Mark Rutledge passed away this week.

Mark was a member of a few exclusive groups. He was an installer who became an electrical engineer, and his work was informed by his experience in the bay. He was a security-and-remote-start pro who transitioned into the audio field (a maneuver which sounds easier than it turns out to be). He pioneered the use of external devices to configure in-car products – first with the Bitwriter alarm programming tool (which I believe was his senior project for his EE degree), and extended to many other devices which were PC-configurable when that was still rare. Finally, he was an engineer who went back to school and learned business, earning an MBA. Mark had a level of credibility with installers, salespeople, buyers, marketers, and business leaders which is very rare – and it was deserved. 

I knew Mark for almost 30 years. He was working in the install bay, and came to a training we did on selling security for cars. Mark took that back to his shop, and reported 6 months later that their average ticket had almost doubled using the techniques he learned. Soon, he was working in our technical support department, taking over the management of the department shortly after I left, and eventually returning to school and progressing to vice-president of engineering. 

He then pivoted, going to iDatalink to head up an audio-focused initiative based on leveraging knowledge of vehicle networks. iDatalink’s Maestro has become a highly respected brand, known for solid engineering, great vehicle-specific plastics, and appealing features. Whether it involved head-unit integration, amplifier replacement, or the sound processor created in partnership with Rockford, products from this group are widely considered to be excellent. 

Seeing him over those years – almost 30 – I got to see him grow as a person and a leader. We have all made mistakes, but the best of us learn from them, and Mark learned from the past and improved. His team at iDatalink hold him in high professional and personal regard. He told me recently that I was a different person than when we met, that I had grown, and I believe this was equally true for us both. He recommended me to my current employer, and I don’t know that I would have this job without his advocacy. 

He will be missed by his many friends and colleagues in the 12-volt community. We offer our condolences to his family. 

A BMW 3-Series gets a multichannel 2-seat upgrade

A few years back, I bought a BMW 335xi GT. I really like the car, but not the audio. The “harman/kardon” system in my BMW should have sounded awesome, but it didn’t. 

The amplifier can generate 100 watts per channel to each underseat 8” flat woofer. The other speakers receive 30 watts per channel. The positioning of the 4” midranges and 1” tweeters up front is excellent. The center channel and rear effects channels are derived by Harman’s Logic 7 upmixer (Harman ended up with Logic 7 when they bought Lexicon). 

However, while it was loud, it didn’t sound great – and I think there were a few reasons. 

  • The center speaker was harsh-sounding. The midrange lacked any low-pass crossover filter, and the tweeter had the standard series capacitor (which filters at the rudimentary 6dB/octave). So the sound from the most critical speaker in the car wasn’t that good.
  • The center channel wasn’t loud enough to create a true center, so each front-seat passenger got their own center image, biased toward their side of the car. This may have been the goal, or maybe Harman didn’t want to drive that center hard enough since it didn’t sound that great. 
  • The midbass was lacking. This was due to the unusual decision to ask the 4-inch midranges to play down to 80 Hz, where the underseat woofers come in. Previous BMW models didn’t do this, and subsequent BMW models didn’t do this – it seemed to be something they tried in the 3-Series, and gave up in the 4-Series. Regardless, the 80-160 Hz octave measured very weakly. 

When I bought the car, I was just beginning to learn about upmixer technology – but it was my first car with a center-speaker grille, and so I really wanted to try it out. So we installed a multichannel system into my car using a discontinued processor with Dolby Pro Logic II. For reasons which still escape me (and which are not mentioned in the manual for this product), the Dolby PLII would not turn on for any inputs other than the Toslink SPDIF input. In my car, this was an easy fix – I used a gateway which converted MOST 25 into Toslink. 

So, that Dolby system was pretty good, but it wasn’t great. It certainly wasn’t nearly as good-sounding as the best upmixer upgrade I’ve been involved with – a 2017 Range Rover with Meridian (those cars use Dolby Pro Logic II or DTS Neo upmixing, and that amplifier/processor did an amazing job). I think some of it was the implementation of the Dolby PL II function, some of it was the center speaker I used (a passive two-way coaxial), and some of it was the limited audio processing available in that older device. 

I eventually removed that system, so I could test other products in a traditional one-seat arrangement – but I always wanted to try again, so recently I designed a new system. I was working with Elettromedia on a training contract at the time (now I’m a Technical Product Marketing manager with the company), so I designed a system using Audison products. I planned on re-installing my h/k processor/amplifier, but I couldn’t find it anywhere. I finally bought a replacement online, and that wasn’t cheap – don’t misplace $1200 OEM amplifiers! (Fortunately I found one used for much less). 

My system used all 9 channels out of the h/k amplifier:

  • Center
  • Front Left and Right
  • Underseat Left and Right Woofers
  • Rear Doors Left and Right 
  • D-Pillar Surrounds Left and Right

We ran all of these into an Audison bit Virtuoso. The Virtuoso has 12 channels in, and it can handle voltages up to 32VAC, so we didn’t need any external attenuation of the bass channels. Then we split those input channels up like this:

  • Center High and Low (active 2-way components)
  • Front Door High and Low (active 2-way)
  • Rear Doors High
  • D-pillar Effects (passthrough)
  • Underseat Woofers Band-passed as midbass (with a higher crossover point)
  • Subwoofer Low

The cabin speakers were all Audison Prima BMW drivers. While these speakers are in the Prima line, they are higher in performance than many of the other Prima speakers. They have cast aluminum frames on the midranges, as well as neodymium magnets and waterproofed cones. There are 2” voice coils on the woofers with aluminum formers, and the tweeters have 29mm domes (this is the largest snap-in BMW tweeter on the market). The drivers themselves are really impressive, but because they were intended for OEM amplified use, the crossover is a basic 6dB design, and this limits their potential performance – you can get higher performance by going with a steeper crossover slope in a true two-way configuration. Since we went fully-active on the midrange/tweeter crossover handoff, we overcame this limitation – and I urge anyone who’s doing a high-performance BMW system to do the same!

We retained the OEM rear-effects speakers in the D-pillar – fidelity is not crucial with these speakers, and very little power is sent to them. 

The subwoofer is an APS10D dual-voice-coil driver in a sealed enclosure.

For amplification, we used three Audison SR amplifiers for a total of 13 channels:

  • Two SR 4.300 85×4 amps
  • One SR 5.600 75×4/550×1 amp

The amplifiers and the Virtuoso processor, along with a fused power distribution block from the Connection line, are in a rack based on the Musicar bolt-in amp rack for the F3x BMW platforms. The basic design was modified by Pierce Barrett, who also did the rest of the electronic installation (Patrick Rollins assisted with the speaker drivers).

The four 75-watt channels power the rear surrounds and the rear doors. The eight 85-watt channels power the front mids and tweeters (actively), the front center mid and tweeter (actively), and the underseat woofers. We used APS BMW-2 2-ohm underseat woofers, so the SR amps actually deliver 130 watts RMS to each woofer. The sub gets 550W from the SR 5.600. 

When I fired this system up, it took me a bit of time to get the levels sorted out, but once I did, I was really surprised at how good the system sounded with the h/k amp as the source. One improvement was the fully-active front end – my previous system had not been fully active, and this was a definite performance upgrade!

When tuning it, I used the new bit Virtuoso 3.0 feature set, including phase equalization on the outputs, to get the final results. I also used the Educar Test & Tune app, which has Left Center and Right Center test tones. These are essential when setting the level of a center channel – too loud and it becomes a black hole that eats all the instruments, but too quiet and you split the center image in two. 

One of the things I have learned is about cancellations. We get phase cancellations whenever one sound arrives at two different times from two different speakers. That can happen when front and rear speakers are different distances away. It can happen when left and right speakers are different distances away, and the recording places the sound in the center and routes that identical sound to both left and right. It can happen at a crossover point in the transition band, where both the high-passed speaker and the low-passed speaker are playing the same note. 

We use delay to address all of these problems in an active one-seat system. Once we switch to an active upmixed system, we can’t use delay to address most of these issues without ruining the two-seat stereo effect we are trying to deliver. I used the phase equalization in the bit Virtuoso to address these issues, and that was another huge advantage this system had over the Dolby Pro Logic II system – I got better up-front bass, more cohesive midbass, and better imaging by using tools which most DSPs still don’t offer. 

More and more premium vehicles have 2-seat presentations – we can upgrade them more easily than we might think!


– In upmixed systems, the center is the most important speaker. Use the best speaker you can, and go active whenever possible. 

– The Audison BMW speakers are capable of great performance in an active configuration.

– The BMW h/k amp is capable of passing great sound to your system (as long as you handle the bass channel voltage properly)!

– For best results with an upmixer, a powerful DSP is needed, with phase equalization as well as parametric equalization. 

– The Educar app helps set the levels of upmixed center speakers easily. 

Speakers are meant to create air-pressure changes. 

Audison AP6.5P midwoofer
Audison AP6.5P midwoofer

 Let’s take one speaker. Let’s use the most popular size in car audio – the 6.5” round speaker. 

That speaker can play an amazing range of notes. It can play from pretty low in the bass – 80 Hz for sure, maybe even lower in some cases – and it can play up to roughly 2kHz or so before it starts to roll off from the listener’s perspective in a car door. 

But one 6.5 just isn’t loud enough. What do we do? 


We can give it more power. But each time we double the electrical power we send to the speaker, we only get 3dB more output – and 3dB isn’t that much. It’s noticeable, but it’s not impressive. 

Cone Area

We need more cone area – another speaker. So now we have two speakers playing the same notes. Double the cone area gets us a potential 3dB increase. 

Two Speakers

Audison AP6.5P midwoofer
Audison AP6.5P midwoofer
Audison AP6.5P midwoofer
Audison AP6.5P midwoofer

Power and Cone Area

Now let’s give that added speaker another amp channel, same as the first. We have doubled the electrical power of the sound system. That’s another 3dB! That’s 9dB we’ve gained! Now, we are getting somewhere. 

However, now we have a few potential problems. 


First off, we had better connect the speakers in proper polarity! If we accidentally get one connected backwards, we will cancel out a lot of the potential sound. If a speaker is connected in reverse polarity, the output is 180 degrees out of phase relative to the other speaker. When one speaker cone is moving out, the other is moving in, and the air-pressure increase we are looking for doesn’t happen. 

Two speakers wired in polarity

Two speakers wired in polarity
Two speakers wired in polarity


Secondly, we must be cautious about where we place the speakers relative to the listener and to each other. If both speakers are the same distance from the listener. If the two speakers are different distances from the listener – if the path lengths the sound must travel are different – then the sounds don’t arrive at the listening position at the same time. If the sounds don’t arrive at the listening position at the same time, they may not be in phase any longer. They may be misaligned. 

Depending on the path-length difference, and the note being played, we may inadvertently cancel out the note almost completely. That’s not supposed to happen – adding another speaker was supposed to make our system louder!

Example - two identical flat signals, after one is delayed 2.44mS - about 33 inches
Example - two identical flat signals, after one is delayed 2.44mS - about 33 inches

In home audio, we can just move one speaker to solve the problem (we may not want to, but we can). In car audio, we used to try to do that by building speakers into the floor, to make the path lengths closer to equal. It’s a lot of modification, and nowadays we have other tools we can use to address this problem – but first we have to know it’s happening. 

Four Causes of Cancellation

When do car-stereo systems suffer from this problem of driver cancellation? There are four chief causes:

  1. Front and Rear speakers. The rear speakers in most audio systems are there to help us play the system louder. In older cars, the rear speakers  were the larger ones, so they could play louder and they could play more bass. In modern cars, that’s rarely true any longer. Regardless, having front and rear speakers play the same notes causes this problem, when the rear speakers are farther away.
Front and Rear Speakers
Front and Rear Speakers

2. Speaker drivers on adjacent sides of a two-way crossover filter, such as a tweeter and a midrange. Wait, I thought crossover filters existed to prevent this? Well, all crossover filter networks have a transition band – a band where the high-passed driver and the low-passed driver are both contributing to the sound. Depending on the slope of the crossover filters used, this band might be wide or it might be narrow, but it exists. Speaker system designers have to ensure that the filters selected don’t introduce such cancellations (especially after the high-passed and low-passed speakers are installed at different distances from the listening position!) This is most commonly noticed at the subwoofer-to-midwoofer crossover point, where cancellations cause a loss of bass (as well as revealing that the subwoofer is in the back of the car).

Transition Band of a Crossover
Transition Band of a Crossover

3. Left and Right speakers. This isn’t just about stereo. A stereo system plays the same sounds from both left and right speakers when the recording engineer wanted to place some sound in the center of the imaginary stage, midway between the speakers.  But if you play a mono signal, both left and right speakers still play the same sounds at the same time – this isn’t just about stereo! When that happens, and the left and right speakers are different distances from the listening position, we get cancellations which ruin our stereo illusion – as well as reducing system volume at several frequencies.

Left and Right Cancellations
Left and Right Cancellations

4. Reflected sound arriving at the listening position later than the direct sound from the speaker. This is a fact of life in a car. Glass is highly reflective for sound, as is dashboard material. We can’t eliminate reflections in a car, even if we want to. Fortunately this is the least critical of the four causes of cancellation. The farther a sound travels, the more attenuated it becomes, and the less cancellation it can cause. 


What do we do? 

Solving #1: For years, my goal was to not use rear speakers – but that prevented me from getting the benefit of the added cone area and added amplifier power. So now I use them, but I use either delay (for a one-seat system) or phase equalization to manage rear-speaker-caused cancellations. 

Solving #2. My solution to this is to always use Linkwitz-Riley 4th-order 24dB/octave crossover filters. Then, I just have to worry about distance-related cancellation, not crossover-filter-induced cancellation. 

Solving #3. For the past decade or more, delay has been the standard fix for this problem. However, it only works for one listening position. Upmixed systems don’t rely on the same sort of summed signal creating the phantom center, so OEMs have started using them more often over the past 10 years or so. Phase Equalization can also fix this – at least partially – and Bose (among others) has been using this for at least 20 years. This is why most DSP UIs ask you for the distance you want to delay the sound – because delay caused by distance can be offset by introducing delay on the other channels, to get everything in sync. (Some DSPs ask you for the path-length difference for that speaker, and they should really make that fact clearer).

Solving #4. Don’t worry about it. We’re good. Some people want to put midranges in A-pillars to change the reflective properties, and that’s fine – but we aren’t eliminating them, we are only changing them. Reflections will still happen. 

Precision, Distance, and Frequency

Some installers have tried to impose high degrees of precision onto the management of cancellations. They’ve tried to use laser distance measurement, or complex formulas, to measure the various distances. Fortunately, we don’t need that degree of precision. As the table below shows, to reach 180 degrees of cancellation at 80 Hz, the path-length difference has to be 82 inches. That’s a long distance for two subwoofers! “Well, isn’t there partial cancellation before we get to the 180 degree point?” Sure, there is, but not at 1% or 10% of 83 inches! At 90% of 83 inches, sure, there’s cancellation! The good news is, you need to be close to 83” of path-length difference for cancellation to be come a big problem. 

As you move to shorter and shorter path-length differences, the initial cancellation frequency moves higher and higher. Those shorter path-length differences are found in cars more often!

But once we get to really small path-length differences – an inch or two – we couldn’t possible hold our heads still enough – we would have these problems all the time! It turns out, we do have these problems all the time – we have just learned to ignore PLD-caused cancellations above some frequency. What frequency? Some researchers say 1500 Hz, some say 3000, some say 6000. 

That’s why, when we have two subwoofers 2” apart from each other, and we wire one backwards, we get almost complete cancellation. At the frequencies involved, a 12” path-length difference (technically, the voice coils of 2 10” woofers would be about 12” apart in this scenario) is so close to 0 that they might as well be coexisting in the same space. So don’t overthink subwoofer distance – close gets you 98% of your benefit. 

But once we have a 6.5 and a tweeter, and one is 2.5” farther away than the other, and we use a crossover point at 3000 Hz, we just put those two speakers 180 degrees out of phase with each other at that frequency. That’s a big deal. Now if one is 5’ farther away, they are back in phase at 3000. If one is 7.5” farther away, they are back out of phase at 3000. Above a certain frequency, the higher we go, the less sensitive we are to these problems.


Once we start thinking of DSP less as a tool for prissy audiophiles, and more as a tool for cancellation management, we can use it to sell more speakers, sell more amplifier channels, and earn more business from happy clients. 

Quality Assurance – Not Just for Factories

I see conversations on social media about how to deal with potential clients who are particular. Most of the comments make me cringe. I want a client who is particular, I want a client who cares a lot about their car – so we get an opportunity to show that we care even more. 

Due to the complexity of system integrations today, we should all have quality assurance procedures in place. We would be horribly angry if our manufacturers shipped us defective product at the same rate that our industry delivers defective installations. 

This three-phase plan is based on a similar process used in the construction industry. I’ve modified it for us. 

This should not be free – no one should do it for free, and it should not be “thrown in as an afterthought”.  This should be baked into your price. 

Phase 1: Preparation

Is there a “work order”? This means different things for different shops – it might be an invoice, or a sales order, or an estimate form. Regardless of what your shop uses to start the job, it needs to be completed. 

Is there a system diagram? If the job has any complexity beyond a speaker swap, a block diagram is really important, even if it’s drawn with crayon on the back of a napkin. If the salesperson can’t block diagram it, I’m not convinced they actually have a plan. 

Are all the parts on hand? If not, what is the ETA for those parts, and how should that affect the sequence of installation tasks?

Is there a wiring diagram printed out? If one isn’t available, that may slow things down. If you don’t have access to Prodemand or Alldata, get that handled. Contact 1Sixty8 Media for a Prodemand subscription. 

Is there any gear you haven’t installed before? If so, read the freakin’ manual. 

Are there any tasks or skills required which are new for you? Let your supervisor know this up front. Research it however you can. 

Have you installed in this vehicle before? What open questions need to be answered? Door depth, channel count, stereo presentation, OEM signal type, etc? Answer these as early as you can, so you can modify the plan if need be. 

What are the client’s expectations ? For sound, for appearance, for vehicle modification, for access and storage? Seemingly small things like USB location and phone storage make a big impact on daily use of the car. It shouldn’t be about doing what’s easiest for you, it should be about making the client’s life easier. 

Is the project estimated correctly? Can it be completed in the time allotted? If not, make that clear to the salesperson up front, and to your supervisor. 

Phase 2: Installation

Follow shop standards. If the shop doesn’t have standards, follow your standards, but hold those up to scrutiny. It’s best to have shop standards on how channels are assigned, how fusing works, how to ground head units, etc. 

If anything throws the schedule into doubt, communicate this to the salesperson and management immediately, so that the client can be kept in the loop. 

If the person tuning the system is not the installer: 

  • The speaker polarity, gain, channel assignment, and noise levels should all be checked before handing it over for tuning. 
  • A system diagram is essential, with channel identification. Don’t pass a system off to be tuned if you don’t know if you got your polarities and channels right – because at that point, the statistics say you probably didn’t. We’re talking about quality control here, not casino night. 
  • Pass along the client’s expectations regarding volume, bass, stereo performance, preferred genres, and seat position. 

Take pictures. Document your job. 

Phase 3: Inspection

This must be performed by someone other than the installer, because if the installer has been missing something, a fresh set of eyes and ears are more likely to pick it up. This may be the same person who tunes it, but they need to come back before the car gets pulled out. My strong suggestion is, the car doesn’t get pulled out until it’s OK’d by someone other than the installer. This forces QA to happen for the day to progress. 

  1. Does the installation accomplish the core functions? Does it sound great, does it hit hard, does CarPlay work, does it detect laser? 
  2. Is the car properly reassembled? Are there gaps or loose parts? 
  3. Do new parts – fabricated or manufactured – look as expected? Are they in the location expected?
  4. Is the clock set? Do the lights match the dash lights? Did someone take the stickers off the radio? 

We had a 996-platform 911 GT3 in for a CarPlay radio installation. The installation was performed properly, the radio was perfectly functional – but the off-the-shelf dash bezel used in the installation was horrible. We ended up making our own to make sure this customer was happy. I don’t know if that customer would have been happy with that horrible dash bezel or not, but we weren’t willing to bet our business’s value and reputation on it. 

Now, pull out the car. 

This process builds huge equity in your brand, because you deliver a great experience at delivery. It builds confidence in your salespeople, because the delivery rarely includes bad surprises.  Finally, it helps you justify – to yourself and your customers – a proper and sustainable labor rate. 

So, create your QA process today, and start delivering more value. 

Phase is really tough to talk about. 

One reason is, we’ve confused the words “polarity” and “phase” for decades. 

The polarity of a speaker refers to the (+) and (-) terminals. If you connect them to the (+) and (-) wires, respectively, you have observed proper polarity. 

Let’s assume we have two identical subwoofers, in the same enclosure, close to each other on the same side of the enclosure, sharing a baffle. For the sake of simplicity, let’s assume they are single voice coil woofers. 

If they are connected observing proper polarity, both cones will move outward when the amp’s voltage swings positive, and both cones will pull inward when the voltage swings negative. 

If you accidentally connect one woofer’s voice coil backwards, so that its cone pulls inward when the voltage swings positive, and pushes outward when the voltage swings positive, you now have two woofers which are fighting each other. 

Both are trying to create air-pressure changes, but the work one is doing cancels out the work the other is doing. Because they are close to each other and on the same baffle, and because the distance they are separated is nearly nothing compared to the wavelengths they are playing, almost complete cancellation occurs. 

And this is a pretty common error made in car audio. Almost everyone has had this experience. We can relate. 

This is an important time to define wavelengths. Here is our friend the sine wave.  If this wave is describing sound in air, the term wavelength refers to the distance between the peaks, one complete cycle. Starting at zero degrees, you can see the progression to 360 degrees.

Now, lets think about two sine waves. These are aligned in phase.

If we sum them together, we get one bigger sine wave, at the same frequency. 

Here are two sines, at the same frequency, but 90 degrees out of phase.

If we add two sines that are not quite aligned, the result won’t be as large in amplitude as we would expect from adding two sines which are in phase. That is a form of distortion, if you think about it. 

Here are two sines 180 degrees out of phase with each other.  

This causes complete cancellation. OK, this takes us back to our hooking up speakers backwards. 

When we swap the (+) and (-) wires, we invert the phase 180 degrees. Instead of pushing the woofer cone out, we are pulling it in. 

Smaller deviations in phase – less than 180 – don’t result in such a complete cancellation. The cancellation may be near complete, or with smaller errors, it may simply prevent the two signals from adding together to be as loud as they could be. You might lose a lot of energy, or you might lose a little, but you lose some. 

This is fundamentally why single-driver speaker systems seem to be relatively insensitive to phase inversion. Some people can hear the difference when you flip polarity, but there is no scientific consensus on which is correct. For the purposes of this article, we don’t care about “absolute phase” – we only care about phase interactions. 

Now, if you think about it for a moment, the odds of two subwoofers being connected opposite-polarity of each other is about the same as the left and right side front speakers accidentally being connected opposite-polarity of each other. But we probably don’t remember this experience nearly as often. Why? 

Well, one reason is, with deck replacements, the harnesses are color-coded and it’s literally harder to make that mistake than it is with most larger clear-jacket speaker wire. 

But there are two more complex reasons at work also. 

When two speakers play the same note, the possibility of the two speakers’ output interfering with each other and causing cancellations always exists. 

Crossovers. A high-pass speaker (say, a tweeter) and a low-pass speaker (say, a midrange) will play the same notes in the transition band of the crossover – the band of notes where the output overlaps as the crossover filter begins to do its attenuation. The output of these two frequency-adjacent speakers can cancel each other because of the crossover filters we have chosen. 12dB Butterworth, for example, probably the most common crossover filters used historically, put the speakers 180 degrees out of phase with each other the crossover point –  in the electrical domain!

If both speakers are the same distance from us, and they are connected observing the proper polarity, they now are cancelling each other out simply because of the crossover filter we selected! So, experienced speaker designers simply flip polarity of one driver, and the problem is solved. Alternately, you can select crossover filter types which don’t have this problem (24dB/octave Linkwitz-Riley filters are often used for this reason). 

Path-Length Difference. If the two speakers are connected observing proper polarity, and they are the same distance from us, and they are sent the same content to play, they are in phase with each other. But what if the two speakers are different distances from the listener? The difference in the length of the paths the sound travels will put the two speakers out of phase at certain frequencies. 

With the subwoofer example above, the wavelength at 80 Hz is about 166 inches.  That’s the distance a wave takes to go from 0 degrees, all the way to 180, and then back to 0 (in a circle, 360 degrees = 0 degrees). That means that half a wavelength is 83 inches – halfway through, the wave is at 180 degrees opposite where it started. That means that if the two speakers are 166 inches apart, they will be back in phase with each other. If the speakers are only 83 inches apart, they will be 180 degrees out of phase with each other. If they are 249 inches apart, they will again be 180 degrees out of phase with each other. 

So, if the two subwoofers are connected properly, but one sub is 83 inches farther away than the other sub, there is a big cancellation at 80 Hz. Fortunately, this doesn’t happen in cars because they aren’t big enough. Even if we put the two subwoofers on opposite sides of the trunk, they aren’t far enough apart to create significant distance-related cancellation. 

However, the wavelength of sound at 125 Hz is only about 108 inches. Half a wavelength becomes 54 inches. That rough amount of “path-length difference” happens fairly often in a car, in two situations:

  • The trunk subwoofer and the door midbass are often very far apart. That is why getting those speakers in phase with each other can be so tricky. 
  • There are rear speakers playing as loudly as the front speakers.

The wavelength of sound at 250 Hz is about 54 inches. Half of that is 27 inches, and that’s a pretty common “path length difference” between the left and right doors from a listener’s perspective. A left speaker and a right speaker are playing the same range of notes in a stereo system. Any content present in both the left channel and right channel, at similar amplitudes,  will be played by both the left and the right speakers, at similar loudnesses. That means that if your left door speaker and right door speaker have a “path-length difference” of 27 inches, they will be 180 degrees out of phase with each other at 250 Hz. That’s a pretty common problem in car audio – getting cancellations starting at around 250 hz. (Note that this won’t affect left-only or right-only content). 

The wavelength of a 3000-cycle wave is about 4-1/2 inches. So if the two speakers have a path-length difference 4.5 inches, they will be back in phase with each other. If the speakers have path lengths differing by 2.25 inches, they will be 180 degrees out of phase with each other. If they have 6.75 inches of path-length difference, they will be back to 180 degrees out of phase with each other.  When can that happen? When we have midranges and tweeters slightly separated from each other, or midbass and midrange speakers. If the two speakers are adjacent speakers driven by high-pass and low-pass portions of a crossover filter network, say mids and tweeters, they will overlap in the transition band, as mentioned above. 

When the path-length difference is 1/2 the wavelength of the crossover frequency, the speakers will be 180 degrees out of phase at that frequency, which is in the center of the transition band. 

Also – and stick with me for a moment – if the path-length difference is an odd multiple of 1/2 the wavelength of the crossover frequency, the speakers will be 180 degrees out of phase. At one-half a wavelength, you’re 180 out. At one full wavelength, you’re back in. At one-and one half wavelength – or 3 halves – you’re back out of phase. At two wavelengths – or four halves – you’re back in phase. 

You do not have to remember all that! Almost all DSPs will do that math for you – you enter the distance to each speaker, and the software does the calculation. The few that don’t do the calculation allow you to enter a raw path-length difference in inches or centimeters (even if they don’t tell you that’s what you’re entering). You can calculate path-length differences by using subtraction on a piece of scrap paper. 

When we select crossover filters, we are doing the job of a speaker designer. When we use delays, we are making up for the limitations of the car cabin. In either case, we are managing phase cancellations – even if we didn’t realize it!

The reason we haven’t noticed speakers accidentally being opposite-polarity as often as we have subwoofers is that the cancellations aren’t complete – they don’t cause near-complete silence, the way two out-of-polarity subwoofers do! There’s enough phase cancellation in “normal” door speakers (i.e., in systems not using delay) that it doesn’t sound perfect either way, and it’s harder to tell the difference.

Now, delay only works for one listening position. There are other ways to manage phase (all-pass filters being one). But we can’t start to use them until we understand the objective. Understanding this allows us to use what the OEM has done to partially manage phase – and improve upon it – without wasting a bunch of time.

Recently I was contacted by a high-school friend who asked if I could assist her husband with selecting an audio upgrade. I’ve been asked recently to show how I quote via email, so here’s that email with the pics and contents I used. 

The final job ended up around $7500 with lighting and sound damping (I forgot to include it in the original proposal, which was my error – I’m a bit out of practice, been out of retail for three years!)

Hi, Mike, nice to virtually meet you. 

Sorry about the delay, I’ve been a little swamped.  So you have a 2014 Tundra CrewMax with JBL. 

I’m going to give you a lot of data. Forgive me 🙂 First some explanation, then some links, then some overall prices, then detailed line-item proposals at the end.

I also need to stipulate something. For ten years, I was co-owner of Musicar Northwest, the best shop in the northwest. They’re here in Portland, I’m on good terms with them, they take care of me all the time, I spun off our training business and I do training for car-audio professionals now. I’ve been doing this 30+ years, and Musicar is the best shop around, but we weren’t the cheapest then and we aren’t now, just as a heads-up. If you want to see some cool projects, check out their Facebook page  . The work they do is amazing. Since I left, they do even more exotics and radar/laser systems, but my old partner Tom loves Toyotas and they do plenty of them as well. 

So, if this stuff isn’t in the price window you had, I understand. Take a look and let me know what doesn’t make any sense.

What You’re Starting With and What You Will Need

As you may know, Toyota/JBL systems use a layout where the “preamp” that controls volume up/down, fader front/rear, and tone controls lives in the JBL amplifier and NOT in the in-dash receiver. When you turn the volume knob on the front of the radio, a command goes from the radio to the JBL amplifier on a data network connection and says “get louder” or “get quieter”. Same thing happens when you use the steering-wheel control for volume. 

So, when we replace the receiver, we need to duplicate the steering-wheel functions. That’s pretty well understood by those of us who know what we’re doing.

If the JBL amplifier is being retained, we need a way to tell it when to turn on, and we need a way to tell it when to get louder or quieter. The commands are sent over that data network. There are a few interfaces which can do this for us. 

If your truck has a back-up camera, these interfaces (with a couple of additional parts) can retain the backup camera. If you don’t have a backup camera and want to add one, these parts are not required when connecting an aftermarket camera to an aftermarket receiver. 

One interface on the market, the Maestro RR made by my friends at iDatalink, will plug into a data port on the back of certain radios and also support front/rear fade, and add some other functions (like TPMS sensor identification, more gauges, and trouble-code identification. Only a few radios support the Maestro interface, and some are a pain in the neck to use. 

If you’re deleting the JBL amplifier in favor of an all-new audio system, then the interface we need may get a bit simpler. We may be able to get away with a simpler steering-wheel control interface.

You will need a dash bezel-and-bracket assembly to replace the stock radio with a new radio. Since you have the metallic columns on the left and the right of the OEM receiver’s screen, there are a couple of bezels that look proper in the dash, one from Metro and one from Scosche, which have the matching metallic columns. (They may not be a 100% match but they will be close). 

So, to replace the receiver, we need an interface and we need a dash bezel-and-bracket assembly to hold the new radio and look stock. 

We will probably want a few other things. I like to install a OEM-looking USB port. I just did this for my sister and brother-in-law’s FJ Cruiser project truck, and I found a port on Amazon that fits in a Toyota switch port and has one USB for Audio and one for Charging. So that should be pretty easy. 

I’ve spoken to a few colleagues who have tried to use the Tundra OEM BT microphone (as you mentioned). They ended up using the aftermarket microphone for performance reasons. We should be able to install the new mic somewhat out of sight, though, so it doesn’t look stupid. 

I’m going to estimate all those parts (not including any new camera) at around $300. 

I’m going to estimate the time spent on a new CarPlay receiver physical installation, microphone installation, USB installation, and the interface programming and installation at around 3.5 hours.

That puts our “Radio Prep Package” – everything needed to put in a new receiver, except the receiver itself – at between $600-650.

So now we need to choose a radio. Pat told me we don’t care about a CD drive. 

The other main differences are:

Screen size (6” class or 7” class) – the screens larger than that either use a “floating” screen which looks obviously not factory and partially blocks the AC vents, or take custom work to integrate into the dash. They look awesome, and my old shop is great at doing that, but unless you tell me otherwise, I’m going to focus on 6” or 7” screens that fit in the OEM location. 

Volume control type – 6” screens can have a knob. 7” screens have a smaller “picture frame” around the outside and need buttons for volume. This isn’t a big problem in your truck, because you’re going to use the steering-wheel volume control 95% of the time. Sony has a 6” with a volume knob. It happens to have a CD drive, and it also happens to have Maestro support. 

User interface outside of CarPlay – this is a bit subjective, but I prefer the Sony user interface, then the Alpine, and the Kenwood and Pioneer are at the bottom of my list. This difference is lessened since CarPlay has a universal user interface. 

Maestro support – there are two Alpine units with Maestro support, both 7” screens, one with native nav. Then there is one 6” Sony with CarPlay support. Then there are some Kenwoods and I think some JVCs. The two reasons I suggest Maestro are really 1) If you’re retaining the JBL amp, and 2) If you really want to know which tire is triggering the TPMS. I have it in our Subaru, and I like it. 

Wireless or wired CarPlay – Finally, a few units have Wireless CarPlay. This is really questionable in value. First of all, Bluetooth can’t support CarPlay – you need to use the WiFi radio. We have a discontinued Alpine with wireless CarPlay in my wife’s Subaru, and it seems to work fine, but it does drain the battery a little faster, so you’re going to plug your phone into the charger anyway, and once you’re doing that, you might as well just plug in with USB and not use wireless. 

Now, some of these radios are a bit hard to find right now. Lots of people are upgrading their cars at the moment, and supply from Asia was interrupted for a while and has not kept up. 

The basic 6.5” Alpine is $300, the 7” Alpine I like with Maestro is $600, the 6” Sony with a knob and Maestro is $500, and the Alpine 7” with native navigation (works when you’re outside of 4G cell networks and CarPlay won’t work for you) is $1300. 

The $300 Alpine is not as nice as the others, and it’s been really hard to get due to backorders. 

My suggestion is the $600 Alpine or the $500 Sony, unless you want native navigation as an option, and then the $1300 Alpine is the way to go. (Note: We ended up using a $600 Pioneer with Alexa, since he was considering an external Alexa unit). 


Alpine 6.5” that’s hard to find:

Alpine 7” CarPlay with Maestro:

Alpine with nav:

If you don’t care about the TPMS, the Sony 7” is my favorite. No CarPlay support, but a great-looking screen and a great UI and a great design to the buttons and the hardware. It’s $600. Here’s the link.  

If you need wireless CarPlay, you’re going to need to go Kenwood or JVC, and I don’t have any direct experience with those units, they are relatively new. 


Pat mentioned that you guys wanted more bass. There are two ways to get this – add a sub to the JBL system, or replace the whole JBL system. 

The subwoofer I would use either way is the JL Audio Tundra Stealthbox that goes behind the seat. Stealthboxes are made in the US fiberglass vehicle-specific enclosures, and they are the class of the field. Check it out here. 

I’m not a fan of the Tacotunes-style wood enclosures. They aren’t well-made (I bought one for a Tacoma project and didn’t use it), and they often take away interior space. 

I would use this Stealthbox with a 600-watt amplifier. Depending on the level of sound quality we want to get, we can use a standard amplifier, or an amplifier with DSP that will let me get the best-quality bass and a better blended transition from the sub to the cabin speakers. The standard amplifier amplifies the signal meant for the JBL subwoofer, but sends that signal to a better subwoofer. The DSP amplifier allows us to optimize the signal for the new subwoofer, getting lower notes and integrating it with the cabin speakers better, so the sub doesn’t seem to be standing on its own.

If you’d like to replace the JBL system and get better sound, that’s my specialty. Here’s the challenge for most people in my industry. There has been a lot of progress the last 10 years at making aftermarket sound systems sound better in one seat. There is not much equipment or skill at making cars sound better from both front seats. 

That’s what I just did for my sister and brother in law, and the speaker system provisions in the FJ are almost identical to the Tundra – the same technique will work. 

The biggest problem for car stereo is that we don’t sit in the center of the car. Because sound travels relatively slowly, when you’re closer to one speaker than the other, you get these weird periodic cancellations that really damage the sound AND where it seems to come from. This graph would be perfectly flat if the measurement was taken exactly between the front seats, over the center console. But in one seat or the other, the sound gets ragged:

Without any phase equalization or delay:

These problems affect both front seats. There are a handful of amplifiers with DSP sound processing in them today, and only some of them have the tools that let me deliver great sound in both front seats at the same time, without compromising one side for the other. Basically, we need to flip the phase of the 

signal on one side within a narrow range, and NOT on the other side, so the worst of these cancellations are corrected. (Our brains are relatively insensitive to the higher cancellations).

We usually use delay to solve these problems in ONE seat, but they make the problems worse in the other seat. So using delay is out.

After use of phase equalization:

In a vehicle that you would be in all the time without anyone else, it might not matter, but it sounds like you guys plan on both being in this truck at the same time for road trips, so a “two-seat” system might be of interest to you.

Your Options

So, the options are:

Add-A-Sub 1: Stealthbox, normal amplifier w/600W

Add-A-Sub 2: Stealthbox, DSP amplifier w/600W

Full System Upgrade 1: 900W 5-channel JL Audio amplifier, Audison AP2 dash speakers and AP690 door midwoofers. JL Stealthbox. The amplifier would go under the seat in place of the JBL amplifier, and we would mount it to OEM bolt points with a custom-made bracket (no drywall screws through the floor of your truck, which is how most installations happen in our business). Same with the speakers – custom mounting adapters, installed using OEM hardware, no permanent mods to the car.

Full-System Upgrade 2: with Two-Seat Optimization, JL DSP 5-channel amplifier with 900W and 5 channels. Same system as above, but with DSP tools that allow me to get a great stereo presentation for both front seats, overcoming the cancellations.

Both of these plans leave the rear speakers stock, on head-unit power, and focus on upgrading the front 4 speakers and the sub.

I was going to go ahead and spend $26 on a custom made switch from CH4x4 that would let you switch from one-seat mode to two-seat mode.

Pricing Overview

The new receiver in these estimates is $600. You can make adjustments up or down if you select a different one.

New Receiver, assuming $600 receiver, $1345 total installed.


New Receiver and Stealthbox added to JBL system: $3985

New Receiver and Subwoofer with DSP amplification: $4445

New Receiver and Full-Range System 1 (no DSP): $5100

New Receiver and Full-Range System #2 (DSP, 2-Seat) $6020

Detailed Proposals

These show you the time estimated and the cost of the parts and labor. The hourly rate is $100, with a $10 consumables supply charge.

Receiver Cost: 

This presumes a $600 radio.

Receiver and basic Add-A-Sub:

Receiver and DSP Add-A-Sub:

Full-Range Upgrade 1, no DSP:

Full-Range System 2 with DSP:

Thanks for looking at all this! Let me know what questions arise from all that.

Best regards,


Before we talk about center channels, it’s probably helpful to understand a bit about how recordings are made and why and under what conditions a stereo recording works with a stereo system.

2-channel recordings are made up of three kinds of “sounds”: 

  1. Pure stereo information: These are sounds that are ONLY recorded in the left or the right  channel. Sometimes these are called “differential.” 
  2. Mono information: These are sounds that are recorded exactly the same in the left and right channels. These are sometimes called “common.”
  3. Off-center information: These are sounds that are recorded in both left and right, but at different levels or different phase.

Here’s what that looks like. This is the seven drum beat track from the old IASCA disc. 

The top half of this display is the left channel and the bottom half is the right channel.

It’s pretty simple to see that the left drum beat (the first one) is only recorded in the left channel and the right drum beat is only recorded in the right channel. The drum beat in the center is recorded in both channels.

If you look carefully, you’ll see that in the left channel, the beats gradually decrease in level and in the right channel, the beats gradually increase in level.

The fourth one (the one in the center is the same level in both channels.

In order for the drum beats to sound like they are the same volume, the total energy of the two channels combined has to be the same. For the center drum beat to be exactly as loud as the left or the right, the sound has to be recorded 6dB lower in the right and the left. If it’s less than -6dB, then the center will be quieter than the left and the right. If it’s more than -6dB, then it will be louder.

The second and third drum beats and the fifth and sixth, are recorded so that they appear in between the center and the left and the center and the right. A 6dB difference in level moves the image HALFWAY.

the sum of the energy in the left and the right channels is the same for each of the drum beats, then the sound of each of the beats will sound like they are the same level.

In the picture below, I’ve mixed the stereo track to mono to see if this is the case for this track.

It isn’t. The second, third, fourth, fifth and sixth are all louder than the first (left) and the 7th (right). The fourth (center) is slightly quieter than the second, third, fifth and 6th.

Nevertheless, in a properly set up stereo system, we’ll hear the beats move pretty evenly from left to right.


To make the definitions above clear, the first and last beats are “pure stereo”. The fourth is “mono”. The second and third and fifth and sixth are “off-center”.

This is called “panning” and it’s how the recording engineer who does the final mix places instruments and voices across the stage in the track.

Raising the level of the mono information compared to the stereo or off-center information helps to move images forward and rearward. Some additional realism can be added using reverb.

OK. A stereo system that consists of two speakers is ideally designed to pass these conditions through the system unchanged. If that’s the case, the the panning and reverb works and we hear the instruments and vocalists in their proper arrangement across the stage in between the two speakers.

I’m not going to discuss ambient sounds or sounds that seem to come from outside the bounds of the speakers in this tech tip–this is about center channels and the processing required to make them work.

The conditions for a properly set up STEREO system are that the levels of the two speakers  and the frequency response must be precisely matched and the sound from the left and right speakers should arrive at the listener at the same time.

In a room, that looks something like this:

 Traditionally, for a stereo system at home, we simply place the chair in between the two speakers we assume are already precisely matched from the factory and that’s it. Simple. If the speakers have reasonably flat frequency response, then what’s on the CD passes through the system mostly unchanged and we hear something pretty close to what’s on the CD.

What if we want to share the experience with someone else?

Or, what if we provide a venue for lots of people to experience something that’s been recorded at the same time. Like here:

What if we’re charging everyone in every seat $25? Obviously, anyone who cares about the whole experience will book their seat early and choose one as close to the center as possible. Everyone in the center in each row will get a pretty good representation of placements during the show, but what about the people on the left and on the right?

The audio track that accompanies video in movies can add a sense of realism. Watching Transformers on an iPad with the iPad speakers, for example, doesn’t provide the realism that we expect when we fork over a bunch of cash to go to the movies.

This is what multichannel was originally designed for: Movies.

In order to provide a sense of realism, it’s helpful for dialogue to appear to come from the location of the screen or the location of the image of the person speaking on the screen. When something flies over, it’s helpful for the sound to also fly over.

This is what multichannel does. It provides better realism.

This is what the ENCODED audio track on a DVD does. It includes six discrete streams that can be DECODED by a decoder into six channels of audio. Newer formats will encode many more tracks–all in the interest of adding realism.

Years ago, movies weren’t distributed on digital media and only two tracks of audio were available–on film and also on VHS tapes.

So, upmixing was designed to “encode” all of the surround information into the two audio tracks that were available.

Multichannel systems that relied on upmixing to place sounds in discrete channels relied on “decoding” the panning between left and right channels.

Early analog versions “Dolby” decoded left right and a rear channel. To encode a sound into the rear, the mixer put the information in the left and right channels 180 degrees out of phase. The decoder would send that to the rear speakers.

That decoder was included in receivers for use at home when watching movies.

Later, the technology was adapted to add an output for a center speaker, and that was designed to place the vocal track at the screen in the center speaker.

The most popular format was Dolby Pro Logic. It worked well for movies, but music listeners were often too lazy to turn off the decoder when they stopped watching movies and listened to music. The original pro logic didn’t sound so good with music because it often placed TOO MUCH information in the center and too much information in the surrounds.

For years, multichannel was maligned by audiophiles for this very reason.

Finally, some of the algorithm was changed and a “music’ setting was added. The damage was done, however.

Because the original objective for multichannel audio was enhancement of realism for video, and because both the encode./decode format (discrete surround) and upmixing (turning 2-channel program material into multichannel material) we both called “surround”, confusion ensued.

There’s a poor understanding of the difference in the audio industry and among enthusiasts and the two are often conflated.

OK, back to the topic: center channels.

In a stereo system, there’s ONE good seat. That means only ONE listener can enjoy high fidelity playback at a time. That listener has to be placed in between the speakers.

What about a car? 

In a car, no one sits in the center.

So, we have to optimize this somehow. We can EQ the left and the right channels independently, so that the sound from both speakers is precisely matched in level and in frequency response.

Then, in order for stereo to work, we have to fix the arrival time problem in order for what’s on the disc to pass through the acoustic system as unchanged as possible.

So, we delay the left channel so sound from the left speaker arrives at the same time as sound from the right speaker.

I see diagrams like this one all the time used to illustrate this. It’s a complete misrepresentation. 

The left speaker isn’t “moved outward”. It just plays a little later than the right speaker.

It’s a stupid graphic. It’s similar to saying that if you and I are going to meet at the Blue Note at 9 PM and I live next door to the club and you live on Long Island, if I leave an hour later than you so we arrive at the same time, my house is farther away. That’s ridiculous.

This is NOT what happens:

 This is what happens:  

We simply hear stuff that’s recorded the same in both channels in the center, as we would if we sat in the center. The center image appears directly in between the two speakers.

When we adjust the levels and the delays to compensate for one listener in the car, we make the problem worse for someone sitting in the other seat.

The sound from the right speaker arrives much earlier and the level of the right speaker is higher–we turned it up a little bit to compensate for the attenuation that happens because it’s farther away.

When we tune a stereo system, we often use delays to set the image.

When it’s correct, the center is in the center. 

No matter what we do, we can only optimize this for one seat.

So, what if we put a speaker in the center? 

If we combine a center speaker with some software that can send the mono information to the center and remove it from the left and the right, then the center information appears in the center no matter where we are seated.

This is what an upmixer does. It EXTRACTS the mono information and sends it to the center speaker. It also removes some (or all) of the center information from the left and the right.

Since the car is symmetrical and so are the listening positions, this works in both seats.

Now, the only information that relies on an acoustic sum that’s correct (like our stereo system) are the images that appear in between the center speaker and the right or left speaker, like this:

The same rule applies, here. The center speaker should have the same frequency response as left and right and should arrive at the same time.

Check out the distance between the left speaker and the listener and the center speaker and the listener. Much more similar than the left and right. This dramatically improves the time problem and dramatically minimizes our need for delay–which is the tool that fixes one side and destroys the other.


So, now we can use simple level adjustments to optimize the image. Easy peasy.

So, what happens if instead of using an upmixer to extract the mono information, we just use a summed signal (L+R) for the center?

Think about the rules.

 And this is why center channels get a bad rap.

If one considers all the mystery surrounding how to determine exactly what the arrival time of a speaker actually is (do I use a tape measure, do I measure impulses, so I use clicks and pops and screw around for a week), this is a much simpler and more straightforward way to tune the front image.

You should hear this:

 What if we use a simple sum (L+R) for a center signal?

Now, if we have a signal that’s recorded in only the left channel or the right and we’ve set our center channel level to be as loud as the left and the right, what happens?

The Left image appears in between the left speaker and the center speaker and the right image appears between the right speaker and the center channel. We’ve  reduced the stage width by half.

If you hear this, turn down the center:

 If you hear the center in front of each seat, turn up the center: 

If you hear this, turn down the right:

And if you hear this, turn down the left:

So, you’ll hear all kinds of crap about upmixing:

1. It’s bullshit, and it doesn’t work

2. It isn’t “real”

3. I don’t watch movies in the car.

4. It sounds over processed

5. It narrows the stage

6. There isn’t enough encoded material

None of that is true or relevant.

It’s just easier. It sounds at least as good as a properly set up stereo system. And, it works in both seats.

Ken Ward: “I’ve been doing this over 30 years. I’ve seen many multichannel approaches. I hadn’t seen one sound good until this past year, using PLII and Andy’s upmixer tuning recommendations. 

The people who have sold these products haven’t done a good job of communicating the purpose of their upmixing. I read the white papers on Logic 7. I read a LOT of stuff. And until this year, I wouldn’t have sold any of my clients multichannel sound because I hadn’t succeeded in making any of it sound any good. I can make stereo sound great, and my reasons for avoiding multichannel sound in cars for 30 years were good and reasonable and based on experience, not ignorance.”

Andy Wehmeyer: Of course, I agree with that. To Ken’s point, I will agree that all of these “advancements” have fallen short in one way or another. Some of them have been terrible, but most of them haven’t been well understood.

So, here is an explanation of why some of these processors have really sucked, and why most of them have included limitations. Many of them, used carefully and set up properly have provided better performance in both seats, but rarely matched the performance of simple stereo in ONE seat.

OK…for “Surround Sound” there are two basic objectives. 

Originally, the objective was to increase the spaciousness or sense of envelopment in a MOVIE soundtrack, and in audio playback at home too. 

As I mentioned in the tip yesterday, the medium for distribution of video to homes and to theaters was film or tape of some kind which contained only 2 audio channels. Whatever audio track came along with the video had to be included in these two channels. 

Watching a movie in stereo, provided the speakers were large enough and the amplifier powerful enough, could adequately provide the full range of audio, including the visceral impact of loud events, but it couldn’t provide a 3-dimensional experience (including sounds behind and to the side of the viewers. Falling rain, for example, is an event that should appear to happen all around the viewer, even as dialogue should be presented from the location of the screen, in front.

So, rear speakers were added. Sending simple stereo to the rear speakers could make the rain sound like it came from all around, but that would also screw up the placement of the dialogue. That’s the same as rear speaker in a car, which, if they play regular stereo, screw up the image in the front. 


OK, so how do you get a separate rear speaker channel into a 2-channel medium?

You encode them. That means you include them in a way that they can be extracted later. Ideally, the encoding would be done in a way that doesn’t compromise the performance if the track is played back in simple stereo. 

So, in a stereo system we have two inputs and two outputs. Those are left and right. 

In an original “surround sound” processor, there would be two additional outputs, left rear and right rear. If we’re just using a stereo system, we would simply hook up to the left and right outputs. 

Remember, our film or tape only has two channels, so this has to work even WITHOUT an additional processor: if you play the movie WITHOUT the processor, it has to sound like stereo.

I think the original encode/decode may have been Dynaquad, which was used on some record albums. It didn’t perfectly preserve the stereo when played back over two channels, but it was pretty close. 

Here’s how that works:

Below are the encode and decode matrices (thanks Wikipedia). 

Don’t worry, this isn’t as hard as it may seem.

On the left, you have Left Total and Right Total. That’s what’s recorded on the left and right channels. On the top, you have the components of that signal. Ideally, for playback over 2-channels, the LT (left total) and RT (right total) would equal L and R. If the only change was additional output level, that would be OK. 

In the matrix at the top, you have ENCODE. That’s what goes into the recording. On the bottom, you have what the processor does to extract the channels. 

So, for ENCODE, on the left recorded channel, we have Left information (1 indicates no attenuation or amplification). We also have some right information (.25) indicates that the left channel is included in the right but attenuated by 12dB (20log (1/.25)). 

Stick with me…

Also in the left channel, we have another left. That’s Left Back. OK. Now we have two lefts (so the left signal is twice as loud). Finally, the right channel is mixed in out of phase and attenuated by 6dB. That’s what -0.5 means. 

The right channel is the opposite of that. 

So, if I play this back on a regular stereo system without the decoder, in the left, I get left x 2 and right out of phase and attenuated by 6dB. However, because i have 2 lefts, the relative difference between the left and right signal in the left is 12dB. So, in stereo, I get a little more separation and a little less mono (phantom center) because some of the common information is cancelled (because a little right is inverted and mixed into left). 

The opposite happens in the Right channel. 

So, when we hook up a stereo system, we don’t get EXACTLY what we expect, but it probably doesn’t suck.

When we play this back through the processor that includes a DECODER, look what happens. The left front gets just Left Front. That’s the 1. The right front gets the Right Front. 

The FRONTS play back in stereo. 

The left back gets attenuated left and some attenuated right out of phase. The right back gets attenuated right and some attenuated left out of phase. That means that MONO information is attenuated in the rear channels. This adds some sense of spaciousness.

So, the result of listening to a Dynaquad system, so long as you sit in the sweet spot, just like a regular stereo system, is stereo with enhanced spaciousness. 

If you don’t use the decoder, you get stereo, basically.

If you don’t sit in the center, you don’t get stereo, just like when you don’t sit in the center with a regular stereo.

OK. So the idea with Dynaquad was to add some spaciousness to the sound for music playback.

OK for the sake of brevity, lets skip ahead to Dolby–there were a bunch of intermediate schemes. No need to go into that.

Remember, this is a way to get surround (rears) onto a 2-channel medium and not screw up stereo playback too badly.

Below is the Dolby Surround Matrix:

So, the original Dolby Surround included a center and one surround channel. The CENTER channel is Left + Right. One divided by the square root of two is .707. That’s equal to 3db of attenuation. So, the center channel plays left plus right and plays that 3dB louder than the left or the right. 

So, if we sit in the center, like we do in a stereo system, the phantom center works OK. Some narrowing of the stage happens. But…hang on…

The +j is a positive 90 degree phase shift and the -j is a negative 90 degree phase shift.

Now, the rear surround speakers are out of phase, they’re both attenuated by 3dB, and they help to restore the spaciousness to the front. Basically, the signal to BOTH rears is L-R (the difference signal).

The ESP-3 did something similar.

Now, when played back over stereo, this works pretty well. When decoded, we have good performance for movies–mono information which is mostly dialogue and stuff that happens in front of us is firmly anchored in front and the out of phase stuff comes from the back, too. 

More spaciousness.

OK…that worked pretty well, but the next improvement was to be able to locate sounds to the sides and to be able to differentiate between left and right in the surround speakers. 

Next came the first Dolby PL2:

In PL2, the center is still 1.4(L+R). That’s Left plus Right playing at +3dB.

But now we have sides and rears, both phase shifted by + or – 90 degrees and the Left total and Right total are similar enough to stereo that when we listen to an ENCODED soundtrack in stereo, this sounds OK.

Remember, this DOESN’T enhance the front stereo very much for an offset listener.

OK…so, we need an additional improvement. Movie theaters are full of people and with all of these, there’s no improvement to left center and right localization for people who aren’t seated in the center. 

So, what to do? 

PL2 does a pretty good job of resolving left and right in the sides and surrounds, but it narrows the stage in the front a little bit. But the front ONLY works for one seat.

Now, at the same time that  all of this is going on, there are some other people working on other ways to add spaciousness for AUDIO playback. Remember, these matrices are designed for VIDEO. 

The other guys are trying to find ways to add spaciousness by adding reverb and delay and applying that to the rear speakers and sometimes to the front speakers, too. Early reflections (or short reverb) added to the front can provide a sense of a space around the instruments. Late reflections can provide a sense of a listening space. 

This was the thinking behind processors like the Yamaha DSP1, which provided lots of SOUND FIELDS, like Church, Hall, Stadium, Jazz Club. These were AUDIO processors.

All of these were ways to enhance the sense of space, but none of them addressed the localization problem in the FRONT.

Now, the problem in the front is crosstalk, or channel leakage. We want to enhance the localization of a center vocal or center dialogue, which is also present in the left or the right speaker, we can make the decoder determine if the dialogue is, in fact, in the center and then we can force the decoder to turn down the left and the right speakers during the dialogue. 

Simple, when something is in the center, the left and right speakers don’t play.

That works fine when only one thing is happening, but if someone is speaking and something else is going on in the left or the right channel, then it’s attenuated along with the center information. That sucks and no one likes it. 

Another way to do it is to use phase to cancel the center signal in the left and right, but that doesn’t work either because cancelling one channel’s content adds inverted signal to the other channel. 

Ugh. None of this works.

So, we’re still stuck. we need a way to preserve ambient sounds but shift the focus of the algorithm to the sound that’s DOMINANT. 

What we need is a vector. We can do front to rear OK and left to right OK, but the combination is a problem. 

So we need a way to determine the ANGLE of the dominant sound, so we can use an ACTIVE MATRIX to turn speakers up and down quickly so that we minimize annoying loss of events in the sound field and provide a better representation of placement in front.

So, things that are recorded out of phase are REAR DOMINANT and in phase are FRONT DOMINANT. Level between right and left determines LEFT and RIGHT DOMINANCE.

Now we can calculate a vector and use the resulting math to turn speakers up and down quickly. In the diagram above, the resulting vector would cause the left, the rear left and side left speakers to be turned down a little and the front right, and side right to be turned up a little bit. 

This happens really quickly so we don’t hear it–kind of like frame rate in a movie.

This is how matrix processing works. Dolby PL2 and Harman’s Logic 7 processors are matrix processors.

Now, all of this was done for MOVIES. What happens when the signal isn’t encoded?

To Ken’s point, none of this was very well communicated and all of it was intended for movies. People are lazy and when they put in music, they didn’t turn off the encoder. Plus, they wanted all the speakers they had paid for to play.

Audiophiles hated the way this sounded with music for a couple of reasons: 

1. The math was designed to firmly place dialogue in the center, so the turning down of the left and right speakers was too aggressive. It resulted in a narrow sound.

2. Out of phase information was steered to the rear. At the time, guys mixing music would sometimes attempt to add spaciousness to stereo tracks by putting some sounds that appeared in one channel out of phase in the other channel. This breaks the algorithm and those sounds would steer rear. Later, when compressed files became available, some of them didn’t preserve the phase relationship between left and right and that would also cause steering artifacts.

These sometimes sounded like the rear and side speakers would turn on and off. With MS-8, moving the fader to the back made this problem apparent and I used to get calls all the time asking what was broken.

Keep in mind that all of this was designed to get a movie soundtrack onto a 2 channel medium and that performance for MUSIC was a secondary consideration for the people who designed this stuff. Improvements for music were all about tweaking around the margins for what improvement was possible for when people forgot to turn off the decoder.

Then, came DVD. Now it was possible for the audio track to contain lots of additional information. In addition to a basic stereo track, there were also five additional digital streams for the additional channels in a surround system. If you wanted stereo, you could choose to play back only the unaltered stereo track–your receiver would do this automatically. If you had a surround decoder, then the surround streams would be played.

The benefit of this is that now the guys mixing the soundtrack for the movie could place events in discrete channels. The surround didn’t have to be included in the stereo track, but it often was for BACKWARDS COMPATIBILITY. 

That meant that if you had an old decoder, you could still hear surround when you watched a movie. That separate track was often available in the DVD menu, and it was called 2-channel downmixed.

And this is where the confusion came from. In an effort to automate all of this, marketers failed to explain any of it. They just figured that the receiver would take care of it and no one would have any trouble. 

The license for the software often included the discrete decoder (Think Dolby Digital), the adaptive matrix (the new Pro Logic 2) and even the original Dolby Surround. All of this was too complicated for casual users and all of it was lumped in together, despite being designed for different systems. PL2 was later updated with a Music setting that worked really well for music too, but it was so late that audiophiles never caught on. 

This confusion is similar to the confusion over bluetooth audio quality. At the baseline, it has to work, so it defaults to the best configuration supported by both the transmitter and the receiver. Some configurations sound good and some not so good, but no one except for the enthusiasts know the difference.

DTS, which is the Dolby competitor had a similar collection of software sold in the same way.

Another upmixer to watch out for: Harman QLS 3-D (Quantum Logic Surround).

I can only find three car brands that use this: Hyundai/Kia, Maserati and Lincoln, and I can’t find any lists or any additional information about it on the Harman site.

Anyway, here’s how it works.

In a stereo system, images are placed across the soundstage in a process called “panning”. Sounds on the left are recorded in the left channel. Sounds on the right are recorded in the right channel. Sounds in between are recorded in both channels at different levels. Sounds that should appear in the center are recorded the same in both channels. Sounds that are left of center are LOUDER in the left channel than in the right channel. A 6dB difference between the channels places the sound halfway between the center and the left or the right.

The sense of space in the recording may be real or synthesized. That sense of space is generated by reflections in the room in which the recording was made, or they may be generated using some kind of reverb (old school) or a convolution process (combining the impulse response of a real or a synthesized room) with the recorded signal. These reflections happen LATER in the audio signal than the initial sound of the instrument. 

QLS separates these signals in two ways, and it’s quite ingenious. To separate the panning across the front stage, it uses a series of filters that extract the common signals for the center, the left and right signals for left and right and intermediate signals for sounds in between left and right. 

Basically, these filters compare the left and right channels to determine whether a sound is the same in the left and right channels and for level differences. I think there are seven filters. Once those sounds are extracted, they can be steered around the speakers in a car like a horseshoe. There are basically two “modes”: In the audience or on the stage. On the stage mode wraps the sides all the way around to the back speakers. In the audience does less wrapping. 

In addition, there is a filter which separates the initial sound from the room response included in the recording. The room ambience is extracted from the recording and it can also be steered to the output channels. The level of the room can be increased or decreased in each of the channels to place the listener in the front of the room or in the back of the room. The ambience can also be eliminated and the listener is left with a completely “dry” recording. 

Depending on the settings, this may or may not sum through a summing processor acceptably. If it cannot be defeated entirely, then grabbing the left channel and grabbing the right channel is NOT going to be sufficient for any system that has to sound great.

If you run into this thing, you should use it or remove it entirely, if possible. In the Hyundai Genesis, you may not be able to remove it unless there’s a MOST adapter that makes it possible. 

So, Ken Ward, you’re right. In audio, Surround Sound may be the most poorly explained product ever. 

And none of it was designed for cars. It was included in head units back in the day because many of them played DVDs and the upmixer was included in the license. Hey, who doesn’t want to add a million additional features to the packaging? 

No one explained it. Few people understood it. Lots of products included it. Lots of customers and installers were confused by it.

Now that super fast DSPs are available, there are lots of possibilities. Instead of the adaptive matrix that turns channels up and down to direct the sound in a particular vector, we can just extract information and send it to the right speaker. We don’t have to rely on phase cancellation and level to remove sounds. 

That means we can upmix differently and design the performance for different spaces and different experiences. 

In the car, we don’t care about resolving a plane flying overhead and we don’t watch movies very often, so maybe we don’t care about the ability to place rear events in the rear.

What we care about in the car is expanding the sense of space so it doesn’t sound like we’re in a car. We want a wider and deeper stage. 

Maybe we don’t care about a stage and we want to distribute sound to all the speakers in a way that sounds kind of like what we hear in a club. 

Most upmixers in the past were designed mainly for movies, but not any longer.

So, try to keep an open mind. Some of these things can simply improve the stereo we are used to in a way that’s in keeping with traditional goals for stereo systems: better believability, a wider and deeper stage and acceptable performance in more than one seat.

I first met Fraser Hiebert and Kurt Porter in 2019 – I was doing training for the Canadian Elettromedia importer at the time, and NextGen of Saskatoon, Saskatchewan, had just picked up Hertz and Audison. We did a “onboarding” webinar with them – back before webinars were cool. 

Well, over the next year-and-a-half, I’d hear from Fraser and Kurt occasionally with questions – always really good, I-did-read-the-manual sort of questions, the kind I enjoy answering. I helped them set up a Logic 7-upmixer retention demo system in Kurt’s C-Class, that sort of thing. 

This particular client ordered a new 2021 GMC Sierra 2500, and wanted a first-rate stereo system to be integrated with the GMC Intellilink head unit. 

The truck came to NextGen almost right off the lot, and great care was taken to protect it in the bay.

The vehicle came with the factory Bose amplified system. This system uses phase equalization to provide good imaging to both front-seat occupants. 

Fraser decided to eliminate that processing, start with good old 2-channel stereo, and deliver a one-seat stereo presentation, so a Zen Audio AVB-GM external preamp was used. Like other Zen Audio preamps, this device connects to the digital infotainment network in the vehicle, intercepts messages sent to the Bose amplifier from various devices, and decodes the audio in those messages. The audio is converted to analog and then volume-controlled on the six RCA analog preamp outputs.

In home audio terms, this removes the factory audio system from the analog signal path completely. The audio is preserved in its digital form until after it reaches the Zen Audio’s digital conversion circuitry. Fraser and Kurt configured the Zen to use output channels 5 and 6 for a non-fading signal suitable for subwoofers, with output level of those channels of the Zen controlled by the bass tone control in the GMC Intellilink head unit. 

The analog preouts fed the Audison APF8.9bit DSP amplifier. Unlike its predecessors, the APF8.9bit (or “Forza”) has dedicated preamp inputs on a different connector than the speaker-level inputs, for maximum signal transfer. The amplifier is rated at 85 watts per channel into 4 ohms, and in this system it was configured in a staggered-six-channel mode: 85 watts by four channels, and 260 watts by two channels. The Forza amplifier/processor was mounted to matching white “Starboard” PVC, on the back wall near the OEM Bose amplifier.

The Forza served as the DSP system controller, and powered all speakers other than the subwoofers. The front stage was the Audison Voce AV1.1 soft-dome tweeter and AP690 woofer, powered actively with 85WPC to the dash tweeters and 260W to the door midbass drivers. 

The Voce tweeter is designed to handle a 2500 Hz crossover point – around an octave lower than the industry average. This provides a warmer sound, it elevates the stage to some degree, and it has better symmetry to the dispersion characteristics in the uppermost octave played by the door midwoofer. That OEM 6×9 midwoofer was replaced by an Audison AP690 6×9 midwoofer, retaining maximum cone area for more midbass immediacy. 

Both the midwoofer and tweeter drivers were mounted using custom-made PVC or ABS adapters made by NextGen. “The Canadian winters get a bit cold and icy – these plastics are proof against water damage or temperature-related breakdowns”, explained Hiebert.

This left two 85-watt channels of the Forza for powering the rear coaxial speakers. The 9th channel of the Forza is designated for subwoofer amplifiers, and in this system was routed to the Hertz HCP1DK mono subwoofer amplifier. This amplifier, mounted under the aft end of the center console, delivered a ton of continuous power into the subwoofer load, which was two JL Audio 10TW3 shallow-frame subwoofer drivers. 

The 10-inch JL subwoofers did their work inside an MTI Acoustics sub enclosure specifically designed for the Silverado/Sierra crew-cab trucks. Christerfer Pate and his crew at MTI have made quite a name for themselves with high-quality vehicle-specific subwoofer enclosures for full-sized trucks the past few years, and these pictures show how “OEM” this enclosure looks in place. 

The NextGen crew did a great job. 


We estimated 18.75 hours and charged him that. The whole thing took us 20-21 hours total, we were behind by about an hour. This is mostly because of our major issues with trying to tune it with a new laptop. We would have finished in 17-18 hours most if it wasn’t for disassembling and reassembling the car a couple times trying to understand the problems we were having with the new laptop filtering everything below 100 Hertz. Finally used another laptop and it was fine. Later, we installed new ASIO drivers on the new laptop, and the problem was solved!

I recently received some PMs asking me about why some industry veterans  said they don’t believe in the “house curve philosophy” because “every car is different”. 

Obviously, I can’t speak for them. 

But I was reminded of an article I read a long time ago by a pro sound guy named Bob McCarthy. I looked it up, and I went ahead and ordered his book –  On Sound System Design –  and here’s a bit of what he said in the article:

“Where do you aim the speakers in a lively hall? At the people. And in a dead hall? At the people again. In what kind of hall do we intentionally aim sound at anything other than the seats? None that I have ever been involved with. Do we approach this differently for pop music than speech in a house of worship? Do loud shows need to aim away from the walls while quiet ones don’t?

This might seem like a silly line of questioning, but I am bringing this up to make a simple but important point. Sound system engineering is not about the room. It is about the sound system.”

He goes on to make some excellent points, some specifically for pro, some I’m going to test out in my own car – but the gist of it is this:

A house curve approach isn’t an approach of where to set the sliders. That truly would ignore differences from car to car. 

A house curve is a frequency response target. You can change that target for personal preference or system deficiencies (or because you don’t want to rattle one trim panel you aren’t going to fix the rattle in, I guess), but you should start somewhere. 

Speaker designers – and I mean speaker-system designers, not speaker driver designers –  start somewhere. They have a goal in mind. They select drivers and build crossover networks and add attenuation and reverse polarity in the wiring in order to reach an objective. 

Room-correction-algorithm engineers – the guys at Dirac and Audyssey –  have a goal in mind. They use EQ and phase processing and delay and level to reach an objective. 

Pro audio guys have a goal in mind, whether they have quantified it, or they know it when they hear it. 

When we design, install, and tune a car audio system, we are acting as speaker-system engineers and also as manual room-correction engineers. If we want to avoid quantifying anything, and simply “know it when we see it”, we can do that to try to preserve the mystery that only a few of us can possibly understand. 

But I don’t believe it’s a mystery. I don’t believe it needs to be and I don’t believe it should be. 

Now, I will meet you halfway. I completely believe that you can have two audio systems which, when measured with all speakers playing with a 1/3-octave RTA, measure the same and sound very different (both tonally and from a stereo presentation point of view). I think that the deficient system can be improved without changing its gear, through better tuning beyond what you see with a 1/3-octave RTA with all the speakers playing. (It might also benefit from better product, of course.) That doesn’t mean much, though. I think that’s just misusing a tool, and I’ve written about the early measurement mistakes I made in this regard. I also believe what Dr. Floyd Toole has said about measurement, which I will paraphrase as “if you want to know why two speaker systems sound different, you need more than 1/3-octave resolution”. There is a question about whether or not you need greater than 1/3 for tuning, as opposed to speaker-system R&D, and I prefer to look at measurements in resolutions higher than 1/3-octave as an educational activity – but I digress. 

The use of a tuning process that includes a target curve is intended to efficiently get you, as a beginner or novice, to the point of fine-tuning without a lot of wrong turns or wild goose chases. 

Now that said, I can’t tell you how many beginners or novices have heard a car I’ve tuned to a target curve, and been unable to find anything to change or tweak afterwards. It was better than anything they’d heard in a car. I could, though. I could show them something to change and improve, and I would explain what I would do. I believe that after those folks listened to fifty or so tuned systems, they would start to notice where to improve as well. It takes practice. 

So, if you have a mechanism that works for you, God bless you. We have too few people in our industry who can tune a car, and we need them all. 

But using a target curve doesn’t ignore the acoustics of the vehicle, or the problems in the vehicle, or the rattles in the vehicle. It’s an objective. For some who’ve never tuned a car, having an objective demystifies the whole thing. 

Wouldn’t it be great if everyone could tune a car by ear and not need to do measurements? Well, in the words of Charlie Brown, “Wouldn’t it be great if potato chips were good for you?” I can’t tune a car by ear. I can’t. But I made a good living tuning a car with test equipment – ONCE I understood what the hell I was doing and stopped spending my time going down rabbit holes. 

I’ve been collecting target curves. I have one, Andy Wehmeyer has one, Harman probably has that same one if they remember, JL has a couple, and on and on. The thing is, until you understand how to reach one of them, you can’t reach any of them. The interactions of phase and time arrival and acoustic summing are what make tuning harder, and if you don’t sort those out, you can’t hit any curve at all. It’s not the shoes. 

I have a tuning process. Some others have tuning processes. If you follow one of the good processes, you’re sorting out the interactions of phase and time arrival and acoustic summing, even if you don’t know it! The point of a good tuning process is to help you correct the problems in that car. 

So, I don’t teach a process with a target curve in it because I ignore that cars are different. I don’t ignore that cars are different, but I also don’t ignore that humans  – and our hearing systems – are very similar. Our basic audio designs need to have a commonality among them so that we can make this into a business. In the words of a forgotten comedian, “we can’t stop serving baked potatoes because the guy with the recipe and the tin foil went home!” After we bake the potato, we can put the stuff on top that we want – but we need a potato first. 

What do we need to measure?

I recently watched a seminar on advanced tuning delivered by my friend and colleague, Steve Turrisi of JL Audio. 

I greatly enjoyed it. Steve takes this sort of thing very seriously, which is all too rare in our field, and any technician would be well-served to watch this two-part presentation. 

Plus, I think that “tuning” is “measuring the system and correcting the problems”, as I’ve heard it described (correct attribution is Andy Wehmeyer’s friend Ulrich, I believe). And it’s hard to describe Steve’s class as anything other than that.

That said, there were a few things I was unclear about, and a few things I disagreed with. 

Soon after, I talked to Steve over the phone and cleared up the biggest questions I had about the presentation.

Fundamentally, I will describe Steve’s advanced process as “measure the impulse response, analyze the resulting amplitude and phase responses, then correct for amplitude and phase both”. Which sounds sensible, right? However, in the commercial car-audio environment, it’s not something I would rush into.

I also recently read Sound System Design and Optimization by Bob McCarthy. It’s all about pro sound, but it’s an interesting read for anyone interested in sound, even if around half of it isn’t applicable to what we do. Bob was a pro sound engineer who then got involved in the development of an impulse-response measurement system at MeyerSound. Steve acknowledges using this book as a resource often, and its influence on the content of his class is apparent.

How impulse measurements work

I’ve said before that fundamentally, real-time analyzers are analog AC voltmeters, and they measure the voltage at multiple frequencies at the same time. We can call this the amplitude, or the magnitude, of the content. 

An RTA doesn’t measure time or phase, just amplitude. 

Impulse-measurement systems measure the actual frequency response in both the amplitude and phase domains. This means that an impulse measurement system can tell you a numerical value for the amplitude and for the phase of a speaker system at any given frequency. 


Phase is measured in degrees. Once you get to 360 degrees, you have traveled in a circle, and are basically back to zero degrees of phase. Engineers often refer to phase changes as “rotations” because of this aspect of phase. 

We don’t hear differences in absolute phase. Play one speaker, and play with the phase. We can’t hear a quality difference. What does that mean? It means that most people can’t hear the difference, and those who could reliably hear a difference couldn’t reliably decide which one was “better”.

But play two speakers at near the same level, and if the sound from those speakers is out of phase at any frequency, there will be cancellations at that frequency – and those cancellations are usually audible. 

Phase nonlinearity vs phase cancellations

When amplitude changes, phase changes. We aren’t sensitive to absolute phase, but we are sensitive to phase cancellations. Below a certain frequency, we are much more sensitive to these cancellations – that frequency seems to be around 1500-2000 cycles, but some believe it is slightly higher.  

When can system-caused phase cancellations happen?

  1. Two drivers playing the same note – in the transition band of a crossover filter
  2. Two drivers playing the same note – front and rear speakers different distances
  3. Two drivers playing the same note – left and right speakers different distances
  4. Two speakers playing the same note – inadvertent reversed polarity
  5. Reflections, when the amplitude of the reflection is nearly as high as the direct sound, but the distances traveled are significantly different. 
  6. When the OEM system is performing phase processing, and we didn’t notice. We assume all signals are in phase right and left at all frequencies, but that turns out not to be the case. We will consider this an OEM Integration topic, and only address the first five. 

Let’s look at each of these in turn. In all five cases, the sounds must be nearly the same loudness.

  1. The transition band of a crossover filter: This is where two speakers overlap. The problem can be caused by crossover filter selection, but if the two speakers are different distances from the listener, the problem is magnified. The first step in preventing these cancellations is thoughtful crossover-filter choice. When choosing active crossover filters, I use 4th-order Linkwitz-Riley filters exclusively. The steeper filter slope means the transition band is very narrow, they are phase-neutral once levels and arrival times are corrected for, and they sum flat (whereas 4th-order Butterworth filters don’t sum flat, because they are literally a mistake). Steve advocated an approach which I will call what I’ve called it for years – “crossover whispering”. There is a lot of time being spent in this process deciding what crossover filters to use based on the interior of the car, and I’m not convinced there is any commensurate benefit.

2. Front and rear speakers playing the same notes, but from different distances: This is probably one of the reasons I’ve hated rear speakers for many years (my formative years were before I knew how to use delay). However, rear speakers are the best way to make a system louder, and that’s important to a lot of us and a lot of our clients. Just make sure the rears are time-corrected, or make sure they aren’t loud enough to ruin your stage. Many upmixer systems use rear door speakers primarily for stereo sound for rear-seat passengers, and rear-deck speakers or D-pillar speakers for ambiance channels. 

3. Left and right speakers playing the same notes, intended for the center of the stereo stage, but different distances from the listener: These cancellations only affect sounds intended to appear in the center of the stage, which is a lot of sounds – including the drum kit in most recordings. Using delay to correct for these differing arrival times became a very popular tuning method around ten years ago, but it is only correcting the problems for one unique listening location. The lower you go in frequency, the bigger this problem gets, and the higher you go, the less pronounced this problem gets. 

4. Inadvertent reversed polarity: It’s not as apparent with other speakers as it is with dual-subwoofer systems. Test your speakers with a polarity tester before tuning. It may seem like a rookie move, but you’d be amazed at how often this is the problem. Apply the science of quality control management to connecting speakers, and the statistics say that the odds are not in our favor. 

5. Reflections: These are often the result of speaker locations, but it’s worth remembering that in many cases reflected sounds which are nearly as loud as direct sounds also travel a very similar distance as direct sounds, so the comb filtering effect is not as bad. When the distance traveled by a reflection is significantly greater, some attenuation also results. That said, most installations use factory speaker locations and accept whatever reflections are present as a result. 

Delay is how we often address 2 and 3, and how we partially address 1. 

Tape measures – do they work to set delay? 

Steve stated that the best way to set delay is to measure phase, and set delay to correct it, rather than to use a tape measure. In our conversation, I double-checked that he wasn’t saying that tape measures don’t work for predicting delay. I will elaborate in a moment, but please understand that “tape measures don’t work” isn’t Steve’s position.

Steve’s position, as I understand it, is that you can correct for phase problems best after measuring the phase response of each driver, and comparing the various amplitude and phase responses. 

This sounds reasonable. However, I find this method for car stereo to be difficult to perform, very difficult to teach, and inefficient compared to other approaches which start by measuring distance.  

The big problem with delay as a phase cancellation correction method is that the delays you need are all interrelated – left and right drivers, front and rear drivers, and adjacent drivers (subs and midbass, for example, or mids and tweeters).

If you start setting delay on a basis other than distance, you can easily end up chasing your tail, as each delay change you make to address one problem causes other problems (either with adjacent drivers, or rear drivers, or drivers on the other side). This is not a problem if you initially set your delays based on distance. Now the relationships in the transition bands are correct, and the relationships in the off-center listening position are correct. 

Far better, in my mind, to accept distance as the primary influence on 2 and 3, (front/rear arrivals and left/right arrivals) and as a major component of 1. (Transition-band cancellations are made worse in most instances by different arrival times). Otherwise, you end up juggling an awful lot of interrelated problems. 

Measure phase, or just amplitude?

I’ve messed around a bit with measuring time and phase, microphonically and also electrically. I rarely do it. The biggest reason is that when I do this, I’m in a production environment, and I don’t have the time. 

In answer to a question, Steve indicated that this can be done in 2 hours once you have some practice. I think he’s being very optimistic there. I might be wrong, but I’m pretty confident that it would take a LONG time practicing to get an impulse-response measurement-based tune down to two hours. I think it would take at least a full day for quite a while, and that’s a tough chunk of time to monetize. The number of guys in this industry who could do this consistently in 2 hours, I bet I could count on the fingers of one hand (I wouldn’t be on the list).

This is a commercial activity, not a contest. At the other end of the spectrum, we’ve got professionals suggesting tuning methods which don’t even use an RTA – which use an SPL meter and a bunch of test tones – because “many shops have loud compressors and impacts going and can’t use an RTA with any degree of accuracy”. That is not my position. We need to measure sound, even if we have to wait for our window. 

 My approach is this – if there’s a phase cancellation we can hear, there’s a phase cancellation we can SEE when we measure the system’s amplitude. If you can’t see it in the amplitude response, it’s not much of a cancellation. I’ve watched Steve explain phase cancellations twice now in trainings, and each time he shows the summed amplitude measurement, the cancellation is plain as day on the amplitude measurement graph (the same measurement you would see on an RTA). It seems really apparent to me that you can still address that cancellation and confirm the result in a quantified manner without the actual phase quantification. 

Play Channel A. Play Channel B. Play both at the same time. If there’s a dip with both, but not with either, you have a phase cancellation problem. 

Now, if I were designing a speaker system for an OEM project, or a home speaker, I would TOTALLY use the full-power impulse-response approach. Systune, SMAART, and REW are all pieces of software that can do it. They are not simple, and I speak as someone who spend a lot of unbilled hours over the last 15 years figuring out how to tune cars with test gear. I don’t think I want to start trying to convince people to sort this out on their own time.

We have a long road of adoption just to get one-channel real-time analyzer systems into most shops and get technicians comfortable with them. That’s my goal. 

Now perhaps I’m wrong. As Bill Murray said, “If I’m wrong, I’ll go to jail – peacefully, quietly, I’ll enjoy it”. But for now, my belief is, learn how to get good results with one channel, with amplitude measurements, with a defined process, and in a minimal amount of time, FIRST. When you feel you’ve reached the limits of that method, then start looking at where you want to go.