# Sticky  The Essentials of Sound Quality IMHO



## ErinH

*Foreword:*

For a while now I’ve been in this hobby, learning and falling on my face with failures in understanding and application. Over the past few years through competition, get togethers, talking with friends, research and testing drivers for my site (in my signature), I feel like this stuff has finally started to really click. 

First and foremost, I'm a fan of music. All sorts. I'm not in this hobby or a fan of posting data just because. Like many of us here, I've been jamming out since a toddler (I have the video of me dancing to Michael Jackson as a 3 yr old to prove it... and I had some awesome moves, lol). Mix that with a desire for learning technical things, it seems like a no-brainer that I'd really have a passion for this hobby because it really does force you to study if you want to excel in it. 

What I notice on the forums is that a lot of people starting off tend to get confused by speakers and tuning… and it’s not easy to weed through all the mess or ignore the bad advice when you don’t know the person it’s coming from. There are a lot of threads where people tell you what to do but they don’t really explain why. Typically, in my opinion, a lot of this is just restating what someone else told us. I’m no stranger to this. This thread is not to be considered an all encompassing summation on sound quality. I am not the foremost authority on sound quality and speaker reproduction and I don't pretend to be. That said, I've spent the majority of my free time over the past 5 years researching and studying acoustics and providing 3rd party data for various drivers and products. I'd like to consider myself as an educated nooB... because I'm always learning, I'll never be an expert by any stretch. Hopefully, this thread will shed some light on the why and how’s of tuning and help you understand why speaker measurements matter. 

Sprinkled in with some technical info are my own trials and tribulations; what matters and what doesn’t. What is a good midrange? Why do I need to cross my midrange at 3khz instead of 8khz? Why is time alignment so important? How do I use time alignment and how is it different from polarity? Now, of course, I can’t speak for all situations and I won’t say that the things I advise against can’t work. 

*Simply put: the following is in my opinion best practice advice. *If you talked to me in person, everything I say below would be said to you regardless of who you are or what gear you run. Therefore, this thread is to be considered an attempt by an enthusiast at listing the basics of speakers along with some tuning tips/methods based on years of experience and reading on various internet sites. I’ll cite sources where necessary but since this is built on years of reading I can’t really give credit where it’s due in every instance because I don't know the first place I read something. 
I can say various forums, Linkwitz, Zaph, Toole, JBL/Harman, Keele, Geddes, and my friends and mentors in this hobby… all of them are my resources. And in many ways, I am my own resource through trial and error.


As of this writing, this thread is incomplete. I will update as time permits but wanted to get a jump on it so people can start reading and possibly go ahead and try to apply some of the content herein as I go.


*Cliffs: *
If you need cliffs already then this thread isn’t for you. You’re going to have to do some reading so consider this a warm up. 
(bet you didn’t see that coming)


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Caution... SCIENCE!!! *

Stereo is rooted in science. So, whether you believe science has a place in furthering the hobby is a moot point. That said, the science referenced herein is more of an off-the-cuff attempt at explaining the things we are concerned with than it is a doctoral candidate paper. I have made every effort to back up information with links or real-world measurements. If something doesn’t look right, contact me. 

Some aspects are left out on purpose; some things I already know aren't mentioned. This is done for the following reasons: a) I can’t tailor my attempts at helping the DIY audio community to those who only wish to nitpick each and every aspect of this presentation nor will I exhaust myself replying to potential comments in advance by providing rebuttals in said fashion, and b) I cannot possibly cover - in detail - all the things I’d like to without making this thread just plain overwhelming. 

Again, consider the purpose: to really make things as concise as possible without leaving out important information.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Basics of Drivers:*

Response Types: 
Frequency
Polar
Power

Beaming
Acoustic (Natural) Roll Off
Distortion:
Linear
Non-Linear



*Tuning:*

Setting Gains
Crossover Frequency and Slope
Time Alignment
Phase/Polarity
Levels
EQ


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Basics of Drivers:*​ 
The following sections are intended to help you understand the important aspects of a raw driver and data which will then lead to helping determine nominal crossover points.




*Response Types:*

*Frequency Response:*

The measure of frequency (Hz) vs amplitude (dB) across a given range 
Multiple frequency response measurements at varying axes are taken to show how the driver behaves in all directions. Those axes are comprised of the following:
[*]On-axis response: Speaker/system response when the listener is directly facing the speaker (SINGLE POINT).
[*]Off-axis response: Speaker/system response when the listener is anywhere EXCEPT directly on-axis (typical range is 15 to 90 degrees off-axis in car).​
Ideally, the frequency response will indicate no hot or cold spots in response as the listener/measurement mic moves around the speaker
The driver or speaker shall maintain it’s general response, other than a decreasing output level as the frequency gets higher

Below is the frequency response on-axis (0 degrees) and off-axis (30 and 60 degrees). Note the graph legend in addition to callouts.

​







*Polar Response:*

Another way of relating frequency response using a particular frequency or frequencies, mapped out in a polar pattern, representing the directivity of a speaker at varying angles (ie; as you move from 0 degrees on-axis to any angle off-axis)

Below is a polar response example of a speaker modeled in LEAP. The model was derived based on a simulated horizontal axis measurement ranging from 10hz to 1.28khz. You can see that as the frequency increases, the radiation changes from omnidirectional to more directional. 

​





*Power Response:*

Single measurement which is the *sum total of both direct and reflected sounds* and is a representative example of what the listener will hear at a given location

*Typically an average of multiple measurements in the listener’s head area*

An “ideal” power response is one with no significant peaks or dips caused by irregularities from any single axis of measure 
The total response shall roll-off smoothly. The rate of roll-off is a matter of directivity index. 
*When you RTA and average the results, power response is what you are measuring.*

The picture below shows a speaker measured both on and off-axis in varying axes. The subsequent picture is an average of all these measurements, which results in the power response measurement.



​


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Beaming:*

Beaming is kin to the acoustic low-pass (discussed in the following post).
Beaming is a function of the effective driver size: the dispersion narrows as the wavelength (frequency) becomes smaller than the size of the drive unit.
Note: Effective driver size is taken from 1/2 surround to 1/2 surround.
*All drivers beam! * Though, some drivers are built to extend further on a given axis than others, but often at a cost (sensitivity, cone breakup, etc). 
Frequency response measurements illustrate beaming easily; *look for the on and off-axis responses to diverge. This is your beaming point.*
The formula to _approximate _the beaming point is: 0.5*(Speed of Sound)/(Driver Size). 
Note: This is for the case of a conical driver. The same formula applies for square/rectangular/oval shaped drivers. In this case you simply use the dimension to determine the vertical or horizontal beaming point. For instance, a 5x7" driver doesn't have a uniform polar pattern; it will beam at the 5" and 7" dimension so therefore you will have a different response on/off axis vertically and horizontally (depending on how the driver is oriented to the measurement device).
Below is a table illustrating approximate beaming points for a round driver with a given cone diameter: 
 




In the example below of a 4” midrange with an *effective diameter *of about 3”, beaming occurs approximately at 2khz.

​


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Acoustic Rolloff:*

Acoustic high-pass determined by:
Fs – Free air resonance of a driver
Qts – Bandwidth of driver resonance


Fs and Qts can be obtained by manufacturer Thiele-Small specs or derived by an impedance chart if available:

​



Acoustic low-pass:
Determined by things such as motor force, suspension, and (namely) inductance. 
If the inductance (essentially resistance to current, for our purposes) is high, the output of the speaker is less, regardless of the axis.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Distortion:*

There are two types of distortion:

*Linear**:*
Any divergence from flat in the frequency response would be considered a form of linear distortion. 
No driver is _completely _flat. Though, typically below the beaming point, drivers are _fairly _flat. Outside of beaming is where breakup occurs and is typically the area where linearity in response is compromised.
Does not change with volume.
This is typically used to determine low-pass values as cone break-up is the worst offender after the beaming point.


Below is an example of *poor linear performance*. Notice after the beaming point of this 4” fullrange driver how it’s on and off-axis measurements differ?





Below is an example of *good linear performance*. Notice after the beaming point of this 4” driver how it’s on and off-axis measurements follow the same trend?






*Non-Linear**:*
Distortion that changes with volume.
Referred to as harmonic distortion, THD, etc.
This is typically used to determine your high-pass values.
While argued as to it’s merit of audibility, a good rule of thumb is to avoid the 3% THD range (*Note: 3% THD is 30dB down from the fundamental*).


Below is the measured distortion of a 4” fullrange driver. Note the THD has reached the 3% mark at about 150hz. In addition, high frequency breakup exceeds 3% THD at approximately 3khz.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Tuning Advice:*​
This section covers some of my “lessons learned” with tips based on my experiences tuning my car and helping tune others. As with anything, I should caveat by saying your mileage may vary. Not everything that works for one will work for another, but maybe some of my failures can lead to your success. 




*Resources:*
Here are a couple recourses I'll reference in the body of text below. Go ahead and check them out. And you may want to go ahead and burn the CD linked below. 


Test CD:
This disc was given out at my GTG in November. There's a lot of tracks on here but for the purpose of this post, we'll only focus on the pink noise tracks in the last half of the disc.
Time Alignment Site:
Robert Mcintosh (Pockets5) helped me make this site. The goal was for someone to easily punch in measurements of each speaker to the listening position and get the numbers needed to delay each speaker so it arrives at the listener at approximately the same time. It's not perfect, but should cut about 80% of your time off the manual way.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Setting Gains:*


*Intro:*

Setting amplifier gains can be as complicated as using an oscilloscope with a dummy load or simply using a $10 digital multimeter (DMM) and Ohm’s Law. A scope will let you know when the output signal is clipping (distorting). The DMM method assumes you know at what voltage for a given load the output signal will clip. I will discuss the latter.
Since most (brand-name) amplifiers will do their rated, continuous power per spec, all you need is to connect the multimeter to the speaker output of the amplifier and raise the gain until you achieve the required output voltage while playing a tone through your cd player
Ohm’s law is used to determine amplifier output voltage. An online calculator can be found here: Ohm's Law Calculator
Use an attenuated tone, such as -10dB or -5dB as this will allow additional volume for ‘quiet’ tracks. 0dB tones may be used in lieu of attenuated tones IF you listen to heavily compressed music.
Tones can be downloaded here: Realm of Excursion

The tone should be in the bandpass of what you are setting the gain for. For example:
If you are setting subwoofer amp gains, use a 40hz tone. 
If you are setting midrange amp gains, use a 1khz tone.


*Setting gains with a DMM:*

Here is a quick guide to properly setting your system gains using nothing more than a DMM.
Reset all your cd player DSP features, balance, fader, etc to ‘default’.
Disconnect all speakers to keep from damaging them (and your hearing).
We will assume your cd player puts out clean, undistorted signal at near maximum. Play your tone (-10dB suggested) and increase the headunit volume to a click or two below max.
Knowing the amps’ rated power at a given load (ohm) obtained from manufacturer’s spec (assuming they are a reputable brand), use Ohm’s law to determine voltage needed. 
For example, with a 4 ohm load and 100wRMS rating, the output voltage should be 20VRMS.
Set the DMM to “VAC”, which is AC Voltage.
Connect the leads of the DMM to the amp’s appropriate speaker output and increase the amp gain until you achieve this voltage
Repeat for each channel.

You’ve now set up your system to maximize output and performance with a wide variety of music.​
Keep in mind, when level matching, you can use your decrease amp gains as opposed to decreasing output level on your DSP. Though, I choose to maximize the gain structure up the chain as much as possible and make adjustments at one place (such as a DSP) just to keep things simple.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*DSP Basics:*


*Intro:*

Most DSPs have these same features:
Crossover
Polarity (0/180 deg phase)
Time Alignment – ability to delay signal so all channels reach the listener at the same time
Levels – essentially gain for each output channel
Equalizer


The difference in DSPs is usually small, in regards to features. The difference typically comes in the resolution of adjustment of these features or (most notably) the interface. Here are some examples*
*Note: This is NOT intended to be a summary of all features of every DSP. I’ve just chosen specific examples.:
Resolution of adjustment:
Alpine PXA-H800 uses 0.5dB increments for levels; Helix DSP uses 1dB
Alpine PXA-H800 uses 1/3 octave frequencies for crossover points; Helix DSP has 1/12 octave (in other words, Helix has a whole lot more flexibility); ARC Audio PS8 allows you to type in any number you wish
There are some exceptions where a DSP has something different (Helix offers adjustable phase in 15 degree increments on the subwoofer channel)

Interface:
Most DSPs are now controlled via a computer interface, though some are still controlled directly through the DSP (ie; Pioneer P99 or Alpine H800)
This is user preference. Some prefer to have a laptop and some prefer to be able to tune ‘on the fly’.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Crossovers:*

Crossover Point/Slope should be evaluated as a set. Using just a number and a willy-nilly slope isn't exactly a good method to use. There needs to be some reason for setting these values. The frequency dictates at which point you want to start rolling the speaker off. The slope will dictate not only absolute and relative phase but also attenuation. 

Crossovers are made of both the frequency and the slope you use. So, let's look at that...

*Crossover Frequency*:

Namely, there are four aspects I am considering for low/high pass crossover values. Each of these are discussed in the driver basics section above but I will recap.

Beaming. Where does the driver's high frequency response begin to separate from on/off-axis? Using the _Polar Response and Beaming_ reference above, you can see some of the math behind it and reasons as well.
High frequency breakup. All speakers begin to break up at some point. It's just an effect of the cone material and shape. Typically drivers don't break up until about an octave or so above their beaming point. Some drivers control break-up better than others through cone design. What you want to avoid is the area where the break up is severe enough to be heard outside of the crossover point. BUT, since this breakup usually occurs above beaming, you should be crossing before breakup occurs. 
Low frequency distortion. You should know what I mean here... take a tweeter for example. If you cross a tweeter at 500hz, odds are, you're going to get all sorts of distortion and ultimately fry it. The general rule of thumb seems to be to cross the tweeter at 2*Fs (Fs=resonant frequency of the voice coil). BUT, this isn't a one-size-fits-all solution. Some drivers may have a low Fs but may not be well suited for a low crossover (ie; some drivers have an Fs of 700hz but I wouldn't run them at 1400hz full tilt with any slope). The key here, really, though is matching the dispersion pattern as well as you can to the driver before it. In the case of a 3" mid which beams at 2.5khz, you'd want to cross your tweeter somewhere in this area to keep from having a null at the crossover point, not fixed by any phase/polarity changes. Again, see the post I mentioned in #1, above. 
Natural Rolloff. On the low frequency end, the driver rolls off naturally. This is dictated by the Qts and Fs (the Qts dictates how much the rolloff is and Fs tells you at what frequency it occurs). These 2 pieces of information can be found in an impedance plot. Most people will try to set their crossover point and slope to essentially follow that same natural rolloff so you combine both the acoustic rolloff of the driver with the electrical crossover and don't alter the phase severely. 



*Crossover Slope**:*

The slope you use depends on the following:

Level of attenuation needed. 
This should be self-explanatory. Basically, the steeper the slope, the faster the rolloff. 
Phase: 
I can't say enough how non-trivial this is. There's the "set it and forget it" method which can be made to work or there's the "spend a lot of time on it until you get it as good as you can" which I propose. The latter option will save you a lot of headaches in the future.
Think of two sine waves. If both are in phase, they play together and the amplitude is increased. If they are out of phase by 180 degrees, they cancel each other out. Your goal at the crossover is to essentially allow one speaker to 'carry' in to the other, without evidence or calling attention to anything in the crossover region. You want in phase sine waves... for a lack of better analogy.

*What you will HAVE to do is EXPERIMENT. * Using the above crossover frequency info, pick a point that makes sense to _start with_. From there, change the slope of one driver with respect to another or even both drivers. Take notes. Which settings sounds better? What happens when you change the polarity of the tweeter but leave the midrange polarity the same? What happens if you change the slope of the midrange from 24dB/Octave to 12dB/Octave? Then try flipping it's polarity. 

FWIW, I typically shy away from anything less than Linkwitz-Riley 12dB/octave (LR2 - "2" for second order). The reason why is pretty simple: power handling. If you know me, you know I like some volume. Then there's the case of out of band EQ'ing that may be necessary and if you don't have this ability with your DSP, it's a problem. A steeper slope keeps less information from overlapping. Especially the high frequency content you don't want playing due to beaming and breakup. And for those convinced that low order is the best way to go in a time domain case, keep this in mind: although a 6dB/octave butterworth has no group delay, it doesn't mean your summed response won't result in such. What you care about is the total response. Not just a resistive load single crossover. 
​
*There's no guaranteed slope or crossover point that will work for you. The reason why is because the crossover network that works best is dependent on the characteristics of a single driver (breakup, natural rolloff, distortion) AND the crossover point/slope of the driver you are mating it to!*

You will have to experiment. You can use your ears or a mic. I think it's useful to try using your ears as it makes you a more attentive tuner. The mic can do a lot of things great, but it can also mislead you as well. So, you have to really listen in between adjustments made via mic. One thing I suggest is to narrow the best settings down to a few. Then from those, listen. Use pre-sets if you can and switch back and forth between them.

I also recommend using correlated pink noise to help you set your crossovers. The CD link I provided in the Resources section has this. 
Mute everything but the mid and tweeter (or whatever it is you're setting the crossovers on) and listen to them as a pair while you alter the crossover settings. Listen for the sound to be fuller and as one. If you hear more of one driver than another keep in mind this may be simple levels, which we will address later. For now, focus on trying to listen for cohesion at the crossover point, though. If you're crossing your mid and tweeter at about 3khz, don't worry about what's going on at 500hz or 10khz. Pay attention to the 2-4khz range and see how that area changes as you adjust your crossovers' settings. Something else you can try, which is what I do, is to use pink noise 'tones' at the frequency around the crossover point. This is really helpful with time alignment which I'll get in to later as well.

Something else I've noticed is when the two drivers are out of phase, I'll hear a phantom image in front of me; away from the drivers I'm tuning. It's a total mind trip. 

Finally, your goal is to get it as close as you can. It likely won't be perfect, because most car installs require a large delta between drivers and thus there will most likely be lobing (this is again why #1 in the _Crossover Frequency _section is important) but what you should find is after spending some QUALITY time doing this, you'll have a much fuller soundstage and more cohesion between drivers. Time alignment can be used to further tweak this. That's for later, though. *Right now, focus on getting the crossover point and slope as dialed as possible between each set of drivers. *



*Additional Crossover Notes:**:*

While the previous section focused on a single driver characteristics and understanding them, the import factor is the implementation of a driver in a system… with other drivers. 

Good power response should be your goal but it’s not easy.

For example, mating a 1” tweeter to an 8” woofer isn’t as easy as mating a 1” tweeter to a 6” woofer. Why? Because the tweeter doesn’t have to cross as low to match the 6” woofer’s dispersion. If you tried to cross the tweeter to match the 8” woofer, you’d increase the tweeter non-linear distortion considerably. Conversely, if you increased the 8” driver’s low pass filter you’d likely exceed it’s beaming point and the dispersion wouldn’t match between the mid/tweeter.



Let's look at another example using a typical 2-way setup:

A typical 2-way setup consists of a 6.5” woofer and ¾” tweeter. The following is a generic analysis based on typical drivers:
¾” Tweeter Crossover:
THD reaches 3% at 2khz (raw driver)

6.5” Woofer Crossovers:
Most 6.5” woofers with moderate linear throw (3-5mm one-way) have 3% THD by 80hz.
 Fs/Qts will drive the enclosure which drives Qtc which drives the crossover as well. A lower value Qtc means less cone control and more attention should be placed here to not damage the suspension or fry the voice coil.
Beaming will occur by about 1.2khz with an effective diameter of 6”. Based on this, you can assume a 2khz would suffice to keep the driver’s on and off-axis response fairly well matched. Much higher above this and the separation is more severe.


Therefore, a nominal crossover point between mid and tweeter in this generic example would be in the 2khz – 3khz range, in order to mitigate woofer breakup and beaming but also to lessen tweeter distortion. Crossing the woofer above 80hz on the low end will help mitigate it’s distortion at higher volumes.​


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*How We Hear: Interaural Time Delay and Interaural Intensity Difference:*

Our ability to determine location of sounds is rooted in what can be boiled down to a few key points (and for the sake of this presentation will be kept very basic):

Interaural Time Difference (ITD) – Localization determined based on the time a sound takes to arrive at each ear
Interaural Intensity Difference (IID) - Localization determined based on the intensity of a sound arriving at each ear 
Also known as Interaural Level Difference (ILD)


Each of these contributions can be roughly boiled down to the following*:

ITD cues contribute mostly up to approximately 800hz and then share with IID through approximately 1400hz where IID then takes over
* This is a summary of various research. There is additional useful information here: ITD and IID Cues


Below is a graphical representation of ITD/IID and the area where they have some overlap:






It should be noted this section does not consider any sound source other than laterally in front of the listener. For further reading on the topic, also consider the impact of the Head Related Transfer Function (HRTF) and Head Related Impulse Response (HRIR) as the “Cone of Confusion”.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Time Alignment and Levels*

*Basic Audio Terms:*

Knowing the basic definitions of soundstage can really help you understand why certain DSP aspects are needed and how they can be implemented. So, let’s look at one of the most basic and important aspects of soundstage:

Balance: Center and Acoustic Boundaries
To properly balance your system is it important to know your stage boundaries.
In acoustics, these boundaries are defined as the left-most stage and the right-most stage. 
You know a center is the point in space between two lines. Therefore, *the acoustic center should be placed mid-way between your acoustic boundaries. The center shouldn’t be an artificial point on your dash: it is the acoustic center of the acoustic left and right boundaries.*
It’s important to note that not all recordings have the same characteristics; some may have a vocalist to the right of center. Others may place a kick drum to the left side of the stage. Judging your center by a sole vocalist or instrument can be misleading if you don’t know for sure where that location on the stage really is supposed to be. So with that said, it’s a good idea to use correlated pink noise or a centered narrator (both provided in my Test CD) to determine center.

Depth, Width, Height
Each of these have importance as well, but I feel focusing on the center and putting that in the appropriate location with respect to your left and right boundaries will allow the other aspects to “fall in place”





*Time Alignment and Levels:*

Typically, in a car, the listener is positioned so that none of the speakers are the same distance from him. 

The below illustrates a standard car setup with only a single speaker on each side creating not only a near-side biased stage, but also incoherency in the sound due to poor time arrival differences. There is a clear stage boundary, but no focus and no way to really pick out a center.




In the example above, the Left/Right speaker delta is 11 inches. In time, 11 inches is approximately 0.808 ms, which means that the left speaker’s sound arrives 0.808ms before the right speaker’s sound. This would create a left side bias due to *ITD*.

The SPL at your seat is approximately 2dB higher from the left side. This would also create a left side bias but due to *IID*. 

The combination of ITD & IID would drive an overall stage that sounds squished to the left; there would be very little focus if any and no well-defined acoustic boundaries or center.


Given what we learned about our hearing based on ITD & IID, we can use our DSP’s time alignment and level features so that each speakers’ sound arrives to you at the same time as opposed to the nearside sound arriving first.
Of course, it should be noted that this really assumes properly acoustic polarity has already been accounted for. A simple 0/180 degree “phase” swap. In some cases wiring your speakers up in electrical polarity the correct way doesn’t necessarily mean you’ll get the correct acoustic polarity. Phase will be discussed more in the following section.

Time Alignment:

Time Alignment and Level Matching’s use is to give the impression that all sounds arrive to the listener at the same time based on ITD & IID. In short, if you want to simulate a sound arriving later from one speaker than another you add time delay and decrease the output which you want to push away. 

Time Alignment can be derived in simple (tape measure) or complex (measurement system with loop-back) manners.

The simple method is:

Use a tape measure to determine the distance of each driver from your listening position. 
Measure approximately from the speaker voice coil (if you can’t see it, add the appropriate amount to the cone) to the center of your ears.
Use the following site to determine the approximate time delay needed to ensure each speakers’ sound arrives at the same time: 
http://tracerite.com/calc.html

The complex method would be one using measurement gear to measure the impulse response of your speaker. But, this will not be discussed due to time constraints. Maybe later, though. 




Levels:

Given that levels are more dominant in regards to staging in higher frequencies, tweeters are typically the most impacted drivers when it comes to level setting between left and right. Additionally, in acoustics, each doubling of distance would result in the halving of SPL; in this case the left side’s output would ***likely be stronger.
***_Note_: If the left speaker is aimed off axis, it would permit the higher frequency content to roll off a bit sooner thus helping to mitigate the IID bias to the left speaker. This is why some home speaker setups are aimed inward. Something to chew on.... 

There is no real easy way to adjust for levels via a calculation. I mean, there is… but real world things cause issues here:

Did you set the gains the exact same?
The environment will take a flat speaker response and make it not so flat.
Are the speakers the exact same sensitivity?
Bigger concern between driver types such as a midrange and tweeter, as opposed to two of the same midranges.


The best way to start off adjusting levels is to do so using pink noise and listening to high frequency content (>2khz) and adjusting levels until you achieve a good balance between left and right stage. Another option is to use an RTA or Phone app that will approximate the SPL at your seat from each speaker.



Using Time Alignment and Levels to create a Balanced Stage:


The ITD aspect: 
Using the simple T/A method, you determine that the left speaker delay should be set to 0.80 ms. 
The IID aspect: 
You also have determined via SPL measurement that you need to attenuate the left speaker 2dB to match with the right speaker.

What you wind up with a more equal representation of left and right stage boundaries, resulting in more width and a more realistic center location for your stage. 


So, what you should now have is something *CLOSE* to this:





Another way to consider it is going from this ...




... to this ...







Recap:

Remember, when adjusting levels and time delay what you’re adjusting with one speaker is *RELATIVE *to the other speakers. You’re adjusting the nearest speaker(s) so that it sounds as if it has the same intensity and time of arrival as the furthest speaker. This would then sound as if the other speaker had moved.

Or think of it like this: you’re moving your ‘center’ with respect to the left and right boundaries. You continue to do this until your boundaries sound equidistant from the center vocalist/pink noise.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Equalization: A Primer*

So, we've gone over the basics of DSP. So far we've covered time alignment and level matching to help you tune. Another major aspect of tuning is using the EQ to help resolve problems. Now, some may say EQ isn't necessary. I say that depends: what is your goal? What do you want to achieve in a car? What speakers do you have, how is your install set up, did you pay attention to the driver basics section and how it impacts your crossover choice? 

Let's put it this way:
If you have a speaker that has few problems outside of it's beaming point then you're ahead of the game. If you cross the speaker before modal issues take over then that's a help. If you aren't doing either of these then you need to know that the speaker is going to cause you all sorts of problems and you're going to be going hard to work on the EQ to correct these problems. Compounding it more is the simple fact that some problems can't be resolved via EQ. You can't fix a cancellation null in a speaker due to surround or basket resonance. Not going to happen. You can't fix a modal issue in one axis and it not affect other axes. What about your installation... do you have resonance in an enclosure? If so, good luck correcting for that with standard 1/3 octave EQ. 

But, let's say you've got the system set up to the best of your ability (install and basic tuning methods already discussed). *When it comes to using your DSP’s equalizer, there are essentially two different methods you can use:*
*Ear*
Correlated Pink Noise or 1/3 Octave Pink Noise: these tracks can be used for tonality adjustments and centering of frequencies that may jump out of band

*Measurement*
SPL Meter used with 1/3 Octave Pink Noise or tones
RTA
Impulse


Each of these methods has some advantage over another, but there is no single ‘best’. What I find is a good mix of ear and measurement will net you the best response. 


*"Why can't I use just one of these methods?"
"Why not just use method only? "*
*Because of the human factor*: We don’t always understand what the data is showing us and misinterpret it, therefore creating more problems than we set out to fix.

"Measurements don’t lie, right?"
Well… sure, they tell you what they tell you. *But what if you are measuring the wrong way? *Doesn’t help you, does it? There are numerous tutorials on how to measure yet there’s still a lot of questions and potential for better ways to measure. Additionally, you may want to believe you fixed a problem because it’s visually not there, but what if you fixed the wrong thing? What if the bump at 1khz you knocked down was merely next to a null that can’t be fixed? So, you knocked down 1khz 9dB and you thought you fixed it only to sometime later realize you didn’t fix the real problem and the truth is that you can’t really do anything about the null anyway. Bummer. 



"Why can’t I trust my ear?" 

No one says you can’t. *You should, however, question your brain.*  
The problem with using your ear is simply that it’s easy to miss things or even to focus too much on something while altering what was good to begin with. 
For example, let’s say you hear a problem at 1khz; it’s pulling to the left. So, naturally, you start cutting the left side at 1khz and boost the right side at 1khz. Problem solved, right? Until you get in your car the next morning and it’s STILL there.  After some playing around you found that the problem really is due to a reflection from your right speaker. You wanted to believe you were fixing the problem and so you did… but you really didn’t. Another potential problem could be the issue at 1khz is due to a reflection; a harmonic of 500hz is causing the issue and if you wanted to fix the root of the problem, you needed to adjust 500hz. Not 1khz.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Measuring "Right"*

Now you're obviously asking how you measure the "right" way, or "listen right". Well, truth be told, it's not very easy to relay over the internet. It really takes time and practice. Luckily there are a few sources already that I think do a good job of illustrating the methods of measuring and listening. Here's some links:

Dr. Sean Olive's "How To Listen" Software
http://www.diymobileaudio.com/forum/how-articles-provided-our-members/163234-first-timers-guide-measuring-your-system.html
My RTA Walk-Through Thread





*Analyzing the Results*:

One thing that really needs to be done is *spatial averaging*. The two measurement tutorials above discuss this but I want to reiterate it because it's just that important. If you'd like additional information, please read the attached PDF in this post . The document is by Dr. Earl Geddes and is a study he did for Ford Automotive. It's not too technically heavy which should keep the "I don't like science" excuses to a minimum. 
Note: The following is plucked straight from my build log. 


The cliffs version is simply this: your car (and home) are wrought with reflection inducing panels/walls. When measuring response in the environment, you have two options: 

"Gate" the response so you obtain only the response of the speaker you're trying to measure and you essentially ignore everything else.
Measure everything: speaker response and reflections.

Doing the first in the car?... good luck. How about... don't bother. At least not until you've gotten really good at measuring and understanding what you're measuring. Let's just say for all intents and purposes you won't be doing the first... like... ever (thank you, Taylor Swift, for making that phrase weird now). Really, it's just pretty much trivial unless you have a very specific goal and understanding of how to achieve it this way. 

So, we do the second option. The issue, then, is the fact every measurement you take is a measurement of _EVERYTHING _occurring at the mic. This is good and it's bad. It's good in the way that there's not a whole lot you can do to the speaker itself so it kind of keeps you from worrying about it. - Although, this is why I really encourage people to study independent tests or do their own to understand the issue(s) with the speakers they've chosen _before they use them in the car_. - It's bad because, thanks to the nature of the reflective environment, you can't really trust a single point measurement (a measurement taken with the mic in one location). If you move the mic as little as one-half inch you'll get a different result. Most notably in the higher frequencies. *This means RTA'ing your car for any desired curve by using one mic measurement is a TOTAL WASTE OF TIME*. It's ideal to take multiple measurements in the "head area" and average them together. TrueRTA, OmniMic, and REW allow you to do this pretty easily. Then you have what is known as a good spatial average. It's not an exact method but it's the most realistic and approximates a very realistic response in the seated position. 

For this spatial average there are a couple methods: one is a 'live average', discussed in my tutorial, the second is using various single measurements and averaging them together. 

For this purpose, I did (6) individual measurements.
Now I've got 6 measurements. What next? Simple: average them all together to get one measurement.

Here's an example....

All six measurements taken by the method described above (no smoothing applied):













Same as above, but with 1/3 Octave smoothing:














All of the above averaged in to one response:















Let's talk about the above... at least my personal take on the above. I'm sure others may key on to some other aspects I might otherwise ignore or just overlook. 

Notice how the response varies more the higher you get in frequency? This is exactly why I said using a single point measurement to tune to a curve is a very bad idea. 

I'm going to ignore the shape of the curve, however, for this post... what I really want to focus on is midbass/subbass response, so let's look below 300hz. Anyone notice the one glaringly different thing about the response below this frequency versus the response above it? No matter where the measurement was taken, the response is pretty much the same. This is the critical frequency area (schroeder frequency (Fs)). Linkwitz gives the most simple definition I can think of here:


> The frequency fs is also called the Schroeder frequency and denotes approximately the boundary between reverberant room behavior above and discrete room modes below.


Which makes sense, right? Look again at the graphs I provided. Reverberation is occurring above about 300hz as evidenced by the diverging responses from the 6 head area measurements. Below this, the response is pretty constant in this area so it is modally (sp?) dominated. What does this mean to us? *You can ignore spatial averaging (multiple measurements) when focusing on low(er) frequency response! This saves you time!* Of course, every car is different so I suggest you always do a spatial average to determine where this Fs occurs in your car, but you can expect it to occur around the 200-400hz area, depending on car size. The larger the 'room' the lower the Schroeder frequency. This means once you do a spatial average you'll know where this frequency is. From then on, when you only care about working on the low frequency response, you can ignore spatial averaging and just put the mic at the seated position and measure, tune, measure, tune, rinse, wash, repeat until you're satisfied. I will caveat this by saying that tuning low frequency response with graphic EQ's isn't easy because modal peaks and dips are often too narrow and too specific of a frequency to effectively be targeted by graphic EQs. This is where my subsequent posts will sort of pick up. 



*Cliffs:*

Tuning based on one mic measurement is a waste of time. This has a caveat...
Take a few measurements in the head area, where you sit. Look at them all overlaid. Where do they really start to diverge? This is your car's Schroeder frequency.
Above the Schroeder frequency you must take multiple measurements and average them if you want to tune via RTA.
Below the Schroeder frequency, one mic measurement will suffice since the response doesn't change enough to matter.
Graphic EQs aren't the best tool for fixing response issues low in frequency. Parametric EQs are MUCH better. But, if all you have is a graphic use it to the best of your potential.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

*Analyzing the Data*

As mentioned earlier, one problem with measurements is the incorrect use of the results. Once you achieve your measurements through the aforementioned methods, here’s some things you should ask yourself before heading straight to the EQ:

Is what I’m seeing audible? Is it a dip or a peak? Here’s a good quote from Floyd Toole regarding this:



> Attenuation of excessive levels appears to be very safe, but avoid trying to fill deep holes. A narrow dip is probably caused by a null in a standing wave or interference pattern. As such it is the acoustical equivalent of a bottomless pit - it cannot be filled. Narrow dips are difficult to hear in any event, and all that will happen if you dial in a lot of gain is that the amplifiers will have reduced headroom, and the loudspeakers will be working harder to no avail. The result will be increased distortion.


*
In other words, don’t worry about dips. But do worry about large peaks.
*
If it’s a peak, use the EQ to drop it down some and re-measure. Did it go away? If not, are you really sure it’s a peak, or is it just a spot next to a null that looks like a peak? A-ha!!!!
If you are looking at the overall system response, try measuring each side individually and see if you notice something that may be causing the issue. IOW, take it down a level and see if the issue is caused by just the left midrange (for example). Then evaluate and tweak (or don’t) as necessary.


Let's look at some real examples...




*Low Frequency EQ'ing (tuning below the Schroeder Frequency):*

Taking off from the above, I'm only going to focus on the response below 300hz (graphs are out to 400hz for the sake of resolution). 

The following is with no EQ. Time alignment and levels have been set, however.

First off, let's take a look at the difference measured from the driver's seat vs the passenger's seat. 











The results show the same response show pretty much the same curves above 70hz. I've seen this numerous times; almost as if the car has varying Schroeder frequencies. One is for the entire cabin; the other is for one location at a time. Of course, I'm not talking about moving the mic to the rear of the car... that's an entirely different can of worms. The point in this measurement, however, is to show that there is actually a sub-band that really needs attention below the seated Schroeder frequency: the midbass band is entirely subject to this. As shown, 70hz is the starting point for different results between seats but 300hz is about the starting point for different results within the same seat. So, 70hz to 300hz is gonna be a total PITA in my car. Through about 5 years of dealing with this same car, my measurements show me what I already know, so it's definitely been vetted. 



Next...






After doing that, it's time to get back to the driver's seat and start measuring response from there.

One might choose to measure the system response as a whole and use the RTA that way, but it's a bit more conclusive to study each individual side's response (left and right side response). This is easy to do: just pan the balance to one extreme or the other and measure. When you do, you'll have the left side stereo contribution vs the right side stereo contribution.

So, here we have just that. Panned left is Green. Panned right is Purple. No EQ. 1/12 octave (to show the crappy little modal stuff that 1/3 doesn't get).











What this really shows me is that both the left and right side stereo contributions have their own problems. Notice that slight dip around 85hz at the driver's seat? Everyone has that problem to some degree because of their proximity to the speaker. Bottom line, that dip is a cancellation mode. There's nothing I can do to fix it, either. I can EQ it up but what will happen is I'll just keep applying more power to the driver's side midbass, causing distortion to ramp up and likely audible issues due to it. And while it may raise the response there, it'll also make resonant modes more problematic. The potential to damage the driver certainly exists. There's just not a whole lot you can do here. Some EQ will help but if you try to flatten it out by adding 4-5dB of EQ you'll alter the response curve in a negative way and create other issues. The only way to really fix a problem like this is to move from the boundary causing the null or move your driver(s). So, I just ignore this. Truth be told, it's not a real big issue when listening. And this is just one more example of why you should not rely entirely on the RTA. You should always use your own ears to accompany what you've measured. If you have a narrow dip it's not as audible as a broad dip; the same goes for a bump in response. 

So, yea... I'm not going to sweat that dip at 85hz measured at the driver's seat. It's a lost cause and serious waste of time to try to flatten it. I just want to smooth it so a bit of EQ here and there will help that. 


Now, look at the rest of the curves. That dip around 85hz on the left side is exacerbated by the rise in response around 125hz. After looking at the decay plot, measured by REW, I see why...











This is a plot of response over time, laid out in 2-D. The highest levels are closer to the initial response time. As the graphs change color below one another, you're seeing 'slices' of the response in time. Look at the legend. It shows time in milliseconds (ms). Each color corresponds to a time slice/section. Ideally want to see is each slice dying out quickly and contributing less and less to the results. However, what you actually get is modal issues showing up... these are the ones that linger around and don't taper off smoothly. Looking at these plots is pretty subjective and really should be used with some subjective listening as well. But, I'll give some thoughts on how I look at it...

The 125hz issue showing up in the left side FR plot... now look at the decay plot around that frequency. See how the darker blue looks pretty mountainous here with a dominant spike at about 125hz? Notice how the shade of blue just before this has the same spike? This is an indication of a modal issue. Luckily, I have an EQ band right here... I can cut it some. The problem, however, is cutting here also affects the tonality in other ways. With a parametric EQ, I can set a narrow Q and cut accordingly. But, I don't have that, so I have to cut here with the 31 band EQ. Here is the result when I use the EQ to cut 125hz by 3dB:











Not surprisingly, there was no miraculous alteration of the issue. It cut the problem by 3dB as it should but it didn't make the ringing issue go away. It did lessen the effect some. This is where subjective listening will tell you if it helped. The drawback here is you also changed the tonality of the system because the Q (bandwidth) of the 1/3 octave equalizer is so wide; it doesn't just change a single frequency. 

This site is a great reference for what frequencies influence what you hear and can help you understand the tradeoffs you deal with when changing EQ bands to fix problem areas:
Interactive Frequency Chart - Independent Recording Network



There are other frequencies that do the same thing. 100hz definitely lingers. 83hz lingers as well. Remember earlier my bit about bumping up 80hz to fill in that hole caused by the left side response? What do you think happens when you do that regarding the modal issues? It's a nasty problem. What you really need is a way to target specific modes without negatively affecting the other areas you want to fix with standard EQ methods. This would be a really good intro in to why parametric EQs are so good. So, I'll stop here and pick up there when I have the chance.

Keep in mind I've only really discussed one component of the system response here. The right side response has it's own problems as well. 


*Cliffs:*

Room modes suck. They muddy up system response as a whole.
When the midbass is muddy it overshadows everything good about the rest of the system. 
All cars have modal issues smack in the midbass area. 
Standard EQ can only go so far. But when properly used, EQ can help tame some of the modes which results in a much more tonally pleasing car stereo and much better blending with sub on the low end and midrange on the high end.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

Tuning tips that might be worthwhile...


*Just a spoonful of sugar: The difference in a single dB here and there...*

At the Vinny (MECA Comp) I got some good feedback from David Hogan (judge). He is probably one of the best critical listeners I've ever met. Not only does he give poignant critiques, but he also provides excellent suggestions on how to improve the sound. 

He gave me a couple pointers here and there on my sheet (which I've linked at the bottom of this post). I had a few minutes last night to implement those suggestions. 1dB cut between 1-1.6khz on the right. A 1dB cut at 400-500hz on both sides. Saved the result. Toggled between my 'Vinny' setting and the newly updated one. While not a 'night and day' difference, it's still pretty apparent the focus tightened up. And when it did, there was more detail in the tracks I used to compare the tunes (Natalie Merchant - _Wonder _& Depeche Mode - _Enjoy The Silence_. IMO, you can't expect to make little changes have a great impact right away. Usually these kind of minute changes with a solid affect happen when you've gotten pretty far down in the tune (or assuming you already have an excellent baseline). If you have a hodge-podge setup and T/A, phase, levels are out of whack then adjusting a single dB here or there isn't really going to do much. It's like adding $1 to $1million. But, once you get to a point where your setup is blending well and things are pretty well tuned - not perfect, but good - these kind of changes can really help out.





*Sub/Midbass tuning. Remember: It's all about the frequency wavelength.*

This info can really take your system from being "cool" to "holy crap!" in about 30 minutes. So, don't just read it and say "thanks, dude". GO OUT TO YOUR CAR AND APPLY IT. Ask questions. I guarantee you there will be an improvement if you apply this info...


First off... *There's no sense trying to EQ something if the phase is out of whack at the crossover.* Using the method I laid out here (Subwoofers And Time Alignment), I focused on measuring the response while adjusting the actual phase on my Helix DSP. 
Note: If you don't have a phase delay, just use time alignment; it's the same thing. The phase adjustments just allow you to do T/A in larger steps. To have an idea how much T/A you need for the sub/midbass adjustment use this link to calculate the relationship between phase angle/frequency/time:
Phase angle calculation time delay frequency calculate phase lag time shift between voltage difference time of arrival ITD oscilloscope measure two signals formula angle current voltage phi phase shift time difference - sengpielaudio Sengpiel Berlin

For example: Using the site above, you'll find that to adjust the phase at the crossover point of 80hz by 15 degrees you'd need to delay in time 0.50ms. This is important to understand.


15 degrees phase adjustment at 80hz requires 0.50ms... Think about that... that's a _big _number in time for such a small delta in degrees. To put it in perspective, think back to the time where you were adjusting your T/A values on your midranges to line them up. Typically, 0.5ms can mean the difference in too far right or too far left; you definitely move a lot of space laterally between midranges with 0.50ms. You typically moved in those 0.03ms steps and worked within a small window of probably 0.20ms to really tighten up your center focus. 

Now compare that to when you were doing T/A on your sub by a measly 0.03ms and couldn't get things to sound right. *It's because you need to adjust the delay for LOW frequencies by a LOT more to get the phase to swing just a little bit, as compared to a tweeter who's phase at 2khz can be a full 360deg out with that same 0.5ms.*


Make sense? Good! 

So, if you want to really adjust your subwoofer, I'd suggest doing so in 0.5ms chunks. Or, even using 2ms chunks (about 45 degrees at 60hz). Adjust incrementally until you've gotten to the point where smaller steps will help. Then switch over to 0.50ms steps. 

This is what I did:
I put the mic at the seat and swept. Saved. Adjusted phase on the DSP again. Swept. Saved. Rinse, wash, repeat. On the RTA screen, I zoomed in so I could focus on the 30-200hz area and I focused solely on the crossover point. I found the phase adjustments that got me the highest SPL. Then I went outside the crossover point and looked at how that blended with the rest of the system. 

After picking a few phase angles that worked well, I swept the system with an impulse using REW. I then looked at the group delay for the 'better' phase angles just to see if anything was out of whack. To my surprise, the GD was very benign. When I compared the new phase setting vs my old one with EQ it was a huge difference in smoothness. 


*Note: You don't have to have an RTA to adjust the subwoofer delay in the manner I discussed above.* It's helpful because it takes a lot of guesswork out of the equation. But, simply, you can play a test tone at the crossover frequency and adjust delay until you have the highest SPL by ear. Listen for the bass to be up front. You'll likely find an area within about 45 degrees that seems to be in phase. To help really narrow it down, go outside of the passband to your midbass by one tone (ie; if you cross your subs/midbass at 60hz, go up to 80hz) and repeat the steps to fine tune the delay. You should now have your sub DIALED.


It's worth noting that, again, thanks to the long wavelength of low frequencies, you're likely not going to get the sub in phase at the crossover AND frequencies outside of this. In my case, I just worked with the trade-off. I took the setting that provided me a high SPL at 70hz (my crossover point) and also good SPL below and above that. Make no mistake, you WILL run in to this same issue. Regardless of your crossover point. More so if you overlap; less so if you underlap. But you'll see the trade offs.

*The most important factor here is realizing the relationship of frequency to time/phase. And how you waste your time (no pun) trying to adjust subwoofer delay in very small increments.* Use BIG TIME STEPS for low frequency delay and alignment. The higher in frequency you go from there, the smaller those time steps in delay can be. Remember what I said above, 0.50ms delay is only 15 degrees at 80hz but it's nearly 360 degrees at 2khz. *Time Delay is not created equal!*


-------------------------------------------------------------------------------------------------


*EQ'ing the low frequency response with midbass and subs playing is the way to go.*

I dumped my EQ settings on my midbass and subs a few weeks ago. I realized that the way I had gone about EQ'ing them was backwards. I had EQ'd each individually. What I should have been doing was looking at them together. Why? Wavelength. You've surely heard about staggering midbass/subs to help smooth the response. It's nothing new. The problem is, when you EQ a driver by itself in this passband you're not accounting for the interaction and influence of the other driver. That can both hurt and help you. The key is, you don't know unless you evaluate the system as a whole. Midbass to midbass this is more important in the entire passband. Sub to midbass, this is really only important at and just around the crossover (maybe 1/2 octave +/- the crossover point). 


So, I measured the response of the subs + midbasses. I then measured the response of each midbass by itself and subs by itself. I did this last bit so I could better analyaize the influence of each speaker. For example, if I had a bump at 150hz, was it a single driver problem or both? For the most part, what I found was the response was pretty nominal, *when evaluating at 1/24 octave*.* I only had to make a few adjustments here or there. 
* using 1 octave or 1/3 octave smoothing will cause you to fix things you can't repair and stuff that doesn't need repair. I'd suggest only using 1/3 or 1 octave smoothing for midrange frequencies and above (namely, tweeter).

Also, if you read the scoresheets I posted at the end of this, you'll notice a couple comments regarding the low frequency response. Matt Roberts made a note of 30hz being out of phase. He was dead on. I noticed this in the GD data but at the time just dealt with it. What the culprit is was a low frequency boost I had; the boost was set at the lowest frequency I could and ran to about 35hz; this was just an attempt to add some extra low end. What happened, however, was a phase null at the seat. When I dropped that particular boost from my tune, the null practically went away. I'm still having to deal with a bit of an out-of-phase characteristic there right now but I don't think it's anything anyone else besides a well-trained judge would have picked up on had I not discussed it here. 


-------------------------------------------------------------------------------------------------




I'll try to update this thread with some graphs when I get the chance to illustrate just what I'm talking about with regard to the frequency response and group delay results.


Score sheets in PDF format:
https://www.dropbox.com/s/la9xj8spfnxr7uz/Vinny 2014.pdf



Hope this helps some folks out.


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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

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## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*

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## ErinH

that's about it.

I'll update the remaining sections as time permits.


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## BlackHHR

Subbed


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## dragonballz

Subbed. You still got your 8thcivic?


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## DLO13

BlackHHR said:


> Subbed


:afro:
:lurk:


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## bobduch

I just had a Lycan flashback. Subbed.


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## ErinH

I wish. For a number of reasons. :/


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## Hanatsu

Great thread explaining the "basics" and common misconceptions.

Don't wanna bring confusion into the thread but I tend not to 'experiment' much with crossovers though. If you use 4th order L-R acoustic slopes everywhere many problems are avoided, I simply set crossovers considering modal nulls and "non-linear distortion levels vs. output" in the lower frequencies and the power response of the mids upper limit. Have both drivers -6dB at the xover point and make sure they sum perfectly. The phase at xover can be confirmed by looking at the group delay for example.


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## whoever

Erin, great article, thank you
for taking the time to write it


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## Victor_inox

Subbed


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## ErinH

I realize I left something out in the crossover section so I'll have to address that before moving on.


The Tuning section; that's going to be my best attempt at relating all the lessons learned to you guys so hopefully you can benefit from it. That said, it's going to take me some time to provide data sets and graphics. I can type up the words on the fly in a relatively short time (a few hours for everything) but I think the graphics will go a long way to really driving the message home. So, be patient with me and it'll get done. 

- Erin


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## ErinH

updated.


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## bigbubba

Could have sworn I sub'd this earlier. Great read so far. Keep it coming!


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## ErinH

bigbubba said:


> Could have sworn I sub'd this earlier. Great read so far. Keep it coming!


you sub'd the other thread. I wound up deleting that one because I couldn't re-order things without making a huge mess of the thread. so, I punted and started this one. :/

thanks for the feedback. hopefully it helps. at some point in this thread I'll start trying to break down my tuning methods. but I need to build a base understanding and explain some key terms before I do that.


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## venki7744

Sub'd...looking forward to more interesting stuff.


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## bertholomey

Awesome job dude. My brother-in-law just started reading through threads on DIYMA and has expressed a desire to learn more of the basics.......I'm sending him a link to this thread right now!


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## Niebur3

Great thread Erin.....very long overdo and MUST become a sticky!


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## SkizeR

subd


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## mires

Read through everything once. Made my head hurt. Gonna try again after dinner.


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## sierrarider

Subbed. Thanks for sharing Erin...


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## Kevin K

Thanks Erin for the time and effort in sharing this. Your input is always beneficial.


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## sbaumbaugh

Sub'd

Thank you Erin,


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## pionkej

Hanatsu said:


> Great thread explaining the "basics" and common misconceptions.
> 
> Don't wanna bring confusion into the thread but I tend not to 'experiment' much with crossovers though. If you use 4th order L-R acoustic slopes everywhere many problems are avoided, I simply set crossovers considering modal nulls and "non-linear distortion levels vs. output" in the lower frequencies and the power response of the mids upper limit. Have both drivers -6dB at the xover point and make sure they sum perfectly. The phase at xover can be confirmed by looking at the group delay for example.


I agree...so long as you are saying 4th order summed response. For those that don't know what I mean, you can't just pick a LR4/4th order/24db slope (just to cover all forms of description) and say it's good. Your speaker has a natural roll-off and you need to combine that roll-off with the electronic slope you choose. So, for example, if your speakers have a -6db slope, you'd want a 18db electronic slope to achieve the 24db slope.

My 0.02


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## pionkej

Oh yeah, and BIG KUDOS Erin.


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## ErinH

I agree with you guys... To some extent. When I get further in to this thread I'll go a bit deeper in to the topic.


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## quietfly

As always, i'm impressed with the attention to details. this is one of the things that i love about this forum, the willingness of people to share the info they have gained via long hard lessons.
Thanks Erin!!


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## bradknob

Damn man this is awesome. Exactly what i need about now, Thanks.

Now I feel like im missing out because i just got an ms-8 and most the tunig is done fro me.


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## svnuss

Great post! Will reference.


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## SQram

Thank you for your efforts sir, great info for all!


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## rton20s

I have a feeling that links to this thread are going to be scattered all over DIYMA in the future. Thanks for all of your efforts on this Erin.


----------



## casey

very helpful, thanks again Erin. The only issue I will probably run into is not having the trained ear that some of you guys have. I want to try and do as much as I can but will definitely be getting some assistance. I just like to know what changes Im making and this definitely helps


----------



## Hanatsu

pionkej said:


> I agree...so long as you are saying 4th order summed response. For those that don't know what I mean, you can't just pick a LR4/4th order/24db slope (just to cover all forms of description) and say it's good. Your speaker has a natural roll-off and you need to combine that roll-off with the electronic slope you choose. So, for example, if your speakers have a -6db slope, you'd want a 18db electronic slope to achieve the 24db slope.
> 
> My 0.02


Yes indeed. I was talking about the acoustic slopes.


----------



## Militarymisfit

My Man! You just saved me from posting new threads. Great info and easier to find than spending hours searching through other threads.

I appreciate the initiative, and time and effort this is going to take. Thanks.

Oh... and sub.


----------



## ErinH

awesome. that's what I was hoping to do here. give one concise source for the more technical aspects of this hobby in a manner that doesn't need a phD to understand. glad to help. 

I will say most of this would be a WHOLE lot easier to explain if we all were face to face and had a whiteboard.


----------



## quietfly

bikinpunk said:


> awesome. that's what I was hoping to do here. give one concise source for the more technical aspects of this hobby in a manner that doesn't need a phD to understand. glad to help.
> 
> I will say most of this would be a WHOLE lot easier to explain if we all were face to face and had a whiteboard.


we could set up a GOTO meeting and have you present........ then you'd have a white board. we could record the sessions and add it to the posts....


----------



## TOOSTUBBORN2FAIL

They actually did that a few years at the national Dodge Neon gtg, with the Chrysler head engineer who oversaw the neon's design process. Had all kinds of how to's to fix problems that weren't supposed to be there, but occurred during production. The guy road raced them in spec neon and scca road touring.


Subbed. Lots of good info scattered over posts Erin has made, nice to have it all in one place.


----------



## cajunner

turn this into a video series, charge a nominal fee, for each section of knowledge blocks. Probably could pull it all into 8 sections, sort of a DSP primer and AutoSound 2000 briefs, mashed up.

maybe get a couple of guest speaker sections, Andy Wehmeyer, Jon Whitledge, that Summers guy, I'm sure there's a few others that could give an hour, maybe even have a little SMD spot for high-decibel, a couple of engineer types, maybe even Scott at Hybrid, whoever...

could go a long way with little budget, Kickstarter the series, who knows?

a full-on tuning clinical, starring industry heavy hitters, and oration/production by the personable party-man Hardaway, Hardaway's online guide to today's higher end...


----------



## mrpeabody

Fantastic stuff.

The flow of the information is really well done too.

Allot of things I've read about over the past few years, but now much more concise in my head, lol.


----------



## schmiddr2

Everyone is going to find something(s) in this that will help them. Stuck!


----------



## Theronh357

Sub'd


----------



## ErinH

updated some posts...


----------



## quietfly

I think this should become it's own WIKI....


----------



## casey

just re-read. overwhelming amount of stuff


----------



## SkizeR

reading the updated posts makes me so happy to have a left/right 31 band graphic EQ AND ten band parametric per channel


----------



## quietfly

Erin, 
a small request, when you update could you include a link to the updated section, its hard to traverse the entire thread on my phone while in meetings and your updates are infinitely more interesting than my meetings... 

thanks!


----------



## ErinH

sure.


----------



## quietfly

Danke schön!


----------



## 94VG30DE

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*



bikinpunk said:


>


I have been reading here for 7 years and this is the first time I have understood ^. Lot's of great content, more great to come I'm sure. Thanks for the concise flow of information. "Accessible" technical writing is hard to do, so A+ there.


----------



## fcarpio

Subbed!


----------



## ErinH

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*



94VG30DE said:


> I have been reading here for 7 years and this is the first time I have understood ^.


awesome!


----------



## Woosey

Wow great work Erin! This will keep me up late the next few days...


----------



## ErinH

Woosey said:


> Wow great work Erin! This will keep me up late the next few days...


You still awake?! lol.


I've let this slip a bit the past few days. I'm about to do some tuning on my car (hopefully) in the next few days and I'll be using that as an opportunity to update this thread with some measurements and analysis... it'll be my attempt at sharing the process and how the data can be viewed in different ways. I'll update with RTA plots and discuss my findings. I intended to do this a couple days ago but just haven't found the free time.



In the meantime, if anyone has any questions or possibly even wants to share some success stories from things learned here, I'd appreciate hearing that kind of feedback. If not, I'll just assume it's waaaaaay too hot for you guys to be inside your cars right now, too.


----------



## ike3000

I've been following a lot of your posts and find them very informative. Success stories? I don't even know where to start. I've tried each one of your techniques with great results. The most beneficial piece of information has to be properly setting xover points to avoid beaming and distortion spikes. Nothing revolutionary there, but for some reason your explanations "clicked" better than anything else I have ever read. I've been able to gain a lot of volume and dynamics just by choosing the proper xover points. Phase and time delay is an ongoing process, but I am finally comfortable with my xover points.


----------



## abusiveDAD

ike3000 said:


> I've been following a lot of your posts and find them very informative. Success stories? I don't even know where to start. I've tried each one of your techniques with great results. The most beneficial piece of information has to be properly setting xover points to avoid beaming and distortion spikes. Nothing revolutionary there, but for some reason your explanations "clicked" better than anything else I have ever read. I've been able to gain a lot of volume and dynamics just by choosing the proper xover points. Phase and time delay is an ongoing process, but I am finally comfortable with my xover points.



Agreed...
thank you kind sir (ErinH)


----------



## ErinH

ike3000 said:


> Nothing revolutionary there, but for some reason your explanations "clicked" better than anything else I have ever read. I've been able to gain a lot of volume and dynamics just by choosing the proper xover points.


that's the reason I started this thread. awesome to hear that.


----------



## BEAVER

I have a really stupid question... How do I open the file that contains the tracks for the disc? None of my programs will open it. :-/

EDIT: Nevermind. Downloaded a program to open the files. Google is your friend. Sorry for cluttering this thread.


----------



## quietfly

*.....Taps microphone..... * Hellooooo, Is there anybody in there? Just nod if you can hear me. Is there anyone at home?


----------



## ErinH

quietfly said:


> *.....Taps microphone..... * Hellooooo, Is there anybody in there? Just nod if you can hear me. Is there anyone at home?


was this for me? not taking offense if it was. just asking.


----------



## quietfly

i was just prodding for more knowledge.....  plus i was listening to some Floyd, and everything just aligned for the perfect idea for a post here. No offense was intended, and hopefully none was taken. You are one of our favorite distributors of knowledge!!!!


----------



## ErinH

I didn't think you meant it in a bad way. There's just no way to word "are you talking to me" (Joe Pesci voice) in a way that can't be misconstrued as such. lol. 


Yea, I'll update this with some more info at some point. I've actually posted some measurements and info in my build thread. Might port it over here.


----------



## Babs

That was Dinero. Taxi Driver.


----------



## captainobvious

Great job Erin. This will make for a great reference for understanding the problems with vehicle acoustics as well as ways to best work with the environment we're stuck with.

Might I suggest to also paste in some of your content from the sound power/power response 101 thread? Specifically the graphical representation of beaming and even sound power. Seeing a visual of 2 drivers playing (one with narrow response (beaming) and one with wide response is great. That was very helpful for me in understanding the concept and why it is so important in crossover selection in the vehicle.

I'm looking forward to further updates here. I know this is a laborious process, but your efforts are appreciated. Well done, sir !


-Steve


----------



## legend94

i just had a flashback to werewolf circa eca

awesome thread!


----------



## fcarpio

Just checking, for gain settings instead of using a DMM (like you did) you could use an oscilloscope and -10db test tones looking for clipping (set gains just before that clipping point) from the beginning of the chain to the end of the chain (HU, DSP, Amps and whatever else is there with gain settings) through all devices. Correct? Do you see any drawbacks with that method?


----------



## ErinH

Most people don't have or have access to an oscope. Using it is fine.

I would suggest not setting your headunit level with the -10dB tone. Set it with 0dB. Most likely you'll max it out either way, but you want to start with the clean output you can. Adjust the things downstream of the headunit with an attenuated tone (-5dB or -10dB is typical).


----------



## cajunner

an odd question, but one that many people do have is about the maxing out of the deck's pre-amp.

conventional wisdom says that when you push a component to 100% of it's clean max, you're running it hot. You're making the most electrical juice go through the parts, you're pushing the deck's limit on capacity, and that is worrisome.

I understand that for S/N or dynamic range considerations, right before clipping should be where the best signal ends up at the amps.

But, how about just turning it back to 2/3 output?

Would it possibly contribute to a cooler running pre-amp, or give the deck more available power on it's power supply, for other circuits to run?

Is running a deck at 0 db, or max theoretical clean gain, somehow driving the deck too hard?

If it's just a matter of saying "the op-amps are designed to push all the way to the stops, so it doesn't matter how hard you crank a pre-amp" then that would be good to know, but if there's a chance that some of us are heating up our units unnecessarily, that would also be good to know.


----------



## ErinH

when you're talking about clipping, at the first instance, you're talking about THD+N values in the hundredths of a percent. maximize the source output up to that point (whether that be full pot or 2/3 pot or any arbitrary number that fits this conversation). then make up any lost volume via the attenuated tones.

that's how I do it. that's how others do it. that's not how everyone does it (whether on purpose or due to simply having no 'method' at all). I used to set gains based on audible noise floor. now that it's not a factor, I set them by a simple 3:1 power ratio. 

YMMV.


----------



## cajunner

so the difference in longevity is nil?

you're saying that pushing a 4V deck to 4V as a normal tuning recommendation, is no different than pushing that deck to a max of 3V?

like, I understand that music is dynamic, and you rarely have a true 4V signal coming out of the deck for any duration, but in averaged power over time, would the deck that is averaging 1V be potentially less likely to fail over one that is pushing 2V?

always wondered that, as my non-scientific experience shows that a deck turned up real high, tends to heat up more, even if you can turn off the deck's speaker outputs, and especially if you can't, and even if they are not hooked to any load.

that's probably where the problem I think exists, in that when you turn the deck all the way to near-clipping levels, you may maximize the noise floor but you also may be shortening the life of the deck's output section, which is tied to the deck's pre-amp driving a set of op-amps that may or may not be happiest at higher voltage levels.


I usually equate power through components heating them up, as being a current issue, and in pre-amps the current is minimal, since we're dealing with high impedances.


but when you have a deck that is putting out it's high volt signal, at low ohm drive levels (a sign of quality in a deck) then maybe it matters?


----------



## ErinH

From a product life POV I can't see that it's an issue.


----------



## cajunner

fair enough, then.

I know today's decks with DSP on board, should have slightly higher total current draw, based on the DSP's sucking of the power tap. That alone helps bring the heat into the deck chassis, and using one of these decks as a dead head, or no power out to speakers off internal amplification, is going to still bring up the average operating temperature compared to a non-DSP deck.

and one of the biggest issues facing DSP product is the temperature fluctuation in vehicles. You see DSP chips getting errors when you raise the temperature, and I think that a lot of problems come from the car environment making it harder for DSP architecture to operate in all temperatures and climates.


----------



## subwoofery

cajunner said:


> so the difference in longevity is nil?
> 
> you're saying that pushing a 4V deck to 4V as a normal tuning recommendation, is no different than pushing that deck to a max of 3V?
> 
> like, I understand that music is dynamic, and you rarely have a true 4V signal coming out of the deck for any duration, but in averaged power over time, would the deck that is averaging 1V be potentially less likely to fail over one that is pushing 2V?
> 
> always wondered that, as my non-scientific experience shows that a deck turned up real high, tends to heat up more, even if you can turn off the deck's speaker outputs, and especially if you can't, and even if they are not hooked to any load.
> 
> that's probably where the problem I think exists, in that when you turn the deck all the way to near-clipping levels, you may maximize the noise floor but you also may be shortening the life of the deck's output section, which is tied to the deck's pre-amp driving a set of op-amps that may or may not be happiest at higher voltage levels.
> 
> 
> I usually equate power through components heating them up, as being a current issue, and in pre-amps the current is minimal, since we're dealing with high impedances.
> 
> 
> but when you have a deck that is putting out it's high volt signal, at low ohm drive levels (a sign of quality in a deck) then maybe it matters?


If you maximize the source output and set the gain with an attenuated tone (-5dB or -10dB), you'll be using less of your HU's pre-amp when playing your system at let's say 85dB of loudness 
VS 
Setting your HU down to 2/3 of its output and setting the amp gain to match the output voltage. Believe that your HU would (if it really does heat up) heat up more if you play it to 85dB of loudness

Am I wrong coz I'm not too sure after writting all of this lol 

Kelvin


----------



## EzMeUc

Really loved the gain settings method you described here. I always tought setting gains was something you had to play with. Now I can match it perfectly with my HU. Thanks!


----------



## cajunner

subwoofery said:


> If you maximize the source output and set the gain with an attenuated tone (-5dB or -10dB), you'll be using less of your HU's pre-amp when playing your system at let's say 85dB of loudness
> VS
> Setting your HU down to 2/3 of its output and setting the amp gain to match the output voltage. Believe that your HU would (if it really does heat up) heat up more if you play it to 85dB of loudness
> 
> Am I wrong coz I'm not too sure after writting all of this lol
> 
> Kelvin


I don't know.

I just know that when I crank the ol' decks to the stops, they did heat up, and I found that getting my deck to clip the amps at half-way on the knob, seemed to be a good setting, there was less heat noticed on the deck chassis.

Some of the newer decks with DSP probably do have ways to deal with heat, better than older decks on the average since DSP does take a small amount of power over the pre-amp or fluorescent lighting. Maybe the observation I make, is no longer valid?

I was asking, because I didn't know!


----------



## subterFUSE

Sub'd


----------



## fcarpio

ErinH said:


> Most people don't have or have access to an oscope. Using it is fine.
> 
> I would suggest not setting your headunit level with the -10dB tone. Set it with 0dB. Most likely you'll max it out either way, but you want to start with the clean output you can. Adjust the things downstream of the headunit with an attenuated tone (-5dB or -10dB is typical).


I see, thank you.


----------



## t8thegr8

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*



ErinH said:


> *Time Alignment and Levels*
> 
> 
> 
> 
> Another way to consider it is going from this ...
> 
> 
> 
> 
> ... to this ...
> 
> 
> 
> 
> 
> 
> 
> Recap:
> 
> Remember, when adjusting levels and time delay what you’re adjusting with one speaker is *RELATIVE *to the other speakers. You’re adjusting the nearest speaker(s) so that it sounds as if it has the same intensity and time of arrival as the furthest speaker. This would then sound as if the other speaker had moved.
> 
> Or think of it like this: you’re moving your ‘center’ with respect to the left and right boundaries. You continue to do this until your boundaries sound equidistant from the center vocalist/pink noise.


I'm thinking this driver focused timing would destroy the experience for the passenger.... is there a way for both parties to share the widescreen? I feel like the passenger gets screwed over in this deal and ends up with the scattered images. Id like to impress as well as enjoy, is that out of the question?


----------



## subwoofery

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*



t8thegr8 said:


> I'm thinking this driver focused timing would destroy the experience for the passenger.... is there a way for both parties to share the widescreen? I feel like the passenger gets screwed over in this deal and ends up with the scattered images. Id like to impress as well as enjoy, is that out of the question?


You need to use a center channel and get some sort of center channel steering processing. 

Kelvin


----------



## captainobvious

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*



t8thegr8 said:


> I'm thinking this driver focused timing would destroy the experience for the passenger.... is there a way for both parties to share the widescreen? I feel like the passenger gets screwed over in this deal and ends up with the scattered images. Id like to impress as well as enjoy, is that out of the question?



Do it with separate presets on the DSP 

You'll never get it to sound as good in BOTH seats as you can get it in a single seat.


----------



## t8thegr8

Right on. I forgot you can change the settings lol


----------



## Babs

I'd guess it'd be like dialing in your TA and EQ to sound best dead center between driver and passenger. It'd be very likely a completely different tune with EQ as well as TA. Might possibly be an 'easier' tune though, as distances would be identical left/right as would 'theoretically' the EQ between left/right drivers. Steering wheel being the one thing asymmetrical really.


----------



## 2DEEP2

Natan Budiano can get both seats to sound the same and just as good as one seat.

He has done it with a center channel and without a center channel.

He had midbass and mids mounted in the kicks with sub center of dash. Now he has mid and tweeter mounted high with a center channel. It's easier with a center channel. 

He also uses TA to get phasing and time alignment right for both seats.

When Andy W. tuned my car for USACi World Final back in 2007 we got both seats to sound, image and stage the same. However, the ambience was more closed on the passenger side than the driver side. 

Andy used the same technique that went into the MS8 with some frequency steering. Although he wanted to lower my tweeter crossover point, I would not let it go below 4k Hz.

My car's setup used 8" deep and wide in the kick, 3" forward with equal path lengths for driver and passenger and tweeters forward and wide in the A-pillar. The tweeter location and aiming could have been improved but crap happens when your 1000 miles away from the build site.

At the time, Andy was like Natan, both believe TA, without polarity flips, should be used to fix phase even in a two seat car.

I change polarity for the best phase response given crossover point and slop (as stated by Erin) and used TA from there.
However, I do listen for phase response issues across the bandwidth of the driver when I flip polarity. I use in/out of phase frequency barks across the bandwidth of the driver. 

Otherwise the main issue that need be addressed with TA in a two seater is putting the tweeter, mid and midbass on the same plain. Mounting drivers on the same plain is great but may not be possible with equal path length in a daily driver. When tweeters are crossed over high attenuation will move them more than time. Still, a system sounds better when you have the plain alignment right.

Easy way to do this is door locations or sail panel/A-pillar mid and tweeter with midbass below in kick/side panel. However, when path length difference are great (>6" I think was the AutoSound 2000 rule of thumb) the vehicle will image and stage better/equal from both seats with a center channel.

In any case, most who have achieved same SQ for both seats at an SQ just as good as one seat, will agree TA is useful in a two seat car. You want to fix phase and vertical alignment (putting Tweeter, Mid, Midbass speakers in the same plain) issues. You want the vehicle to be symmetrical with either two sweet spots (center channel method) or one large sweet spot (no center channel method). 

If you can't put speakers where they work best for both seats, add a center channel.

Erin, I really like how you explained Qt. Great write up.


----------



## ErinH

updated Post 17:
http://www.diymobileaudio.com/forum/2107545-post17.html


Cliffs:
You want to align your subs/midbasses in time? Use long delays. That little 0.03ms ain't gonna do anything.


----------



## subterFUSE

Great info here.

I was reading today in the REW forums about using Impulse Response to determine time alignment for different drivers. In the case they were discussing it was for aligning a tweeter to a midrange within the same cabinet, but I'm sure the same principles apply to a car.

Here's a link to the thread. Specifically, the final post has the instructions: time-alignment-impulse-response.html

The crux of my question centers on the chronology of tuning processes.

It would appear the recommendation is working on the higher-frequency drivers first, setting the desired Xover point, and then adjusting the time alignment so the initial rise of the impulses is aligned, and then flipping polarity of the lower-frequency drivers so that the phase alignment results in max SPL at the crossover point.

Would this mean that the suggested order of tuning be tweeters first, then mids, and subs last? Adjusting TA/phase for the lower frequency drivers at each step until finished?

Or does the order not matter? Personal preference, etc...


----------



## 2DEEP2

My preference is Midbass First.

I work up from there and add in the subs last.

Because I used an 8" and tweeter set up for many years and phase issues are most pronounced in the midbass frequencies, I've always started with midbass.

However, I also learned that lower fundamental frequencies can create color at higher harmonic frequencies. So, my experience would suggest it is better to get lower frequencies right first and move up.


----------



## Rusty

2DEEP2 said:


> My preference is Midbass First.
> 
> I work up from there and add in the subs last.
> 
> Because I used an 8" and tweeter set up for many years and phase issues are most pronounced in the midbass frequencies, I've always started with midbass.
> 
> However, I also learned that lower fundamental frequencies can create color at higher harmonic frequencies. So, my experience would suggest it is better to get lower frequencies right first and move up.


I will try this once I get my install sorted. thanks for the tip.


----------



## peenemunde

I've spent years surfing, trying to understand what all these big words mean.... Hell, even little words sounded like jibberish before spending days learning the fundamentals of acoustics and how a DSP can alter response so well. 

But until my first read through, I merely thought I had a good grasp of how it all comes together to give you aural sex. Very intuitive so far and extremely user-friendly. 

Thanks for taking the time to change DIY history 
This thread will be legendary and live on through the ages


----------



## yogegoy

Halloo, don't mind me just subbing.


----------



## DavidRam

Amazing thread! Thanks for all the time you put into this...


----------



## chucko58

Thank you ErinH. Subscribing in the hopes that there will be more of this good info down the road (pun intended).


----------



## 1fishman

Don't know how i missed this incredible thread!
Thanks sooo much for this.


----------



## ErinH

No problem. I keep meaning to add some things to this but I always forget or don't have the time. In the meantime, I'm glad this is helping folks.


----------



## Tritty

Excellent thread! Sub'd


----------



## Muzak4Life

Erin,

This is fantastic. I am very green to this and attempting my first active front stage with DSP.

I have a really elementary question regarding setting your gains.

How do I play a tone (-10db)? Is it a file I can download?

One step at a time for me as I am new to everything, but absolutely love it.

Thanks


----------



## Muzak4Life

Sorry, one more question....sorry for the basic questioning and understanding..

My Zapco ST-4D is rated at 100wrms @ 4
My SB17MFC35-4 is rated at 60wrms
My D3004/6020-10 is rated at 100wrms

Would it be best practice to set the amp gains at the 100wrms level of the amp, given the 60wrms of the SB's?

I hope this is relevant to the thread.

Thanks


----------



## subterFUSE

Muzak4Life said:


> Erin,
> 
> 
> 
> This is fantastic. I am very green to this and attempting my first active front stage with DSP.
> 
> 
> 
> I have a really elementary question regarding setting your gains.
> 
> 
> 
> How do I play a tone (-10db)? Is it a file I can download?
> 
> 
> 
> One step at a time for me as I am new to everything, but absolutely love it.
> 
> 
> 
> Thanks



It's a test tone from a utility CD. I run mine from a CD set by Autosound 2000.
There are various tones at different frequencies and levels.


Sent from my iPhone using Tapatalk


----------



## Babs

Found a great app for tones. Called Tone Generator. Works nice!





























Sent from my iPhone using Tapatalk


----------



## Muzak4Life

Thanks Guys...Really appreciate it.


----------



## subterFUSE

REW also has a tone generator, and it's free.


----------



## Muzak4Life

So I downloaded the frequencies onto an SD card and spent morning adjusting gains and DSP per Erinh guide.

Holy Cow! Where has this been all my life....I havent even began the EQ and tweaking. I think I still need to fine tune my crossovers/ slopes, but it can only get better.

Thanks for the assistance....Still trying to reign in the crossovers if anyone has advice, always listening. System below.

Thanks!!!!


----------



## ErinH

If you have left speakers on one amp gain and right speakers on a separate amp gain, one thing that may be worth considering is to level match the left/right outputs at the amp. 

instead of adjusting the DSP level at the processor to attenuate the nearside speakers (typically I attenuate anywhere from 1-5dB depending on the speaker, location, and aiming), you may want to simply make up for that at the amplifier. This would help to lower the noise floor if that's a problem, too. 

So, let's say you set left and right speaker gains with a -10dB tone. After you do this and you find in the DSP that you're having to cut the left side by 3dB, just go to the amp, use a -7dB tone and 'level match' at the amp. that will help lower the noise floor. 

I don't do this myself, but it's just something that may be worth considering.


----------



## Lou Frasier2

i saved this in my documents ,definitely going to implement this stuff in my system


----------



## ndm

ErinH said:


> If you have left speakers on one amp gain and right speakers on a separate amp gain, one thing that may be worth considering is to level match the left/right outputs at the amp.
> 
> instead of adjusting the DSP level at the processor to attenuate the nearside speakers (typically I attenuate anywhere from 1-5dB depending on the speaker, location, and aiming), you may want to simply make up for that at the amplifier. This would help to lower the noise floor if that's a problem, too.
> 
> So, let's say you set left and right speaker gains with a -10dB tone. After you do this and you find in the DSP that you're having to cut the left side by 3dB, just go to the amp, use a -7dB tone and 'level match' at the amp. that will help lower the noise floor.
> 
> I don't do this myself, but it's just something that may be worth considering.


The dropbox account says that the tones have been moved. What happened?


----------



## ErinH

I removed them. They were part of a disc that I didn't want to share forever. 


Others have noted you can get tones elsewhere so you have some other options. RealmOfExcursion.com also has free tones.


----------



## ndm

ErinH said:


> I removed them. They were part of a disc that I didn't want to share forever.
> 
> 
> Others have noted you can get tones elsewhere so you have some other options. RealmOfExcursion.com also has free tones.


Gotcha

I think I already had them but I was going to download just in case.


----------



## rton20s

There is also free software available to generate your own tones. I recently used Audacity for that purpose.


----------



## ndm

rton20s said:


> There is also free software available to generate your own tones. I recently used Audacity for that purpose.


I am trying to figure out how to generate 0db, -5db, -10db tones in Audacity. 

I would prefer to use audacity. 

Any help would be nice.


----------



## rton20s

This is how I did it. 

Generate > Tone

Waveform: Sine
Frequency (Hz): Whatever you need (40, 125, 4000, etc)
Amplitude (0-1): 1
Duration: Whatever you need (I used 1 min.)

Once the track is created, do the following...

Effect > Amplify...

Amplification (dB): Whatever you need (0. -3, -5, -10, etc)

I would then export to .WAV format. 

That was it. If this is incorrect, hopefully someone else will chime in.


----------



## ndm

rton20s said:


> This is how I did it.
> 
> Generate > Tone
> 
> Waveform: Sine
> Frequency (Hz): Whatever you need (40, 125, 4000, etc)
> Amplitude (0-1): 1
> Duration: Whatever you need (I used 1 min.)
> 
> Once the track is created, do the following...
> 
> Effect > Amplify...
> 
> Amplification (dB): Whatever you need (0. -3, -5, -10, etc)
> 
> I would then export to .WAV format.
> 
> That was it. If this is incorrect, hopefully someone else will chime in.


Awesome Thanks


----------



## Muzak4Life

I hope I am not turning this thread into swiss cheese here. I apologize if I am and will start a new one if appropriate.

Question. Set the gains on my amps using DMM. Do I leave the HPF/Full/LPF nuetral and use the Helix to process? Do I leave the frequency alone as well and allow the Helix to pass? Same with Bass?

I am wondering if I am canceling out the DSP downstream adjusting the amp settings. 

My first time doing this, lots to learn and I could not find a clear answer to this.


----------



## funkalicious

Great info. Subscribed for further enlightenment.


----------



## Tom Westling

Great info. I will be using this when I assemble and tune my new system. Thanks very much for putting it together!


----------



## subterFUSE

Muzak4Life said:


> Question. Set the gains on my amps using DMM. Do I leave the HPF/Full/LPF nuetral and use the Helix to process? Do I leave the frequency alone as well and allow the Helix to pass? Same with Bass?



If your goal is to run active crossover with the DSP, then yes.... you want to disable the crossover inside the amplifier. Set the amp to Full.

Just be aware that when running active you must be mindful of your DSP settings so that you don't damage your tweeters with too much bass.


----------



## damonryoung

Just a late sub here...


----------



## jonah1810

This is so useful!!! 
I feel stupid for asking so many questions now, a large majority of them are outlined in this guide! 
Thanks erin!


----------



## Babs

subterFUSE said:


> If your goal is to run active crossover with the DSP, then yes.... you want to disable the crossover inside the amplifier. Set the amp to Full.
> 
> 
> 
> Just be aware that when running active you must be mindful of your DSP settings so that you don't damage your tweeters with too much bass.



Yep. Reason I'm hoping more and more amps go back to old school.. Without any crossovers in the signal chain at all. 


Sent from iPhone using Tapatalk


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## GenGeek

Such an amazing thread, Erin! This helped me out in my (still building) build and I feel like I can conquer my own audio system now because of you. Thank you SO very much!!


----------



## sqnut

Essentials of sound quality? 

1. Get a ref sound.
2. The ability to hear a difference between the ref sound and your car.
3. The ability to tweak and say better or worse accurately.

....imho.


----------



## RXZILLA

ErinH said:


> *Foreword:*
> 
> For a while now I’ve been in this hobby, learning and falling on my face with failures in understanding and application. Over the past few years through competition, get togethers, talking with friends, research and testing drivers for my site (in my signature), I feel like this stuff has finally started to really click.
> 
> First and foremost, I'm a fan of music. All sorts. I'm not in this hobby or a fan of posting data just because. Like many of us here, I've been jamming out since a toddler (I have the video of me dancing to Michael Jackson as a 3 yr old to prove it... and I had some awesome moves, lol). Mix that with a desire for learning technical things, it seems like a no-brainer that I'd really have a passion for this hobby because it really does force you to study if you want to excel in it.
> 
> What I notice on the forums is that a lot of people starting off tend to get confused by speakers and tuning… and it’s not easy to weed through all the mess or ignore the bad advice when you don’t know the person it’s coming from. There are a lot of threads where people tell you what to do but they don’t really explain why. Typically, in my opinion, a lot of this is just restating what someone else told us. I’m no stranger to this. This thread is not to be considered an all encompassing summation on sound quality. I am not the foremost authority on sound quality and speaker reproduction and I don't pretend to be. That said, I've spent the majority of my free time over the past 5 years researching and studying acoustics and providing 3rd party data for various drivers and products. I'd like to consider myself as an educated nooB... because I'm always learning, I'll never be an expert by any stretch. Hopefully, this thread will shed some light on the why and how’s of tuning and help you understand why speaker measurements matter.
> 
> Sprinkled in with some technical info are my own trials and tribulations; what matters and what doesn’t. What is a good midrange? Why do I need to cross my midrange at 3khz instead of 8khz?  Why is time alignment so important? How do I use time alignment and how is it different from polarity? Now, of course, I can’t speak for all situations and I won’t say that the things I advise against can’t work.
> 
> *Simply put: the following is in my opinion best practice advice. *If you talked to me in person, everything I say below would be said to you regardless of who you are or what gear you run. Therefore, this thread is to be considered an attempt by an enthusiast at listing the basics of speakers along with some tuning tips/methods based on years of experience and reading on various internet sites. I’ll cite sources where necessary but since this is built on years of reading I can’t really give credit where it’s due in every instance because I don't know the first place I read something.
> I can say various forums, Linkwitz, Zaph, Toole, JBL/Harman, Keele, Geddes, and my friends and mentors in this hobby… all of them are my resources. And in many ways, I am my own resource through trial and error.
> 
> 
> As of this writing, this thread is incomplete. I will update as time permits but wanted to get a jump on it so people can start reading and possibly go ahead and try to apply some of the content herein as I go.
> 
> 
> *Cliffs: *
> If you need cliffs already then this thread isn’t for you. You’re going to have to do some reading so consider this a warm up.
> (bet you didn’t see that coming)




So I tried the Time Delay Calculator and it didnt sound correct. I took the measurements and reversed them and the sound moved to the center of the stage. Am I doing something wrong?


----------



## TwistdInfinity

Okay, so if your dsp/Hu takes time delay input in milliseconds, then you need to delay each speaker according the difference in distance from the furthest speaker.

Ignoring your sub, your furthest speaker will likely be the passenger side mid bass. 

Then measure the distance between you and the p/s mid bass and you and the p/s tweeter. The difference in them is the value you convert to milliseconds to delay the p/s tweeter by. 

Repeat for d/s tweeter and mid bass both compared to the p/s mid bass


I use an app on Android called Delay Calculator. It accounts for temperature also (speed of sound changes at different Temps as the air is thicker and thinner). 

From there it usually requires a bit of playing around to get it perfect. 

I'm not sure with the distance input dsp/hu's if you have to enter the amount of distance you want to delay it or simply enter how far away it is and it will automatically work it out? Have only used millisecond entry format 
Sent from my D5833 using Tapatalk


----------



## Ainuke

TwistdInfinity said:


> Okay, so if your
> 
> I'm not sure with the distance input dsp/hu's if you have to enter the amount of distance you want to delay it or simply enter how far away it is and it will automatically work it out? Have only used millisecond entry format
> Sent from my D5833 using Tapatalk



On the Alpine H800, it's rather confusing, as it appears that you should be adding the distance in, and it calculates the appropriate delay. However, it's the opposite. The distance needs to be added into the _opposite_ side. So you're looking for the _virtual_ distance you want; and the _difference_ in the distances, to boot. 

So the difference between the far mid bass (65") and the near mid bass (25") is 40", you want to _add_ 40" to the _near_ side, for a virtual 65" distance to match the far side's physical distance. 
Tweeters should be calculated by the distance difference from the far mid bass as well (assuming the far mid bass is the farthest one)

I _*think*_.
It took me forever to reason this out. The manual is no help at all, only saying that you can choose time or distance to input.
I only haven't used the auto time function on the dsp because I want to keep the rears out of the equation to improve staging. Haven't figured out how to isolate the fronts without changing the speaker setup to fronts only (all other settings are lost when you do that).


----------



## Ainuke

TwistdInfinity said:


> Ignoring your sub, your furthest speaker will likely be the passenger side mid bass.


Should you be ignoring the distance to your sub?

If it's the furthest driver from the listening position (especially if mounted in the trunk), it should be the one the rest are time adjusted for, no?

Mine's under the floor between the driver and passenger seat, so I don't have this issue, but inquiring minds want to know.


----------



## Babs

RXZILLA said:


> So I tried the Time Delay Calculator and it didnt sound correct. I took the measurements and reversed them and the sound moved to the center of the stage. Am I doing something wrong?


The page simply tells you how much delay in ms to apply to each side for the shorter distance speakers to all match up to the longest speaker. So if something was terribly off but swapping fixed it, somewhere the numbers got swapped.


----------



## TwistdInfinity

Ainuke said:


> Should you be ignoring the distance to your sub?
> 
> If it's the furthest driver from the listening position (especially if mounted in the trunk), it should be the one the rest are time adjusted for, no?
> 
> Mine's under the floor between the driver and passenger seat, so I don't have this issue, but inquiring minds want to know.


Yeah you're correct in that the sub will have zero delay and everything will be delayed against that, but to simplify the explanation of how to delay near side vs far side I was leaving it out. 

Ideally you would set the sub to zero, then with a sine wave tone that's in the middle of your sub and woofer crossover region playing, start delaying your far side woofer until the tone is at it's maximum volume (or flip the woofer phase 180 degrees and delay until it's quietest if that's easiest to hear). This is the point your sub and woofer are in phase with each other. 

Now you find out how much delay to apply to near side woofer, punch it in then fine tune with your ears till happy. 

If you're running a three way front you would probably want to do the phase matching of woofer to mid as well (woofer to tweet or mid to tweet isn't as important). To do this - turn everything except the far side woofer and mid off. Again, play a tone in the crossover band between woofer and mid and delay your mid until the tone is at its loudest. 

Then turn woofer off and turn near side mid on. Delay near side mid until image is in the same spot as where the woofer image was. If you get them in the same spot both your near side mid and woofer should be in phase with each other too. 

Then delay/level match tweeters till image is centred. 

So you end up with a centred and phase aligned system, not just a centred system. If you don't phase align your driver's to each other you will get some weird phase issues in the crossover regions. 

Sent from my D5833 using Tapatalk


----------



## Ainuke

TwistdInfinity said:


> Yeah you're correct in that the sub will have zero delay and everything will be delayed against that, but to simplify the explanation of how to delay near side vs far side I was leaving it out.
> 
> Ideally you would set the sub to zero, then with a sine wave tone that's in the middle of your sub and woofer crossover region playing, start delaying your far side woofer until the tone is at it's maximum volume (or flip the woofer phase 180 degrees and delay until it's quietest if that's easiest to hear). This is the point your sub and woofer are in phase with each other.
> 
> Now you find out how much delay to apply to near side woofer, punch it in then fine tune with your ears till happy.


Gotcha. 
I like this method; I'm going to have to give it a try. I've always wondered about using the straight measurement way. Depending on where you measure to, you can be off enough to make a difference. I've heard both to use the closest edge of the cone, and to estimate distance to the voice coil. Even harder when said cool is behind a door panel...

Between this thread and Hanatsu's one on REW, I have my work cut out for me.


----------



## mbradlawrence

TwistdInfinity said:


> Yeah you're correct in that the sub will have zero delay and everything will be delayed against that, but to simplify the explanation of how to delay near side vs far side I was leaving it out.
> 
> Ideally you would set the sub to zero, then with a sine wave tone that's in the middle of your sub and woofer crossover region playing, start delaying your far side woofer until the tone is at it's maximum volume (or flip the woofer phase 180 degrees and delay until it's quietest if that's easiest to hear). This is the point your sub and woofer are in phase with each other.
> 
> Now you find out how much delay to apply to near side woofer, punch it in then fine tune with your ears till happy.
> 
> If you're running a three way front you would probably want to do the phase matching of woofer to mid as well (woofer to tweet or mid to tweet isn't as important). To do this - turn everything except the far side woofer and mid off. Again, play a tone in the crossover band between woofer and mid and delay your mid until the tone is at its loudest.
> 
> Then turn woofer off and turn near side mid on. Delay near side mid until image is in the same spot as where the woofer image was. If you get them in the same spot both your near side mid and woofer should be in phase with each other too.
> 
> Then delay/level match tweeters till image is centred.
> 
> So you end up with a centred and phase aligned system, not just a centred system. If you don't phase align your driver's to each other you will get some weird phase issues in the crossover regions.
> 
> Sent from my D5833 using Tapatalk


I utilize this roughly but level match the all drivers first. I note that you point out level matching the tweets, do it for mids/midbase as well?


----------



## TwistdInfinity

Yeah I actually measure with a mic each driver first separately and level match and EQ matching drivers so they have as similar a response to each other as possible. 

Then do the TA. 

If you don't match the drivers in both response and level first you will find your image is very unfocused or drifts a lot at different frequencies as one side will be louder than the other 

Sent from my D5833 using Tapatalk


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## mbradlawrence

Ok, so, while others have different opinion on the order, at least two of us are on the eq then TA process!


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## Ainuke

mbradlawrence said:


> Ok, so, while others have different opinion on the order, at least two of us are on the eq then TA process!


I would offer this up:
Since the distance of the drivers to the listening position is fixed and not affected by EQ, but that the phase alignment _can_ affect EQ, then TA would be the better place to start. 

My $0.02

-Erik


----------



## SkizeR

Ainuke said:


> I would offer this up:
> Since the distance of the drivers to the listening position is fixed and not affected by EQ, but that the phase alignment _can_ affect EQ, then TA would be the better place to start.
> 
> My $0.02
> 
> -Erik


beat me to it


----------



## Babs

Ainuke said:


> I would offer this up:
> Since the distance of the drivers to the listening position is fixed and not affected by EQ, but that the phase alignment _can_ affect EQ, then TA would be the better place to start.
> 
> My $0.02
> 
> -Erik


Individual drivers can be EQ'd before any timing and phase work, because the individual driver singularly is only measuring the response of one driver. However, EQ work for pairs, sides and overall (basically more than just one driver) should only be done after dialing in TA and phase.


----------



## sqnut

The world is round, doesn't matter if you sail from west to east or vice versa, you will still circumnavigate the globe . It doesn't matter what you do first, in any case to start with you will be going back and forth between them, even after you have set TA and EQ.


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## gregerst22

sqnut said:


> The world is round, doesn't matter if you sail from west to east or vice versa, you will still circumnavigate the globe . It doesn't matter what you do first, in any case to start with you will be going back and forth between them, even after you have set TA and EQ.


Yup, that's been my experience. In the beginning there can be a lot of back and forth until a good formula is figured out. But once crossovers and phase / TA have been pretty well established for the install its pretty much all EQ from there. Maybe with minor tweaks to TA. 
But there have been more than a few times that after all that I want to go back and change crossover points. Then it's starting the process over but still much easier than the first time because things are still within the ballpark.


----------



## TwistdInfinity

Yeah individual driver EQ first then ta then overall system curve EQ (what you do to one side match it on the other if you don't have ability to do input and output eq) 

And yes, your likely going to go back and readjust ta slightly 

Sent from my D5833 using Tapatalk


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## subterFUSE

The tuning technique I used before Finals was to do crossovers first (in REW), then EQ individual drivers for good left/right balance (in REW), then do time alignment (by ear), and then do minor EQ for tonality (by ear) and a final REW look for visual confirmation.


Crossovers affect phase. EQ affects phase, too. That's why I chose to do TA last.


For my time alignment, I created a kick drum loop in Ableton Live on my computer. It does a pretty good job, at least to my ears. It's probably because of my years of DJing that I am just very used to time aligning drum sounds. That's what beatmatching records is all about. 



However, for 2016 I'm going to be trying to switch up the order of operations in the tuning process. I'll probably be doing TA first, then crossovers, then EQ, and then..... something else.


----------



## Babs

subterFUSE said:


> For my time alignment, I created a kick drum loop in Ableton Live on my computer. It does a pretty good job, at least to my ears. It's probably because of my years of DJing that I am just very used to time aligning drum sounds. That's what beatmatching records is all about.


Kick drum loop.. I found one somewhere but if you're willing to share I'm sure I'd love it. I also use a track I found called "polarity pulse" which is simply a click-track of sorts. When it's nailed, the drivers disappear and it's like someone rapping a screwdriver handle dead center on your windshield. 

At this stage in a new tune, I need to do ear-EQ work I think next with 1/3 octave noise tracks to see where any image shifting is happening between sides. Maybe it'll help further balance out and narrow in the center further than FR-matching the sides initially. That's the stuff with EQ I guess you can't do with a mic and REW.. Where ears become the tool.


----------



## Babs

John, speaking of overall tuning I stumbled through the playlists on the phone and gave some 1/3 octave pink noise a try to see where the stage image floats. 

Though this tune is only TA and individual driver EQ, done to match sides to a curve, which did a decent job, the noise tracks showed the center dancing all over in a few spots. So the sides are only partially balanced. Thus I've found my ghosting image issue from the last NCSQ meet that absolutely perturbed me. How we get better, right? 

Until I measure sides again I speculate this is due to constructive and destructive stuff going on between sides from existing filters applied, where areas have phase issue and also reflection as well. Lest I forget this is tuning in a glass and plastic fishbowl. 

So for the crowd here I guess the prudent course of action is Kyle Ragsdale's method from his EQ by ear YouTube video, applying cuts in those areas to center up those problem regions. Basically manually steering the image back in those areas. 

I don't have the vocabulary for this.. Group delay maybe? But I guess due in small or large part from the EQ FIR filters already applied. 

Just thought I'd share. 

John I like how you put it to me on Facebook. The struggle is real. 


Sent from iPhone using Tapatalk


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## davewpy

I personally do individual drivers level matching, individual EQ, then system XO followed by rough/measured TA. Then repeat the same by ears to verify.

The key is in what to listen for and the devil is in the details. The sequence will be back and forth.

When I started out, the first thing I did was XO. Then TA with EQ last.

Sent from my ONE A2005 using Tapatalk


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## mbradlawrence

My 2cents.....

This thread at its beginning discusses how time and volume both affect perception (ITD and IID). So.....if you t/a matching drivers (e.g., the mid one side to the mid on the other) by ear (e.g., adjust t/a of one mid) so that the pink noise in the focus under the rearview before level matching, you are only considering the time effects. When the volume of each is matched (I use an RTA on my phone or REW if my friend is available) from the driver seat, that matching will then change the location of where pink noise is centered and t/a is done again. 

If volume (e.g., level) is matched first, that is done independent of t/a. volume is volume and as each driver is done by itself, it is independent of arrival times (e.g., level matching is not affected by t/a). Once the levels are matched, then the t/a is done and 2 rather than 3 steps are needed.

Now, im sure I missed something here and am actually looking for criticism as to what it is. 

Maybe my thoughts are only relevant where ITD and IID are both present (e.g., around 1khz). Actually, looking back at the start of this thread is informative in that levels may only really matter for tweeters (see post 13).

Thanks for advice, I just have theories and only 2 builds of experience so looking for the those with more to set me right!


----------



## subterFUSE

mbradlawrence said:


> My 2cents.....
> 
> 
> 
> This thread at its beginning discusses how time and volume both affect perception (ITD and IID). So.....if you t/a matching drivers (e.g., the mid one side to the mid on the other) by ear (e.g., adjust t/a of one mid) so that the pink noise in the focus under the rearview before level matching, you are only considering the time effects. When the volume of each is matched (I use an RTA on my phone or REW if my friend is available) from the driver seat, that matching will then change the location of where pink noise is centered and t/a is done again.
> 
> 
> 
> If volume (e.g., level) is matched first, that is done independent of t/a. volume is volume and as each driver is done by itself, it is independent of arrival times (e.g., level matching is not affected by t/a). Once the levels are matched, then the t/a is done and 2 rather than 3 steps are needed.
> 
> 
> 
> Now, im sure I missed something here and am actually looking for criticism as to what it is.
> 
> 
> 
> Maybe my thoughts are only relevant where ITD and IID are both present (e.g., around 1khz). Actually, looking back at the start of this thread is informative in that levels may only really matter for tweeters (see post 13).
> 
> 
> 
> Thanks for advice, I just have theories and only 2 builds of experience so looking for the those with more to set me right!



That is essentially why I didn't do the TA by ear at the beginning, because EQ/Levels can affect the left right balance and shift an image out of center. By matching the speakers first with EQ, you are effectively removing that potential skew in the results from intensity differences. Like you said, fewer steps.

IID/ITD ranges are not brick walls. They overlap like a crossover. So even though a midbass might be ITD dominated, there can still be an IID influence.

Now, if we are talking about time alignment by more scientific means like IR, and we are only interested in arrival time then it's inconsequential if the levels are matched. Arrival time is arrival time to a computer. It doesn't care about the psychoacoustic sod center image location.


Sent from my iPhone using Tapatalk


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## Ainuke

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*



ErinH said:


> *Setting Gains:*
> 
> 
> *Intro:*
> 
> Setting amplifier gains can be as complicated as using an oscilloscope with a dummy load or simply using a $10 digital multimeter (DMM) and Ohm’s Law. A scope will let you know when the output signal is clipping (distorting). The DMM method assumes you know at what voltage for a given load the output signal will clip. I will discuss the latter.
> Since most (brand-name) amplifiers will do their rated, continuous power per spec, all you need is to connect the multimeter to the speaker output of the amplifier and raise the gain until you achieve the required output voltage while playing a tone through your cd player
> Ohm’s law is used to determine amplifier output voltage. An online calculator can be found here: Ohm's Law Calculator
> Use an attenuated tone, such as -10dB or -5dB as this will allow additional volume for ‘quiet’ tracks. 0dB tones may be used in lieu of attenuated tones IF you listen to heavily compressed music.
> Tones can be downloaded here: Realm of Excursion
> 
> The tone should be in the bandpass of what you are setting the gain for. For example:
> If you are setting subwoofer amp gains, use a 40hz tone.
> If you are setting midrange amp gains, use a 1khz tone.
> 
> 
> *Setting gains with a DMM:*
> 
> Here is a quick guide to properly setting your system gains using nothing more than a DMM.
> Reset all your cd player DSP features, balance, fader, etc to ‘default’.
> Disconnect all speakers to keep from damaging them (and your hearing).
> We will assume your cd player puts out clean, undistorted signal at near maximum. Play your tone (-10dB suggested) and increase the headunit volume to a click or two below max.
> Knowing the amps’ rated power at a given load (ohm) obtained from manufacturer’s spec (assuming they are a reputable brand), use Ohm’s law to determine voltage needed.
> For example, with a 4 ohm load and 100wRMS rating, the output voltage should be 20VRMS.
> Set the DMM to “VAC”, which is AC Voltage.
> Connect the leads of the DMM to the amp’s appropriate speaker output and increase the amp gain until you achieve this voltage
> Repeat for each channel.
> 
> You’ve now set up your system to maximize output and performance with a wide variety of music.​
> Keep in mind, when level matching, you can use your decrease amp gains as opposed to decreasing output level on your DSP. Though, I choose to maximize the gain structure up the chain as much as possible and make adjustments at one place (such as a DSP) just to keep things simple.


Okay, here's a newbie question:
Let's assume that my DSP _doesn't_ put out a clean signal near maximum (0db on my H800). Let's say 1k tone clips at -22db setting on the dial (running through digital optical via AirPlay, which I _do_ assume to not clip at maximum). Higher tones clip down to -25db. So let's be safe and say that max unclipped signal is -25db.
This is the DSP volume level that I should be setting amp gains to with a -10db 1K tone?

And for tweeters, I should be using something more like 4K tone, right? The h800 won't let me lower the tweeter crossover below 1K, and it seems that being more into the range it will be playing is better to test than right at the crossover point, no? (generally have HP at 2.2K 4th order).

Mainly asking because I set the gains using -10db tones with DSP level at -0db, and had to turn the midbass amp down to about 1/3 to get 21V (output on amp is 115W @ 4ohm). It sounded like dogs**t, distorting the top end so much it was unbearable. It's only listenable with the volume (DSP) set to <-20db, which makes me think is due to signal clipping. At that level, you can carry out a conversation above the music, it's so low...

Any advice welcome.


----------



## Ainuke

*Re: The Essentials of Sound Quality: Advice Based on My Trials and Tribulations*



Ainuke said:


> Okay, here's a newbie question:
> Let's assume that my DSP _doesn't_ put out a clean signal near maximum (0db on my H800). Let's say 1k tone clips at -22db setting on the dial (running through digital optical via AirPlay, which I _do_ assume to not clip at maximum). Higher tones clip down to -25db. So let's be safe and say that max unclipped signal is -25db.
> This is the DSP volume level that I should be setting amp gains to with a -10db 1K tone?
> 
> And for tweeters, I should be using something more like 4K tone, right? The h800 won't let me lower the tweeter crossover below 1K, and it seems that being more into the range it will be playing is better to test than right at the crossover point, no? (generally have HP at 2.2K 4th order).
> 
> Mainly asking because I set the gains using -10db tones with DSP level at -0db, and had to turn the midbass amp down to about 1/3 to get 21V (output on amp is 115W @ 4ohm). It sounded like dogs**t, distorting the top end so much it was unbearable. It's only listenable with the volume (DSP) set to <-20db, which makes me think is due to signal clipping. At that level, you can carry out a conversation above the music, it's so low...
> 
> Any advice welcome.


Okay, answered my own question. 
-10db tone at -25db on the dial doesn't get me anywhere close to 21V on the midbass amp with the gain maxed. The sub amp won't even _pretend_ to want to hit 30v. Even with -1db tone and -15db on the dial, the sub amp gets a bit over 22v with max gain. 

However, it does sound awesome now...


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## caraudiopimps

Phenomenal write up!


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## brother_c

Thanks for all the information it is greatly appreciated. You did a great job of explaining many of the more complex details as well as the interactions of changes that are made when tuning. Instead of reading countless threads where the replies seem to be the revolving 'my way is better than yours' or 'i can tune a 500 system an it will blow your doors off tuning is all that matters' It's nice to have something meaningful to revisit. Everyone started somewhere it's cool that you took the time to provide an excellent starting point for others.


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## bnae38

Test cd link is bad, looking for 1/3oct correlated pn.

Any chance anyone can help me out?


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## SkizeR

bnae38 said:


> Test cd link is bad, looking for 1/3oct correlated pn.
> 
> Any chance anyone can help me out?


PM me your email


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## Jheitt142

SkizeR said:


> PM me your email


Me to me to

Sent from my SM-G930W8 using Tapatalk


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## SkizeR

street.terror said:


> Me to me to
> 
> Sent from my SM-G930W8 using Tapatalk


Pm me your email 

Sent from my SM-G920V using Tapatalk


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## sareea

Hi Erin 
I read in one of your other comments that you stoped tuning for a house curve long time ago. 
Can you please post your new insights from the past years ? 
I have reached to a level where i know how to get a flat curve with umik and rew. 
I understood from sqnut (he gave awesome advices that correspond to what i am experiencing ) to further pinpoint my own perceived flat curve and from there to tune for tonality. 

Once I understand what I like, would it be good to re edit the housecurve (i use andy) to my liking and re-EQ using rew ? 
Or would it be simpler just to attack certain frequencies like you said from the beginning without tuning to a flat first?

I am currently training my ears with harman how to listen


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## Holmz

Subscribed


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## Swaglife81

Everyone should read this thread. You can always learn more and never stop learning. All the beginner tuning threads guys have started could have been prevented just reading this. What I love about this hobby is... It's part science, part subjective and part objective. Everyone has a different preference, style, and opinion. Regardless it's a good read even a refresher for guys that are experienced but done tune all the time or for a living


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## dcfis

davewpy said:


> I personally do individual drivers level matching, individual EQ, then system XO followed by rough/measured TA. Then repeat the same by ears to verify.
> 
> The key is in what to listen for and the devil is in the details. The sequence will be back and forth.
> 
> When I started out, the first thing I did was XO. Then TA with EQ last.
> 
> Sent from my ONE A2005 using Tapatalk


How do you keep from blowing tweets?


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## High Resolution Audio

Great write up. Subbed for reference.


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## davewpy

dcfis said:


> How do you keep from blowing tweets?


Before doing anything I input a basic set of xo settings. Usually 3500/18 for the tweeters and move it around from there.

Sent from my ONE A2005 using Tapatalk


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## subterFUSE

Pink noise at reasonable volume will not blow a tweeter, even at full range.


Sent from my iPhone using Tapatalk


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## JohnKoz

Damn, you people know things and stuff.


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## SkizeR

this needs to be bumped


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## RockamyZ

ErinH did a great job on this...a must read Imo. Thank You.


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## moodue

Just wanted to add my thanks to ErinH on this fantastic write-up!


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