On behalf of Allen & Heath, we are pleased to announce new V1.6 firmware for QU, and the new QU-YOU Personal Mixing app for all QU series!
The new V1.6 Firmware includes;
- Expanded channel & effects Libraries with 56 factor points for EQ & compression
- New Status Logging for the QU Drive recorder
- Latching Talkback Mic Option
- New Option for PAFL to follow Mix
Up to 7 of the QU-Apps can run and one instance of QU-PAD, as well as unlimited ME-1 making this the most powerful all-in-one ecosystem available on the market!
The App is available on the Apple store for FREE!
Improving Intelligibility In A Challenging Acoustical Environment: Part I - An Introduction To a Process
If you have ever had the task of improving sound system speech intelligibility in a large, acoustically challenging space, you might find the blog entries to follow interesting. This project was brought to our attention by a Systems Integrator (SI) who was awarded a contract to install a new audio system. Along with the new audio system, there is a budget for acoustics, for which an opinion of probable cost was provided by the SI. The SI provided me with information about the existing loudspeaker system, which consists of a group of three (exploded array?) low Q, smaller, curved, multi-coned driver boxes, which would typically provide for very low intelligibility in speech reinforcement. The loudspeakers are a famous brand product, and this type of system is well accepted at the proposal stage of many projects, even where the space is acoustically challenging.
How bad is it? What are the potential solutions? How do we quantify those "solutions?" This blog is going to answer the three questions in detail, and attempt to provide you with an outline of an approach that removes the guesswork and you risking your good reputation.
Our friendly SI has a good deal of experience in these types of projects, and they did do a loudspeaker demonstration at a facility function for this client, as both a service and a proof of concept. The demonstration was successful in showing that speech intelligibility would be substantially improved by using a digital steerable column loudspeaker. Then the SI built an EASE model to investigate where a good location for the column would be in terms of providing good coverage for the specified listener area. Next, I received a call re. possible recommendations on types of and amount of acoustical treatment. They had already provided a rough amount and cost, so this was more of an exercise of confirmation for the SI. My response to the questions provided was a series of questions. What, where, why, and how? The how was, how much is budgeted for acquiring acoustical data? Why guess or simply trust a simulation, when the space exists? Knowing the room volume, and types of surfaces (absorption coefficients), I could use a Sabine or Eyring equation, but that still leaves the variables of how accurate is the data on the surface materials? There is also the case of occupancy, where those equations simply fall apart and become inaccurate, due to non-homgenity of surfaces. Most important, we can quickly acquire some acoustical data and have most of the answers we need, without guessing! Let's measure it!
Acquiring Acoustical Data - Get Our Yer Balloons & Starter Pistols
The truth is, acquiring acoustical data is a simple process. For this project we decided to get multiple data sets that would provide for answers on "how bad is it", while also possibly creating more questions. Here we go!
Let's start with some pictures of the space. In order, the first three pictures are the receive locations for our measurements, named positions R01, R02, and R03, with R03 being closest to our (test speakers) energy sources. Our energy sources are a Dodecahedron loudspeaker (farthest right), a small powered 8" two-way/low Q (middle unit), and a 56" column speaker (farthest to left). All three were augmented by a 15" subwoofer that was placed on the floor. We added the existing loudspeakers to our group of energy sources, and you can see two of three units in the fourth slide, near the the ceiling.
Now that we have all the sources in place, we need to choose are weapons of choice for collecting the data. First up are our measurement microphones. An omni, a stereo X-Y mic, and a tetrahedral mic. The Tetra mic uses the shown Roland 4-channel recorder to gather data. The X-Y mic uses an application to record with my iPAD, and the omni mic is capturing Room Impulse Response (RIR) measurements using EASERA SysTune (software) via a firewire mic pre. Once everything was set-up, it took about 30 minutes to gather all the data sets.
You Got Data, Now What?
The first thing we can do with this data is do the needed post measurement processing to have all the recordings converted to Room Impulse Response (RIR) files in a .wav format. Since we were in the space, we took the time to simply listen with the best measurement device we can access, our ears. To experience the room again, we can use our RIR's and a convolution software to listen to dry speech, with the room "added in." To better match the on site listening experience, the following convolutions are of the stereo X-Y mic capturing data from the energy sources. Our receive location is what we label as R02, which is the center of the room.
First up, the column speaker, then it's the 8" two-way, and finally the existing system.
What did you hear? It should be evident that the column speaker provided for a much better Direct to Reverberant ratio D/R in comparison to the existing speakers. What becomes quickly evident is that the existing sound system can amplify an acoustical signal, but it actually lowers the intelligibility in doing so! This certainly isn't what the owner of the space wanted when in purchasing a sound system for their facility! Our friendly SI is on the right track with their loudspeaker recommendation of a digital steerable column.
What other data do we get from the measurements? More than this forum allows for showing, but lets look at the IEC 60268-16 metric more commonly know as the Speech Transmission Index (STI). This metric is used internationally to determine the suitability (intelligibility of) for sound systems in various applications. For Emergency Communication Systems (ECS), a minimum STI is "the law" in regard to a sound system meeting the requirements as determined by the local AHJ. Let's look at the STI scores for both the existing and column loudspeakers.
The column speaker STI score is 0.645, which is good for a reverberant space. The existing speaker STI score is 0.42, which is not a passing score for communications of any sort in this authors opinion, and according the NFPA 72-2010.
Room Decay (Reverberation)
The last piece of measured data to look at for this post, is the room decay time, often referred to at RT60, T60, or T30. These are all one in the same, as being the rooms statistical reverb time. This needs to be considered, as do other room acoustical properties, because not all communications in the space will be reinforced by a high Q column loudspeaker. In this space, two people can't have a conversation once they move more than say 10' apart, without saying "what?" often. Add some other energy into the space, like another pair of people having a different conversation, or many others having conversations …. .well, you get the picture. Here's the room decay times below.
The above room decay times are acquired using the Dodec as the source, since it has the most omnidirectional pattern, i.e. a Q of 1. The three plots are my three receive locations, R01, R02, and R03. For the data to be valid, the three plots should be nearly the same. The horizontal line(s) represent the average of the times for the octave bands centered at 500 Hz, 1 kHz, 2 kHz, and 4 kHz. These are the single number "reverb times" that are referred to often in describing the room decay of a space. From looking at the graph, we see the room decay time is 3.5s.
The final pieces of todays puzzle is the start to looking at prediction software. In this case it's the rough or "base" model created in EASE by our friendly SI. The model is very basic, but it does have the surface materials inputs as standard EASE materials. Here's the "wire-frame" provided in EASE.
Statistical Room Decay From EASE
In EASE you can view the statistical room decay (reverberation time) in the form of a graph, as shown below. It is important to note that this data is derived from either the Sabine or Eyring equations, and is ONLY accurate for rooms with homogenous surfaces. If this room had a carpeted floor, or a "soft lid", or any other large surface that wan't a highly reflective surface, the equations would be less than accurate.
Measured v. Modeled Room Decay
If you compare the measured room decay with the modeled room decay graph, you will see some correlation. I could average the data of the octave bands centered at 500 Hz, 1 kHz, 2 kHz, and 4 kHz for the modeled space and compare to the same average for the measured space. Modeled 500 Hz - 4 kHz average is 3.45s. Measured average is 3.51. They are almost the same, but if you look at the data at 250 Hz, you'll see the modeled room has a decay time of 4.4s, and the measured room decay at 250 Hz is at 3.7s. At 8 kHz, the modeled room shows a room decay of 1.25s, when in reality it is 1.7s, and that's a substantial difference!
At this point we could altar the room surface material to get the model closer to the measured data, but that would only be useful for gathering data for the room without any treatment, and/or without a large amount of people. The model will once again become less accurate as we change the surfaces. The answer to this problem is to move from looking at the simulated statistical data that is based on the geometric model, and to move to a more advanced acoustical study by using either ray tracing or particle emissions in the modeling program.
Stay tuned as we cover that in the beginning of our next post. We will then also introduce the proposed loudspeakers into the equation, and derive new data sets!
Originally published on: 7/22/2014
Improving Intelligibility In A Challenging Acoustical Environment: Part II - From Measured To Modeled
In Part I (see Part 1 below!), we introduced the concepts of the value of gathering acoustical data in an existing space as a reference. The data can be convolved and listened to at a later date, which is useful simply for the concept of not trusting our "acoustic memory" of what a space sounded like days, weeks, or even months ago. With a few mouse clicks we can take ourselves back to the space on an aural journey. The reference also becomes very important when we venture into the world of room simulations, such as EASE and CATT Acoustic. We can create more accurate models through a process of comparison of modeled and measured data results. Once we have an accurate model, i.e. one that is close to what we measure, we can then move on towards providing design solutions for both loudspeakers and acoustics.
A Newly Installed Loudspeaker System
As stated in Part I of this blog, our friendly SI had already done some proof of concept work in demonstrating the virtues of a loudspeaker that the owner agreed was an improvement over the existing system. The value proposition was made and the owner signed on the dotted line. A new steerable loudspeaker system was installed.
In my world that simply means more data that can be measured to compare to our room simulation (model). Based on the results of the models room decay data vs. the measured data, I also decided to get more "physical data" on the room. More on that later in a discussion of energy scattering. In acquiring Room Impulse Response (RIR) data for the newly installed system, I had a chance to audition the new loudspeaker in the space, and then take it back home by way of RIR, for the purpose of convolution. Like the audio files in Part I, I've convolved RIR's with dry speech to compare the existing or now "old system" with the "new system." For reference, I also include the column loudspeaker from the initial test
For reference, below is the model showing the three receive locations (R01, R02, and R03), with R01 being farthest from the loudspeakers. You might also notice the addition of the digital steerable (newly installed) loudspeaker above the other shown (test) loudspeakers. The SI steered and tuned the loudspeaker. Below is a comparison of the new loudspeaker ("Steerable Column"), the "Existing" or "Old" speakers, and the test "Column" speaker. I've convolved the RIR's with dry speech. The files are for R01, which is in the farthest position from the loudspeakers.
Speech Intelligibility Vs. Speech Quality
What did you hear from auditioning the above speakers "in the room?" I think we can all agree that the "old system" is unintelligible and the sound quality is poor. The column speaker used in my initial measurements has what would "score" as good speech intelligibility. Remember, this is a gymnasium withe a 3.5s room decay time. The "steerable column" that was recently installed also has good intelligibility, considering the space, and some might say it has possibly better speech "quality", in that it has more upper octave energy. So, does it have better speech quality or speech intelligibility? Let's look at the STI (IEC 60668-16).
First lets look at the existing "Old" speaker system, then the Column speaker, and finally the Steerable Column speaker, all at position R01.
The old installed system scores at or below 0.45, which on the STI scale is on the border between poor an fair. I would consider the existing system to have poor intelligibility, and even poorer speech quality. The Column speaker we used during the first tests scored at 0.575, which is at the upper end of the fair scale. Not quite "good", but "fair." The newly installed steerable column speaker scores at 0.54, which is also considered just a "fair" score.
Now go back and listen to the column and steerable convolved speech files again. In terms of being able to understand the words spoken, they are really equals, and to me ears the column does sound a bit more intelligible. One thing to consider here is that the column was unequalized, while the steerable column was equalized. The added high frequency content of the steerable might be perceived as improving the quality, but it doesn't improve the intelligibility. What were talking about here is whether or not you can understand what is said, and to what degree.
At this point we might start asking ourselves why the simple passive "column speaker" scores higher than the much more costly digital "steerable column" speaker?? Here's my take on this. There is a causal relationship between D/R, C50, specular reflections, and the resultant STI score. The math is complex, using 11 modulation frequencies of the 7 octave (approximate) centered bands of energy. The weighting of the Modulation Transfer Index (MTI) gives us the STI score, which is the horizontal dashed lines on the graphs. My belief is the lower score for the installed speaker is partially due to its placement in a soffit of sorts. The soffit provides for coherent and non-coherent summation of energy of the direct energy form the steerable column, and the angled side walls of the soffit. At some frequencies the soffit walls are in fact creating added sources, and the Q of the system decreases with the number of energy sources. Doubling of sources of like Q, lowers the D/R and C50 scores by 3 dB per doubling of sources. Those side walls also act as the side walls of any waveguide might, providing reflection paths for some energy, and diffraction of energy at wavelengths closer to the dimension of the opening itself. Another consideration is that the steerable column is mounted at a higher elevation, which results in the need for a larger vertical opening angle than the lower positioned column (test) speaker. The column has a nominal vertical opening angle of 14 degrees, and is actually around 35 degrees at 550 Hz. The digital steerable column has a user selectable opening angle. In this case there are two beams of 10 degrees and 20 degrees, resulting in the steerable column having a lower concentration of energy in the higher octaves, where the column speaker narrows up to 14 degrees in the vertical plane. Keep in mind that the weighting for the STI scale is more towards the upper octave band energy. The preceding is a very interesting topic, and you are very welcome to chime in! I'm going to leave this subject now and move on to what we are trying to achieve with our room model.
Adjusting The Model
We started with one model, and from that we acquired a statistical RT for the room, and we compared that to the measured RT or room decay. Given that the room is an almost ideal room for trusting the statistical RT (Eyring Equation), given all the surfaces have an absorption coefficient of well below 0.3, we would expect the modeled v. measured to be very similar. For the most part that proved true (see Part I of this blog), but it wasn't very accurate for all octave bands of energy, specifically the lowest and highest octaves of energy. This is why we measure whenever possible! We want greater accuracy! So, what to do now? Adjust the model!
We might consider this as much art as it is science. Maybe a better explanation is that we can use empirical data and common sense to make our changes to the model. The first thing you have to consider is what acoustical properties did you assign the surfaces in your model. I used what is labeled as hardwood court, 2x5/8 gypboard, and painted masonry. Truth is that I don't really know what is behind the gypboard walls in terms of how it is constructed, and it would likely vary for exterior walls verses interior walls, which this room has both. The painted masonry, isn't painted brick, it was the closest thing our SI could find to put in the model. The material was actually concrete masonry unit (CMU). Well maybe those are one in the same? The bottom line is the room decay is much longer at the lower octaves than what the model shows! Looking at the data for 2x5/8 gypboard, we find it shows absorption (transmission) by diaphragmatic action in the lower octaves, so that seems like a logical place to make an adjustment.
Making minor adjustments to the surfaces provide us with a "better model." Directly below is the room decay for the adjusted model, and below that, is the room decay for the measured room.
In terms of the one metric, room decay or T30, we are now very close to reality with our model!
At this point we could stop refining our model and start inputting acoustical treatment into the model. The treatment will help improve intelligibility for both reinforced and non-reinforced speech, and help control the ambient room noise levels that will interfere with speech intelligibility. My take is that is stopping short of what needs to be done! Why? I would agree that you could at this point take the above data, use the Eyring and even the Sabine equations to determine the sabins needed to attain a goal room decay, but that wouldn't be the best solution when considering the reinforced sound. The reason it isn't the best solution for the reinforced sound is because the SI has installed a highly directional loudspeaker, and a cursory look at the room decal in detail (ETC or Reflectogram) shows us that there are some high level reflections that arrive back at the listener area at deltas in time and level that are cause for lowering speech intelligibility and speech quality. In other words, high level reflections are cause for low intelligibility scores.
By Nick Shively
All too often I've been called in to run sound at an event as a hired gun, only to step up to a mixer configuration that flies in the face of logic or reason. Sure, I can grit my teeth and power through it, but if I had my way I would rip out the snake and repatch everything. With the advancement of digital mixers, getting what I want where I want it has become a ton easier. This topic maybe be old-hat for the experienced live engineers, but for the upcoming wave of newbies, I may be able help users bypass the school of hard knocks.
More and more I am witnessing corporate production companies make the move to supplying sound for concerts, and with multiple band changeovers involved, the mixer layout becomes paramount. Moving from the boardroom to the stage can be overwhelming the first few times out.
Typically there will be one main snake-head that makes a home run back to mixer. Stemming from said snake-head can and often will include sub-snakes and breakout boxes to different sides of the stage. On paper, your stage plot makes sense, however, reality quickly sets in as the front of house engineer walks up to the mixer for a line check and the instrument he is expecting on channel 1 ends up on channel 17 like an intern is playing a cruel joke of audio Boggle with your inputs.
Now you're stuck having to over-think, losing precious time trying to find that microphone that is feeding-back. You need a system that you are able to standardize upon, being able to reach for a channel without looking and make adjustments. Coming from a studio background, I tend to layout my channels as I would see them in my recording software:
When called in to run sound, first thing I do is standardize my mixer layout: drums first, followed by bass, guitars, keys (or just DI boxes in general) and then vocals. This allows for quick, almost second-nature esque adjustments. Sounds simple enough, yet I see it overlooked all too often; kick drum on channel 9, snare on channel 15, bass guitar on channel 2, etc. With that, we welcome the advancements of the digital mixer.
Most digital mixers nowadays allow the user to move channels around on customizable layers, or at the very least, allow the user to pick and choose which input socket the channel is receiving it's source from. In the case with the above pictures, a GLD allows the user to fully customize the layout of the mixer by dragging and dropping channels without the need for digging deep into menus or going cross-eyed with a matrix patch window
So now you have your mixer laid out how you like, everyone is dialed in and everything sounds great. Enter the world of DCAs (click here for the uninitiated). Consolidating a large number of channels down to a manageable few can be an important part of one's workflow of an event - you are now able to access a single fader to control a myriad of channels. So for example, set your drums, guitars, keys and vocals on DCAs 1-4 and you are now able to easily massage the mix in real time during the performance. This becomes especially handy with the era of iPad control; how are you expected to quickly navigate all of your inputs on a tiny little screen? Just pull up your DCA faders and you're golden.
Stuck on an analog mixer? No fear, you're not left out in the cold. The same basic idea can be achieved via VCA or Group channels. Click the above link for details on the differences.
Was this information helpful? Lets hear your thoughts in the comments. Hopefully this has pointed some aspiring engineers in the right direction, or acted as a refresher course for the veterans. In the coming weeks we will be covering different aspects of live events, from band changeovers, handling monitor mixes, to my biggest pet peeve question: "Cool speakers, how many watts are they?"
The question of accuracy of a acoustical room (model) simulation is found when we can compare modeled to measured data. Building an EASE model is easy, but does the geometric model hold up under scrutiny when considering a variety of acoustical metrics? The first thing to keep in mind is that this author is unaware of any modeling software that treats sound energy like sound energy. I'll qualify that by saying that "the programs" use ray tracing or particle emission from energy sources into the modeled space as if the energy were of infinitely short wavelengths. Considering the energy to be more like emitting light sources into rooms. There are a few problems with this model, and it becomes more apparent at longer wavelengths, i.e. lower frequencies. The modeling software works "good" for higher octave band energy. Where this comes into play is how the programs deal with surfaces, and surface irregularities. Here's two images that illustrate the issues
The images are Energy Time Curve (ETC) plots for the same source and the same receive location. First the measured data, then the data from the room simulation.
Note that at about 80ms after the initial impulse that the room simulation (lower graph) has several higher level peaks of energy that are not present in the measured data! Is this important to consider? The answer is yes if you are considering room acoustics in your design. This is true even if the room acoustics aren't going to be altered, because your data for various loudspeaker system designs will be affected by the erroneous data. The culprit for these errors is the surface data for the surfaces that are cause for these reflections. In room simulations we can try to find these via ray tracing or graphs that show the direction of specific peaks (reflections) of energy. In EASE AURA, these are found in the pulse directivity plots, as shown below.
The above pulse directivity graph shows the direct (red) source energy arriving at the receive (0 dB attenuation) position. Also shown is a chosen single (yellow) pulse that is arriving from behind the energy source, and lower in elevation then the energy source. From the data we know that this reflection comes from the front (rear wall of altar in this case) wall. Other data not shown in the above image, but included is that the reflection is -10 dB, and arriving 28ms after the energy source.
In cases (like the ETC measured v. modeled example above) where this pulse is "false" data we have means to remove it from our model. If it is too high in level, we assume that either the absorption or scattering coefficients need to be adjusted. If we built what most would consider a geometrically correct model with acoustically "correct" surface materials, the answer is found by adding scattering.
On 7/3 we recorded log sweeps and acquired RIR's using the previously mentioned sources, a column loudspeaker, a low Q speaker, and a dodecahedron speaker. All with a subwoofer to extend the test speaker(s) response. We also collected data on the installed loudspeakers, (3) low Q devices mounted at the apex of the ceiling, midpoint between transepts, over the front of the altar area. The following information contains snippets of the data, including pictures, descriptions, and audio files.
Below is a floor plan view of an EASE model "wireframe" of the room. The light green icons are test microphone receive positions, starting at the far end of the nave, and moving "down" towards the altar (positions 1, 2, and 3). A third position is shown in one of the transepts.
Below are my first attempts to upload some audio files. The files are convolutions of RIR's and "dry speech". There are in the following order:
Dodecahedron @ position 1 - RIR acquired using XY microphone
Dodecahedron @ position 3 - RIR acquired using XY microphone
Column Speaker @ position 1 -RIR acquired with mono mic and EASERA SysTune
Column Speaker @ position 3 - RIR acquired with mono mic and EASERA SysTune
Special Note: Although I intend to continue with the process, and follow through to the finish of designed, installed, and measured new system, I am going to start another Blog (same topics) for a different space. This space is a Catholic Church, and upon viewing plans. looked like it would be a good candidate for the blog. Unfortunately, or fortunately for the church, the room acoustics aren't nearly as interesting as the simple room geometry would suggest. The room has decay time of sub 2s, i.e. I wouldn't consider this a difficult space to achieve good intelligibility using many design approaches! Onward. I'll test to see if my audio files play. If they don't on first try, I'll seek a remedy. My apologies ahead of time for being a newbie at "blogging."
Todays pairings are the Tannoy VX12HP and VSX12.2BP sub. Both powered by a Lab.gruppen IPD amplifier. Below we show a screenshot from SysTune, while using the audio delay alignment tool within the Virtual EQ tab. It's almost like cheating!!!
Below we see the delay analysis tool in Systune being utilized. The top plot shows the magnitude response of the top and the sub independently, with the combined response for a chosen delay time. The options are shown in the plot below. I've chosen the "SUB" TF as the reference, since the high order low pass filter will "push it back" in time. Note we have choices of "Pos" and "Neg" summation. I chose positive, delayed the top 7.66ms, and the results are it measured as predicted. BTW - The rig sound amazing! Clean as a whistle, with superb definition across the spectrum.
Prepping some boxes for measurements and future demonstrations. Measurements to include IR capture, to show magnitude and phase response. This post will start the series that will include posts on measurement (best) practices, including driver alignment, carded and end-fire LF array measurements, how to read phase plots, i.e. what is and what isn't a non-minimum phase event. Hope to see you follow, comment, and add in your data/experiences!!!!
Infocomm 2013 was a great experience! I complain every other year when we have to go to Orlando, and yes that's still a big time suck, but the show always is great. Getting a chance to see and talk to colleagues, friends, and new acquaintances from across the world makes the trip worthwhile. Here we see the Allen & Heath booth is 'rockin', with the worlds most innovative digital mixers!