# Measure time delay (T/A) with aRTA and RoomEQ



## Hanatsu

Hi,

messed with this pretty much lately (feel free to correct me if I've done something wrong, I'm not superpro at this lol). I don't take full credit for this guide as there are several more knowledgeable people than me which have been helping me get it right. Thought I might as well share "my" method of measuring time differences between speakers. There seem to be a major interest how to do it, it's quite simple really once you get the hang of it.

Here's how to do it with both aRTA and RoomEQ (REW). It can be done with HOLMimpulse as well but it seems like it can be inaccurate under certain conditions. All these programs are free to use. 

First you need a soundcard with a Line-in and a speaker output. You need a loopback reference as described by the following picture, without it, it won't work.

_Place the microphone at listening position, in the middle of where the head is located. Measure one driver at a time. I measure outside the stopband of the crossovers, you should probably make a bandpassed measurement when measuring sub/mid (RoomEQ allows this). You'll get a better impulse with high frequency data._










First off aRTA, download it here: ARTA Download. Just run it in demo mode.

This how ARTA looks.










Click on; "Setup/Audio devices" for soundcard configuration. Select which channel the mic is connected to. Check the soundcard settings so I/O works as it should. Looks different on each computer, but I assume you know how to do this.










Now click on the measure tab. (The little red arrow symbol is a shortcut to the same menu). 










Now measure, make sure the "use the secondary channel as reference" box is marked. Choose preferred input channel to the microphone channel, i.e left in this case.










After the measurement is done it should like this.










Click "set as overlay"










Now mute the speaker you measured and turn on the speaker you want to T/A against. Measure again. It should look something like this now;










Zoom in if required. Turn on "View/Gate time"










Left click on the highest peak of the first impulse, right click on the other peak of the second peak, this will create a gate. You can check the delay and distance at the bottom of the screen.










We now know that one speaker need to be time delayed by 0.75ms (25,3cm distance). Set T/A accordingly, done!

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

Now RoomEQ. Download it here: REW - Room EQ Wizard Home Page. You need to register first, it's free to use though.

This is how it looks. Go into preferences first off.










Click the "use loopback for timing reference" box.










Then the soundcard tab. The timing boxes only showed up in ASIO mode on my stationary computer, dunno why really. Creative's soundcards have crappy drivers. Should look like this anyway. Choose the channels for reference and measurement in/outputs.










Now measure. Here you can choose to do a bandpassed measurement. I went with these settings (measured midbass to midrange drivers).










Repeat measurement for each driver with all other muted. You can observe all measurements under "overlays".










First click at the "Impulse" tab. Choose %Fs viewmode (thanks for the tip mojozoom). Now we can observe the time difference.










1,13-0,496 = 0,634ms

0,634ms = 21.7462cm (343m/s).

Here's a page to convert distance/time if you want. 

Time/distance calculator




















Note that measurements in aRTA / RoomEQ were NOT the same, therefore there's different distances in the examples. 

Btw, anyone know how to calculate the time difference directly within RoomEQ? Would be awesome...


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## Sonus

Thank you 

Will have a go at this when my dsp and amps have been installed.


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## masswork

Hanatsu,
have you try comparing the result between band passed excitation signal vs adjusting by looking at the knee insted of the peak? That's where the impulse start to peak... Point where it left 0v mark...


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## quality_sound

Excellent write-up. Is it just the numbers that are different or were the measurements totally different? Did they both give you the same delay calculations?


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## Hanatsu

quality_sound said:


> Excellent write-up. Is it just the numbers that are different or were the measurements totally different? Did they both give you the same delay calculations?


Measurements were completely different. The method is accurate though. Tried a speaker at 10cm distance, then moved it 0.5cm away and both programs picked it up easily.

Sent from my Samsung Galaxy 3 via Tapatalk


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## Hanatsu

masswork said:


> Hanatsu,
> have you try comparing the result between band passed excitation signal vs adjusting by looking at the knee insted of the peak? That's where the impulse start to peak... Point where it left 0v mark...


Eh not sure I follow. Compare what? 

Sent from my Samsung Galaxy 3 via Tapatalk.


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## JMichaels

Looks good. Will have to give it another read and a try.


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## bbfoto

Nice! Thank you!


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## nabman

Thanks for this, Hanatsu! This is very timely for me...

One problem I had was with the "use loopback for timing reference" switch. Since I use a USB mic, it was not clear how I can set a loopback and use it for a timing reference. My output is the PC soundcard but my input can either be the mic *or* the input of my soundcard. It seems that what I would need to make this work would be to have the program (REW, aRTA, HolmImpulse, etc) take one input channel (e.g. Left) from the mic and the other input channel (e.g. Right, in this example) from the soundcard. Then I could loop back the right channel and use that for the timing reference. But that doesn't seem possible ... 

Since I ran into the above constraint, I've tried an alternative approach as described in Advanced Crossover Design. The approach is not as direct as using the impulse measurement but it seemed to be the only reliable approach available given that I was limited to a usb mic. The basic idea is to measure identical sweeps using each driver separately and also one with both operating concurrently (with no delays). From the individual responses (both amplitude and phase needed) the sheet computes what the summed response would look like for any given delay between the 2 individual responses. All that is left is to choose the delay that makes the computed trace look identical to the one we measured with both drivers firing.

This works, but is more convoluted/involved and if possible I'd like to be able to do/check this using REW, aRTA, etc.

Which leads me to the question : Is there a way to have an appropriate timing reference that REW/aRTA can use for those of us that have usb mics?


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## Hanatsu

nabman said:


> Thanks for this, Hanatsu! This is very timely for me...
> 
> One problem I had was with the "use loopback for timing reference" switch. Since I use a USB mic, it was not clear how I can set a loopback and use it for a timing reference. My output is the PC soundcard but my input can either be the mic *or* the input of my soundcard. It seems that what I would need to make this work would be to have the program (REW, aRTA, HolmImpulse, etc) take one input channel (e.g. Left) from the mic and the other input channel (e.g. Right, in this example) from the soundcard. Then I could loop back the right channel and use that for the timing reference. But that doesn't seem possible ...
> 
> Since I ran into the above constraint, I've tried an alternative approach as described in Advanced Crossover Design. The approach is not as direct as using the impulse measurement but it seemed to be the only reliable approach available given that I was limited to a usb mic. The basic idea is to measure identical sweeps using each driver separately and also one with both operating concurrently (with no delays). From the individual responses (both amplitude and phase needed) the sheet computes what the summed response would look like for any given delay between the 2 individual responses. All that is left is to choose the delay that makes the computed trace look identical to the one we measured with both drivers firing.
> 
> This works, but is more convoluted/involved and if possible I'd like to be able to do/check this using REW, aRTA, etc.
> 
> Which leads me to the question : Is there a way to have an appropriate timing reference that REW/aRTA can use for those of us that have usb mics?


I'm not 100% certain, but I don't think you can use USB mics. HOLMimpulse have a function which can be used for time aligning without loopback, dunno exactly how reliable it is though.


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## quality_sound

From what Erin was saying, a mobile pre or transit would be needed. I picked up a pre but haven't tried it yet.


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## nabman

Hanatsu said:


> I'm not 100% certain, but I don't think you can use USB mics. HOLMimpulse have a function which can be used for time aligning without loopback, dunno exactly how reliable it is though.


I was afraid of this - I've searched but haven't found any thread yet that describes how to do this with an usb mic. 
At least the solution using ACD works - I guess the good news is that even if it's a bit more complex, it really needs to be done just once.
Thx!


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## Brian Steele

Hanatsu said:


> I'm not 100% certain, but I don't think you can use USB mics. HOLMimpulse have a function which can be used for time aligning without loopback, dunno exactly how reliable it is though.


Seems to be quite reliable, in that the results are consistent and repeatable. I just tried it tonight. 

The simple instructions:

1. Connect your measuring apparati (I use a USB mike and a USB soundcard) and start HolmImpulse
2. Under "Device and Signal", select the "Square Noise (Improved MLS) response (I seemed to get the best results with this). Also ensure that the "Keep in/out stream active" option is selected.
3. Under "Data Analysis", select the "Detect Time Zero" option. (I also selected the "First Positive Peak" option too).
4. Under "Measurements", perform your first measurement.
5. Go back to "Data Analysis", select the "Time zero locked (time alignment)" option, then select "Use".

You can now perform your additional measurements. You can view the distance offsets from the first measurement under the "Impulse Response" window.

Night's still early so I'm going to have some more fun with it


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## Dillyyo

quality_sound said:


> From what Erin was saying, a mobile pre or transit would be needed. I picked up a pre but haven't tried it yet.


"mobile pre" or "transit"? Can you elaborate on these? Also, I could have sworn NPdang had done impulse measurements with a USB mic. It would really suck if my EMM6 mic can't be used for setting TA via impulse.


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## Dillyyo

Hanatsu-

It seems from your diagram that if I had a laptop with one "mic in" jack and one "headphone out" jack, that I would need 2 dual jack splitters to give me a total of 4 jacks, correct? if so, aren't "mic in" jacks on laptops usually mono signal inputs?


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## Brian Steele

Dillyyo said:


> "mobile pre" or "transit"? Can you elaborate on these? Also, I could have sworn NPdang had done impulse measurements with a USB mic. It would really suck if my EMM6 mic can't be used for setting TA via impulse.


Use HolmImpulse as I outlined in a previous message. Works fine with USB mics.


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## Brian Steele

Brian Steele said:


> Night's still early so I'm going to have some more fun with it


I forgot to mention, HolmImpulse gives the delay directly in cm, so no need for any conversions involving the speed of sound. If the impulse for speaker A is 10 cm before speaker B, simply add 10 cm of delay to speaker A.

BTW, during my measurements tonight, I discovered that my front left midbass seemed to have a 5 dB "trough" between 200 Hz and 1kHz. Grr... looks like I'm going to be taking apart that door this weekend. Hopefully it's a simple issue with the x-over and not a problem with the driver.


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## Dillyyo

Brian Steele said:


> Seems to be quite reliable, in that the results are consistent and repeatable. I just tried it tonight.
> 
> The simple instructions:
> 
> 1. Connect your measuring apparati (I use a USB mike and a USB soundcard) and start HolmImpulse
> 2. Under "Device and Signal", select the "Square Noise (Improved MLS) response (I seemed to get the best results with this). Also ensure that the "Keep in/out stream active" option is selected.
> 3. Under "Data Analysis", select the "Detect Time Zero" option. (I also selected the "First Positive Peak" option too).
> 4. Under "Measurements", perform your first measurement.
> 5. Go back to "Data Analysis", select the "Time zero locked (time alignment)" option, then select "Use".
> 
> You can now perform your additional measurements. You can view the distance offsets from the first measurement under the "Impulse Response" window.
> 
> Night's still early so I'm going to have some more fun with it


Reproducibility might be robust, but nothing to clearly indicate accuracy is. Without a loop-back reference to measure the inherent latency between the various devices, it seems it would be difficult to have consistent and accurate results. I guess it would depend on whether that inherent latency had much variability to it or if all the measurements would be off by pretty much the same amount.


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## Dillyyo

Brian Steele said:


> Use HolmImpulse as I outlined in a previous message. Works fine with USB mics.


I just questioned that process in a post prior. How does it correct for the lack of latency referencing without any loop-back? Is there inherent variability within this latency that the loop-back referencing is supposed to adjust for?

Just set this up and did a test run in my office with my EMM-6 mic and using my laptop speakers as output device. My first initial measurement puts the first peak at 0 on the distance axis, but it does so for each subsequent measurement, even when I am moving the mic further away from the source. Am I missing a switch here or something?


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## Brian Steele

Dillyyo said:


> Reproducibility might be robust, but nothing to clearly indicate accuracy is. Without a loop-back reference to measure the inherent latency between the various devices, it seems it would be difficult to have consistent and accurate results. I guess it would depend on whether that inherent latency had much variability to it or if all the measurements would be off by pretty much the same amount.


I think that, using the process I gave, HolmImpulse uses the first measurement as a benchmark and basically assumes that everything from the point the measurement starts to the first positive pulse recorded is caused by system latency. Of course if that latency varies, then the measurements will be off, but (1) if the system latency is that variable, any process you use for measurement will be inaccurate, and (2) I would not have gotten consistend results if system latenct was that variable.


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## Dillyyo

Brian Steele said:


> I think that, using the process I gave, HolmImpulse uses the first measurement as a benchmark and basically assumes that everything from the point the measurement starts to the first positive pulse recorded is caused by system latency. Of course if that latency varies, then the measurements will be off, but (1)* if the system latency is that variable, any process you use for measurement will be inaccurate, and* (2) I would not have gotten consistend results if system latenct was that variable.


i don't know if I agree with the bold. Isn't that what the loop back function is for; to measure differences in signal latency? Why would variations affect anything if corrective measures are taken every time referencing is done? Maybe i'm under the wrong assumption as to what the loop back function actually does. I'm wondering if it is only used to determine when the signal leaves the sound card and then compare to when it is received in the mic input side of the card. If that is the case then I wonder how not using a loop back would determine actual time the signal left the card? Maybe a default standard based on trended latency data?

Now i'm confusing myself


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## Brian Steele

Dillyyo said:


> i don't know if I agree with the bold. Isn't that what the loop back function is for; to measure differences in signal latency? Why would variations affect anything if corrective measures are taken every time referencing is done? Maybe i'm under the wrong assumption as to what the loop back function actually does. I'm wondering if it is only used to determine when the signal leaves the sound card and then compare to when it is received in the mic input side of the card. If that is the case then I wonder how not using a loop back would determine actual time the signal left the card? Maybe a default standard based on trended latency data?
> 
> Now i'm confusing myself


Ok, I think I understand where you're coming from. Using the "loopback" for every measurement ensures that every measurement is corrected for system latency, no matter what it is. Makes sense.

I think HolmImpulse tries to address variations in system latency by keeping the in/out stream active (that's why that option needs to be selected), rather than starting it up for every measurement. If this does work, then measurement of latency derived from the first measurement can be used as a reference. I performed a number of measurements with it set that way tonight, and the results seemed to be pretty consistent. If they were not, then I'd have seen varying time offsets for different measurements of the same speaker.


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## Hanatsu

Dillyyo said:


> Hanatsu-
> 
> It seems from your diagram that if I had a laptop with one "mic in" jack and one "headphone out" jack, that I would need 2 dual jack splitters to give me a total of 4 jacks, correct? if so, aren't "mic in" jacks on laptops usually mono signal inputs?


Yes you would need 4 jacks in total. Mic input on my laptop is 2ch as far as I know. Be sure to set the input as 'Line-in' and not 'microphone'.

Sent from my Samsung Galaxy 3 via Tapatalk.


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## Dillyyo

Brian Steele said:


> Ok, I think I understand where you're coming from. Using the "loopback" for every measurement ensures that every measurement is corrected for system latency, no matter what it is. Makes sense.
> 
> *I think HolmImpulse tries to address variations in system latency by keeping the in/out stream active (that's why that option needs to be selected), rather than starting it up for every measurement.* If this does work, then measurement of latency derived from the first measurement can be used as a reference. I performed a number of measurements with it set that way tonight, and the results seemed to be pretty consistent. If they were not, then I'd have seen varying time offsets for different measurements of the same speaker.


I don't think I know enough about in/out streams and how they relate to a chip or program documenting accrued data, to support or argue against your proposed hypothesis.  With that said, your position seems logical, so I might have to give it a try. I just have experience with REW and that lessens the learning curve.

Edit: I just re-read your post and I understand where you are getting at. The real question then is, does taking an initial one time "standard" measurement and using it as a corrective factor for the duration of the session, give _perceptibly_ lesser results than using a newly obtained factor for each measurement taken?


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## Dillyyo

Hanatsu said:


> Yes you would need 4 jacks in total. Mic input on my laptop is 2ch as far as I know. Be sure to set the input as 'Line-in' and not 'microphone'.
> 
> Sent from my Samsung Galaxy 3 via Tapatalk.


Would you or anyone happen to know where in Windows 7 I can alter those variables? In XP it use to be easy to get to, but no matter where I go audio related on my T500 w/ Windows 7, I cannot see an option to setup channel inputs. 

I know Lenovo hardware crippled the ability to record streaming music, but I don't if that has any correlation with my issue of not being able to locate where I would assign channels.


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## Hanatsu

Dillyyo said:


> Would you or anyone happen to know where in Windows 7 I can alter those variables? In XP it use to be easy to get to, but no matter where I go audio related on my T500 w/ Windows 7, I cannot see an option to setup channel inputs.
> 
> I know Lenovo hardware crippled the ability to record streaming music, but I don't if that has any correlation with my issue of not being able to locate where I would assign channels.


You know which soundcard you're using? Some software have that option built-in. My laptop have a soundblaster x-fi soundcard, easy to change in creative's software.


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## nabman

Brian Steele said:


> Seems to be quite reliable, in that the results are consistent and repeatable. I just tried it tonight.
> 
> The simple instructions:
> 
> 1. Connect your measuring apparati (I use a USB mike and a USB soundcard) and start HolmImpulse
> 2. Under "Device and Signal", select the "Square Noise (Improved MLS) response (I seemed to get the best results with this). Also ensure that the "Keep in/out stream active" option is selected.
> 3. Under "Data Analysis", select the "Detect Time Zero" option. (I also selected the "First Positive Peak" option too).
> 4. Under "Measurements", perform your first measurement.
> 5. Go back to "Data Analysis", select the "Time zero locked (time alignment)" option, then select "Use".
> 
> You can now perform your additional measurements. You can view the distance offsets from the first measurement under the "Impulse Response" window.
> 
> Night's still early so I'm going to have some more fun with it


Actually I've tried this approach (with the exception of not using "Square Noise"). My experience has been that the results were consistent _within_ a single set of measurements when I kept the kept the stream active but less so _across_ different sessions - the same approach done at/on different times/machines yielded slightly different results. i.e. stopping everything and restarting HolmImpulse yielded differences as did the use of a different computer (I tried using a desktop machine I dragged into the garage as well as a notebook). The numbers were in the same ballpark but not the same (usually within 0.25msec) and made me conclude that the approach had some other confounding variability. I have no idea why though...

I'll try again with the "Square Noise" option.


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## Dillyyo

Hanatsu said:


> You know which soundcard you're using? Some software have that option built-in. My laptop have a soundblaster x-fi soundcard, easy to change in creative's software.


Conexant 20561 SmartAudio HD

I have an external USB soundcard too, but its bigger and I'd rather not lug it around nor go and buy cabling/jacks.


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## Dillyyo

nabman said:


> Actually I've tried this approach (with the exception of not using "Square Noise"). My experience has been that the results were consistent _within_ a single set of measurements when I kept the kept the stream active but less so _across_ different sessions - the same approach done at/on different times/machines yielded slightly different results. i.e. stopping everything and restarting HolmImpulse yielded differences as did the use of a different computer (I tried using a desktop machine I dragged into the garage as well as a notebook). The numbers were in the same ballpark but not the same (usually within 0.25msec) and made me conclude that the approach had some other confounding variability. I have no idea why though...
> 
> I'll try again with the "Square Noise" option.


Concerning...:worried:


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## Dillyyo

nabman said:


> Actually I've tried this approach (with the exception of not using "Square Noise"). My experience has been that the results were consistent _within_ a single set of measurements when I kept the kept the stream active but less so _across_ different sessions - the same approach done at/on different times/machines yielded slightly different results. i.e. stopping everything and restarting HolmImpulse yielded differences as did the use of a different computer (I tried using a desktop machine I dragged into the garage as well as a notebook). The numbers were in the same ballpark but not the same (usually within 0.25msec) and made me conclude that the approach had some other confounding variability. I have no idea why though...
> 
> I'll try again with the "Square Noise" option.


Disregarding the variability in testing group results, what were your perceptible results like? Maybe not dead on accurate, but as accurate as manually measuring would give?


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## Hanatsu

Dillyyo said:


> Conexant 20561 SmartAudio HD
> 
> I have an external USB soundcard too, but its bigger and I'd rather not lug it around nor go and buy cabling/jacks.


That soundcard using the builtin Realtek HD driver or is it using some other interface? If so, you should have the option. I can see if I find it


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## Brian Steele

nabman said:


> Actually I've tried this approach (with the exception of not using "Square Noise"). My experience has been that the results were consistent _within_ a single set of measurements when I kept the kept the stream active but less so _across_ different sessions - the same approach done at/on different times/machines yielded slightly different results. i.e. stopping everything and restarting HolmImpulse yielded differences as did the use of a different computer (I tried using a desktop machine I dragged into the garage as well as a notebook). The numbers were in the same ballpark but not the same (usually within 0.25msec) and made me conclude that the approach had some other confounding variability. I have no idea why though...
> 
> I'll try again with the "Square Noise" option.


Did you repeat step 5 every time to stopped the stream (e.g. switched PCs, restarted HolmImpulse, etc)?


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## Dillyyo

Hanatsu said:


> That soundcard using the builtin Realtek HD driver or is it using some other interface? If so, you should have the option. I can see if I find it


It is using the built in conexant sound card from what I see in the audio device management interface. If Realtek and Conexant are one and the same, then the answer is yes. I have looked everywhere and I see nothing. I have the option to use the Conexant or an SPDIF interface.


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## Hanatsu

Dillyyo said:


> It is using the built in conexant sound card from what I see in the audio device management interface. If Realtek and Conexant are one and the same, then the answer is yes. I have looked everywhere and I see nothing. I have the option to use the Conexant or an SPDIF interface.


I'll look into it tomorrow. See if I can find out if it's possible.


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## Brian Steele

nabman said:


> Actually I've tried this approach (with the exception of not using "Square Noise"). My experience has been that the results were consistent _within_ a single set of measurements when I kept the kept the stream active but less so _across_ different sessions - the same approach done at/on different times/machines yielded slightly different results. i.e. stopping everything and restarting HolmImpulse yielded differences as did the use of a different computer (I tried using a desktop machine I dragged into the garage as well as a notebook). The numbers were in the same ballpark but not the same (usually within 0.25msec) and made me conclude that the approach had some other confounding variability. I have no idea why though...
> 
> I'll try again with the "Square Noise" option.



I ran into another "variable" tonight. When using the procedure, don't have any other programs running on the PC. I was trying to put together an illustrated procedure by doing some screen captures of the various settings and measurements in HolmImpulse and couldn't understand why the reference point kept shifting, until I realised that it shifted every time I switched to the MS Paint program after taking a measurement to save the screenshot. Doh!

FWIW, when realised that and started taking measurements without taking the screenshots, I was able to duplicate the results I got yesterday.


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## 14642

So long as the latency doesn't change between measurements, then latency doesn't matter, since we're concerned about the RELATIVE delay between speakers, not the ABSOLUTE delay of any speaker. This means that the loopback measurement isn't necessary. It eliminates the latency from the displayed measurement. 

Making a loopback won't eliminate the error caused by additional latency caused by running other programs, etc.

There's one HUGE issue that no one is talking about here and it's really important. Picking the top of the peak is NOT the way to set delay, unless every speaker is a high-frequency speaker. The slope of the peak IS high frequency content. When we measure a tweeter, we can pick the peak because the peak is sharp and it's close enough to the picking the point at which the impulse starts. This is NOT the case with low frequency signals, where the slope is more gradual. The tip of the peak for a tweeter is NOT the origin of the sound and neither is top of the hump in the measurement of a low-passed signal. The beginning of the rise is the origin. If you're using the %FS in REW, for example, back up (leftward) on each measurement line to a point that corresponds to something like 5% and compare the measurements there.


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## Brian Steele

Andy Wehmeyer said:


> So long as the latency doesn't change between measurements, then latency doesn't matter, since we're concerned about the RELATIVE delay between speakers, not the ABSOLUTE delay of any speaker.


Good point.




Andy Wehmeyer said:


> There's one HUGE issue that no one is talking about here and it's really important. Picking the top of the peak is NOT the way to set delay, unless every speaker is a high-frequency speaker. The slope of the peak IS high frequency content. When we measure a tweeter, we can pick the peak because the peak is sharp and it's close enough to the picking the point at which the impulse starts. This is NOT the case with low frequency signals, where the slope is more gradual. The tip of the peak for a tweeter is NOT the origin of the sound and neither is top of the hump in the measurement of a low-passed signal. The beginning of the rise is the origin. If you're using the %FS in REW, for example, back up (leftward) on each measurement line to a point that corresponds to something like 5% and compare the measurements there.


Great point. My main speakers are basically "full range" (more precisely, coax), so this was a non-issue for me.  However this will be an issue when doing TA between active component speakers, or main speakers and subwoofers.


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## Brian Steele

Brian Steele said:


> FWIW, when realised that and started taking measurements without taking the screenshots, I was able to duplicate the results I got yesterday.


I've put up a draft of the process (time alignment with HornResp) here. It includes a number of additional details that I reminded of when repeating the procedure today.

Time Alignment using HolmImpulse


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## Hanatsu

Andy Wehmeyer said:


> So long as the latency doesn't change between measurements, then latency doesn't matter, since we're concerned about the RELATIVE delay between speakers, not the ABSOLUTE delay of any speaker. This means that the loopback measurement isn't necessary. It eliminates the latency from the displayed measurement.
> 
> Making a loopback won't eliminate the error caused by additional latency caused by running other programs, etc.
> 
> There's one HUGE issue that no one is talking about here and it's really important. Picking the top of the peak is NOT the way to set delay, unless every speaker is a high-frequency speaker. The slope of the peak IS high frequency content. When we measure a tweeter, we can pick the peak because the peak is sharp and it's close enough to the picking the point at which the impulse starts. This is NOT the case with low frequency signals, where the slope is more gradual. The tip of the peak for a tweeter is NOT the origin of the sound and neither is top of the hump in the measurement of a low-passed signal. The beginning of the rise is the origin. If you're using the %FS in REW, for example, back up (leftward) on each measurement line to a point that corresponds to something like 5% and compare the measurements there.


Thanks for the explanation, think I get it now. I'll try this instead. 

You should perhaps add this quote in the OP. I can't edit after 4 weeks.


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## Dillyyo

Andy Wehmeyer said:


> So long as the latency doesn't change between measurements, then latency doesn't matter, since we're concerned about the RELATIVE delay between speakers, not the ABSOLUTE delay of any speaker. This means that the loopback measurement isn't necessary. It eliminates the latency from the displayed measurement.
> 
> Making a loopback won't eliminate the error caused by additional latency caused by running other programs, etc.
> 
> There's one HUGE issue that no one is talking about here and it's really important. Picking the top of the peak is NOT the way to set delay, unless every speaker is a high-frequency speaker. The slope of the peak IS high frequency content. When we measure a tweeter, we can pick the peak because the peak is sharp and it's close enough to the picking the point at which the impulse starts. This is NOT the case with low frequency signals, where the slope is more gradual. The tip of the peak for a tweeter is NOT the origin of the sound and neither is top of the hump in the measurement of a low-passed signal. The beginning of the rise is the origin. If you're using the %FS in REW, for example, back up (leftward) on each measurement line to a point that corresponds to something like 5% and compare the measurements there.


So, you mean I should take the point at which the peak is at 5% rise off of baseline? Also, are you saying REW can be used for impulse measurements without the need for a loop back?


----------



## nabman

Dillyyo said:


> Disregarding the variability in testing group results, what were your perceptible results like? Maybe not dead on accurate, but as accurate as manually measuring would give?


Tough to say - I never used a tape measure since it is very difficult for me to get a proper measurement to my sub. The measurements were in the same ballpark across measurements - the max deviation I got was 0.25msec (which is 3"?).



Brian Steele said:


> Did you repeat step 5 every time to stopped the stream (e.g. switched PCs, restarted HolmImpulse, etc)?


Yes - I was careful with this one!  Every time I started/restarted HolmImpulse or wished to do a fresh new set of measurements, I'd unlock, do the first measurement and lock the zero based on that one.



Andy Wehmeyer said:


> So long as the latency doesn't change between measurements, then latency doesn't matter, since we're concerned about the RELATIVE delay between speakers, not the ABSOLUTE delay of any speaker. This means that the loopback measurement isn't necessary. It eliminates the latency from the displayed measurement.
> 
> Making a loopback won't eliminate the error caused by additional latency caused by running other programs, etc.
> 
> There's one HUGE issue that no one is talking about here and it's really important. Picking the top of the peak is NOT the way to set delay, unless every speaker is a high-frequency speaker. The slope of the peak IS high frequency content. When we measure a tweeter, we can pick the peak because the peak is sharp and it's close enough to the picking the point at which the impulse starts. This is NOT the case with low frequency signals, where the slope is more gradual. The tip of the peak for a tweeter is NOT the origin of the sound and neither is top of the hump in the measurement of a low-passed signal. The beginning of the rise is the origin. If you're using the %FS in REW, for example, back up (leftward) on each measurement line to a point that corresponds to something like 5% and compare the measurements there.


Interesting ... could the load that background programs have on the system/cpu etc., impact the relative measurements? e.g. if the load is different when the woofer is being measured relative to when the mid is being measured? Maybe this is why my weaker notebook had different results from my desktop? i.e. the notebook's cpu etc were more easily overloaded?

Yes, I agree re your second point - using the peak doesn't quite work universally when you have different drivers. The width of the waveforms are also not the same so picking the max point (even assuming you got it right) may not represent the correct delay. I tried to compensate for this by doing exactly what you suggest here - I used the *beginning* of the first rise (or "fall" depending on your polarity) as the "starting" or "zero" point. This choice made a bigger difference when setting the sub relative to the woofer than when setting the tweets relative to the mids as expected.

In my case I have a 3-way+sub setup in my car (tweeter/mid/woofer/sub).
I used the average of the last two measurements I made (notebook and desktop) to set the delays but I felt that the uncertainty in each measurement could contribute to much larger "errors" in the entire setup. 
This is what prompted me to look at the alternate approach I mentioned. There was more work (and learning) involved *after* making the measurements (I used REW to make my measurements this time). The measurements need to be dumped into a .frd file (with phase info included), then converted (not needed but preferable so I did it) to minimum phase and then imported into the ACD sheets.

I took measurements across every pair of "adjacent" drivers (sub/left-Woof, sub/right-Woof, left-Woof/right-Woof, left-Woof/right-mid, right-Woof/left-mid, left-mid/right-mid, left-mid/right-tweter, right-mid/left-tweeter, left-tweeter/right-tweeter) and for each pair, I used a sine-sweep over a range that both drivers could handle. *Exactly* the same range was used with each pair. For example, I would send a 80-1khz sweep to my left woof, the right woof and then both woofs. This would yield 3 measurements from which the relative delay between the left and right woofs could be determined. For the woof/mid measurements, I used 200Hz-2kHz, for the mid/mid I used 400Hz-8kHz, etc.
For the most part the delays I computed using this data were consistent. I think I screwed up on my right-tweeter/left-mid measurements since they seem to be *slightly* different from what the other measurements imply for my delays. The difference was small - as I type this I forget exactly what it was, but iirc the direct measurement was lower than what was implied by the others by about 0.2msec. All other readings implied delays for the individual drivers that differed by less than 0.05 msec. I have a Minidsp board so can theoretically set delays to 0.01msec accuracy. 
The new measurements were different from the ones I had from using the impulse method with my usb mic (Dayton EMM-6 with MSL USB Mic Mate) for the sub and tweets. The woof/mid delays were similar (but not the same). My ability to localize instruments within the sound-stage feels a lot better also though that could be psychological!

But ... I don't want to hijack this thread with a different approach! I plan to do another careful round of measurements now that I better understand the method. However, due to work related travel I won't be able to get to it till early May - I'll try and post a walk-through and my results in a separate thread if no one else has already done so by then.


----------



## nabman

Brian Steele said:


> I've put up a draft of the process (time alignment with HornResp) here. It includes a number of additional details that I reminded of when repeating the procedure today.
> 
> Time Alignment using HolmImpulse


Thanks Brian - this is very clearly outlined. I will try using the "Square Noise" option next time - that may have been the weakness in my earlier attempts.


----------



## fcarpio

The part I do not understand is if you do this in pairs like align tweeters, then mids, then midbasses, etc. with respect to each pair only in isolation from the group. Is that assumption correct? Or do we at some point need to align a tweeter with a mid, and/or a mid with a midbass for a tighter overall time alignment?


----------



## Hanatsu

All drivers should be T/A to each other. In which order you do it doesn't matter. I like to T/A each mid, each tweeter etc to eachother first then move both sides with an equal amount to another set of drivers until the align properly...

Tapaaatalk!!


----------



## eddieg

Hanatsu, Andy

Still have some reading to do here but just wanted to say thanks!

I tried to figure it out on the rew where the time measurements would be but never got back to it and on arta I got some redicilous results, they did work but I must have multiplied something somewhere. 

You guys sure saved me a lot of time!

By the way I noticed you did not enter any calibration files, I guess that for time response it does not really matter.

I remember that I created a calibration file that has the loopback response in it so that is what I am using as calibration file when tried to test for the impulse. Need to try it again sometime.


----------



## 14642

you don't need a loopback because all you need to find is relative delay. The delay through the soundcard is a constant. (A+X)-(B+X)=A-B

Oops. Edited. I already wrote this.


there's one other thing to consider because all of this focus on accuracy to the tiniest increment isn't necessary. You can only delay the signal by one sample. So...if your DSP samples at 48k (most do), then the adjustment increments are .28 inches. Once the destructive interference caused by the phase error is pushed out of the passband of the speakers you're aligning, then the error doesn't really matter in terms of altering frequency response. For imaging, getting the midbass and midrange right is important. Backing up from the peak by 12dB is absolutely sufficient to determine the approximate location of low passed speakers. For really high frequencies, we don't hear phase very well, so errors in tweeter alignment are audible as frequency response aberrations, which can be "fixed" well enough with EQ.


----------



## Hanatsu

You need loopback if the software reset IR at 0ms for each measurement. HolmImpulse can lock the relative delay through the soundcard but I'm unsure if REW/aRTA can do it.

Tapaaatalk!!


----------



## scoobyman

This is a very good topic and a great way to measure TA down to a ms, worked like a charm, thanks for that.

However, i would like to emphasize on a point that EQ and general db output of a speaker can have a much greater influence on perceived sound than TA.

As Andy has mentioned, when it comes to high frequencies and some of the low range spectrum, we pick up sound location from whatever source is the loudest, so if you EQ your tweets or midbass incorrectly, you WILL end up with a wandering sound stage even if your TA is spot on.

If someone could provide more info on how to combat this issue by other means than avoiding using separate channel EQ'ing in these frequency ranges, i would be really grateful as i'm struggling with this atm.


----------



## sqnut

scoobyman said:


> If someone could provide more info on how to combat this issue by other means than avoiding using separate channel EQ'ing in these frequency ranges, i would be really grateful as i'm struggling with this atm.


The best way to tackle L/R balance is to have an eq per driver. At the very minimum you need an eq that is L/R independent.


----------



## Hanatsu

sqnut said:


> The best way to tackle L/R balance is to have an eq per driver. At the very minimum you need an eq that is L/R independent.


Yep ^^


----------



## scoobyman

sqnut said:


> The best way to tackle L/R balance is to have an eq per driver. At the very minimum you need an eq that is L/R independent.


Well here is the thing - when i started with level adjusted speakers and no EQ'ing and time aligned them correctly as shown in this guide, i ended up with a perfect centre image for every channel, then i went ahead to RTA the system and adjusted the EQ accordingly on every channel and now my perfect centre image is gone. How do I deal with that ?


----------



## Hanatsu

How did you measure FR?

Verify that the EQ is centered by listening to these tracks. Lower frequencies like 200-400Hz should originate from the same spot as 4000-8000Hz for example. 

https://www.dropbox.com/s/zoc2vgb4e44ys2a/Disc%201%20%28Tuning%29.rar

https://www.dropbox.com/s/mi6lx03nqpp7lxy/Disc%202%20%28Testing%20%26%20Verification%29.rar


----------



## scoobyman

> How did you measure FR?
> 
> Verify that the EQ is centered by listening to these tracks. Lower frequencies like 200-400Hz should originate from the same spot as 4000-8000Hz for example.
> 
> https://www.dropbox.com/s/zoc2vgb4e4...8Tuning).rar
> 
> https://www.dropbox.com/s/mi6lx03nqp...ication).rar


I use a laptop with external sound card and a good mic running spectraRTA software.
I use the 7 drums track to check for centre image

https://www.dropbox.com/s/e0qql0pulk0cqpw/7 drums.flac

Anyhow, after EQing seprate sides of my system that had a perfect centre image, it has shifted both in tweeter range and in midbass range, midrange was unaffected and i was still able to get a good centre image on midrange drivers despite heavy EQing.

Is my EQing technique flawed ? From what i can only assume is the reflection problem, midbass area requires pretty drastic EQ changes in different areas on different drivers, i wont go into detail as this topic is not the place for it, but I have read plenty of threads about RTA tuning to make sure i do everything correctly, but my L&R speaker response differences are pretty drastic without any EQing even though they are level matched and have a very similar response when measured on-axis.


----------



## eddieg

Well my question was not really if I need a loopback - I've mentioned the loopback as a part of my calibration file. 

The question was if really there is a need when measuring impulse response for a calibration file at all - but I guess the answer is no as all we are measuring is relative time so it does not matter if the MIC and PC are calibrated correctly. 

Thanks!

Eddie


----------



## Hanatsu

Don't understand that part. Which calibration file? For the soundcard? How can the loopback 'be part of the calibration file'?

Differences of 10dB is common between L / R midbass drivers even if they measure the same in nearfield. It's the modal region of the car, where the speakers are mounted in relation to the listening space affects the response.

So how do you measure exactly? Are you in the car? Averaging? Sweeps or noise? One driver at a time or?

Tapaaatalk!!


----------



## eddieg

There is a way to create a calibration file that takes in acount the loopback and the MIC calibration file - I did it a long time ago and I have both files to choose from. 

The measurments I am doing is when the MIC is inside the car placed where my head is usally at when I'm driving. 

As for the Time delay - of course I measure one speaker at a time and then compare the overlay. 

Thanks

Eddie


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## 14642

Noi, the mic is also a constant. If your mic rolled off at 1k, then it might not be appropriate for measuring impulses from tweeters, but it is probably not the case.


----------



## fcarpio

Subbed, will try this later on today.


----------



## Kevin K

fcarpio, how did your TA session go? 
Which route did you use?


----------



## Alextaastrup

I used TA software for quick calculation of time delay from Acoustic Power Lab. Amazing 3D pictures! TA soft can be downloaded free (demo). Takes very short time to optimize time delay for the whole system. Measured only with PC, external sound card and mic with 48V. Took less than halv an hour to adjust passive front and sub.Twitters and midbasses (in the passive setup) were automatically TA aligned with the help of another sofware from the same manufacturer. It is important to EQ each speaker before time alignment. My tweeters were rather different in sound before EQ correction, therfore it was not possible to place scene just an a center. They should play equal for optimal scene.


----------



## Kevin K

tell us more about the TA software.


----------



## Alextaastrup

Kevin,
TA software is rather complicated and demands Mathlab installation first.
But then everything will be done quickly and easy. I placed the mic just between two front seats in order to make scene both for driver and front passanger. Sub plays now just in front of me (despite actual placement in the back of the car (sedan). Sound from it comes through two holes in a shelf.
Please find this information from the manufacturer site.
By the way - with some impressing results (2 golds in 2014, Eurofinals in Austria and best sound - in 2013):
Acoustic Power Lab :: Home
From this site:
TDA is a first of its kind speaker analyzing software that gives you access to speaker and PA-system parameters that were unobtainable in the past. Measuring a speaker is blazingly fast with perfect accuracy. Thanks to a proprietary short sweep technology and a new time domain analysis algorithms, it takes just a matter of seconds. After this the program automatically calculates accurate AFR, DFR, IR and TDA visualizations for use in speaker and PA-system calibration from small studios up to major concert venues. Yes, you heard right. No more measuring in the lab needed to know your gear in and out.

If you are a touring sound engineer, TDA will make it a breeze to recalibrate PA on the set as it is possible to find damaged speakers or elements quickly. It also allows fine tuning of crossover frequencies with accuracy impossible with competing products. This results to greater sonic fidelity and higher max SPL. Logarithmic IR analyzing is also possible giving invaluable information about the quality of speaker elements.
APL TDA offers these groundbreaking features:
•AFR - Amplitude Frequency Response
•DFR – Delay Frequency Response
•IR, both linear and logarithmic
•Delay visualization in 2D and 3D
•Precise parameter viewing in mouse location
•Easy to use analyzing tools
•Measurement window up to 300ms
•Accurate down to 1/96ms


For best result on any given speaker or PA setup:
1.Tune-up crossover system – delays, phases, levels (by use of APL TDA as measurement tool)
2.Measure SPFR and apply correction (APL WORKSHOP and APL1 equalizer)
3.Measure residual DFR (delay frequency response) by use of APL TDA
4.Do DFR correction based TDA delay measurement (DFR correction synthesis and APL1 as equalizer with SPFR and DFR corrections merged)


----------



## Alextaastrup

Little bit more info could be found here:

https://www.facebook.com/aplaudio


----------



## subterFUSE

Sub'd


----------



## subterFUSE

Yesterday I finally had a chance to try measuring the time alignment with REW.

I'm just not sure if I am doing it correctly?

First, I checked all my speaker polarity by using a polarity pulse signal and the polarity checker in AudioTools on my iPhone. I checked all my wiring at the amplifier to get the polarity pulses in sync. Had to flip a few wires, but got everything testing positive.

Next I started measuring my midbass. They are currently crossed over from 70Hz to 800Hz @ 24 dB slope. Here's the impulse overlay I got:

microphone in driver seat, at head level.










I chose to compare the peak for Left Mid @ 236.97ms to the right mid. I think the corresponding peak for the right midbass was at 238.282ms. Does that look correct?


Then I did my horns:











Looking at the horns, I am thinking it might be possible one of them is out of polarity. It's hard for me to tell, though.

Anyway, when I looked at my horns I used the following times:

Left Horn = 237.285ms
Right Horn = 237.396ms


I think the physical distance from the driver's listening position to the left and right horns is almost equal, so the time difference would be expected to be minimal.


Anyway, thanks for any help.


----------



## fcarpio

I got similar results and I gave up on this method since tuning T/A by ear works pretty good for me. I hope you get a response that helps sort things out as I would like to try again, the right way.


----------



## fcarpio

subterFUSE said:


> Yesterday I finally had a chance to try measuring the time alignment with REW.
> 
> I'm just not sure if I am doing it correctly?
> 
> First, I checked all my speaker polarity by using a polarity pulse signal and the polarity checker in AudioTools on my iPhone. I checked all my wiring at the amplifier to get the polarity pulses in sync. Had to flip a few wires, but got everything testing positive.
> 
> Next I started measuring my midbass. They are currently crossed over from 70Hz to 800Hz @ 24 dB slope. Here's the impulse overlay I got:
> 
> microphone in driver seat, at head level.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> I chose to compare the peak for Left Mid @ 236.97ms to the right mid. I think the corresponding peak for the right midbass was at 238.282ms. Does that look correct?
> 
> 
> Then I did my horns:
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Looking at the horns, I am thinking it might be possible one of them is out of polarity. It's hard for me to tell, though.
> 
> Anyway, when I looked at my horns I used the following times:
> 
> Left Horn = 237.285ms
> Right Horn = 237.396ms
> 
> 
> I think the physical distance from the driver's listening position to the left and right horns is almost equal, so the time difference would be expected to be minimal.
> 
> 
> Anyway, thanks for any help.


I just noticed your delays and they seem like a lot. The highest delay I have is 1.81ms and I have a large pickup truck.


----------



## subterFUSE

fcarpio said:


> I just noticed your delays and they seem like a lot. The highest delay I have is 1.81ms and I have a large pickup truck.


The numbers I posted are not my delays. They are the x-axis coordinates of the peaks I was using to compare the waveforms.

Therefore, with my measurements my delays would be:

Right MB = 0
Left MB = 1.312ms
Right Horn = 0.886ms
Left Horn = 0.997ms


Oh, and one last thing to mention. My measurements were taken at 97 degrees F. I had a small thermometer in the car. I'll need to convert to 74 F for normal temperature.


----------



## SPLEclipse

My suggestions are below. Yellow lines indicate where I would try to line them up first. The horns are more difficult because the response looks to be different on each one. Somewhere in that yellow box would be a good starting point. You might get better results using a limited frequency sweep, maybe only up to 2khz or so to eliminate some of the high frequency "ripples" that make it hard to see clear peaks.


----------



## subterFUSE

SPLEclipse said:


> My suggestions are below. Yellow lines indicate where I would try to line them up first. The horns are more difficult because the response looks to be different on each one. Somewhere in that yellow box would be a good starting point. You might get better results using a limited frequency sweep, maybe only up to 2khz or so to eliminate some of the high frequency "ripples" that make it hard to see clear peaks.


Thanks for the reply.

That's very possible as I have not yet applied any EQ to my channels. Was trying to do the time alignment first. Tomorrow I will try to EQ the channels so they match a bit better, and then try again with the TA.


----------



## subterFUSE

I was giving this TA measurement another go using REW.

Microphone position is center of driver's head, approx ear level.

I used a single sweep from 2000-2100 Hz for all 4 drivers.
(2 horns, 2 midbass)

I got the following measurement times for the peaks:

Left HLCD = 236.978ms
Right HLCD = 237.812ms

Left Midbass = 238.083ms
Right Midbass = 238.999ms


Here are pics of the measurements:

Horns (Right horn is Red)









Midbass (Left midbass is Red)












So I have a couple of questions:

1. Why are the midbass registering a longer time than the horns? They are obviously located closer to the microphone than the horns are, so I would expect them to register a shorter time than the horns. Midbass are in the doors, while horns are way up under the dash. The horns are a good 12" forward under the dash.

Perhaps it has to do with the freq. response of the mids vs. the horns?

Are the impulse response measurements only good for comparing "like" drivers?
i.e. midbass can only be compared to midbass, and horns to horns?


2. All of the peaks in my measurements were in the top part of the Y-axis. I'm assuming this means correct phase? But the Right side horn peak was in the bottom of the Y-axis. Does that mean that driver is out of phase?

I tested all of my drivers for correct polarity. Did it with a polarity pulse checker, and also did measurements with REW. (Impulse measurements with the microphone placed right up in front of each driver, and no loopback)
Should I be concerned with the phase in the impulse response measurements? Or is that a false indicator?


----------



## subterFUSE

Following up on this thread after spending a fair amount of time on the REW forums learning more about measuring delays via Impulse Response.

Here are a few things I learned:

1. When measuring for TA via impulse response, it's important to only use 1 measurement sweep. I had my REW software performing 2 sweeps by default and this was causing some strange fluctuations in the IR readings.

2. You want to align the initial rise of the first peak in the IR, not the actual IR peak (i.e. the 100% mark). This was really confusing me, as you can tell in the post above. It had me thinking the midbass drivers were reading further away than my horns when that wasn't the case. Now that I'm aligning the initial rise, the timings are better and the passenger side horn is registering as farthest away as expected. Here is a pic of my latest IR alignment.










3. You can hold down CTRL and Right-Click then Drag to get the difference in time and distance calculated for you on screen.

4. You can use the Impulse tab Controls menu to add or subtract delay to the IR charts to visually align them and help confirm the correct amount of delay needed. This is kind of similar to using the manual EQ filters to match a house curve. You can test it out before applying it to your DSP.


----------



## Hanatsu

Good post! Wasn't aware of the ctrl+right click feature. Thanks for the tip.

A steep slope of the IR peak is basically high frequency data. To get a proper anylysis we need to align the actual rise of the slope to eachother, not the peak as you said. The T/A measurement is indeed only valid for one point in space, so no averaging. You can also limit the frequency range to the crossover region for a better reading.


----------



## squiers007

Brian Steele said:


> I've put up a draft of the process (time alignment with HornResp) here. It includes a number of additional details that I reminded of when repeating the procedure today.
> 
> Time Alignment using HolmImpulse


I know this post is somewhat older, but had a couple questions regarding this method. I went to try it out today and got everything setup alright, but when i went to take my first measurement (which happened to be my right side mid) it would play the noise, but my mic would only register it as -60db. I even cranked it up to 90% max volume and only ever hit -58db even though it was super loud coming out of the speaker. 

I was using my cheap Dayton imm-6 mic hooked up to the mic input on my laptop, then using the headphone out to the aux on my HU. All volumes for input and output on my laptop where set to 100%. Anyone have any ideas? I didn't think you needed an expensive mic for this test but maybe I'm wrong. 

Also thinking i may just try REW too to see if my mic or laptop are just being dumb...

Sent from my Galaxy S5 using Tapatalk


----------



## Hanatsu

I'm sure there's a driver/soundcard issue causing that. Holm can override windows settings, make sure all inputs/levels are ok in the software.


----------



## squiers007

Hanatsu said:


> I'm sure there's a driver/soundcard issue causing that. Holm can override windows settings, make sure all inputs/levels are ok in the software.


I set everything up so it matched your screen shots. I'm not very well versed in computer stuff. Anything in particular I'm looking for to know it's setup correctly?

Sent from my Galaxy S5 using Tapatalk


----------



## fcarpio

I have a usb mic and a notebook, how would I go about making the loopback reference connections?


----------



## subterFUSE

fcarpio said:


> I have a usb mic and a notebook, how would I go about making the loopback reference connections?


You can't do loopback with a USB mic. You need a regular mic with XLR cable connection to a soundcard with at least 2 inputs. One input for the mic, and the other input will be the loopback connection.

Dayton Emm-6 is what I have currently. Got it from Cross Spectrum Labs, for a better calibration.

I also have a UMIK-1 which is USB. I use that for freq. response measurements. But for time alignment, you can't get the proper impulse response with loopback as reference with a USB mic. Sucks.


----------



## sqnut

Set to measured distance, tweak by ear, done. It's not rocket science


----------



## fcarpio

sqnut said:


> Set to measured distance, tweak by ear, done. It's not rocket science


I know how to do that, just wanted to validate my results.


----------



## subterFUSE

fcarpio said:


> I know how to do that, just wanted to validate my results.


And it's kinda fun, and helps to learn even more about the subject.


----------



## sqnut

fcarpio said:


> I know how to do that, just wanted to validate my results.





subterFUSE said:


> And it's kinda fun, and helps to learn even more about the subject.


 I understand, but seriously it's a 10 mts job and at the end of the day your ears will tell you if the timing is right or not. Just listen for the right cues.


----------



## Jazz80

If I using M audio Transit for the sound card which is there’s only 1 input and 1 output. How would I go making loop back?


----------



## Hanatsu

1 input and 1 output is enough. Each I/O should be 2ch, you need to separate each channel.


----------



## Jazz80

Hanatsu said:


> 1 input and 1 output is enough. Each I/O should be 2ch, you need to separate each channel.


Ok Hanatsu, I will try. Thank you


----------



## subterFUSE

Jazz80 said:


> If I using M audio Transit for the sound card which is there’s only 1 input and 1 output. How would I go making loop back?


I'm not sure that the M-Audio Transit can do a loopback measurement with Room EQ Wizard. It does not have separate Left/Right inputs.

It also does not have any phantom power for an XLR microphone.


I was previously using the M-Audio M Track. That's a much better option for use with Room EQ Wizard. It has separate inputs/outputs and phantom power for using a condenser mic.


----------



## SPLEclipse

Inputs/outputs that use a single 3.5mm connection can still be "broken out" into L/R if they are, indeed, stereo connections.

On a standard 3.5mm cable there will be three visible bands on the male connector. The on at the very tip is for left channel "+", the middle one is for right channel "+", and the one closest to the cord is the shared ground. This type of connector is referred to as TRS (tip, ring sleeve). You can google TRS pin-out or something of that nature to see how it all works if you'd like to construct some yourself, or buy L/R breakout cables. Here's a link to what they look like (obviously you don't need 10 feet, this is just for reference and ease of finding product):

Amazon.com: Hosa Cable CMR210 Stereo 1/8 Inch to Dual RCA Adapter Cable - 10 Foot: Musical Instruments

I use the left channel of both the inputs and outputs for their intended use, and the right channel of both the inputs and outputs bridged to form the loop-back. If you are using breakout cables similar to the ones I listed above, you would need a F-F RCA coupler like this:

Amazon.com: Hosa GRA101 RCA Female to RCA Female Coupler: Musical Instruments

You would simply plug each right channel RCA into the coupler to form the loop, and use the left cables for their intended output/mic input.


----------



## dannyboy100

Hanatsu said:


> Hi,
> 
> messed with this pretty much lately (feel free to correct me if I've done something wrong, I'm not superpro at this lol). I don't take full credit for this guide as there are several more knowledgeable people than me which have been helping me get it right. Thought I might as well share "my" method of measuring time differences between speakers. There seem to be a major interest how to do it, it's quite simple really once you get the hang of it.
> 
> Here's how to do it with both aRTA and RoomEQ (REW). It can be done with HOLMimpulse as well but it seems like it can be inaccurate under certain conditions. All these programs are free to use.
> 
> First you need a soundcard with a Line-in and a speaker output. You need a loopback reference as described by the following picture, without it, it won't work.
> 
> _Place the microphone at listening position, in the middle of where the head is located. Measure one driver at a time. I measure outside the stopband of the crossovers, you should probably make a bandpassed measurement when measuring sub/mid (RoomEQ allows this). You'll get a better impulse with high frequency data._
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> I just got my 3 way active system finished up (tweeter, mid, midbass and sub) and I'm getting ready to tackle this. Great write up btw!
> 
> So I think I have everything I need. I just had a question regarding the loopback.
> 
> I'm using an ecm8000 into an mbox 2 to the laptop via usb.
> the mbox has 2 channel inputs. Channel 1 being left and channel 2 right. It also has a monitor output left and right.
> So if i understood the diagram correctly i'd need to connect it as follows:
> 
> Mic into input 1 via XLR
> A TRS cable into input 2 Line input to right monitor out.
> and then the left monitor out to my headunits aux input.
> 
> Sound about right?


----------



## subterFUSE

dannyboy100 said:


> Hanatsu said:
> 
> 
> 
> Hi,
> 
> 
> 
> and then the left monitor out to my headunits aux input.
> 
> 
> 
> Sound about right?
> 
> 
> 
> 
> You also need a Y cable so that the 1 output can be split to feed the left and right inputs on the AUX
> 
> 
> Sent from my iPhone using Tapatalk
Click to expand...


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## dannyboy100

subterFUSE said:


> dannyboy100 said:
> 
> 
> 
> You also need a Y cable so that the 1 output can be split to feed the left and right inputs on the AUX
> 
> 
> Sent from my iPhone using Tapatalk
> 
> 
> 
> 
> Yeah, I tested the connections out last night on my home receiver and they worked. But like you point out the sound was only coming out of speaker.
> 
> The aux on my head unit is a 3.5mm input no left and right. So I have to figure out how to get both channels into the aux.
Click to expand...


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## subterFUSE

dannyboy100 said:


> subterFUSE said:
> 
> 
> 
> The aux on my head unit is a 3.5mm input no left and right. So I have to figure out how to get both channels into the aux.
> 
> 
> 
> Adapters.
> 
> Get a 3.5mm to RCA adapter, and then use a Y adapter after the sound card.
Click to expand...


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## dannyboy100

So I've got the loopback working and I'm getting ready to set the time alignment in REW before doing the levels,xovers and eq (following Hanatsu's other thread).

I have a fully active 3way plus sub. 
Using the sub as a reference since it's farthest away I plan on:

Measuring the sub and the passenger midbass, aligning the midbass
Measure the passenger midbass and the driver midbass, align the driver midbass
Measure the driver midbass and the passenger mid, align the passenger mid.
Measure the passenger mid and driver mid, align the driver mid.
Measure the driver mid and the passenger tweeter, align the passenger tweeter
and then finally passenger tweeter and driver tweeter, align the driver tweeter.

Does that cover it? Or am I missing something?


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## sqnut

1. Measure distance to all drivers. Left ear to left side drivers and right to right. *Measure the shortest distance to each cone.*

2. Calculate delay on each driver relative to the one that is furthest. So let's say sub is at 72", left mid bass is 45", right mid is 57" etc. Delay on sub = 0. Delay on left mid bass = (72-45)/13.54 = 2 ms. Delay on right mid = (72-57)/13.54 = 1.10 ms and so on. 

3. Once you measure and input delays, make sure all drivers are in normal polarity.


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## dannyboy100

sqnut said:


> 1. Measure distance to all drivers. Left ear to left side drivers and right to right.


LOL. I did that already. Just trying to mess around with REW now! 

I'm interested to compare my "by ear" setup to the REW setup!


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## 14642

subterFUSE said:


> Following up on this thread after spending a fair amount of time on the REW forums learning more about measuring delays via Impulse Response.
> 
> Here are a few things I learned:
> 
> 1. When measuring for TA via impulse response, it's important to only use 1 measurement sweep. I had my REW software performing 2 sweeps by default and this was causing some strange fluctuations in the IR readings.
> 
> 2. You want to align the initial rise of the first peak in the IR, not the actual IR peak (i.e. the 100% mark). This was really confusing me, as you can tell in the post above. It had me thinking the midbass drivers were reading further away than my horns when that wasn't the case. Now that I'm aligning the initial rise, the timings are better and the passenger side horn is registering as farthest away as expected. Here is a pic of my latest IR alignment.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 3. You can hold down CTRL and Right-Click then Drag to get the difference in time and distance calculated for you on screen.
> 
> 4. You can use the Impulse tab Controls menu to add or subtract delay to the IR charts to visually align them and help confirm the correct amount of delay needed. This is kind of similar to using the manual EQ filters to match a house curve. You can test it out before applying it to your DSP.


Everyone please take note of this. This is the most important post in this thread. 

The peak is NOT the initial arrival. For signals that contain high frequencies, using the peak is an appropriate approximation. For signals that have been low passed or that don't contain high frequencies (like a subwoofer), the peak is NOT a good approximation.

The slope of the peak IS high frequency. It takes longer for a 100Hz sine wave to reach the peak than it does for a 20kHz wave and that's exactly what you're seeing. Attempting to determine the actual arrival time using these graphs is very difficult. A tape measure is JUST as accurate.


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## 14642

You could get closer to the true initial impulse using a trick that's incorporated in MS-8--well, sort of. Output the graphs to a chart and do a search in excel for the value of time where the level is equal to about 10%-- just higher than the pre-ringing. Or, just find that value on the graphs. 

The peak is NOT the the initial arrival. This is very simple to prove. Make a measurement like this of your sub with the LPF. Then compare what you measure with yout tape measure to the location of the peak...


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## Babs

subterFUSE said:


> Following up on this thread after spending a fair amount of time on the REW forums learning more about measuring delays via Impulse Response.
> 
> Here are a few things I learned:
> 
> 1. When measuring for TA via impulse response, it's important to only use 1 measurement sweep. I had my REW software performing 2 sweeps by default and this was causing some strange fluctuations in the IR readings.
> 
> 2. You want to align the initial rise of the first peak in the IR, not the actual IR peak (i.e. the 100% mark). This was really confusing me, as you can tell in the post above. It had me thinking the midbass drivers were reading further away than my horns when that wasn't the case. Now that I'm aligning the initial rise, the timings are better and the passenger side horn is registering as farthest away as expected. Here is a pic of my latest IR alignment.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 3. You can hold down CTRL and Right-Click then Drag to get the difference in time and distance calculated for you on screen.
> 
> 4. You can use the Impulse tab Controls menu to add or subtract delay to the IR charts to visually align them and help confirm the correct amount of delay needed. This is kind of similar to using the manual EQ filters to match a house curve. You can test it out before applying it to your DSP.





Andy Wehmeyer said:


> Everyone please take note of this. This is the most important post in this thread.
> 
> The peak is NOT the initial arrival. For signals that contain high frequencies, using the peak is an appropriate approximation. For signals that have been low passed or that don't contain high frequencies (like a subwoofer), the peak is NOT a good approximation.
> 
> The slope of the peak IS high frequency. It takes longer for a 100Hz sine wave to reach the peak than it does for a 20kHz wave and that's exactly what you're seeing. Attempting to determine the actual arrival time using these graphs is very difficult. A tape measure is JUST as accurate.


Having heard this killer S6 Audi, I'd say he certainly has T/A figured out. Some of the best imaging I've heard in car. I concur. Not to mention about the dead sexiest Audi I've seen yet. 

Makes sense though, you're concerned with the "start" of a signal pulse hitting your ears at the same time.. Peak I guess being frequency dependent in terms of milliseconds from 0db start to peak isn't the thing to dial in between subs verses tweeters or even mids unless they're both doing a pulse in the same exact frequency.. Guessing here.

Also, I found last night a polarity pulse tuning track (an extremely short snap sound every 2 seconds I think) where I kinda did this by ear and was amazed that even up to just a few clicks away from perfect you could still tell a delay between two drivers (mids with LP defeated and tweets in the case I tested). 

What's blew me away was when I hit delays perfect at the magical alignment, they suddenly dissappeared and became one. A single fast snap/thump in my center windshield while the track played. And not only did the drivers dissappear and become one, even in the high freq's of the "thump" sound, the tonality changed and you could tell you were no longer hearing two speakers but one in unison playing one sound at full in-phase intensity. From there, toggling delay one notch either direction and it was off again. That's how powerful phase and alignment is. 

To a newb like me, I'm way excited about it having experienced that myself. I know you guys have heard this a thousand times over hehe


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## DDfusion

Andy Wehmeyer said:


> You could get closer to the true initial impulse using a trick that's incorporated in MS-8--well, sort of. Output the graphs to a chart and do a search in excel for the value of time where the level is equal to about 10%-- just higher than the pre-ringing. Or, just find that value on the graphs.
> 
> The peak is NOT the the initial arrival. This is very simple to prove. Make a measurement like this of your sub with the LPF. Then compare what you measure with yout tape measure to the location of the peak...


Can you go to the tech section and look at my MS-8 post please.


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## 14642

It's certainly possible to make a car that sounds great even if everything isn't quite right. I know--I used to make great sounding cars before i understood this--and I made the same mistake--thinking that the peak is the initial arrival. 

If I made a pie and told you it was an apple pie but I had made it with pears, that wouldn't make it a bad pie, but it also wouldn't make it the best apple pie you'd ever eaten.


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## 14642

DDfusion said:


> Can you go to the tech section and look at my MS-8 post please.


I just went there and looked at several posts. Which one?


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## 14642

Lemme add something here. Because the slope of the line determines the absolute location of the peak, the deviation from correct INCREASES as the low pass filter frequency decreases. For the difference between the location of a mid and a tweeter with a 3k or so filter, the peak is still a close enough approximation. As you move down into the midrange and midbass, the deviation is worse.

Now, with that said, we don't hear phase very easily at high frequencies, but we do hear frequency response. If your tweeters are out of polarity, you'll hear frequency response problems at high frequencies and at the lower frequencies you'll begin to hear poor imaging because the lower the frequency the more acute your ability to hear phase.

Below 100Hz or so, where the deviation is HUGE if you're picking peaks, it doesn't matter much because the wavelengths are really long. changing the delay by a foot doesn't affect phase enough to matter. At 200 Hz? Yes. At 40 Hz, no.


What's been posted here is certainly useful, it just needs to be considered a little more thoroughly.


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## DDfusion

Andy Wehmeyer said:


> I just went there and looked at several posts. Which one?


The one titled MS-8 boosting


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## mbradlawrence

Old thread but I wanted to use a USB mic so started looking at this.....

Anyways, the author of REW is working on a solution to allow for t/a with usb. The below link includes an outline of the solution (go to page 1) and just a few weeks ago he indicated that it was in process.

UMIK-1 timing reference and Phase - Page 4 - Home Theater Forum and Systems - HomeTheaterShack.com

BTW, that site is SCARY informative on REW. I'm a Electrical Engineer and it makes my head spin in some cases!!!!!!!!!!!!!!


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## Babs

Wow. That'd be way cool!


Sent from iPhone using Tapatalk


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## mbradlawrence

John M. of REW just posted the below. Looks like I can use my driver side tweeter as reference (it is the closest) and all other speakers will be later in time. Yea USB mic! Any errors in my thought process?

Thread (from hometheatreshack)

I have uploaded V5.15 beta 3 to the installers directory with initial support for an acoustic timing reference. The choice of timing reference is in the Analysis preferences. If an acoustic timing reference is selected the measurement dialog shows a control for the level of the timing reference signal and, when using Java drivers, which channel to use for the timing signal (with ASIO drivers REW uses the timing ref channel selected on the Soundcard preferences). A limitation of this initial implementation is that the reference channel is only used for the timing signal, it can't be measured while using it as the timing reference (e.g. if the left channel output is being used for the timing ref, the measurement sweep will only be on the right channel output). I'll look at addressing that in a subsequent build.

The timing reference signal is a linear sweep from 5 kHz to 20 kHz, so the speaker being used to provide the reference needs a tweeter (a sub cannot be used as the timing ref). Keeping it to high frequencies helps with accurate timing and means the signal will usually be reproduced by the tweeter alone, so the derived timing signal peak location will not be affected by timing differences that may be present between drivers in a multiple driver speaker. The sweep is quite short, about 0.7 seconds, so it shouldn't present any power handling issues. The actual peak levels detected for the timing reference and the measurement can be read from the "Source" section of the Info panel. The delay relative to the timing ref is shown in the "System Delay" section of the info panel, as a time and equivalent distances in metres and feet.

I have only had a chance to test this on a Windows machine. As ever, please let me know of any unexpected behaviour.

Here is the cumulative list of changes since V5.14:
•Moved preview generator into a separate thread to avoid delaying display of file dialog
•Added sample rate, start freq and end freq to default filenames for sweeps saved as WAV
•Removed the confirmation dialog after signal WAV generated
•When meas sweep is selected start and end frequency spinners reflect chosen sweep start/end rather than being half/double
•Removed redundant check for 0..1 input volume range
•Increased range of impedance Y axis to 10k
•Changed signal generator linear and log sweep fade in and fade out to 10 ms (fade in was 8k samples at start freq, fade out was 8k samples)
•Start delay is available for both SPL and impedance measurements, relocated the start delay controls below the Start Measuring button
•Added an acoustic timing reference option

•Bug fix: Drop Small filters used a threshold of 1 dB instead of half the flatness target in some circumstances
•Bug fix: Apply Windows could fail when the overlays window was being used
•Bug fix: Loading cal file could fail when using UMIK under Linux
•Bug fix: Controls in graph axis limits dialog could disappear under some circumstances
•Bug fix: Var, Psy and ERB smoothing did not work on RTA spectrum trace


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## Jscoyne2

Ive gotten majorly lost in this thread. I know that alot of info was added in the last 4 pages. Is there a newly written with updated info on anything important to know?

Also any suggestions for cheap soundcards/ can you use a USB soundcard (and which one?)(not usb mic)

and can you give a more detailed diagram of how to wire it all up. im not sure what you mean by speaker out. 

i guess what im asking is can you simplify it so a 5th grader can do it >.>


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## Jscoyne2

Hanatsu said:


> Yes you would need 4 jacks in total. Mic input on my laptop is 2ch as far as I know. Be sure to set the input as 'Line-in' and not 'microphone'.
> 
> Sent from my Samsung Galaxy 3 via Tapatalk.


^^^^^^


----------



## Babs

So messing around I found an app that fixed the driver issues. I have output sound again from the laptop.. Hmmmmm where shall I begin. Sweeps for sure but what now. Holmimpulse, something else?


Sent from my iPad using Tapatalk


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## jonah1810

Hey I was just wondering where you measure for this sweep? do you just take one measurement at say right in front of your nose?


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## Babs

jonah1810 said:


> Hey I was just wondering where you measure for this sweep? do you just take one measurement at say right in front of your nose?



Valid question. I refer you to this thread for info on spacial averaging. Post #16 begins discussion on mic positions:

http://diymobileaudio.com/forum/showthread.php?t=163234

To put it over-simplified, for sweep measurements in REW, you measure several times with the mic stationary, moving the mic slightly but around where your ears will be.. From that you'll save several plots which you will average. It's far more tedious than pink noise and the RTA function but supposedly is more accurate, especially for high frequencies. One way is to prop the mic pointed forward held down at the headrest. Take a measurement, then move the mic a couple inches, take another, etc.


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## jonah1810

Babs said:


> Valid question. I refer you to this thread for info on spacial averaging. Post #16 begins discussion on mic positions:
> 
> First-timers guide to measuring your system - Car Audio | DiyMobileAudio.com | Car Stereo Forum
> 
> To put it over-simplified, for sweep measurements in REW, you measure several times with the mic stationary, moving the mic slightly but around where your ears will be.. From that you'll save several plots which you will average. It's far more tedious than pink noise and the RTA function but supposedly is more accurate, especially for high frequencies. One way is to prop the mic pointed forward held down at the headrest. Take a measurement, then move the mic a couple inches, take another, etc.


Can you use an average of sweeps for phase measurements? for measuring my FR ive just been doing pink noise and rta. im just trying to figure out to get my time aligntment done most accurately

Edit: I did read through most of hanatsu's guide there, I just didn't realize I could use an average of the eight measurments to do my time alignment


----------



## Hanatsu

No phase averaging.... :<


Sent from my iPhone 6 using Tapatalk.


----------



## edouble101

It has been a few years since I measured impulse response.

Does my feedback loop look good using a Behringer U-Phoria UMC204HD?


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## edouble101

In second thoughts, this will not work as I have pictured. The Behringer U-Phoria UMC204HD will not output both A and B outputs at the same time. That would have been too easy.


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## subterFUSE

edouble101 said:


> In second thoughts, this will not work as I have pictured. The Behringer U-Phoria UMC204HD will not output both A and B outputs at the same time. That would have been too easy.




Use a Y cable?


Sent from my iPhone using Tapatalk


----------



## DonGato

subterFUSE said:


> Following up on this thread after spending a fair amount of time on the REW forums learning more about measuring delays via Impulse Response.
> 
> Here are a few things I learned:
> 
> 1. When measuring for TA via impulse response, it's important to only use 1 measurement sweep. I had my REW software performing 2 sweeps by default and this was causing some strange fluctuations in the IR readings.
> 
> 2. You want to align the initial rise of the first peak in the IR, not the actual IR peak (i.e. the 100% mark). This was really confusing me, as you can tell in the post above. It had me thinking the midbass drivers were reading further away than my horns when that wasn't the case. Now that I'm aligning the initial rise, the timings are better and the passenger side horn is registering as farthest away as expected. Here is a pic of my latest IR alignment.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 3. You can hold down CTRL and Right-Click then Drag to get the difference in time and distance calculated for you on screen.
> 
> 4. You can use the Impulse tab Controls menu to add or subtract delay to the IR charts to visually align them and help confirm the correct amount of delay needed. This is kind of similar to using the manual EQ filters to match a house curve. You can test it out before applying it to your DSP.


What frequency do you use?


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## Jakub220

So today I was using the HolmImpulse for the first time. I set up the time alignment with a tape measure first and I input the numbers in my dsp. 
Then I closed all of the programs and run Homeimpulse. I was using Umik 1 microphone and the Laptop was hooked up to the headunit with an aux cable.

The numbers don't make any sense. How this is possible that the distance is so far that the scale has to be in meters to see all 3 measurements?!

The black measurement is a right midrange in the passenger door. 

Does anybody know what I could have done wrong?


----------



## subterFUSE

Your loopback reference is likely bypassing the DSP which means your measurements are probably showing you not only the flight time in air, but also the latency of the processor.

I don't know how to use HOLMimpulse, but this is true with all dual FFT measurement softwares since they are essentially doing the same thing. Most of the softwares even have a button to re-center the 0 ms axis to remove the latency and/or the flight time in air from the measurements and shift the impulses to zero.

Another option is to run the loopback reference signal through the DSP also, by tapping an otherwise unused output. This will remove the processor latency from your measurements completely since the reference will be going through the processor.


----------



## Jakub220

subterFUSE said:


> Your loopback reference is likely bypassing the DSP which means your measurements are probably showing you not only the flight time in air, but also the latency of the processor.
> 
> I don't know how to use HOLMimpulse, but this is true with all dual FFT measurement softwares since they are essentially doing the same thing. Most of the softwares even have a button to re-center the 0 ms axis to remove the latency and/or the flight time in air from the measurements and shift the impulses to zero.
> 
> Another option is to run the loopback reference signal through the DSP also, by tapping an otherwise unused output. This will remove the processor latency from your measurements completely since the reference will be going through the processor.


I ran everything through the dsp, I also set the first reference signal to zero. 

The holmimpulse was glitching sometimes and I had to restart it and erase previous measurements. I was always starting from the scratch to eliminate any possibile variables that would negatively affect the measurements. 

Would connecting the laptop directly to the dsp help? I have a long jack to RCA cable. 

I'll try again today, thanks


----------



## Petererc

https://www.diysubwoofers.org/misc/holmimpulse/ta.html


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## Jakub220

Yeah, I have followed those instructions. I partially solved the issue by not minimizing the Holmimpulse window, I'm still getting results that absolutely don't match the real distance but the first 2 measurements are on point. I started with my woofers and they came up on point. The midrange speaker was a third one I measured and the results were way off.
I'll the updates once I figure all of it out.


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## tjk_bail

Jakub220 said:


> Yeah, I have followed those instructions. I partially solved the issue by not minimizing the Holmimpulse window, I'm still getting results that absolutely don't match the real distance but the first 2 measurements are on point. I started with my woofers and they came up on point. The midrange speaker was a third one I measured and the results were way off.
> I'll the updates once I figure all of it out.


BUMP.,,,,,,,


Did you get it figured out ???


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## Libba

i dont know what im doing wrong but i get the peaks of both measurements on 1 single line.


----------



## ftmsmohan

Im trying to TA Sub + LMB + RMB as you can see below.
Could someone please guide me how to calculate the distance between these three.
I'm using Focusrite Scarlett 212i + Dayton Audio EMM6.

I'm posting the configurations as well, so please review and comment if i had done anything wrong.















I'm trying to Right Click + Drag in the Overlay-Impulse window, it does gives some values.. and it quickly disappeared, not sure correct or not.








Tried the option"Estimate IR Delay" from Impluse tab, still not sure how this works.

Appreciate if someone could help. Thanks in advance.


----------

