While not directed related to audio measurements, a Pin 1 Problem is something worth mentioning every once in a while since it still plagues the professional audio industry and many still don’t know about it but suffer the consequences.
More than 15 years ago, I was introduced to the concept of a “Pin 1 Problem” on a user list by Jim Brown. At the time I realized that Pin 1 Problems were a serious issue and while some manufactures addressed it immediately once the solution was explained, others ignored the matter and have kept on designing gear with a pin 1 problem. Consequently, I create this website as a place to gather and share information about pin 1 problems:
If you want to understand what a Pin 1 Problem is and how it can affect your audio, read this recent article by Jim Brown:
prosoundweb.com – Pin 1 Revisted by Jim Brown page 1
prosoundweb.com – Pin 1 Revisted by Jim Brown page 2
prosoundweb.com – Pin 1 Revisted by Jim Brown page 3
This is the original document that Jim Brown is referring to:
rane.com – Pin 1 Revisted PDF
A “Hummer” is a device meant to help find Pin 1 Problems. It’s parts, construction and use are explained in this PDF:
Build a “Hummer” to help find “Pin 1” ground problems PDF
I intend to test all of my gear for Pin 1 Problems soon and share my discoveries but before I go to the trouble of building a Hummer, I thought I would verify that it is still the appropriate device to test modern equipment for a pin 1 problem. I’ll report back once I’ve heard back from my friend and expert on the subject, Bill Whitlock of Jensen Transformers.
While RF (radio frequencies) are outside the realm of the audio spectrum we are dealing with for audio measurement purposes (roughly 20Hz to 20kHz), audio frequencies are low frequency cousins to the RF frequencies (mHz to gHz) used to transport audio from point A to B without wires. Just as we can use measurement mics, audio interfaces and software to analyze our sound systems, we can measure an RF environment with RF scanning hardware and analyze the incoming data with RF coordination software in order to optimize an RF system and choose the best frequencies available in a given location for our needs.
Like it or not, as audio professionals, RF is and will continue to be a huge part of live audio production work, whether it be wireless vocal mics, wireless instruments or in ear monitors (IEMs). There really is no excuse for being RF illiterate. Especially if we intend to rely on RF gear for measurement purposes.
Some good news! Understanding RF measurement, coordination & optimization is a simple pursuit when compared to sound system design and optimization. RF scanning (measurement) hardware and RF coordination software is readily available, inexpensive and fairly straight forward to use. I intend to provide the necessary details for anyone who is interested to be able to assemble an RF “measurement rig” and learn how to use it.
On a personal note, the more RF based audio measurement work I do, the more I realize how these two “art forms” (audio measurement and RF measurement) run parallel to each other and compliment each other.
I will begin to store what I consider essential RF related information here:
This article is worth reading:
eaw.com – driving long cables PDF
This article by Merlijn on delay speakers is a great explanation for when and why we need them.
prosoundweb.com – To Delay or Not To Delay – page 1
prosoundweb.com – To Delay or Not To Delay – page 2
There are only a few wireless measurement platforms on the market that can be used right out of the box. They all rely on a digitized signal to avoid a compander circuit (rendering most RF systems useless for measurement purposes). Even with a digital RF system useful for wireless measurement purposes, each system I am aware of exhibits various frequency and phase related artifacts. Beyerdynamic recently began to offer a measurement capsule compatible with their existing TG1000 digital wireless rig. Below is a review of the system by Fedele de Marco including measurement results.
Over the last few weeks I have been getting acquainted with my new Lectrosonics measurement gear which consists of:
(1) Venue 2 – (6) channel wide band receiver
(1) R400a – (1) channel narrow band receiver
(2) HMa – (1) channel wide band plugon transmitter
I purchased the (6) channel Venue 2 receiver (A1 band) in order to build a multi channel measurement rig. I purchased the single channel R400a receiver (block 19) so that I can travel with a single channel of measurement kit when necessary. The HMa plugon transmitters will work with either the Venue 2 or R400a receiver since the A1 (wideband) transmitter covers block 19 (narrow band). Now I just need more HMa transmitters 🙂
A few notes regarding using Lectrosonics digital hybrid gear for measurement purposes.The system has an inherent latency of approximately 3 ms compared with a cable. Just something to be aware of.
The Lectrosonics digital hybrid receivers have a high frequency noise reduction function that is set to NORMAL at the factory. It should be set to OFF for measurement purposes or your measurement results will be tainted at frequencies above 2k. The following image illustrates how the high frequency response of the system is affected by the 3 different noise reductions settings when measuring “pink noise”. Note that the NR setting should be restored to either NORMAL or FULL (depending on the circumstances) for non measurement purposes. The procedure for making these adjustments can be reviewed in the user manuals:
Venue 2 receiver user manual PDF – page 11
R400a receiver user manual PDF – page 12 / 13
The next image shows that with the noise reduction setting turned off, the response of the RF system (ignoring slight mic position differences) matches the response of the same mic with an XLR cable.
The HM & HMa plugon transmitters have an adjustable high pass filter labeled as “LF” that should be set to 35Hz (the lowest setting available). Other setting include 50Hz, 70Hz & 100Hz which would be useful when using the plugon transmitter with a handheld cardioid mic.
HMa transmitter user manual PDF – page 10
I have been involved with various measurement projects that took advantage of other people’s Lectrosonics RF gear and while I was very impressed, I wasn’t able to afford the gear until now. Having used the single channel HMa TX / R400a RX kit on a recent travel gig, I can undoubtedly state that having the ability to measure without dragging mic cables around the venue and risking mic damage due to a snagged cable is a game changer. The cost of the gear itself, the cost of batteries and having to manage batteries during each measurement process are considerations. On the flip side of the coin, the amount of possible time savings is no trivial matter. T=M (time = money) & there is never enough time to measure. Wireless measurement gear is an investment in your end result. I also purchased (2) Lectrosonics handheld transmitters and (2) Lectrosonics beltpack transmitters so that once I’d done with the measurement process, I can use the same receivers with different transmitters for production work.
One of the nice things about the HMa plugon transmitters is that they can be used as a handheld transmitter once you swap out the omni measurement mic for a cardioid dynamic or condenser mic. Certainly won’t roll off the table!
My first attempt at multi language support for this website was based on a wordpress plugin that requires a fee after the trial period. It is not out of the question to pay for a translation service but if a fee can be avoided, why not. So I read some reviews of various paid and free wordpress plugins and decided to try one that simply relies on Google Translate to do the translation work.
Unlike the previous plugin which created a new website for each chosen language, the google translate plugin just translates the whole website AS IS in place. There may be a downside to this approach but at this point I can’t think of one so.
Here it is…
audiomeasurements.com – with multi language support
I’m curious to see how it works for visitors in other languages. Give it a try and let me know what you think!
To select different languages, there should be an orange “TRANSLATE” tab at the bottom right of the website. It looks like this:
If you want to add multi language support to your wordpress website via the Google Translate plugin, this link explains how.
wpbeginner.com – how to add google translate in wordpress
For anyone interested in getting the ebook version of Bob McCarthy’s latest edition of Sound Systems – Design & Optimization, the publisher has it on sale right now for around $55 USD. A 30% discount!!! Not sure how long the sale will last so if you’ve been wanting the eBook version, now is the time. I own the paperback and the kindle version. Glad to have both of them for different reasons. Follow this link: Sound Systems – Design and Optimization 3rd Edition
If the eBook is no longer on sale try using “SSD20” without the quotes as a discount code for a 20% discount at check out
Dave Rat recently revealed a new 30″ subwoofer design based around the Powersoft M-Force moving magnet linear motor driver.The M-Force driver is fundamentally different than previous driver technology. Should be a game changer in the high SPL concert sound arena.
ratsound.com – dave’s post 010817
ratsound.com – dave’s post 010617
powersoft-audio.com – M Force datasheet PDF
powersoft-audio.com – M Force User Guide PDF
AES 9060 PDF – A novel moving magnet linear motor
AES 91646 PDF – Subwoofer design with moving magnet linear motor
Interested in owning some SDS30 yourself?
ratsound.com – SDS30 information form
During a recent exchange between myself and Fedel de Marco about our measurement related websites, I realized that there might be way via a wordpress plugin to have translations of our sites created automatically. I’ve been copying text from his website and pasting it into google translate manually to be able to read his Italian blog paragraph by paragraph.
Fedele De Marco’s “Il lato oscuro della fase / the dark side of the stage” website
Surely there must be a better way…
weglot.com – Make your website multilingual
Via the trial version of the weglot translation plugin (limited to one language & 2000 words), I was able to create a mirror website in Italian in a matter of a few minutes with very little effort on my part.
To access the Italian version of the site you can either select it at the bottom of any page of the English version or you can access it directly with this link. Note that due to the 2000 word limit of the trial version of the plugin, very little of the website is being translated into Italian. From what I can tell, the home page is. Not sure what else.
audiomeasurements.com – Italian version
If you are interested in accessing the AM website in a different language, let me know which one. I assume I can shut off the Italian version and add support for a different language without cost to test how well it works for someone. There is a cost involved if I decide to support multiple languages so I’m going to take it slow in implementing other languages until I understand how it works and how much interest there is and how many languages I would need to support. It is not out of the question for me to fund multi language support myself. Here are the various costs involved. I may already have enough words on the AM website to be forced to move to a professional weglot account. Here is the various levels with pricing.
Here is a list of languages that the weglot plugin supports:
weglot.com – supported languages
For anyone interested in building a measurement rig and getting their feet wet, Rational Acoustic’s Smaart DI has been a good place to start. Picture Smaart for teenagers. Purposely simplified to make the process quick and easy. Rational Acoustics has just announced that Smaart Di2 will be released on January 24th, 2017.
This is what the website has to say about Smaart Di2.
“The latest addition to the dual-channel Smaart “Di” platform – Smaart Di v2 – will be released on January 24th, 2017. Smaart Di v2 takes the simplified, streamlined two-channel interface from Smaart v7 Di and modernizes it with many of the features and enhancements developed for Rational Acoustics’ flagship multi-channel Smaart v8 program.”
read more here:
rationalacoustic.com – Smaart Di2 coming soon
Pricing and upgrades:
rationalacoustic.com – Smaart Di2 pricing
Once Smaart Di2 is released, I would recommend anyone that doesn’t already have a measurement app, download the appropriate demo here and take it for a drive. If you have questions on how to set it up or take measurements, contact me.
rationalacoustic.com – Smaart demos
I’m on the hunt for a driver that can produce full range sound so that I can build a reference speaker for my bench measurement setup. Why a single driver? No crossover, no inherent phase issues, no time alignment discrepancies, etc…For this purpose, I am considering the Aurasound NS3 193 8a:
partsexpress.com – AuraSound NS3-193-8A
AuraSound NS3-193-8A Specifications PDF
Below is a link to a white paper PDF that explains the technology used to manufacture AuraSound speakers. Well worth reading…
AuraSound NRT Whitepaper PDF
Meet Fedele de Marco. Below are links to his facebook page, facebook videos, Youtube videos and his website where he shares his perspective on FFT Analyzers, Design, Testing and Calibration of sound reinforcement systems.
Website – “Il lato oscuro della fase / the dark side of the phase”
I recently configured this website so that posts I add to here also appear on Facebook. I hadn’t thought much about the implications of doing this but one of them is that where as before I would make an initial post and then refine it / edit / add content / etc…, now whatever I initially post is somewhat written in stone on Facebook. Consequently I have stopped posting out of fear for not wanting to post before the information is ready. For example, I have some posts that are in the works but not public yet. Once I am through with my work load for the holiday season, I will catch up and figure out how to have Facebook involved without it hindering the creative process. Thank you for your patience.
My introduction to combining speakers properly came from reading Bob McCarthy’s book “Sound Systems – Design & Optimization” but it was during his SIM3 class when he demonstrated the approach that I clearly understood.
On paper the concept seems rather simple.
“-6dB + -6dB = 0dB”
Combine speakers at their -6dB point and they sum back to 0dB.
A key piece of information is that manufacturers state speaker coverage based on their -6db point. If a speaker has a stated coverage of 90 horizontal x 60 vertical, that means that traveling from 0 degrees on axis (ONAX) in an arc to being more and more off axis (OFFAX), when you get to 45 degrees in either direction horizontally or 30 degrees vertically, the mid / high frequencies will have dropped by 6db. If you need more horizontal or vertical coverage than that, you will need another speaker. The not so obvious question to ask is “at what frequency am I trying to combine them?” Speakers are not inherently linear in directionality. In general, speakers are omni directional at lower frequencies and more and more directional at higher frequencies so you have to consider which frequency you’re paying attention to. In order to clear up my own doubts, I recently wrote Bob to get his take on choosing the right frequency to combine speakers and this was his reply:
“6 db – the flat part of the beam width – typically 1k above but with small boxes could be higher. E.g UPA above 1k, UPJ 2k, junior 3k , UPM 4K”
I’ll add that speaker directionality is a moving target. Here is an example of what I am talking about. A directional plot for a QSC AD-S12.
Pay attention to the red part which represents odB and where the yellow starts which represents -6dB. Between 2k and 15k, this speaker is rather consistent in directionality but it is by no means linear. At the two extremes, the cabinet is omni directional below 100 (not shown) and coverage begins to narrows above 8k. Where is -6db. I would take Bob’s suggestion and choose around 2k for this cabinet.
Let’s combine some generic speakers and see what happens. One of the mistakes we can make when assembling a sound system is to put two or more full range speakers side by side in a horizontal array and not to splay them at all. Like this (0 degrees):
The result of a 0 degree splay is broad band comb filtering, something to methodically avoid. You’d be better off unplugging one of them.
(INSERT TRACES TO ILLUSTRATE)
If you don’t already know what comb filtering is, please read this article:
soundonsound.com – What Exactly is Comb Filtering?
The opposite scenario would be to aim the speakers in opposite directions (180 degree splay) which would be silly. Like this:
Believer it or not, 180 degree splay between two cabinets would sound better than a 0 degree splay. In this case you would have no high frequency combing and extra in the low end. If it were me, I’d still unplug the one facing the wrong way.
Options beside 0 and 180? Some might logically assume that a trapezoidal cabinet’s sides indicate the proper splay angle and simply butt them together like this:
In some special cases the sides of a speaker cabinet do indicate the correct splay angle but in general the side angles of a trapezoidal box do not indicate the proper splay angle so it’s not an assumption we want to make.
As a starting point for discussing proper splay, I chose the Meyer UPA as a point of reference because it was the first trapezoidal cabinet.
Here is an article that explains what made the UPA special:
mixonline.com – 1980 Meyer Sound Labs UPA 1 Arrayable Trapezoidal Speaker
“From the outside, the UPA-1 was decidedly different: It was the first trapezoidal speaker (U.S. patent D271,967), now a common practice within the industry. Its sloped sides allowed the creation of tightly packed, wide-coverage horizontal arrays to minimize the comb-filtering effects that occur when spaced drivers reproduce the same frequencies.”
How does one splay Meyer UPA speakers for the right amount of overlap? Meyer Sound has done all the research for us and published the results. The proper splay angle for combining UPA-1P cabinets is 70 degrees. Like this:
On the other hand, the proper splay angle for combining two UPA-2P cabinets is 30 degrees which just happens to be the angle of the cabinet side. Like this:
The UPA-1P & UPA-2P share the same cabinet dimensions but use a different horn, have different coverage patterns and require different splay angles for proper combining.
The UPA-1P has a coverage patter of 100 degrees horizontal x 50 degrees vertical (-6dB).
meyersound.com – 2 UPA-1P spec sheet PDF
The UPA-2P has a coverage pattern of 50 degree horizontal x 50 degree vertical (-6dB).
meyersound.com – 2 UPA-2P spec sheet PDF
Because the sides of a UPA-2P indicate the proper splay angle, you can place them with their sides touching in what Meyer calls a “tight pack”.
meyersound.com – 2 UPA-2P tight pack
meyersound.com – 3 UPA-2P tight pack
With a UPA-1P, there has to be a gap at the front of the cabinets.
This PDF article explains how to properly combine a pair of UPA-1P.
meyersound.com – 2 UPA-1P 70 degree splay
Seems simple enough but in the following PDF, Meyer states that the same cabinets can be splayed 85 degrees apart. What is the catch? You sacrifice a linear frequency response for more coverage. This suggests that while there is a correct splay angle for proper combining, there is some wiggle room.
In the owners instructions for a UPA-1P, Meyer Sound has the following to say regarding designing arrays using UPA-1P cabinets :
meyersound.com – UPA-1P operating instructions
Creating an effective array with the UPA-P requires a precise understanding of how to combine the coverage
area and SPL of the individual speaker with those of adjacent speakers. Array design is a trade-off between
increasing on-axis power and creating smooth transitions between the coverage areas of adjacent speakers.
As the splay angle (the angle between adjacent cabinet faces) decreases below the coverage angle of the indi-
vidual speaker, the power at the center of the array increases, but the coverage overlap between adjacent speakers
causes comb filtering and other frequency response variations. As the splay angle increases toward the coverage angle, the power at the center of the array decreases, but the variations in frequency response diminish. As the splay angle increases beyond the coverage angle, noticeable gaps begin to form in the array’s coverage area.”
Before we move on, it’s important to recognize that some cabinets have a rotatable horn. In those cases, obviously we need to verify the orientation of each horns in order to properly combine them and if necessary we may need to rotate them for our purposes (more vertical coverage or more horizontal coverage). A Meyer UPJunior has a rotatable 80 x 50 horn.
meyersound.com – UPJunior spec sheet PDF
meyersound.com – UPJunior operating instructions PDF
This is what Meyer has to say about splaying UPJunior’s properly for horizontal and vertical arrays:
As you can see by now, Meyer Sound goes out of their way to make sure their users have the information they need to make good decisions. This is not true of most speaker manufacturers. In fact I can think of only one other company that might provide those details as part of their product literature. What this means is that for most speakers, you will have to figure it out for yourself. The first thing to do is to find the spec sheet for the speakers you will be combining so that you know what the stated horizontal and vertical coverage pattern is. From there you can either make your decision based on rules of thumb or use your measurement rig to find the proper splay angles.
For measurement demonstration, I have access to some QSC KW122 cabinets to illustrate the concept of splay & how proper splay affects your end result (seen via the measurement results). Note that unlike the Meyer UPA, the KW122 cabinet is not a trapezoid so the there is no method for achieving a “tight pack”. This means that the horns of KW122 array can only be so close together which limits your ability to combine them optimally:
qsc.com – KW series spec sheet PDF
A KW122 has a 75×75 degree horn. What QSC called Asymetric 75 degree. If you want to combine two KW122, a good starting point is 1/2 of the stated coverage pattern. In this case each KW122 is worth 75 degrees of coverage and combing two cabinets side by side would suggest a splay of 37.5 degrees apart.
INSERT 0 degree splay trace
INSERT 37.5 degree splay trace
INSERT 75 degree splay trace
What if the cabinets you have to work with don’t have a make and model shown (possibly home made) and your time is running out? This is where having a measurement rig is helpful. You would establish a base level and frequency response on axis (ONAX) of one of the speakers and then move the mic in an arc or better yet, leave the mic in place and rotate the speaker until you find the -6db point @ 2k. This experiment would indicate an approximate splay angle. If the -6db point was 25 degrees off axis @ 2k, you would need roughly 50 degrees of splay between the center of two of those speakers to combine them properly.
To illustrate the point that there is no “right” way to splay speakers but instead many opinions and many variables, you may enjoy reading the following thread on the prosoundweb.com forum
Note that some of the posts on that thread come from widely respected individuals in the industry. “It depends” could be the stated consensus. Distilling what I read, let’s go with this.
1. Ask the manufacturer for guidance. If anyone has a vested interest in making sure their speakers are used together properly it will be the manufacturer.
2. Use the stated coverage pattern of the cabinets as a starting point and adjust as necessary.
If the manufacturer states how to properly combine their speakers, that is where I would start. I would verify that their stated method approximates the process previously described. If the manufacture doesn’t provide any information regarding combining their speakers (typical), I would ask them for guidance. If they can offer none, do the measurements and figure it out for yourself. Then share your findings publicly.
Ignorance is NOT bliss and with a measurement rig, you can stop being ignorant and stop relying on the opinions of others. Once you have learned for yourself the proper splay angle for a given speaker cabinet combination, you will be on your way to better sound and the confidence it takes to make good decisions about combining speakers in general.
I just stumbled across SpectraPLUS by Pioneer Hill Software. The PC based application seems to be in use by various industries.
Note that there are (3) versions of the software and each one is available as a 30 day unlimited functional demo.
spectraplus.com – downloads
The websites technical articles reveals some of what the software is being used for,
spectraplus.com – technical articles
Audio equipment testing
Frequency Response tests
Room Acoustic measurements
Noise measurements and monitoring
Precision signal generation
Sound Power measurements
Musical instrument manufacturing and testing
Underwater acoustics, pile driving noise levels
Rotating machinery analysis
Real-time spectrum analysis of live input
Record, Playback and Post Process WAV files
Displays: Time Series, Spectrum, Spectrogram, 3-D Surface, Phase
Full Featured Dual Channel Signal Generator
High Resolution FFT Analysis up to 1,048,576 pts
Octave Analysis from 1/1 to 1/96 octave
Up to 24 bit precisionShenand
Digital Filtering, Distortion Analysis, Transfer Functions
Acoustic Tools: RT60, Equivalent Noise Level (Leq)
THD+N versus Frequency
Acceleration, Velocity and Displacement
Advanced Programming API
This past week I was able to install a miniDSP 2×4 balanced dsp unit in a rack that powers a 2 speaker rehearsal sound system for the ballet company I work with. The device worked out perfectly. Right price, right size.
In order to use the device you have to choose and purchase a plugin.
minidsp.com – 2 x 4 plugin options
In my case, I needed basic routing and EQ so I went with 2×4 Advanced option:
minidsp.com – 2 x 4 Advanced plugin
While the 2×4 Advanced plugin did what I needed to do, the user interface could use some refinement.
Here is a Smaart 8 trace showing the response of the left speaker before and after processing with the minidsp device. I eqed each zone for a relatively flat response via the input EQ and then after combining them made a small low mid cut to both. I’ve used 1/3 octave smoothing on the first trace to make it clear what is what. Brown is pre eq, green is post eq. Note the coherence issue (red) between 1k and 2k.
This is the same trace but with 1/48 octave smoothing instead of 1/3:
This following image indicates the EQ settings used in the miniDSP 2×4 advanced plugin app. Note that there are 5 parametric eq bands available on the inputs and I used 3 filters:
Once I measured the system with both speakers eq-ed as show above, I added one more filter in the output section of the miniDSP plugin to balance out the LF.
After I configured the single speaker EQ, I measured at different places on axis of that speaker. The following trace indicates those results. First 1/3 octave smoothing and then 1/48:
I may replace the power amp in the rack with an amp that has built in DSP soon and if so, I won’t need the miniDSP 2×4 balanced device for that purpose any more and can use it for something else like tuning my studio monitors. Neat device!
A local venue that I support has recently suffered some serious RF (radio frequency) interference for the first time. Not surprising since here in the DFW area, there is less and less RF spectrum for wireless mic purposes. I contacted RF Venue and explained the RF circumstances. RF Venue staff recommended using (2) RF Venue RF Spotlight antennas with their Distro 4 antenna distribution device as a starting point to avoid further RF interference at the venue. This is what an RF Spotlight looks like:
The next link below will take you to a webpage where you can watch a demonstration comparing a RF Venue CP Beam directional antenna with an RF Spotlight. Very impressive.
My initial tests are very promising. The following images were created using a RF Explorer hand held RF scanner with RF Venue’s VANTAGE scanning software for Mac. The first image is of the immediate RF environment using a whip antenna (like those provided with most wireless mics and in ear monitors). The second image is the substituted RF Spotlight antenna. Notice the drop in almost all levels and a significant drop in general noise floor.
Tomorrow I will do a scan at the venue with their wireless mics on as a reference point and see what sort of reduction there is in hostile RF conditions. More soon…
I have been looking for a small dsp unit to use on some projects where space is a consideration and I finally decided on minidsp’s balanced 2×4 device.
minidsp.com – balanced 2×4
I recently got to play with the non balanced version of the same device (RCA inputs and outputs) but wanted the balanced version to interface with my other balanced equipment. Out of the box, the unit doesn’t actually do anything. You have to purchase a “plugin” to give it a purpose. In my case, I selected minidsp’s most straight forward plugin, “2×4 Advanced” for $10. There are more elaborate plugins that can perform 4 way crossovers and such but I just needed some basic EQ out of the box. For a complete list of available plugins, this link explains them all.
minidsp.com – 2×4 plug-ins
Once you purchase a plugin and install it, the plugin becomes your interface for the device. Once you’ve connected to the device, you can change parameters in almost realtime via the USB connection. Once you’ve finished making your adjustments, you can store the configuration to the device and unplug it from the host computer. Then the device will stand on it’s own. This coming week I will be installing the 2×4 on a small portable sound system I use on my ballet gig to EQ the speakers. Here is a video that explains the minidsp concept:
Today I visited the Montgomery Arts Theater at Booker T Washington High School in Dallas Texas to do a check up on their sound system before their upcoming production of “Oklahoma” begins the tech process.
A few things that slowed the process down. First, the dbx 4800 dsp is located in the amp rack on the 3rd floor of the theater and without a way of controlling it remotely, a lot of time was spent climbing stairs and ladders. Next time I would be extend the dbx 4800 network cable to FOH measurement position. Secondly, during the measurement process, it was discovered that the Main L / R speakers were out of polarity. How do you troubleshoot a connection that runs all over the building? Easy! You use Soundtools XLR Sniffer / Sender tester. What is known in the industry as a “Rat Sniffer”. Unlike a normal cable tester box, the two parts of a Rat Sniffer / Sender kit can be used at any location all over the building. This is one of the 2 most important audio tools I use on a daily basis. Get one or more if you don’t have one already. If you have more than one kit, you can troubleshoot multiple lines at the same time.
Using a RAT SNIFFER we narrowed it down to a bad XLR cable between the catwalk patch point and the self powered speakers. Resolving that issue made the system much more usable:)
measurement results to follow:
minidsp.com – rePhase FIR tool
Quote: “rePhase is a Windows-based freeware program written by Thomas (aka “pos”), a long time miniDSP community member. rePhase generates finite impulse response (FIR) filters that “reverse” the phase shifts introduced by a loudspeaker crossover. rePhase can also generate linear-phase crossovers. With the aid of a real-time FIR filtering engine or “convolver” such as miniDSP’s OpenDRC or miniSHARC, the result is a linear-phase loudspeaker system.”
This SIM3 machine must have an interesting story…
“The powerful software enables you
to measure your audio system
to display, interpret and process measurement data
to carry out multiple mathematical calculations
to generate crossovers and other FIR filters
to identify reverberation times
to establish correction filters for speaker drivers and the listening room
to identify harmonic distortions
to filter WAV tracks
to play music tracks for test purposes”
prosoundweb.com – FIR-ward Thinking: Examining Finite Impulse Response Filtering In Sound Reinforcement Systems page 1
prosoundweb.com – FIR-ward Thinking: Examining Finite Impulse Response Filtering In Sound Reinforcement Systems page 2
prosoundweb.com – FIR-ward Thinking: Examining Finite Impulse Response Filtering In Sound Reinforcement Systems page 3
At the link below, there are videos, mp3s and PDFs of some presentations by Bob McCarthy, Jamie Anderson, Dave Gunness, Merlijn van Veen, etc…
The application the video below explains can be secured here:
A friend of mine has (16) Beyer MM1 measurement mics with calibration files but they are not formatted to load directly into Smaart without be manipulated.
As soon as I figure out how to get the file formatted to load into Smaart, I will post the details.
I just learned about Audyssey and it’s various technologies. One of them being MultEQ. I cannot speak of whether the concept / system works or not but it’s interesting to consider.
“At the core of everything Audyssey creates is our first technology, MultEQ™. Based on 6 years and $6 million of research at the USC Immersive Audio Lab, MultEQ delivers accurate, enveloping, and distortion-free sound in any listening environment. It has won numerous awards and is the basis of several other Audyssey technologies. MultEQ has been established as the standard room correction technology for today’s leading audio products.”
This is an interesting find.
The Audyssey Tuning System is the newest and most advanced way for manufacturers to optimize sound. Exciting stuff, if you’re the type of nerd who gets excited by ground-breaking audio engineering.
The ATS technology bundle is designed to be implemented at the factory level, so automobile and electronics companies can tune their products directly off the line to provide their consumers with optimized and consistent sound. We’re bringing innovation to industry by making it a fully integrated part of any industry’s manufacturing process.
ATS begins with MultEQ, Audyssey’s revolutionary space equalization software. Except a traditional MultEQ uses a single microphone. MultEQ on ATS uses 8 microphones and records chirps simultaneously. So if you’re a geek, you already love it. ATS is processing audio information in multiple locations, from multiple directions, not just averaging for a sweet spot, but precisely mimicking the actual listening experience. That’s a lot of information. So a job that used to take literally days, now can be done in about 45 minutes. And by the time audio engineers take and record all the acoustic measurements they want in all the places they need, they’ve not only lost time and added work, but they’ve also ended up with a number that’s significantly less accurate. Our technology is making manufacturers’ jobs easier and smarter. We call that applied intelligence.
And with ATS the whole thing is done in real time. Pretty cool. See Audyssey Tuning System lets audio engineers experiment and hear the changes as they make them. Applying filters to one output affects every other and a manual adjustment is not only tedious, but with the 64 channels that ATS is built to handle, it’s downright impossible. Every time a filter is applied to one part of the audio chain, ATS technology reads the cluster and rebalances all the other affected aspects. It’s like ten thousand techies, nerding-out at precisely same time.
Now add in technologies like Audyssey’s Volume Extension and Dynamic Volume, and the speaker performance kicks in to a warp-drive.
Audyssey Tuning System is the result of a list of technologies combined with years of research into psychoacoustics. While manufactures need to balance their sound perfectly, at Audyssey, we have a passion for making sure that perfectly is how it’s heard. From how the audio is produced to the way it’s processed, Audyssey is the journey to great sound.
Watch this Meyer Sound webinar:
meyersound.com – Interpreting Delay Finder (Impulse Response) Data
And then watch Merlijns video:
merlijnvanveen.com – Why The Impulse Response Won’t Work For Subwoofers
I have known since the beginning that the final frontier of audio measurements and system optimization is a firm grasp of how phase works and how to manipulate it properly. It’s also one of the hardest parts of the pursuit for me. Maybe I am special:) we cannot ignore our weaknesses and continue to make progress so the rest of 2016 I will study phase harder than ever before. Below are the top results when I do a google search for “understanding phase”. I’ve included one entry about phase diagrams that in theory will have nothing to do with audio phase but maybe it will. No reason to limit the exploration until we know that “phase” isn’t a universal concept whether it be related to the moon, gases or audio.
Here are links to the top results I get in Google when I type in “Understanding Phase” on 081616.
youtube – Understanding Phase – Part 1 by Bob McCarthy
youtube – Understanding Phase – Part 2 by Bob McCarthy
soundonsound.com – Phase Demystified by Mike Senior
uaudio.com – Understanding Audio Phase and Correcting Issues by Daniel Keller
mixonline.com – Understanding Phase Part 1 : Making Sound Visible by Bob McCarthy
musicthinktank.com – Understanding Phase by Barry Gardner
khanacademy.com – Phase Diagrams by Sal Khan
ask.audio – Audio & Music Production Concepts: Understanding Phase by Rishabh Rajan
I recently sent Howard Page some questions and his response included a reference to reading an article he wrote for Audio Technology magazine.
The article is entitled “The Howard Page Method – Tuning & Optimizing Large-Scale Concert Sound Systems” and starts on page 21 of the document available at the link below. Enjoy!
This is an online article by Barry Jackson from 2010. Interesting read: