During the Dracula production, we have some student shows performing Peter and The Wolf. I decided to use the main PA (Renkus Heinz) since the system is going through some changes and I thought it would be good to make sure it works before we leave the building (while there is still time to fix it if it doesn’t).

Before we last Sunday afternoon after Dracula, the house crew brought the main PA back in to show position. All the Renkus Heinz systems were fed signal via COBRANET until Dave Lawler & Craig Doubet came in. Now all signaling is done via analog lines. COBRANET is only being used to control the speakers via RHAON. So I was the first person to hear the main PA work since the DSP units were swapped out (BIAMP AUDIA for BSS LONDON BLU 160). The good news is that it all works. The bad news is that the gain structure of the main PA is way off. I had to turn my main feed down roughly 32db to get back to where I am with the smaller Iconyx PA. Yikes! Today after the last student performance, Craig and I compared the Iconyx level with the main PA level and adjusted it back -20. In order to make sure I wasn’t guessing I placed a mic at the rear of the house (before the balcony) and took measurements of the HR Iconyx. Then the HR main array. No guess work. Craig adjusted so they are the same and I was able to return my Qlab channel back to unity. I’ll post traces when I’m back in the venue tomorrow. How did I ever get by without a measurement rig???

I’m back at the Winspear Opera House with Texas Ballet Theater for Ben Stevenson’s production of “Dracula”. This is a first in many way. It’s the first production Texas Ballet Theater has done with an orchestra in the Winspear pit. It’s also the first time Texas Ballet Theater has performed with the Dallas Symphony.

The sound system is in the process of being upgraded by Dave Lawler and Craig Doubet. The same team that upgraded & now maintains Bass Performance Hall’s sound system in Fort Worth.
docktrdaveaudio.com
Craig is currently rewiring the racks in the basement. Previously all patching was done in the basement and getting there and back is a 10+ minute round trip so the tie line patching has been moved into the sound booth. The original main system DSP (Biamp Audia) has now been replaced with BSS London BLU-160 units. In order to make sure work could be done but the Dracula production wouldn’t be affected, we coordinated what I needed in advance so only the systems that are required for my production were replaced first.
For Dracula I need over stage monitors for the dancers which is a simple tie line feed. I also needed the existing Renkus Heinz Iconyx main PA for preshow announcements and tech rehearsals without the orchestra. The lobby and backstage paging systems have been left alone since they must function through out the production.

I have to get ready for a tech rehearsal with QLab playback but I’ll post more information soon.

After taking Bob McCarthy’s SIM 3 / System Design class it is apparent that moving one or two measurement mics around the room (as I have been doing) is an inferior method compared with using more mics and leaving them in place during the measurement / tuning process. Logical but who owns more than one or two measurement mics? I happen to own 10+ Earthworks omni mics. No excuse not to start using more mics!

Bob stated that (5) is a mimimum for combining (2) speakers. These are the (5) positions. I will use my recent measuring opportunity for reference.

Going from one side of the room to the other:
OFF AXIS – A
ON AXIS – A
XOVER – A/B
ON AXIS – B
OFF AXIS – B

Bob calls these mic position classifications:

OFFAX
ONAX
XOVR

There is one more mic position classification Bob calls SYM (for symmetrically opposite element) that I’ll ignore for the moment since this post is about (5) mics.

The ONAX positions are used for EQ decisions. The XOVR position is used for delay decisions and to verify that a proper spatial crossover.

I’ve been watching a Meyer Sound SIM 2 machine being auctioned on eBay.

Having recently taken Bob McCarthy’s class and asking a few questions, I know that SIM 2 uses a keyboard only (pre mouse) and so everything is done with key commands. Having grown up with DOS, this isn’t as scary to me as it might be to a younger person. Meyer Sound always provides a lot of good information in their user manuals and I couldn’t find a manual online for SIM 2 so I wrote Meyer to ask. Voila!

SIM 2201 User Manual

Published February 01, 2015 @ svconline.com

The hard part about equalization is the guessing game, but Bob McCarthy has finally found the answer.
Bob McCarthy – On Sound System Tuning: Putting Equalization In Perspective

Published December 24, 2013 @ prosoundweb.com

There are lots of disagreements but all agree on one thing: You can’t change…
Bob McCarthy – Equalizing The Room – What It Really Means
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Page 3
Page 4
Page 5

Published June 11, 2014 @ prosoundweb.com

Matching loudspeaker coverage shape to the audience shape — the audience shape is visible, the loudspeaker coverage, not so much…
Bob McCarthy – Horizontal & Vertical: Attaining Uniform Coverage
Page 2

Published August 07, 2014 @ prosoundweb.com

A dartboard of 30 things I’ve learned in 30 years of being hyper-focused on one thing..
Bob McCarthy – 30 For 30: Lessons Learned From Years Of Tuning Sound Systems
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Published October 30, 2014 @ prosoundweb.com

The question of “to be, or not to be” reinforced should be weighed in the appropriate context…
Bob McCarthy – All The World’s A Stage…But Does It Need Sound Reinforcement?
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Published February 24, 2015 @ prosoundweb.com

Getting loudspeakers to play leap-frog in the forward direction and demolition derby on the back side…
Bob McCarthy – The End Fire Cardioid Subwoofer Array Made Visible

Published June 01, 2015 @ prosoundweb.com

What’s actually happening with a line of loudspeakers when it’s broken or tapered…
Bob McCarthy – The Impact Of Breaking The Line
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Published August 10, 2015 @ prosoundweb.com

A look into main loudspeaker system trade-offs involved in dealing with balconies, accompanied by design examples…
Bob McCarthy – Balcony Battles : When To SPlit Upper & Lower Mains
Page 2
page 3

In my quest to understand how speakers work, how various parameters are used to calculate building a speaker enclosure of a given size and shape, I am considering this software / hardware combination:

DATS Measurement and Accuracy PDF

True Audio – DATS

Parts Express – DATS

QUOTE:
Highly accurate measurement of loudspeaker impedance and parameters
Easy-to-use, fully-featured measurement software with intuitive interface
Compact USB measurement module with molded test leads and alligator clips
Software includes signal generator, scope, and measurement of inductors and capacitors

Data can be saved to create a driver parameter library or exported to popular box design programs
END QUOTE:

The more I learned about Sound Systems : Design and Optimization, the more I realize that until I understand how speakers are designed, how crossovers frequencies are chosen, how boxes are ported, etc… I can’t possibly grasp what my measurement rig is showing me. One of the things that is clear when you’re taking Bob McCarthy’s class is that Meyer speakers are well designed boxes. I see (1) Meyer speaker out of 1000 other brands, most of which are passive, non self powered and no where near the quality of a Meyer speaker. Meyer speakers are nice but if you don’t know how to optimize and tune a lessor speaker, how well can the process go for you? Long before I will see a Meyer, I’ll see a Mackie or a Yamaha or a Renkus Heinz or a EAW or a QSC. While I would prefer to work exclusively with Meyer speakers, it’s not gonna happen any time soon. What I am left with are typically 2 way passive designs. My current mission is to learn about how to design, build and optimize a 2 way loudspeaker. Fortunately I have a sibling that is way ahead of me. True, owner of Toby Speakers.

tobyspeakers.com

The first pair of speakers are going to be based on a coax 8″. Why? Because I already have (2). I’ll pull the spectral crossover board off the back and extend the 4 wires to locate it on the back of the box. This way I can play with both a passive configuration and also an active configuration where I can use DSP (digital signal processing) to time align / phase align the two drivers and also flatten the frequency response. Next I’ll build a non coax pair with the same design concept. Once I understand what to expect out of a two way loudspeaker, I have a three way design I want to experiment with.

One of the things that SIM 3 can do (using branches) is select the desired mic when you change to a different branch. For those of us who don’t have SIM 3 or if you do, don’t have the Meyer Sound mic selector, Radial Engineering’s Gold Digger mic selector may be a welcome addition to your rig.

Radial Engineering – Gold Digger

YOUTUBE – Gold Digger explained

Other than 48v phantom power and some relays there is no other circuitry. 4 mics in / 1 output. You can only select one of the 4 mics at a time so no concern of having more than one mic active. Since phantom power is provided, there is no loud explosive sounds coming from the mics when you switch. If you have Smaart 7, you can have more than one active live trace which is arguably better than having to view the results from one mic at a time but I can still see how being able to switch between mics may speed up the measurement process. If you have a 2 in / 2 out interface like the popular Sound Devices USB Pre 2, the Gold Digger mic switcher is a no brainer.

I’m reading user guide for Meyer Sound’s SIM 3 and thought I would share…

Meyer Sound – SIM 3 UG PDF

On PDF page 57 “Chapter 3: SIM 3 Configurations and Branches”:

QUOTE:
3.1. SETTING UP BRANCHES 3.1.1. Overview
Within SIM 3, the setup and management of the signal paths from console through processor to the loudspeaker and room are organized into Branches. A branch consists of three measurement points:
■ Console: The Console is the audio input into the sound reinforcement system (source), typically the output of a mixing console or other source. This measurement point is exactly the same as the processor input.
■ Processor: The Processor is the audio output from the signal-processing chain that lies between the console output and the amplifier input. Typically this chain includes equalizers and delay lines.
■ Microphone: The Microphone is the output of a measurement microphone located at some position in the venue that captures the audio from the loudspeakers within the room.
Within a typical SIM measurement session, the user will create many branches to have complete information on all the elements of the sound system and the interaction between them at different locations.
3.1.2. Defining Branches
A branch is a series of three connected measurement points that monitor the flow of a signal from origin to its destination. The normal flow is from a console output through a processor to a speaker in the room. The three access points are Console — the point of origin, Processor — the output of the signal processing chain, and the Microphone.
The primary requirement is that all three points are driven by the same original source signal. With this satisfied, the measurement points can be used to make transfer function measurements that will characterize the processor electrical response and the acoustical response of the driven speaker in the room.

END QUOTE:

Here is a diagram showing the “three measurement points in SIM 3”

One of the things that I have NOT been doing is taking measurements at the processor outputs. My typical measurement system is based on “CONSOLE / ROOM” as Meyer SIM 3 would label it. Without the “PROCESSOR” node, you have to guess how to shape your parametric EQ to match the room response. I know some system engineers who think that is fine but I’m having second thoughts. More information and less guess work saves time and time is something that is never unlimited.

SIM 3 lets you take measurements between CONSOLE, PROCESSOR & ROOM at the same time. If you’re using SIM 3 in tandem with a Meyer Sound Galileo or Galileo Callisto DSP unit, you have access to it’s input / output signals as well as muting using a single SIM BUS cable. SIM 3 relies on what are called “BRANCHES” to manage signal path routing and measurement points. Once the branches are setup correctly, you can move fast and efficiently between any 3 measurement points.

With Smaart 7, it’s possible to view an unlimited number of live traces (within the processing power of your computer). For example, one live trace could show CONSOLE TO ROOM response. Another trace could show the CONSOLE TO PROCESSOR response. Another trace could show the PROCESSOR to CONSOLE response, etc…
This functionality also allows for viewing the response of multiple measurement mics at multiple locations at the same time. Something SIM 3 cannot do. Within the limitations of your audio interface and the qty of mics you have, this opens up a lot of possibilities.

One of the measurement obstacles we are left to face with modern DSP is how to access the processor inputs / outputs. With analog processing, a Y cable is the solution. With DSP, the same Y cable may be used but there may also be a more efficient way of accessing the processor signals. For example, there may be a routing block that allows for selecting different signals routed to the same output. This is something I will be working on over the next few weeks. Having seen Bob McCarthy take great advantage of SIM 3’s overlapping CONSOLE / PROCESSOR / ROOM traces, I’m convinced. My days of guessing frequency, gain and Q settings to provide a complimentary EQ trace are over.

Consequently this evening I began assembling and cabling up a DSP rack based around an XLR patchbay and a recently purchased BSS London BLU 160. Once the rig is complete, using Smaart 7, I will be able to measure the (8) analog outputs of the DSP unit as well as the CONSOLE output and whatever measurement mics I’m using. Pictures of wiring explanation to follow.

Exciting news!!!

It appears that Bob McCarthy has revised Sound Systems : Design and Optimization and is releasing a Third Edition in the near future.

How do I know? It’s listed as available for pre-order on Amazon.

Amazon – Sound Systems : Design and Optimization Third Edition

Now we wait.

Now that the 4 day class has ended, I am left with my class notes and Sound Systems : Design and Optimization Second Edition to revisit the information Bob presented in class. One of the statements Bob made that seems vitally important is that he places the mics where he wants the speakers aimed. This is a much different concept than what I normally get to do which is tune the system AS IS. Where I put the mics where the speakers are already aimed. One means your speakers are aimed correctly when you’re done and the other (my current method) is a wild card left up to the previous installer / sound provider / etc…

Many times I have zero control over anything but system tuning via the console output EQ. Some times I am provided with zone control and have the option to adjust delay times between front fills and mains, underbalc delays and mains, etc…

The only time I have the opportunity to control the entire system is when I build it myself. This may be an install I’m working on or it may be a temporary system I’m providing for an event. Regardless, 75% of the sound systems I use are not mine to aim. I guess in that 75% section of my opportunities, tuning the system without the option of aiming it is all I can hope for.

What Bob provides in his book but explains in a way that I understand it better via his class is how to understand whether things are done correctly or not (regardless of whether I can do anything). If I can analyze a system and understand what is right and wrong while still having a successful audio event, that is a valuable experience.

Bob repeatedly states that the goal is to anticipate what you want to see when you measure something and be able to recognize (via a library of IF / THEN experiences) what is causing a different measurement result. In one case we fought trying to time align one speaker to another and the results kept on changing in the extreme HF range. Eventually Bob pointed out that the HVAC had recently came on and now we were seeing the results of a temperature change and air movement in the room. “A moving target” of sorts. Bob would adjust the time between the speakers for near perfect phase and then things would move the slightest amount. He would readjust the delay time and then it would change again. What to do in a case like this? Optimize as best you can and move on.

I am looking forward to trying what I witnessed Bob do in class on my own. I have a feeling Bob makes it look real easy. Hopefully between his book, my class notes and having watched him, I’ll be able to adopt the various protocols he follows to design and optimize a system. A system could be as simple as (2) speakers. Bob stated that once you understand how to combine (2) speakers correctly, a more complicated system is tackled using the exact same concepts and procedures.

I’m sure this post will continue to grow and divide so I’ll stop here and get back to reading Sound Systems : Design and Optimization again.

The last day of class was spent working on time aligning a sub to it’s main and time aligning a delay to it’s main. We also learned how some high end functions of the SIM 3 machine. For example, how to measure THD and how to do a “console check”. How to verify polarity. At the end of the day, we received a certificate of attendance and Bob signed books.

It’s always fun to be a participant in a class full of experienced audio engineers when it’s time to help pack up the class rig. I can’t think of a more copacetic group of professionals. Like ants in a colony…

If only audio gigs in general were that well staffed and organized

🙂

DAY 3

Current Temp: 67 degrees F
Humidity: 77%
Precipitation: 6%
Wind: 6 MPH
Pressure: 29.96 Hg
Dew Point: 61 F

I woke up earlier than I need to this morning. Lots of thoughts running around in my head about phase. Today we learned how to take a set of speakers and combine them correctly. Both as a unity pair and as a main / sidefill sort of configuration where the sidefill is covering a much shorter distance. The most important discovery for me was that a set of speakers may require a splay angle of X when coupled at unity but a different splay angle when coupled in a main / sidefill configuration. My assumption would be that the splay angle is fixed based on upon the cabinets used but this is simple wrong.

I’ll review my notes and add more soon.

Day 2

Current Temp: 99 degrees F
Humidity: 57%
Precipitation: 6%
Wind: 12.46 MPH / S
Pressure: 29.83 Hg
Dew Point: 61 F

There is so much important information coming in but a few highlights of Day 2,

We learned to decipher the SIM3 phase trace. With LF on the left and HF on the right, the general concept is that when you’re going down hill (breaking) you’re in the past. When you’re on the flat lands you’re in the present (coasting) and when you’re downshifting and applying throttle (climbing), you’re in the future. If you find a point on the trace where it goes vertical, you’ve experiences a rotation of phase. For example 180,360, etc…

For a phase corrected loudspeaker such as the Meyer ????, you may have only (1) vertical point which is near perfect for a 2 way loudspeaker.

INSERT TRACE

and how to find the -6 db contour of a speaker starting with ON AX NEAR, OFF AX NEAR, ON AX FAR, OFF AX FAR, and then working along a curve where you lose spl by traveling off axis but gain it by moving closer to the speaker.

If I can get some photos of the tape on the floor we used to graph out the contour, I’ll add it here. Same for one of the phase traces.

Today it finally cooled down a bit in the afternoon and then started to rain a bit in the evening.

Bob & Jason had dinner with Bill Eckenloft & his associate Declan and myself and discussed Broadway sound design.

DAY 1

Temp: 67 degrees
Humidity: 77%
Precipitation: 6%
Wind: 6 MPH
Pressure:
Dew Point:

Today is the first of 4 days with Bob “606”.

I’ll do my best to gather as much information as possible and report back each day.

The end of our Day 1 class morphed into a meeting between Harry Brill Jr’s Smaart 7 class and our group at a nearby tavern with a few guest SIM3 engineers attending (Dave Lawler & Craig Doubet). It was great to talk with people from the Smaart class and hear what they’re going over. If the classes hadn’t overlapped, I would of taken both classes…

Next time!

FYI, Harry Brill Jr. has some good reference stuff on his Tiger Audio website.

Rational Acoustics – Harry Brill Jr bio
Harry Brill Jr – Tiger Audio Inc

But enough about Smaart & socializing…

What did Bob talk about on Day 1?

1. Frequency / Time
2. Transmission speed
3. Transmission loss
4. Temp & Humidity
5. Frequency range
6. Wavelength
7. dB scale
8. Coverage angle
9. Coverage shape
10. Beamwidth
11. Summation amplitude
12. Summation phase
13. Coupled arrays
14. Uncoupled arrays

Even with lots of questions, we apparently made it through the desired information for day 1!!!

In case you’re wondering, pretty much everything Bob covered today is included in his book, “Sound Systems – Design & Optimization”
bobmccarthy.com – Sound Systems Design and Optimization book

We may have covered a few things that aren’t in the book due to questions but 99% of the information came out of the book. If you don’t have it, GET IT!

Stay tuned for DAY 2 post…

After hunting for some rigging hardware (from a rigging supply) yesterday, this morning I met with the ISD crew about the game plan for rigging the new speakers. We used 1/8″ aircraft cable, 1/8″ thimbles & the end terminations were made with 1/8″ copper sleeves using a Nicopress 63V-XMP crimpe. Each tool & sleeve combination has different specifications. For example, a different crimping tool may need more or less crimps to hold the total weight. For 1/8″ sleeves, you use the oval P groove & the termination requires (3) crimps. Each of the 3 crimps should measure .353 inches wide with a digital caliper. We used 1/4″ Crosby screw pin shackles which are rated for .5T / 1000lbs @ a 5:1 minimum safety margin. What this means is that you could hang 5000lbs from the 1/4″ Crosby shackle before it would break. Rated for a SAFE WORKING LOAD of 1000lbs, means that this 100lb speaker is going to be safe in the air and stay put. Since we have (2) cables on each cabinet, either one could fail and the full weight of the cabinet would be easily supported.

These EAW AX3 series cabinets are very back heavy so using only the front support points they hang upward toward the ceiling. We used a 3rd rigging point on the cabinet with 1/4″ shackles and rated chain for the rear pullback point which allowed for setting the proper downward angle.

Here is a photo with the (6) EAW AX3 cabinets flown and the old Peavey PA system uninstalled.

I mentioned in the first post covering this project that the existing PA was flown when it wasn’t designed for being hung. Here is a picture of one of the cabinets the ISD crew took down.

measurement traces to follow…

I fellow audio engineer invited me to go with him to figure out how to hang some better speakers in the gymnasium at his child’s school.

The current system has a few things wrong with it. It appears all the HF drivers are blown. The speakers are mostly aimed at the walls and ceiling and not at the seating areas. The existing speakers didn’t have rigging points to begin with so the ISD staff “customized” them and hung them anyway. In case you’re wondering, this is NOT acceptable. NEVER HANG A SPEAKER THAT IS NOT DESIGNED BY THE MANUFACTURER TO BE HUNG. You could kill someone.

Here are a few shots of the current 15 band EQ curve. Of special note is the 12db boost on the input gain followed by almost all boost on the EQ filters. Also note the LF cut fighting against the 25hz, 40hz & 63hz boosts. Good stuff!:) Maybe I can measure the current rig (flat and with the current eq settings) before we dismantle it.

Today we did a coverage analysis and marked the pending ceiling points using a self leveling laser. We also took some measurements using one of the EAW AX cabinets at center court. Tomorrow we’re going back to assist the ISD staff in hanging the new speakers, wire up the rig and do some temporary processing so the school has a working system for the first pep rally of the year.

We have (7) cabinets to work with so were using (3) per side in a L/R/L configuration.

More after tomorrow.

I’ve installed a number of Meyer Sound pa systems over the years. The Meyer loudspeaker I’ve hung the most is the UPQ-1P. The UPQ-1P is a two way self powered loudspeaker. I’m always amazed at how good a pair of UPQ-1P sound with or without Meyer subs.

Meyer Sound – UPQ webpage
UPQ-1P spec sheet PDF

With the pro sound industry using line array’s for literally everything regardless of whether it’s the right tool or not, it’s refreshing to hear a system that covers well, sounds great, doesn’t cost a fortune and is free from the inherent comb filtering of horizontal &/or vertical arrays. Fortunately there are designers that agree and aren’t afraid to use a single main box per side where that is the suitable solution for a given venue.

Here a few examples of UPQ based designs that Meyer has featured in the past on their website:

Oakland’s Mills College – Littlefield Concert Hall
Austin’s Continental Club
Houston’s Improv Comedy Club
Copenhagen’s Rhythmic Music Conversatory
Germany’s Nurburgring auto racking course

The Meyer UPQ series is muy bueno!

I was recently asked to take a look at the sound system at a small church across town.

Upon arrival it was immediately obvious that acoustic treatment is needed in order to make the space work with a sound system. Excessive low / low mid energy is an understatement. In discussing the matter with church staff, it was revealed that the space is rented and the church may move to a different space if there can’t be some agreement about long term leasing. What this means to the project is that whatever acoustic treatment is added needs to be portable and inexpensive. With that on the table, we have done some experiments in the space to figure out which speakers might be a good fit for the space. We tried a pair of QSC KW122 as a 75 degree sample and a pair of borrowed QSC K10s for a 90 degree sample. Arguably the KW122 pair sounds better than the K10 pair but the K10 pair covers better. It was decided that a pair of K10 will be purchased.

The current configuration is a pair of Mackie Thump TH-12A at the outside corners of the room.

A few issues with this setup. In order to cover the space, the current mains are cross firing which is washing the side and back walls. The speakers are actually audible from the podium so there is a chance for feedback. The speakers are so far apart that the imaging of the PA is outside the entire direct source zone (the stage). Sadly this is a typical situation. Where speaker placement is dictated by the layout of the space. The solution is to fly speakers overhead.

Today I returned to swap out their AudioTechnica podium for an Earthworks Flex mic (huge improvement) and also to do some corrective parametric EQ on the current system. The church owns a stereo 15 band graphic EQ which up until today was inline with the main L/R speakers. I loaned them a Yamaha YDP2001 parametric digital EQ / delay to cover the mains which freed up the graphic eq for the monitor send. The orange trace below is the flat trace measuring only the HR speaker. The Pink trace is a post eq measurement from the same mic position and with the same single speaker. Note that we took a few measurements with the new EQ and the results were fairly consistent.

The orange trace below is the same measurement as above (flat). The green trace is with the mic in the middle of the two speakers with them both producing sound.
One would expect a rise in low end when you measure both mains at the same time instead of only one.

After some corrective eq, the PA doesn’t sound terribly bad. It certainly sounds better than it did without corrective EQ. When you listen to music, it’s a bit on the thin side but that is a condition dictated by the acoustics of the space. Any low / low mid source sings in the space without any amplification. Consequently, the PA needs to avoid those frequencies. Once the church purchases the new speakers, we will install them overhead and locate them in toward the center of the space so the imaging is more accurate. Hopefully there will be support for adding some acoustic treatment which is going to take the venue to a new level.

Some lessons to take away from this project.

1. If the acoustics of a venue are bad, no amount of sound system is going to fix it
2. If the acoustics of a venue are bad, resolve them before you make final decisions about any sound system upgrades

In this case, the existing speaker positions are not ideal and the exiting speakers can’t be hung (no rigging points) so it’s a safe decision to get new speakers that can be rigged overhead and located inward. The existing pair of self powered speakers can become floor wedges. Stay tuned.

I just came across some product documentation that supports a concept I learned many years ago from Toby Guynn.

WIKI – midwoofer-tweeter-midwoofer (MTM)

The basic concept is that with a typical 2 way midwoofer-tweeter (MT) design, you have lobe tilting away from what you would expect to be the center axis of the loudspeaker. MTM

WIKI – Acoustic Lobing

WIKI – Loudspeaker Time Alignment

Take a look at the following image. It comes from the spec sheet for an EV FIR-2082 loudspeaker. Most manufacturers give us useless information like “40hz to 20k” “90 degree dispersion”. This is basically useless information.

Questions:

What is the frequency response between 40hz and 20k on axis? The loudspeaker may reproduce frequencies down to 40hz and up to 20k but at such a low volume that it’s irrelevant. Some manufacturers will state another spec that is “usable frequency response”. Yes please. Please tell me useful information instead of fantasy marketing specs.

In case it’s not already very clear, no speaker is direction at all frequencies and even the best speakers money can buy do not radiate sound at every frequency equally in the same dispersion pattern. A general rule of thumb is that LF is omni directional (a sphere shaped dispersion) and HF is narrow. MF are somewhere in the middle depending on the crossover frequency, size of the various drivers, spacing of those drivers, size of the loudspeaker baffle, etc…

QSC provides a -6db spec for their passive speakers. Let’s take the ADS12 speaker as an example:

Here is what they provide for the specs of that box.

While I can’t verify that this is a marketing ploy, their own graphs state clearly that the frequency dispersion is NOT stable through out the frequency response of the loudspeaker.

An industry friend and I were talking about the importance of having matched measurement mics and he mentioned this lab:

NWAA Labs – website

QUOTE:

MACH™ Testing
Multi Angle Computerized High-speed Testing™

Originally developed by NASA scientist, Ron Sauro, in 1968 for use in testing the Pioneer, Explorer and Voyager spacecrafts in addition to the mannned Apollo lunar program, the MACH™ Testing method has been reformulated into a revolutionary process for evaluating loudspeakers and materials. This cross-pollination of technology from the realm of space research to the world of sound and acoustics research has produced a high-speed testing system unlike any other.

The MACH™ testing method uses a proprietary array comprised of 19 matched Earthworks M-30 microphones, three custom modified Presonus microphone preamps and a Linear X precision turntable capable of one tenth degree accuracy during the rotation of the device under test. It simultaneously samples impulse responses within an extremely large, free field environment and uses these to create loudspeaker directivity balloons in a fraction of the time used by other testing methods. This data is gathered by a multi-channel version of EASERA, using appropriate stimuli, and then is converted to EASE V4, EASE V3, and CLF 1 and 2 formats for transmission to the client. EASE GLL formatting is also available for an additional charge.

END QUOTE:

Seems like a field trip is in order…

I know Meyer Sound – SIM3 measurement engineers who use a quantity of Josephson C550 mics with their rig.

Josephson C550 spec sheet PDF

This eBay listing is interesting because the seller is auctioning off (21) mics in various cosmetic states. I thought the images they provided on the auction were worth saving and sharing.
Without knowing if these mics have been as abused as they appear to be, I would be hesitant to take the chance unless I could get them really cheap and hope that some of them are ready for measurement purposes.

Years ago I helped a school replace it’s mono center cluster with some self powered QSC KW122 speakers configured as a R / L / R / L system.

The center of the old cluster was a JBL AS4735NW 3 way speaker. The NW has a passive crossover which allows it to be powered from a single amp channel whereas the STD version must be biamped.

JBL 4735 – spec sheet PDF
JBL 4735 – service manual PDF

The AS “architecture Series” is the install version (without handles but with rigging points). The same basic design is also available in the SR series is the portable version. In this case the SR cabinet is called an SR4735

JBL SR4735 – service manual PDF
Here is a product
JBL – SR series brochure

When it was taken down, it was put into storage and forgotten. Last year I tested it and discovered there was something wrong in the mid range. It went back into storage until today when I dragged it out again and remeasured it. When configured for passive crossover use, the cross over frequencies are 600hz and 2800hz. When configured for biamp use, you feed the LF 15″ driver directly an the MF / HF passive crossover covers that frequency transition.

Here is what the cabinet measures like AS IS.
This is a measurement with the LF driver unplugged
This is a measurement with the LF & MF driver unplugged (HF only)
This is a measurement with the MF driver unplugged (LF & HF)

In removing the 8″ MF driver, I discovered a few things. First, it’s housed in a sealed box inside the cabinet. Secondly, the paper cone has broken almost completely loose from the piston which explains why most of the mid range are mostly absent but not completely absent.

Having done some recent measuring at a venue with some passive 3 way cabinets (separate amp for each speaker & separate processing) and realizing how little I understand about how to configure an active crossover (analog or digital), I thought that once this JBL AS4735NW is @ 100% again, I can do some modifications to it to make it possible to easily turn the (3) speakers on and off externally. Also to choose between the passive crossover and an active (3) way setup. It will require some switches and rewiring but should make it possible to quickly test and understand what does what. How does a 3 way speaker behave and measure when the (3) drivers are in polarity and out of polarity? How does changing the crossover frequencies affect the sound of the loudspeaker?

My hope is that once I have seen how everything works in a controlled environment and I understand what to expect when things are wired correctly and incorrectly, I will be better prepared to recognize the same attributes in the field. If I need to verify a speaker after replacing drivers or discover something strange, it would be helpful to know that I’m on the right track and not chasing my tail.

If you can’t tell what is what working with a 2 way passive box, there is little hope of managing a 3 way active rig. I’m still in the “lots of questions” category of understanding a 2 way passive box.

While doing some recording in Steamboat Springs Colorado, I had the opportunity to measure a Bose L1 line array and B1 sub.

Bose L1s with B1 bass cabinet

Bose L1s with B1 bass cabinet – owners guide

The rig sounds pretty good. The blue trace below is the L1-II system without the B1 sub plugged in. The green trace is with the B1 sub plugged in.

I’m curious about the peak just above 250 and the hole below 250. Maybe the 24 x 3″ speakers in the line array are being high passed to protect them. Note that without the sub, there is nothing below 250hz and with the sub, there is excess low end. Also note that the 3″ speakers seem to naturally roll off around 8k. I was curious how the rig might respond to eq. Here is the curve I came to:

EQ result is the pink trace:

You will notice the HF shelf I used really doesn’t do very much. The 3″ speakers just don’t go up that high. I think it sounded a better with that slight boost but it’s really not necessary. The clarity of this system is surprisingly good. The strange thing going on at around 250 didn’t respond well to my EQ adjustment. Could be a room issue. My coherence trace suggested the issue wasn’t eq-able but I gave it a shot anyway. The important eq adjustment I would repeat is the LF cut centered at around 70hz to balance the sub with the line array. If parametric eq isn’t available, it should be possible to use a high wattage resistor to reduce the sub volume by a few db passively. One could build an assortment of inline speakon to speakon adapters that would allow for adjustment of the B1 sub level based on the acoustic environment. For example, if the rig is in a corner you may have even more excessive LF. If outdoors you may need to run the sub wide open.

I recently came across this Meyer Sound SIM 3 rig on Ebay. It had already sold but the listing gives a glimpse of how valuable a rig still is and shows some of the accessories that can be added to a base system to expand the number of inputs and outputs that can be compared:

QUOTE:

“This listing is for a used Meyer Sound SIM 3 audio analyzer system in perfect condition. The system includes one(1) SIM 3 3022 unit, one(1) SIM 3 3088 multi-channel line switcher and one(1) SIM 3 3081 8-channel mic switcher. Also provided is two 10’ DB-25 to XLRF and 2 10’ DB-25 to XLRM cables, Powercon connectors, a 2U drawer and a 6U rack case. This is a self-contained system that doesn’t require a computer as the proprietary software is running from the hardware. You only need to plug a VGA monitor on the back of the 3022 unit and a mouse on the front. The perfect solution for anybody wanting a self-contained portable system not tied to a computer.”

Here is a quote from “Loudspeaker Design Cookbook – 7th Edition (Vance Dickason) taken from page 8:

“Cone Directivity – As frequency increases all speakers become more directive and the high frequencies begin to “beam” like the light from an automobile headlight. At frequencies where the wavelength of sound (wavelength being equal to the speed of sound divided by the frequency = C/F, i.e. 1kHz has a wavelength of 1.13 feet) is large compared to the circumference of the cone (about 3 times the diameter), the radiation is spherical. As the frequency increases to the point where wavelength is equal to the circumference of the driver or smaller, the radiation patter becomes progressively narrower. The chart in FIG. 0.14 gives the =6db off-axis points for different diameter speaker diaphragms (after Daniels with changes, JBL Pro Soundwaves, Fall 1988)

WIKI – Directional Sound

QUOTE:

“The larger the speaker array, the more directional, and the smaller the size of the speaker array, the less directional it is. This is fundamental physics, and cannot be bypassed”

This is a link from that page and explains some higher reasoning behind the concept if you’re the type of person who can understand such things:

WIKI – Huygens Fresnel Principle

It appears that sound veteran Mark Dowdle will be @ FOH this weekend for Steely Dan’s concert at Gexa Amphitheater in Dallas Texas.

What does Steely Dan’s engineer use to check the PA system?

Find out here:

Pro Sound News – Pursuit Of Perfection: Concert Sound For Steely Dan’s Summer Tour

QUOTE:

“As with most FOH engineers of Dowdle’s vintage and background, Steely Dan’s catalog and Donald Fagen’s solo album The Nightfly are standards for tuning and evaluating sound systems. To answer the perennial question, “what’s used on a Steely Dan tour to tune the PA?” he replies that he uses Steely Dan live every day during sound check.
But prior to that, he plays Thomas Dolby’s “My Brain is Like a Sieve” from Aliens Ate My Buick and Frank Zappa’s “Lucielle” from Joe’s Garage, which has a very natural sounding vocal. A Shure KSM9 condenser mic then helps in the fine-tuning of its response.”

END QUOTE:

Funny. Aliens Ate My Buick is easily in my top 10 recordings of all time and I used it just last weekend in Allentown PA to check my PA.

Surprisingly I just got an email from Bill Whitlock of Jensen Transformers in response to an email I sent to Jensen probably a year ago about Pin 1 Problems. He explained to me that Radial Engineering recently acquired Jensen… You can read articles about Jensen and the acquisition here:

FOH Magazine – Jensen Transformers’ 40th Anniversary

CEPRO – Jensen Transformers acquired by Radial Engineering

In the meantime, the reason I was writing Jensen originally was to request permission to share an article Jensen produced regarding what was coined a “PIN 1 PROBLEM” and for which I secured and designed this website more than a decade ago:

pin1problem.com

Expect more regarding this matter as Bill and I exchange information on the topic.

WIKI – A/B Sound System

QUOTE:

“An A/B Sound System is a type of Sound reinforcement system or Public address system. Unlike a more typical sound reinforcement system, an A/B Sound System provides two electrically isolated signal paths from microphone to speaker, resulting in a system where signals from two microphones only interact acoustically and never interact electronically. This is accomplished by placing two separate loudspeakers at each speaker position and feeding the two speakers separate signals from separate microphones.[1] The purported benefit of such a system is a reduction in phase cancellation and intermodulation distortion, and an improvement in speech intelligibility when two microphones are used simultaneously. A/B Sound Systems are unusual because they require double the speakers and therefore have double the cost.”

The first time I ever heard an A/B system was on Disney’s first touring production of Lion King @ Bass Hall in Fort Worth. During the sound check and went to the farthest balcony to listen and was amazed that I could understand every syllable. When two actors were speaking, each one’s voice was routed to a different set of speakers covering the same zones. I spoke with the audio engineer at the time and learned that the choir was coming from a different PA and the orchestra from another.

Speaking of Lion King, the Broadway show as designed by industry heavy Tony Meola. Here is a Meyer interview with the man himself from 2002.
Meyer Sound – Interview with sound designer Tony Meola

Here is a run gear rundown and interview with Tony regarding the rig itself.
Meyer Sound – The Lion King on Broadway

Dave Rat – Demonstrates a Little known but very common Loudspeaker Distortion

This concept makes me lose sleep at night. This is solid justification for using different PA speakers to reproduce different material. For example having a PA for amplifying the band and another PA for amplifying the vocals. We should all be thinking about how low frequencies are affecting the rest of our frequency response.

I like Dave Rat. He is willing to invest his time in better and better sound and take what he learns from his own experimentation and incorporate it into his work flow.

His current Red Hot Chili Peppers PA is proof.

QUOTE: “With the double hung PA concept I was able to implement and demonstrate that having instruments and vocals coming from separate sources is not only an advantage for stage monitors but can be applied on a grand scale. Also, since I could move any instrument to either the outer or inner PA in real time, I became very aware of the important advantages gained in using separately located sources. With monitors, I would find out after the show if separate sourcing worked, with the double hung main system, I was in the listening position personally (along with 20,000 of my best friends) and could hear the immediate and direct effect.”

What Dave is doing with his PA system x 2 is called an A/B sound system and is sometimes used on Broadway shows. Dave may be the first rock & roll mix engineer to take the leap though.

WIKI – A/B Sound System

QUOTE:

“An A/B Sound System is a type of Sound reinforcement system or Public address system. Unlike a more typical sound reinforcement system, an A/B Sound System provides two electrically isolated signal paths from microphone to speaker, resulting in a system where signals from two microphones only interact acoustically and never interact electronically. This is accomplished by placing two separate loudspeakers at each speaker position and feeding the two speakers separate signals from separate microphones.[1] The purported benefit of such a system is a reduction in phase cancellation and intermodulation distortion, and an improvement in speech intelligibility when two microphones are used simultaneously. A/B Sound Systems are unusual because they require double the speakers and therefore have double the cost.”

The first time I ever heard an A/B system was on Disney’s first touring production of Lion King @ Bass Hall in Fort Worth. During the sound check and went to the farest balcony to listen and was amazed that I could understand every syllable. When two actors were speaking, each one’s voice was routed to a different set of speakers covering the same zones. As I recall, the choir was coming from a different set of speakers and the orchestra from another.

Speaking of Lion King, the Broadway show as designed by industry heavy Tony Meola. Here is a Meyer interview with the man himself from 2002.

Meyer Sound – Interview with sound designer Tony Meola

It may seem trivial to test a common mic cable but there is more to a mic cable than 3 pins. Depending on how those pins are connected on both ends, you will either have good audio, no audio, bad audio or low level audio. A incorrectly wired cable might be acting like an antenna.

Meyer Sound has provided us with a SIM3 procedure for doing just that.

SIM3 measure XLR cables

Not all of us have a SIM3 machine sitting around in which case the Rat Sound – Rat Sniffer / Sender Cable Test Kit is a good option instead.

The RAT SOUND – XLR Sniffer / Sender Cable Tester kit

SoundTools – XLR Sniffer / Sender kit

I have 4 of these floating around. Since the two parts can be located away from each other, it makes it simple and fast to test things between the stage and FOH without running a long XLR cable which would be necessary with other cable testers. A set cost around $50. Well worth it.

This is a guide to what the 3 LED lights indicate:

Here is a youtube video about the Rat Sniffer / Sender with Dave Rat.

Dave Rat – Rat Sniffer & Sender – A remote end XLR Cable Tester

The Miller Symphony Hall is an interesting venue. Starting out as a farmers market and becoming a theater in 1899, it was one of the leading burlesque halls in the US in the early 1900s.

WIKI – Miller Symphony Hall

Miller Symphony Hall

Some history about the venue

As the photos above reveal, in addition to the orchestra level, there are (2) relatively deep balconies. There are no rigging points to hand a PA over the stage. Consequently my biggest challenge was no running the PA too loud for those up close. Anytime you have a ground stacked or stage stacked PA, the people up close to the stage are going to be subjected to excess volume if the level is at show volume in the rear of house seats. That is physics and there is no way around it. Ideally a main PA is flown which evens out the volume discrepancy. In the case of the Miller Symphony Hall, there are no rigging points.

What to do? We would need to under balcony speakers and over balcony speakers to allow us to turn down the main PA and make up the different. This may actually be possible thanks to the lighting positions at each level.

My ultimate goal is never to hurt anyone ears and to avoid sound complaints. I’d rather get a “it’s not loud enough” complaint than “it’s too loud” complaint. In this case, I went out of my way to make sure the symphony and venue staff understood my concerns, understood what I could and couldn’t do and to know that I was receptive to any complaints at any time. I even met with the head ushers to explain what to expect and what to do if someone complained.

The fact that wasn’t a single reported sound complaint can be attributed to a few things. A PA that was optimized for the room (linear frequency response to the best of my ability) and the choice to leave everything a little on the dull side where I was standing. It’s counter intuitive to mix where you can’t really hear clearly what is going on but that was necessary in this case. I made probably 50 trips to the front row orchestra level to get a sense of how my mix would translate up close to the PA during the band sound check and orchestra rehearsal. Two things were obvious issue. Loud up close compared to the mix position and more present. So over the course of the sound check and orchestra rehearsal, I cut some 3150k from pretty much every channel. In hind site, I could of done the same thing at the stereo L/R buss but I never know what is needed when I start and after a while I’ve tweaked everything at the channel. Fine. More important to get through the rehearsal process and be ready for the show than to second guess every choice that is made along the way.

A few things that made the show work. First was the use of IEMs by (3) of the (4) band members. The 4th band member used hot spots which are great because they sit on a stand at chest height so they don’t have to be very loud. They also don’t reproduce a bunch of low end so there is no inherent mush involved.

There is never enough rehearsal time with the band. There is never enough time with the orchestra. How does one deal? I keep on working until I know I’ve resolve the issues I know are going to pop up. After the band and most of the orchestra left the building, a first violinist came back to practice. I politely went up on stage and said, “if you’re going to play, will you put your mic back on so I can tweak your channel?” He said sure. He probably played solo violin for 30 minutes which gave me a really good chance to walk around and listen and fine tune the EQ on his channel. A trombonist also returned and I got to work on his channel which needed some work. Lastly the harp player came back and she played for probably 30 minutes. Another rare opportunity to adjust her channel. Harps are a hard instrument to manage. They can be overly percussive at times and overly soft / dull sounding too. They can play very low in pitch and play very high in pitch. Consequently, a harp is one of the instruments I need the most time with (if I can get it). Those (3) musicians serendipitously made my job easier and made the show sound better. Thank you!

The other thing I did was spend the remaining time until dinner reconfiguring the custom layers on the Yamaha CL5 console (my favorite new console). I made a few poor decisions when building the input list which came back to haunt me. Using the custom fader layers, I was able to correct those to some degree. I was also able to setup some custom layers for a few specific songs that require some faders being close. For “Yesterday” I have (6) active channels.
Violin 1-1
Violin 2-1
Viola 1
Cello 1
acoustic guitar
Paul vocal.

I used a custom fader layer to put those channels side by side.

For Eleanor Rigby, I build a custom fader layer as follows:
V1-1
V1-2
V2-1
V2-2
Va 1
Va 2
Cello 1
Cello 2
Paul
George
Ringo
John

Custom fader layers allowed me to mix a show I didn’t really have mastered without making any major mistakes.

The Yamaha CL5 is cool in the fact that you can add channels or DCA to a custom fader layer. Very flexible. If you haven’t spent any time with a Yamaha CL or QL console, you owe it to yourself to try them both. Every console has it’s benefits and downsides. Yamaha did a good job with the CL and QL series. My only wishes would be a higher channel count (96 please instead of 72) and 24 DCA instead of 16. Seems selfish of me but that is what I need to do this gig well.

My typical measurement rig consists of (2) Earthworks TC40K (a matched pair), my Mac laptop and a Metric Halo 2882 audio interface. Software could be either SpectraFoo Complete or Smaart 7.

On a recent gig, I put out the two mics. One about 1/3 of the way back in the house from the HL line array. I put the second mic near the FOH position to get an idea of how I would need to mix and to see how different those two positions were. I adjusted the gain on both mics to match which I realize now isn’t always the right choice. If a mic is close to a speaker and you move the mic back, you would expect to see a drop in overall level as you move further and further away from the speaker. So what? This decrease in level tells us the decrease in level between those two points. Where before I had a gain setting of 35 on the close mic, I had a gain setting of 40 on the second mic. 40 minus 35 = 5 db of SPL between the first mic and the second mic. While matching the gain of the mics allows to see the traces overlap for comparison, knowing how much the SPL drops off by the time you’re standing at FOH is equally important.

In sound terms, this means that each time to double the distance away from the source, you 1/2 the volume. Moving from being located 1 foot away from the speaker to to 2 feet is 1/2 as loud. Moving from 2 feet away to 4 feet is 1/2 as loud again, etc… This doesn’t work exactly like this in reality because some frequencies are supported by near boundaries while other frequencies are not. So the drop in SPL with the doubling of distance is frequency dependent. Indoors, you can expect the low end to drop off slower than the mid / high frequencies.

This is why it’s better to use more speakers instead of one set of huge speakers unless you can fly them high enough to even out the distance between those up close and those far away. This is where delay rings come in.

If I were to make the same measurements again, I would make the gain for both mics equal which would reveal how much sound is being lossed by the time it arrives at FOH (back corner / under a balcony / etc…)

INSERT PHOTOS

I just finished a Classical Mystery Tour show in Allentown PA with the Allentown Symphony Orchestra. This gig was yet another reminder that without my measurement rig and more importantly the skills I have acquired to know how to use it, I would be blind. It took (8) well placed parametric EQ filters to make the EAW KF730 PA play nice with the room.

The mix position was near the back wall on the HL side of the hall. I was at least 20 to 30 feet underneath a balcony. I placed a mic at about 1/3 the distance of the room away from the stage stacked HL array. I used that mic for my “what do people hear up close to the PA” reference. I located a second mic next to FOH for my “what am I going to hear” reference.

Orange trace indicates HL array without any board EQ.
Light blue trace indicates the system response after EQ in the nearfield.
Yellow trace indicates the system response after EQ in the far field.

This indicates the impulse response in the nearfield

This indicates the impulse response in the farfield

I used the near field mic to EQ the main PA as follows.

Band 1 – 80hz / -18 / Q 6.3
Band 2 – 100hz / -6 / Q 4.0
Band 3 – 200hz / -9 / Q 2.5
Band 4 – 300hz / -9 / Q 5.0
Band 5 – 900hz / -4 / Q 3.2
Band 6 – 1.7k / -4 / Q 4.0
Band 7 – 10k / -8 / Q 4.0
Band 8 – 14k / -6 / Q 4.0

After finishing the EQ adjustments, I did a good bit of walking in the house to make sure that chosen EQ worked through out the venue. Things seemed relatively even and other than the expected SPL loss in the far seats.

A local tech school is in the market for a new PA system. Someone in the organization is leaning toward a line array. The space has 20′ ceilings (low) and is short but wide. Sometimes a line array PA is the best option. Sometimes it’s not. How does one know? There are a few well known facts about line arrays. They need to be a certain length to behave well. The number I continue to hear is 10 to 12 feet. Longer is OK but shorter is not. There is a trend to use line arrays regardless of the circumstances and this post aims to reveal why that is a bad decision to make.

Here is a model of a Meyer UPQ-1P

Here is a model of a Meyer 3 box JM1P 60 degree x 60 degree array (can be rotated in either direction with the only difference being the location of the overlap seam)

Here is a model of a Meyer 4 box Leopard line array

Here is a model of a Meyer 5 box Leopard line array

Do people use short line arrays? Absolutely. Do they sound good? Questionable. If you choose a PA system based on sound quality and you don’t need the high SPL levels a line array can provide, a pair of UPQ-1P with some front fills is the optimal PA for the space. If you look at the models, the same is true. If you neither listen or model and go based on current trends, you’re likely to have a short line array even though it doesn’t sound or behave as well and costs more.

Obviously the most important aspect of choosing a PA system is sound quality. When you can’t model the PA you are considering, I would suggest you demo the PA systems you are considering and then let your ears be the deciding factor. Not your eyes. Not your expectations of what looks impressive. Not what a sales rep suggests. Use your ears…

I’ve always wondered what it would cost to get into a Meyer SIM3 rig and recently I decided to check into it. After all, “knowing is half the battle” as GI Joe says.

What does a Meyer Sound – SIM3 machine cost?

As of 2015, retail is $13,430…

Now add measurement microphone / microphones, VGA monitor, softrack or case, etc…

The engineers I work with who have SIM3 rigs have racks that fit inside a pelican case for transport.

Spent the morning and afternoon flying to Allentown PA for a Classical Mystery Tour show tomorrow. Fortunately the crew was already mostly set up when I arrived so tomorrow will be a much more enjoyable experience. Same day loadins, setups, sound checks, orchestra rehearsals, shows, strikes are to be avoided when ever possible.

Sound provider – Bluechip Sound.

Blue Chip Sound

The main PA consists of a (6) box per side ground stack EAW KF730 Line Array, (2) EAW subs. EAW processing. Monitoring is a mixture os IEM, EAW Microwedges and Galaxy HotSpots. (2) Yamaha CL5 consoles (FOH and monitor land), (2) RIO 3216 stage boxes for 64 x 32 inputs / outputs @ the stage. I used (2) local inputs to feed stereo music and pink noise into the system. I placed (2) Earthworks TC40K measurement mics. One on axis of the HL array about 1/3 of the way back in the house. The other on axis but next to FOH. This way I could easily compare the difference and understand whether or not my adjustments in the nearfield worked near the back of the room. FOH position was a sort of worst case scenario. Near the back wall, in a corner, under a deep balcony.

During the setup period, I was able to get a basic idea of what the system measured like without any board EQ. I was shown that the CL5 has an 8 band parametric EQ which I inserted on the stereo L/R bus. I’d already tried using the stock 4 band but couldn’t achieve what I needed to do.

The venue is an old burlesque theater built in the late 1800s. It has been renovated but the actual theater itself is mostly untouched.

One important distinction between Meyer SIM3 and the other measurement systems I’m aware of is SIM3’s ability to display a PROCESSOR (EQ) trace at the same time it is displaying the console to microphone trace.

SIM3 refers to the (3) measurement nodes as “CONSOLE, PROCESSOR, MICROPHONE.”

Here is what Meyer has to say,

QUOTE:

“3 is the Magic Number

A capability unique to the SIM 3 audio analzer system, these three transfer functions work simultaneously to provide real-time data acquisition. The SIM 3 system starts by measuring the effects of the electronic or acoustical signal path through a comparison of two points in the signal chain, most commonly combinations of the mixing console output, equalizer or digital signal processor output, and a microphone placed in the room to capture the sound as heard by the audience. Three transfer functions using state-of-the-art, dedicated hardware and software processing make possible a slew of computations and measurements, delivering a wealth of information in both frequency and time domains.

You can select three different frequency response measurement views:

Room + Speaker – the unfiltered system response, measured by comparing the signal processor output and microphone
Processor – the signal processor and inverse processor response, measured across the processor from input to output
Result – the corrected system response, measured by comparing the processor input and microphone”

The CONSOLE to MICROPHONE trace displays the difference between what is leaving the console and what is measured by the microphone. The PROCESSOR trace provides a reference between the CONSOLE output and the PROCESSOR (EQ / DSP) output. The PROCESSOR EQ / DSP output is inverted so that when you adjust your EQ, if you create a curve that matches your console to measurement mic response, you will have created an exact opposite EQ curve. This EQ curve is called a “complimentary EQ curve”. With SIM3 you can visually see when you have achieved a “complimentary EQ curve” and the result at the microphone. This is obviously an important feature of SIM3 and has set it above the rest.

One of the things I have learned from SIM3 users is that once SIM3 has been configured using branches (preconfigured measurement nodes), one can manage large measurement projects with ease. When using SIM3 mic selector

With programs such as Smaart and SpectraFoo, you are able to see information but not at the same time. Basically with SIM3 you’re able to look at the whole picture at the same time. With other measurement apps, you must set up 2 different measurements and move between them.

Here is a diagram of what SIM3 does.

One of the things I learned long ago is to tap on the sides of a speaker enclosure to get a sense of how well built it is. An ideal speaker would transfer 100% of the vibrations from the drivers into the air and not into the cabinet walls. Concrete would be a superior speaker enclosure material but the weight factor renders it useless except in the extreme high end HIFI realm. For most us, MDF / plywood / hardwood are going to be the materials used to build an enclosure. There are other enclosures built out of similar materials. They’re called drums. While a resonant drum is desirable, a resonant speaker enclosure is not desirable. In order to avoid speaker enclosure resonances (when the enclosure starts to resonant along with the sound it reproducing), the speaker enclosure needs to be properly braced and dampened. Expensive speakers are typically well braced and dampened. Inexpensive & lightweight enclosures typically are not well braced and dampened.

With the tap test, you can get an idea of how an enclosure will behave when it’s reproducing sound. If a speaker sounds like a drum when you tap on it, it will resonate the same way when reproducing sound which is NOT a good thing. Speaker enclosures are NOT drums and drums are NOT speaker enclosures. One requires different construction and design than the other.

Meyer Sound has acoustic modeling software called MAPP and recently released a new version called MAPP XT.

MAPP XT

I spent more than an hour last night learning how it works. MAPP allows you to move speakers and mics around in a virtual venue and see what happens via the frequency response / impulse response and amplitude / SPL windows.

This model is of a single Meyer Sound – UPQ-2P in a medium sized room with one measurement mic. The model is showing what the response of that cabinet @ 10k. Pretty nice.

This is what you get when you enable the 4 boundary reflections. Not such a pretty design.

One of the reasons MAPP XT can provide such useful information is because Meyer has done all the data gathering in advance. When you click on “PREDICT” in MAPP, it’s not calculating things locally on your computer. It’s actually calling into a server at Meyer Sound, doing the number crunching and then sending the results back to your screen. The true test of a model is whether or not what the model shows reflects what happens in the field. Does the install work as designed and modeled? How different is the final install compared to the model? My experience installing a lot of different Meyer PA systems in a lot of different venues (all modeled in MAPP) is that things are typically within a few degrees and a few percentage points of what the model showed regarding positions, angles and coverage. Amazing!

Meyer MAPP XT is free. They do require registration but that’s a simple process.

Here is a step by step instruction for getting up and running quickly:

MAPP XT tutorial

Once you have the software, you’ll want to watch the video tutorials to learn how to use it. Meyer’s video tutorial library for MAPP can be found here. Click on the Educational Videos link to access the videos:

MAPP XT resources

In a previous post I explained how the existing house speakers were found to have issues and had to be temporarily replaced. Anytime you get to try something different in a space is a great opportunity to learn whether the new rig works better or worse than the previous rig. In this case the original rig consists of (2) Renkus Heinz TRC151/9 acting as a stereo L/R. These speakers have a 90 x 40 horn that can be rotated. The horns have been mounted with the slot in the vertical position up to this point but is that right horn orientation for the space? Is 90 degrees of horizontal coverage enough or too much? If we rotate the horn so that we have 40 degrees of horizontal coverage is 90 degrees of vertical coverage too much?

The speakers used to temporarily replace the damaged TRC151/9 are QSC KW122 which have a 75×75 degree horn. Obviously if a 90×40 is the correct horn for two speaker coverage of the space in the horizontal plane, two 75×75 isn’t enough. If two 75×75 is adequate, 90×40 is too much in one plane and not enough in the other.

To make matters more complicated, a while back a center speaker was added. A QSC KW122 to be specific. So previously the rig consisted of (2) 90×40 TRC151/9 and (1) KW122 in the center of those. Now the rig consists of (3) KW122s. When the change was made the decision was made to abandon the “stereo” concept because there is very little “stereo imaging” to be had in the space. The current configuration treats the space like a (3) zone mono PA. There are (3) sections of seats separated by (4) isles. Anytime you have the same sound source covering the same territory as another sound source, you’re creating comb filtering at the overlap zone. If you want “stereo”, you deal with this sonic hit but once you can make peace with the fact that for very few “stereo” isn’t stereo at all but instead a sonic mess, you can begin to design and configure without false notions as your guide. Currently the Scott Theater PA is mono and (3) zones. Fine. Not surprisingly the PA sounds better in this configuration. The FOH position is covered by only (1) of the (3) speakers (a first). There is minimal overlap and most of it is happening in the isles.