone

Red = sound spreading mostly straight away from the center of the cone. This would be like the laser pointer you mentioned as a tool for aiming A-pillar speaker pods.

Putting it another way

Green = if you were holding the 6.5″ woofer in your hand, while it was playing bandwidth limited pink noise centered at 500 Hz, it should sound very similar or the same whether it is aimed directly at your face or if it’s aimed off to one side of your head.

With 3-way components the ideal approach with our crossover filters is to focus on the left column of frequencies. Our crossover type to begin the DSP setup is Linkwitz-Riley, 4th order(LR4). There is no overlap or underlap. More on that later.

  • 6.5″ woofer Low Pass Filter frequency ~ 500 Hz.
  • 3″ mid-range speaker Low Pass Filter frequency = 1,130 Hz.

How does this look in the DSP?

  • 6.5″ LPF 500 Hz
  • 3″ HPF 500 Hz
  • 3″ LPF 1,130 Hz
  • 1″ tweeter HPF 1,130 Hz

What’s a problem with this? This plan violates the rule of keeping the system reliable. The tweeters will likely sound distorted and soon they will break before you hit volume 42 of 50. A better starting point:

  • 6.5″ HPF 100 Hz
  • 6.5″ LPF 500 Hz
  • 3″ HPF 500 Hz
  • 3″ LPF 3,000 Hz
  • 1″ tweeter HPF 3,000 Hz

Why do we start with a Linkwitz-Riley 4th order crossover, and the mating speakers have the same crossing frequency? Electrically this results in a flat transition between woofer and mid-range, and flat between mid-range and tweeter. Predictability is what we’re after.

Simulation of the electrical response of the LR4 crossover with a 3 kHz LPF and a 3 kHZ HPF

Does this look like a difficult approach to planning your component speaker installation? Aiming component speakers doesn’t need to be a mystery. Leave your questions in the comments below.

Barry Schanz
Barry Schanz Enterprises, LLC
dba Rubyserv

Mastering Mobile Audio Systems Begins With a Plan

Posted on Leave a comment

You Want Huge Power for SQ?

huge power for sq

Huge power wasn’t a “thing” when I got started with car audio in the mid-late 90s. My family owned a retail store in small town North Dakota and good sounding vehicles were a draw. The good equipment was expensive and people bought it eagerly. Beginning in 2002 I took a break from car audio and completely shifted my focus on cars to the performance scene. For the first time I tried drag racing and dyno testing. Audio equipment was a burden to slow me down. It had to go and I wasn’t working in the electronics business.

I re-entered the world of car audio in 2010 by going back to my family’s store, and it was really obvious that something had changed about amplifiers. Generally they were cheaper, more powerful for the same money, and everywhere I looked I saw 1,000W or more for bass. Turning to Google to learn, I discovered the online forum scene had changed and the “SQ” world of outboard digital signal processing with individual speaker equalization, Time Alignment and so much more was revealed. This technology wasn’t new in 2010, just new to me. It was new to me to have a separate amplifier channel for tweeters and mids, as before I had used passive crossovers for that.

The Internet SQ crowd was debating amplifier headroom to answer the question of “How much power is needed for the best sound quality?”. No longer was it cool to have 35W x 4. If you wanted a really “ballsy” SQ system it seemed like you ought to multiply that by 10-20X. POWER! DYNAMICS! MORE HEADROOM! If that snare drum snap doesn’t make me blink it must mean my amp is too small, right?

If we can’t turn it up loud and keep it there reliably then what is the point?

Aren’t there consequences to greatly exceeding the manufacturers’ power rating recommendations? Yes! Why do people disregard the power handling numbers in their quest for the ultimate in sound quality? As this is about potential damage to speakers versus amplifier power ratings, and not the evolution in audio equipment technology or price per watt of power I’ll next share an excerpt from an article I found.

Why Do Tweeters Blow When Amplifiers Distort?

Bigger Amplifiers
A persistent myth in the audio industry is that clipping damages tweeters, so you should use a bigger amp to ensure more headroom so the amp won’t clip. This claim is simply bollocks! Take the 100W amp described above, and replace with an amp big enough to prevent clipping … even with the additional 12dB input signal as shown in Figure 7. Since a 100W amp was just below clipping with an average output of 16W, if we add 12dB that takes the peak amp power to 1.6kW (near enough) and the average power will be 254W.

Rod Elliott, Elliott Sound Products

The article is required reading if you want to master mobile audio system design. Bookmark it to save it for later studying. Let’s focus on the last statement, “the average power will be 254W“. If we choose amplifiers large enough to meet a goal of never clipping on our music, a likely consequence is melted speakers. The electrical system burden increases. The complexity of the system installation increases rapidly.

Stop focusing on huge power for sound quality. There are many more important matters to focus our efforts on to increase our enjoyment of listening to music through our mobile audio systems.

See also:

Keep It Simple – Don’t Make This Mistake In Your Sound System

Power Ratings: What Do They Mean?

Barry Schanz
Rubyserv Mobile Audio Systems

Posted on

I Was Told I Don’t Need a DSP

need a dsp

I Was Told I Don’t Need a DSP

Since I’m Running Passive Crossovers

The right tools for the job.

One thing that affects us all at some time is advice that is misleading, completely wrong, or half correct. A recent conversation with a client brought up a question that can be translated to “Which tools does my mobile audio system need to meet my goals?”. No, this isn’t about tools to install your equipment. Let’s get into the tools used to adjust the electronic signals to suit our goal of good sound quality.

Do I need a DSP?

Students of Mobile Audio
Students of Mobile Audio
Join the Facebook Group NOW!

What are passive crossovers? What do these have in common with a Digital Signal Processor? 

Before I get into that let’s look at a list of some of our tools we need for good sound and reliability within an audio system:

Tools of the Trade (within a mobile audio system)

  • Filters – low pass, high pass, shelving, etc.
  • Level controls
  • Equalizer
  • Tone controls
  • Bass boost
  • Signal delay aka time alignment
  • Phase control
  • Polarity switching

The Basic Reason for Passive Crossovers

Diagram of speakers connected to an amplifier
Diagram of speakers connected to an amplifier

Small speakers need to have the bass signals removed to reduce distortion and to protect them from damage. Do we need a DSP to do this? Certainly it is a powerful option. Many DSP solutions have the entire list of audio tools in one package.

Back to passive crossovers. At the most basic level they will have a high pass filter to protect a small speaker, usually a tweeter. As designs of passives scale up in complexity we next add a level control to reduce the loudness of the tweeter, in relation to a mid-woofer. This is a valuable tool to have as there are so many places we can install tweeters into a vehicle that adjusting levels is a must. Also, it may be necessary to bring down the loudness of the tweeter in relation to the woofer, such as in a 2-way component speaker system. 

Four Divisions of Sound We Hear, Color Shaded

As frequency goes up we want no abrupt changes. The woofer and tweeter transition normally happens in the midrange. A passive crossover and a DSP both offer ways to ensure that transition happens smoothly. 

Dividing the Signal

Why would the signal need to be divided? What signal am I referring to? Let’s look closer at the diagram pictured above. In the upper left corner the boxes represent passive crossovers, with blue lines representing the speaker wires carrying the output signals from the amplifier(the red box).

passive component speaker system diagram
passive component speaker system diagram

Ok. Two amplifier channels. Four speakers. The next vital reason for passive crossovers is to split an amplifier output channel to allow the use of 2 or more speakers. Does this sound like a familiar concept from your experience of audio systems? 

Ok, but do I NEED a DSP??

Filters, level controls, equalizers, delays, and more. If I may make an analogy, there’s a lot that’s been done with very basic woodworking tools. Give a skilled builder a jigsaw and some sawhorses and you might be stunned at the quality and creativity of mobile audio system that can be constructed. Add more tools, such as a router table with a variety of bits, a table saw with a fence to cut perfect straight edges, and the build time is reduced and precision scales up rapidly. 

When we can add more tools to the mobile audio system before and after the amplifier the potential to enhance the experience can be enormous. A DSP can be thought of as a complementary array of tools to give the system designer / builder many more options.

I want to mention also, as it can’t be ignored if you ever intend to master mobile audio systems, the undeniable influence of the interior of the vehicle on the sound system. Reflections happen off of every surface and we sit very close to all surfaces inside our vehicles. We can’t throw the toolbox at this problem, but in the “Passive Crossover vs DSP” debate it’s obvious the DSP route can give us a big advantage.



Do not underestimate the power. Do you need a DSP? I want you to let me know what questions you have from this article. Leave a comment below and subscribe to get updated of future posts.

Barry Schanz

Posted on

Ohms, Subwoofers and Amplifiers


Describing Flow Through a Mobile Audio Pathway


The topic today refers to the last pair of modules of a total of four that complete the Mobile Audio Pathway. Refer to the previous article. I believe if we can’t have an audio system that plays loudly, clearly and with good balance, for a long time reliably then what is the point? 

Ever heard the saying “It’s so hot you could cook an egg on that?” That’s the sort of condition I would like you to plan to avoid when planning to run subwoofers and an amplifier to power them.

car amp cooking eggs
Click for the video.
Published on Jun 21, 2012
Once again we see Hank Veach showing off. His amps got hot enough to fry an egg on.

Review: Why do we need an amplifier? 

Answer: Small signals must be multiplied to get the loudness we desire out of our speakers.

Why? Speakers, as we use them in mobile audio systems, are very inefficient. Most of the signal, the voltage, that gets to the speaker is lost as waste heat. 

Amplifiers are much more efficient than speakers, yet they still lose a great amount of energy as heat.  

Students of Mobile Audio
Students of Mobile Audio
Join the Facebook Group NOW!


Review: What are “Ohms” and why is it so important with subwoofers and amplifiers? I am not going to get into all of the “Whys” as it will end up requiring many thousands of words. I do strongly recommend you follow the link in the carton image below to refresh your knowledge of Ohm’s Law. 

Disclaimer: If you want to be able to confidently design and build out a reliable mobile audio system you cannot do it without understanding electricity.  ‘Basic Installation Technician Study Guide’, published by MECP under the Consumer Electronics Association, is a book I strongly recommend for anyone who wants to learn to earn a living with mobile audio systems.

Ohms law cartoon
Click to learn “How to Use Ohms Law”

Common mistake: mixing up where the Ohms come from in the amplifier+subwoofer relationship

RIGHT: We speak of Ohms with the subwoofers, specifically the voice coils.

WRONG: An amplifier does not “have” Ohms.


MTX Audio has provided a web site to help you to know if your subwoofer and amplifier combination is compatible. You can go through this exercise to be a more empowered DIY shopper, or you can use this to double check my recommendations for you.

Subwoofer Wiring Diagrams

Subscribe to stay informed of the next article on Mobile Audio Pathways.

Barry Schanz
Schedule a Meeting Now

Posted on 3 Comments

Mobile Audio Pathways

Mobile Audio Pathways

Mobile Audio Pathways

Students of Mobile Audio
Students of Mobile Audio
Join the Facebook Group NOW!

The purpose of this is to lay the foundation of how we think about flow through a Mobile Audio System. Mobile Audio Pathways is a concept that you must understand like the back of your hand if you want to master sound system design. 

Sketching out the pathway by hand, by putting a pencil or pen onto paper and making it become whole in front of you, helps make it more solid and real in your mind. You can do this on a tablet or computer also. If you’re worried about what program or app you need, do not overthink it. More on that later.

Have you ever gotten deep into a project which you started without a plan, and you started to doubt yourself? You’ve got parts scattered all around, you’re stepping over and around things, you’re wondering Where Did I Put That Thing?! I’m not promising that won’t happen again once you learn and implement Mobile Audio Pathways. What I’m laying out is a proven way of planning a sound system that will serve you well.

Whatever Mobile Audio System you decide to start, start something good. Become more confident. Understand why it is you’re buying what the salesman or the discussion group or the shopping site recommended. The pathway works whether it’s a very small and simple sound system or it’s massive and pushes you past what limits you thought you have. 

Describing Flow Through a Mobile Audio Pathway


These are the 4 primary modules of a Mobile Audio System. 


This is all about playback of a recording. We’re not generating music in the vehicle, like singing into a microphone karaoke style or plugging in an electric guitar and playing a solo. The Player could be a radio, or a head unit, a receiver, your smartphone or a portable digital audio player. Did you think of another form of player? 


Signal processing means something is changing the voltage(the signal in a Mobile Audio System) in some constructive way before passing it along through the Pathway. 

An equalizer is a signal processor. This might not be a separate piece of hardware. Your player likely has an equalizer, even if it’s just a button or slider to increase the bass. That’s signal processing.


The Player and the Signal Processor work on Low Voltages that are not yet ready for the final module in the Mobile Audio Pathway. First we need the Amplifier. This is where Low Voltage is multiplied, or increased by many times in amplitude, up to High Voltage. 

The Player often has an amplifier. A head unit usually gives us the option of connecting speakers directly to it. Your smartphone may have a headphone jack, and if it does it has an amplifier.


This is the final module of a Mobile Audio System. The job of a Speaker is to take voltage and convert it into pressure against the atmosphere, which is then heard by our two ears and deciphered by our brain. 

Pictured below is a Mobile Audio System Diagram. Do you see a Player, a Signal Processor, an Amplifier, and Speakers?

mobile audio system diagram
Mobile Audio System Diagram
completed in Microsoft Word
Touch to zoom in

Subscribe to stay informed of the next article on Mobile Audio Pathways.

Barry Schanz
Schedule a Meeting Now

Posted on Leave a comment

More Power – Amplifiers

more power

More Power – Amplifiers

Do you find your sound system never seems to get loud enough? Maybe it gets loud but the details in your music keep getting more crunchy and grungy as you rotate the volume knob to the right. Distortion rises to an offensive and speaker-threatening level. Instead of giving you pleasure the music makes you mad.[1] 

Often the determined audio enthusiast turns to a bigger amplifier(MORE POWER). 

More Power, More Channels

I believe if we can’t turn it up loud and keep it there reliably then what is the point?

Might we still be in the danger zone after increasing amplifier power? It depends.

More Power Justification:

  • Increase the watts and dB SPL must go up
  • I want nothing but clean power for my highly dynamic recordings
  • The bass is overpowering everything else(more power for mids and highs)
  • Adding more speakers and I want or need more amplifier channels
  • More power means greater headroom

What is ‘Headroom’?

Headroom is the difference between the maximum output power of the amplifier and the maximum signal you ask it to output. If you want 50W at most and the amp can put out 100W, it has 3 dB of headroomHeadroom is a good thing, because it means when you push the amp hard, it won’t distort. [2]

I stopped experimenting with this extra headroom several years ago. The thirst for more power for my speakers peaked with a maximum of 250W RMS per 6.5″ midbass woofer, which I stepped up from 90W RMS.

That’s almost 3X the power. Must’ve kicked butt, right?

It was not what I hoped for. Nothing broke and I didn’t harm myself or nearby small animals. Something happens as we add more power to our speakers that I don’t think can be understood until it’s tried.

Apparent Power vs Distortion 

This article is not going to take a deep dive into that subject. I will say now that if we double amplifier power that theoretically gives us another 3 dB of clean dynamics, assuming the music has adequate dynamic range. Along with that is the thought that some distortion is completely tolerable to most people, and distortion from the speakers can not only be enjoyable but it adds to our perception of how loud the music seems to be. 

During the years that have passed since that 250W per woofer experiment I have also dropped the power all the way down to 18W RMS on similar woofers. How was it?

18W per speaker got bizarrely loud. Distortion increased a lot more obviously and earlier as the volume at the radio increased.

You might hate me for ending the article here, but remember that this is an ongoing message. Maybe it’s strange to say it could even become a discussion. Leave a comment with your questions below.


  1. There’s a Biological Reason Why Some People Get Chills When They Listen to Music
  2. amplifier “headroom” explanation? AmigaPhreak 

Barry Schanz
Schedule a Meeting Now

Posted on

Clean vs Dirty Power – Car Audio Talk

clean vs dirty power

Clean vs Dirty Power – Car Audio Talk

Matthew Allen Clark administrator of Real Car Audio Help on Facebook
Last night I was doing some research and came across an forum argument about “clean” and “dirty” power so without us starting an argument I am going to ask a question without adding my opinion (until later) I want to know everyone’s views on this especially you elder statesmen like my self who have been doing this for decades.

So question:
Give me your best definition of what is meant by “clean” power when talking about an amplifier?

The foundation of this discussion is Distortion.

Distortion is any departure from the original. That’s not a dictionary definition.

Listen to examples of audio distortion, provided by 

How to Prevent Distortion

I believe this entire debate is full of holes, and how deep do we want to take this?

In terms of audio amplifiers “Dirty” vs “Clean” power tends to mean
1) A broad judgement of quality of the amplifier.
2) The way in which the gain or input sensitivity was adjusted

I have never seen or heard this lingo used by audio amplifier designers. That doesn’t mean it never happens but this is street talk, not engineering talk.

this is street talk, not engineering talk

Back to distortion.

What is the job of an amplifier?

It takes a signal of some lesser amplitude and multiplies it to greater amplitude. By the nature of the devices required to make that happen there will be a change other than increased amplitude away from the original signal, aka distortion.

This isn’t always a bad thing. How much distortion can we hear? Does it change if we’re listening to a sound system in headphones vs a car sound system while driving on the highway?

That is to get you thinking about perception, which is an aspect of audio playback systems that I find extremely fascinating.

Clean vs Dirty power

Remember this is laymen’s terms and not engineering talk. I’m not an engineer and I would be happy to reach out to someone if the interest is great enough.

My perspective on this is “Clean Power” is amplifier output of sufficiently low distortion that it is not an influence on sound quality. In other words, it’s just making the music louder and we call that characteristic “transparency”. It is assuming the gain or input sensitivity is set in a way that does not introduce excessive clipping under normal use.

As an aside, I said “excessive clipping” very deliberately. I do not promote setting an amplifier so that it never clips.

Further reading:

Barry Schanz
Schedule a Meeting Now

Posted on 1 Comment

Amplifier Gain Setting – Introduction

Amplifier Gain Setting introduction

Have you ever said any of these things about amplifier gain setting?

  • I set it to halfway
  • It wasn’t even turned up all the way!
  • I didn’t even touch it
  • It’s supposed to be louder than that

Disclaimer: Before you get into this, please understand I do not know your individual skill level. This is a primer to a topic that brings in a lot of science, and the science is not going to be discussed in depth. This is skimming the surface. Got it? Good. Let’s go.

Why does this thing we call Setting Gains even matter? 

The purpose of an amplifier gain setting is to match the input range of the amplifier to the output voltage of the source unit.

Why? The thing we get to listen to by hitting Play starts as sound waves, which are changed into electricity, which eventually get changed from electricity(we will measure this as Volts) back to sound waves.

Back to gain setting. The source unit is the radio. Output from the radio goes to the input section of the amplifier. 

The output voltage swings up and down as we adjust the volume setting of the radio, and also as the program(music, talk radio, podcast, etc.) changes in loudness. 

We know the bottom of the range. That’s Volume at ‘0’ or when the music is paused. The top of the range can be assumed to be 3/4 of maximum. That’s for the output of the source unit

The input of the amplifier is also adjusted. When the gain setting is too low you’re going to run out of volume, or the bass won’t get as loud as you think it should. 

A Note About Loudness

Turning the gain knob to the right makes the sound louder. This is similar to the result of turning up the volume at the radio. We are doing different functions. The gain knob is not the volume setting of the amplifier.

gain knob on amplifier
Another label for amplifier gain setting might be ‘Level’.

Rob Haynes, Product Training Specialist at JL Audio in Miramar, FL, explains the four primary reasons we must have a repeatable and accurate method for amplifier gain setting. This might also be called the Input Sensitivity setting.

  1. Optimize performance from the amplifier without excessive clipping
  2. Reduce distortion
  3. Improve sound quality
  4. Avoid damaging the speakers

The following video gets right into some technical terms and it assumes you already have experience of successfully installing car audio amplifiers and the use of basic tools that are shown in the video. If you want free step-by-step guidance, and you’re on Facebook, now is the time to join our group Students of Mobile Audio.

Top 3 Gain Setting Methods (in no particular order)

  • By ear with music
  • By combining tone test tracks and a digital multimeter (set to measure volts).
  • Use tone test tracks and an oscilloscope

I’ve done it all three ways with success. Each method must include reference quality test tracks so you or a technician can go back at any given time and get things back in alignment. Turning the knob up halfway is not a method I will endorse. That’s guessing.

There have been many sources of reliable test tracks for audio systems over the years. One that came up as a free download is Focal Tools CD

Focal Tools CD
Focal Tools CD

Advanced amplifier gain setting

  • Balancing multiple amplifiers
  • What to consider in low power versus high power amplifiers
  • Clipping indicator lights and other in-built amplifier setting aids
  • How much clipping is too much?

Would you like to contribute your own amplifier gain setting article? Ask me how to be a guest author!

Barry Schanz
Schedule a Meeting Now