# Tuning your car using a pc based measurement setup



## npdang

This article is a very basic, step by step look at how to tune your car using a pc based measurement setup.

First, position the mic in the listening area. Here, I placed the mic just in the middle of where my head would normally be.










Now we will measure the time arrival of all the speakers.

Start by disabling/muting all the speakers. Starting with the left side tweeter, enable or un-mute it. Begin your mls measurement, looking at the time response (impulse response). I use WinMLS software for this application, a calibrated pre-amp/mic combo from Germany, and the M-Audio transit usb card. Be sure to read the other article on assembling a pc based measurement system for other options. Repeat again, testing each speaker by itself with the others muted. 

It is critical when doing this that you make certain your soundcard is "time synched". Most sound cards have a delay between the time that you begin playing a signal, and you begin recording. To make it even more frustrating this delay can vary between measurements. You would just want to make sure that your software+hardware is setup to compensate for this delay, so you can get accurate time measurements.

You should see something like this, which shows you the time arrival of the left tweeter in RED, and the right tweeter in BLUE.










Now, notice the first big spike with the RED line tweeter is going negative. That mean's your tweeter is wired in reverse polarity. A quick trip to your tweeter or amp to rewire and the problem is fixed. Also look at the BLUE line tweeter. The first initial spike, which corresponds to when sound from that tweeter first reaches the microphone, is exactly .6 milliseconds behind the RED tweeter. Now you know that you need to delay your RED tweeter by .6 seconds.










Now look at both tweeters. The initial spike is going positive, which means the tweeters are in the correct polarity. Also, the spikes are "almost" overlapping. The time difference between the 2 tweeters is now about .1 milliseconds which is good enough.

Basically, you would want to continue doing this until all the speakers are in the correct polarity, and the initial spikes are all overlapping.

Next up, we want to set the equalizer. To do this we would measure the frequency response of each driver separately and then take a look. You want to make sure that you include strong early reflections in your measurement, but not later ones. Generally, early reflections are responsible for tonal aspects, while later low amplitude reflections account for a sense of space.

RED = left tweet
BLUE = right tweet
BLACK = left mid
GREEN = right mid

If you're not using an accurate mic+preamp with a known, generally flat response make sure to take your frequency response measurements with a grain of salt.

(NOTE: in the bottom time window, the green/black lines are the mids time response. See how the initial spikes lineup? That shows you that the drivers are correctly time aligned. Look at the second large spike. That is possibly the cone breakup or a strong reflection from the Seas Excel w18 mid/bass. Also, note that although this driver shows a +12db spike when measured in an anechoic chamber (in the spec sheet), in MY car door there is no audible cone breakup at the listening position, and thus no need for a notch filter.)










Let's do an example starting with the left side. I like to begin with the tweeter. Looking at the RED line, we can see that below 3khz there's a bit of a dip, and above 10khz there's a -5db rolloff. So you would probably want to cut out a bit between 3 and 10khz, and boost a little bit around 2khz in order to flatten out the response. I level matched the mid and tweet pretty good by ear so I'll just leave that alone (BLACK and RED lines) Here are the settings I used, inputted into my Behringer dcx2496 processor. 










Now let's do the left side mid (BLACK line). It's pretty obvious there's a big hole centered around 750hz, and some bumps around 100hz, 200hz, 400hz, and 1800hz...probably from door cover diffraction and resonance.

Here are the settings I used for the mid. Notice how the boost and cuts corellate to the peaks and dips in the (BLACK) line frequency response:










Now to setup the crossover points. Look at the black line and the red line around 1-2khz. We know that for the Seas w18 Excel used in this example, distortion rises above 2khz. So keeping this driver below 2khz would be a good start. Also, the ribbon tweeter used in this example begins to show higher distortion around 2khz as well. So it looks like 2khz is a good starting point. I also like to use steep crossover slopes in order to minimize distortion, and make sure that the driver really isn't stressed too far beyond the crossover frequency.

Now that I've decided on a 4th order Linkwitz Riley crossover at 2khz, let's look again at the frequency response of the RED and BLACK lines at 2khz. The BLACK line or mid/bass already naturally rolls off at 24/db octave roughly at 2khz. So we don't need to set anything here. The RED line or tweeter "almost" seems to rolloff at 2khz 24db, but actually starts a little later at 1.5khz. So, I decided to try a 1st order (-6db) rolloff at 2khz to see what would happen.

Here is the final response of the left side mid and tweet. Notice the frequency response is much smoother overall, the mid/bass plays down flat to almost 40hz (no highpass set), and the crossover point actually ended up being around 2khz, 4th order (-24db). You also want to make sure that the tweet and mid are playing at the same level.










And here is the final summed frequency response for the left side. Tweet and mid/bass measured together. Notice around 2khz, the transition is smooth and seamless.










Now you basically want to repeat all that for the right side. Make sure that the right side is as good a copy of the left side as possible. Here is the frequency response of both the right and left side independently:










And here is the summed frequency response for the entire system, compare it to the unequalized, un-tuned frequency response at the very start (below). The optimized response is about +/- 3db ... so not bad. The system with only "textbook crossover" of 2khz 18db, and level matching by ear and no equalization has a response of about +/- 10db.










Finally, a good place to highpass the mids would be around 80hz -12db. It keeps the mid from distorting at high output levels, and it's generally a good place to bring the subwoofer in.

Here is a frequency response of the sub (located in the trunk rear firing) with no eq or crossover overlayed with the rest of the system:










Notice the dip around 60hz, and the natural 24db rolloff at 80hz. Doesn't it seem redundant to use a 24db slope on your sub if it's already got a natural rolloff at that point? I'll leave the issue of sub integration as an excercise, but my first thought would be to lower the output a little, boost the response at 60hz, and highpass the mid/bass at 80hz -24db. 

Now that was pretty easy wasn't it? 

Some points and conclusions I'd like to sum up that I think are fairly obvious. 

First off, "one size fits all" passive crossovers and/or texbook crossovers just don't work. That's because we can see that the in car frequency response is completely different from the frequency response on the spec sheets, which were taken in an anechoic chamber. Also, we have to take into consideration the actual natural rolloff of the driver before implementing a crossover.

Also, equalization is nearly a MUST have for typical car setup. Here I have the mids in the doors, and the tweeters in the kickpanels. Everything is heavily dampened. But still notice the somewhat extreme, and strange eq settings I had to use just to get the frequency response flat. It could take weeks or even months of discriminating listening to find all those flaws. 

Also, notice the shape of my equalization curves above. Those would be almost impossible to achieve unless you are using a parametric equalizer. Notice that I am equalizing the tweeter and mids separately from each other, as well as from each side. In order to get the best possible/flattest response and good imaging, an equalizer with this kind of flexibility and power is absolutely needed... and chances are you won't find it in an analog unit.

And make sure that even though you have a flat frequency response, you want to cross the speakers over where they will not distort or strain. Always bear that in mind when choosing crossover points, or boosting on the eq.

Lastly, notice how the summed system response looks a bit different from the left/right side response. If that happens, you can go back in and eq both sides to adjust the response flatter. However, although your tonality may improve from having a flatter summed response, the further apart your left and right side frequency response is from each other the worse your imaging will be. Try and fool around with small adjustments and balancing the need for tonal improvement with imaging together, and of course ALWAYS let your ears be the final judge!

In this case, I was very pleased with the sound right out of the box. Definitely one of the best cars I've heard so far. Imaging and staging is nearly perfect (for a car), and tonally it's very good. With the ribbon tweeter response easily goes out flat to 20khz unlike most tweeters. However, I still feel that it will take weeks of hard listening and fine tuning by ear to get that last 5%.... although you could certainly stop at this point since it sounds quite good.


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## npdang

Equipment used:

WinMLS 2000 software
M-Audio transit usb soundcard
microphone and mic preamp from http://www.ibf-akustik.de/2005/index05.html

And let's not forget the Behringer dcx2496 for eq/time alignment/crossover


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## Hobbes26

Awesome writeup.

Were you sitting in the drivers seat at the time of measurement? Legs get in the way of speakers in the kicks or doors quite a bit when you're doing the measurement. Especially when you've got the seat moved forward. ...Though this IS just the first/basic step...


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## npdang

Actually no I don't. My head would block the mic 

The only difference I can really notice when sitting in the seat though is the imaging smears a small bit to the right side. Moving my leg a little more out of the way or pulling the seat back so that I'm not blocking the mid clears up the problem. Tonally, I don't know what the effect would be although it sounds balanced to my ears.


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## aaron

Definitely required reading! Thank you for all the info. Now I know why such few cars ever seemed to truly impress me. Not many people were doing these kind of measurements ( that I knew about ) when I was involved with car audio about 10 years ago. It was all RTA.


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## FaintReality

When checking the frequency response of the drivers, the tweeters in particular, is it safe to not have the crossovers set? I am assuming you set the crossovers on the tweeters to protect them, just lower than ideal so you can find out where there natural roll-off is? Then again, the software does not use test tones so it makes me think they might be ok with no crossover during testing?

Also, when performing the test, is there an ideal volume setting on the HU like 1/2 way up, or can you just turn up the HU a few clicks since the frequency response should be the same regardless of volume?

Thanks, Dave


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## sqkev

Please bear with me, a few noobish question

What should I gate all the measurements at?

In your summed results, are they actual measurements or simulated?

I see that you measured tweet + tweet and then woofer + woofer 
THEN adjusted TA on each and xo and eq afterwards. 
For someone who doesn't have a dcx and considering passive, is it better to:

do gated FR of tweet and mid on one side, adjust level, xo, do measurement again

then
repeat the same for the other side and EQ and TA both sides?



one more question that is slightly off topic, 
when going active and setting all different levels for tweets, mids and possibly midbasses and subs, would the system be all linear at all different levels? 
Let's say that we match the voltage on all channels, yet not all amps produce the same amount of power at that certain voltage. In theory, this is the disadvantage to a all active system, correct?


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## npdang

sqkev said:


> Please bear with me, a few noobish question
> 
> What should I gate all the measurements at?
> 
> In your summed results, are they actual measurements or simulated?
> 
> I see that you measured tweet + tweet and then woofer + woofer
> THEN adjusted TA on each and xo and eq afterwards.
> For someone who doesn't have a dcx and considering passive, is it better to:
> 
> do gated FR of tweet and mid on one side, adjust level, xo, do measurement again
> 
> then
> repeat the same for the other side and EQ and TA both sides?
> 
> 
> 
> one more question that is slightly off topic,
> when going active and setting all different levels for tweets, mids and possibly midbasses and subs, would the system be all linear at all different levels?
> Let's say that we match the voltage on all channels, yet not all amps produce the same amount of power at that certain voltage. In theory, this is the disadvantage to a all active system, correct?


I didn't gate any measurements, however I did use 1/3 octave smoothing. These are actual measurements, not simulations.

The steps I used were:

1. time alignment for all drivers
2. level matching
3. handling driver to driver transition which does include use of eq as well as xover
4. equalization 

It wouldn't matter if you went active or passive, the steps should be the same.

Your last question I think is a bit confused (or perhaps I am). Level matching is matching the sensitivity of all the drivers... not the output voltage of your processor or amplifier. 

You can measure a change in system frequency response with volume, however it isn't *that* big of a deal Imo... and it would be due to things like driver non-linearities, reflections, power compression, etc.

And for things like the sensitivity of our ears to different frequencies changing with spl, that's fine as well. As long as you playback the recording at the original volume, it should sound accurate. You shouldn't expect it to sound the same at a significantly lower, or higher volume Imho.


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## corrado

hi npdang,

I have already bought all the hardware require for the measurement, now just deciding on which software to get.

one question, must I point the mic to the rear when measuring the sub, assuming my sub is in the trunk?

please advice, thanks.


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## 300Z

Cool, another Corrado owner here...  how did you found out about this site?

As for your question about the mic placement when measuring the sub, it shouldnt matter since low freq arent controlled dispersion...

Regards
Leo


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## corrado

hi there, thanks for the response....

but I do not own a corrado...how I wish I do....

got to know this site via a friend....

btw, my SS exact should be in next week....

bought it after reading review from sounddomain....I believe you are there

cheers.....


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## 300Z

Yes i'm on sounddomain, ECA and some other forums as well...

What sized of the Exact you got?

Leo


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## Grand Masta

FaintReality said:


> When checking the frequency response of the drivers, the tweeters in particular, is it safe to not have the crossovers set? I am assuming you set the crossovers on the tweeters to protect them, just lower than ideal so you can find out where there natural roll-off is? Then again, the software does not use test tones so it makes me think they might be ok with no crossover during testing?
> 
> Also, when performing the test, is there an ideal volume setting on the HU like 1/2 way up, or can you just turn up the HU a few clicks since the frequency response should be the same regardless of volume?
> 
> Thanks, Dave



Nguyen, I am curious about this as well.


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## daitrong

Grand Masta said:


> Nguyen, I am curious about this as well.


i would guess you have your xovers set before measuring... You're not going to need the Frequency response at 350hz for your tweeters since you're never be playing your tweeters down that low. Plus if you were to measure it that low, the FR would like all jacked up.. lol Maybe nguyen will chime in and let us all know for sure.


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## npdang

I do set the highpass crossover when measuring tweeters.


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## Grand Masta

Do you set the highpass on the midrange or midbass drivers to prevent them from overexcursion in the lowest octaves?

Also, why is the tweeter present in this graph? Looks like the "midrange" of the tweeter was properly attenuated by the filter and then the tweeter began playing again in the lowest octaves. I am confused.


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## npdang

It's background noise... you see it's present for the mid's as well.

For the mid/bass I don't highpass, but you can. It's really not even necessary for the tweet as long as you're not driving it too hard.


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## Grand Masta

Hmmm... What is causing this background noise in the lowest octaves? Does it affect the subwoofer FR measurement as well?


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## npdang

It's not a big deal since it's mostly under 20hz. If you really need accuracy around that area you can always use a different stimulus along with pre/post emphasis to remove noise.

Imo when tuning bass in the car, it's not always about having a flat response but having a smooth response and seamless integration between mid and sub that makes it sound good.


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## Greg200SE-R

npdang said:


> Start by disabling/muting all the speakers. Starting with the left side tweeter, enable or un-mute it. Begin your mls measurement, looking at the time response (impulse response). I use WinMLS software for this application, a calibrated pre-amp/mic combo from Germany, and the M-Audio transit usb card. Be sure to read the other article on assembling a pc based measurement system for other options. Repeat again, testing each speaker by itself with the others muted.
> 
> It is critical when doing this that you make certain your soundcard is "time synched". Most sound cards have a delay between the time that you begin playing a signal, and you begin recording. To make it even more frustrating this delay can vary between measurements. You would just want to make sure that your software+hardware is setup to compensate for this delay, so you can get accurate time measurements.
> 
> You should see something like this, which shows you the time arrival of the left tweeter in RED, and the right tweeter in BLUE.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Now, notice the first big spike with the RED line tweeter is going negative. That mean's your tweeter is wired in reverse polarity. A quick trip to your tweeter or amp to rewire and the problem is fixed. Also look at the BLUE line tweeter. The first initial spike, which corresponds to when sound from that tweeter first reaches the microphone, is exactly .6 milliseconds behind the RED tweeter. Now you know that you need to delay your RED tweeter by .6 seconds.


npdang, I am trying to measure the time arrival for each individual driver in my system, just like the measurement in your first graph above. I am having trouble getting such a nice, "clean" and distict spike to appear in my measurements. 

My setup is: M-audio Mobilepre, Behringer ECM8000, demo versions of lsplab, WinMLS and Freeprobe running on a current model Vaio notebook. In my car I have a Clarion DRZ9255 with active 2-way front comps and a single sub. All I want to do right now is accurately measure and set my time-alignment. 

Can you go over the initial settings for WinMLS (Seems like you use this app often)? AFter calibrating the output/input levels, are there any special settings that need to be changed? 

the results I am getting are jagged lines, and then a not-so-distinct spike... not a flat line leading up to a clearly defined spike like in your graphs. The readings can change somewhat between measurements as well. I know it's not easy to elaborate on such a complex subject but any info you (or anyone else) can provide would be appreciated!


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## JohnSmallberries

I am also trying to set up a system like this. I have the ECM8000 as well and what I want to know is, how do I calibrate this microphone, or do I really need to in order to get tests that would work for setting eq etc for my pxa-h701?

Not being much of a german reader I don't have any idea what the site that npdang linked has to do with the mic...

Thanks a bunch!


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## dennisp

Anyone who has the MobilePreUSB and WinMLS, could you tell me what settings you have here? Thanks.








I'm just confused as to how to calibrate the sound card.


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## JohnSmallberries

i have a similar issue.. im not sure how to use the mobile pre in winmls...

I have a ecm8000 and tried using truerta as well but I think it was not setup right as the adjustments I was making in the car made the audio sound change but the response in these programs didn't seem to change.


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## 03blueSI

That is wierd, I have the ECM8000 and a behringer mic preamp and I have used TrueRTA in the past and I could easily see the changes. I am now using FuzzMeasure (MLS for the mac) and can see changes fairly easily as well. One of the big things to do though is to use multiple mics if you can afford to and sum the response


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## audionutz

Greg200SE-R said:


> npdang, I am trying to measure the time arrival for each individual driver in my system, just like the measurement in your first graph above. I am having trouble getting such a nice, "clean" and distict spike to appear in my measurements.
> 
> My setup is: M-audio Mobilepre, Behringer ECM8000, demo versions of lsplab, WinMLS and Freeprobe running on a current model Vaio notebook. In my car I have a Clarion DRZ9255 with active 2-way front comps and a single sub. All I want to do right now is accurately measure and set my time-alignment.
> 
> Can you go over the initial settings for WinMLS (Seems like you use this app often)? AFter calibrating the output/input levels, are there any special settings that need to be changed?
> 
> the results I am getting are jagged lines, and then a not-so-distinct spike... not a flat line leading up to a clearly defined spike like in your graphs. The readings can change somewhat between measurements as well. I know it's not easy to elaborate on such a complex subject but any info you (or anyone else) can provide would be appreciated!



Ditto!!!! Anxiously awaiting the answers


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## npdang

The settings above look correct, except perhaps the input line should typically be a "line in". I'm not sure what an analog connector is.

Also, the "yes, my sound card is synched" refers to the fact that there is a non-changing time delay between when the mic begins playing and when it begins to record. Usually I prefer to select NO, use loopback. What happens is you can use one end of your stereo plug as a loopback to the soundcard input to calculate the delay before each measurement. This enables you to get accurate time measurements for soundcards whose record and play time vary with each measurement.

As far as the jagged lines, non-distinct spike... make sure the output level is turned up to a reasonable level (above whatever noise you may have). I'd also look at the vertical and horizontal scales... maybe zooming in or out can help bring out a clearer picture. And lastly, make sure your mic/setup is actually working! Running a quick FFT postprocess should show you the frequency response. If it doesn't look right, chances are your impulse response isn't right either.


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## audionutz

Figured out one problem...I am trying to use Lsplab, NOT winMLS! What about the proper settings for lsplab???


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## npdang

Do you have any screenshots? Does it show the correct soundcard and input channel being used?


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## audionutz

np, it works now. Unfortunately I cant send screenshots b/c the "puter i'm using for this is not networked via LAN yet...Thanks so much for the help! 
As an update I am awaiting the eval codes to activate the trial version of WinMLS. Maybe that one will be alot easier to decipher in terms of arrival times.


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## npdang

Keep us updated. Maybe it's just the scaling on the plot that makes it difficult to read.


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## audionutz

OK, got the WinMLS codes, it is installed and operational. I calibrated the mic, sound card, and good to go. 
NOW, with reference to the impulse response window....THe window is too large and I cant see the exact arrival time (x axis goes all the way to 400 ms!) in the help tutorial, it says you can set your window to a small increment using the impulse response toolbar. BUT, heres the problem----WHERES THIS TOOLBAR!?!? I cant find it, it wont open from anywhere for me, and it is not listed under "toolbars". ARGH! Did I d/l an incomplete file? Is it only available on the purchased version, and not the trial version? Am I just an idiot? 
Inquiring minds wanna know!
LOL


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## npdang

I believe you can right click on the window and there should be a menu allowing you to change it.


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## audionutz

Will try it, thanks man!


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## t3sn4f2

Hi Npdang, can you help me with something. According to your tutorial a .1 millisecond difference in arrival time between left and right wave is good enough. What is the maximum that I can move my head around in the car once the car has been optimized the way you say?


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## z_accoustics

I have a UA-5 which has done a fantastic job at showing me frequency responce and i was able to accomplish wonderful time alignment, but not by impulse response.

The problem: everytime I do a measurement, the impulse response shows up at a different start time even when i've made no adjustments at all. The time is +/- up to 2ms difference between identical mesurements. I read the part of the post that says make sure your soundcard is "time synched" but i'm not sure how to make sure of that. Is there some setting or diagnostics program that lists the card as being "time synched" or not. Is it a driver adjustable setting? Right now i'm happying with the setup because I have coaxils, so it's not too complicated t/a wise, but I plan to upgrade to a 720prs 2-way and then i'm going to need to be able to measure accurate impulse response arrival times. Any solution to getting my current computer setup to do that? thanks.


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## npdang

If you run one of the output channels back into one of the input channels, your program should be capable of detecting the time delay (and sometimes frequency response and phase) of the system with each measurement and correct for.


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## z_accoustics

Yepp, i did the loopback calibration but the problem is that I get different delays by repeating the same measurement. Meaning it's inconsistent delays


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## npdang

I'm not familiar with the software you're using, but WinMLS will correct the time delay with each measurement as long as you leave the loopback connected.


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## z_accoustics

Oh... you leave the loop back connected? I don't have a loop back connected. When i calibrated the audio device I plugged the mic input into the output. Now I have my E800 mic plugged into the mic input. Where does the loop back go? I am using digital coax output to my head unit. Do i plug the headphone output in to the 2nd mic input?

The UA-5 input/output details are listed here: http://www.roland.com/products/en/UA-5/specs.html


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## npdang

You have a stereo line out, and a stereo line in. Take the left channel line out and plug that into the left channel line in. Go to winmls and configure it to "use loopback" and set the loopback channel to left. 

Your playback channel is now the right output channel. Connect that to your amp, hu, etc. Connect the mic to the right channel line input. That's it.


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## z_accoustics

excellent directions, thanks! I should probabbly still be able to use the digital out to digital in on my DEC instead of using the RCA right output.


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## durwood

I know this thread is old, but I was just pondering some things.

1) When making impluse measurements to use for time delay, where is the mic placed? Is it Directly centered in the middle of your headrest or is just off to the right side of your headrest? The reason I'm asking is because I think it will make a difference to where your center stage images...I think. Maybe I'm way off but I'd like to know where you placed it because I can't tell from your pictures.

2) If it's placed in center of your head rest, then you should theorically end up witha center image directly in front of you. If its placed off to the right side, would you end up with a center image more in tune with a center image centered somewhere between your mirror and directly infront of you? 

3) When making the frequency response measurements, do you keep the mic placed facing forwards or do you turn it or angle it to each side as you take measurements for each side of the car? The reason I ask this is because our ears don't point directly forward and I have noticed drastic differences regarding facing forward vs turned to the side. When I turn it to the side I get a frequency response that agrees more with my ears than with facing directly forward. (Example, I had a huge hole in my response at 1Khz when facing the mic forward, but turned to the sides there was no 1Khz hole) So that brings me to wonder what the best angle is to place the the mic for measuring frequency response, 0, +/-90deg or something in between 0 and 90? At what angle are our outer ears postioned? (not the holes in the side of my head ) 

I know the ECM8000 mic is supposed to be omnidirectional, but I'm not sure I trust using it that way for measuring incar frequency response.

Thoughts?


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## handy

so what mic you suggest too?
cause ecm 8000 is standard for soundcard measurement.


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## npdang

durwood said:


> I know this thread is old, but I was just pondering some things.
> 
> 1) When making impluse measurements to use for time delay, where is the mic placed? Is it Directly centered in the middle of your headrest or is just off to the right side of your headrest? The reason I'm asking is because I think it will make a difference to where your center stage images...I think. Maybe I'm way off but I'd like to know where you placed it because I can't tell from your pictures.
> 
> 2) If it's placed in center of your head rest, then you should theorically end up witha center image directly in front of you. If its placed off to the right side, would you end up with a center image more in tune with a center image centered somewhere between your mirror and directly infront of you?
> 
> 3) When making the frequency response measurements, do you keep the mic placed facing forwards or do you turn it or angle it to each side as you take measurements for each side of the car? The reason I ask this is because our ears don't point directly forward and I have noticed drastic differences regarding facing forward vs turned to the side. When I turn it to the side I get a frequency response that agrees more with my ears than with facing directly forward. (Example, I had a huge hole in my response at 1Khz when facing the mic forward, but turned to the sides there was no 1Khz hole) So that brings me to wonder what the best angle is to place the the mic for measuring frequency response, 0, +/-90deg or something in between 0 and 90? At what angle are our outer ears postioned? (not the holes in the side of my head )
> 
> I know the ECM8000 mic is supposed to be omnidirectional, but I'm not sure I trust using it that way for measuring incar frequency response.
> 
> Thoughts?


1. Put it where you want it to image.

2. Yes.

3. Good point. I think experimentation is key. You can also do a spatial avg. between the left and right side doing that, or a number of other things. Sitting in the car certainly impacts the response as well.

What's really important is to play with everything. The car is really the worst possible place to measure anything, and really requires alot of experimentation with time windows and mic placement to get a useful measurement.

Try removing the capsule from the mic tube and placing it on a headband over your ear. Do an avg. from left to right side while sitting in the car. The results will be interesting. Also, experiment with a flex time window or something similar.


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## durwood

npdang said:


> 1. Put it where you want it to image.
> 
> 2. Yes.
> 
> 3. Good point. I think experimentation is key. You can also do a spatial avg. between the left and right side doing that, or a number of other things. Sitting in the car certainly impacts the response as well.
> 
> What's really important is to play with everything. The car is really the worst possible place to measure anything, and really requires alot of experimentation with time windows and mic placement to get a useful measurement.
> 
> Try removing the capsule from the mic tube and placing it on a headband over your ear. Do an avg. from left to right side while sitting in the car. The results will be interesting. Also, experiment with a flex time window or something similar.


Cool thanks! Great idea with the headband. I might have to try that. I've already experimented with a 7" nerf ball. I just need to make my rig more solid and I'm going to do some comparing.


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## freeride1685

I am so excited that we have all this technology to mess with these days!

Some more questions:

1. It is an active crossover, and can differentiate between the tweeters and the mids, etc, as shown....but i am a bit confused....this will be used in line between the source and the amps correct? this is NOT a replacement for the passive crossover networks that come with my speakers sets, correct?

2. I have been having trouble wrapping my head around this, as i understand the concept of tuning for the driver, but as i usually have passengers (who are fellow audio enthusiasts) riding with me, how can i best tune for the two front seats together? obviously, it would be a rough estimate of the same concepts, but i just cannot really figure out what would be best for both of us.

3. This Behringer UltraDrive Pro is so awesome and yet not very expensive....i am a bit confused as to why my Rockford 3sixty.2 cost $700 and does not have the same levels of precision as this.....or maybe i am mistaken and i should look deeper into the RF....i am just tempted to sell it and get this one instead...recommendations?


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## freeride1685

another couple questions.....

how did you connect the DCX2496 into the car's power supply?

how did you run the mic cord out of the car to the laptop with all the doors closed? did you crack the window a tad?


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## npdang

Search general audio, there's a power supply for it. I run the mic cord out the trunk or a side window usually.

1. this is a replacement for passive network. It goes in between the amp and the headunit. You need a separate channel of amplification for each driver.

2. take a spatial average of both seats.

3. car audio products are always expensive... thats why we have "diy" mobile audio.


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## freeride1685

hmmm...that would mean that i need 11 channels to work with...and that's just a whole lot.....ehhh i suppose this is not the product for me just yet...although that is nice since it leaves something exciting open for the future!  

i'm sure i will still be relatively happy with the flexibility of the 3sixty.2 and i can still use MLS to analyze the system, albeit to a lesser extent.


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## freeride1685

although.....there is no reason why i can't get this unit now and treat it just like the 3sixty.2....i have a feeling the build quality of this device may be better, lending itself to longevity and top notch SQ. Also, it will give me the flexibility to mess with it more in the future.

so i will need to get cables that go from male RCA to female XLR, correct?


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## michaelsil1

Did anyone figure out how to get a more distinct graph with the time alignment? Mine seems all squished together.


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## npdang

michaelsil1 said:


> Did anyone figure out how to get a more distinct graph with the time alignment? Mine seems all squished together.


You need to change the x-axis (time) scale. Usually, you're only interested in the first 5 or 10 ms not the whole 300ms+ etc.


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## cvjoint

Has anyone gotten MLS to work with Vista? I have a the 64bit version of Vista and I find it impossible to install. I tried the compatibility feature per developer suggestion but it did not work.


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## michaelsil1

cvjoint said:


> Has anyone gotten MLS to work with Vista? I have a the 64bit version of Vista and I find it impossible to install. I tried the compatibility feature per developer suggestion but it did not work.


Try installing Vista's Service Pack 1 (400+ MB) it fixed a large number of issues users were having.


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## cvjoint

It is updated I believe. You run Vista?


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## michaelsil1

cvjoint said:


> It is updated I believe. You run Vista?


No I don't (Vista is a bloated pile of **** IMO); the Service Pack was just released a couple of days ago it.


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## cvjoint

michaelsil1 said:


> No I don't (Vista is a bloated pile of **** IMO); the Service Pack was just released a couple of days ago it.


Not going to disagree with you on that. I don't think anyone will.


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## cvjoint

Update: I did have the service pack and everything else up to date on my Vista laptop.

WinMls2004 and M-Audio Transit are NOT compatible with Vista no matter what compatibility feature you select or admin privileges. 

I hope these issues will be solved in the near future as XP has been announced to be off the shelves this summer.

If anybody does crack this bastard let me know.


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## michaelsil1

cvjoint said:


> Update: I did have the service pack and everything else up to date on my Vista laptop.
> 
> WinMls2004 and M-Audio Transit are NOT compatible with Vista no matter what compatibility feature you select or admin privileges.
> 
> I hope these issues will be solved in the near future as XP has been announced to be off the shelves this summer.
> 
> If anybody does crack this bastard let me know.


Have you dumped Vista?


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## cvjoint

Nope, I just have limited access to tuning as I had to give the XP laptop to my mom.


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## BigRed

well just played with lsp lab and its really frustrating me. I take a measurement and get a reading. I don't change anything or move and take another reading and its completely different. I don't get it. Its the trial version, but according to npdang its supposed to be really good.??


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## freeride1685

a couple questions.....

i don't recall reading about time alignment for the sub...and if not, why not? i assumed that the sub would be your point of reference, being that it is usually the furthest away of all drivers, and that all other drivers should be delayed based on the sub. this seems especially important to help the mids bring the illusion of sub bass to the front stage.

i checked the link for the calibrated mic/mic preamp from germany and it was dead. is this the behringer ecm8000 that people have mentioned? i liked the initial setup that npdang was using but i couldnt find out about that one piece.

EDIT: found the page, and found the calibration combo...damn that is more expensive than i had hoped...and it has to ship from germany 

another question too...referring to my post awhile ago about trying to get the best compromise for the two front passengers, npdang suggested taking measurements for both seats and then summing them together. now i wonder if it would be legitimate to just put the mic right in the middle of the two seats in the first place..


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## 14642

Wow, great post.

I have a few of things to add.


1. Don't sweat the arrival time of the tweeters too much. We use level at high frequencies to determine their locations. Also, measuring the phase of the tweeters is tricky. that initial peak may change + and - with repeated measurements. Phase between the tweeter and mid is important and you can check that by looking at the frequency response of the combination.

2. Once you've done the EQ, move the microphone to a few different locations (around the driver's head postition) and make sure there aren't any huge differences in the 1k-10k region. If there are, make an average of those curves (plot them on graph paper or use apost-processing routiine to average them. For those of you who are ezxel users, output the measurements as a text file, import them, average them and graph them either in excel or export them to the analyzer program. A spatial average should be magnitude only with no phase measurement. 

3. If you're running an active setup, you should ALWAYS install a capacitor on the tweeters and an octave or two below your intended crossover frequency. It won't affect the tuning but it will protect the tweeter. Measuring the arrival time of tweeters with the high pass filter engaged is OK.

4. the initial spike in the impulse response measurement is high frequency information. the steeper the spike the higher the frequency. When you're measuring impulse responses for setting time alignment, low pass filters can make identifying the arrival time more difficult. the spike will look like a hump rather thn a spike and the real arrival time is not necessarily the top of the hump--it's a point where the level of the rise (the level of the speaker's output is high enough to be heard relative to the other speakers. Use a full range signal for midbass and subwoofers--that will make the spike steeper--you're trying to identify the location of the speaker without regard to group delay caused by irregular frequency response. 

5. If your measurement is 20dB above the background noise, that's loud enough.


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## Ge0

Andy Wehmeyer said:


> Wow, great post.
> 2. Once you've done the EQ, move the microphone to a few different locations (around the driver's head postition) and make sure there aren't any huge differences in the 1k-10k region. If there are, make an average of those curves (plot them on graph paper or use apost-processing routiine to average them. For those of you who are ezxel users, output the measurements as a text file, import them, average them and graph them either in excel or export them to the analyzer program. A spatial average should be magnitude only with no phase measurement.


My methodology towards taking frequency response measurements is rather different to what people here consider "convention". I don't use a single microphone set on a tripod in the listening position. I've never obtained pleasing results in doing so. However, I do feel my method stays true to what you state above and could not agree with you more.

I prefer to take my bodies attenuations, reflections, etc. into account when taking frequency response measurements. After all, your body does create obstructions in the path between your ears and the loudspeakers while in vehicle. This WILL affect frequency response will it not?

I use binaural microphones. One mic mounted as close as possible to each ear. My mic preamp has two microphone inputs. Software is set to sum the response from each microphone. Software is also set to collect long averages of 100 readings. I make sure to rotate my head, or make "normal" movements in the general region while collecting readings. In my opinion this is equivalent to "waving the microphone around the head region" that others have stated. That, and I am averaging in readings from various microphone positions that are relevant to where your ears may be as others have said is so important...

I can't say this is right or wrong but does seem to work VERY well for me. I've never achieved smoother tonality than what I do now. Can you comment? Am I truely a nut case or do I have valid points? Holes in my theory? 




Andy Wehmeyer said:


> 4. the initial spike in the impulse response measurement is high frequency information. the steeper the spike the higher the frequency. When you're measuring impulse responses for setting time alignment, low pass filters can make identifying the arrival time more difficult. the spike will look like a hump rather thn a spike and the real arrival time is not necessarily the top of the hump--it's a point where the level of the rise (the level of the speaker's output is high enough to be heard relative to the other speakers. Use a full range signal for midbass and subwoofers--that will make the spike steeper--you're trying to identify the location of the speaker without regard to group delay caused by irregular frequency response.


Thank you!!! I was pondering this very thing yesterday when tweaking in new speaker locations for a four way setup. My entire time alignment strategy would change if I had crossovers enabled. For instance, with no crossover enabled on my midbass, difference in arrival the arrival time between midbass and opposing side midrange was 14.7ms. With crossovers enabled, the peak looked like more of a slow moving HUMP than it did an impulse peak. If measuring the top of the hump, the difference in arrival time to the opposing midrange was 15.3mS. I was stumped. What was the "correct" setting. This is confusing simply because I truely don't understand what I am physically looking at when reviewing an impulse response plot. Where did you dig this information up? Can you recommend some good reading to help drag the mystery behind these plots?

Thanks again. Glad to see you being an active contributor to DIYMA. I liked hearing what you had to say on the Carsound forums, but haven't frequented them in quite a while.

BTW, did you get my PM regarding binaural microphones? 

Ge0


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## michaelsil1

I would like to hear more regarding binaural microphones.


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## Ge0

michaelsil1 said:


> I would like to hear more regarding binaural microphones.


I have found minimal info so far googling the term "binaural microphone". The few that come to mind are overly priced mediocre solutions.

There is a good site describing how to build your own for $20 though. Nice, but I'd like to track something down that is prepackaged rather than go into yet another fabrication cycle. It's hard enough finding time to work on my car. Spending that time in my basement constructing a mic array will need to wait until fall.

Ge0


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## The Drake

Great thread. I have been wanting to do some of these tests for a while and just had time to do it lately. I am trying out WinMLS right now and picked up a mobilepro USB and wasn't really sure which microphone to pick up cause guitar center or Sam Ash didn't have the behringer mic that was mentioned int his threat, but the person at Guitar Center suggested a condenser microphone so I used a Sterling Audio ST51 for my tests. 

I just want to know if I am getting the correct responses here. I have a Alpine W200/H701 running full 5.1 system btw.
View attachment 5905


I know its a bit crowded, I can post individual responses as well if needed. I really dont have much idea what I am doing other than reading through these posts.

Oh yeah I did have the car running so that's why there is a lot of activity in the lower frequencies on all responses.


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## Jopop

Sorry for this stupid post but  Here are my results, do they seem reasonable?



Code:


--- Actual distances measured with software ---

L mid 4.78ms 5.39ft
R mid 7,55ms 8.51ft
Sub   6.03ms 6.8ft
L tw  3.56ms 4.01ft
R tw  4.67ms 5.27ft


--- What i think each speaker should be delayed by in the time delay menu if i am getting the manual right---

L F (Left mid)    = 8.51 - 5.39 = 3.12ft = 95.2cm
R F (Right mid)   = 0ft
Subwoofer         = 8.51 - 6.8 = 1.71ft = 54.4cm
L R (Left tweet)  = 8.51 - 4.01 = 4.5ft = 136.0cm
R R (Right tweet) = 8.51 - 5.27 = 3.24ft = 98.6cm

I have an alpine 9887 and the manual says to subtract the distance from one of the speakers from the distance to the farthest speaker, and that is the value the speaker should be delayed. Is this correct, and does my notes reflect this?

Also the values in centimeters are not the same you'd get when you convert feet to centimeters, they are rounded to the nearest value available on the head unit.


If interested i used Smaart v6 software, a huge behringer mixing console as a preamp (i am a cheap bastard so i went with what i had), a cheap behringer measurement mic and various crappy wires. I took several readings for each speaker and they were the same every time, except for the sub which started out a bit higher but eventually stabilized on 6.03ms (it stayed at 6.03 for 8 measurements or so). Oh and for the sub to measure correctly i had to turn LP off.


Oh and my frequency response was fairly flat 80-14khz, there was a just a wide dip in the 300-500hz range. I take the reading with a large grain of salt though as the mic isn't all that (esp. below 100 and above 15k).


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## michaelsil1

Jopop said:


> Sorry for this stupid post but  Here are my results, do they seem reasonable?
> 
> 
> 
> Code:
> 
> 
> --- Actual distances measured with software ---
> 
> L mid 4.78ms 5.39ft
> R mid 7,55ms 8.51ft
> Sub   6.03ms 6.8ft
> L tw  3.56ms 4.01ft
> R tw  4.67ms 5.27ft
> 
> 
> --- What i think each speaker should be delayed by in the time delay menu if i am getting the manual right---
> 
> L F (Left mid)    = 8.51 - 5.39 = 3.12ft = 95.2cm
> R F (Right mid)   = 0ft
> Subwoofer         = 8.51 - 6.8 = 1.71ft = 54.4cm
> L R (Left tweet)  = 8.51 - 4.01 = 4.5ft = 136.0cm
> R R (Right tweet) = 8.51 - 5.27 = 3.24ft = 98.6cm
> 
> I have an alpine 9887 and the manual says to subtract the distance from one of the speakers from the distance to the farthest speaker, and that is the value the speaker should be delayed. Is this correct, and does my notes reflect this?
> 
> Also the values in centimeters are not the same you'd get when you convert feet to centimeters, they are rounded to the nearest value available on the head unit.
> 
> 
> If interested i used Smaart v6 software, a huge behringer mixing console as a preamp (i am a cheap bastard so i went with what i had), a cheap behringer measurement mic and various crappy wires. I took several readings for each speaker and they were the same every time, except for the sub which started out a bit higher but eventually stabilized on 6.03ms (it stayed at 6.03 for 8 measurements or so). Oh and for the sub to measure correctly i had to turn LP off.
> 
> 
> Oh and my frequency response was fairly flat 80-14khz, there was a just a wide dip in the 300-500hz range. I take the reading with a large grain of salt though as the mic isn't all that (esp. below 100 and above 15k).


Do you drive on the wrong side of the road?


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## Jopop

michaelsil1 said:


> Do you drive on the wrong side of the road?


Hmm i don't get this question - my car is a LHD.. you can clearly see the left speakers are closer than the right speakers. What were you referring to?


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## michaelsil1

Jopop said:


> Hmm i don't get this question - my car is a LHD.. you can clearly see the left speakers are closer than the right speakers. What were you referring to?


It appeared that you were measuring for a right hand driver.


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## S3T

I think it's a BIT egoistic approach  What about passengers? They will suffer from asyncronous, out of phase sound and strange frequency response a lot more than before...


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## 03blueSI

if you are trying for a 2 seat car take impulse measurements at a central point between the seats at ear level and time allign to this point. For eq average a series of measurements from both the driver and passenger position


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## Asentaja

This is probably a silly question, but would it be possible to adjust the phase (if you have variable phase shift) based on a phase response measurement?


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## digitalhifi

Asentaja said:


> This is probably a silly question, but would it be possible to adjust the phase (if you have variable phase shift) based on a phase response measurement?


Nope. Phase response measurements (at least traditionally) are measurements of the linearity of a driver, or speaker system, and are in the frequency domain (ie. they don't have relevance to time without using the inverse Fourier transform) To measure the phase delay you have to know that your drivers are in alignment time wise. The difference in peaks (or any equivalent position) of a sinusoid from both speakers is the phase delay. Generally we just invert phase or time align to adjust for these differences in cars. Variable phase is more common in home stuff.


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## Asentaja

Ok, thanks for clearing that up.


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## StickToRhythm

A few thoughts regarding the comments in the original post about the redundancies of applying a crossover slope to a driver that already naturally rolls off around the intended frequency: 

While the natural rolloff of a driver may behave the way you intend for the first couple octaves beyond the crossover point, what happens after that? Does it continue dropping off like it would if a crossover were applied, or does it even out a little bit, just at a lower level? One of the images in the original post appears to indicate the latter.

While I agree that it seems unnecessary in terms of frequency response, it seems to me that there is still value in implementing a crossover as it will still likely attenuate much more signal outside the pass band. By applying the crossover, you will be making the most efficient use of your amps by only requiring them to amplify the portion of the signal intended to be reproduced on that channel. And there would probably also be a positive affect on the power handling capability of the driver, however small.


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## Dillyyo

I need to bring this thread back from the dead and maybe someone can clarify for me. My question is in regard to the sound card calibration. I have two options for this part of the process: 

1. I can use my Lenovo T500 laptop running Windows 7, which has a Line Out (headphones/stereo) with standard headphone jack and a line In (Mic) with standard headphone jack. Problem I am having here is that I cannot seem to alter anything with my soundcard in Windows 7. The sound card is a "Conexant 20561 Smart Audio HD and likely a pretty crappy card. I have found no way to alter variables with it like I use to be able to in XP. 

2. I can also use an external sound card, a Native Instruments Audio Kontrol 1. I use to use this for some DJ applications and hardware. It is a very good quality sound card with Mono XLR input, 1/4 balanced Mono input and 1/2 & 3/4 outputs, along with a headphone out put. 

My problem with both if these set ups is that I don't know how to apply the "loop back" set up to either one. For the laptop, there is only one input (mic) and output (headphones). It seems easy enough to loop those two together, with the mic coming in through a USB, but where in software or OS can I designate only Left or Right for the signal?

With the external card, can I take a stereo output signal from the Headphone out and loop that back into a Mono input jack (mic) or should I use one of the 1/2 outs and connect only one side into the Mono input jack? 

Th rest seems simple whereas this setup is confounding me


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