In order to understand what a transfer function measurement is displaying, you need to understand what the impulse response / phase response / frequency response means.

Abbreviated, IR / PR / FR:

In my ignorance, I basically ignored the IR and went straight for the FR (incorrectly thinking of a FFT measurement as a RTA of sorts). Knowing better now, the IR is the very first and most important trace. The IR trace reveals acoustics and secondary reflections that may be caused by poorly aimed speakers, poorly placed measurement mics, etc…

The IR can also provide information about polarity. If the impulse trace shows a peak that goes negative instead of positive, you may have to resolve that issue before doing anything else.

It’s essential to be able to read and understand what an IR trace reveals.

Once you have an acceptable IR (which also provides a time offset between source and response / reference and measurement), then you can move on to the PR and FR traces with confidence that you aren’t wasting your time.

There must be a better way to tell a house person that their PA has issues than just telling them their PA has issues because doing so wasn’t met with much applause.

During the Dog Days loadin / setup it was discovered that the HR main had an issue. What issue? No lows and no highs but nothing blown???? Strange but very noticeable.

Ultimately we took both the HL and HR speakers down (after labeling them) to attempt to fix the HR speaker and to measure and verify what the issue is. With a working HL speaker that is the same make and model, it will be obvious if there is something wrong.

I have no doubt there is something wrong. I just don’t know what it is yet. Once I measure them both, I’ll post the results and hopefully correct the issue and provide new measurements once they are both fully functional again.

More soon.

On Thursday we began the load in process for Dog Days (the opera). On Friday the traveling crew arrived. The sound designer discovered that something was wrong with the HR main speaker. No lows and no highs. The box is a Renkus Heinz (????) which has a 15″ woofer and 2″ driver on a horn. I’ll know more once I measure the cabinet against it’s working counterpart but my guess is that a component on the crossover has failed and the phase between the drivers is off now. Both components are working so it’s not a blown speaker / driver. What do you do when you find a malfunctioning house speaker? Take it down and put up a different one of course!

The system consists of the following:

(2) QSC KW122 speakers hung as L / R for mains.
(1) TOA HX5 4 box mini line array for a center cluster graciously left behind by TCU’s Bill Eckenloft.
(2) QSC KW122 speakers hung for side of stage monitors.
(4) Mackie SRM150 hotspot type speakers in the orchestra area.
(2) Mackie SRM350 rear of house speakers for the helicopter

Today we tuned the system.

PHOTOS TO COME

Even if there are only a handful of visitors who are actually interested in audio measurments and the rest are google and microsoft searching for information, audiomeasurement.com has hit a milestone of 10,000 + visitors.

Cheers!

I’m back in a local Dallas venue that has a Fulcrum Acoustic sound system.

The system consists of:

(2) DX1265 for upper main L / R
(2) DX1295 for lower main L / R
(4) US212 subs located in cavities under the stage.
(3) CX5D for under balcony fills
(7) CX5D in the face of the stage for front fills

There is a center cluster that is not a Fulcrum product.

(1) D&B TI10L 5 box line array center cluster

d&b audio – Ti10L webpage

This is my 3rd visit to the venue with Texas Ballet Theater. Previously I have known that the system wasn’t optimized but didn’t know enough to know what to do about it. I also didn’t have a relationship with the house crew to start digging into their settings. I have simply measured and Eqed at the board as best I could.

This time, there is enough trust between myself and the house crew to get into the DSP settings and discuss what might be wrong and what might be done to make things better.

What are the issues? The PA appears to have issues of speaker placement and aim. Poor coverage on the sides, uneven response, uneven levels, overlapping coverage, possibly incorrect delay times, speakers that are reflecting off near by surfaces, subs located too close to specific seats, subs mounted in resonant chambers, etc…

If the proof is in the pudding, it simply doesn’t sound very good. The good news is that I think the existing system can be greatly optimized with proper aim & correct processing. Neither of which requires much effort and cost to correct.

Since I haven’t finished gathering the necessary information to go forward with the verification process yet, I’ll hold my final judgement but in the meantime, here are some interesting images…

This is the DSP block that is processing one of the DX1265 cabinets.

HPF

LPF

LF EQ

HF / LF EQ

HF / LF FIR (Finite Impulse Response)
WIKI article – FIR / Finite Impulse Response

This is interesting,

The upper DX1265 Main L / R boxes have a delay between the HF/LF & LF driver of .438 ms:

The lower DX1295 Main L / R boxes have a delay between the HF/LF & LF driver of 1.479 ms:

Same box, same dual 12″ LF drivers. The DX 1265 has a 60 x 45 horn. The DX1295 has a 90 x 45 horn. What accounts for the 1 ms+ difference in delay times between lower and upper boxes (between the LF and HF / LF drivers) ???

The Fort Worth Opera festival loadin starts tomorrow. Dog Days is one of the operas in the festival but it’s being produced in a different venue. The Dog Days load in starts on Thursday.

WIKI article – Dog Days opera

NY Times – Dog Days

Here is the input list that came with the show:

I have some of these mics and some will be substituted for arguably “better” mics. Some will be substituted with arguably “comparable” mics.

This afternoon I setup a mic test rig using a QSC KW122 self powered speaker. I used the speaker because it is relatively flat and being self powered, took about 1 minute to have pink noise coming out of it. I set up a mic stand at roughly 6′ away from the speaker. The speaker’s HF horn was located at roughly 6′ off the ground with no objects between the mic stand and the speaker.

Note that the KW122 has a 3 way switch on the back. The choices are “Ext Sub” (reduces low end), “Normal” (flat response), “Deep” (extends low end). After measuring those 3 settings with the first mic, I selected “Normal” for all subsequent measurements.

This trace shows what the switch does:

INSERT TRACE HERE

Then I proceeded to take some reference measurements using my matched pair of Earthworks TC30K omni mics. Then I measured with my newly acquired Audix TR40 pair (non matched). After that I went through the rest of my measurement mics including an Earthworks M30BX (self powered), TC25K omni and then moved on to the mics to be used in the production.

This includes:

AKG 535
AT 4033
Earthworks SR20 / SR69 (same thing but older model)
Earthworks FM500 (flex mic)
Oktava MC012 / MK012 (same thing but newer model)
Sennheiser E604
Shure SM57
Shure SM58 / SM58S (switchable)
Shure Beta 52A

The Oktava bodies can work with different capsules so I compared all (6) cardioid, (4) omni (2) hyper cardioid capsules.

All in all it took about 3 hours to go through the process of documenting the serial numbers of the mics, measure and save the data.

INSERT TRACES HERE

The Fort Worth Opera festival loadin starts tomorrow. Dog Days is one of the operas in the festival but it’s being produced in a different venue. The Dog Days load in starts on Thursday. Maybe by next week I have some time to update this page and add the measurement data…

My AMPROBE SM-CAL1 Sound Meter Calibrator just arrived and I thought I would figure out how to setup an accurate SPL meter in Smaart 7. Fortunately, Rational Acoustic’s “Calvert Dayton” has already provided instruction on how to go through the process via a post on the Rational Acoustics forum. Calverts response is #7.

Re: SPL –measurement does not what it should

Instructions as follows:

1. Open the Amplitude Calibration dialog window by clicking CalibrateSound Level Options button in the dialog (Options > SPL/LEQ…).
2. Select the audio device and input channel that you want to calibrate.
3. Turn on your calibrator and wait a moment for the average to stabilize — say 10-15 seconds for a microphone calibration using an SLC, maybe 20 to be on the safe side. 20 seconds would be enough time to build from -138.5 dB (all 0’s in a 24-bit integer) to 0 dBFS (all 1’s), given a 0 dBFS reference tone.
4. Click the Capture button.
5. Type your nominal reference level (e.g., 94, 104 or 114 dB) in the Calibrated Level field — when performing a microphone calibration in the field you generally want to use the highest output level that your sound level calibrator offers.
6. Hit the Enter key to recalc the Calibration Offset.
7. Click the Apply button.
8. If you want to calibrate additional inputs, go back to step 2 and loop through again. Otherwise click the Done button to exit the dialog.

Here is another website that described the method:

Smaart V7 calibration for SPL

I have a production coming up where the visiting engineer wants a Shure Beta 52a for the kick drum. I could substitute an arguably “better” mic but I happen to like the sound of a Beta52A on kick drum and it around $125 street price, it doesn’t hurt to have an industry standard mic in my collection. One of the things that manufacturers do to “kick drum” specific mics is to PAD them down so they don’t overload an input. There is usually some extra low end involved.

As a reference, there are many kick drum specific mics on the market and then some non kick drum specific mics that have been used for that purpose such as the EV RE20, Sennheiser MD421 and Beyer M88.

Here are some articles on kick drum mics, none of which provide more than opinion about the difference between the mics they compared:

recordinghacks.com – kick drum mic shootout

gearslutz.com – 11 kick drum mic shoot out

soundonsound.com – bass drum microphones

What does it all mean without measurements to see what is what?

INSERT Beta52A measurement data
INSERT D6 measurement data
INSERT RE20 measurement data
INSERT MD421 measurement data
INSERT AKGD112 measurement data
INSERT E901 measurement data
INSERT Earthworks SR25 data
INSERT Earthworks TC30K data (for reference)

Unfortunately, there is a limited amount of useful information currently on Youtube regarding audio measurements in general. Consequently, I’ve geared up to build a Youtube video channel specifically related to audio measurements.

I’ll be using Camtasia screen recording software to capture the measurement process, add titles as well as edit and assemble non screen video and still media together.

Camtasia webpage

If you have an idea for a video that explains something you want to learn regarding audio measurements, contact me.

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I just came across some old install photos and thought I would share.

A while back I hung a new sound system at “Pirates Voyage” in Myrtle Beach South Carolina.

piratesvoyage.com

We replaced the existing EAW PA with a Meyer Sound rig designed and provided by Pro Sound out of their Orlando Florida office.

In reviewing the photos of the old rig, I was reminded of what appears to be “acoustic treatment” on the backsides of the EAW arrays.

I can only guess that someone was trying to reduce some of the low end / low mid content coming off the back of the cabinets. The thickness of the padding appears to be about 2 inches. I would argue that 2 inches of padding will cause zero reduction of any sound coming off the back of the cabinets as that sound would be omnidirectional anyway and completely ignore the padding. Whatever resonance was the culprit might of been better served by putting the padding on the inside of the cabinets.

If you want to know how a speaker will behave when it’s producing sound, tap on the sides of the speaker (any speaker / any side) with your knuckles. If it is relatively muted and of higher pitch, that suggests a well braced speaker. If you excite the walls of a speaker when you tap on them and it sounds like a drum, it will sound like that same drum at certain frequencies when it produces sound.

A perfect speaker cabinet wouldn’t vibrate at all. Think of a speaker cabinet being made of concrete. There are some home stereo speakers made of concrete but for professional applications, concrete speakers just don’t make any sense. Speaker design is a balancing act. Cost, weight, performance, etc… IF you brace a speaker cabinet correctly you don’t need to use concrete to get a relatively vibration free cabinet.

If you mount a speaker inside a structure that doesn’t vibrate at all, then all the energy of the speaker will be transferred into the air. Instead we settle for some cabinet vibration but you certainly don’t want cabinet resonances where the cabinet literally sings along. Inexpensive molded speakers typically make for hand good drums. The better the speaker, the better the cabinet design and bracing. Some inexpensive speakers are mounted in a wooden box with no bracing. That might as well be a hand drum. Speaker design is more than porting and cabinetry. A well designed cabinet lets the speaker does all the work.

There is a balance between light weight and good sound.

In the sound reinforcement industry, flown subs is a relatively new trend. In the early days of large scale sound reinforcement even the main PA was typically stacked on the ground, stage or located on scaffolding to raise it up.

What is the benefit of flying subs?

For one when they’re on the ground those closest to them are getting a lot of low end. For instance, those standing right up against the barricade. If you’re 3 feet away from a sub array, it’s going to be really loud in comparison to those who are 30 feet away or 300 feet away. By flying your subs you equalize the distance at which the audience can be in proximity to your subs. Another benefit is that when your subs are flown with your mains you have the opportunity to time align the subs for a larger portion of the audience. When you fly your main speakers and put your subs on the ground, you’ve got a limited amount of time alignment choices and none of them are going to work for the entire audience.

I’ve been looking into the concept that a line array needs to be approximately 12 feet long for directional control down to about 100hz.

As you shorten your array, the directional control of your array starts to climb in frequency.

In order to know what frequency your short line array becomes omnidirectional, you can use this website calculator:

MC2 System Design Group – Wavelength calculator webpage

Here are some lengths of frequencies or “wavelengths”.

If the goal is directional control down to to crossover point where the sub kicks in, obvious 100hz / 12feet / 3.44 meters is pretty much a minimum. What if we can get by with a sub crossover point @ 125hz?

So we still need 9 feet / 2.75 meters of line array length.

What if we want our directional control to extend below 100hz? How about 50hz? We need 22.6 feet of line array.

25hz equals 45 feet! I don’t think I’ve ever seen a 45 foot long line array but I have seen some that are 20+ feet long.

What is the benefit of having directional control down to the sub crossover frequency? I’ll have to do some more research to answer this question fully but I can tell you that your stage sound will benefit from not having a bunch of low end and low mid energy coming off the main PA and loading your stage. The concept of putting sound where the people are means the audience and not the stage. The last thing I want is low end from the PA system bleeding into all my stage mics.

Along these same lines are the use of cardioid subs. I would guess that with a long enough array (at least 12 feet) and cardioid subs, you may have very little PA sound on stage which is an obvious benefit to the band / talking head, etc…

With a sound meter calibrator, you can verify the output level of a measurement mic so that you know that your SPL measurements are accurate. Yesterday I secured an AMP SM-CAL1 with 1/4″ adapter for around $100 on Ebay. Between the 1/2″ and 1/4″ adapters I should be able to maintain all of my measurement mics.

AMPROBE SM-CAL1 sound meter calibrator webpage

AMPROBE SM-CAL1 sound meter calibrator data sheet PDF

AMPROBE SM-CAL1 sound meter calibrator product manual PDF

QUOTE:

“The Amprobe SM-CAL1 is a sound meter calibrator with two output levels of 94 dB and 114 dB. The calibrator generates these fixed sound level signals for calibration of sound level meters. The unit ships with a ½” adaptor installed to accommodate sound meters and microphones with ½ diameter.

Battery: 9V, 006P or IEC 6F22 or NEDA 1604.
Electrical (sound) standard: ANSI S1.40-1984 and IEC942 1988 Class2.
Output sound level: 94dB and 114dB re 20 uPa under reference.
Accuracy: ± 0.5 dB.
Output frequency: 1 kHz ± 4 %
Two output levels of 94dB and 114dB
Output frequency of 1000Hz
Fits ½” microphones
Easy one handed operation
Low battery indicator
CE, conforms to ANSI S1.40 – 1984 and IEC942 – 1988 class 2
Includes ½” adapter, 9V battery and users manual

This is taken from the user manual (ENGLISH)