Delay settings soundsystem
Printed From: Speakerplans.com
Category: General
Forum Name: Advanced Discussion
Forum Description: Advanced discussion area for higher lifeforms
URL: https://forum.speakerplans.com/forum_posts.asp?TID=107290
Printed Date: 20 April 2024 at 11:38am
Software Version: Web Wiz Forums 12.06 - https://www.webwizforums.com
Topic: Delay settings soundsystem
Posted By: 303addict
Subject: Delay settings soundsystem
Date Posted: 16 August 2022 at 12:05am
|
Hello,
I got 2 wbin 18inch speaker jbl 2242h (sub 37,5hz to 80hz) Then i got 4 scoops 15inch with jbl 2225h speakers (kick 80hz to 250hz) Then the tops are 2x jbl srx 715f (250-20khz) What will be the best delay set up for this stack? The wbins are 1,26m delay ,so i put it on the jbl srx 715f. But what do i do best with de scoops? Use them as a reflex bin,or use them like horn? What would be the best delay settings i can use? Would be great if some one could help me. |
Replies:
Posted By: 303addict
Date Posted: 16 August 2022 at 12:06am
|
https://imageshack.com/i/pm0uNSzdj photo of setup
|
Posted By: Lucasdude
Date Posted: 16 August 2022 at 1:52pm
|
What LMS are you using? |
Posted By: 303addict
Date Posted: 16 August 2022 at 3:41pm
|
still using the DBX PA for setting delay en x over.... with 3 amps :-Berhinger ep2500 for tops 800w rms 8 ohm -the t amp tsa 2200 for the 4 scoops parralell wired ,so 2x(2x 800w) rms 4ohm -the t amp 4-1300 for the wbins 800w 8 ohm
|
Posted By: kipman725
Date Posted: 16 August 2022 at 4:07pm
|
It's best to measure to find delay, which will vary with position (changing path lengths due to acoustic sources not co-located). I find for a mono stack like this a ground plane measurement 6-10 meters from the stack, centered, is a good point to align to. The REW find IR delay function works well if the measurement isn't contaminated by reflections, the drivers are connected directly to an amp channel for the measurement and the measurement bandwidth is broad. E.G for a reflex sub I measured up to 1kHz. |
Posted By: 303addict
Date Posted: 16 August 2022 at 4:18pm
|
Okay thx,will try that for sure,but we have this saturday a party so it will be later then. We already did a soundcheck openair,and it sounded pretty good So for this party we will use this specifications for the sound. Later i will work with REW to fine tune al the system and time align everything like it should be!
|
Posted By: Lucasdude
Date Posted: 16 August 2022 at 7:09pm
|
Although not perfect, the auto EQ function of the Driverack is worth playing with. It's a quick way to set up the system in different environments. I'm not sure who makes the DBX RPM, but it can be found for a reasonable price. |
Posted By: Lucasdude
Date Posted: 16 August 2022 at 7:11pm
|
this one https://www.jpleisure.co.uk/DBX_RTA-M_Reference_Mic.html" rel="nofollow - https://www.jpleisure.co.uk/DBX_RTA-M_Reference_Mic.html
|
Posted By: 303addict
Date Posted: 16 August 2022 at 8:16pm
|
Indeed can try that also ,but i have berhinger ecm800 mic Think that that no problem I suppose.
|
Posted By: 303addict
Date Posted: 16 August 2022 at 8:19pm
|
What would be de best mic setup for this soundsystem? How far away,what position of the mic 90° or straigt to the sound system?
|
Posted By: toastyghost
Date Posted: 17 August 2022 at 12:07am
A set of EQ filters generated from an RTA measurement has nothing to do with time alignment. RTA is time-blind. Also, be very wary of doing any alignment without the final low pass and high pass filters in place for each pass band. Those filters inherently affect phase (as does every IIR filter) so the alignment probably won't hold up once they're added in unless you're very careful to align the rise time — not the peak! — of quasi-anechoic IR data & then choose suitable complimentary electrical filters to achieve the desired target acoustic crossover. That might not necessarily be achieved by setting the same corner frequency and slope type on both channels. Especially if you're trying to align a ported passband to a horn or sealed enclosure, for example. If you’re always going to be stacking the rig with the grilles aligned, then at least you only have to do this process once. Assuming you don’t change the crossover values or EQ filters on the output channels of your DSP, that is. Follow this article: https://www.merlijnvanveen.nl/en/study-hall/166-subwoofer-alignment-the-foolproof-relative-absolute-method" rel="nofollow - https://www.merlijnvanveen.nl/en/study-hall/166-subwoofer-alignment-the-foolproof-relative-absolute-method And/or this one: http://www.excelsior-audio.com/Publications/Subwoofer_Alignment.pdf REW would be my preference on zero budget, thanks to the delay synthesis tool & frequency-dependent windowing. It also has a good display of group delay as used in Charlie Hughes' guide. However, it doesn't do real-time dual-channel FFT display like the Smaart software used on Merlijn's guide - but OpenSoundMeter does & is also free. Make sure you verify all channels sum as expected with a measurement of the whole rig once you've set the delay values and filters. That step is often missed. |
Lucasdude wrote:Posted By: 303addict
Date Posted: 17 August 2022 at 8:55pm
| Okay thx for this explanation!! |
Posted By: kipman725
Date Posted: 18 August 2022 at 11:25am
|
If you perform the measurements with no crossover in place you get higher resolution data and data that is universally applicable; I.E it's the actual delay. If you use these measured delays as a starting point for crossover design and don't vary them you should be able to ensure good summation if the drivers have roll-offs near the crossover points by using a mix of: Asymmetrical crossover slopes, all pass filters, filters with different HP and LP points. A brute force approach (only works with floating point processors due to internal signal levels) is to flatten the drivers to an extra half octave either side of the crossover point using PEQs and then apply standard LR4/LR8 crossovers. In most cases that should result in good summation. A free tool to simulate your crossovers is https://kimmosaunisto.net/%20" rel="nofollow - VituixCAD |
Posted By: toastyghost
Date Posted: 18 August 2022 at 12:37pm
That generally only holds true if you align using the group delay outside of the device passband, as shown in the Charlie Hughes guide. Experience users can do it other ways, but their process is established and they know what they’re looking for when making the measurements. The problem with running devices with no filtering in place is that most people will align the IR peaks, which is not the true time of flight or first arrival. Intrinsically, the IR peak will be the high-frequency data - useless if you are trying to align a sub and full range cabinet. Same for the ETC peak, although less so. The rise of the ETC is where you need to be looking - but this is difficult for seasoned folk, let alone beginners. The other issue is that even if you get it correctly aligned without filters, the filters will then change the alignment through the acoustic crossover region. Unless they are carefully chosen — and parametric EQ or alternative filter shapes used to achieve a measured version of the actual target response of something like 4th order Linkwitz Riley — the delay values will cause crossover ripple at best, and cancellation at worst. I should also point out that every DSP unit implements the same ‘standard’ filter shapes in varying different ways. Your simulated biquad filters aren’t always what comes out once the values are shoved into the actual hardware doing your processing. Verification, verification, verification! There is a series of 3 papers from the late 70s/early 80s by the illustrious Dick Heyser on determining the correct delay to observe the ‘true’ phase response of a loudspeaker if you’re interested in the deeper background. |
Posted By: kipman725
Date Posted: 18 August 2022 at 2:18pm
|
REW doesn't use the IR peak if you use the estimate IR delay: "Estimate IR delay calculates an estimate of the time delay in the
measurement by comparing it with a minimum phase version. The delay it
calculates can be removed from the impulse response by pressing the
Shift IR button on the panel shown after the delay is calculated and
can additionally be applied as a timing offset for subsequent measurements." https://www.roomeqwizard.com/help/help_en-GB/html/graph_impulse.html#top" rel="nofollow - https://www.roomeqwizard.com/help/help_en-GB/html/graph_impulse.html#top
To me this is more of a difference in methodology to achieve the same outcome. There are many delay values that will work to achieve summation but only one that is the distance to the acoustic center (ignoring frequency variance) and this may not easily result in summation due to relative phase shifts between the boxes. so if tuning in the field you may want to perform a more iterative process with crossovers in place. I am interested in the Dick Heyser papers if you have them? |
Posted By: 303addict
Date Posted: 18 August 2022 at 11:04pm
|
I am interested in the Dick Heyser papers if you have them? also intresested in that how more info how better i will understand it beter and beter. thx so mutch to ever one for all the posts! |
Posted By: toastyghost
Date Posted: 19 August 2022 at 10:55am
That’s definitely better than auto-aligning the energy peak, but it’s still not ideal in many real world situations. A minimum phase plot can only be accurately calculated from the Hilbert transform of the magnitude response if the data is anechoic. That means you have to ensure you know the limitations of the measurement environment, make a large number of averages to improve the SnR and reduce the effects of stuff like wind, apply carefully chosen windowing to remove reflections, and then assess the results before you do the alignment auto-magic. Using the group delay is much easier in my experience. Especially in many real world situations, where you might not have access to the DSP to remove the filters or be risking damage to components by running a high enough level log sine sweep to get a signal to noise ratio of >20dB at the absolute minimum. When doing log sweeps, I use a local shelving filter at the audio interface outputs to protect HF drivers (and my ears). That doesn’t affect the results of the SnR is high, because the shelf is applied on the loop back too.
Sure, there are different approaches when designing a system from scratch (or integrating mixed configurations of existing speakers) and working at a gig or venue. There is always going to be some passband ripple, even if it’s only 0.2dB. Of course, moving off-axis does that more as the time of flight distance to each source changes. Since the majority of listeners are off-axis — even for hifi — it’s also important to align or verify the results where people will actually be. What is also often overlooked is the fact the choice of acoustic crossover slope affects the directivity. You can end up steering the main lobe up or down, or creating side lobes (vertically) that result in nasty reflections from the roof. The ‘yellow bible’ book by Davis & Patronis Jr, Sound System Engineering, is a goldmine for this stuff with lots of nice diagrams. I got a second-hand copy of the third edition for a tenner on eBay.
The three-part series is titled "Determination of Loudspeaker Signal Arrival Times" but unfortunately it doesn't seem to be open access on the AES e-Library. Someone has uploaded the first part here, though it isn't the juiciest or most practical section: https://sdlabo.jp/archives/Determination%20of%20Loudspeaker%20Signal%20Arrival%20Times.pdf This four-page article isn’t about alignment explicitly, but it’s mostly plain English and has good diagrams regarding acoustic measurements in general. Start with that, before making too many choices based on what the computer shows: https://www.prosoundweb.com/accuracy-applied-part-1-in-a-series-on-the-keys-of-loudspeaker-measurements/ |