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Hello!

To start off with, I have never taken any classes in audio engineering and do not have a lot of expertise when it comes to actually knowing how various settings effect sound.

However, I do know what sounds good and can tell when something doesn't sound too good. =)

So, I wanted to get some advice concerning a few things that we run into at our church.

To start off with, here is our system...

Yamaha 01v96 Digital Board w/ 32 channels
HEAR In Ear monitors
(2) Larger Peavey Floor Monitors on platform
(6) Time delayed house speakers, not sure on make/model

Overall the system is amazing. What we are able to do on the fly, the overall sound that comes through the house, etc, is great.

The problem I have is that when I solo out a channel in my headphones I hear something totally different (In a better way) than what I hear in the live house. I realize that acoustical treatment in the room will help and that listening to one channel through headphones is a lot different than the whole house. But the difference isn't in the ability to hear the channel, it is the clarity, the precision.

For example...Our bassist plays a lot like Victor Wooten. He is extremely talented and uses a lot of that style of play, accentuating high and low ends as well as stacatto and other forms. When I listen to his channel on the board everything is crisp and clear, you can hear every minute detail. However, through the house you just hear the low end and not much more than a rumble.

He is going direct into an active direct box. I've tried every EQ setting I can think of...had him crank all sorts of knobs on the bass, but it doesn't get much better in the live sound. Is this a total acoustical treatment thing in the room or is there a magical EQ setting for this style of play?

-

On another note...When our youth band plays, they are on the platform where there are only the two large monitors. So, no in ear stuff. We have the bass, guitar 1, and guitar 2, microphoned with SM57s. Though the house probably sounds okay like this, since I play in that group, I have no clue how it sounds out there as compared to up on the platform.

On the platform it is just sort of a mess of sounds coming from each amp as well as some more sound coming from the monitor (Normally vocals). Is there a better way to control the sound on the stage? Should we have our amplifier levels set low, high, or just enough to hear ourselves?

As I said, I'm no engineer (Though I am going to be taking some training). But I know that some things don't sound too good. I appreciate any responses that I may receive. Please let me know if you have any tips at all.

Thanks!

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Comments

moonbaby Wed, 02/22/2006 - 09:02

Hi, Arvida!
As far as the bass tone is concerned, from your description, it sounds like (1) of (2) problems may be involved:
Either your system has some sort of subwoofer arrangement that is not described and that the bass channel has been mis-assigned to (that kind of whacko thing happens more than you'd think in churches!), or
The EQ "library" setting for that channel has been erroneously selected or programmed, and all the bands are set incorrectly. I am not sure whether that particular boards' PFL/solo listens pre- or post-EQ. That is something you will need to address. If it is PRE-EQ, than the solo'd signal from the DI will be fine in the phones, but the EQ'd signal will be going to the house.I am suspecting that is the problem...
I was going to suggest, too, that where you "tap" the signal from the bass or bass amp, to GO to the DI may have something to do with this issue, but not if the phones' sound OK. Example, some bass amps have a seperate subwoofer output to "biamp" with. If you are taking the signal from that point, yeah, there'll be nada but rumble. Don't think that's the problem,though.
So, to recap problem#1, I'd check into the EQ settings and where the solo'd signal is tapped in relationship to the signal flow.
Now, to problemo#2:
It is ALWAYS a good idea to keep the onstage guitar (and bass) amps as low in volume as you can, and in your case, maybe off stage entirely. You can always bring them up in the floor wedges. That will tend to create a "single point" distribution of the monitor mix. Instead of guitar#1 coming from the left (after it reflects off the right wall!), guitar#2 coming at the back of your heads (after it bounces off the ceiling!), and the bass envelopes you in a cloud of mud(!), you are "focusing" the sound from a central (or at most 2) points. This can help you with the psycho-acoustic antics of sound that wreak havoc on your vocalists when the sound is bouncing around the room. Make sense?

anonymous Fri, 02/24/2006 - 08:45

Hello,

Thanks for the advice!

We have two large subwoofers that are placed towards the front of the stage on the right and left.

I think that the DI are coming into the phones pre-EQ because it sounds pretty raw, but nice. Then again, it may not be...It may be that the reason the solo'd channel sounds better is because it isn't mixing with anything else. I'm not totally sure.

I'll check a few of the things you mentioned and I'll let you know for sure.

As far as the onstage stuff...If we move those things off stage, would you suggest pointing them towards the stage still or somewhere else?

Something I did the other day at rehearsal was to crank the gain on the channels that were using microphones on amps...This actually seemed to help a lot for monitor mix. Before the only way I could get the mic'd amps to come through was to turn up the amp or to crank the levels of the channel way up.

So, is this acceptible? Should we do anything different with compression or anything?

Thanks again!

moonbaby Sat, 02/25/2006 - 11:46

OK, let's start with Audio 101: GAIN STAGING...
You have a mic in front of an amp, right? You have the performer play the guitar/bass/zither/kazoo plugged into the amp, and have them play it at the volume level that they're going to be playing at during the live performance. Go to the channel that the mic in front of the amp is plugged into. Hit the Solo button (this might be called PFL-"Pre-Fader Listen") button. Watch whatever metering that this function has on your board. On my Tascam DM24's, it's a multi-segment LED array, on my studio's Wheatstone it's a backlit VU meter, on your Yammy it's probably a function of some LCD display, whatever. Adjust the INPUT gain control to drive the metering so that the loudest parts of the playing don't send the meter into "the red zone", but don't set the gain so low that the meter doesn't register very much,either. You want to get a strong enough signal so that you are above the inherent noise level of the mixer, without "clipping" the signal with distortion. And "clipping" a digital mixer sounds absolutely horrible. Once you've obtained a proper input gain setting, you use the channel's fader to set the level in the mix. Now you do this with EVERY signal source that you have-vocals, guitar, keys, bass, drums....even kazoo!
You may also find that adjusting the channel EQ will vary that signal strength. Why? Because you're varying the "cut/boost" of the band(s) by +/- so many decibels. Boosting, say the bass@80 Hz., will mean that you will probably have to reduce the input gain by a similar amount. Otherwise, you run the risk of "clipping" that channel. Do ya see where I'm coming from?
So to answer your last question, yes,it's acceptable to "crank" the channel's input gain...if that's what it takes.
OK, now for that first question....Where do you "fire" the amps?I have no idea, because in some venues, aiming them away from the stage will cause more problems due to their sound ricocheting off some wall and "showing up" twice as loud as they are onstage in the 5th row! I'm serious about that. Ideally, it would be great to have them in a dressing room backstage.... Realistically, I prefer to use a device called a "gobo", which is a sound-controlling barrier wall that is placed in front of the amp to keep it from blasting out onto the stage. You make these, or you can order custom-built ones from Moonbaby Industries :wink:
Basically, these are a portable, aesthetically-correct 36"H X 36"W panel made of 3/4" MDF or plywood that you cover with sound-absorption material (fiberglass insulation, Dacron,etc) on the side that the amp is firing into, and paint/upholster the outer "visual" side to match your decor.
This greatly assists you in keeping the blast from an amp contained. I have made/designed sooo many of these over the years, it's scary. For churches, bars, studios, you name it. I think that it's ironic that the stuff we use to get good sound in a bar is used to do so in a church, don't you?
Well, I've gone on for long enough today....

Jim "Wake Me Up Before You Gobo" Moonbaby

anonymous Mon, 03/06/2006 - 06:41

Thanks for the response!

I did check on gain levels and they seemed to all put the channel at the max without clipping, so I think we were okay there. Overall, setting the gains up and using the faders to control actual output worked really well for our sound. That way, most of what we heard on stage was the actual monitor mix.

As for the amps, we just sort of had them positioned pointing towards the middle/out of the stage at low levels and things seemed pretty good.

=)

On a side, note, which you may know about, but maybe not...

We are getting an Alesis HD24 to multitrack record. So, we were initially going to take ADAT optical outs from the Yamaha 01v to the recorder and then mix the channels later on. However, if it is possible to hook up another mixer to the HD24 and mix it live to the recorder as well as maintain a house mix with the Yamaha, we'd like to do that.

Do you know if this is possible? Or, would it be better to use my laptop and just use software as the live mixer for the HD24? Thanks again!

moonbaby Mon, 03/06/2006 - 08:14

Why do you want to mix your live recording "on the fly"? The whole reasoning behind using a multitrack recorder as a live recorder is so that you can keep different sources seperated to REMIX after the fact.
I think that you are putting a LOT of pressure on whoever is doing the sound to come up with (2) seperate decent sounding mixes simultaneously, but in a different way then, say, house and monitors are done.
This is because the live PA mix will be adjusted by someone who is hearing the band in an accoustical setting, and is making adjustments to the house and/or monitor mix accordingly. For example, no matter how low a volume the guitar amps are set, the congregation will hear some of them, and the mix level will be adjusted to reflect that. Ditto for any "monitor wash" coming off the stage.
But the recording side of things is different. ALL you have is what is plugged into the system and fed to the HD24. Trust me, they are totally different animals! At our church, I do a live mix on a nice Midas Heritage 3000 board. We split the signals off the desk to a Pro Tools rig in a control room behind me so that the recording engineer(s) have total isolation (accoustically) from the live mix. Even then, they are not attempting to get the mix "right" during the performance. They are mainly trying to get the LEVELS right! And they have the additional advantage of doing it all on a Control 24 console, not a laptop!

Jim "Never Bite Off More Than You Can Chew or Somebody's Gonna Choke!" Mooney

anonymous Tue, 03/14/2006 - 09:16

You sort of answered the question I meant to ask, in a round about way. =)

My desire is to have our main guy run the house sound. Then have a second person manage and maintain our recordings. So, we'll have our 01v split sending all channels over to an Alesis HD24.

My understanding is that you can't really do anything with the outs going to the HD24, you just get the audio that is being inputted to that channel, right?

Or is it possible, like it sounds you have setup, to have someone mix levels that are sending to the recorder so we have less legwork later on? If we do this, we'll have to have a second mixer right? Or hookup a PC to the 01v?

I guess a lot of this will be trial and error. My desire is to, like you said, dump it all to the recorder and mix it all later on when we have time. I guess I just want to make sure that what is going to the recorder is acceptible for post-production and I'm not really sure how to monitor it to make sure it is good.

As far as the post-production, what would you recommend having in the setup? Obviously the data from the recorder. A PC to do editing...Other than that, what else would you setup if we weren't able to do a ProTools rig? Should we invest in a good mixer to do post-production, or should we use our 01v for that as well? Any other hardware that you may recommend?

Thanks again. I really appreciate it.

Oh, also, do you know of any good books about post-production/recording?

moonbaby Wed, 03/15/2006 - 04:55

OK, my working knowledge of the 01v96 is not that great. Are you able to take 32 sources and mix them at once? I thought that the limit was 16 mics...? What is the capability of each input channels direct output? Is it "pre-fader" or "post-fader"? What you will need to pull off what you want to do is have enough MIC input channels with PRE-fader direct outs to go to the recorder. Then you can use your laptop/PC to control the recorders track levels, I would think. But maybe I think wrong. Me Tarzan sometimes.
Actually, if you get the right board, you may be better off ditching the Yammie for the live sound entirely. Instead, get an analog mixer like a Midas Venice24 or an Allen&Heath GL2400, and use the boards direct outs to feed the recorders tracks. Mix the house and monitors with the Aux sends and buses on the board, feed the recorder with the DO's, and mix the recorder through the Yammie! The mic trims on the main boards inputs would set the levels to the DO's. If further level-trimming at the track inputs is required, I think you can do that via laptop/PC,right?
Make sense? Got $$$?

Jim "Me Tarzan, Where's Jane?" Mooney

anonymous Thu, 03/16/2006 - 12:00

Hah, I think my head just exploded.

=)

Update though...

Yammie is using two optical ins for our external pre's which are sending 16 channels to the Yammie. So, that is where we get the ability to manage 32 total channels of audio on two different layers. =)

Our musician monitor system is using one of the three optical outs. Which leaves us with only 8 channels of optical to mess with.

So...Scrap everything above entirely! We've got to go with a new board if we want to do both recording and live mixing on the same device....

Now, my thoughts are as follows....

We have our old A&H GL2400 (Suprise!)...So, I was thinking that since we are already setup digitally for live sound...why not take all 32 channels coming in via our snake (I believe all 32 are XLR)....Can we take those 32 XLR channels and split them to 64 (32 original and 32 new). Then send the second half of them to the A&H.

We could then use 1/4 TRS to connect the A&H to our recorder for as many channels as we need. =0 Then, we can live mix as well as live record and not have to intermingle the two.

So, my question would be....Is it possible to split those 32 XLRs when they get to the booth? Right now all FOH instruments and mics are going to XLR bays (Three of them), which are wired to the snake that comes to the booth.

So, easiest for us would be taking those 32 XLR ins and splitting them to 64 XLR ins, obviously just doubling our inputs so we'd have two of each channel.

Is this possible? Signal degredation? Concerns?

moonbaby Fri, 03/17/2006 - 14:22

OK, sorry about the delay in getting back to you....
You keep talking about "splitting" the audio. If you mean that you want to take your 32 sources (instruments, mics, etc) off the stage, through the snake, then splitting each of them to 2 seperate ways (one to the PA mixer, the other to the recorder)...yes...and no.
You are probably NOT going to want to actually "split" all these signals to go two seperate ways. This will require a 32-input splitter system box, and a seperate recording board to feed the recorder. A splitter is an arm-and -a -leg,$$$. Not to mention that 2nd board. If you thought that you could split the mics to go to the PA mixer and then to the recorder's XLR ins....NO! Those inputs on the recorder are NOT mic inputs, they are "line level" ins. A mic signal is too weak to drive them, dig?
So what I was suggesting you do is take the (say) 32 mics (or 24 or whatever), run them to a decent board that has TRS insert/direct outs on each channel (and the A & H is fine for this), mix your sound for the PA on that board. Then you use the direct outs on each channel to feed each track on the recorder. The Direct Outs are sourced AFTER the mic preamp portion of the input channel, so THEY will be able to drive a proper signal to those track inputs. Do ya follow me so far? Good! I knew that you would!
I think that the A & H board you have has the ability to operate the Direct Outs either before the channel fader (pre) or after the fader (post). You definitely want "Pre" so that the fader adjustments you make to mix the PA don't affect the levels to the recorder! But your Gain trim controls at the front end of each channel strip on the board DO affect everything-PA AND recorder. That's OK, though, because if you have that control properly set to prevent overloading the input channel, the signal will probably NOT overload the recorder. And I think that you can use your laptop to control track recording levels on the recorder, anyway. But you will have to consult the recorders operators' manual. And check with A & H about the "pre" or "post"options for the Direct Outs. There may be a small "DIP" switch or a blob of solder that you'll have to deal with to select the mode, inside the board. Testing it is easy, though. Simply hook a mic up to the channel, plug the Direct out into a guitar amp (!) and work the fader...does the volume change? No? GREAT! Get it? Good. I knew that you would.
NOW we get to the Yammie. You use that to MONITOR your tracks during the recording process and MIX your tracks when you are ready to do so.
How? Simply route the track outputs from the recorder to it. Via optical, ADAT, whatever interfacing you have. Leave it there. It makes a better recording/mixing tool than it does a live sound mixer, anyway. And you'll find that the mic pre's on it ain't all that!
Next week, hooking up your Direct Outs to the recorder with different type connectors....Later.

moonbaby Fri, 03/17/2006 - 14:31

BTW, the reason that a splitter is soo expensive is because there has to be either a special transformer ("passive") for each input, or an "active" electronic circuit for each channel, in the box. If you simply 'Y' the mics out, you will pick up al sorts of interference (can you say "Radio Cuba"?),noises, and the mics will not "see" the proper preamp load, resulting in degradation of the fidelity...MUCHO! And all sorts of other crapola you don't want to deal with...trust me. Been there, done that.
My first splitter was a homemade jobbie that used a set of "bridging resistors". I was living in Florida (still do). Every night we used the damned thing for remotes, we'd pick up AM stations from the Carribean, along with Fidel Castro and his 3-hour speeches!

anonymous Mon, 03/20/2006 - 07:50

Thanks for the great info.

I found last night that I can successfully patch 16 channels optically through to the recorder. So, I've done that....

Now, to get the 8 other channels, I can go direct from my pre-amps to the recorder. Just need to find a good 16 or 8 channel TRS snake.

Also...You've been so helpful. I really appreciate it.

I put some sound isolaters on our drum mics, which helped tremendously in bleed. However, the ride is really bleeding into the snare, which is annoying. Any thoughts on this? We are using an SM57. The rest are getting a lot of bleed too, but he is in a cage, what do you expect, right?

Thoughts on the best electronic drum sets?

anonymous Mon, 03/20/2006 - 07:52

Thanks for the great info.

I found last night that I can successfully patch 16 channels optically through to the recorder. So, I've done that....

Now, to get the 8 other channels, I can go direct from my pre-amps to the recorder. Just need to find a good 16 or 8 channel TRS snake.

Also...You've been so helpful. I really appreciate it.

I put some sound isolaters on our drum mics, which helped tremendously in bleed. However, the ride is really bleeding into the snare, which is annoying. Any thoughts on this? We are using an SM57. The rest are getting a lot of bleed too, but he is in a cage, what do you expect, right?

Thoughts on the best electronic drum sets?

anonymous Wed, 03/22/2006 - 07:35

When you say the null of the pattern are you talking about the backside of the mic? Right now I have the mic on the left top of the snare. With an isolating foam peice behind it. Then the ride is over to the right of the snare.

So, if I move the mic to the right hand side, the ride will be behind the mic and thus all sound from the ride will go into the foam hopefully and be absorbed.

Yeah, I hate the cage...But churches...they think it is the best way to go.