Measuring horn reflections back into the throat

Status
This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.
In the last few years I've got several (cheap) DSPs but did not program it myself, I just used the buttons on the unit or an application with a GUI to create my settings, I did nothing as complex as you are planning, I just encountered their limitations pretty quickly. I solved that by minimizing the number of applied filters or by using more DSPs for different channels (instead of one for all) but quickly came back to the basics as in 'what you fix physically on the speaker doesn't need processing power' and 'find your compromise, don't fix EVERYTHING'. Especally the latter is very important. Absolute perfection can't be reached in acoustics. Well, at least so far, maybe that changes in the future.

See, I'm not saying 'it doesn't work', I'm saying 'maybe check if can be realized with that complexity'. You can't force everything, sometimes it's better to change or optimize some of the physical circumstances (driver, horn, adapters or, very important, the room!) to reach your goals. So far I don't think your path is unreasonable, there's a lot we all can learn. I just like to check from time to time if it's realistic what I do or if there isn't a more simple way to get to the goal. I don't know how critical the processing power of your dsp is, maybe it's not even worth a concern, I simply don't know, I don't even know what you are using. I just wanted to say, it's probably worth to check that first.

Yep, I ran into the same hardware limitations when I began playing with FIR.
Started with a miniDSP DA-8, which is probably still the best beginning hardware solution I know of. But its consumer I/O, just OK DACs, and desire for more taps eventually led me to a bank of 4 openDRC DIs.
With balanced I/O and could use my own DAC's, since they are digital in/digital out. That gave me 8 channels of 6144 taps at 48kHz....lots of play power.

Learned a ton with that platform. Achieved awesome sound with exceptionally smooth mag and phase on 4-ways.
But always sticking to the principles of acoustic solutions come first, and don't over correct to a single spot (because it's so tempting to make perfect looking measurements ;)

Currently I've moved to QSC's q-sys and its Core110f. Awesome platform for experimenting...you can lay out virtually any processing design you want. Tremendous processing power...even let's you embed a dual channel FFT's reference and measurement points anywhere in the signal chain you like.

Below is an electrical impulse experiment I am toying with, for if and when a measured reflection worth trying to neutralize pops up.
The impulse is a full range signal, mixed with a 1ms time delayed small 'notch-only' signal.
signal. I'm studying how FIR files combine, making a single FIR for the impulse shown, vs making FIRs for the individual signals and combining them in REW.

I don't think i could try stuff like this without the q-sys design flexibility, and awesome capabilities of REW and other measurement programs.
OpenDRCs are listed in Swap Meet (shameless plug) :)

So yes, I totally get what you're saying about scoping out capabilities before we start a project.
 

Attachments

  • full and 1ms notch.jpg
    full and 1ms notch.jpg
    54.9 KB · Views: 167
Oh boy, wish you hadn't told me that.....that's shows an incredible over concern with pre-ringing imo!
The previous 'notched' signal measurement was without any FIR.
It was just from normal dsp, albeit trying to do something a bit tricky.

Have you looked at impulses from pure soundcard loopbacks?
Here's a straight loopback from a rme baby face pro...I don't see its impulse as all that different from the 'notched signal' impulse, in terms of pre-ringing.

Would you say the rme impulse below has pre-ringing?
Hope not, cause if so, I gotta just shake my head..:confused:
 

Attachments

  • soundcard.jpg
    soundcard.jpg
    45.4 KB · Views: 169
@Mark, I believe preringing has been covered already in your thread:
Pre-ringing: Who has heard it?

Aside from the impulse display view being heavily weighted towards high frequencies and that we are discussing bandlimited signals, the end result using music is not audible: https://www.diyaudio.com/forums/multi-way/329993-pre-ringing-heard-5.html#post5604334

If there is more discussion to be had about preringing, perhaps continue in the above linked thread. Otherwise, please continue with measuring horn reflections :)
 

ICG

Disabled Account
Joined 2007
You can see it like you want but if the amplitude is twice as high and twice as long as the correction signal, it clearly has more influence on the sound than the correction does. Or in other words: If you can ignore the pre-ring, then you can even more ignore the reflection.

I can only say I've heard music on certain speakers where you actually can hear the difference (which is the pre-ring) between FIR and IIR filters.
 
@Mark, I believe preringing has been covered already in your thread:
Pre-ringing: Who has heard it?

Aside from the impulse display view being heavily weighted towards high frequencies and that we are discussing bandlimited signals, the end result using music is not audible: https://www.diyaudio.com/forums/multi-way/329993-pre-ringing-heard-5.html#post5604334

If there is more discussion to be had about preringing, perhaps continue in the above linked thread. Otherwise, please continue with measuring horn reflections :)

Could not agree more, Mitchba :)
 

ICG

Disabled Account
Joined 2007
If there is more discussion to be had about preringing, perhaps continue in the above linked thread. Otherwise, please continue with measuring horn reflections :)

So you both deny it's worth discussing if it is an issue or not? As I said, if the amplitude is twice as high and twice as long as the correction signal, it clearly has more influence on the sound than the correction does. So why is a signal error much higher not important anymore? If that's acceptable, then the reflections are a non-issue anyway, that makes then no sense what-so-ever.

Remove the rubble before you start vacuuming! :mad:

Okay guys, I'm out.
 
Thanks for that … a device I'd love to play with.

btw, I got to briefly meet Dave G. at trade show in the summer.
I told him how at least two of us were trying to crack his patents for our own DIY:D
And about using the 2 sections of a coax CD, one as source the other as mic to explore reflections..... he said with a grin, 'That'll work!'
Very nice guy.

But alas, I haven't done a dang thing down that path since last post here...
 
Administrator
Joined 2004
Paid Member
I wonder if HOLMIpulse would be a useful tool here. It allows you to choose the part of the impulse you want to look at. For example looking at only the impulse after the initial spike is as easy as sliding a few markers. I've done that to look at room or box effects.

You can also export just the portion of the impulse you want and use that in other tools.
 
So it sounds like you're saying it does more than just visually truncate the first part of the impulse.
If so, I'm guessing it must make some kind of recalculation that essentially gates out the first x amount of time.... ????
That would be very useful.

I was using a single sample dirac pulse as the test signal impulse, to be able to visually identify such a short spike, and then look for reflected response back into the second voice coil after that.

If HOLMIpulse could gate out the dirac pulse, it would really help.
 
Thanks.

I've never been able to resolve crashes with HOLM Impulse on a couple of prior PCs.
But since I just built a new Win10 Pro install, ground up, new license and all, I'll try again.

I was just hoping you might know how it works to give me some extra motivation :)
 
Status
This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.