image description


Forum Index » Profile for chrisatrational » Messages posted by chrisatrational
Author
Message
32.4.2AI » UC - Room Analyzer Question » Go to message
sjc193 wrote:

I had written up a big explanation on this but somehow hit a button that refreshed my screen and I lost it all


You know Steve, I was about to write an explanation why we only measure one system at a time, but I stopped myself. If you aren't using Smaart v7 or v7 Di, you can't measure what the differences are with VSL (namely time variance), and being that the frequency smoothing in VSL is 1/3rd octave, it is difficult to tell the exact frequency response, other than an approximation.

I will attempt to boil it down for those who are curious:

In order to generate a transfer function measurement, we must compare the reference signal (the noise coming out of the console) to the measurement signal (the RTA microphone picking up the system) and compare them in time with each other. This requires a delay offset for the reference signal. The delay is set internally and timed to the HF energy coming out of the system (you would see this in the Live IR and Phase trace of Smaart). If there were multiple systems on at the same time (think left and right simultaneously), smaart would see this as two distinct arrivals and would time itself to the system the microphone is closest to, however the interference from the system that is not in time with the analyzer will introduce interference that will through off the measurement.

This is a known fact in measurement world, and why we measure systems one at a time. The goal of aligning PA systems is to take multiple systems and make them one cohesive system. Measure the individual components and add them together to create a cohesive sum of all parts. The idea is that system a + b +c = system A.

Feel free to poke me whenever if you are more curious about this stuff.

Cheers!

- C

32.4.2AI » UC - Room Analyzer Question » Go to message
Hey GT - the Main output is stereo linked. Try using two subgroups as your main output, this way you will be able to have separate control over their eq, level, and delay.

StudioLive General Discussion » sub's off the mains. EQ? » Go to message
With the aux send you can adjust exactly how much of whichever source is being sent to the subs, rather than on/off and master level (you use the aux output knob as your master instead when feeding the subs with an aux)

I think subs on an aux are a bit more flexible, as well as musical. This way you are actually mixing the subs - not just turning all instruments to them up or down.

but don't take my word for it, give it a try on a couple different gigs and decide for yourself.

FWIW, When using Presonus gear I only put subs on an aux when requested, I like everything on the main fader and prefer channel eq/HPF to get rid of any low frequency energy I don't want reproduced. the bigger consoles I have a dedicated mono out, or center output, which I will engage on channels I want in the subs, and again, I'll use channel eq to do the fine adjustments. do whatever makes your shows works and sound best for you!


- C
StudioLive General Discussion » sub's off the mains. EQ? » Go to message
Because the subs are powered, you don't need to worry about HPF/LPF, they have their own internal processing. It sounds like you want to do the "subs on an Aux" thing. Just assign an aux out to go to your subs, which should then be daisy chained together. Mix in the instruments you want in the subs to taste. You can get into analyzing and whatnot, but for your first try, I would just stick with getting the signal flow right

Often times people put a little filtering on the 'one note subs' ie -3 db at 50 Hz if that frequency gets a little crazy. for your purposes now though, I would just 'turn em and burn em'.

- C
StudioLive General Discussion » Save EQ after SMAART Room Analysis? » Go to message
Hey kgbak -

There are two reasons why I do this, one is that I want control over left and right, the other, is that I don't want the headphone output to be subjected to a system eq.

So,I measure usually the left side first, apply an Eq, copy that to the right side and then run the analysis with the eq on the right side and see if works out, if not I'll adjust the right side para eq accordingly. Then, channels all get assigned to sub group 3/4 for the main PA, as well as the main out for my headphones/cue.

I was in another thread trying to deal with the issue of keeping in ear monitor eq adjustments separate from any eq adjustments going out of the mains


Are you using the aux sends for the in-ears? if so, each aux out has it's own parametric and graphic eq (my 24.4.2 does anyway, I'm not up to date with what the smaller desks do). However, this is a global thing, and I'm feeling like you want to be able to adjust channels independently from the main out and aux sends. Channel adjustments made will be heard wherever they are routed to, that's where having a second monitor desk and engineer become interesting because you can tailor channels for the musicians to hear, separate from the mains.

FWIW, I know of one product that can have a different mixer for each system; Air Console, AKA SAC (Software Audio Console). These systems are pretty diy and take a good amount of hardcore nerding out to put together, but when you have it, you essentially get something like 24 simultaneous mixers all at once. The possibilities are pretty cool as you can have virtual separate mixing consoles for each monitor send, each with their own channel eqs, effects, etc. I have only attempted this a few times, and for one off stuff it is always way too time consuming. As you can imagine, having to get a kick drum sound for 5 mixes, plus FOH, then snare, then HH, Toms, Bass, GTS, Keys, Percussion, etc etc etc etc.

I hope this was some help to you. I think it's safe to say that most of us get the sound right in the house system, and make adjustments for the house system. Although we send the same signal to our musicians on stage, if you are using one console for FOH and MON, the MONs get what they get, and often times for me that just means 'don't let them feedback' and 'don't let them interfere too much with my house mix'. If a musician is giving you a load of crap for their mix and it sounds good in the house, take a listen to what they are hearing and try to get the sends to their mix more balanced, and help them out with applying a nice EQ on the aux out to get rid of any harshness/boxyness/whatever. Also, definitely try compressing their mix to give a more polished, smooth sound. You don't need too much, a little will go a long way.

Cheers!

- Chris
Live Sound » Outdoor rig recommendations? » Go to message
Sir Melvis Bacon, Knight of BaconHam Palace, is right on. What may work well indoors will be terribly insufficient outdoors. For 300 people outside, 2 double 18's a side would be very practical. On top of those, you can get away with two trap cabs well splayed, you can delay the outer cabs by a few ms to create a more cohesive horizontal array. However, to do this alignment, you are looking at getting Smaart Di to do it accurately.

This is a daunting initial investment, especially if you are buying new. You may want to look around, make some calls, see if there are any companies around that you can rent from or purchase some used but usable equipment. Also, If you can rent any part of the system until you can afford to buy, you will be better off and provide a higher quality experience. Bigger shows need bigger rigs, more is more. Bigger rigs require larger amounts of power, so as Sir Melvis Bacon stated accurately, there will be other pieces of gear to make your speakers a system. (AC distro, cabling, cases, transportation etc)

just a little rant - It drives me crazy when my town puts on shows in our 'green' and hires somebody that doesn't have the gear to do the coverage, resulting with too few speakers being pushed too hard to sound good; leaving people in the crowd grimacing and struggling to hear anything clearly. Please don't be that guy.

Good Luck!
Live Sound » Smaart Room Analysis not working » Go to message
Gadget, I often wish I could 'like' some of your forum posts. We keep it modest here I guess...

StudioLive General Discussion » Question on setting up SMAART » Go to message
RichR wrote:

I have the HPF on the 312AI on. When setting up this way and using SMAART, in two different rooms so far, SMAART is cutting a ton of low end on the PEQ.

Is this the correct way to set up the speakers when running SMAART, or should I just run to the top (with HPF off) for the room analysis, and then hook up the subs?



Think of the transfer function magnitude trace which smaart generates as a general representation of what the frequency response of the system is at the measurement point. in other words, 'what goes inta' and what comes outa' the system - Smaart isn't cutting any low end. Actually smaart isn't doing anything to the system at all, it is just showing you what you got. You can screen shot the trace with the HPF on, and an again with it off, and see what that filter is doing to the frequency response.

For what you are doing, you are correct in your set up. The HPF is meant to be on when the subs are connected inline, that's standard for any system.

Have Phun!

- Chris
24.4.2 » SMAART unavail when configured w FireStudio » Go to message
AFAIK the Smaart features are only supported in the SL desks, regardless of the order in which they are connected.
StudioLive General Discussion » Question about SMAART » Go to message
I feel your pain, and often have to deal with l/r not sounding the same when I fire up a rig. This is one of the things that Smaart excels at, being able to measure how different (or same) systems are. You can do a near field measurement to get more of a direct idea of a speakers FR.

Also, try running L/R from two subgroups so you can have a fat channel for each. Run the wizard on both and see what is going on frequency wise. If you suspect a polarity reversal, play both at the same time, reverse the polarity on one side - if the system gets louder when you reverse the polarity than you will have found a fault. The VSL Smaart cannot tell the difference with polarity as it doesn't effect the frequency domain.

With regards to your console and possible solutions - the L/R bus share the same fat channel - and are essentially linked aside from the bus panning. In my experience if the L/F pa doesn't sound the same, and I know my console processing is off, than cause is outside of the console. Polarity could be an issue, but it would result in a loss of level when combined more than frequency issue. Other issues could be amp channels not set the same (if powered speakers), or if active - different presets on the speakers or different parametric/graphic setting in the system processor, or malfunctioning drivers (if you are using more than one speaker a side, it may not be obvious at first. It could have even just been physical placement and interaction with boundaries.

StudioLive General Discussion » SMAART does not detect enough audio and fails » Go to message

My best suggestion would be to return to a previous OS; as with any audio application, often the latest and greatest OS's render the applications we rely on useless. As an audio guy, being an early adopter is most often not in your best interest - and to be avoided whenever possible. I'm still on iOS 6 and mac OSx 10.7.5 and plan on staying here for quite a while longer!!

- C
Live Sound » live vocal reverb on 16-0-4 » Go to message
try using a hall type, 1.5-2 secs with a long pre-delay, like 80ms or more. Makes the reverb sound more like a subtle, well controlled delay, adds length to notes without being obvious. and don't forget, EQ THE DELAY. you have a fat channel on the fx engines, take advantage of that.
StudioLive General Discussion » Capture - Saving Sessions » Go to message
I created a new session in capture, under the "Capture" root folder that the application creates in my documents folder on the mac HD. When I saved it, I named the session the name of the song (killing grounds), but in the capture session it says "Killing grounds - Barley Hoppers". The Barely Hoppers are the band that I recorded in a previous capture session, the band that wrote Killing Grounds is The Brotherhood Of Thieves. I can't figure out how to get rid of barley hoppers in the session file name on the screen. I'll attach some screen shots that I hope will bring better understanding to what I'm trying to say - even I'm getting confused as I type this!!

Does anybody know what I'm doing wrong? When I go to save as, it just gives me the option to save the song name, not the band name as well.


Many Thanks!!!!

- C
Live Sound » SMAART microphones » Go to message
Thanks for the kind words Steve!!
Live Sound » SMAART microphones » Go to message
you can use pretty much any microphone for delay alignment. The software is just timing the peak arrival and that doesn't require anything special. In fact I've used an sm58 to time delay systems when it's really windy and they are far away, since the polar pattern can be of some help rejected noise. However, I wouldn't use a 58 for anything involving eq adjustment/tuning.

A pencil condenser can be a pretty fair substitute, some are really quite flat. Just make sure you don't have any high pass filters on. in general, any low-cost meas mic (the offereing from dbx, behringer, audix, RTA420 (Rationals budget mic), Presonus, etc) are all the same type of capsule, and offer similar-ish responses depending on how long ones been in the sun, or in the cold, or bashed around etc.

The smoothing algorithym in VSL is so high that you won't be able to see the fine detail of the differences of the microphones like you can in Smaart. Also, yes there are what is called 'mic correction curves' which are supplied by most mid to higher end microphones. With Smaart, you can use the correction curve to calibrate the input of the microphone, essentially making the microphone a truly invisible probe, for transparent microphones. This is a super user feature, basically, if you've got a decent mic and know what to look for - you are going to get results that are just great. I keep a cheap mic in my kit as well as some nicer ones. When a RTA420 tomahawks I go "darmnit", if an Earthworks m30 does the same, my response would be much more....intense... Luckily I haven't had that issue!!
 
Forum Index » Profile for chrisatrational » Messages posted by chrisatrational