How to figure how much delay is needed

[The Dude]

OK, if I want to obtain system delay I need another setup, usb mics are good for taking frequency response but that is it. Not good for all the other stuff I am trying to do. So I will have to look at options for the right set up.

In the mean time I will be doing some more research, as hopefully I can some how understand all of this. I need to go back and try to read through all the post, I need to understand what I am doing as to why I am doing it, and how to properly do it. Rudy has giving me some good threads to read, which I think will help along with reading the rew help files.

[NBPK402]

What kind of AVR do you have? Won't the AVR take care of the delays for you automatically?

[Rudy81]

What kind of AVR do you have? Won't the AVR take care of the delays for you automatically?

I'm not talking about AVR controlled system delay, for example Audyssey. I am talking about actively crossed speakers, where each driver is individually controlled by a crossover and has separate amplification.

[The Dude]

What kind of AVR do you have? Won't the AVR take care of the delays for you automatically?

I'm not talking about AVR controlled system delay, for example Audyssey. I am talking about actively crossed speakers, where each driver is individually controlled by a crossover and has separate amplification.

Yes what he said.

[Rudy81]

duder1982.....you got me thinking about the distortion plot so am looking into that right now. I pulled up my original open baffle measurements when I brought the OBs up to my room.

This is what I got....

[The Dude]

OK so when PQing I want to lower the peaks right. So i would dial in the peak frequency such as 120 hz, then I need to figure out how much q is need or band width correct, then how about db drops.

[Rudy81]

OK so when PQing I want to lower the peaks right. So i would dial in the peak frequency such as 120 hz, then I need to figure out how much q is need or band width correct, then how about db drops.

I do very little eq on my own since Audyssey Pro does most of it for me. I'm sure someone can suggest a good way to do that.

[The Dude]

So audyssey pro eqs the Oris horn separately and the open baffle separate, then you manually go in and adjust your time delay, and add your crossover. I will say tonight I played around with my settings, this setup is sounding pretty good.

I was looking at a post from Cask, on tri amping the Belle, and Lascala. He had some eq settings posted so I plugged them in to the Minidsp, and left the P audio 4525 with the K69 un equed, it as well sounded pretty good, alot better then it did before. Even better when cranked up.

This is where I am really having fun, I might leave them alone for a bit so I can do some work in the basement. I often wonder if I am trying to eq some room responses. Maybe thats why is sounds good but measure bad,

Edited April 30, 2014 by duder1982

[Rudy81]

No, no, no. Audyssey can only treat each speaker as 'one' unit. I use REW to time align the Oris horn and OB bass with each other. I also use REW to set the amplifier gain in order to match levels at the crossover point (This is important since my Oris is way more efficient than my 15" woofer drivers). This, in order to present my Oris, OB and individual actively controlled left and right subs as a 'single speaker'. I do not use REW to eq any of the drivers or speakers, I leave that job to Audyssey since it can compensate for room modes and speaker deficiencies as well.

Once the Oris, OB and sub are actively crossed and aligned, I don't mess with the crossover any more. Again, no eq is done at this stage since Audyssey runs will come next. If I change the crossover point on any of the drivers, I do have to go back and do it all again to ensure the timing and phase alignment is correct, then run Audyssey again....yes, it can be a pain in the arse. However, I don't normally change things unless I have a good reason.

[The Dude]

OK so when PQing I want to lower the peaks right. So i would dial in the peak frequency such as 120 hz, then I need to figure out how much q is need or band width correct, then how about db drops.

I do very little eq on my own since Audyssey Pro does most of it for me. I'm sure someone can suggest a good way to do that.

When you run Audyssey to auto eq, do you do it to the lf and hf separate, or the speaker as a whole. Sorry if my question sound redundant but I am having problems wrapping my head around this whole active thing. I am slow when it comes to reading, as sometimes helps me to learn hands on.

One thing that makes me think you would eq the speak as a whole, is back in the why did people eq at all, to make up for their individual listening space. So they would by speakers with passive networks then would go through and eq the speakers with the passive networks in place. We are just removing the passives here and adding in the eq with the crossover, with now the ability to electronically time align eliminating the need for physical time aligning.

[Rudy81]

Audyssey is a separate process and is done to the speaker as a whole....it is last in the process I use.

The active crossover is used in place of the passive xo in order to 'build' the speaker from the individual drivers, but allowing us to set our own XO point and EQ if you so desire. The active XO also allows us to play with the timing of each driver using that XO.

You are not slow at all, it just takes a bit to understand the various items we are discussing. For the time being, take Audyssey out of your thought, since it is something that is used AFTER you have put each 'whole' speaker together. The timing we are discussing here relates to the relationship between drivers in a given speaker. The timing Audyssey does is between whole speakers and how sound effects reach the listening position.

[The Dude]

Ok that helps,

I have made it to page 8 of your thread your posted a link to, I hope I have made it past Ben and Dr. whos discussion in regards to which is necessary or better. Looks like I have made it to the point where you picked up the EV one.

[Rudy81]

Ok that helps,

I have made it to page 8 of your thread your posted a link to, I hope I have made it past Ben and Dr. whos discussion in regards to which is necessary or better. Looks like I have made it to the point where you picked up the EV one.

As you can see in the other thread, it took me a long time to wrap my head around the concepts and how to implement them. That is not to say I am fully up to speed. I learn stuff all the time as I read different articles etc. BTW, the RANE site has a lots of good articles on XOs and EQ.

[The Dude]

Got to this point,

Backing
up a step...the goal of the xover is to sum the two drivers together so
that the final passband is flat. Just picking a xover frequency doesn't
guarantee a flat passband because you have phase rotation from the time
arrival differences, and from the natural phase response of the
individual drivers...and even from the type of xover slope too.

Here's a crash course in how I personally go about doing it...

When you pick a xover point (based on polars/distortion
numbers), what you want to do is put in the filter and then measure the
MF and HF separately, but plot them on the same graph (use the "all
measured" tab in REW). The frequency where the two lines overlap is the
acoustic xover point, which can be different from the electrical xover
point since your speakers won't be perfectly flat. Then take a
measurement of both drivers combined...what you want to have happen is
for the combined response to be 6dB louder than each driver is
individually at the acoustic xover point. Usually what happens
(especially with horns) is that you'll measure a dip instead of +6dB.

To
fix the phase, what you want to do is add delay to the tweeter to
time-align the two signals. You can actually measure the time delay
using REW, which would be a good starting point, since that compensates
for the propogation delay. Using a LR xover type ensures that the xover
is phase aligned (since it is by definition). So all that is left is to
compensate for the individual driver phase. The easiest way to do this
is to adjust the delay on the HF.... Start by playing a test tone at
the acoustic xover point and measure the SPL. You'll notice that you'll
be able to make the SPL at the acoustic xover point go up and down as
you change the delay...so stop when you hit +6dB. Also make sure you
limit yourself to within +- 1/2 cycle so that you don't mess up the
time alignment (every 360 degrees of phase the on-axis magnitude will
be +6dB).

Now measure again and you should see that your dip got filled in real nicely and your passband is flat.

Keep
in mind that this is just the basics...there's a lot more involved with
voicing a speaker than just setting the xover frequency and getting
proper summation through the passband. There's a few other nuances that
show up with the xover settings too that are hard to describe without
having measurements handy.

My question is what does he mean by frequency over lap. If I run a test on the lf from 20-1000, and the hf from 300-18000, then wouldn't there be overlapping from 300-1000.

[The Dude]

When you pick a xover point (based on polars/distortion numbers), what you want to do is put in the filter and then measure the MF and HF separately, but plot them on the same graph (use the "all measured" tab in REW). The frequency where the two lines overlap is the acoustic xover point, which can be different from the electrical xover point since your speakers won't be perfectly flat. Then take a measurement of both drivers combined...what you want to have happen is for the combined response to be 6dB louder than each driver is individually at the acoustic xover point. Usually what happens (especially with horns) is that you'll measure a dip instead of +6dB.

This was very helpful, I haven't gone through this the fullest extent, but just following some easy steps I am back to listening to music.

I have once read when you want to know the answer to question a you ask question b, kind of what I did here but by accident. I know delay is important as I have done physical delay it made a difference.

But after following these little steps I have a way better sounding system.

Thanks guys for the good reading material. Once I get my room together, and better measuring system, I will give this another try.

[NBPK402]

What kind of AVR do you have? Won't the AVR take care of the delays for you automatically?

 

I'm not talking about AVR controlled system delay, for example Audyssey. I am talking about actively crossed speakers, where each driver is individually controlled by a crossover and has separate amplification.

I understand that... I was under the assumption that if you ran from your AVR to your MiniDSPs then to your amps that you would setup the crossover in the MiniDSP and that Audyssey would take care of the delays... After reading some more of your posts I can see I was wrong.? I am just getting ready to set mine up and I am unsure of where to start. If the AVR won't take care of it then it sounds like I will have an even harder time with doing this. I wish I had someone who was close to my location that could come over and help me get these setup. I have the mic, minidsps, and can download the latest version of REW. Edited September 4, 2014 by ellisr63

[The Dude]

What Mic are you using.

[NBPK402]

What Mic are you using.

UMIK-1 from Mini-DSP

[The Dude]

As far as I know you are going to need a different Mic set-up for the delay. If you want to set the crossovers refer to one of the last post. It's not perfect but it works. Then you can run audyssey and see how it sounds. Going through the long detailed steps is for optimal sound.

[NBPK402]

As far as I know you are going to need a different Mic set-up for the delay. If you want to set the crossovers refer to one of the last post. It's not perfect but it works. Then you can run audyssey and see how it sounds. Going through the long detailed steps is for optimal sound.

What mic do I need for setting up the delay?