Jump to content

Timing Measurements


jcmusic

Recommended Posts

"Ok what exactly am i looking at in your diagram? We are talking about khorns here."

What part of
 

"Digital delay is the only fix."

Didn't you understand?

What part of "If the horns are deeper than the woofer, then an all-pass delay will fix it." and "I've done the above (balanced) with dual 12's"

Didn't you understand?

Link to comment
Share on other sites

I wonder how audible a 4.8 ms delay is?

I believe that most people will not notice a 1 / 200th of a second delay in itself, but when the sound of the squawker (or tweeter) constantly arrives a few frequency cycles EARLIER than the other parts of the music, there are varying phase shifts and a loss of accuracy. Room boundary reflections affect the perceived arrival of sounds, so time delay correction might be more beneficial with certain room/speaker combinations than others, but it is fun for me to pursue the theoretical ideals in the quest to build a system that works really good.

Edited to change word LATER to EARLIER.

Edited by Khornukopia
Link to comment
Share on other sites

"what is an all pass delay? "

See the schematics.

The all-pass has delay without changing the frequency response of the signal it's passing (all-pass).

http://en.wikipedia.org/wiki/All-pass_filter

Q: What are allpass filters and what are they used for?

A: Allpass filters are filters that have what we call a flat frequency response; they neither emphasize nor de-emphasize any part of the spectrum. Rather, they displace signals in time as a function of frequency. The time displacement accomplished by an allpass filter is specified by its phase response. Allpass filters are used in circuit design to perform various frequency-dependent time-alignment or time-displacement functions. Audio applications include filter banks, speaker crossovers, and reverberators. Allpass filters appear in both continuous- and discrete-time applications.

Edited by djk
Link to comment
Share on other sites

I wonder how audible a 4.8 ms delay is?

I believe that most people will not notice a 1 / 200th of a second delay in itself, but when the sound of the squawker (or tweeter) constantly arrives a few frequency cycles later than the other parts of the music, there are varying phase shifts and a loss of accuracy. Room boundary reflections affect the perceived arrival of sounds, so time delay correction might be more beneficial with certain room/speaker combinations than others, but it is fun for me to pursue the theoretical ideals in the quest to build a system that works really good.

The only thing wrong with this is the sound from the  tweeter/squawker arrivers first...

Link to comment
Share on other sites

I wonder how audible a 4.8 ms delay is?

I believe that most people will not notice a 1 / 200th of a second delay in itself, but when the sound of the squawker (or tweeter) constantly arrives a few frequency cycles later than the other parts of the music, there are varying phase shifts and a loss of accuracy. Room boundary reflections affect the perceived arrival of sounds, so time delay correction might be more beneficial with certain room/speaker combinations than others, but it is fun for me to pursue the theoretical ideals in the quest to build a system that works really good.

The only thing wrong with this is the sound from the  tweeter/squawker arrivers first...
Thank you for pointing this out, I will edit my original posting. I wouldn't want to be the one to create confusion on the topic because of the time delay in my thought process!
Link to comment
Share on other sites

 

 

Yesterday after taking some measurements of all my drivers which are well balanced across the board, I did notice that the woofer is about 4.8ms behind the mid/tweeter. Is there nothing that can be done to correct this other than going active? Did I read some where about some delay being in the xover?

 

How, specifically, did you measure the drivers' time delay?

 

Hi Rudy,

I used REW and it's impulse feature, doing one driver at a time on each side. So that's six total measurements.

 

 

Here's a good article on using (or not using) impulse response for time alignment:

 

http://bobmccarthy.wordpress.com/2010/02/08/phase-alignment-of-subs-why-i-dont-use-the-impulse-response/

Link to comment
Share on other sites

The reason I asked how you did it is so I could understand the methodology.  Based on what you said, I'm not sure you have an accurate result. The best way to use REW to determine the time of the drivers is to use the loopback feature of the unused channel so that the system delay time is taken into account. That is, the unused channel on the sound card you are using.  Then, you should be able to just hit the 'i' button on the REW program and it will tell you the delay time of the measurement with the system delay taken into account.  IIRC, the REW manual tells you how to do that. 

 

The other issue is that the passive crossover may be adding delay as well.  I don't know for sure since I have not looked into taking measurements when drivers are hooked up to a passive XO. I am assuming you disconnected the drivers not being tested and sent the signal to the XO with only one driver hooked up at a time.  Correct?

Link to comment
Share on other sites

The reason I asked how you did it is so I could understand the methodology.  Based on what you said, I'm not sure you have an accurate result. The best way to use REW to determine the time of the drivers is to use the loopback feature of the unused channel so that the system delay time is taken into account. That is, the unused channel on the sound card you are using.  Then, you should be able to just hit the 'i' button on the REW program and it will tell you the delay time of the measurement with the system delay taken into account.  IIRC, the REW manual tells you how to do that. 

 

The other issue is that the passive crossover may be adding delay as well.  I don't know for sure since I have not looked into taking measurements when drivers are hooked up to a passive XO. I am assuming you disconnected the drivers not being tested and sent the signal to the XO with only one driver hooked up at a time.  Correct?

Yes Rudy that is correct and I did use the loopback setup to do this...Also disconnected the other drivers...

Edited by canyonman
Link to comment
Share on other sites

Rudy,

This can get really deep and have no end, I am no expert but; I do know what sounds good when I hear it. I may not get it perfect electriclly but it sounds damm good right now!!!

 

That's all the counts. As I have mentioned, I'm no expert.  Just know enough to be dangerous.  

 

One of these days, I'll take a few hours to play around with timing and see if I can hear the difference. 

Link to comment
Share on other sites

Does anyone know if the passive crossover will add a time delay to the driver being tested?  I am pretty sure it will affect the phase you observe since my understanding is that a filter will affect the phase.

 

I think that your answer is yes. Looking at the group delay curve of a low pass Butterworth filter (red line)...it's variable with frequency.  The purist idea of "time delay" sort of breaks down - think of it as time delay at the frequencies of interest.  Danley uses the 90 degree phase shift of his crossover filters to add time delays in his Synergy Horn series to the drivers that are closer to the horn's mouth by stacking the Butterworths in series. Bessel filters have flat delay curves.

 

Butterworth3_GainDelay.png

Edited by Chris A
Link to comment
Share on other sites

Thanks Chris, I figured it might add some sort of delay as the signal went through the filters. I know the phase is also affected as the filter is used, which is why everything I have read recommends not EQing after you have time and phase aligned drivers.

 

I have also found in my messing with my drivers that if I use a different frequency range to test my drivers, it affects the time reading. I have noted that when using the same frequency sweep for both drivers, I get a more accurate measurement.  

 

Not sure why that happens, or if i am making some mistake along the way.  

 

I do know that if I use the same frequency sweep for both drivers.  Then apply the time difference to the faster driver. Then run the sweep again and check the phase, it is nearly perfectly aligned, confirming the timing to be correct......at least that is how i understand it.

Link to comment
Share on other sites

My experience is that using digital active crossovers is an education in filtering in itself, when used with REW or something that can show unintended changes in phase or amplitude after using filters. 

 

Mark F. (Mark1101) made a comment about "too many PEQs" in the Belle thread using a Dx38 that led me to completely overhaul my settings - and I found that I could use far fewer PEQs and still get the desired results (except for the bass bin which requires 3 PEQs to tame its FR).  The midrange wound up requiring no PEQs, and the Beyma CP25 tweeter required only two, and now has flat FR out past 20 KHz.

 

post-26262-0-65900000-1407595619.jpg

Link to comment
Share on other sites

It has been a while since I last chimed in but this is a subject I think is really very relevant for a Klipschorn owner. Other than i.e. a KEF107 these reproducers have built in time alighment flaws. Luckily these don't show up in big rooms with KH's set apart wide, where you merely hear reflections and less direct sound. On the other hand in a small room, listening from a short distance and powered by a very fast amp without any distortion (like the F3) one hears the slap on a drum or the stick on the brim not as one sound but composed out of two or three seperate sounds. Also staccato singing is perceived like that. But your brains adapt to it and you get used. BUT once you fix it the difference is very obvious. 

I did so by means of a MiniDSP and Y-ing together hi and lo output of the pre-amp. A solution for the books one of the forum members kindly called it. And also by setting the HF horn right to the back. 

 

This brings me to a new angle on the discussion: DIRAC is a DSP that corrects room response but also time-aligns speakers upto 10 msec. I really like that idea and was wondering if any member has any experience with this solution. It seems so perfect for our beloved Khorns. MiniDSP offers a hardware solution with built in DIRAC Live. But one could also use the music server (PC or Mac).

Edited by Malleman
Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

×
×
  • Create New...