# rePhase,a loudspeaker phase linearization,EQ and FIR tool v.2 for car



## oabeieo (Feb 22, 2015)

Hello everyone , 

In this thread we will discuss rePhase and it’s uses on many different platforms to be used in a car. Let’s not talk about running jriver or cpu convolution unless it’s going to be used in a car for the most part this will be discussing FFT convolution and relatively lower tap filter making techniques.
General fir filtering , we will go over the concepts, and the uses for corrections to be made in a car with fir capabilities. We will go over how rePhase and REW are interchangeable and how to make good measurements to be applied to your speakers using REW and how to make a measurement that would normally not mean much matter. 

I have invited Thomas Drugeon (a.k.a Pos) from DIY to join us., Pos wrote rePhase and I hope he chimes in from time to time to help keep my info accurate  I urge anyone who downloads rePhase and uses it to please go to surge force and contribute to Thomas for such an excellent software, same with John from REW this is a professional gesture we all should do to show our appreciation

I would rather not let this thread be crashed by noobs , if anyone doesn’t know what fir filtering does or is , please educate yourself first on what the general terminologies are. In no way am I trying to drive anyone away however fir filters are complicated and it takes a lot of time to explain the basics and a simple search will give you all the answers you could possibly want. Once familiar with the basic terminology than please join in discussion.  


So, I’ll get some of you up to speed on what I personally have done. In my car I use 4 dsps, 1 Dirac Live ddrc22di and 3 minidsp 2x4HDs. My system consists on a clairion headunit NX-706 running 96khz toslink with volume control embedded so volume is done completely upstream. That feeds into the Dirac Live 2ch Dirac room correction box that I use basically as my global EQ and the first stage of my correction. Toslink out of that into an active 3way toslink splitter that divides the toslink into three that goes into each of the 3 minidsp 2x4HD minisharcs and than analog single ended RCA out to 4 amplifiers in a 4way system. 

The 2x4HDs is where I use the minisharcs and make 8 seperte fir for each driver (channel) to have , eq, crossovers, phase corrections , and magnitude corrections, spots where Dirac needs small adjustments all in each fir. That is where I will be talking about rePhase and how it is setup in my car.. however as far as rePhase , my system is unique and I could get the same results with just rePhase alone, I chose to use the Dirac box so when I change equipment or drivers out setup time is minimized. (I’m a freak and change speakers a lot) in a normal persons system no Dirac box would be needed. RePhase can get everything done right the first time. 

So now I would like to talk about what it is exactly in a car that makes fir filtering really make a lot of sense. Mostly a linearized mid-bass HPF and the subwoofer LPF is a must have. But beyond that basic everyone should have in a car the principal of how “phase” works in a car, how to read the measurements and how to apply a measurement using rePhase. And lastly after all the corrections are done for each side (left and right respectively) how to apply another correction on top of all the separate corrections to offset not just the timing differences between Left and Right but to correct for the comb-filters caused by interaction of the Left and Right channels caused by the time of flight delay. :kid:

I will share this article from Dirac Research first to get everyone up to speed on exactly what that correction looks like and what has to be done to achieve it realistically in a car with the use of fir and rePhase! :devil: 


:drummer: without further ado the article from Dirac Research: This is a must read for anyone trying to use fir in car. As I can’t go into everything in one post this thread will be evolving over the next weeks and months.
I’ve dug myself into every rabbit hole possible but once I was able to start to apply these type of filters is when I fell into the dam glory hole, and 2ch audio in car becomes an entirely different animal and takes on things never thought possible. Once one has begun to get this type of filtering, it becomes silly to think about all the ppl bickering about trying to get the “time alignment” controls set perfectly, when actually “time alignment “ controls alone seem to collapse the stage and compress the recording so it’s right biased. As inducing a delay on the signal causes a serious comb filtering issue between left and right , than the actual time of flight distance compounds the problem even worse.

Here’s the article for everyone to read. I’ll stop here for the night and continue tomorrow or the next day. 

https://static1.squarespace.com/sta...+Obtain+a+Good+Stereo+Sound+Stage+in+Cars.pdf


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## oabeieo (Feb 22, 2015)

Here is a couple links we should all read to save me time in explaining some basics that REW and rePhase do. 

https://www.computeraudiophile.com/...nerate-amplitude-and-time-domain-corrections/



And the link to the multi parent rePhase thread over at DIY for anyone who’s interested 
rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - diyAudio


I will start gathering screenshots tonight to get into building the basics on how to make a lovely 2 seat listening position car or a down right deadly 1 seat listening position car, with standard door / dash speaker placement without minimized PLDs (path length differences) and start to jump right into it


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## dgage (Oct 1, 2013)

oabeieo said:


> Here is a couple links we should all read to save me time in explaining some basics that REW and rePhase do.
> 
> https://www.computeraudiophile.com/...nerate-amplitude-and-time-domain-corrections/
> 
> ...


But we like your explanations. 

"Daddy, can you tell me another story?"

LOL. Thanks for the links! Have a Happy Thanksgiving everyone!


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## oabeieo (Feb 22, 2015)

I will be speaking now in regards to minidsp 2x4HDs , or minidsp mini-sharks tied to a conventional DSP like a helix or with any range of added DACs or with any minidsp platform dsp like the 6x12 with added mini-sharks. 


The analog devices shark processors are used in minidsp products and in other products like the najda boards. The ADSPxxxx is basically a 32bit floating DSP most are 8ch output and use a AKM codec as the DA converter which is quite ample for a run-of-the-mill chipsets. 

Hopefully everyone has download a copy of rePhase and started to play with its controls. 

You open rePhase to a blank setting. You can import your REW auto eq right into rePhase and maximize the low filter tap at 2042taps per 2 outputs (4096 on 4ch) at 96k (24/96) input or get 4084taps per 2ch at 44/48k inputs. So if your using a mini-shark ( a raw ADSP chip on a board connected through the I2s link to a compatible DA or input bus from minidsp) instead of the 2x4HD you could probably get away doing the entire correction in the fir by itself without the need to deploy any iir based biquad banks for PEQ or APF functionality. A purely fir based correction would be sure the have no recursion. Meaning, the silence of filter ringing caused by iir infinite amount of ringing in the attenuation cycles as samples are fed back into the input endlessly.
This “noise” is almost impossible to hear unless you’ve been making filters an obsession (like me  ) otherwise using iir peq banks for minimum phase adjustments or even as all pass functions go as well. Using as much iir controls when running 96khz on a mini-shark (2x4HD) will be a must to be able to get the taps needed to make usable change to the system in the fir banks. 

In that instance the user can start in rePhase and do the Minimum phase portion first and export raw coefficients into the iir peq slots so there’s no need to jump platforms, the entire correction can be worked from rePhase alone. After your done working the minimum phase parts of the correction you can simply move forward and generate some fir! 

If running a multiway and using multiple fir, make sure to give file names that include the tap setting, bit rate, general description on target import bank....that way when your sending 8 files you can manage what is actually loaded in the bank and can help when needing to use the delay setting in dsp to offset the fir fft length. For example a file name could look like ;
Left6G40ND-8ms-LR2LPF150hz-LR4HP1600hz-linphxo-.bin 

That would tell me I’m working with a 6.5” Beyma 61/2 midrange with an fir that has 8ms of delay , a band pass linear-phase crossover and there crossover points and alignment in the file name. So when it’s a live fir Loaded in dsp I can have enough info about the functionality of the fir when bouncing around channels and not forgetting. Think of the long file name as leaving yourself a breadcrumb trail as you go *WAY down deep* in the rabbit hole. That file name will be your only way back to reality after going all night at it  

Think of it as a guiding fairy 


*Oh come on andy! Where did that come from  *_*”more fairy fun than ever!”!Lmao!! *_
Making sure to manage and save a file or configuration to your platforms is vital to working an fir. Spitting out a 10min fir will get you in the ballpark , but to hit home plate going back and making small adjustments to your dsp and fir is in the order of things so getting good habits will only make it much easier to see into the depths of it when working a correction. 


Filter output delay can be managed and integrated as part of each fir if there’s no outboard delay controls available, you just have to move the impulse x amount of ms forward in the impulse centering controls on rePhase. I find it easier to use the delay of the fir ? And whatever speaker (most likely the sub) has the longest fft in “ms” will have no delay added , examples;

If sub fir is 10.63ms because you used a 2042tap fir and use a rectangular window and center the impulse on the peak energy if the ir and use a 1024t fir on midbass with rectangular windows And center on peak centering with a delay of 5.35ms you would subtract 10.63-5.35 and plug the answer (5.28ms) into the 
Midbass delay setting to get both speakers to play at the same time. If you add the 5.28ms before the ir peak they will both be 10.63ms delayed fir. I like to use delay , it’s faster for me and easier to manage. So bring a calculator to the car with you. You’ll need it for this and much more to come 
That’s enough for tonight, I’ll gather screenshots and go more tomorrow 

Chao


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## oabeieo (Feb 22, 2015)

dgage said:


> But we like your explanations.
> 
> "Daddy, can you tell me another story?"
> 
> LOL. Thanks for the links! Have a Happy Thanksgiving everyone!


Hehehehe


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## Jscoyne2 (Oct 29, 2014)

Nice. Thanks for your effort 

Sent from my SGH-M919 using Tapatalk


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## Elgrosso (Jun 15, 2013)

Keep going I think for once I understood 50% 
Good strategy to go slow and step by step.
But in the same time I don't think I'm ready for manual fir yet...


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## oabeieo (Feb 22, 2015)

Elgrosso said:


> Keep going I think for once I understood 50%
> Good strategy to go slow and step by step.
> But in the same time I don't think I'm ready for manual fir yet...


Yes, 
Step by step, 
It will take few weeks to get to the red meat. But will be worth it for a lot of ppl just strtin out Or thinking about taking the plunge .


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## oabeieo (Feb 22, 2015)

So we’ve talked about different convolution engines 
And talked about file naming and general organizational aspects of of filter management . 
We’ve read the basic principles on the time domain and the near side bias problem in cars.

So it comes to a fork in the road where one has to decide which approach to take. Solving the near side bias problem for one seat or two. It will take a completely different approach on each method. This is where we need to start running some numbers to see what’s achievable. 

We need to measure distances of the car dimensions, and the speaker positions relative to the car dimensions and where you or two will be sitting. 
So let’s make an example car. Let’s say the car is 

5’ wide (door panel to door panel)
4’ tall (Floor to ceiling )
7’ long (corner of windshield to back glass corner) 

(You can take some 1/2space measurements and 1/4space measurements as well to help make sence of some odd reflections that may appear in measurements)
For example we’ll just stick the the basic room mode problems.

So if we had a door mounted midbass/midrange combo driver we could say there 5’ apart. 
Sound will cancel when they are 180° Out of phase. Let’s run the math and get those basic numbers and determine what frequency we will start to see a combfilter start to emerge.

So 5’ wife car and we sit let’s say 36 inches from the left speaker and 60 inches from the right speaker. Wait that’s more than 5’.......exactly, if the speakers are mounted in front of you it will add a few inches to the total measurement, so that will change the “math” a bit and will change the way the combfilter appears on measurement, but we can still see where the direct on axis energy will collide and get in the ballpark.


Keep in mind REW has a room sim feature that does all this for you, it’s excellent in fact but someone needs to ask John at REW to make the room size variations smaller because I can’t get the smalles setting to match the size of my car. If you have a big truck you might get lucky. 

So let’s take the distance the speakers are offset from center to center now , about in front of the lower part of the driver seat directly in front of the speakers , let’s say we measure 36’ driver and 48’ passenger , okay now the numbers add up to our total 5’. Let’s run that and see what we get. Any frequencies that wavelength in ms or feet whatever you use when the ratio is 1:1/2 it will cancel, if you sat exactly double the distance away from side to side it would be easy to calculate the frequencies. Find the frequency at the 1st comb and work your way up to the 3rd comb that’s where we will focus an fir correction, at higher frequencies we will use IID principles and HRTF in general to get it right. Because both ears matter more for each side up high and down low even SPL through the magnitude is dominant between left and right respectively relative to where your seated 

So let’s get the basics on what we’re doing 
*Speed of sound = 1125.33ft~sec *

Knowing that we can use any online calculator and determine just about anything or use your calculator manually. 
Let’s determine how many ms the car is wide first 
At 1125.33ft sec that means there 0.89ms in one foot (approx 13.5” is 1ms) some ppl say in car a ms is equal to a foot, which is almost right, and definitely works good but once you get farther than 1foot it drifts off so keep the math tight and do it the right way. 

*At 5’ wide that 4.45ms wide car*. Let’s determine what frequency is 4.45ms long now. That roughly 252hz (4.465’ 4.99ms) it’s close enough to 250 to sadly say you could calm it the 250hz band. So what happens with a car 5’ wide at 250hz? Each speaker will be able to play one full cycle without any obstruction, this will be the beginning on the “Schroeder” or room mode frequency on the X axis. Now what happens in between that 250hz time path is where things get dicey. It’s good to know the math in “ms” how far you are from each speaker, and how many ms the car is in dimensions so we can calculate where to start and stop out room correction, once the frequency is longer than the room than we will start to be more and more certain the speakers will behave more in a minimum phase way. Also the pressure in the car will create nodes of high pressure and low pressure at your position in a different manner so. So any changes we make in those lower frequencies we make to amplitude will also affect phase directly and proportionally. At frequencies above that we will have to start to look at the combfilter issue caused by the two speakers interaction mostly in the x axis which we determined to be 4.45ms So 250hz and up. 

Andy W. Has stated many times that 250hz is a fundamental frequency in most cars and almost all cars it’s safe to say 250hz is a good spot to start any all pass functions. And he’s right. I’ve discovered this as well and 250hz is a great place to start the first all pass to combat the 1st combfilter. We know that the car is 4.45ms so the first cancellations that are at frequencies shorter than 4.45ms we could call it the 1st combfilter. This combfilter will appear in different spots around the car and should be the *inverse* of itself on the opposite side of the car. This first combfilter is where we need to really focus a lot of what we do in the fir on and all eq work needs to be worked around this 1st big visible notch in the magnitude. 

This leads us to get to know our cars. We can’t just cut a measured frequency in half and say arbitrary that’s where it will cancel , we need to take measurements of each Left and Right separately than measure Left and Right together and watch the notches appear. Because of funky 1/4space loading and the *transfer function * of each Left and Right respectively we might see the 1st notch show up +/- 1/2 octave Or so. 

So one has to look at the data collection of measurements of frequency response and look at the interaction between left and right using balance controls, and the tape measure measurements and the physical spaces and put together a picture of the transfer function of the car itself as a whole. Once you can see what frequencies cancel and what frequencies are reinforced because of the room and the numbers are pretty close to the hard math you can start to generate a correction filter that will work. The less PLD there is the more achievable a two seat car is the the HF. As frequency rises the need for less PLD becomes a serious issue. But on the same token the higher in frequency we go the more HRTF takes over and some IID properties of hearing take over where amplitude can make or break a two seat car before PLD will. So determine how many ms. That is past 1khz. 1khz is mostly a frequency where depending on PLD but most cars you can start to begin to slowly just get an even mag response between channels and get a clear image for both sides. The worse PLD is it might get into the 1.2-1.4khz before we can start to apply that. Anything under about 2ms is good to use 1khz as a great reference to start making amplitude the driving force for localizations. So basically between 250hz and 1khz is where I call the “diffuse” region and good minimum phase behavior is the main driving force To get a good transfer function. 

Obviously speaker placement should be considered first, if the speakers can be placed in a way the will radiate there energy or sound power evenly through its passband entirely evenly on both sides for both passengers or at least the driver that will be the single most important aspect to getting a good fir to be easy to manage and not fight you every step of the way through every peak and null.


This may seem like a big ramble but it’s important on how we approach the fir, we will see in further posts how this all is key to getting a correction filter to do what you ask of it 

Still getting screenshots prepped. More tomorrow after turkey 


Happy thanksgiving to everyone!


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## oabeieo (Feb 22, 2015)

Sorry guys , Black Friday at the shop is tearing the arse outa me. 

Haven’t had time, got to get through this weekend. 

One thing I also wanted to talk about is the 70-80hz botch we all have. This notch has a cause but also an illusion, but truly no different than the notch you have between 250-600 with the big fat peak behind it. 

Being we mount subs in the back of our cars mostly I’ll be talking about a rear mounted sub , although omnidirectional yes, but placement will lend to the transfer function on the loudspeaker. The sub frequencies are much longer than any dimensions of the car, so as the sub starts to compress the air in the back of the car the air will build momentum as the air molecules get tighter together, they start to look for a path out and *transfer* towards the other possible escape paths with low air resistance, the size of the car that is longest will is the last escape path for the high pressure air to move to, once the long part of the car starts to compress it will be proportionate to the amount of air the speaker pistoniclly can produce at one fourth cycle. (We can hear in 1/4waves under pressure) as the pressure itself will cause our eardrums to make the rest of the motions to complete the cycle (but at 1/4 the volume as well) that’s why we need large subwoofs. Need to move lots of air to hear the bass proportionately. So when it’s at half cycle at that frequency the air that had had already started to build pressure is rebounding back towards the opposite node (pressure zone in cabin depending on transfer function cant say where that will be) but can say as it rebounds it will interfere with the 1/2 cycle and cancel itself out. This is why there’s a huge dip in the impulse response in a car. The car body pushes back the air and acts like a diaphragm and causes a big ripple in the IR. 

This also means where it’s roughly 1/8th we will get some reinforcements. Say hello to cabin gain. So cabin gain makes the 70-80hz dip “seem” a *lot* worse than it actually is. Even though the dip looks severe and everything you to to try to EQ it doesn’t work we will go over some ways to make the transfer function more even with fir so you can get it to sound smooth without the need for gobs of time smearing PEQ notches or boosts to make things worse. 

Once the cabin gain has been either damped or EQed down , you’ll be surprised how much more power you can give the sub without blowing it up. The big dip can be salvaged by properly addressing cabin gain , gain structure, and using hass techniques to “move” the bass forward using the midbass while breaking up the cancellation node a little bit using the midbass fir. 

The notched can fairly easily be made to go from -15db to -3db without adding any other drivers or using bigger drivers. The one thing I’ve noticed about when using hass technique, is that the leading speaker needs to sound very good and have *no breakup modes or fs issues* even if it is down -10db compared to the speaker on the trailing side of timing it will make all the nastys from a speaker come out with laser precision. Also don’t go for absolute max hass effect , yes you can get the bass to move up front but push timing back .2-1ms , keep the nastys hidden by the cancellation. Amplifying fs breakup by 10db isn’t pleasant


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## oabeieo (Feb 22, 2015)

WELL ! 

I got a bunch of screenshots and started writing a post in PhP and discovering that was not the direction I want to go with this thread , 

I was basically demo-ing REW rePhase filters in auto EQ and decided I’m going to skip magnitude correction from 20-20k and go through it in a multiway. 
There’s that link I posted that goes through it anyway, so I sorta wasted tonight but still got a couple screenshots I’ll share , 

Sorry for the setback, i want the screenshots to not be confusing and to not go way off on. Tangent and have to spend weeks clarifying details because I went in wrong order. 

Back to the drawing board to capture screenshots. This time I know a way to explain technique and reasoning that will be much more compatible with any magnitude correction approach or any mag correction using various DSPs. So I’ll build on this thread and just do a standard magnitude correction using EQ from 80hz to 20k flat using whatever method as long as it’s averaged multiple times and separate left and right EQ work starts at 250hz and up. Under 250hz just averaged Left and Right simultaneously and eq -ed together. 

Weather auto eq or manual a flat mag is a good baseline to start. 

This is what I would do as a suggestion but obviously you know how your car behaves and sounds with your gear and install , so I want to leave out my personal preferences on things that have different routines. 


So , multiway eq work goes as follows 

1 measure each speaker with the crossover turned either off or at least 1 oactave past its desired crossover point. close mic with RTA and moving mic method in a slow averaging window and circle the mic around the edge of speaker , to capture its off axis output and all around the speaker. In the area where the crossover will slope knock down any peaks with PEQ. Whatever your method, this will ensure the linear phase crossovers that you will apply in rePhase will do for one , but also keep the phase of the system from making a change that would be difficult to track down and isolate later. Don’t worry about making the crossover slope area super flat, just no peaks, the rest leave alone unless it’s drastic and more than 6db, maybe consider a steeper filter or different crossover if speaker can’t play to full power an oactave below its crossover point. This will also ensure that any radical eq changes we make later in the pass bands won’t move the crossover slope shape to a frequency that is unsafe of too low for the speaker. 

If your wondering how that can happen , think about it, if you cut most of the pass band and than gain it up to compensate for the lost signal the crossover area that may not have eq would be untouched thus making the amplitude in the crossover higher in relation to the passband that has most of its energy cut. That would extend the crossover acoustically way past it’s crossover point. 

We going to sorta try to avoid having to stack a bunch of IIR crossover on top of fir crossovers to acquire an acoustical slope. But sometimes just adding the extra iir crossover on top of a fir crossover might be needed to get the gain structure required to fill some nulls. If combing is severe where we will be using more than 6db of eq you may want to start with a little more gain on that driver compared to other drivers when eq work is done. 
I’ve had as much as 10db difference in gain between drivers in crossovers pre eq just to get them to work right at crossover. So do what you do with your gear, get a flattened mag and leave the nulls alone. I always EQ everything down and never go past -12db and I get the mag down within 3-6db of the bottom of null. Figure if you eq the mag down to the bottom of each null , at the bottom the speaker will get spitty if driven to hard and a soggy - not snappy non transient sound will be the result , although more spectral and better sounding at low volumes , leave the nulls down just a wee bit, but definitely eq everything else down a bit as much as well. 

I’ll put more up soon, I’m organizing!


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## Saturnsl2lover (Mar 26, 2017)

So oabeieo you have 3 hds with an optical splitter and that works no signal degradation am i correct? I was thinking running 4 hds with an optical splitter in my bmw using toslink on the optical ring bus. Ill just use a mobridge converter. I have 14 speakers now if i dont add a seperate sub if i did that will bump me up to 16 channels so 4 hds. Maybe i can run dirac on each hd later i wonder if that's overkill lol


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## oabeieo (Feb 22, 2015)

Saturnsl2lover said:


> So oabeieo you have 3 hds with an optical splitter and that works no signal degradation am i correct? I was thinking running 4 hds with an optical splitter in my bmw using toslink on the optical ring bus. Ill just use a mobridge converter. I have 14 speakers now if i dont add a seperate sub if i did that will bump me up to 16 channels so 4 hds. Maybe i can run dirac on each hd later i wonder if that's overkill lol



yes 3 HDs and a dirac 2ch dig in/out in front of them 

its toslink! cant degradate light, Its just optical light. No degradation what so ever and no noise what so ever. 

Only think with toslink is jitter, and that has nothing to do with a spliter. 
or any other digital source. Its a problem with clock syncing. and power supplies for digital clocks etc etc ...


no it works marvelous. The active split-er allows syncing. The passive spliters work without sync but may be a sample off from one another.

The HDs are excellent. At first I though they were a bit janky howver after really getting into them , they are quite good. I just thought a bigger box with nicer case meant better engineering. not so. for the size and the fact they are 12v and car work with zero mods and have zero noise. I love them


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## oabeieo (Feb 22, 2015)

Okay here is a screen shot of rePhase Start up screen.

You can select a output file from just about any format you could think of.
CPU convolution, stereo convolution, mono, 24bit fixed, floating, 32bit,,,etc etc ..


you can see many different tabs for the controls, I wont post screen shots now of all of them as the thread develops I'm sure we will see them all

you can do phase EQ, normal EQ, linearize a existing crossover, or make a linearized crossover in fir, add delay, get rid of that annoying box rise and GD caused by a port with the box tool. So many thinks to do. 

you can invert a measurement (or anything on the screen) so you can manually draw out a correction, you can invert a correction, it can compensate for any existing passive crossover alignment with the compensate function (which I have found many more uses for than just that) 

Once you hit the "generate" button it makes the fir to the directory listed and named what you name it. Once generated the result and prediction will be overlaid. As long as the result is close (meaning a lot of things by close we will talk more about later) it will sound good. You can simply change windowing, centering values, tap amounts toill you can get a result that is acceptable. At first I though it had to be "perfect" to what the graph shows or ringing will happen. That's not entirely true, and again we will talk more later about that. Generally with small tap high sample rate filters a mixed phase filter is ideal. A combination of either iir and fir filters or a combination of minimum phase filters and linear phase filters, (one in the same, just outside of small tap boxes the ability to everything in fir becomes a issue so iir will be used in combo to get rid of ringing, per-ringing, and get the acoustical shape to match the fir. 

You can also have REW do the eq work for you. I will not go deep into that, as I stated earlier but will include a screen shot to see it..

Here's a few screen shots to get started. I need to do some trial measurements now so I can import some very basic loudspeakers responses so the phase isn't a mess so as I explain things, no one will get lost when It comes to the more complicated things like summing two driver responses and how to identify problems and what to work and not work on...


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## Saturnsl2lover (Mar 26, 2017)

Well i am following. Of course now the hds have the dirac add on so maybe quad dirac AND rephase and REW?


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## oabeieo (Feb 22, 2015)

Saturnsl2lover said:


> Well i am following. Of course now the hds have the dirac add on so maybe quad dirac AND rephase and REW?


 Sound a lot like myself  A little bit freaky about DSP

In all honesty Dirac is a excellent tool hands down , 

But for the guy that can settle on a set of speakers and settle on not changing out his or her equipment every month and spent the time with rePhase one could really nail down a correction that performance is better than a Dirac tune with no mods, a tweaked Dirac tune is very good , but for those with the manual fir rePhase can do wonders . 


Can’t wait to use Dirac car platforms, not available as of yet but from what I hear is in the works, I have very high hopes for it.


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## Saturnsl2lover (Mar 26, 2017)

If i can keep up with the manual process as you outline then i dont need dirac. 4 hds and four 4 channel amps to get me going with factory speakers. Ill keep an eye here for help ?


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## Elgrosso (Jun 15, 2013)

oabeieo said:


> Can’t wait to use Dirac car platforms, not available as of yet but from what I hear is in the works, I have very high hopes for it.


What do you know about that my friend? I just saw few posts on the mini forum.


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## oabeieo (Feb 22, 2015)

I read a lot and ask a lot and have lots of ideas 


Don’t wanna say much more than that


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## oabeieo (Feb 22, 2015)

Okay before I move forward , I need to make sure EVERYONE understands this. There is a 100% chance you will get completely lost and the rabbit will eat you for lunch.

So , take 15min out of your day and read this entire page start to finish. Don’t be a ***** and get board 1/8th the way through or tell yourself you can do this without understanding this principle. . Read the whole thing. 

https://www.roomeqwizard.com/help/help_en-GB/html/minimumphase.html#top


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## oabeieo (Feb 22, 2015)

:worried: Okay so (that was a joke BTW he he)


More coming soon. 

Once I get going and get all my info organized so it makes sense we will see how things need to be *invertable* 

Meaning , you have to be able to reverse the polarity and have the opposite happen. If there’s a reflection that has *diffraction * attributes to it , we will see that diffraction can not be un-diffracted. 

We will be discussing how to pull phase around and get a reflection to behave a bit better so that it’s high and low air pressure spots are not in a offensive spot.

Keep in mind moving phase can act like moving sound forward or delay but it all comes out at the same time. Nothing is delayed , it’s just structured a bit different that’s all  

More soon


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## oliverlim (Dec 5, 2016)

So you are using rephase to generate the filters for the minidsp unit. How does it sound? I believe you used APL1 before. How does that compare with what APL1 does?


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## oabeieo (Feb 22, 2015)

oliverlim said:


> So you are using rephase to generate the filters for the minidsp unit. How does it sound? I believe you used APL1 before. How does that compare with what APL1 does?


YES with minidsp 2x4HDs, I go over a few minidsp products in the first few posts that are compatible. But with focus on the minisharc (2048t at 96k or 4096t at 48k) 

APL1 , almost had the trial up and running , still want to try it. Got massively sidetracked and had a revelation with rePhase. The APL1 trail is on back burner. 

It sounds stellar (I hit the glory hole by gosh) (I’m such a nerd) 


My car is definitely that car that sounds nothing like it should relative to where the speakers are placed. It definitely sounds like a ton of DSP is applied , but not in a bad way at all, quite the contrary in fact. Can tell lots of dsp is in use, but dsp done well. Very dynamic , no harsh spots, goes very loud with stage staying exactly the same at all volumes. 


Once I went to a concert (smallish venue about 2500ppl) and it had a very similar effect. Except in the car it’s reproduction and has all the artifacts of a recording engineer, solid center , etc.... but anyhoo my car reminds me of that venue. I remember the concrete building it was in seemed to have no reverberations and seemed like i couldn’t hear any echo. 




———-


Sorry for slow to post, been very busy at work, the holiday season in car audio is slammin , we’re doing a lot of installs working long hours, 

But after January I should start having more time, I stepped down from management to move to a store closer to home, (5hr a day drive too hard on the Fam) so I’ll finish the year in my store than transfer. Makes me pretty sad, I love that position and worked pretty hard to attain it, but need to open new chapter. So, I’ll have more time for everything else. (Should be a good thing for now) 



I saw this pic and stole it from a site and wanted to share it , it shows the different zones as far as acoustic behavior is concerned in a room. Each “zone” needs to be addressed separately as they all react differently to the overall manipulation of the audio signal. 

I have a lot of screen shots made and am making posts in HTML and PHP , as soon as I feel the info is as accurate as I can explain I’ll post. I sometimes get ideas and thoughts mixed so I want to be careful to not have misinformation or convey a point that is right but comes across incorrectly. I assure everyone it will be through enough to make good use of it


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## oabeieo (Feb 22, 2015)

so, here are a few pics of some measurements at the listening position. I did a left and right , no-EQ just crossovers (mixed phase, some iir some fir and some half and half) you can see the phase wrap wheres there's iir crossovers or a time delay. 

I added a 5cy IR window to everything. Than I rand Dirac Live , No pre-tune, with a flat response (cant remember if I had tilt, either way just for conversation. Lets look at what is going on before and after. 

Ill post about it after so I can write from my phone instead of this bulky computer..Also a view of the rePhase files and minidsp configs


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## oabeieo (Feb 22, 2015)

So , 
I used Dirac as a fast example of what one of the big goals is just for speed 


If you look at the left and right no correction (can see what measurement I have open) 
And left and right with correction it’s pretty easy to see what’s going on. 

Phase is not pancake flat. There are wraps still , and that’s OK! You can not hear a phase change on a single speaker, and that’s the point of this post to get you going on what to do. 
It’s when you introduce the other speaker/channel (depending on what your doing) the goal is to have left and right phase and magnitude to be as close as symmetrical as possible (at least for now, later we will go apeshit on APFs and shape the stage and completely disregard the symmetry, but that’s way way way down the road. For now, just get them looking symmetrical)

(Sorry for not making my limits the same but you get the drift) 

As long as the Left channel vs right channel are making the same shapes in magnitude and phase it will be “phase coherent “ (not a real word but can be used) 

Keep in mind if your spl meter is calibrated on your measurement software and you take measurements from the same point in space on the driver side , the passenger side will look like it’s phase is more “positive “ throughout the magnitude. That’s what you want. You don’t want to try to “correct” the magnitude of the phase to match the phase on the left by correction. I moved my scroll bar and so you can’t really see it because I added windows because I took these measurements a while back. The reason the right side will be positive is it has more delay, (time of flight, unless you window it out which you will) but at first the right will look a few degrees positive in the linear view and simply more positive in log. 


If you were to try to correct for that the center image would be directly in front of you instead of between the speakers, you want to let the magnitude in “db” move phase for you throughout the FR magnitude. By simply reducing the output on the left side will move the phase with it in the opposite direction. After all your signal delays are done and FR is made to your liking and is symmetrical on both sides, let the minimum phase behavior move the phase magnitude. We will only be focusing on the shape of the phase vs Left and Right and how the shape relates to the FR magnitude. 

If we look the Dirac Software made Left and Right look dam near exactly the same on Left and right in FR and phase (BTW FR means frequency response, but you knew that) 

This view is in the linear axis just for viewing. Even in the linear axis you can see some issues where phase isn’t exactly the inverse of FR. It could be just the view , the measurements, the window, but it’s always a good idea to start to pay attention to those anomalies with each view window your in. Start to get a feel for the room. 

But even with this , one could set the magnitude so that FR has the most symmetry as possible between channels, asymmetrical crossovers can be very useful and asymmetrical level setting between channels in a multiway can be very valuable before running to the eq. 
Your crossovers can make a lot of good changes for you and save a lot of backtracking later down the road. 

After EQ Left and Right you should have a good baseline on what to do next. There will be back tracking, be ready, trust your ears also, Ive skipped over hundreds of validation measurements by just using my ears, once you get a hang of what a phase change will sound like when played against its other stereo half you can get a knack for what’s off and needs fixed to get left and right to be more symmetrical. 

Once you get some real symmetry happening with crossovers and levels and eq , than its time for a few more sets of measurements. The phase changes you make you will want to compare what individual drivers are doing as far as phase behavior vs when it’s paired with its fellow drivers on like channel and that will let you get an idea of what’s happening when they sum. 
When you take your 2nd set of measurements, and have a decent flat (or reference target magnitude) between left and right its time to do a different viewpoint. You’ll want to get rid of the time of flight delay and generate the minimum phase response. Getting rid of time of flight delay I like to just remove samples from the IR, the acoustic timing method will use the HF driver as a dominant reference point. I like to loop back and move the IR manually, I like to see what happens as I move the delay out of the IR window. If you center before the peak, in middle of peak, or where ever , find out how long your foriour is, make sure Left and Right have the same windows applied and same settings in your analysis. That way if you remove 2 samples or 200 samples. 

Once you generate a minimum phase you can now see inverting that’s happening as amplitudes change and how the phase should look a lot like the mag inverse. 
In any spots that do not look the same between left and right and that do not follow the general inverse shape of the FR those are areas we will want to look deeper into. 

For a two seat tune simply (ha simply it’s no easy thing) make measurements of the passenger side as well, the the driver and passenger to have the similar phase response that is a mirror of itself. Does not have to be perfect (it can’t!) just not swing to -180deg to +180deg between combfilters. Get it so they both +/-90deg on both sides to mirror each other. Some spots will be achieved some won’t be so much but just make it best you can. This will get into more later, let’s start with a one seat tune before we whip out our shlongs. 


Any phase change will affect FR as well, and any phase change in one area will move phase all the way through the spectrum in some areas. After a correction is made its pretty important to listen and RTA with some noise and re adjust eq or levels. Once you change phase the way the speaker behaves in its space will change the FR as well. Could make some areas louder with reinforcement or quieteter with interference. A phase change will change the interferences all over the place and a new measurement must be made after levels are re balanced. Once you get few the first few you’ll notice some cancellations actually sound better and should be kept as it may add to the entire room character better it’s sound envelopment or spacial impression. 

Next posts will be covering those types of views and how to import the measurements into rePhase and start making corrections!


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## Miguel mac (Sep 28, 2009)

Hello, I run carpc win10, i have only 2 ch output (Essence ST card). I can play active with Amp xovers (arc audio SE4200 + 2300 amps) and run Rephase in my carpc? , 

my settup is tweter, midwoofer + sub. (seas magnum + seas W15cy001 + L26roy)


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## LumbermanSVO (Nov 11, 2009)

oabeieo said:


> But after January I should start having more time, I stepped down from management to move to a store closer to home, (5hr a day drive too hard on the Fam) so I’ll finish the year in my store than transfer. Makes me pretty sad, I love that position and worked pretty hard to attain it, but need to open new chapter. So, I’ll have more time for everything else. (Should be a good thing for now)


I knew you've eventually jump stores, the drive on 25 is absolutely insane.


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## oabeieo (Feb 22, 2015)

Miguel mac said:


> Hello, I run carpc win10, i have only 2 ch output (Essence ST card). I can play active with Amp xovers (arc audio SE4200 + 2300 amps) and run Rephase in my carpc? ,
> 
> my settup is tweter, midwoofer + sub. (seas magnum + seas W15cy001 + L26roy)



Yes in fact that’s a sweet way to do it (if your ok with car pc lag) but yes with 
Jriver as a convolution engine and rePhase you get get a very high tap count.

Keep in mind CPU convolution and FFt convolution are a bit different, but a lot of ppl use a pc as a DSP and run half million tap filters with good success. I tend to like fft better with shorter impulses (the smearing stays manageable)


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## oabeieo (Feb 22, 2015)

LumbermanSVO said:


> I knew you've eventually jump stores, the drive on 25 is absolutely insane.




YES it is downright brutal ! 

1week at new store so far so good, it’s a bit tough at times going to a new store because no one knows how intense I am with car audio so I have to throttle way back to mind others egos and feelings to a degree. But so far so good , 
24hrs a week more with the fam...









——————————————————————————

More to come later , after the Super Bowl. I’ll be back on my game and get to more rePhase fun .


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## Bnlcmbcar (Aug 23, 2016)

oabeieo said:


> Sound a lot like myself  A little bit freaky about DSP
> 
> In all honesty Dirac is a excellent tool hands down ,
> 
> ...


If I run the DDRC-22D before a Helix DSP Pro and perform the Dirac Live measurements while the Helix is outputting a R+L summed center channel, will the output of the center channel be similar to the Dirac Virtual Center feature from their automotive sound systems?










“Dirac Virtual Center can be applied even when a physical center speaker is mounted. Depending on the input mix to the center speaker, the soundstage tends to be rather narrow, in particular when the center speaker is fed with a left + right signal. Dirac Virtual Center allows for a tuning process where the physical center speaker is used to stabilize the center image rather than creating it. This results in a more stable and wider soundstage and a more distinct phantom center compared to when only a physical center speaker is used.”

https://www.dirac.com/dirac-virtual-center/


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## Elgrosso (Jun 15, 2013)

Nope, Dirac live doesn’t have all the fancy new stuff for oem yet.
And the 22d is only stereo so no way to create a filter for your center only.
But you could try an ms-8 for that, Logic7 works very well!
Best next step I guess would be the future Audiofrog dsp.


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## Bnlcmbcar (Aug 23, 2016)

I was hoping through some awesome Dirac magic that it could help alleviate the mono center channel that most DSP users are stuck with due to companies not investing the pricey licenses for center channel processing. 

Something along the lines of:

Since the center is R+L...

Only the R signal would come out of the Center in conjunction with the right speakers when performing the right side tune. And only the L signal would come out of the Center in conjunction with the left speakers when performing the left side tune..

So when played together through some magic that oabeieo could explain we would in a way derive the Dirac Virtual Center

Sighs.. patiently waiting for that Audiofrog unit.


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## Holmz (Jul 12, 2017)

Bnlcmbcar said:


> ...
> Sighs.. patiently waiting for that Audiofrog unit.


What AF unit?

Will this be as good or better than a DDRC-88A?


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## Bnlcmbcar (Aug 23, 2016)

Audiofrog is working on a unit that can upmix stereo to surround sound including proper Center channel by signal steering via Penteo. No details on whether it will have fir filter capabilities. 



GotFrogs said:


> There are a bunch of differences. First, in the matrix upmixers, the center is a summed L and R and the steering angle computer turns the levels of the channels up and down according to the calculated vector to maintain stage width. Second, the rears are a L-R signal.
> 
> In our Upmixer, the sound field in the stereo recording is separated into mono (same in L and R), Intermediate information (similar but not exactly the same in left and right--the sounds between the center and the left or the right) and differential information (only right and only left). The mono info is sent to the center. The intermediate and differential info is spread over the left and right, the sides and the rears--kind of like a horseshoe if you were to turn all of the dials to 11.
> 
> This provides great stage width, a stable center and no artifacts in the sides and rears.


Only Andy has the knowledge of when it will be out for retail. So until then I guess let’s not stray too far since this thread is for RePhase and Dirac Live.

As far as the DDRC-88A, it does not have an up mixer to my knowledge. It can perform Dirac Live calibration to 8 seperate channels. I’m curious to know how many taps per channel? Same as DDRC22 divided amongst 8 channels or is itvmore like 4 DDRC22’s in one?


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## oabeieo (Feb 22, 2015)

Elgrosso is right , Dirac isn’t release anything for car as of yet

As far as a center ch. I would simply buy a logic 7 up mixer and insert that in the chain if you want a true center that has the proper decoding for what you want to do. 

If you have a ddrc 22d before the helix i would run the correction without the center ch turned on (rears ok) so it doesn’t mess with the 1st measurement that sets delays, but also would leave the eq work improper. I would just use rew after your Dirac run and eq the center in the helix to match the response of the left and right *summed * response . Meaning play both left and right together, use PN and do a moving mic average and just manually eq the center to match that response down to 250hz.or so if tiny center at least to 400. 

For your center you will want to insert a toslink splitter and get a signal split before the ddrc so you can add a 2x4HD or DDRC24 to not be a part of your Dirac run , you won’t want to try To upmix a Dirac correction as left and right will have different electrical responses respectively to the original source and up mixing is dependent on phase of left and right so that can’t be altered to get a proper upmix. 

With that dsp you could get the center to shape the way you want, obviously your upmixer would have to go between the source and any Dirac 


The ddrc88A or BM is ideal for a updstream upmixer as it’s seperate corrections tied to a single target. That how I would do it 

Ditch the helix , use a ddrc88xx and a logic7 upmixer before it. Pretty easy and would be one hell of a sweet setup, and the BM add on you can get Dirac to use all the channels to make a dedicated sub output and also mix portions of each correction to any channel, 
You could mix 25%of left correction and 25% of right correction on top of the center correction if there’s massive differences in the drivers FR pre correction and mix parts of each correction and send it to other drivers to blend the sound if the locations are radically different as far as sound characteristics go.

The same exact acoustic response can sound completely different if one loudspeaker is playing let’s say into glass and another is playing into carpet or a soft padded seat.....
So the 8&BM allows you to mix each Dirac correction onto other drivers to balance that in 1% increments. Pretty sweet. The combo of mixed corrections and seperate Dirac targets on each ch but tied to a single target would give you the ability to make it do whatever you want as far as how it sounds. 

But again as far as upmixing , again just needs be done upstream than run corrections on each respective channel.


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## Bnlcmbcar (Aug 23, 2016)

Thanks!! 

I couldn’t discern what effect would occur when performing the Dirac Live calibration while a mono sum center speaker was turned on.

It’s interesting you say to ditch the Helix. I’ve actually been toying with the idea of fitting a surround upmixer unit in my glove compartment. I snagged an Outlaw Audio 975 from eBay that I was thinking of powering off a mini pure sine wave inverter upstream of the DDRC-88A with the BM plugin.

If you can fit and power the Outlaw unit it opens the door to:

-Analog and digital inputs including Optical,coax, and hdmi (perfect for iPad or tablet source).

-192 kHz 24-bit DAC's for each of its 8 output channels that can be processed to a surround sound matrix galore such as:

Dolby TrueHD, Dolby Digital Plus, and Dolby Digital decoding

DTS-HD Master Audio, DTS-HD High-Resolution Audio and DTS decoding

Dolby Pro Logic IIz, Dolby Pro Logic IIx, Dolby Pro Logic II, and DTS NEO:6 processing/upmixing from Stereo in either Movie or Music modes

It can even set crossovers for each channel. Very tempting. As is your ms8 Logic 7 route. I’ve only heard BMW’s modified implementation of logic 7 for their vehicles. Not the true Lexicon or Harmon Logic 7 upmixing that’s often touted as the best.

Do you know how many taps are available to each channel on the DDRC-88A?

In regards to the DDRC-22D + Helix Route, could I perform the calibration while Helix sends R-L and L-R signals to the rears? Or is it best to perform the calibration with a stereo rear and then modify the matrix post calibration?


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## captainobvious (Mar 11, 2006)

Good stuff, thanks for posting. I'll have to download RePhase and play with it.


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## SSSnake (Mar 8, 2007)

Good read. I am working a CarPC based FIR setup currently. I am sure this will help a ton. Thanks!


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## oabeieo (Feb 22, 2015)

So I’m back (after a year of hiding )  

So rephase is going good a new release is out 1.4.1 
It has some new features worth checking out. 


However on the topic of phase correction and phase manipulation I would like to chat about all pass filters 

Many of your dsps have all pass filters and maybe I can shed some light on there operations 


What is an all pass? It’s a filter. There’s a few different types , I’ll talk about 1st and 2dn order all pass (for audio) 

So let’s dig in. 

First we need to discuss a few things. First the complex plane. The complex plane is an imaginary line that is where each pole on the filter is placed. Poles are exactly what they say. Poles like positive and negative right....not exactly but close. If a input and output signal have a shift in phase to or more than 360° another pole could be said to be added , but in coefficients it’s dosent quite work like that , it’s the calculation that can add poles 
For instance a filter with 90deg would have one pole at -1/rc and a zero at rc. Which makes them an inverse of each other. So the delay at the frequency would add the pole. 

So in a 2nd order filter for instance the shift is frequency dependent. And the shift center frequency is at 1/4th the wavelength, which is called quadrature. So on your all pass control which is most likely a 2nd order all pass, the frequency you pick is when the input and output are 1/4th wavelength apart or 90° respectively . The Q of the all pass says how fast the input and output rise and fall from 0° to 360° . 

So when you add a all pass filter , the signal goes in , and when it reaches its center it is like a mirror effect putting the output to 360° , so if it goes to 360 at lets use a center of 400hz , that means the input and output have a one cycle delay of the wavelength of 400hz right? Almost, it has a delay, but not quite a full cycle, it’s a quarter cycle delay between poles. So what’s the wavelength of 400hz 33.9 inches or 2.51ms a quarter of that is .62ms. So that added group delay can be a added or subtracted or by 1/2 from the input or output to “alignment “ with another crossover or speaker, but in a car , that’s not what ere listening for. 

You could certainly measure your car, window the response and do all that playing around with measurements and apply the all pass and it could definitely make things worse sounding. 

I would suggest using an all pass filter as something you can listen to real time. 

So where do you place these all pass filters ? 

You have to get to know how your car behaves. It’s not something rew can listen for it don’t know if your stage is high or the vocal is high and center. Is there that one annoying frequency that pulls your phantom center to the left or right, or is there a severe combfiltering notch that just won’t quit eating up all the ambiance and sucking the impact out of your system. An all pass might be a great help to solve some of these issues for one seat. 

Knowing the all pass goes “in phase” at 360 degrees which is essentially zero degrees with added group delay, you can move only some frequencies phase in a line, how steep that line is is the Q. A Q of 50 would move the phase from zero to 360 inmaybe a span of just a few frequencies where a Q of .5 will slowly shift the phase over the span of maybe entire octaves depending on how high or low the center is placed so if you have a deep combfiltering notch at 350hz or 400hz or 500hz (wherever that critical midrange and midbass notch appears) you can add an all pass to one of the two speakers and play with the centering and Q and it may or may not cooperate with what your trying to do. There’s a lot of different things happening) if it does help great! You should now have either reinforcement at that frequency or you have shifted the aural cues on that frequency only to help support imagining and centering. Another trick is to use the all pass in the peaks to drive them out of phase , which will subsequently help fix the dips which are out of phase (that’s why there dips) so if everything is out of phase, it’s actually back in phase, sometimes a simple invert polarity switch along with an all pass is just what is needed. 

I know the Macaroni .....I mean masconi and the JLvxi have all pass so does minidsp and a few others. Use the all pass , do it real time and listen.


Another cool thing with all pass, you can apply it at the knee of a crossover, as the crossover rolls off the phase moves , if you add a all pass at a frequency that is let’s say 15° from zero (could be 1db down or so on let’s say a high pass) then it would shift the entire pass Band by 15° +360 . Doing that and a little trickery between inverting the polarity you could shift the phase of the entire left or right channel , by small increments, that along with signal delay can really make them dash pods come together just great....


Or cascade several all passes with delay between them for reverb for that rear fill you always wanted to add space to ..... possibles are endless.....


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## oabeieo (Feb 22, 2015)

IF You apply an allpass and it dosent do what you want. 
Add or subtract a tiny delay to that speaker, it could just be the group delay that’s messing with what you think is going to happen.


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## oabeieo (Feb 22, 2015)

Here is a link to Swiss bears from DIy complete write up to do your own room correction using rephase . 

This process works great, probably would want a opendrc or a few of them to get it done in a car. But this is the manual way to do a room correction 


http://forums.melaudia.net/attachment.php?aid=22240


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## Elgrosso (Jun 15, 2013)

Cool a new forum, that I can read 
Is it from POS?


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## oabeieo (Feb 22, 2015)

Elgrosso said:


> Cool a new forum, that I can read
> Is it from POS?


Yeah, that guys is complete brainiac and super kind. He helps everyone, and has plenty of patience for noob questions. Super awesome dood. 

But yeah I just got caught up on the 115 pages of thread reading I missed


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## oabeieo (Feb 22, 2015)

one Thing rephase is very useful in a car is getting any unwanted ringing to go away. 


If I turn off all eq and processing and listen to a speaker play in the environment it’s in 
Via a door or pillar or a box or kick or anything, I’ve noticed any colorations caused by the speakers mounting almost always has phase issues that a simple adjustment might make that location better or enclosure better or pod or door etc 


You can definitely do near field measurements the see exactly however 
If you have a ringing from a location you can move the phase back a little and heat it go away. 


It’s a good idea to unmount your speakers off there mounting and measure the speaker in free air and try to replicate that , it’s pretty easy to hear frequencies when there delayed or advanced after a few go’s at it and after the correction when you fix it and it sounds and measures better, 

I would suggest most of the eq done in the 120hz to 400hz range isn’t necessary if you have a flat playing driver and a decent baffle and when the phase is fixed the peaks go down and the speaker plays like it should...


Even the best of installs ever would still need a little rephase in a car to just fix that last little bit. 

It works amazingly well. 

I’ll try and post some measurements


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## oabeieo (Feb 22, 2015)

Okay gang here it is 

This is the article that explains meaningful 
Phase and transfer function and magnitude, 

The stuff I’ve been saying for about a year but falls on def ears 
It took me a long time of figuring **** out and this kinda explains it all in one read


Pay the fu&$ attention to it when your reading tho it’s easy to let your mind wander. If you listen to what this man is explaining it all makes sense and than you guys might realize that rephase actually is awesome companion with REW and you don’t need no silly time domain software if you have a clue on how it all works together. REW is very good software by itself 

This is why transfer function analysis is more important and must be considered when giving advice to someone about there system. The time domain can not be ignored especially in a car. Once you realize what it all means than statements like 
“Purposely moving time delay +/- .5ms on a speaker away from tape measure distance can be a good thing” because you don’t know cyclically where the speaker your “lining it up to” is at unless you have a understanding of how to read a transfer function to begin with. 

This is the kind of stuff that makes us better 


https://www.prosoundtraining.com/2016/11/14/meaningful-loudspeaker-phase-response/


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## JVD240 (Sep 7, 2009)

Thanks for sharing that. I've never read it despite being a member. Lol.


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## oabeieo (Feb 22, 2015)

JVD240 said:


> Thanks for sharing that. I've never read it despite being a member. Lol.


Definitely! 

And thanks for all the thanks everyone. It took some diggin to find that one.



I remember I posted a pair of transfer functions about a year ago (I think it was a Dirac live correction for left and right taken in REW) it had a three wraps in the phase but was all beautiful downhill. And left and right looked very similar. Anyway , I remember someone said something like ~”what exactly am I looking at” and “I thought the phase was supposed to be flat” 

Getting flat phase on a graph isn’t exactly what we’re trying to achieve exactly, 
It’s more about getting left and right to have the same phase and magnitude shape from the listener spot. The transfer function. Which placement and aiming of the speaker strongly makes it achievable. Sometimes even with massive amounts of dsp some parts just can’t be fixed and still either 1. Sound good or 2. Get very loud. 

A lot has been written that a speaker radiates below beaming at all directions and it’s dosent matter how it’s aimed. Well that kind of logic is very easy to prove true but unfortunately isn’t how things pan out. The aiming of a speaker absolutely contributes to its transfer function. Getting symmetrical Impulses before any dsp put a car leagues ahead of the game


The electromechanical nature of a transducer and the amount of air it needs to displace per given input signal and the motion of the speaker is sorta in my mind how the time domain starts at the acoustical level. 

If a speaker is moving 1mm at 1k (depending on a whole lot of electromechanical variables) to get to zero degrees at 1k (based on the input signal at that specific magnitude) if more power is given to that speaker at 1k it moves positive in phase (because it’s getting louder and going uphill) when you look at it from right to its peak , however when you look at it from its peak down the phase becomes negative. That is the angular part of the phase that gives a minimum phase peak have those properties in its phase response. So if the amplitude is raised the speaker has to move faster to stay vibrating at 1000hz because it has to move farther from peak to peak because it has to move the suspension farther to make the amplitude greater. Longer travel makes the walls (or axis) of each angle longer as well. Imagine a triangular shape that gets bigger and bigger the louder you make a frequency compared to the other frequencies around it. So even though the speaker is moving 1000x a sec it’s amount of travel is increased to make up for the higher amplitude. We all know a speaker is not a perfect piston at all amplitudes and frequencies however usually does a very good job in its operating range. It’s the non linearities in the speaker and it’s suspension that affects the time domain and it suggests that those differences is where a loudspeaker gets parts of its timbre or sound and subsequent phase response when measured closely other than its own natural reverbance from its build materials. So arbitrarily saying a speaker radiates the same as long as it’s not beaming is false. The cone itself size dictates beaming and the cone itself is one of the axises on the angles of which it’s displacement can produce either faithfully or not.


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## tonynca (Dec 4, 2009)

This is the best thread I've read in the past few months.


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## mfenske (Feb 7, 2006)

Apologies if this has been answered but can I use this with the C DSP 6to8? Would it need some additional hardware?


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## oabeieo (Feb 22, 2015)

mfenske said:


> Apologies if this has been answered but can I use this with the C DSP 6to8? Would it need some additional hardware?


No unfortunately not


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## oabeieo (Feb 22, 2015)

I am officially off Dirac live 
And have migrated fully to opendrc and ran my 1st full tune via rephase room correction. So far , better. I can officially beat Dirac live now. (Finally) 

And it takes hundreds of measurements and time and patients 
But godam. It’s very nice to choose which path to take to my liking. 

I think this thread will become very active in future days. 

My friend on DIY rephase thread gave me the missing link to extreme sq on my own tunes

Cascaded firs. Everything has the exact same polars as if nothing had a crossover turned on at all. 


I will go into depth soon. I need to finalize a few things first



Out with Dirac sharc hello opendrc @12,288 taps 





upload pics


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## Bnlcmbcar (Aug 23, 2016)

oabeieo said:


> ...it takes hundreds of measurements and time and patience..
> 
> My friend on DIY rephase thread gave me the missing link to extreme sq on my own tunes
> 
> ...


Sounds interesting! Looking forward to this.
:snacks:

But wanted to check... 
Does the process require hundreds of measurements? 
or 
Have you mastered the process after hundreds of measurements? :scholar:


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## oabeieo (Feb 22, 2015)

Bnlcmbcar said:


> Sounds interesting! Looking forward to this.
> :snacks:
> 
> But wanted to check...
> ...



Thanks , 

So the cascade wasn’t working as well as I thought it would, it works in therory perfect from his pics of xsim it’s awesome. But it seems my HDs lack the power to pull it off and the way the cascade is blocked out I don’t have anywhere near enough fir. I would need a opendrc on each pair. So 5 opendrc boxes to pull it off 
In my 10ch system. 

So I’m working on doing the room correction now. As of now the mids and highs sound exactly like Dirac. It’s very very close. 

The sub integration is a little tricky I’m getting help on it tho from some smart fellas. 


I dare you to say smart fella as fast as you can. Say it faster !


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## oabeieo (Feb 22, 2015)

Got the sub dialed in better than Dirac ......only tiny bit better but still better for me 


Here’s what’s up.

I have to thank BTRTT and FLUID on the DIY side of home audio for the help super great guys!


I basically asked a few questions and pretended to be more noob than i am to try to get them to talk to me in a scaled down way, as these guys are way smarter than I am and say some things that are still a bit Chinese to my book of terms.


So after a few friendly posts I learned how to get my sub completely nailed down to blend with the midbass in a way that is just unparalleled.


Here’s how 

1. The room gain adds a tremendous amount of boost in the subsonic range all the way down to DC. In fact there’s a chance that your “room mode” at 50-120hx (wherever it is) isn’t a room mode at all, it very well could just be you have so much room gain below that spot. For example a 70hz dip could be side to side cancellations from doors and not necessarily the room. Not always. There still is room modes that are related to the size of the car , however the big one sometimes may not be as intimidating as it looks once the proper sub eq is performed. 

I had to use a 6db BW @ 20hz to straighten the responce to DC 
This subsonic range being eq really makes the bass get in time with the music 
It has something to do with a transfer. So this subsonic must be gloabal and also all sub eq work must be done gloabally. 

2. The phase 

You basically take a single measurement at listen position 
Remove the time of flight. REW impulse centering works great. 
Generate minimum phase 
Checkbox off the minimum phase version so only excess phase and measurement is on the screen 
Than export the measurement as txt
Than import into rephase 
Make corrections 
And move excess phase to match the minimum phase plot.

Generate the fir and load into DSP. 

You can see in this measurement the phase at 80hz 
That is a reflection or boundary. (The car shape and size and it’s loading characteristics) below that the phase moves up due to the subsonic rolloff. 
The natural rolloff should be left. There is no reason to linearize natural roll off.
The excess phase can be manipulated to better correspond with the minim phase and the response


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## subterFUSE (Sep 21, 2009)

Sub'd


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## dgage (Oct 1, 2013)

You say better than Dirac, which version? 

Looks like you need to make a video! 

Thanks for sharing your learnings. You are persistent and we’re very fortunate you are.


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## oabeieo (Feb 22, 2015)

dgage said:


> You say better than Dirac, which version?
> 
> Looks like you need to make a video!
> 
> Thanks for sharing your learnings. You are persistent and we’re very fortunate you are.



Both. 

So I need to go slow on this because we all know I get ahead of myself. 

So I’m starting with the sub and working my way up and we’ll all see how the transfer function will sum beautifully on all combined in a 4 way. 


So better, yes. And this is why. Dirac is excellent. It does a downright amazing job for an auto tune. It’s so very very good at what it can do. It’s only shortcoming is it has rules that are fixed. 

We all know in tuning there’s more than one way to achieve the same measured outcome. One way may have better sonic attributes and are more musical than technical. Like weather to boost or cut, which route to take to achieve a given result. Weather to treat a reflection with phase or not, weather to leave the high Q dips alone or not, or even work on them a little. The software works within its framework of rules period. Although laid out extremely well, arguably the best room correction software and easiest to use for a complete auto tune experience that gets you tuned extremely well in a tiny amount of time. 

I’m okay with taking the time to take all the measurements required and I like that I have the say in what is corrected or not. 


I’ve spent 4hrs on the mids and highs and I haven’t even scratched the surface and honestly it sounds the same as Dirac, but it sounds the same as my very best Dirac tune I’ve ever had. And that’s without really spending as much time as I know I could have. 

So I stopped there and started with the sub, that was the hardest to get right and now I got it. That subsonic really massively made a big difference.


I’ll try and quote his post from another forum so we can understand what he was saying about it. Very interesting and I can’t wait till I can tune as good as this guy. Frikin brilliant guys on this forum.


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## oabeieo (Feb 22, 2015)

Here’s the important parts of that conversation....super awesome stuff 







Oabeieo said:


> so here is some measurements of just the sub at listening position
> 
> 
> 
> ...






Oabeieo said:


> so I did this in rephrase






BYRTT said:


> First will say remember room is global so don't count its corrections dedicated for the lowest pass bands but as global for the summed system passband, and yes room gain is of minimum phase except the reflections that arrive late at listening position, one can repair late reflections but think forget about it because correction only works at that particular spot in space where microphone was positioned. Reflection free is no room but we can get superior sound with a perfect dialed in speaker system corrected for room gain and a house curve that is tweaked to suit the speaker/room profile.
> 
> 
> Below is sim model your system HP stopband with the red curve, you can see the numbers in first row, next i set the program WinSpeakerz model for auto cabin response with the numbers in second row. Now look the blue curve it is hot and worthless in you get a flat boost all the way down to DC, this will sound bloated with DC thumbs and mud sound signature way up into higher frq.
> ...






BYRTT said:


> No box linearization for the Q0,83 @45Hz stopband only linerize subs LP filter : )
> 
> Here is a block diagram example of 4-way system where i used numbers for your sub because you told those numbers, the rest is for example. As we were around pages back to hinder destructive ripple calculate final target slope per transducer to be a cascade that includes the side band transducers slopes plus stopbands exactly as in block diagram. As you see stopbands are of minimum phase and not linearized so we end up a true system sum minimum phase curve as was it a single point wide band transducer 45Hz(Q0,83)-20kHz(Q0,707).
> 
> ...






BYRTT said:


> Hi Oabeieo,
> 
> Know it can be hard to grasp therefor suggest you take some virtual study to educate your self, it is not that hard copy my example pass bands in Rephase export them as frd files import to REW where you can sum them on "ALL SPL" tab using the wonder math it can offer in "Controls" dialog. You will then see example sum perfect as was it one pointsource transducer, and when you do the same exercise but omit side and stop bands you get varius errors in amplitude and phase domains no matter you use FIR or IIR XO slopes.
> 
> ...





fluid said:


> That was pretty much the discussion that was had above. If you linearize the sealed box you will change the phase away from minimum and end up with a phase that does not follow the amplitude response.
> 
> Measure the result you have and use REW with a FDW to clean up the phase, generate a minimum phase version in REW and compare to what you have. If there's excess phase you can export that and use rephase to try and correct for it and see if you like that any better.


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## dgage (Oct 1, 2013)

Can you share the link or links so we can see the source pictures they reference? Thank you!


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## oabeieo (Feb 22, 2015)

dgage said:


> Can you share the link or links so we can see the source pictures they reference? Thank you!


It’s the rephase that’s bread on DIY audio 

It’s like 300pages long. And a killer read 

Very easy to find


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## oabeieo (Feb 22, 2015)

So this is another how awesome rephrase is...

My sub amp has a dial crossover that cant be turned off.
RF bd1000a1 it has a dial from 50-250hz BW24

So I made a 269hz LR4 linearization than in phase eq was able to make phase flat 
than added my LR4 linearization on to it. I sent that to the sub channel FIR and the FIR that does EQ is on the opendrc global EQ. Two separate fir. 

Now the sub is perfect. I turned the knob on amp crossover up all the way to 250 added the linearization I custom made for the BW filter


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## oabeieo (Feb 22, 2015)

Got the sub dialed and styled 

This week I’m doing this again 3rd attempt , this time I think I’m getting the hang of it. 

The only part that’s still sucking for me is left and right coherence and the minimum phase filters on left and right de-correlation


I have questions out now to some experts to get a bit more info 
I was just told today my soundcard I’ve been using for this might be erroneously spinning my phase backwards on some measurements. I’m hoping it was just the 2nd sub crossover causing all the headache, I have to work through Black Friday installs this weekend so won’t get to it till next week. 

I’m bringing home my DMRTA from work this weekend and I’m starting all over again. That might be my only issue this whole time is the creative usb sound card (which is extremely flat response) I think it’s still fine but I’ll bring home the audio control sound card and bring out my big guns. 

If all goes well , *the link below *is the procedure to use that is better than Dirac if you have the time. Even without doing seperate left and right eq as of now and just going off what phase I have been able to measure with the small sound-card it’s still better than Dirac: so we’ll see. I’ll post up every measurement and a entire ride along with the dmrta 

Till then 

https://www.dropbox.com/s/10xdhh83jokzbxv/REW_rePhase_tuto.pdf?dl=0


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## oabeieo (Feb 22, 2015)

cant wait till I can try this. Especially with the amazing help I’ve been getting on the DIY side from fluid. 

I have a brand new set of beyma 8G40-4s arriving tomorrow. The 2118h is coming out and I’m also going to build up my kicks to be a lot thicker fiberglass and a bigger vent to the outside. I’m also going to eliminate my vent holes in the kicks and try to get the response more uniform and get rid of some of the back-wave interference

So I have a busy Black Friday weekend at work, next week the beymas go in and I start making my own midrange corrections. 


I have an argument out now on comb filters and time delay. We’ll see what comes back and see what the rephase experts say about unequal path lengths. 
I’m hoping for a easy to adapt procedure to the one I posted above. 

We’ll see


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## dcfis (Sep 9, 2016)

What didn't the legendary 2118 not do for you?


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## oabeieo (Feb 22, 2015)

dcfis said:


> What didn't the legendary 2118 not do for you?


Short answer , 2118h plays to 70hz in .5 but don’t even think about anything more than 40w 



dcfis said:


> What didn't the legendary 2118 not do for you?



The 2118h has been a fabulous midrange, in fact I can already say with confidence the 2118 will outperform the G40 above 800hz and have better midrange. I can’t think of an 8” that is as good as the jbl in the midrange 

It’s been proven that the 2118h “breaks” at 300hz . Meaning it starts to drop off because it can’t act pistonic below 300 *while keeping the same linearity as it’s upper band.*

It makes no sense to eq down that magical midrange down to the midbass output when what little midbass can be made from the 2118h is riddled with distortion. 

JBL states that it “shall” play down to 70hz. Which it can in a .7cuft bass reflex tuned around 45. the natural resonance gets lowered a tiny bit in that type of enclosure which also the port will start to control cone not allowing it to move as much as it would in IB. So that’s where they get it’s good down to 70. 

The 2118 has a extremely thin diaphragm and a stiff compliance. It’s not mean to to move that much. The distortion in IB goes way way up under 300hz IB.


So I’m trading exceptional midrange for a little more bottom end performance. I have the 2118h in a a-periodic kickpanel pod that I have stuffed with fiberglass AP membrane that gets the jbl down to 200 using LR8 and that’s the bleeding edge for my kicks in AP. The kicks have a nasty ringing at 111hz very high q that has been difficult to tame. 

I’m changing the kicks to IB and venting to the outside and want them to play down to 150. (80 would be great but it’s not possible while maintaining high enough efficiency for my goals) so the 8G40 is good down to 125hz and it has superb midrange under 1khz. The 8G40 has the best lower end vocals I’ve ever heard out of a speaker. I kicked myself several times for selling it. Now I have another brand new set from the 2014 2015 run of them. (The ones that has 6mm xmax vs the new “improved” that has 4.5mm). 

Ive been looking for awhile now for a new old pair. 

The 8G40 is also in 4ohms and has more motor than the jbl. The G40 is a amazing sounding loudspeaker. The 10G40 and the 6G40 don’t quite capture the same type of “big vocals” as the 8”. It’s quite a good speaker for what I am going to be changing into. 

If I can get down to 150 and still be able to use a 4th order filter I’ll be estatic but I doubt I’ll be so lucky. I think it’s great potential but I don’t think I’ll get much below 150, maybe 125hz with 8th order. 

Only one way to find out


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## oabeieo (Feb 22, 2015)

So when dealing with fir filters you have a finite amount of time windowing to address phase delay and group delay. 

With rephase you can invert the time and invert the polarity separately 
This can be useful especially with unequal path lengths.

So I would like to talk about something Dirac does quite well and will be key into following along with what’s what we’re going to be doing in coming days. 

So Dirac can take a 2ch measurement and each of the channels could have substantial phase delay or group delay in the measurement and it can still correct for it very nice. You can have speakers with incorrect delay settings. Example the left tweeter could be .5ms out of arrival from the mid and the midbass could be 2ms out of time with the mid. That would show up as a bunch of wraps and. Much more than 180deg of shift. 

The ddrc22d is limited to 23ms of output delay where it’s counterpart the opendrc 2x2 has 64ms of output delay at 6144t per channel. So the ddrc22d must be using iir type custom order allpass filters to deal with the phase delay. 

The group delay and phase delay could be considered the same thing, phase delay I am speaking of is any delay that is in the measurement except time of flight delay. 

I would like to share two videos on this subject so we all know how to read delay on a logarithmic time to frequency chart. This gives highlights on how to add or subtract time out of a small part of a measurement and repair it within rephase 

Rephase allows you to export iir biquads to a minidsp file and use the rest of the correction in fir. It can split up the coefficients to iir and fir outputs on a single correction. Weather you use a mixed phase approach or seperate fir and iir biquads to do the time correction and get each channel to be doing the same thing. 

Rephase has 16banks of eq and 7 or 8 banks (IIRC) of iir curves for all pass types. Pay attention to these and how to do the calculations within it. The calculations can greatly help figure out weather it’s better to keep a few wraps in the measurement as it might simply be group delay 


Part 1 


https://youtu.be/_KZ1_Je9fA8


Part 2

https://youtu.be/IcYBP51eCMk


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## Holmz (Jul 12, 2017)

oabeieo said:


> ..
> I have questions out now to some experts to get a bit more info
> I was just told today my soundcard I’ve been using for this might be erroneously spinning my phase backwards on some measurements...
> ...


Phase and freq and related to each other by on integral and a derivative...

So play a tone, like 440 Hz... a "phase spinning backwards", will make it 439 etc.

Basically... trust your gauges, as the sound card is probably right!


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## oabeieo (Feb 22, 2015)

Holmz said:


> Phase and freq and related to each other by on integral and a derivative...
> 
> So play a tone, like 440 Hz... a "phase spinning backwards", will make it 439 etc.
> 
> Basically... trust your gauges, as the sound card is probably right!


Lol it is right . 

It was my amp whole time. So I’m looking for new bass amp ....probably going with matching alpine type X the XA90M 

Yeah very strange , even after the crossover linearization it still had super goofy polars


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## zacjones99 (May 11, 2009)

Oabeieo you're a wealth of information and I really appreciate your sharing your experiences with us. I just stumbled on this thread right after installing a CDSP 8x12 DL. I've been following the CDSP 8x12DL threads, and in the last couple days I've been trying to figure out a workaround for the phase shift on the non-defeatable 10hz SSF and variable LPF on my sub amp. Funny to see you're kind of dealing with that right now too. It sounds like the best solution really might be to use a different amp with no x-overs. 

The Dirac Live version still has appeal to me as although I'm pretty familiar with the basics of x-overs, slopes and time alignment, I start to lose it at phase and transfer function. But I have to wonder if it would make any sense to try and add one or two of these OpenDRC-DI's to my setup. It seems like you could do a bunch of tuning on the sub in an OpenDRC-DI, then (because you're supposed to apply all filters and eq applied to the sub to the entire front stage as well) forward that tuned signal via SPDIF to the CDSP inputs and allow it to run DL on the midbass, midrange and tweeter, or maybe even the sub again as well. 

This may seem a silly question, and please pardon my ignorance, but does forwarding the tuned signal to the CDSP as its main input signal solve the need to apply all filters and EQ from the sub to the mains (as far as phase etc is concerned)? Or do you actually need to send the original signal again to the mains, and actively apply all EQ and filters that were applied to the sub to the each individual channel in the mains?

And one step further, could you work on both the sub and the midbass in a single OpenDRC if it was just part of the chain before a CDSP? Or would you need an OpenDRC for the sub and another OpenDRC for the midbass?


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## oabeieo (Feb 22, 2015)

zacjones99 said:


> Oabeieo you're a wealth of information and I really appreciate your sharing your experiences with us. I just stumbled on this thread right after installing a CDSP 8x12 DL. I've been following the CDSP 8x12DL threads, and in the last couple days I've been trying to figure out a workaround for the phase shift on the non-defeatable 10hz SSF and variable LPF on my sub amp. Funny to see you're kind of dealing with that right now too. It sounds like the best solution really might be to use a different amp with no x-overs.
> 
> The Dirac Live version still has appeal to me as although I'm pretty familiar with the basics of x-overs, slopes and time alignment, I start to lose it at phase and transfer function. But I have to wonder if it would make any sense to try and add one or two of these OpenDRC-DI's to my setup. It seems like you could do a bunch of tuning on the sub in an OpenDRC-DI, then (because you're supposed to apply all filters and eq applied to the sub to the entire front stage as well) forward that tuned signal via SPDIF to the CDSP inputs and allow it to run DL on the midbass, midrange and tweeter, or maybe even the sub again as well.
> 
> ...



Thanks zacjones for the kind words, excellent questions also.



So an opendrc would definitely be able to untwist that phase up to 64ms with the 48k plugin. 10hz is 100ms so you would be able to do most of it. Most of it is inaudible however the air between 10hz and 20hz really helps the entire spectrum get in tune. It’s crazy how much the sub keeps the system together. It’s easily the easiest driver to get right and the most important speaker in the system (if I had to pick one and label it as such) I would go for a different amp if possible and use the dsp on other things.



If you can add the same exact subsonic filter to all other channels without having to buy anything else and without massive dsp losses that would solve the issue as well.



A pair of opendrcs would be quite the system. So far my favorite configuration has been my opendrc upstream of 3 other dsps and one of the 3 being a ddrc24 Dirac that is only for midrange. It is excellent sounding.



To be completely honest after studying what Dirac is doing, it’s very hard to beat the algorithm. A computer in charge of the biquads with a missive amount of programming and fine tuning to the algorithm is hard to top when staring at the controls of rephase and some of the answers you need just can’t be thought up as fast. If you like tuning and figuring out how’s and why’s rephase is excellent and does produce a better sounding tune.



So loudspeaker crossover linearization is by far the most beneficial thing a system can have and the removal of any group delay. That’s half of the room correction right there.

A pair of opendrc might be two much dsp for 4 potential channels. But if you used one as a sub dsp and the other as a highs dsp that would work excellent as a pair of global dsps that work on everything. So a few ways to do it. I would like to make a suggestion though that maybe will help.



Focus on speaker linearization first, than do the room correction. Getting the crossovers to work together perfectly is by far the biggest improvement I’ve had above all else as far as time domain goes. So much so when I was on Dirac I opted to use 3 2x4hds with a ddrc22d upstream because of linearized crossovers. Instead of using the all in one 8x12DL that does so much except driver linearization.

I haven’t had a need to cross below 80hz and you simply don’t need to. The 2048taps at 96k works just fine for crossovers and you also have 4ch of fir power per unit. Minidsp makes a 8ch version that has even less taps so a pair of 2x4hds is plenty, or a 3rd for the sub if you want to take it that much farther but it’s really not at all needed. I only have a third because of my rears.



So having fir channels at every output is the very first thing to acquire. You really can’t do a linearization on a gloabal dsp. It needs to be done at the driver level. And here’s the thing.



Being that hear the sum of all reflections and all the speakers playing there’s a lot going on and it needs to be broken down and sorted out to get a proper linearization. With Dirac and the room correction side of a correction your correcting for the sum of all frequency and time related problems. So in that instance you can linearize the sum of all speaker and reflections (room correction). At the driver level you linearize each individual sound source before they sum. That’s where things make the biggest difference and makes dealing with the room a lot easier.



When we’re talking about the sum of multiple drivers the crossover time distortions would be viewed as a group delay and not a linearization. So adjustments to the time domain in a global output would affect the sum of everything and should be measured as such.



When you correct the group delay in the time domain gloabally without doing any loudspeaker linearizations first you can’t truly linearize the crossovers only the sum. The sound quality is still massively improved vs doing nothing at all. In a global correction if you for example fix the group delay of a high pass you making the low pass time smear worse as a low pass and high pass go opposite direction (except a few even order filters in that case the polarity is reversed on the low pass). If the crossovers sum perfect and let’s say your car breaks all the problems of a car and the Linkwitz-Riley alignment actually works the way it should in a home setup it than it would be okay. The problem is in a car, the alignment topologies don’t quite operate to the textbook they were as much written. They still perform mostly as textbook however the farther the drivers are mounted away from each other and the more path length issues are presented the alignments perform worse and worse, or shall I say different and unpredictable, because it may work out based on luck. So while you can still get superior tonality and sound with a single global correction , driver linearization is where the cake is baked and the global correction after is the icing.



Doing separate linearizations you can get them to sum perfectly no matter how you have them mounted or what axis there on or the path length differences. All you do is make the driver flat for an octave past crossover at listening position and turn on your textbook linear phase crossovers in a nutshell. There’s more you can do of course but doing just that is 75% of the time domain distortion.



Once the driver linearization is done you go to do your global correction. You can count of the the speakers summing right so no worries about crossover cancellations as a result of any global correction. Than you measure each side and sum the measurements. Do minimum phase eq until the sum of L and R are flat. Than do separate left and right measurements and look at excess group delay and make it minimum phase. No de-correlation.



Using separate eq settings on left and right is the worst thing one can do because of unequal path length until I prove otherwise or someone can prove otherwise. I’m still working on that part of it and have been using linear phase eq after doing summed L and R eq and have been having some good luck in the higher frequencies. Pre-ringing is audible in the lower frequencies with linear phase eq. 
It works excellent in stop bands or at the filter knee. I can’t hear pre-ring at high frequencies. Linear phase eq also helps shape the DI or power response. More on that later.

Hope that helps


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## zacjones99 (May 11, 2009)

oabeieo said:


> Hope that helps


It absolutely helps. Thanks for the detailed explanation. Well for starters I guess I will go ahead and ditch the Zapco Z-3KD with non-defeatable variable SSF and LPF. Why they made a top tier amp without a defeatable filter switch I cannot imagine.

I will likely run the C-DSP 8x12 DL for a while to get a feel for it and see if I feel the need to make improvements. The concept of linearizing each driver individually at the listening position sounds pretty amazing. It sounds like this could be most easily accomplished by simply adding a couple 2x4HD's into the end of the mix. The only downsides are the additional A/D and D/A conversions required, and with the C-DSP we're still stuck on Dirac 1.7. 

Will a single 2x4HD have enough taps to process a sub and a pair of midbass drivers? 

Maybe a great place to start would be with adding a single 2x4HD for the sub and midbass drivers. That way I'm not overly concerned with the additional A/D and D/A conversions since we're only talking about 20-300hz of audible frequencies. We're still going all digital through the midrange and tweeter, and I can let Dirac handle those for now. If I find I'm able to handle tuning with the single 2x4HD and want to try my hand on the mid/tweet, I can always add another at any time on the mid/tweet, and see if the benefits of driver linearization outweigh the additional A/D and D/A conversions.

Anyway I hope I haven't strayed too far off topic here. When I actually get knee deep into rePhase I'll be back. Thanks again!


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## oabeieo (Feb 22, 2015)

zacjones99 said:


> It absolutely helps. Thanks for the detailed explanation. Well for starters I guess I will go ahead and ditch the Zapco Z-3KD with non-defeatable variable SSF and LPF. Why they made a top tier amp without a defeatable filter switch I cannot imagine.
> 
> I will likely run the C-DSP 8x12 DL for a while to get a feel for it and see if I feel the need to make improvements. The concept of linearizing each driver individually at the listening position sounds pretty amazing. It sounds like this could be most easily accomplished by simply adding a couple 2x4HD's into the end of the mix. The only downsides are the additional A/D and D/A conversions required, and with the C-DSP we're still stuck on Dirac 1.7.
> 
> ...


“Only 1.7” .....oh man any Dirac is downright amazing and the 8x12DL is one powerhouse that is just downright badass. 1.7 is a amazing platform.

An 8x12DL will sound impeccable. Remember the crossover distortion is not very much audible. It’s still audible, but it’s small. And if the sum (what you actually hear) is linear (phase and frequency) than honestly what difference does it make. The differences are tiny. If you like to chase perfection than do 3 2x4hds with a ddrc22d upstream. I’m trying to beat that setup right now manually and being honest I’m not sure if I’ll ever get there it’s that dam good.
And I only say that because I measured a perfect Dirac impulse on both left and right channels. So it’s hard to beat perfect. I think I just want to learn how to get there on my own all the way.

A 2x4hd fir bank is good for a crossover linearization and not really much more than that. Maybe some small eq in the fir but it falls apart fast with much more than a crossover. It’s really obvious they designed it to be a linear crossover. Just not enough taps for anything else. It work great for a crossover though. And it’s a great dsp on top of that. At low frequencies you can center the impulse ahead a few ms and get a low frequency crossover to work with the 1024taps. One you add eq on top of it it just isn’t powerful enough as eq would require different centering in most instances if your trying to eq out of band.

If you have a digital source honestly a ddrc22d and a pair of 2x4hds is downright amazing hands down. Otherwise , the 8x12DL if your only wanting Dirac.

I would avoid the ad/da da/ad nonsense.
The quantization errors is audible as it’s all floating point. If it was fixed point it wouldn’t be such a big issue at 24/32bitAd but with floating point I can hear the difference. It’s grainy and looses depth. At least not over this, you want the signal to be right and stay digital with this much dsp being done.


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## tonynca (Dec 4, 2009)

It's good to see you're still at this Andy. I've been enjoying my system for the last couple months and taking a break. I need you to get your hands on a Bit One HD and give me your opinion. 

I really like the fact that I could choose any crossover slope and retain phase correlation between the active speakers. An example of how I was able to use this to my advantage was that I had an issue with 250-300hz where there was excess group delay due to car interior and speaker mounting position. It caused 250-300hz to steer to the right excessively. I could never get low mids to sit in the center. Even when I tried to EQ it out it just didn't work. The left and right amplitude were match but there was excess group delay on the right speaker which EQ couldn't fix. I used a 48/db slope and crossed mid+midbass over at 300hz and the issue went away completely. Solid center image for male vocals. I basically resorted to having my midbass playing 250-300hz by using a very steep slope while maintaining phase correlation between the active speakers. I totally agree on what you're discovering because I think I noticed that as well when I'm using the Bit One HD with FIR filters.

Your ears are keen to be hearing A/D conversions haha. But that's what makes you good at this. Every time you pump a signal through a device you're degrading or coloring it in some ways. Less is more there. 

Sent from my iPhone using Tapatalk


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## oabeieo (Feb 22, 2015)

Thanks Tony 

yeab I would love to hear your car I bet it’s amazing. My buddy Evan has the n7 3 ways in in Ram with pods a lot like yours and it’s very nice to listen to his car with a helix and brax amps. Very different than horns and 8s but good. Dam good actually. (Just can’t get loud without blowing a 500$ 3” speaker which is what sucks) aside from that man oh man the sq is insane. 

But yeah it’s a lot of fun. 

I’ve finally got a few good base tunes going. 
I finally figured out a MASSIVE problem I have been having and completely unaware 

My new laptop had installed windows drivers and made my soundcard run at 96k and I’m taking 48k measurements. 
Also rew says if using a usb mic a soundcard calibration isn’t necessary. 

Only after digging on the rew forums I’ve found where John has stated it is still a good idea to calibrate it for phase related measurements, so I did and everything is working much much much better. 

It was off at 10k so it moved the phase only a tiny amount, but that tiny amount up high is a substantial amount down low when your alignment of things is sample by sample.

Gang, *make sure your soundcard has a loop back, is calibrated,and is running at the right sample rate in windows and in rew. 

I metered my web configuration so it can’t update anymore and hopefully no more drivers update and change themselves. 

I can’t express how important it is to make sure your rig is calibrated and has the right sampling within entered. Also make sure your RTA is using a rectangular window ONLY if using PN noise and the fft lengths match! So so incredibly important *


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## tonynca (Dec 4, 2009)

Hmm might invest in a USB sound card for the MacBook. I don't think I could get loopback with stock sound card.


Sent from my iPhone using Tapatalk


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## dgage (Oct 1, 2013)

Focusrite 2i2 is a good microphone preamp if one wants to run an external microphone. And it has 2 inputs and 2 outputs so a loopback can be setup.


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## oabeieo (Feb 22, 2015)

dgage said:


> Focusrite 2i2 is a good microphone preamp if one wants to run an external microphone. And it has 2 inputs and 2 outputs so a loopback can be setup.


I was just lookin at that haha

I have a old creative sb! Soundcard that has a nice flat responce and all kinds of in/outs
It was 50$ and has worked great , I have the dmrta that I can’t seem to get asio to work right in rew. So I might get one, but so far the sb! is working again now that I figured out what was wrong.

If I can’t get the dmrta to work with rew I think that or the us-366 tascam will have To be the next thing


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## oabeieo (Feb 22, 2015)

tonynca said:


> Hmm might invest in a USB sound card for the MacBook. I don't think I could get loopback with stock sound card.
> 
> 
> Sent from my iPhone using Tapatalk


Do you have a line in? Like a aux in or mic in ? If so you can loopback , you have to just configure it .....in a Mac......Idk . 

The loop back is the only reliable way I’ve been getting good phase measurements 
And when doing more than one speaker it’s critical. The acoustic timing reference works on tweeters pretty worthless on anything else. 

Using loopback I can take measurements and they actually do what it says. No goofy sounding corrections. 

There’s nothing worse than taking a bunch of measurements just for them to trick you into thinking there accurate and than not accurate at all. The phase is a dead giveaway something is wrong. The FR can completely send you on a wild goose chase. 

Even tho my soundcard plays flat adding the calibration made my phase spot on again. 
It was riddled with problems for like 3 weeks to a month before I caught it. Such a headache going from excellent sounding tunes to what the f*** happened.


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## tonynca (Dec 4, 2009)

oabeieo said:


> Do you have a line in? Like a aux in or mic in ? If so you can loopback , you have to just configure it .....in a Mac......Idk .
> 
> The loop back is the only reliable way I’ve been getting good phase measurements
> And when doing more than one speaker it’s critical. The acoustic timing reference works on tweeters pretty worthless on anything else.
> ...


MacBook only has a headphone output. No line in. 


Sent from my iPhone using Tapatalk


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## oabeieo (Feb 22, 2015)

tonynca said:


> MacBook only has a headphone output. No line in.
> 
> 
> Sent from my iPhone using Tapatalk


Does the headphone output also a mic in 

I’ve seen some have both on the same


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## tonynca (Dec 4, 2009)

I think it may but I would still need two ports to do a loopback. MacBooks only have one port. 


Sent from my iPhone using Tapatalk


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## oabeieo (Feb 22, 2015)

So Tried a new one with the new old re calibrated sound card

The result is not worse. The center is super defined...The center steals the show by far, vocal is super real and placed amazingly good now. The stage depth is good....a little different than dirac... still need to tweak that a little, maybe a few more tries. stage placement is good. Midbass response is better but different....not sure yet what..... Its good....
I need to listen for a few days and pick it apart


overall. Its good. I'm happy. It works!

Heres a pic of the left ch in rew and rephrase and the right ch in rephrase ( didn't take screenshots of all the tabs... you should get the jist of it


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## oabeieo (Feb 22, 2015)

So changed a little bit 

This time I moved the excess phase the same amount on both sides equally under 2.6ms long (the width of car) which is the narrowest dimension of the car 

What I didn’t like is now better and it sounds excellent. 

I also added eq work to the right to better match the left and simply ignore the dips 
The left side has a dip at 1k and the right side has dips at 150,630,1k,.....

It’s very close to a Dirac tune now.....
I love the freedom to do it how I choose 

I can’t get over how awesome the center is.
Vocal sounds astonishing real. The far rear stage on left is still a tiny bit wonky. After driving to work today I think I need to lessen the minimum phase correction on the left or lower the Q on one of the bands. 

I should post up the spectral responce , very nice very even 

I’ll do that in a few after work


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## oabeieo (Feb 22, 2015)

I got it working excellent now 

I tried every combination of phase eq that I could possibly do that makes the excess phase as flat as possible while keeping the same settings on left and right channels 

What was left was a little at 300-630-5000

It’s badass now. Stage is right, sounds great, I ignored the big dips and the dip a 1k on the left. 

The music has more snap and the sharp transients rip oh so nice now. 

I am working on screenshots now


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## oabeieo (Feb 22, 2015)




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## oabeieo (Feb 22, 2015)




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## oabeieo (Feb 22, 2015)

I would say that's a pretty close match....They look very alike. Its sounding excellent!..

As good as dirac now........I need to listen for a few days to really be sure and switch back to my Dirac Tune and examine. Super stoked I'm finally getting it


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## oabeieo (Feb 22, 2015)

the right spectrogram shows a little more meat in the 400-1k range than the left..That was a little more energy as its about 1db louder in that range to get the center pushed center instead of in front of me..... 
It shows I have it "balanced right" about 1 click.... Crazy how much that graph reveals about the system and its linearity


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## oabeieo (Feb 22, 2015)

It’s very good.....but I can’t beat Dirac....

After two days of listening to this attempt. (Number 10) maybe 11 will be the magic one for me ....we’ll see 

As of now it’s not better than Dirac....it takes a long time to get it right but is very fun to try to beat Dirac. The sub bass I like better on my tune. But that’s it , the mids and highs just aren’t better yet.


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## oabeieo (Feb 22, 2015)

Got a. Excellent sounding tune (15th iteration)

So for a car I’m thinking this works good so far , left and right transfer functions match very nicely now and the sound is very similar to Dirac ....except I was able to set the magnitude more to my taste and use live RTA (which in my opinion is the best sounding in a car)

So I used REW and got all speakers time aligned and ignored the fact that the imaging was off and waited.

Than I used live RTA pink PN with 8k bins
And set the amplitude from 20k to 1.3khz on the left channel to flat as possible with only the left channel playing. Than I applied the same settings to the right.

I basically did the left channel and worked that side only and applied everything to the right that I did on the left .

So working the left after using live RTA from 20k to 1.3k I than used sweep measurements. I basically copied the Dirac way of measurements in the box shape except I added 3 measurements at each point and each of the 3 measurements in the box were approximately 2” apart. So it was the normal 9 measurements with an additional 18 measurements so the box shape had 3 spots at each measurement point. That allowed me to get a better average at each spot, in case one of the spots had a funky spot.

That’s why I also used live RTA above 1.3k is because each measurement point would drastically change if I moved the mic even a tiny bit making the average non consistent.

Even though under 1.3k that phenomenon was not a problem I still added the 3 measurements at each point on the box just in case.

I carefully made the amplitude flat under 1.3k ignoring the dips in the average. (Btw rephase eq is badass it works so so good by far the smoothest responce I’ve ever hear in my car) I kept my Qs under 4. That was a big sound quality aspect most of all was keeping all my Qs low. They are mostly 1.7 to 2.3 and a coupe in the 3s and one is 4 using two 16band eq banks. The rephase eq can have up to 7 or 8, 16band banks of eq so the amplitude was made extremely flat.

It was more a process of using lots of wider Q eqs and wiggle the amplitude into being flat instead of trying to focus so much on making an exacting Minimil eq where is limited to let’s say 10bamds or less. Having all those eq banks really made it amazing.

I’ve never tuned like this but gosh dam it sounds so much better having a lot of eq bands all lower proportional Q or constant Q
I like proportional Q better I feel like the interpolation works better!

After that was done I took a new set of measurements this time only 3 at the center moving from back of my head to middle of my head to front of my head in a straight line
And applied a aggressive FDW (4cy) and cleaned up the excess phase on the left channel only. I used one one of the 3 and picked the one that had the flattest phase so I would do the least amount of phase EQ 

Than I turned on the right channel with all the same exact eq as the left. Any peaks above 1.3k were brought down using real time RTA. Below 1.3 the only thing that was different was the modes (80hz was obviously louder than the left so was 150 and 300 and 630) I left those alone! I let them be louder. Those peaks need to be left. 

I noticed leaving those louder not only keeps the center where it should be , but on the other side of car the left is doing the same thing. So the energy is there!

I took the 3 measurements on the right and applied the same FDW and cleaned up the excess phase.

Measured left and right and the IRs are very very close to the same. The two wraps are in the same spots and have the same general shape. The coherence is excellent. And the tonality is to die for.

No more poping sound from left midbass can turn it up nice n loud and the 80hz response sounds just fine. Not trying to adress the left door and just ignored it completely and worked around it works to me the best. I lost a bit of gain from working around that dip but the sum of left and right show a nice smooth response.

So when I measured left and right together the dips are still there but the right channel fills in a lot of those dips. Where the left is 10db deep the right is flat the sum is only a coupe dB of dip. Working then eq from the worst side (responce wise) made the easy side fill in and those room dominated frequencies you really can’t tell so much which side is playing and has the dip. The responce is there and that’s what matters.

You can’t change the room dimensions, so parts of that will be there no matter what. Working on the left side first knowing the right side didn’t have those issues again just made the center better. The added gain from those frequencies also made it to where I did not have to attenuate the left channel at all. Both left and right have the same output gain in dsp and left and right sound to me the same


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## dgage (Oct 1, 2013)

That’s a lot to digest but I could see how that might be really nice. When I’m integrating subs into home theater, I also try to go with wider DSP changes to feather the audio instead of hitting it with narrow-Q hammers.

You said you applied an aggressor FDW (4cy). What do you mean there? What freq range is that?


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## oabeieo (Feb 22, 2015)

Aggressive 4cy FDW

That means I’m windowing out almost everything, if you open rew and add a FDW 4cy window is eliminating almost all the reflections out of the responce. So the excess phase and minimum phase are very similar except a couple areas. That way your not over correction on phase eq which makes a whole bunch of new problems and the sq goes to crap real fast .

I’m only interested in areas that are not minimum phase and to make it minimum phase you can only move so much phase around. It’s easy to get lost I’ve found out the hard way.

I only want to see where the minimum phase and excess phase move apart from each other and I’ve only been moving no more than 15deg of phase at a time. And as little a 1 or 2 degrees of phase under 1khz with again wider Q phase changes as well. Smooth transitions seem to sound a lot better. And as long as the peaks are brought down
The small little ripples are just not worth chasing.....seems the more you try to make a measurement look good the worce and harsh your filters sound.

It’s so easy to get lost with rephase because you can work it to make your measurements look perfect, and than your like “hot dam this is going to be so spaceship “ than you load the filter and it sounds like complete ass.

So yeah I was at the point of frustration and I went back to basics and used all rules of tuning I learned and applied that. So I went with low Q filters because you can’t go wrong and lots of them and it finally is super smooth sounding.


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## oabeieo (Feb 22, 2015)

......


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## oabeieo (Feb 22, 2015)

So a few points I’ve sorta learned about all this that maybe is worth noting. 

Phase is very position dependent. 

It’s easy to get lost trying to chase down phase issues that simply are not there. Most of the time the measurements are not enough to do anything with as far as super fine tuning phase. So the thought of super fine tuning the phase is not a practical way to think and isn’t something that should be thought of as possible. The measurements simply are not good enough to try to get perfectly flat phase in a car. 

When measurements are made , and time aligned and averaged you can get a good representation on the big phase issues and those are obvious and can be fixed easily and they sound great. Wraps that are in all the measurements are fixed easily with either a crossover linearization tool and some phase eq with the linearization tools. 
Cabinet resonances can be fixed as well however the ringing from a cabinet or a dash pod or door pod can be fixed however the ringing artifact that causes coloration to the sound can not be fixed with dsp. That is something that a whole thread should be dedicated to. Cabinets that are of lower Q and have better damping obviously will color the sound the least and can be made to sound absolutely excellent with phase adjustments. High Q cabinets or pods can be tamed , however the dry, rough, and simply unpleasant ringing high Q pods and cabinets exhibit can simply be the death of any real sound quality and completely kill the balance between harmonics and the music will always have that dry harsh ringing. A true IB or low Q pod with thick walls and heavy damping is not a choice. It is the only way! 

As far as excess phase, there’s a lot of excess phase. In a car so much early reflections makes up so much of what is heard it takes a lot of measurements to actually see what’s happening. Taking close measurements and working your way with measurements every 2” or so from the speakers to the seat will definitely help “see” what’s happening. 

Using a FDW seems to me the only good way to get a good idea on what to correct. 
A gate is not the ideal way to see the excess phase in a way that is meaningful. The excess phase will in most measurements be no where near the minimum phase. What I did was look at the minimum phase and look for areas that do not follow the magnitude, than look at the excess phase in that spot. The sound can’t break the laws of physics and can’t come out ahead of everything so it can only be lagging. So if a spot is ahead that really means everything else is lagging, which can be the case being that there is so much excess phase and a large portion of the sound energy has at least rubbed against something, maybe an underdash or a door or a window. 

What I mean by rubbed against. The sound pressure may have had an increase due to 1/2,1/4,or 1/8th space loading. And the area where your head is the biggest open space in the car, so it goes from higher pressure to lower pressure. Down near your feet in the footwell and the bottom of the door there is the seat, a console, a firewall , a dash , and other obstacles for the air pressure molecules to find resistance. That resistance will speed things up. Imagine trying to fix phase issues of mids and highs that are mounted inside of a bass reflex port on a sub box. 

This is where car audio ppl and the home audio ppl will simply have very different looking excess phase plots that don’t resemble anything like the minimum phase plot. Let’s think about what excess phase actually is. 

You have measured phase, excess phase and minimum phase. All are a component of measured phase. The excess phase is only the difference between the minimum phase and the measured phase. There isn’t such thing as minimum or excess spl. How loud something was measured should therefore only be associated with the minimum phase. 

This is why I think it’s very useful in a car to use the spectrograph and look at the energy content along with the waterfall and see how long something is hanging on. That’s another topic but still a very useful tool to help understand why and what when looking at minimum phase and excess phase plots. 

Lastly, the home audio ppl like to say things like “don’t do this” or “doing this is only good for one listening position”. Be careful when understanding what is being talked about here and where we may only care about one seat and will only ever care about one seat and our rooms are not even possible to “walk around in”. 

As far as one seat goes, I have (of course) drifted away from some of that only to try to get to the bottom of how to do this in a car and have had some good successes. I haven’t been able yet to fully say with great certainty because I’ve only been able to do this in one car. The one thing I’ve have good luck with in car deviating from the home audio norms is in regards to using phase eq to regain phase coherency after doing seperate left and right eq. 

Basically what I did was work the left and right seperate as far as magnitude and phase wraps (crossover linearization) and the normal twisting things from a multi-way. 
Than after left and right magnitude are flat use phase eq and make the measured phase match left and right. This has worked quite well actually and I know I’m experimenting into uncharted territory but I want to think that the measured phase plot at the center of my head (3 measurement averages) and use phase eq, iir all pass filters, fir all pass filters, and linearization tools and all the tools within rephase to make the measured phase plots match after doing left and right eq has worked good so far. 

The only problem I have had so far was after making some phase adjustments some frequencies changed in amplitude and would require a few tries and some sober listening to see why the amplitude changes happened, some were a result of fighting crossovers as a result of the phase changes and some were results of different interactions between the room and the speakers. So it took a few tries to get it and it works pretty good. 

I’m not of the opinion that it’s better than not doing seperate left and right eq , I have had much better coherence and a more accurate and proper sounding stage without the use of seperate left and right eq, however I’m starting to feel that above about 2k seperate left and right Eq with a post eq phase eq to make the left and right measured phase match is definitely better


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## oabeieo (Feb 22, 2015)

Iteration 21 now......

And need to retract some of what I said in above post. Not a whole bunch , but had a discovery that was in front of me the whole time. So the GD and excess GD. That is all I was missing. 

The excess GD really tells a lot so does GD. 
I was blindly using rew averaging and impulse centering (which works good on a single measurement.....with HF in the measurement) duh. Silly me: I was letting rew center the impulse and the impulse looked centered on the IR window. So I was like , thinking that’s it right....

So the excess phase now is not that far off from minimum phase. I was needing to use excess GD and get the “all” flat areas that don’t have a bump or peak to be at 0ms line. It was a matter of using signal delay, and re shaping the crossover after to accommodate for that change so the crossovers sum correct again. 

Than I was also thinking the peak is the beginning of the IR, forgetting about rise time! Oh yes rise time! I have heard ppl talk about it but never really cared about it I thought they were talking about the speakers transients that were factored into the design of the speaker. No no no no ...no I was wrong, rise time is how fast a speaker can start playing a given frequency. And midbass and subs obviously are before the peak of HF. I already knew that but never thought how I would have to consider that into any measurement. Yes. So the IR actually “starts” at the first sample any polarity movement is made in the IR. Not the peak.:.... 

So needless to say, I re did that entire thing. Luckily I saved the measurements before I changed anything in Rew so they were all Raw measurements. 

Now I can repair the excess phase and get excellent sounding stage. My own rephase tune is finally as good as a Dirac tune. And I have a level of confidence now because I can see, and hear the results and it’s working. 

Upon this realization I had tonight, I came across something strange. It appears to look like a LR4 is one cycle away on the LP side of a complimentary LR4 with the HP..... I am going to have to investigate this....I am probably wrong, but I triple checked everything and it come out the same every time. The wrapping from the Lr4 the way it wraps puts the HP to have exactly 1cy delay and the delay is made up from the “wraps of phase” caused by the filter. It seems to be and I’m just speculating at this point, but instead of using a fir filter to linearize the LR4 a 2nd order allpass IIR filter should be able to to remove that delay. So I’ll find out.
If my hunch is right, the IIr could be very powerful tool in this case


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## oabeieo (Feb 22, 2015)

It’s been while now since I posted this 

I still 100% believe fir on every driver is the cats meow ..... 

have you ever heard a 6.5 in a car that just gets with it.... and has super smooth midbass and plays low effortlessly and has a fun punch to it that sounds killer

Then stick the same driver in a different car and it’s not even close.... sounds spitty and boomy and just ****tz the bed.... well there’s a fix 

an fir can be applied to change the phase so that the speaker can play lower and sound way better!

how can it play lower? Well it can’t ...it still have the same excursions and limits , except the time in which those frequencies come out relative to others becomes the same with an fir ...

What happens is other frequencies playing out of time will cancel lower harmonics if the time is wrong.... that depends a ton on the door cavity ( Q of the system) and the size of car and reflections and loading space.... 

All of which can be measured and simply getting the speaker to not be so out of time with itself will make it sound like it plays lower , smoother, and a hell of a lot more punchy as the rise time is now a straight line up.
The step response will also show massive gains.... 

Another thing .... 

Ever had a dip at like 2.5k area caused by a cancellation..... you can use time alignment to solve some of it but you don’t want the rest of the band out of time to get a dip to come back up 

a simple fir applied to move some phase at 2.5 k on the offending side and dip gone , everything back in proper time .... and again measurable gains .....


rephase is soso so valuable tool and of course everyone should be using Minidsp above all else ..... something that allows fir


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## CrimsonCountry (Mar 11, 2012)

oabeieo said:


> It’s been while now since I posted this
> 
> I still 100% believe fir on every driver is the cats meow .....
> 
> ...


Great info throughout this thread! Always enjoy your deep dives even the parts that whoosh right over me head. I really need to get off my butt and mess with RePhase to learn how to use it. I'm still running the CarPC too so I could host all the FIRs I'd need via JRiver.

I was so close to buying either an upstream Dirac device (DDRC-22D, even the 24A but I'd prefer to keep it digital until my PS8) or the 8x12 outright last week but I may use this thread to get my feet wet with the free version (RePhase) for now. I know it should do the same thing in theory albeit with MUCH more time required.


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## oabeieo (Feb 22, 2015)

Oh my goodness 

send me your crossovers amd I’ll make some textbook firs for you

All I need is what slope etc and some close driver measurements with no crossovers applied ....


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## oabeieo (Feb 22, 2015)

So briber uses .wav iirc 
How powerful is your car pc 

It’s not fft based fir so the 13k sample rule sorta don’t matter , no smearing will happen .... otoh fft based fir is more direct as far as samples go so in a big way it’s better .... sorta , except you can use 256k taps on a 2ghz machine without much issues

if it’s like a 1ghz 256ddr machine it will use a lot of power ..... if bus speed low


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## preston (Dec 10, 2007)

Been reading this forum off and on for 13 years. Built and tuned a few systems. I just dropped into this last page because it came up as a new post and I'm honestly just like "uh....what ?"

Guess I should try a mini-dsp and this DIRAC thing. Don't think I could take the manual tuning anywhere near where you're at !


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## CrimsonCountry (Mar 11, 2012)

Awesome Andy! I'll try and get out there sometime soon to take some measurements for you to look at. I'll admit, I probably need to reevaluate my current xovers but its sounding decent to me at the moment so I'll send you what I have. With no EQ either or just no xovers? I'm still running the 2118s and Ultra Mini Horns setup and they're really messy in my truck with no corrections. The 2118s are peaky as heck in my door IB.

As for my CarPC, it's a (now) older 1150 mobo setup but its plenty powerful by car pc standards: i3-4160 dual core @ 3.6ghz (4 virtual cores w/hyper threading) and 16gb ddr3 running Windows 10. I've actually got the larger I7-4790 cpu I was going to use for overkill but my PicoPSU wouldnt play nice with it (assume TDP was too high).


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## oabeieo (Feb 22, 2015)

preston said:


> Been reading this forum off and on for 13 years. Built and tuned a few systems. I just dropped into this last page because it came up as a new post and I'm honestly just like "uh....what ?"
> 
> Guess I should try a mini-dsp and this DIRAC thing. Don't think I could take the manual tuning anywhere near where you're at !


Thanks , yeah once you get the hang of it it’s really not that bad .... it’s just plugging in numbers ..... definitely need a pen pad when setting up

Dirac is good ..... manual fir is evn better in many ways .... I use both ....
and it seems to be absolutely stellar 

all the tonality issues with Dirac are gone with a good linear crossover.
Try Dirac on some point source speaks you’ll be amazed .... no joke or a good two way book shelf


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## oabeieo (Feb 22, 2015)

So I think I’m on the edge of a new breakthrough with rephase...

i have successfully got rid of the left side dip from my door woofer.... the dip is at 71hz and it’s deep .... it’s right at my 80hz crossover and it’s still deep at 80 so I’ll call it 80) 

I now have that dip almost completely filled with a subsequent peak at 100 that came with it..... but that’s ok !

I basically added another 12db of linear I action to that driver , it uses a LR4 electrical slope with a 36 db acoustic slope ..... so I made a 36db linearization

I was just messing around and trued a 48db linearization and to my dismay that dip got filled in a lot.....

This is my hypothesis.... what if adding the extra 12db of phase rotation in the opposite way from “normal” crossovers turn is somehow have the similar effect of the driver as using a 48db slope of a crossover.... using a 48db slope of a crossover can improve off axis response..... with extra emphasis on “can”

I’m not exactly sure what’s happening here .... and I’m going to dig a little deeper and make some measurements to figure out what’s going on.

it’s been my understanding that the left side dip is attributed to the width of the car...... altering the timing shouldn’t do much for a room mode, however if the cancellation is caused by a harmonic and fundamental being out of time maybe something with that also....
(Meaning 160hz also moves 80x a sec in one direction. So if 160 and 80 aren’t playing together properly one could wipe the other out....hypothetically of course)

anyway it’s all conjecture at the moment.... I’ll get some solid facts and report back soon


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## oabeieo (Feb 22, 2015)

OK so basically what was going on I did some measurements of turns out to not be what I was saying .... lol 

what it did was remove the time alignment at 80hz ..... yeah .

so just at the dip area the left and right door speakers were basically playing in the dip at the same time ... it removed the 1.3ms I have on the left door but literally only at the dip.... so both doors playing at same time ..... no wonder what it’s louder and plays lower ..... TA ruins low response between door mounted woofers.... the wavelength is too long and adding delay simply causes cancellation.

the 48db slope on the linearization is such a fast attenuation it’s keeping pulling the time delay out of that driver literally only in the dip....

I will definitely use this trick more often 

it’s verycool .... definitely something an APF can’t do


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## oabeieo (Feb 22, 2015)

Here’s a couple videos I made , using Dirac and rephase filters in my car ...

this is the next step up from an 8x12dl , just very complicated (not rocket science either)

in the first video I talk about overlap , all overlapping filters are -6db down in the overlap area.... I didn’t mention that


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## bertholomey (Dec 27, 2007)

oabeieo said:


> Here’s a couple videos I made , using Dirac and rephase filters in my car ...
> 
> this is the next step up from an 8x12dl , just very complicated (not rocket science either)
> 
> in the first video I talk about overlap , all overlapping filters are -6db down in the overlap area.... I didn’t mention that


Boom! I’m looking forward to reading through this thread 


Sent from my iPhone using Tapatalk Pro


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## oabeieo (Feb 22, 2015)

bertholomey said:


> Boom! I’m looking forward to reading through this thread
> 
> 
> Sent from my iPhone using Tapatalk Pro


the older stuff has errors ..... but it’s a progression..... lol ....


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## oabeieo (Feb 22, 2015)

deleted ..... you tube deleted because copyright


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## oabeieo (Feb 22, 2015)

So when I talked about summing up close , I should have added , it actually sums in all different places......

and completely backwards on the other side of the speaker.


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## oabeieo (Feb 22, 2015)

Oabes tuning night the 102 , live

I’m really impressed with the wife’s camera mic , shows my system halfway decent


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## bertholomey (Dec 27, 2007)

oabeieo said:


> Oabes tuning night the 102 , live
> 
> I’m really impressed with the wife’s camera mic , shows my system halfway decent


Cool little video! And very cool that she is willing to run the video while you do the audio geek talk  She probably knows a lot more about phase than I do 


Sent from my iPhone using Tapatalk Pro


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## oabeieo (Feb 22, 2015)

bertholomey said:


> Cool little video! And very cool that she is willing to run the video while you do the audio geek talk  She probably knows a lot more about phase than I do
> 
> 
> Sent from my iPhone using Tapatalk Pro


lol exactly. I just got her this canon rebel DSLR, man what a camera..... she loves it , I had no idea what it even was or could do. ..... the video looks just as good as any tv channel. We’re never had anything this nice....


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## oabeieo (Feb 22, 2015)

preston said:


> Been reading this forum off and on for 13 years. Built and tuned a few systems. I just dropped into this last page because it came up as a new post and I'm honestly just like "uh....what ?"
> 
> Guess I should try a mini-dsp and this DIRAC thing. Don't think I could take the manual tuning anywhere near where you're at !


you can actually..... download rephase amd just play with it for two hours

import a measurement, just turn on amd off all types of filters and can see what happens in the time domain....😎

Just playing with rephase shows someone so much of how things work. It’s very easy, nothing different then what we’re been doing for years and years, just a couple new features to learn. Nothing crazy 🥳🤪

I’m actually a bit of a dummy also....I’ve done everything completely wrong thinking it was right for so long that I had to learn from my mistakes when ppl on here were telling me all along, but I was too stubborn to see it....😞


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