i dont understand fundamentally how to use parametric eq
i dont understand fundamentally how to use parametric eq
This tutorial is a pretty good start. The controls aren't exactly the same as SAC, but the principles apply.
Basically, instead of having each EQ fader control a fixed 1/3 or 2/3 octave frequency band, you can vary the center frequency and bandwidth of the filter to achieve very wide or very narrow filters. This allows you to create EQ curves that you can only mimic on a graphic EQ.
SAC Host: Custom built i3 / Gigabyte based rackmount PC, MOTU 424/2408(2), Profire2626(4),. up to. on up to 6 monitor mixers.WinXP Home.
Plugins/Processing: RML, Antares, ReaPlugs. Recording with Reaper.
System Load - 25-30%, at 1x32
99% of the time, things that aren't being done aren't because they don't work. The other 1% is split evenly between fools and geniuses.
BE your sound.
When using pink noise, instead of "acoustically flat" I would rather go for "acoustically smooth". In other words, ruler flat is not the goal, rather a smooth profile, which could be curved. What I look for on the RTA is peaks or valleys that need adjustment, although some of that can also be from room cancellations or additions from all the sound bouncing around. RTA is only a marginally accurate way to adjust the system to the room, but the more capable stuff like SMAART and EASE can be quite expensive to the beginner or guy just learning time-based measurement software. So, many of us are stuck with RTA, but you have to know its strengths and weaknesses.
As most feedback comes from monitors (unless you have your mains behind your musicians! Ack!), you typically want to ring them out for feedback rather than the mains. A quick and dirty way to do this is to briefly (and quickly) point the mics towards the monitor wedges as a singer might do and observe and notch out the frequencies that feed back. For stand mounted mics, you can turn them up until they feed back and notch them that way. If you *are* getting feedback from your mains, try to re-position them to reduce the likelihood of feeding back and, if necessary, carefully notch them for feedback as well.
You should also learn what frequencies are expected from each instrument and experiment with the channel EQ to cut out those frequencies that are not important to the instrument, or unpleasantly mix with other instrument frequency ranges to muddy things up. I.e., piano, vocals and guitar occupy similar frequency ranges, which can sound very muddy if not separately EQ'd with the channel EQ to bring out the important tonal ranges of each instrument while de-emphasizing those ranges that subtract, rather than add, to the overall tonal mix. This can also increase your gain before feedback as when you reduce the available frequency range for a particular mic, it can also reduce the likelihood of that frequency to build up and feed back.
Channel gates are also a good thing to get a handle on as they will shut the mic signal off when no signal is present, thereby reducing the amount of open mics on stage at any given time.
HTH,
Jeff
7th Voice FOH/Mon/Sys Tech
www.7thVoice.net
www.reverbnation.com/7thvoice
SAC Installation:
Dell Optiplex 990, Intel i5/4GB RAM
3xMOTU 2408, PCI-424x card, 6x ART TubeOpto8, 2x Motormix
SAC Portable Rig:
IBM ThinkCentre M50 3.2GHz/4GB RAM
2x MOTU 2408/PCI-424x card, 2x ART TubeOpto8, 2x M-Audio Profire 2626
Yamaha Promix01 controller, EWI TourcaseCUDJ-P-22.
I use Bob's EQ on the Front end. I use SATLive and have generated an EQ curve derived from dual FFT measurments. Feedback from FOH is hardly ever a problem as I always set up my FOH Mains ahead of the Band. Feedback from monitors is handled using either SATLive or Bob's Frequency Analyzer plugin on the monitor output, notched with Bob's EQ as well.
I am going to do another measuring session in a week or 2 to play with the Sub / Mid/ Hi phase alignment. I know I can do better now that I have a bit more experience with SATLive.
-SAC,SAWStudioLite,Midi Workshop,SATLive, Reaper
-SAC Host (24 channel): Various Laptops via Digiface into APPSYS Adat extenders into (3) ADA8000,(2) BCF2000 controllers, 1x64 resolution
-SAC Host (32 Channel): Intel Core 2 Duo E8400 3.0Ghz , 4 Gig DDR/800mz RAM, ASUS PK5PL-CM MotherBd,XP Pro SP3, RME Raydat, (4) ADA8000's
-SAC Remote: Various Tablets via AMPED Router
-SAW Host : Asus Laptop, i7 12g RAM
For the money REW (Room EQ Wizard) from the hometheater group is hard to beat. http://www.hometheatershack.com/roomeq/
Find a calibrated mike on ebay (usually for less than 50 bucks) and since the software is free your cost is minimal for a pretty capable program. What I do is loop around one of the adat outputs to an input and then choose one of the input channels that I looped around as the output from REW, this way the output can have an EQ on it to feed the room. There are many reasons why I like this method but the highest are that the EQ can be copied to the main outs once you determine what the best curve is, the EQ can be easily adjusted for the smoothest response from the output (instead of trying to set on the input side). REW has impulse response which is way more accurate for setting a system that just RTA or pink noise. It also has a special option just for setting subs. Both RTA and pink noise include resonances and reverb trails neither of which you can do anything about no matter how much you chase it in EQ. If you smooth out your impulse response curve that is the best you can hope for in a room.
+1 on Room EQ Wizard.
Yogi, maybe you know how to do what I've been trying to figure out...
Can I use REW to find delay times for speakers? I've been using pencil and paper and a click track for years now. Lately I've been using a laser measure to find the distance variation between main speakers (or source from the stage, etc...) and the speaker I wish to delay. That plus using a click and my ears gets me very close.
I know some other packages like SMAART will do this for you. It seems like I should be able to figure something out like that with REW, but I've only tinkered with it a few times so far.
If you have any suggestions, that would be great.
I guess I should just search the forum as well.
Richard B. Ingraham
RBI Sound
http://www.rbisound.com
Email Based User List: http://tech.groups.yahoo.com/group/sac_users/
No, but I have a method that seems to work really well and you can do with any DAW that can measure time.
Set up the mic in the listening area of the delay stack, play a single sharp click or pop, then measure the distance in time between the peaks on the waveform in the DAW. There's your delay time. I very rarely have to tweak much with this method. I use Audition for this task primarily as it's very easy to highlight on the waveform, but I imagine most DAWs would work as well.
SAC Host: Custom built i3 / Gigabyte based rackmount PC, MOTU 424/2408(2), Profire2626(4),. up to. on up to 6 monitor mixers.WinXP Home.
Plugins/Processing: RML, Antares, ReaPlugs. Recording with Reaper.
System Load - 25-30%, at 1x32
99% of the time, things that aren't being done aren't because they don't work. The other 1% is split evenly between fools and geniuses.
BE your sound.
+1
RTA is much better than people using a 58 and repeatedly saying "yup" .."check 1,2,3" as a method to set EQ. There is still a huge world of people out there that still do this because they cant afford or understand the tools. I take multiple measurements with Smaart throughout the venue to get an better profile of how my speakers are interacting with the room. To some people, a BBE sonic maximizer is the only time alignment tool they know.
Well,
there are a lot of different approaches to FoH equalization.
I know people doing it perfect using a SM-58 and saying something.
An RTA can be very usefull for finding feedbacks, but it is no good choice for EQing, because it is 'time-blind'.
Dual-FFT takes care of time, so their results are much closer to the human hearing (which is not time blind).
Tomy
3 * TIO1608 + AIC-128 + X-Touch + Dante -> AES + DADC-144DT
SATlive is my measurement software
DIN 15905-5 (German SPL Limit)
I agree, no argument. I have seen very very few with golden ears that could set the system well with the aforementioned method of using just a mic for EQ only. I do something somewhat similar to using a click signal to listen for time if I'm on someone elses rig.
The original OP was stating his lack of understanding in the use of a parametric EQ. Most everything in this field is built off of having a good understanding of how to use the tools that are available to us. My feeling is that if he doesn't know about how to use a parametric, he also won't have much understanding yet as to how a dual fast fourier transformation analysis can help him. I'm not picking on anyone here, just asking. How would you suggest someone with ,say , a pair of music store speakers setting on top of some subs powered full range off 1 or 2 amps get a good jump on setting their EQ ?
Thanks -Eric
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