1) I noticed especially with acoustic electric guitars that sometimes a thump noise is introduced into our system... It appears to come from the guitar being bounced on the players knee (if sitting) or maybe the hand is striking the body of the guitar while being played... This thump is coming through the subs (must be below 128 Hz) and for me this thumping can be very annoying... Last night I was very tempted in enable the high pass filter cut (100Hz and below) on that channel during the performance but decided not to... The problem is...I don't know what the frequency range is for acoustic guitars,...and I don't want to filter out any legitimate tone?...
2) When dialing in the input channel gain (trim attenuator) for microphones and instruments (using PFL) I'm uncertain on what the preferred or target peak point on the meter should be?... Is there a \"best\" or \"tried and proven\" procedure for setting up the input channels before a performance?...
Due to the Memorial Day weekends our Sunday night youth service was very small and calm last night... The crowd was only about half of normal size and the makeshift praise team only consisted of my son (acoustic guitar and vocals) and another young adult (vocals)... Let me tell you...this kid has a vocal range that is unbelievable and is slated to audition for next seasons American Idol… Anyways,... this worked-out great for me because I used the replacement mixer (live) for the first time... Last weekend the youth put on a drama in the main sanctuary... As luck would have it...the house was jamed packed with standing room only on the evening when the other mixer crapped-out...
My son Daniel playing his graduation gift and our \"future American Idol\" (I hope) were jaming last night...
Guys,...needless to say,...I'm a very blessed man and proud father!...:D
With a system capable of good low end reproduction anything you don't want in the subs can be eliminated with the aux fed sub method. This method however requires an additional crossover channel and we don't have that here. So how do we limit the amount of LF in the mix so that we don't end up with a muddy hollow mix? Even in the Aux fed method you would engage the HPF, and use the LF frequency control where additional cuts are necessary for that band. We have already established that each channel is capable of full frequency response in our super duper gain structure and system tuning. Now consider that every single open mic is not only hearing the full spectrum it is being subjected to, but is also \"feeling\" it...the vibrations that are induced through the floor manifests itself as rumble and LF howl and can really muddy up a mix. Multiply that times the number of open mics and you can see that if we don't do something about it we will end up affected by it.
Here is what I would suggest... The only instruments that REALLLY belong in the subs are:
1. Kick
2. Bass
3. Keyboards
4. some floor toms perhaps
therefore engage the HPF on all channels that are NOT in this group! ESPECIALLY vocal mics. Because a source has the capability to do a thing, does not mean it should be allowed to do a thing... Like, because a sub is capable of 25 hz doesn't mean it should be allowed to do it in all instances. One very good argument here is that if the amp isn't capable of powering the subs properly they shouldn't be allowed to have the lower registers.. There is very little usable info in this region anyway, and ends up wasting power in useless and sonically displeasing rumble and speaker luffing. (where the speaker pumps but does not reproduce any musical information) In a reverberant space the usable info below 45 hz can cause room loading and modes that can degrade the intelligibility of the mix. Your mixer has a 100 hz HPF.. it also has an 80 hz shelving control (a \"tone\" control fixed at 80 hz that allows cut or boost of that frequency...BTW that is the typical center frequency of the first harmonic of the kick drum) you also have a swept lo mid that is capable of 35 hz. This is very nice, and can be used as a swept HPF if no LF tayloring is needed... in fact with a swept HPF and Swept LPF I could be very happy with a properly tuned, good sounding system...this mixer has that capability...if you ignore the Hi and LO shelf's.
So. with vocal mics, engage the HPF, turn the 80 hz shelf down fully and if you still have unwanted LF in the system use the LM controls.. turn the LM gain down and sweep through the lower frequencies till you find the offending point then work the gain control to set the amount of that cut that you are using. Remember... even though we have the 100 hz HPF engaged, the frequency only starts falling off at that point and is probably in the area of 15db per octave.. so there will be content in the 80 hz range even with the HPF engaged. If you want to add additional warmth to a voice on a given channel on quieter passages, use the 80 hz shelf or the swept LM to do so.
Now, channel gain setup using the PFL ... Yes this mixer has a 4 ladder channel meter and is useful for monitoring a given signal and can indicate if a signal has increased during the performance ( geez I can't hear myself .. I'm gonna crank up my amp!) but the master section has a 12 segment meter that is inserted when the PFL is depressed. Set the channel main fader at the design center...the -0- right next to the LR button that routes that channel to the mains. with the signal present set the input gain for that channel to average about the -0- point.. a 6 db overshoot will still give you 10 db of headroom..(plus a little more actually to the point the mixer amps flatten) and should ensure sufficient headroom while maximizing headroom. This PFL is also great for monitoring individual channel levels against the overall mix, and can help diagnose a missing signal, or errant peak that is eluding you otherwise...
The percussive sound you speak of on the guitar is treated with the same methods we spoke of above...an acoustic guitar should have it's low end treated in such a way as to incorporate fullness but still not promote howl and feedback...this usually means setting the HPF, and possibly further tailoring the LF with the LM control.
BTW if you are using the FOH mixer as the monitor feed be sure to use the \"PRE\" position an all sources sent via the aux sends. That way you can make adjustments to FOH that won't affect the monitor mix...The \"input gain control\" is however before the \"PRE\" buttons so don't adjust the \"gains\" after you have set the monitor mix! VERY IMPORTANT!!!!!
All for now gents.. I have sheetrock to put up! Doesn't that sound like fun?
I understood everything you wrote (absorbed it like a dry sponge) and you told me exactly what I wanted to know!... I can't wait for the next praise team practice so I can set this thing up properly... Speaking of the "Super Duper Gain Staging" I am looking forward to your critiques of the procedure I PM'd you... I understand it (I think) very well up to the DriveRack and after the DriveRack... I'm just uncertain about the correct method or sequence inside the DriveRack itself... Please straighten me out on this and let me know your method for setting it up inside the DR...
I'm planning on setting up two programs inside the youth DriveRack... One will be for live music as it is now and the other will be for recorded music (DVD's) to use for "Friday Night Flicks"... I'm thinking this program will be identical to what I have now in regards to the GEQ, crossovers, delay, and limiters,…but I don't see the need for the AFS, notch filter, and PEQ's... What are your thoughts?...
You realize that the quality of your sound is moving up when your system is so clean you start to notice and are able to distinguish these types of adjustments are necessary! This is total validation that you are significantly improving the sound.
Gadget -
Excellent article!
I have one more thing from my experience which may help clean up the low end stuff that Kevin is starting to notice.
In a church I was setting up, they had crown CT1200's and on the amp there was a selectable 35Hz or 70Hz filter. This turned out to be an absolute boon for getting rid of the muddy sound we were having every week due to unnecessary stage monitor artifacts.
I noticed that on Kevin's amps you could do the same thing at 30 or 50Hz. I think that may help to roll off only the amps powering the stage monitors.
We actually saw this make a difference on our RTA at the church as we ran pink before and after we set the amp filters.
Hi guys...
Thanks we'll get into some other mixing techniques later as time permits.. That one is the biggie though...Oh and Wolfie.. Sorry I wasn't available earlier, call when ever you can...Oh and I believe that was one of the first things Kevin and I did... set the amp filters.
Kevin, I will respond to the PM soon.. I have been very busy working on re-doing the old studio. I have mangled my hands to the point that I can barely type though...
I noticed that on Kevin's amps you could do the same thing at 30 or 50Hz. I think that may help to roll off only the amps powering the stage monitors.
Hello Wolfie,
It's great to hear from you and I hope you had a great weekend!... The amps for the flying mains (JRX115s), floor mains (JRX125s), and monitors are set to roll off at 50Hz and the amps for the subs roll off at 30Hz...
Kevin, I will respond to the PM soon.. I have been very busy working on re-doing the old studio. I have mangled my hands to the point that I can barely type though...
Gary,
At last!!!... Maybe I can offer you some "sound advice"!... HIT THE NAIL with the hammer!... If done properly you should hear a distinct 500Hz ping...instead of a painful 30Hz thump!... And the vocals might punch thru cleaner!...
After studying the mixer literature again...it appears in the FOH mode I don't have EQ control (via the mixer) over my monitor feeds... If this is true...then what would you recommend (with current set-up in mind) in regards to a decent 2-band graphic equalizer... It would be great to have AFS as well...but cost is a major consideration... I have four monitor feeds and it would be nice to be able to low cut and/or to EQ down frequencies as needed... I'm thinking the DBX 215 (X2) might be sufficient for this...but I would love to get your opinions...:?:
After surfing...it appears that Mackie is currently the only manufacturer of a quad-band GEQ...?
Well, It appears that EQ 101 is in order.. class is in session...
First an observation...
\"Using a 15 band EQ is like carpet bombing where a sniper rifle would suffice...\" Not my quote but as usual borrowed without permission and since I can't remember who said it .. without credit as well... but a very accurate statement. I do know it was on the PSW 'LAB' forum though...
The ISO centers on a 15 band EQ very from 60-6000 hz covered by a single fader... that means if (for instance) you were having an issue with the first harmonic of the kick drum, you would have to cut both the 63 and 100 hz faders effectively cutting frequencies from 40 to 160 hz! ALL those frequencies and all we wanted to do was cut 80 hz...We would probably have had to do more cutting than if we were just treating 80 hz also because of the nature of the location within the filters location and the nature of the slope on each adjacent band and the fact that 80 hz falls between bands... NOT good for the sound... not good for the feedback elimination...
A 31 band EQ is better... but still has the problem of ISO centers that often don't fall on the feedback locations and cut way more than is necessary to address the feedback. This is often the solution, but there is a better way... Lets examine fully perametric equalization.
In fully parametric EQ's even though you typically have fewer bands you can do more and be more precise in doing so. A parametric EQ allows you to zero in on the frequency you want to address (with digital eq's to within a few hertz) and allows you to apply a very very narrow cut or boost all the way up to a nearly 20-20,000 hz cut or boost.. with a single band...So with a PEQ you can zero in on the problematic frequency...start with a narrow cut and adjust the \"Q\" (amount of the frequency spectrum you want to affect) till the problem is addressed with the minimum amount of invasiveness to the frequency spectrum.
I typically monitor the FBX screen to identify feedback areas and adjust them with a PEQ on the fly...the FBX screen gives you the frequency in hz the feedback occurred at and shows the amount of cut and \"Q\" of the cut so it's simple to mimic with a PEQ channel.
A variable high and lo pass filter can be a great asset in an EQ as well... and can take actually add many bands of effective EQ in a single knob.
Ok for the sake of argument here I will add a potential cost effective solution.. evebn though it is an off topic piece...If DBX had such a piece I would suggest it.. but alas... http://www.zzounds.com/item--BEHDEQ2496
check out the feature list... and yes.. it's disposable should you have any issues out of warranty but hey I have not had a failure in any of their stuff I posses yet...
Kevin, do you currently (with new board) have feedback issues at any monitor position? If so, which ones and why? Dra
Hi Dra,
Since we learned how to correctly position our monitors (straight ahead towards the vocalist) in regards to using the SM58 mics we have not had any feedback issues... Thank you Gadget!... But our minister wears a countryman headset and he wanders all over the place while he's talking... I currently turn the monitors all the way down just before he speaks... I only mentioned AFS because it might come in handy for one of those forgetful moments... At this point it would be more of a "luxury" than "necessity"...
I'm thinking by EQing the monitor feeds I could low cut some of the lower frequencies (possibly 100-125Hz and below) and possibly get a more pleasant bass response overall?...
Ok for the sake of argument here I will add a potential cost effective solution.. evebn though it is an off topic piece...If DBX had such a piece I would suggest it.. but alas... http://www.zzounds.com/item--BEHDEQ2496
check out the feature list... and yes.. it's disposable should you have any issues out of warranty but hey I have not had a failure in any of their stuff I posses yet...
Holy Moses Gadget,...It looks to me like it would cost approximately $500.00-$600.00 to use two of these for our four monitor feeds?...:shock: I might be able to sneak away (and escape a flogging) for about $300.00-$400.00 maximum... You got any other suggestions?...
My primary reason for EQing the wedges would be to eliminate any mud in the system... My secondary reason might be for feedback elimination...if needed... Please let me know if I'm on the right track, if that's reason enough,...and if/or I have my priorities screwed-up...:?:
Just to let you know...I copy and paste your lessons to word documents because I think they're invaluable...:brown nose: Oops...they need to add another emoticon...:D
Laugh if you want.
Option #1. Find a board that offers 4 band channel EQ's (preferably with mid and low sweeps) and order all the pots in the eq section. Heck, get the filter switch as well. Build an outboard unit yourself.
Option #2. Find (on EBay or whatever) a mixer with 4 sub groups and use the channel EQ's, etc. and put it in line from your aux outs.
As a stop gap you could also see how EQ would work by running your aux into an open input channel on the board and use that EQ to see if it does what you need before you invest more $$$.
As a stop gap you could also see how EQ would work by running your aux into an open input channel on the board and use that EQ to see if it does what you need before you invest more $$$.
My two bits.
W
Hello Wolfgong,
Great idea!... But will that work?... Has anyone here tried this before?... I guess I could use the "line in" on the input channel but how would I get back out?... Ohhh well...back to the manual...:)
I have \"line in\" and \"direct out\" for each channel... The direct out is pre-fade (following the pre/post-EQ setting)... Hey,...this just might work!... I will have to study it a little more...maybe call Allen & Heath tech support to verify... Want to be certain that the line level/impedance of the aux outs matches the line in and direct out of the input channels...:)
If everything looks good then I can dedicate four channels to monitor feed EQs and have 100Hz HPFs built right in... If so it would be nice to be able to disable the phantom power and polarity switches on those channels... Those puppy's will fry the AUX feeds...:?
After some more reading I would have to change an internal jumper on those four channels for the direct-outs to be post-EQ...:)
My main concern about trying this...are the phantom power and polarity switches for those input channels... I don't like having a disasterous situation lurking around in the background just waiting to happen...:twisted:
I'll call Allen & Heath tomorrow to see what they can offer...
That board is amazing! I give you $250 CASH and a used 12\" Black Widow basket for it. OK, I'll throw in a handful of rack screws too. When do you want to meet?
Tomorrow evening we are kicking-off “Friday Night Flicks�!... We didn’t get to it last Friday because our youth minister had something to tend to… In preparation and in addition to my feudal efforts as a “perfectionist� I decided to set-up two separate programs in the DriveRack… I ‘m using one program for “live sound� and the other for “recorded sound� i.e. CDs and DVDs… Currently the recorded sound program is identical to the live program with exception to AFS and notch filters (NF is probably not needed in the live program either)… Last night while I was at the youth center...I wanted to check-out and fine tune (I should say...play with) the system a little more…
There’s something about the new mixer that has been bugging the crap out of me… It seems like I’m not getting adequate gain out of it; in comparison to the old desk,… This makes no sense to me… I decided just for heck-sake to follow the “super duper gain structure� process one more time…and only if it puts my mind at ease… Gadget,…I’m very sorry but this is why I have been persistent (or irritating) in trying to verify this procedure with you… Anyways,…somehow I missed something on my last attempt… I’m pretty certain it’s that I didn’t enable the AFL on my master channels on the previous effort… Because of this I was 3dB short of barely bouncing the yellow on the output gains of the DriveRack… So this is what I did: I kept the input fader at unity (in the DriveRack) and I raised the master slider up by 3dB… Obviously this made a huge difference overall!...
Anyone,...was this the proper approach?..
Since I was the only person in the building (and I really wanted to play with the toys!) I decided to try some other things… Just out of curiosity I wanted to see if I could fine tune the timing delay between the mains and subs by putting-up tones (at the crossover boundaries) and looking for the peaks using the RTA mic… Bad idea!... It doesn’t work...even when I increased the crossover overlaps between the mains and subs… I’m writing this one off due to the unique frequency response characteristic of each speaker(s)… For now it’s definitely going to be an ear thing!... But hey…I learned a little more about this stuff…in how it pertains to the “live environment�… Now maybe if two channels were used with respective tones that are on the center of the overall response of each speaker?…
While listening to a variety of tunes (loud of course) I decided to mute groups speakers so I could evaluate each one individually… I found that certain sound tracks would drive the subs (more prevalent on the right channel) into this annoying fuzz/buzz like sound at the peaks… After trying different things I found that the limiter was causing the problem… I disabled both the “peak stop+on� and the “auto on� and the problem vanished… With the limiter enabled it still did its job (but without the fuzz/buzz) even when running the system wide open…
Anyone,...what are your thoughts on this?.. Was this the proper approach?.. Even though both amplifier attenuators are dead-on 26 should I also back-off the right channel just a tad?...
The highs and mids are incredibly clean and crisp but I think they are overpowering the low end by a little… I lowered the crossover gain from 0dB to -1.5dB on the flying mains and from +3dB to +1.5dB on the floor mains… To my ears (maybe a bad idea) this was a notable improvement… I must say that without a doubt: the floor mains adds a \"fullness\" that really ties this system together!... The two subs are 8 ohms and driven by two amps in bridge-mono… So here’s an incredibly stupid question: If two more identical subs were placed on top of the current ones and wired in parallel (4 ohms)…what would I realize in regards to low end response and on this system?...
Master,…would I not increase those deeBees (at the least) by six?...
For now I think the system is probably 99% optimized for “recorded sound�.. I still need to do some homework and setup the PEQs for “live sound�… I have been very fortunate in not having a feedback issue except for one time…when our minister sat down on the stage right beside a wedge…:oops:
ADDICTED! thats what you are! You will never be able to walk into another place with sound production without RIPPING the system apart! :evil: You guys are all making me feel like a very proud papa 8). You listen, you, you learn, you take the information and run with it, and help others too! Very good gentlemen very good. You have in the process taken some of the burden off me and I appreciate that! I am very busy with things right now and so its good that you can keep the information hi-way flowing :!:
Ok, the only dumb question is the one that doesn't get asked :shock: A mistake is proof positive that at least an effort was made :oops:. Kevin, I have a live preset, and a DJ/ Karaoke preset. You need a whole lot more headroom with live music so it's a very good idea to have it that way.
When dealing with line level sources...there is no \"right and wrong\" way to do it.. there are better ways, there are ways that offer advantages over others... there are ways that elicit a better gain structure, or better signal to noise factor... but having been a recording studio nut I have done the unthinkable..taken things and made them do things they were NEVER meant to do.. the cool thing about line level sources...whether it be mic level or line level.. you cant really hurt them no matter how you connect them because your dealing with high impedance and low voltage.. ostensibly below 3 vrms. I believe that if you can dream up a way to do something you should try it and see what happens...for example, it's common practice to take a send (usually a POST fader send in this case) and send it to an effect processor, say a reverb, or delay...and return it to another channel where we can EQ, apply gain, have a fader to use for control versus a knob, and even an insert point for post process.. processing (like dynamics, or gating, or whatever! I've sent mike signals through a guitar amp...so I could get a distorted sound for the recording.. that didn't sound so canned.. I've sent signals through an am transmitter and into a tiny little transistor radio for its sound... I've sent vocals and even a xylophone through a stomp box for the effect it gave...I was using speakers as microphones long before Yamaha came up with its \"kick\" mic...Ahhh. the stories I could tell...
As far as the apologies.. no need mate! ( as Mic Dundee would say.. \"No worries\") actually the only things that aren't right at to actually clipping in the gain structure are the input stage of the mixer, but you want a good strong signal here...just not clipping no clipping in the Driverack... and then turn the amp down a couple db for headroom when the rest of the system is showing clipping. Remember that in the driverack where the signal is digital.. it no longer acts as an analog signal where signal to noise is key and you need as much signal as possible to push the noise floor down...so get that mixers masters to right at clipping.. occasionally for the gain setup.
The Fletcher Munson equal loudness curve illustrated that our ears hear hi mid frequencies best and lows and highs with diminishing amounts the higher and lower we go...3-5 khz is the area that is most annoying and most fatiguing to the ears.. so watch those.. they also have a lot of energy and tend to bounce around a lot, and thus can cause feedback.
As for the limiter those are called artifacts as the limiter \"hunts\" for something to do, and as it reacts to them. As usual Kevin .. spot on...in thought process and implementation!
As for the subs... 2 subs together... 3 db added gain.. FREE.. not to mention the added drivers.. put all subs together... more gain free, and the added benefit of having no comb filtering! BTW as you already know.. wall..and corner loading also add free db's...As for the deal about adding speakers to lower the impedence.. thai is true, but you need to remember you need to divide the available power by the number of drivers... sure you might get a 50% increase in power but you have also doubled the number of drivers...
I'll tell you whats next... lots more learning.. and experimenting... remember there is no right or wrong way of doing things.. only the results that determine if a thing was worth doing or not!
Keep up the experimentation! I am convinced it is extremely valuable even if we 'bump a few settings' while we are in there learning.
I have had the 'EXACT' same experience with the peak-stop and auto-on functions in the DR. I would like to know what is up with that, but haven't had time or priority to dig deeper into that. Let me know what you find.
Another thought on the DR setting for playback only, CD & DVD, etc.
When you autoEQ'd your room, what did you select as your response curve? I'm guessing flat or 0. The options are: Flat (0), and Response A-D, and Low, Medium and High Precision.
If that is true you may want to copy your current playback program into a new 'development' space and try autoEQing the room again but with a different response pattern like A, B, C or whatever to see what sounds best. I have noticed that for strictly playback these curves can sound much more like cinema sound that we are all used to.
Comments
Click on Nipper... He's got a PM for you...:twisted:
EDIT: PM removed...
Kevin
I have a couple of elementary questions:
1) I noticed especially with acoustic electric guitars that sometimes a thump noise is introduced into our system... It appears to come from the guitar being bounced on the players knee (if sitting) or maybe the hand is striking the body of the guitar while being played... This thump is coming through the subs (must be below 128 Hz) and for me this thumping can be very annoying... Last night I was very tempted in enable the high pass filter cut (100Hz and below) on that channel during the performance but decided not to... The problem is...I don't know what the frequency range is for acoustic guitars,...and I don't want to filter out any legitimate tone?...
2) When dialing in the input channel gain (trim attenuator) for microphones and instruments (using PFL) I'm uncertain on what the preferred or target peak point on the meter should be?... Is there a \"best\" or \"tried and proven\" procedure for setting up the input channels before a performance?...
Due to the Memorial Day weekends our Sunday night youth service was very small and calm last night... The crowd was only about half of normal size and the makeshift praise team only consisted of my son (acoustic guitar and vocals) and another young adult (vocals)... Let me tell you...this kid has a vocal range that is unbelievable and is slated to audition for next seasons American Idol… Anyways,... this worked-out great for me because I used the replacement mixer (live) for the first time... Last weekend the youth put on a drama in the main sanctuary... As luck would have it...the house was jamed packed with standing room only on the evening when the other mixer crapped-out...
My son Daniel playing his graduation gift and our \"future American Idol\" (I hope) were jaming last night...
Guys,...needless to say,...I'm a very blessed man and proud father!...:D
Kevin
Well... I think it's time for mix lesson # 1...
With a system capable of good low end reproduction anything you don't want in the subs can be eliminated with the aux fed sub method. This method however requires an additional crossover channel and we don't have that here. So how do we limit the amount of LF in the mix so that we don't end up with a muddy hollow mix? Even in the Aux fed method you would engage the HPF, and use the LF frequency control where additional cuts are necessary for that band. We have already established that each channel is capable of full frequency response in our super duper gain structure and system tuning. Now consider that every single open mic is not only hearing the full spectrum it is being subjected to, but is also \"feeling\" it...the vibrations that are induced through the floor manifests itself as rumble and LF howl and can really muddy up a mix. Multiply that times the number of open mics and you can see that if we don't do something about it we will end up affected by it.
Here is what I would suggest... The only instruments that REALLLY belong in the subs are:
1. Kick
2. Bass
3. Keyboards
4. some floor toms perhaps
therefore engage the HPF on all channels that are NOT in this group! ESPECIALLY vocal mics. Because a source has the capability to do a thing, does not mean it should be allowed to do a thing... Like, because a sub is capable of 25 hz doesn't mean it should be allowed to do it in all instances. One very good argument here is that if the amp isn't capable of powering the subs properly they shouldn't be allowed to have the lower registers.. There is very little usable info in this region anyway, and ends up wasting power in useless and sonically displeasing rumble and speaker luffing. (where the speaker pumps but does not reproduce any musical information) In a reverberant space the usable info below 45 hz can cause room loading and modes that can degrade the intelligibility of the mix. Your mixer has a 100 hz HPF.. it also has an 80 hz shelving control (a \"tone\" control fixed at 80 hz that allows cut or boost of that frequency...BTW that is the typical center frequency of the first harmonic of the kick drum) you also have a swept lo mid that is capable of 35 hz. This is very nice, and can be used as a swept HPF if no LF tayloring is needed... in fact with a swept HPF and Swept LPF I could be very happy with a properly tuned, good sounding system...this mixer has that capability...if you ignore the Hi and LO shelf's.
So. with vocal mics, engage the HPF, turn the 80 hz shelf down fully and if you still have unwanted LF in the system use the LM controls.. turn the LM gain down and sweep through the lower frequencies till you find the offending point then work the gain control to set the amount of that cut that you are using. Remember... even though we have the 100 hz HPF engaged, the frequency only starts falling off at that point and is probably in the area of 15db per octave.. so there will be content in the 80 hz range even with the HPF engaged. If you want to add additional warmth to a voice on a given channel on quieter passages, use the 80 hz shelf or the swept LM to do so.
Now, channel gain setup using the PFL ... Yes this mixer has a 4 ladder channel meter and is useful for monitoring a given signal and can indicate if a signal has increased during the performance ( geez I can't hear myself .. I'm gonna crank up my amp!) but the master section has a 12 segment meter that is inserted when the PFL is depressed. Set the channel main fader at the design center...the -0- right next to the LR button that routes that channel to the mains. with the signal present set the input gain for that channel to average about the -0- point.. a 6 db overshoot will still give you 10 db of headroom..(plus a little more actually to the point the mixer amps flatten) and should ensure sufficient headroom while maximizing headroom. This PFL is also great for monitoring individual channel levels against the overall mix, and can help diagnose a missing signal, or errant peak that is eluding you otherwise...
The percussive sound you speak of on the guitar is treated with the same methods we spoke of above...an acoustic guitar should have it's low end treated in such a way as to incorporate fullness but still not promote howl and feedback...this usually means setting the HPF, and possibly further tailoring the LF with the LM control.
BTW if you are using the FOH mixer as the monitor feed be sure to use the \"PRE\" position an all sources sent via the aux sends. That way you can make adjustments to FOH that won't affect the monitor mix...The \"input gain control\" is however before the \"PRE\" buttons so don't adjust the \"gains\" after you have set the monitor mix! VERY IMPORTANT!!!!!
All for now gents.. I have sheetrock to put up! Doesn't that sound like fun?
Good looking kid Kevin...you are blessed...
Gadget
I understood everything you wrote (absorbed it like a dry sponge) and you told me exactly what I wanted to know!... I can't wait for the next praise team practice so I can set this thing up properly... Speaking of the "Super Duper Gain Staging" I am looking forward to your critiques of the procedure I PM'd you... I understand it (I think) very well up to the DriveRack and after the DriveRack... I'm just uncertain about the correct method or sequence inside the DriveRack itself... Please straighten me out on this and let me know your method for setting it up inside the DR...
I'm planning on setting up two programs inside the youth DriveRack... One will be for live music as it is now and the other will be for recorded music (DVD's) to use for "Friday Night Flicks"... I'm thinking this program will be identical to what I have now in regards to the GEQ, crossovers, delay, and limiters,…but I don't see the need for the AFS, notch filter, and PEQ's... What are your thoughts?...
Thanks a billion!!!
That boy has girls tripping all over themselves...
Gary,...thank you for the compliment!...:)
Kevin
This is exciting to see!
Kevin -
You realize that the quality of your sound is moving up when your system is so clean you start to notice and are able to distinguish these types of adjustments are necessary! This is total validation that you are significantly improving the sound.
Gadget -
Excellent article!
I have one more thing from my experience which may help clean up the low end stuff that Kevin is starting to notice.
In a church I was setting up, they had crown CT1200's and on the amp there was a selectable 35Hz or 70Hz filter. This turned out to be an absolute boon for getting rid of the muddy sound we were having every week due to unnecessary stage monitor artifacts.
I noticed that on Kevin's amps you could do the same thing at 30 or 50Hz. I think that may help to roll off only the amps powering the stage monitors.
We actually saw this make a difference on our RTA at the church as we ran pink before and after we set the amp filters.
Keep up the great work guys!
W
Thanks we'll get into some other mixing techniques later as time permits.. That one is the biggie though...Oh and Wolfie.. Sorry I wasn't available earlier, call when ever you can...Oh and I believe that was one of the first things Kevin and I did... set the amp filters.
Kevin, I will respond to the PM soon.. I have been very busy working on re-doing the old studio. I have mangled my hands to the point that I can barely type though...
Be well guys!
Gadget
It's great to hear from you and I hope you had a great weekend!... The amps for the flying mains (JRX115s), floor mains (JRX125s), and monitors are set to roll off at 50Hz and the amps for the subs roll off at 30Hz...
Gary,
At last!!!... Maybe I can offer you some "sound advice"!... HIT THE NAIL with the hammer!... If done properly you should hear a distinct 500Hz ping...instead of a painful 30Hz thump!... And the vocals might punch thru cleaner!...
Kevin
After studying the mixer literature again...it appears in the FOH mode I don't have EQ control (via the mixer) over my monitor feeds... If this is true...then what would you recommend (with current set-up in mind) in regards to a decent 2-band graphic equalizer... It would be great to have AFS as well...but cost is a major consideration... I have four monitor feeds and it would be nice to be able to low cut and/or to EQ down frequencies as needed... I'm thinking the DBX 215 (X2) might be sufficient for this...but I would love to get your opinions...:?:
After surfing...it appears that Mackie is currently the only manufacturer of a quad-band GEQ...?
Thanks!
Kevin
Dra
First an observation...
\"Using a 15 band EQ is like carpet bombing where a sniper rifle would suffice...\" Not my quote but as usual borrowed without permission and since I can't remember who said it .. without credit as well... but a very accurate statement. I do know it was on the PSW 'LAB' forum though...
The ISO centers on a 15 band EQ very from 60-6000 hz covered by a single fader... that means if (for instance) you were having an issue with the first harmonic of the kick drum, you would have to cut both the 63 and 100 hz faders effectively cutting frequencies from 40 to 160 hz! ALL those frequencies and all we wanted to do was cut 80 hz...We would probably have had to do more cutting than if we were just treating 80 hz also because of the nature of the location within the filters location and the nature of the slope on each adjacent band and the fact that 80 hz falls between bands... NOT good for the sound... not good for the feedback elimination...
A 31 band EQ is better... but still has the problem of ISO centers that often don't fall on the feedback locations and cut way more than is necessary to address the feedback. This is often the solution, but there is a better way... Lets examine fully perametric equalization.
In fully parametric EQ's even though you typically have fewer bands you can do more and be more precise in doing so. A parametric EQ allows you to zero in on the frequency you want to address (with digital eq's to within a few hertz) and allows you to apply a very very narrow cut or boost all the way up to a nearly 20-20,000 hz cut or boost.. with a single band...So with a PEQ you can zero in on the problematic frequency...start with a narrow cut and adjust the \"Q\" (amount of the frequency spectrum you want to affect) till the problem is addressed with the minimum amount of invasiveness to the frequency spectrum.
I typically monitor the FBX screen to identify feedback areas and adjust them with a PEQ on the fly...the FBX screen gives you the frequency in hz the feedback occurred at and shows the amount of cut and \"Q\" of the cut so it's simple to mimic with a PEQ channel.
A variable high and lo pass filter can be a great asset in an EQ as well... and can take actually add many bands of effective EQ in a single knob.
Ok for the sake of argument here I will add a potential cost effective solution.. evebn though it is an off topic piece...If DBX had such a piece I would suggest it.. but alas...
http://www.zzounds.com/item--BEHDEQ2496
check out the feature list... and yes.. it's disposable should you have any issues out of warranty but hey I have not had a failure in any of their stuff I posses yet...
All for now gents...
G
Since we learned how to correctly position our monitors (straight ahead towards the vocalist) in regards to using the SM58 mics we have not had any feedback issues... Thank you Gadget!... But our minister wears a countryman headset and he wanders all over the place while he's talking... I currently turn the monitors all the way down just before he speaks... I only mentioned AFS because it might come in handy for one of those forgetful moments... At this point it would be more of a "luxury" than "necessity"...
I'm thinking by EQing the monitor feeds I could low cut some of the lower frequencies (possibly 100-125Hz and below) and possibly get a more pleasant bass response overall?...
What do you think?...
My primary reason for EQing the wedges would be to eliminate any mud in the system... My secondary reason might be for feedback elimination...if needed... Please let me know if I'm on the right track, if that's reason enough,...and if/or I have my priorities screwed-up...:?:
Just to let you know...I copy and paste your lessons to word documents because I think they're invaluable...:brown nose: Oops...they need to add another emoticon...:D
Thanks!
Kevin
Nipper's got a PM for you!... He just returned from the water park...
EDIT: PM removed...
Kevin
Option #1. Find a board that offers 4 band channel EQ's (preferably with mid and low sweeps) and order all the pots in the eq section. Heck, get the filter switch as well. Build an outboard unit yourself.
Option #2. Find (on EBay or whatever) a mixer with 4 sub groups and use the channel EQ's, etc. and put it in line from your aux outs.
OK, let me have it!
Dra
As a stop gap you could also see how EQ would work by running your aux into an open input channel on the board and use that EQ to see if it does what you need before you invest more $$$.
My two bits.
W
Hello Wolfgong,
Great idea!... But will that work?... Has anyone here tried this before?... I guess I could use the "line in" on the input channel but how would I get back out?... Ohhh well...back to the manual...:)
Kevin
Dra
Example - Aux (mon mix) #1 ouput to channel 14 (?) input, aux #5 (?) out (new mon mix #1)
Dra
I have \"line in\" and \"direct out\" for each channel... The direct out is pre-fade (following the pre/post-EQ setting)... Hey,...this just might work!... I will have to study it a little more...maybe call Allen & Heath tech support to verify... Want to be certain that the line level/impedance of the aux outs matches the line in and direct out of the input channels...:)
If everything looks good then I can dedicate four channels to monitor feed EQs and have 100Hz HPFs built right in... If so it would be nice to be able to disable the phantom power and polarity switches on those channels... Those puppy's will fry the AUX feeds...:?
Kevin
This is how I see it....
AUX1--->LINE IN--->HPF--->EQ--->DIRECT OUT--->AMPLIFIER
....................................
AUX2--->LINE IN--->HPF--->EQ--->DIRECT OUT--->AMPLIFIER
....................................
ETC., ETC...
Kevin
After some more reading I would have to change an internal jumper on those four channels for the direct-outs to be post-EQ...:)
My main concern about trying this...are the phantom power and polarity switches for those input channels... I don't like having a disasterous situation lurking around in the background just waiting to happen...:twisted:
I'll call Allen & Heath tomorrow to see what they can offer...
Kevin
Dra
If you can add a 2 on the front of that number I can hook-you-up...with free delivery!...:)
Kevin
Dra
You drive a hard bargin!... I know I'll regret this...but I'll have to pass on your generous offer...
Now...if only those baskets came perpetually filled with EVH...:)
Kevin
Dra, you are in fine form today.
Thanks for the laughs!
W
Tomorrow evening we are kicking-off “Friday Night Flicks�!... We didn’t get to it last Friday because our youth minister had something to tend to… In preparation and in addition to my feudal efforts as a “perfectionist� I decided to set-up two separate programs in the DriveRack… I ‘m using one program for “live sound� and the other for “recorded sound� i.e. CDs and DVDs… Currently the recorded sound program is identical to the live program with exception to AFS and notch filters (NF is probably not needed in the live program either)… Last night while I was at the youth center...I wanted to check-out and fine tune (I should say...play with) the system a little more…
There’s something about the new mixer that has been bugging the crap out of me… It seems like I’m not getting adequate gain out of it; in comparison to the old desk,… This makes no sense to me… I decided just for heck-sake to follow the “super duper gain structure� process one more time…and only if it puts my mind at ease… Gadget,…I’m very sorry but this is why I have been persistent (or irritating) in trying to verify this procedure with you… Anyways,…somehow I missed something on my last attempt… I’m pretty certain it’s that I didn’t enable the AFL on my master channels on the previous effort… Because of this I was 3dB short of barely bouncing the yellow on the output gains of the DriveRack… So this is what I did: I kept the input fader at unity (in the DriveRack) and I raised the master slider up by 3dB… Obviously this made a huge difference overall!...
Anyone,...was this the proper approach?..
Since I was the only person in the building (and I really wanted to play with the toys!) I decided to try some other things… Just out of curiosity I wanted to see if I could fine tune the timing delay between the mains and subs by putting-up tones (at the crossover boundaries) and looking for the peaks using the RTA mic… Bad idea!... It doesn’t work...even when I increased the crossover overlaps between the mains and subs… I’m writing this one off due to the unique frequency response characteristic of each speaker(s)… For now it’s definitely going to be an ear thing!... But hey…I learned a little more about this stuff…in how it pertains to the “live environment�… Now maybe if two channels were used with respective tones that are on the center of the overall response of each speaker?…
While listening to a variety of tunes (loud of course) I decided to mute groups speakers so I could evaluate each one individually… I found that certain sound tracks would drive the subs (more prevalent on the right channel) into this annoying fuzz/buzz like sound at the peaks… After trying different things I found that the limiter was causing the problem… I disabled both the “peak stop+on� and the “auto on� and the problem vanished… With the limiter enabled it still did its job (but without the fuzz/buzz) even when running the system wide open…
Anyone,...what are your thoughts on this?.. Was this the proper approach?.. Even though both amplifier attenuators are dead-on 26 should I also back-off the right channel just a tad?...
The highs and mids are incredibly clean and crisp but I think they are overpowering the low end by a little… I lowered the crossover gain from 0dB to -1.5dB on the flying mains and from +3dB to +1.5dB on the floor mains… To my ears (maybe a bad idea) this was a notable improvement… I must say that without a doubt: the floor mains adds a \"fullness\" that really ties this system together!... The two subs are 8 ohms and driven by two amps in bridge-mono… So here’s an incredibly stupid question: If two more identical subs were placed on top of the current ones and wired in parallel (4 ohms)…what would I realize in regards to low end response and on this system?...
Master,…would I not increase those deeBees (at the least) by six?...
For now I think the system is probably 99% optimized for “recorded sound�.. I still need to do some homework and setup the PEQs for “live sound�… I have been very fortunate in not having a feedback issue except for one time…when our minister sat down on the stage right beside a wedge…:oops:
Guess what’s next on the agenda?…:)
Kevin
Ok, the only dumb question is the one that doesn't get asked :shock: A mistake is proof positive that at least an effort was made :oops:. Kevin, I have a live preset, and a DJ/ Karaoke preset. You need a whole lot more headroom with live music so it's a very good idea to have it that way.
When dealing with line level sources...there is no \"right and wrong\" way to do it.. there are better ways, there are ways that offer advantages over others... there are ways that elicit a better gain structure, or better signal to noise factor... but having been a recording studio nut I have done the unthinkable..taken things and made them do things they were NEVER meant to do.. the cool thing about line level sources...whether it be mic level or line level.. you cant really hurt them no matter how you connect them because your dealing with high impedance and low voltage.. ostensibly below 3 vrms. I believe that if you can dream up a way to do something you should try it and see what happens...for example, it's common practice to take a send (usually a POST fader send in this case) and send it to an effect processor, say a reverb, or delay...and return it to another channel where we can EQ, apply gain, have a fader to use for control versus a knob, and even an insert point for post process.. processing (like dynamics, or gating, or whatever! I've sent mike signals through a guitar amp...so I could get a distorted sound for the recording.. that didn't sound so canned.. I've sent signals through an am transmitter and into a tiny little transistor radio for its sound... I've sent vocals and even a xylophone through a stomp box for the effect it gave...I was using speakers as microphones long before Yamaha came up with its \"kick\" mic...Ahhh. the stories I could tell...
As far as the apologies.. no need mate! ( as Mic Dundee would say.. \"No worries\") actually the only things that aren't right at to actually clipping in the gain structure are the input stage of the mixer, but you want a good strong signal here...just not clipping no clipping in the Driverack... and then turn the amp down a couple db for headroom when the rest of the system is showing clipping. Remember that in the driverack where the signal is digital.. it no longer acts as an analog signal where signal to noise is key and you need as much signal as possible to push the noise floor down...so get that mixers masters to right at clipping.. occasionally for the gain setup.
The Fletcher Munson equal loudness curve illustrated that our ears hear hi mid frequencies best and lows and highs with diminishing amounts the higher and lower we go...3-5 khz is the area that is most annoying and most fatiguing to the ears.. so watch those.. they also have a lot of energy and tend to bounce around a lot, and thus can cause feedback.
As for the limiter those are called artifacts as the limiter \"hunts\" for something to do, and as it reacts to them. As usual Kevin .. spot on...in thought process and implementation!
As for the subs... 2 subs together... 3 db added gain.. FREE.. not to mention the added drivers.. put all subs together... more gain free, and the added benefit of having no comb filtering! BTW as you already know.. wall..and corner loading also add free db's...As for the deal about adding speakers to lower the impedence.. thai is true, but you need to remember you need to divide the available power by the number of drivers... sure you might get a 50% increase in power but you have also doubled the number of drivers...
I'll tell you whats next... lots more learning.. and experimenting... remember there is no right or wrong way of doing things.. only the results that determine if a thing was worth doing or not!
BE well guys..
Gadget
Keep up the experimentation! I am convinced it is extremely valuable even if we 'bump a few settings' while we are in there learning.
I have had the 'EXACT' same experience with the peak-stop and auto-on functions in the DR. I would like to know what is up with that, but haven't had time or priority to dig deeper into that. Let me know what you find.
Another thought on the DR setting for playback only, CD & DVD, etc.
When you autoEQ'd your room, what did you select as your response curve? I'm guessing flat or 0. The options are: Flat (0), and Response A-D, and Low, Medium and High Precision.
If that is true you may want to copy your current playback program into a new 'development' space and try autoEQing the room again but with a different response pattern like A, B, C or whatever to see what sounds best. I have noticed that for strictly playback these curves can sound much more like cinema sound that we are all used to.
Carry on good and faithful servant!
Wolfie