Please login or register.
1 Hour 1 Day 1 Week 1 Month Forever
Login with username, password and session length

I wonder what the bass singer in the Oak Ridge Boys uses live.

They were Shure endorsing artists.  I recall both SM-58 and SM-87C capsules on UHF-R the last time I worked with ORB.

Another bass singer I worked with used a Neumann KSM-84.  He'd have floor wedge speakers jumping around on the deck, then they'd fail.

Ok, I'll try to get outside next week and refine my pre-alignemt EQ and see what happens.
IIRC it's a linear phase brick wall filter, 1.2Khz/79db/oct.


Thanks

You bet !
Yep, looked like a brick wall.
Hope flattening works, please see some further comments about timing made in posts with Russ.
Also, must say I agree with Russ you're unlikely to hear a change at that freq, but it does make for nice looking traces haha. 

Sorry, something still isn't working for me, but my experience with zero-phase filters is basically, er, zero, so let me see if I have this correct:

• Setup a speaker and a measurement system somewhere where no one will want to punch you if you run pink noise for an hour straight
• Put pink noise through HF driver
• Use Delay Finder, note the delay compensation time
• Adjust EQ/phase response of the driver until everything is as flat as you can get
• Put pink noise through LF driver
• Use Delay Finder, note the delay compensation time
• Adjust EQ/phase response of the driver until everything is as flat as you can get
• Insert zero-phase crossovers to taste

Right so far? If so, here's the problem I'm seeing:




Personally, I think I'd rather do what Helge's done and then phase-tweak the combined response. It just seems easier to me!

-Russ

Aaah, my apologies Russ, i wasn't clear in last post when to use Delay Finder's readings..
And just to be clearer backing up further to first post, I was trying to answer Helge's question about how to remove the final remaining phase wrap, not suggest an overall different methodology to what you two were discussing.)

May I rearrange your sequence:

• Put pink noise through HF driver
• Adjust EQ/phase response of the driver until everything is as flat as you can get
• Put pink noise through LF driver
• Adjust EQ/phase response of the driver until everything is as flat as you can get
• Insert zero-phase crossovers to taste
• Use Delay Finder, note the HF delay compensation time
• Use Delay Finder, note the LF delay compensation time

   *Use the Delay Finder difference as the delay to be put into processor for delaying whichever driver had the lower Delay Finder reading.
   *Done, in hopefully a lot less than an hour 

I guess we each have learned our methods, and they become the easiest for us. I use phase trace overlay a lot too, because dang, we most often have to.
Honestly, I consider guys like you and all the others that have mastered aligning sloping phase traces to have learned the universal tool, the more difficult tool to learn.
Because the additional flattening steps i proposed only work when both flattening and linear phase xovers are practical.
But when they are practical, alignment gets easier than phase overlay ime/imo.

I wonder what the bass singer in the Oak Ridge Boys uses live.

The first two photos Google Image Search pulled up that were large enough to tell showed different makes of wireless handheld, but nothing obviously esoteric. In the one with what looks like Shure UHF-R, the capsules appear to be good old SM58s.

-Russ

I wonder what the bass singer in the Oak Ridge Boys uses live.

{...} The question I would ask though is if front fills would help improve GBF...  Naturally the lower the speaker plays the more omnidirectional the LF becomes.  I can fight that a bit with a high pass but would really want to ensure that there would be a tangible improvement here to offset the cost and complexity of adding this. {...}

Like Brian mentioned, anything that will allow you to put more distance between your microphones and your PA will increase your gain-before-feedback, with the amount of the difference (at least in the LF) being roughly calculable using the inverse-sqare law: if the neatest speaker to the lectern mic was only 3' away and you increase that distance to 6' (i.e. double) the sound from the PA arriving at the microphone will decrease by ~6 dB, resulting in roughly that much additional GBF; conversely, if your lectern is already 30' away from the nearest speaker and you increase that to 33', you'll only see a ~1 dB increase in GBF. In the HF, where speakers are actually directional, you might see additional GBF gains beyond the simple inverse-square if the increased distance also places the microphones further outside of the speaker's coverage area.

Ultimately your best bet would be to do some quick mock-ups in software to see if you'll achieve your desired outcomes. As I said above, almost all PAs benefit from some kind of front fill, but GBF isn't usually the goal.

-Russ

{...} has to have the mic directly in front of him and no more than 6" away {...}

Just for reference, traditional technique with a microphone like an SM58 is to keep the microphone <1" away from the mouth for most singing.

{...}  I can add a highpass when I get the new system, however I'm greatly concerned if I use a basic SM58, J will still have the proximity issue.  Is there any mic type that help with this?  I've seen reference to, but couldn't figure out if, something like a dual diaphragm mic would work or what would be a good one for this particular use case.

There are several types of microphone that don't exhibit proximity effect, but they all come with significant tradeoffs in usability or cost (or both). Given that your description doesn't mention feedback issues, I would start by ensuring that the people in question are using proper, consistent microphone technique (including how they are holding the microphone, if applicable) and then applying corrective EQ as necessary, and see how far that gets you. Proximity effect on its own is easy to adjust for using EQ (ideally with a wide-band parametric filter, although a low-shelf is often faster and will do a fine job too) and is only really a problem when a singer's inconsistent technique causes the amount of the effect to vary considerably.

-Russ

{...} with both drivers having flat phase on their own, it still takes some time delay to tie them together so that the entire trace is mag and phase flat. {...}

Sorry, something still isn't working for me, but my experience with zero-phase filters is basically, er, zero, so let me see if I have this correct:

• Setup a speaker and a measurement system somewhere where no one will want to punch you if you run pink noise for an hour straight
• Put pink noise through HF driver
• Use Delay Finder, note the delay compensation time
• Adjust EQ/phase response of the driver until everything is as flat as you can get
• Put pink noise through LF driver
• Use Delay Finder, note the delay compensation time
• Adjust EQ/phase response of the driver until everything is as flat as you can get
• Insert zero-phase crossovers to taste

Right so far? If so, here's the problem I'm seeing:

We have our beautifully flat HF trace and our beautifully flat LF trace, but the LF is arriving early. As you say, no problem, we just delay the LF channel by the Delay Finder difference. Now everything lines up, but the LF trace is no longer flat, because adding delay to the LF causes its phase trace to start sloping.

The only way to avoid this entirely that I can think of would be to do all the component-level flattening at the single, fixed delay compensation setting found with the latest-arriving component. Done this way, any needed alignment delay would be already baked into any phase-altering processing.

In the case of Helge's traces, flattening the phase trace of the HF is going to take some significant processing muscle since it's lagging by ~180 degrees at the crossover point. An FIR maximum-phase all-pass filter will do the trick, but will effectively result in the highest HF frequencies arriving later than they originally were, which will necessitate re-setting the delay compensation. (Doing this IIR with a whole mess of minimum-phase all-pass filters would have the same effect since, of course, you can't actually make the sound at 1.2 kHz arrive earlier, you can only make the sound at higher frequencies arrive late enough to match.)

Now that the HF is flat it's time to face the bad news waiting for us in the LF. Since the delay compensation has increased from 20.90 ms to ~21.31 ms (or more), the LF trace that was once within 15 degrees of in time through the crossover is now significantly leading, so we'll have to add one (or more) minimum-phase all-pass filters to re-flatten it, effectively using all-pass filters instead of channel delay.

With all that done, we will have two drivers that are in in time with each other throughout both pass-bands, but it'll take a while.

Personally, I think I'd rather do what Helge's done and then phase-tweak the combined response. It just seems easier to me!

-Russ

I've not had experience with anybody like him and his brother.  He (J) has a soft, deep voice in speaking and singing but can be loud singing where necessary.  His brother (B) has a "normal" voice range but he again has to have the mic directly in front of him and no more than 6" away otherwise it drops off quick.  B isn't so much of an issue because I'm getting him an on the ear mic and he doesn't sing.
 
I can add a highpass when I get the new system, however I'm greatly concerned if I use a basic SM58, J will still have the proximity issue.  Is there any mic type that help with this?  I've seen reference to, but couldn't figure out if, something like a dual diaphragm mic would work or what would be a good one for this particular use case.

That will work but....  the main outputs of your mixer could potentially overload the mic inputs on the Yamaha. It's generally better practice to connect a line signal to a line input but inserting a DI between the mixers is usually a good idea as it transformer isolates both system, level and impedance matches the line signal to a mic input and gives you the option of a ground lift if there is noise on the link.

According to the specs, the main outs on the front of the EMX512C are line-level but unbalanced, which makes the DI a good bet. Being able to easily lift the ground between the two systems isn't a bad idea, either.

FWIW, any of the inputs on the MG12XU with a gain knob and a "line" label (either the XLR combo jacks with the pads enabled or the unbalanced TS connectors) will take a +30 dBu signal without clipping, so on its own level shouldn't be a problem. With a short, clean cable run (and no grounding concerns) you could probably get by with just a cable.

-Russ

ETA: Just for fun I ran some numbers, and if my math is right the MG12XU's +30 dBu maximum line-level input is only ~7 dBu below the EMX512C's maximum speaker-level output. Not that I'd recommend that particular connection, but as long as you weren't running full out there's a very reasonable chance you wouldn't blow anything up...