While working on another post about digital consoles with built in audio measurement instruments, I was reminded that some audio DSP devices offer audio measurement tools already.

For example, the BIAMP Audia device has a transfer function tool for trouble shooting signals withing the device.

Biamp – audiaflex website

Audiaflex manual PDF

Here is what the manual says about the transfer function tool on page 167 of the PDF.

So we’re getting there. Having a tranfer function tool inside a DSP is really helpful. Most of the time on a large DSP system, you have to figure out how to measure your various signals pre and post DSP. If in the long run we can get the internal transfer function tools to work with our external tools, that will make optimizing a large and complicated sound system that much easier. If the console and the DSP can share signals digitally and allow for transfer functions between those signals, even better.

After taking Buford Jones (2) day Mixing Workshop, I was inspired to schedule the first Sound 101 workshop for local stage hands who had previously shown interest. About 15 people turned out today. A diverse group of people (some with no past audio experience) with some very perceptive questions. I was actually amazed that some of the most logical questions were asked immediately as information was presented. We discussed acoustics, phantom speakers, time delaying a PA, stitching speaker sub systems together to act as one sonic image, filtering to make a PA work without too much overlap. I guess really, these aren’t “101” concepts at all. How do you EQ a system? Channel? If the PA isn’t aimed correctly? How do you manage your mix if the subs are 15dB too loud in general?
Understanding the nuts and bolts of acoustics and physics obviously has it’s place but I might of done better to show them how to put a simple PA together and use the mixing board. Maybe system design, tuning, etc… should wait. What do you think?

Fortunately there was enough interest after the class to do a follow up workshop tomorrow and try to cover Equalization.

It’s amazing how fast the clock moves when you’re trying to explain things to a group of people.

Using house gear we setup the following speakers:
(2) Meyer Sound UPJ for mains L/R
(2) Meyer Sound 650R2 for subs L/R
(2) Meyer Sound M1D for front fills L/R
(2) Meyer Sound UM1P for stage monitors

We set up the equivalent of (4) PA systems all tied together to make it possible to control signals from multiple locations and with multiple levels of functionality.

The first PA consisted of a single Mackie SRM150 which has a build in mixer, amp and speaker with a master volume and 3 band fixed EQ. One SM58 and a 1/8″ TRS cables to dual RCA for Iphone use. That system was completely self contained to demonstrate how simple a PA might be and how simple it would be to put together and use.

The second PA consisted of an API-3124m mic preamp / mixer. It has (4) inputs, a stereo output, a stereo return and an aux send.

Channel 1 – Wired SM58 @ Stage
Channel 2 – Wireless Sennheiser handheld mic
Channel 3 – Tascam CD L
Channel 4 – Tascam CD R

The outputs of that console landed on a Midas Venice 240 right behind it.
The same wired SM58 and wireless mic also landed direction on the Midas Venice 240 (via passive splitters) so that one system could be completely independent of the other
The same wired SM58 and wireless mic also landed on the house Soundcraft Vi6 digital console @ the true FOH position.

The PA was processed via a Metric Halo 2882+DSP.

Input 1 – FOH measurement mic
Input 2 – Roaming measurement mic
Input 3 – Midas Venice 240 Main output R
Input 4 – Midas Venice 240 Main R output
Input 5 – Midas Venice 240 Monitor 1 output
Input 6 – Midas Venice 240 Monitor 2 output
Input 7 – Loop (from output 8 – for measurement SOURCE purposes)

Output 1 – UPJ HL main
Output 2 – UPJ HR main
Output 3 – 650 Sub L
Output 4 – 640 Sub R
Output 5 – M1D HL front fill
Output 6 – M1D HR front fill
Output 7 –
Output 8 – Loop (right back into Input 8 – for measurement purposes)

Here is a shot of the setup we used.

We also setup a Gallien Kruger RM800 bass amp / Hartke 410XL bass rig and Fender Twin Reverb as sources which we didn’t get to today. Maybe tomorrow.

A Fender Twin Reverb (Silver Face) is a classic guitar amp. It’s one of the most widely available amps for rent.

I’ve always wondered what sort of interaction is happening between the speakers in a multi speaker combo amp.

INSERT PHOTO OF AMP
INSERT PHOTO OF AMP WITH MIC
INSERT GRAPH WITH MIC AT EDGE, CENTER, MIDDLE, CENTER, EDGE

The combination of a GK RB800 bass amp and Hartke 4×10 XL bass cabinet is one of the most popular combinations of all time. You can rent this rig in any major market. What makes this rig so popular?

INSERT PHOTO OF RIG
INSERT PHOTO OF RIG WITH MIC AND DI
INSERT GRAPH OF DI / MIC CLOSE
INSERT GRAPH OF DI / MIC DISTANT

Link to Prosoundnetwork article

Dayton Audio Test System – website

How Earthworks Measures Mics

Time to Shelf The Low End |Why Your Subs Are Ruining Your Live Mix

I’ve purchased (4) Metric Halo devices brand new. I’ve purchased (6) Metric Halo devices used. Benefits of buying new include having a warranty, knowing your gear is at 100% and if you make your purchase from someone who knows the Metric Halo product line, Mio Console and such, you’re also getting a support system that doesn’t logically come with used gear. There are times when buying used makes sense. There are also times when you can pay too much. I know from experience.

The “future proof” factor of any Metric Halo device means that any device they’ve ever made can be current (if the right hardware is added). Typically this means adding a 2D card which expands the 2882 or ULN2. The ULN8 and LIO 8 are new enough to where there is only one version. Current.

I have one used device that falls into the “paid too much” category. It’s my UNL2. I purchased what I thought was a 2D model which turned out to be a legacy unit. After paying for the cost of the 2D card (even during a Metric Halo sale), I have invested too much invested in this box to sell it without losing money. As long as I keep it, it’s useful. Lesson learned is make sure you know what you’re buying before you buy it.

Let’s say you’re considering a used box on EBAY. If so, you want to make sure you’re getting what you desire to get. I routinely write EBAY sellers to make sure I know that they know the difference between a legacy box and a 2D box. Whether they know the difference between a 2D unit that has onboard DSP and a unit with a +DSP license that is still a legacy model. If you are clear on this and the seller isn’t clear on this, you could be paying way too much for a box that needs to be updated.

Consider the following costs (taken from Metric Halo Store)

2D upgrade card for 2882 = $329
2D upgrade card for UNL2 = $439
+DSP license = $549
Rack ears for 2882 / UNL2 = $37.50
Rack ears for UNL8 = $40
Power supply for 2882 / ULN2 = $65
Power supply for UNL8 = 75

Obviously if you want a current device and you pay too much for a legacy unit, you’re looking at between $329 and $439 in addition to the purchase price of the device itself.

I’ve actually witnessed Metric Halo boxes sell used on EBAY for more than the sale price of a new unit. Amazing. This speaks of how passionate (and maybe short sighted) Metric Halo users are sometimes. What I can tell you is that if you’re the only one bidding and you win without any competition, you might of bid too much for the wrong device. In general Metric Halo users are smart.

There are (2) types of sellers on EBAY listing Metric Halo gear. Those that know and use the product and then those that have no idea what +DSP means or what 2D means. In the case of a Metric Halo user, they will generally list all the facts about the unit because they know what is important to a Metric Halo user. Those selling Metric Halo gear that don’t know what is what are likely to misrepresent what they are selling and even answer questions wrong. Buyer BEWARE!!!

I recommend writing the seller with the following questions:

What is the serial number of the unit (if it’s not shown in a picture). A low serial number might indicate a legacy box that hasn’t been upgraded.
Are you the original owner?
Are you sure the unit is a 2D model (if that is part of the listing)
Are you sure the unit has a +DSP license (the only way to verify this is either if the unit has +DSP on the front panel near the left or if you get a screen shot of the Mio Console window open). Otherwise there is no visual indication. At the same time, any unit could have +DSP added which means the front panel wouldn’t show +DSP. The only real way to know is to ask for the serial number and a screen shot of the Mio Console window.

When in doubt ask the seller for a screenshot of the Mio Console window showing the serial number, firmware, etc…

When you download Mio Console, it includes firmware update files for both 2D and legacy devices. This is what the Mio Console package folder looks like.

This is the file for 2D firmware updates

This is the file for Legacy firmware updates

Remember that a Legacy device can be upgraded to be current but there is a non trivial cost involved and you have to take the unit apart to do the upgrade.

If you make sure you know what the AUCTION is for and how much the unit will cost to upgrade (if any), you’ll be able to bid with confidence that you’re getting a good deal and not the seller.

I recently ran into a fellow sound engineer who works with another local sound engineer who has a medium sized JBL / Cerwin rig he uses on outdoor gigs. The rig is comprised of up to (8) JBL SR4732X 3 way boxes and (8) to (16) Cerwin Vega L36 / B36 subs although this evening rig used (6) and (6).

JBL SRX catalog
Cerwin Vega – L36 sub

Last time I saw the rig in action was years ago and I wasn’t paying that much attention but there were only (2) mid / high boxes per side and (2) subs per side on an outdoor event. That was probably 20 years ago.

In case it’s not already quite clear, as you add more and more boxes together, your chance for phase issues goes up exponentially if they aren’t placed and splayed correctly.

POP QUIZ:

What sounds worse than a (2) box a side PA splayed wrong?

ANSWER:

A (4) box a side PA splayed wrong!

The specs for the JBLSR4732X says that the HF horn is 90 x 50. The spec sheet also reveals that there is a UHF driver that has a 100 x 100 dispersion pattern. How do you mix 90 x 50 with 100 x 100? I’m not sure but even more important, how do you put 2, 3, or even 4 90 x 50 horns next to each other correctly? There should be a splay of approximately 35 to 45 degrees between cabinets and the final splay should be based on measurements. A 90 x 50 degree horn might really be a 92 x 48 or a 88 x 51. The splay should avoid holes in frequency response between cabinets but it should avoid overlap that causes comb filtering. The cabinets (more than 1) should be delay corrected in order to line them up. Where do you put the mic to measure the delay time between (2) cabinets that are splayed correctly? That is a question I will have to answer once I have a bit more experimentation under my belt. I will ask someone who splays similar cabinets for a living and measures their response. In the meantime, here is the way the PA was configured this evening for an outdoor show.

You will note that the (2) raised cabinets are “slightly” splayed but obviously not by 35+ degrees from each other. Then add in the 3rd “front fill” cabinet and you’ve basically got (3) 90 x 50 horns all aiming in the same direction.

Here is what the specs are for the MF driver:

JBL 2447J 1.5″ Titanium Horn Driver 16 Ohm

The Model 2447J is a 1.5″ exit diameter addition to JBL’s family of professional quality compression drivers. The 1.5″ exit allows the Coherent Wave™ phasing plug to directly couple with Optimized Aperture™ Bi-Radial® horns to provide lower distortion and better coverage control to 20 kHz than previous designs. In addition to improving performance, the 1.5″ exit design also reduces size and weight. The Coherent Wave phasing plug structure has four equal length passages to provide in-phase summation of diaphragm output at the 1.5″ exit. This optimized configuration produces coherent acoustical power up to much higher frequencies than more conventional designs. The diaphragm design includes JBL’s exclusive three-dimensional diamond pattern surround tuned to reduce fatigue-inducing stresses in the membrane and support structure. This provides predictable normal resonance modes, and radial reinforcing ribs increase diaphragm stiffness. This diaphragm design combined with the Coherent Wave phasing plug increases the 2447’s output in the 5 kHz to 20 kHz range.

High temperature voice coil former materials and adhesives enable the 2447J to handle high power levels over extended periods of time. The voice coils themselves are identical to previous JBL models, so that impedance and network matching will be the same. After manufacture, the frequency response of each transducer is tested for conformity to JBL’s rigid performance standards. The model 2447J is ruggedly constructed to withstand the rigors of both fixed installations and touring applications. All tolerances are held to the same high levels traditionally associated with JBL designs. The JBL manufacturing process permits the use of rim-centered diaphragms for instant interchangeability and ease of field service.

Specifications: • 100 watts continuous above 500 Hz, 150 watts continuous above 1 kHz • Voice coil diameter: 4″ • Frequency range: 500-20,000 Hz • Sensitivity: 112 dB (1W/1m on 2352 horn) • Nominal impedance: 16 ohms • Net weight: 23.5 lb. • Dimensions: 9-1/4″ dia. x 4″ depth.

Product Specifications

Mounting Type4-Bolt
Exit Diameter1.5″
Diaphragm MaterialTitanium
Impedance16 ohms
Power Handling (RMS)150 Watts

JBL 2447J 1.5″ Titanium Horn Driver 16 Ohm

BrandJBL
Model2447J
Part Number294-438
UPC844632066894
Product CategoryHorn Drivers
Unit of MeasureEA
Weight26.5 lbs.

Here are the specs for the HF driver

Professional Series
Key Features:
Bi-Radial
®
Constant-Coverage
horn design
Constant 100° x 100° dispersion
from 3 kHz to 20 kHz
40 watts continuous program
3 kHz to 21.5 kHz response
Annular-ring diaphragm ferrite
motor structure
44 mm (1
3/4
in) edgewound
aluminum ribbon voice coil
105 dB sensitivity, 1 W, 1 m (3.3 ft)
2404H
Designed for use as an ultra-high
frequency driver in multi-element,
full range loudspeaker systems, the
JBL Model 2404H delivers an un-
matched combination of wide, tightly
controlled dispersion, extended fre-
quency response, high power capac-
ity, and high efficiency
One key to this outstanding perfor-
mance lies in the unique geometry of
the driver
’
s Bi-Radial horn
¹
Devel-
oped with the aid of the latest compu-
ter design and analysis techniques, the
horn provides constant coverage from
its recommended crossover point of
3 kHz to beyond 20 kHz. The Bi-Radial
compound flare configuration main-
tains precise control of the horn
’
s
wide 100° x 100° coverage angle, and
the horn
’
s rapid flare rate dramatically
reduces second harmonic distortion.
The uniform coverage of the horn is
illustrated by the detailed polar data
and the isobar (constant sound pres-
sure) contours included in this speci-
fication sheet. The polar curves of the
2404 exhibit soft-edge pattern charac-
teristics, due to the gradual drop-off of
level with increasing off-axis angle.

Here is an interesting thread about the JBL SR4732X boxes HF characteristics.

Live Sound Int forum – JBL SR4732X crossover quesiton (long)

I finally got around to working with Smaart 7 long enough to get comfortable with the interface. I watched all the tutorial videos as well as read the available literature to get there.

There are (2) things that Smaart 7 can do that in my opinion are worth the price of admission.

Multiple transfer functions (as many as your hardware can handle)
LIVE IR – (Live Impulse Response)

In the case of being able to show multiple TF traces in the same window, this allows for realtime view of a setup like the following:

TF 1
REFERENCE SIGNAL – audio interface output
MEASUREMENT SIGNAL – console output

TF 2
REFERENCE SIGNAL – console output
MEASUREMENT SIGNAL – DSP output

TF 3
REFERENCE SIGNAL – DSP output
MEASUREMENT SIGNAL – Measurement mic

Or for a large setup using multiple mics, maybe something like this:

TF 1
REFERENCE SIGNAL – audio interface output
MEASUREMENT SIGNAL – measurement mic 1 / FOH MAIN

TF 2
REFERENCE SIGNAL – audio interface output
MEASUREMENT SIGNAL – measurement mic 2 / FRONT FILL

TF 3
REFERENCE SIGNAL – audio interface output
MEASUREMENT SIGNAL – measurement mic 3 / UNDER BALC

TF 4
REFERENCE SIGNAL – audio interface output
MEASUREMENT SIGNAL – measurement mic 4 / OVER BALC

TF 5
REFERENCE SIGNAL – audio interface output
MEASUREMENT SIGNAL – measurement mic 5 / DELAY

TF 6
REFERENCE SIGNAL – audio interface output
MEASUREMENT SIGNAL – measurement mic 6 / LOBBY

Here is a screen shot of (3) overlapping measurements in Smaart 7. Green, Blue and Pink. These are just junk measurements but the concept is sound. You can see everything you’re interested in knowing on the same screen at the same time and watch as you make adjustments system wide (if you’ve got enough measurements going). I don’t think the importance of this can be overstated.

What is so cool about the LIVE IR window? Without the LIVE IR window, every time you move the mic or adjust a delay time, you might have to FIND the delay offset time again. With LIVE IR, it constantly updates the impulse response. I’ll give you an example of where this could be really helpful.

Let’s say you want to ring out 12 wedges on stage and your measurement rig is at FOH. You feed pink noise down a return line to the monitor console. Now you can send your REFERENCE SIGNAL to each stage monitor. With LIVE IR, you don’t have to run back and forth between FOH and the stage to FIND DELAY.

On a recent show, I used the venue projectors to project my measurement window onto all the screens in the room. This way I could see my measurement window from the stage on 10’x14′ screen! Live IR would of saved me a lot of time.

I’ve been looking at XLR Y splitter boxes for my measurement rig for some time and when I saw a stereo Avid Audio Splitter listed on EBAY, I ordered one. The cost was around $20 and I couldn’t build one for much less than that (ignoring labor). I didn’t know what to expect inside the box but there is no power connection so the device is obviously passive. Regardless of the circuitry, I could modify the box if necessary. The device has the following connections:

LINE IN (Left / Right)
LINE OUT (Left / Right)
MONITOR OUT (Left / Right)

Of special interest is the massive metal piece inside the box that serves no purpose other than to add weight to the device. Everything else is plastic so it must of been decided to add the weight to keep the box from falling over from the weight of (6) XLR connectors. Counter ballast if you will.

I checked the circuit board and the LINE IN to LINE OUT connections are parallel connections (straight wire). I had hoped that the MONITOR outputs were also parallel connections but instead there are some passive components involved in that signal path.

Here is the trace when I measure the LINE IN to LINE OUT path. Note the perfect impulse response as well as perfect frequency and phase response. What we would expect from a piece of 3 conductor wire with connectors on the ends compared with another piece of 3 conductor wire with connectors on the ends.

What do the resistors and capacitors do in the MONITOR OUT circuitry? Here is a trace of the LINE IN to MONITOR OUT path

Together, the capacitors and resistor networks cause an 18db reduction in gain and some low end phase drift (due to the capacitors). The capacitors also appear to elongate the impulse response. Certainly not what I want to use as a reference point!

Here is a overlay of all (4) snapshots. Left / Right traces match but there is the same -18db reduction in gain.

Since the goal is to have a stereo 1×2 splitter box with parallel connections, I’ve installed some wire jumpers across the LINE OUT and MONITOR OUT XLR pins which in theory will bypass the resistors and capacitors.

Here is a measurement overlay between the original LINE OUT and the MONITOR OUT with the jumpers installed between LINE OUT and MONITOR OUT pins. Identical.

Now I’ll install permanent jumpers between the LINE OUT pins and the MONITOR OUT pins and put this box back together. I’m using buss wire which is solid and relatively non flexible for the connections.

I’m also using clear shrink wrap to avoid any shorts. I shouldn’t need it if I bend the buss wire correctly but I have the shrink wrap and it can’t hurt:)

Here is the device with (6) buss wires installed.

Let’s measure again to verify we have achieved the desired goal.

(4) snapshots that overlay perfectly. I’m going to put the device back together and move on…

Details

The Smaart I-O is a measurement grade 2×2 USB audio interface designed and built specifically for use with Smaart v7 Measurement Software. The I-O features two high-quality, active balanced inputs with 50 dB of computer adjustable gain in precision 1 dB steps. These input gains are monitored directly by Smaart allowing the user to retain accurate SPL calibration while varying measurement signal input levels.

The inputs employ a Neutrik combo jack (XLR / ¼”) to accommodate both mic and line level input signals, each with switchable 48V phantom power on the XLR inputs for measurement mics and a 20 dB pad on the ¼” to accommodate line level signals. The Smaart I-O also features two active balanced XLR outputs capable of providing +8.2 dBu (2 Vrms) max level signal for playback or excitation sources.

Powered by the USB port of your computer (or via the optional 5 VDC jack), the Smaart I-O provides a compact and portable measurement-quality input for your Smaart rig.

A simple control program, included with the product on CD-Rom (or available via download from our website) sets preamp gains and phantom power selection, and automatically integrates with Smaart v.7. The control program also provides firmware update capability and can rename the Smaart I-O at the hardware level – an important feature when using multiple devices with Smaart.

Smaart I-O Features:

•Two (2) Neutrik XLR – ¼“ combo jack inputs:
oComputer controllable 50 dB analog gain in precision 1 dB steps
o Max input (at minimum gain): +6 dBu (XLR), +26 dBu (¼ ”)
o Active balanced with 2k ohm impedance (XLR), 65k ohm (¼ ”)

• Two (2) Active Balanced XLR Outputs
o 150 Ohm impedance o Max Output : +8.2 dBu (2.0 V rms) w/ 100k ohm load

• Frequency Response o Magnitude 16 Hz – 20 kHz, +/- .25 dB
o Phase 16 Hz – 20 kHz, +/- 10 deg

• Sample rate clocks of multiple I-O’s can be linked to allow for device aggregation in Smaart’s dual-channel measurements.

• Bus powering via USB for convenient, single-cable connection to the computer.

• Optional 5 VDC jack in cases of insufficient USB power

• Dimensions 1.75” (h) x 7.25” (w) x 4.25” (d)

© 2014 Rational Acoustics, LLC. All Rights Reserved.

Introduction to the Six Basic Audio Measurements – Part 1-1
Introduction to the Six Basic Audio Measurements – Part 1-2
Introduction to the Six Basic Audio Measurements – Part 1-3

Introduction to the Six Basic Audio Measurements – Part 2-1
Introduction to the Six Basic Audio Measurements – Part 2-2

The PA system at Fort Worth Academy of Fine Arts has gone through many different stages since the venue was taken over. Originally the space was a church sanctuary with a small stage and more seats. When the church built a new building and sold the old one to the school, the PA remained AS IS. This was a bad deal because the school extended the stage which put the overhead PA behind the stage:(

At one point the over head speakers (LCR center cluster) were moved to the front of the new stage but they were still high above the audience so the imaging was 90 degrees overhead for the front row of seats. In 2011, the school purchased some QSC KW122 self powered speakers for mains and monitors and they were hung just high enough to stay clear of the FOH lighting positions. About 1/2 as high as the original center cluster.

Much better imaging, a true stereo PA for the mains & side fills, delays, subs processed by a Media Matrix Xframe 88 and (2) 8802 Break out boxes.

The house PA consists of the following:

House Left
House Right
House Left sidefill
House Right sidefill
Subs (2) under the stage
Delays (2) about 2/3 back in the house and in line with the House L/R speakers

Each of these zones is fed from a separate output on Xframe 88. The subs are fed from an Aux send via a 3rd Xframe 88 input.

coming soon…

coming soon…