'Perceive' Contruction Diary

[RyanC]

Ok a couple of things-

In cubase or Nuendo the delay incured by the process (plugin) is compensated for automatically- So if you have a steep filter on one driver and a soft one on another the difference in delay incured by the plugin "should be" handled for you. I do not believe this is possible with art teknika but either way it does not matter because-

I just setup a sine at the -6dB point and use the (free) voxengo sample delay to allign the drivers at the XO freq. I am alligning them at the listening position (That is what vil is doing too it looks like). I'm not really ready to drop the $ on a LSP so I don't have anything now that will give me a full phase allignment test, so this is all I have right now. But you could use my technique and then measure the overall allignment with lsp. Reguardless you are left with only the phase inconsistancies of the drivers/power amps if they are alligned at the XO f6. I do this by recording the signal back in to the computer from each driver- this way you can assure that you are not delaying it by a full wavlength or more to achieve allignment (and that you are doing it to the right driver).

This is as I see it the primary advantage of digital XO's you can delay any band (easily) to achieve the best inter driver temporal allignment.

I am not familier with the tact stuff- but AFAIK only an FIR process could tailor phase independant of FR within a given driver (It makes sense because only FIR can do one independant of the other in reverse). I have been begging for a parametric phase adjuster though, who knows, maybe soon. The curve EQ is phase linear so it does not effect phase of the signal. As is the case with the Lineq's. You could create an FIR XO within an XO to shift a certain part of the signal later in time- it's hard to say what that would do for you though as they would comb filter when mixed back together (so you would need steep slopes).

A couple of thoughts- you might be suprised how much better it sounds to use either a steeper filter or a lowQ parametric centered well above the XO freq to completely eliminate those modes from the seas drivers. At least on my sub drivers, TC2+'s, the difference was quite audible, but suprisingly not very measureable. This may not effect overall FR that much (due to the hi q nature of break up modes?? I dunno) but reguardless your driver does not sound good at 8k (I would guess by lookin at the FR plot) So it would be better (to me at least) to have it not produce any 8k.

Also my personal feeling is that dome tweeters (or any tweeters really) are rated for what they can hadle heat/xmax wise, not where they sound good + are low in distortion. I discoverd this by listening to just tweeters and adjusting the XO freq- I find that tweets in general tend to get very harsh at the bottum of their range especially in higher SPL situations.

Cone mids and mid-bass's produce a much preferable 3k-5k to my ears than tweets in general (now to find one with good off axis response). But then inter driver spacing becomes a problem. I can't speak for your tweets tho as i have not used them- and I don't have a ton of experience with the higher end drivers (yet).

Anyway yea- use lspcad, just don't get caught into the mindset involved in the inherinet limitations of passive filters as many of these are not present on a digi XO.

Good luck-

RC

[ShinOBIWAN]

I've had some playtime with both the Behringer and the Cubase setup.

Both offer fantastic quality, I'd give the nod to the cubase implementation regarding quality but TBH you have to sit and really listen to notice the differences without room correction. Turn room correction on and the cubase setup really moves up a gear. Everything is cleaner, clearer and very sorted.

I've got issues with both setups however. The behringer has no pre-amp section and the cubase with 3-way DRC, FIR and time/phase alignment is a massive CPU hog. Games stutter, frames are occassionally dropped in theatertek and it has a steep learning curve at first.

I really don't know which to go with actually. For all its evils the cubase XO is incredible. The behringer XO is good but without the DRC its lagging.

Because I feel the difference is very small between the cubase and behringer WITHOUT DRC. I think I'm going to go with a hybrid approach, let the PC do room correction and the behringer can do the XO. I still haven't tried this yet but I can't see a problem.

[mbutzkies]

It looks like you are going to need a quad athlon.

RyanC: The problem of delay is not with music, it is syncing video to audio, and the fact that video would get ahead of audio, you would have to timeshift your video for that. Hopefully you could fix with limited rigging. I don't use nuendo, but I guess you can play video through that but using a multi-track mixer would be an extremely cumbersome setup. If its all audio, you can just add additional delay to sync up like you did. The problem with FIR filters there are two delays, a windowing delay that will not change with hardware, and a processing delay which is hardware dependent. For simplicity the Waves LnEq has equal delay whether your Q=1, or 6.5, that is also why the eq is not completely adjustable.

5th Element might have the idea of just using two different crossovers.

[RyanC]

Cool yea-

Shinobiwan- The room correction is pretty damn awesome - I think if you are the "set and forget" type then the DCX or somthing like it is probably best. But for the endless tinkerer (me) I like the computer. The other thing to keep in mind is that the DCX runs on its own clock with no sync (it always runs at 96k) therefore it runs free. And some $2 SRC chip upsammples at an non multiple rate (EG it would be better if it ran at 88.2 for CD playback) and meshes the two clocks. Your performance now might only be a bit better- but if you bought a big ben or the antelope box, the difference would be even bigger due to the DCX's lack of proper clocking.

mbutzkies-

Right, but if you are inside the 20ms window you will not precieve the latency- Of course if you sit 10ft from your speakers there is already 8ms of latency due to the fact that sound moves alot slower than light. If you brain didn't allow for this sound would always be out of sync with sight, as light always moves faster then sound.

Reguardless that is why we are talking about 2 XO setups- a realtime IIR setup (I have mine down to about 8ms round trip) and a FIR setup for non realtime applications. This way when i am playing my keyboard through the system I use the IIR setup and when i am mixing/mastering i use FIR.

RC

[ShinOBIWAN]

Ryan,

I've got a friend interested in setting up what you did for me with those nuendo files. Problem is he's running a copy Cubase VST 5.

Would it be ok if I passed on your email so you could have a chat with him?

[RyanC]

Sure-

But Unfortunatly I don't have a ton of experience with v5. I was into sonar in those days. I don't know what the routing imitations are-

RC

[ShinOBIWAN]

Ryan,

He has access to a copy of Magix Samplitude 8 at his workplace.

I've told him that the Cubase 5 is rather old now - 5 years now I think.

Would this be better?

[Vil]

he can easelly get Console .
http://www.console.jp/eng/download.html
cheap and very flexible .

abuot $50 , and you can share that program with your friend .

[ShinOBIWAN]

Originally posted by Vil
he can easelly get Console .
http://www.console.jp/eng/download.html
cheap and very flexible .

abuot $50 , and you can share that program with your friend .

Vil thanks for that.

I saw your post earlier in the thread but I dismissed the setup as inferior to the cubase one.

I've had a play with the demo version and had an identical setup to the cubase one in under 5 minutes! The sound is identical too. I think Ryan said that SX2+ compensated for the latencies of plugins but console sounds no different to me.

I've also noticed that CPU usage is slightly lower

All in all I'll be building a dedicated PC for this XO and use console instead of cubase and bin the DCX

What's the best plan of attack? Do I move the RME into the dedicated machine and buy a prodigy or Revolution for the main machine. I'd like to keep the signal digital if possible all the way to the amps as I have done now with the DCX.
Really just tell me how you'd do it.

[Vil]

plugin delay compesation is not important , because you need individual regulated delay for every driver anyway , so you can adjust right delay time for every chanel. also automatic delay compensation somethimes works not good and you can have very strange things going on (thats my personal expierence with Steinberg products ergh)

yes it would be nice to have a digital signal up to DAC .
if you wanna use S/pdif or ADAT digital interface for data to DAC transmission , you will not have best possible result . those two interfaces are very jitter sensitive .I started with s/pdif few years ago and finished with 4 balanced pair interface I2S , transmitted using LVDS transmitters .
Also dont forget lowest possible jitter master clock oscilator for clocking your DACs and sound card .No way to use asynchronous sample rate conversion , especially with Crystal CS8420 chips , that one kills music . those from Analog (AD1896A) are better but still some compromisses .