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what's the correct delay settings for es18's/hogs

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SouthwestCNC View Drop Down
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Post Options Post Options   Thanks (0) Thanks(0)   Quote SouthwestCNC Quote  Post ReplyReply Direct Link To This Post Posted: 23 October 2017 at 12:53pm
Originally posted by Aman Gebru Aman Gebru wrote:

Unbelievable thread.

Are they really people out there who have no idea how to setup their systems. Many have been doing it the correct way 30 years ago when MLSSA first came out in 1987, so there really is no excuse.

Using physical distance to apply delays is about as useful as leaving your car in first gear and calming your getting the maximum performance from it. Sort this out guys. It only takes a few youtube videos and a demo version of Smaart or the free version of HOLMimpulse and a 30 quid soundcard to do it right.



Obviously never been to a free party then Aman .. Neither have I for that matter
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Post Options Post Options   Thanks (0) Thanks(0)   Quote Aman Gebru Quote  Post ReplyReply Direct Link To This Post Posted: 23 October 2017 at 1:09pm
Here's a few links to get you going.

This is the manual for LinearX LMS version 4. It doesn't cover much on alignment as phase has to be derived using minimum phase transform, but the section on how to take measurements of individual speakers is well worth a read. The fun starts at page 289.

http://www.physical-lab.com/images/pdf/LMS410_MANUAL.pdf

The next link is to a paper that has an in-depth section on time behaviour. Read from page 19 onwards.

http://www.kirchner-elektronik.de/upload/15522386-Loudspeaker-MeasurementTechnology.pdf

The last link is to a video by John Atkinson. This is probably the best explanation you will ever get on loudspeakers and should be essential viewing to anyone in the industry. If you watch this and your understanding doesn't improve by 1000 times, then give up.

http://www.youtube.com/watch?v=j77VKw9Kx6U

I suggest Ian makes this a sticky as I'm not normally in the mind frame to give out such classified information.


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Post Options Post Options   Thanks (0) Thanks(0)   Quote Aman Gebru Quote  Post ReplyReply Direct Link To This Post Posted: 23 October 2017 at 1:15pm
Originally posted by SouthwestCNC SouthwestCNC wrote:

Obviously never been to a free party then Aman .. Neither have I for that matter

I would never have guessed....
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Post Options Post Options   Thanks (0) Thanks(0)   Quote SouthwestCNC Quote  Post ReplyReply Direct Link To This Post Posted: 23 October 2017 at 2:08pm
Originally posted by Aman Gebru Aman Gebru wrote:

Here's a few links to get you going.

This is the manual for LinearX LMS version 4. It doesn't cover much on alignment as phase has to be derived using minimum phase transform, but the section on how to take measurements of individual speakers is well worth a read. The fun starts at page 289.

http://www.physical-lab.com/images/pdf/LMS410_MANUAL.pdf

The next link is to a paper that has an in-depth section on time behaviour. Read from page 19 onwards.

http://www.kirchner-elektronik.de/upload/15522386-Loudspeaker-MeasurementTechnology.pdf

The last link is to a video by John Atkinson. This is probably the best explanation you will ever get on loudspeakers and should be essential viewing to anyone in the industry. If you watch this and your understanding doesn't improve by 1000 times, then give up.

http://www.youtube.com/watch?v=j77VKw9Kx6U

I suggest Ian makes this a sticky as I'm not normally in the mind frame to give out such classified information.





Excellent links. Thank you! Its interesting how John Atkinson explains anything within 7ms is interpreted by our brains as a single arrival. The hog falls slightly outside of that with any non horn loaded cabs, but kind of suggests that the es18's we could not differentiate the correct delay setting applied to no delay. Therefore how essential is measurement and getting within 1-2ms or so.

From my own experiments I believe I can hear a noticeable difference down to sort of 2ms so I'm not sure id agree with john on that but interesting point is our brains are good but they have their limit. so there must be a kind of tolerance where it makes no odds.


Edited by SouthwestCNC - 23 October 2017 at 2:13pm
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Post Options Post Options   Thanks (0) Thanks(0)   Quote Aman Gebru Quote  Post ReplyReply Direct Link To This Post Posted: 24 October 2017 at 1:18am
What the meter says and what our brain tells us are 2 very different things. Modern measurement systems are many 1000 times more accurate than we are at detecting differences.

Say we have a sub crossed over into an upper bass cab at 80Hz. L-R 18 slopes on both hi and lo pass sections. By adding only 1 to 1.5ms to the upper bass cab you might see a corresponding change in amplitude at 80Hz of up to 6 or 7dB. It might be a gain or a subtraction, depending on the phase relationship. It will definitely be enough to feel there is a difference even though you might not hear the change in amplitude.

Take the same case and move it up into the midrange at say 2kHz. Same L-R 18dB slopes on both hi and lo pass at 2kHz. Adding 1.5ms to the HF section won't see anything like the difference you will see at lower frequencies. There will be a difference in amplitude, but maybe only around 3dB at the crossover point. You won't be able to hear this difference in amplitude, but you will detect a change in flavour to the sound of the system and this will be down to a change in phase more than amplitude. Even 1ms if placed at the right frequency is enough to turn a system that had a phase wrap at its crossover point into a system with a phase response with no more than say 70 degrees variation about 100Hz. The system with minimum phase deviation will sound infinitely better and more lifelike that a system with a 360 degree wrap at its crossover point, even if the amplitude is way off from being flat or near its target response.

So yes, to us 7ms might only serve to change the character of a system, but to sensitive measurement system its enough to put you in the next field.

You really should go to more parties in fields, there's a lot to be learnt there.


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Post Options Post Options   Thanks (2) Thanks(2)   Quote Aman Gebru Quote  Post ReplyReply Direct Link To This Post Posted: 24 October 2017 at 1:42am
To add to this I will say that the above dB changes I was taking about are in amplitude. To align phase at lower frequencies requires more time offset (delay) due to the longer wavelengths involved. A much shorter time variation is required the higher up you go.

Here we start to see the problem of phase vs amplitude and why you can't just look at the whole thing as amplitude. Small time delays between low frequency bands can appear to change the amplitude a lot whilst the phase response doesn't change much at all. This is kind of ok as we are less sensitive to phase changes lower down in frequency. You might suffer from a degraded transient response, but to most as long as the system is not being destructive in amplitude between bands, it will appear to sound good. 

Midrange variations in phase are more apparent as we are more sensitive in that band. It doesn't take a big difference in time delay between bands to tell something's not right. The amplitude might not swing as much as the phase does here, but phase will be the determining factor of sound quality this high up.

This is where FIR filters come in as they can separate phase from amplitude. You can adjust for a great phase response then adjust the amplitude whilst leaving the phase untouched. Its the wholly grail in sound really and I'm even more surprised that more haven't taken it on board. 

You could tell the systems that were using FIR filters at this years Notting Hill Carnival. Anything by L Acoustics, D&B and a few others sounded really good. In fact so good that we can now hear how bad the rest of the systems have really sounded for the last 30 years.

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Post Options Post Options   Thanks (0) Thanks(0)   Quote 4D Quote  Post ReplyReply Direct Link To This Post Posted: 24 October 2017 at 9:19am
Originally posted by Aman Gebru Aman Gebru wrote:

Here's a few links to get you going.

This is the manual for LinearX LMS version 4. It doesn't cover much on alignment as phase has to be derived using minimum phase transform, but the section on how to take measurements of individual speakers is well worth a read. The fun starts at page 289.

http://www.physical-lab.com/images/pdf/LMS410_MANUAL.pdf

The next link is to a paper that has an in-depth section on time behaviour. Read from page 19 onwards.

http://www.kirchner-elektronik.de/upload/15522386-Loudspeaker-MeasurementTechnology.pdf

The last link is to a video by John Atkinson. This is probably the best explanation you will ever get on loudspeakers and should be essential viewing to anyone in the industry. If you watch this and your understanding doesn't improve by 1000 times, then give up.

http://www.youtube.com/watch?v=j77VKw9Kx6U

I suggest Ian makes this a sticky as I'm not normally in the mind frame to give out such classified information.




Cheers boss..

DMZ. "The bass was intense. Girls were literally running up to stand next to the subs"
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Post Options Post Options   Thanks (0) Thanks(0)   Quote SouthwestCNC Quote  Post ReplyReply Direct Link To This Post Posted: 24 October 2017 at 11:31am
Originally posted by Aman Gebru Aman Gebru wrote:

To add to this I will say that the above dB changes I was taking about are in amplitude. To align phase at lower frequencies requires more time offset (delay) due to the longer wavelengths involved. A much shorter time variation is required the higher up you go.

Here we start to see the problem of phase vs amplitude and why you can't just look at the whole thing as amplitude. Small time delays between low frequency bands can appear to change the amplitude a lot whilst the phase response doesn't change much at all. This is kind of ok as we are less sensitive to phase changes lower down in frequency. You might suffer from a degraded transient response, but to most as long as the system is not being destructive in amplitude between bands, it will appear to sound good. 

Midrange variations in phase are more apparent as we are more sensitive in that band. It doesn't take a big difference in time delay between bands to tell something's not right. The amplitude might not swing as much as the phase does here, but phase will be the determining factor of sound quality this high up.

This is where FIR filters come in as they can separate phase from amplitude. You can adjust for a great phase response then adjust the amplitude whilst leaving the phase untouched. Its the wholly grail in sound really and I'm even more surprised that more haven't taken it on board. 

You could tell the systems that were using FIR filters at this years Notting Hill Carnival. Anything by L Acoustics, D&B and a few others sounded really good. In fact so good that we can now hear how bad the rest of the systems have really sounded for the last 30 years.



Thanks Aman. Very concise explanation This prompted me to have a look at the the minidsp. I can definitely see the benefits of using Fir. Thanks for bringing to my attention. Infact ive ordered a minidsp to have a play with :). Apart from the minidsp, what rack mount processors have fir? seem a search of the subject doesn't come up with many results.

Edited by SouthwestCNC - 24 October 2017 at 11:33am
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Post Options Post Options   Thanks (0) Thanks(0)   Quote toastyghost Quote  Post ReplyReply Direct Link To This Post Posted: 24 October 2017 at 11:48am
Using FIR is NOT a magic bullet. If you don't have serious experience you can do all sorts of stuff that will look amazing at one spot but be god awful at another. Get the basics right using minimum phase filters first, and when you're good at that in a variety of circumstances, then look at FIR.
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Post Options Post Options   Thanks (0) Thanks(0)   Quote SouthwestCNC Quote  Post ReplyReply Direct Link To This Post Posted: 24 October 2017 at 12:50pm
This why dsp is frowned upon as a bandaid to not getting the horns right in the first place.
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