Archive for the ‘Tuning Techniques’ category

Toning Your Sound System

September 10, 2010
No this is NOT a typo. I did not mean to write “Tuning your sound system” because that is entirely a different subject. So what is the difference between toning and tuning?

 Here is a simple example from the muscial side: This is my son Simon. He has a guitar effects pedal that has exactly the TONE of Eddie Van Halen. One small thing though: he can’t TUNE his guitar.

A legend in his own mind

 

Sound systems also have a similar contrast between these two concepts. Tuning  a sound system (in my estimation) is where you adjust the system so that it has uniform response over the listening area, with minimal distortion, maximum intelligibility and best available sonic imaging. Tuning is about making the far seats similar to the near seats. An objectively verifiable – but verifiably unattainable goal of same level, same frequency response, same intelligilbility throughout the room. Making the underbalcony as similar as possible to the mix position (which hopefully is NOT under the balcony). It is about making sure every driver is wired correctly, still alive, aimed at the right place and cleanly crossed over to the next one. It is about making it so the mix engineer can mix with confidence that theirs is a SHARED experience. Because it an objective pursuit, the use of prediction tools, analysis tools and our ears all play important roles in the process.  It does NOT, however mean that it sounds GOOD. “Good” is subjective.

Toning, on the other hand, can’t be done wrong. It is entirely subjective. Toning a system is the setting of a bank of global equalization filters at the output of the mix console that drives the sound system. If you want to set it by ear fine. If you want to set it by 10,000 hours of acoustical analysis containing mean/spline/root squared/Boolean averaging then go for it. If I am the mixer and I don’t like it, I will change it. Too bad. I like MY tone better. Deal with it. I don’t like flat. Deal with it. I like flat. Deal with it. There is nothing at stake here. Nothing to argue about. And no need to bring objectivity, or an analyzer to the table. The global equalizer is just an extension of the mix console eq. In the end the mixer will choose what they want to eq on a channel by channel  basis and what they want to eq globally. But also in the end there is no wrong answer, because it is entirely subjective. I have worked shows where, in my opinion the mix sounded like a cat in heat. That’s my opinion, and therefore not relevant, unless asked for. I asked the mixer “Are you happy with that?” They say “Yes”.  As long as I have ensured the cat in heat is transmitted equally to everybody in the room (i.e. TUNING the sound system), my work is done.

Good toning enhnaces the musical quaility, or natural quality of transmitted sound. Good tuning ensures that the good (or bad) toning makes it beyond the mix position.

 Piano Tuning…. and Toning

One does not have to know how to play a piano to be a competent piano tuner. It is an objective pursuit. Numbers. It can be done with an analyzer and/or a trained ear. The toning of a piano, a subjective paramater, cannot be wrong. John Cage opens up the piano and scatters nuts and bolts on top of the strings. This “tones” the piano. Is it wrong? Of course not. But before John Cage plays the “prepared” , i.e. toned piano, do you think he has it TUNED?  You bet.

John Cage Prepared Piano - a subjectively "toned" piano

 Below is another example of a “toned” piano.

I always wanted to find a way to work a deer head into my music

 Below are the tools for TUNING a piano. Similar to the ones we find our artistic auto mechanics using to TUNE up our car.

Tuning Forks

Hmmm..... Digital calipers: Objective or subjective?

Strobe tuner: otherwise known as a frequency analyzer

 

Just semantics or more?

So why do I make this distinction?  Because I have recently experienced several cases where people are confusing these concepts. In one case a guy wrote an article about how much better systems sound if they don’t have a flat response. Better to have peaks and dips. He notes that people that tune sound systems with analyzers do the clients a disservice by making thr system “flat”. Who am I to argue with this. He doesn’t like flat. OK. However, in the course of putting down acoustic analyzers for global equalization, the article never mentions the OTHER things that we use analyzers for: checking polarity, aiming the speakers, adjusting splay angles, adjusting relative level between speakers, setting crossovers, phase alignment, intelligibility analysis, treating reflections or most importantly: working to make it sound uniform throughout the room. The article compares equalizing your church sound system to your home hi-fi, which is to say TONING the system.  Maybe this guy’s approach is great for toning the system, but it is useless for tuning the sound system. The article “The fallacy of a flat system” can be found here

———————-

Then I received a question from one of my recent students from Asia:

Dear Bob:

Last week I join the BRAND X SPEAKER COMPANY seminar, they use another method to alignment the line-array system.

1) the whole line-array should be same EQ & same level.

2) they use room capture software to alignment the line-array system. They capture about 15 trace at difference mic position in the venue but not on axis speaker position and finally they sum average of the trace to 1 result then EQ it. What do you think?

————–

This was my reply:

1) the whole line-array should be same EQ & same level. 

I cannot find any good reason for this. The lower area is covered by the lower boxes, the upper area by the upper boxes. They are in very different acoustic environments, they are very much at different distances. Why lock yourself into a solution with no flexibility? If the end result is a perfect match… then great. If not…what can you do besides make excuses?

2) they use room capture software to alignment the line-array system. They capture about 15 trace at difference mic position in the venue but not on axis speaker position and finally they sum average of the trace to 1 result then EQ it. What do you think?

This solves NOTHING. The end result is the same eq to all speakers. If it was an average of 2 positions or 20,000 positions the average is still just ONE set of parameters. If it sounds different in the front than the back before you average then it will sound exactly the same amount different AFTER the average. Why bother to take samples all around the room if you are not going to do anything about the DIFFERENCES around the room? It is just a waste of time.

The only reason to use an analyzer is to get objective answers such as: is it the same or different?  Not for subjective ones such as – does it sound good?

Example: Let’s say you average 20,000 seats and put that in as the eq for all speakers. Then the mixer hears it and wants a boost at 2 kHz.  What are you going to do? You are going to boost 2 kHz or get fired. Who cares about the average now?

6o6

———-

In this case a manufacturer is using toning techniques without dealing with the tuning part. BOTH must be applied if we are going to bring the tonal experience to the people that pay to hear our sound systems.

Conclusion:

Keep an eye on both sides of the issue, but bring the right tool for the job:

Recommended system tuning attire

System toning outfit (Women only PLEASE!)

 

Toning apparel for men

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If you have ever tuned a sound system – watch this video – 3 min.

September 2, 2010

This is a highly informative  – and realistic – short video on the how a system optimization engineer and FOH engineer relate in their job roles.

We have Bennet Prescott to thank for this. I only wish I had done it myself.

SIM3 Optimization & Design Seminar at UC Irvine

June 29, 2010

We just completed a 4-day SIM3 training seminar in the south side of southern California. UC Irvine is located very near the ocean, which makes one wonder how folks could study when the surf’s up. It is also right next to John Wayne Airport. Naturally I flew in and out of LAX, and drove the hour down to the other airport. Why? Because I live in St. Louis, which USED to be an aviation town (ever heard of Charles Lindbergh, McDonnell Douglas or TWA? – all just museum stuff now.).

Measuring, measuring, measuring

We had a good sized class of 19, including grad students and professors from UCI, some engineers for Creative Technologies ( a rental house specializing in corporate), some freelancers and two special guests: Daniel Lundberg  and Jamie Anderson. There were 3 people (not including Jamie) who had attended my seminar previously and were returning. This is, for me, the highest honor and I am very grateful for the support of Will Nealie (whose photos are shown here), Chuck Boyle and Szilard Boros.

The Venue

We were fortunate to get to do the seminar on the stage of the 300 seat Claire Trevor Theater. This allowed us to measure first in the controlled circumstance of the near-field on stage and then work our way out into the house. As an added bonus we were allowed to measure (and re-design and retune) the house system, which had an up-to-date line array type system of 8 x Meyer M1-Ds.

The class moved along very smoothly. We covered LOTS of ground and the acoustics of the hall were very favorable so that students could get a look at what real systems can do in a good hall.

The class progresses in complexity over the 4 days, beginning with measuring a processor, then on to a near field single speaker, adding a subwoofer , near field arrays, distributed arrays and then out in the house where we design a full system and tune it. All the while the progression of complexity is underscored by the theory behind the data. The number one focus point of a SIM3 seminar is understanding what the data says and WHY they data says that. Proper diagnosis must ALWAYS come before treatment, and all treatments need follow-up testing. If they don’t work then get started on a new diagnosis. SIM school tunings are never rehearsed so when something shows up on the screen, we all are seeing the data for the first time. There are always surprises and this was no exception.

In the course of the tuning here we found that our original design had too much coverage for the room. If we had gone to MAPP On-line or even used a simple protractor on the plan view of the room this could have been seen in advance. But PURPOSELY we did not use those tools to find the answer. It is better for the learning process to see how unkowns can be decoded by the analysis methods. The “Main” array was 2 x UPJ-1P in a coupled point source, located at the house left stage edge. Our goal was to cover evenly across the room – a straight horizontal line along the 3rd to last row. As we measured the 1st speaker across the row we could see that it cover ALMOST the entire width…. almost. Adding the 2nd speaker was WAY too much, leaving it off, left us 4 seats short of the aisle. Conclusion: Our design was flawed. (This made it just like a REAL gig except that the designer’s ego was not at stake).

 It is much better (as a learning experience) to use the SIM 3 Analyzer to prove the design was wrong and to force us to consider the optimization options that had the highest prospect for success. If only we could wave a magic wand an turn this 80 degree speaker into a 50 degree speaker!  Oh….. WE CAN!     In this case we rotated the UPJ-1P horn on the 1st speaker (they are 80×50 rotatable) so that we got 50 degrees of horizontal coverage for the “A” speaker (the longer throw). This covered enough of the room to make a successful, smooth transition to the B speaker. Then we added the “B” speaker – too wide again until we rotated its horn as well. The end result was even coverage across the straight line  of the 3rd to last row within 1 dB.  The process involved measuring on axis, at crossover and off axis until the splay angles were optimized, the eq’s set (individually and together), levels set and delayedso they were phase-aligned at the crossover. Then we added the subwoofer to the array in both an overlapping and non-overlapping mode (different delays were needed for this). Finally we added a small delay speaker to extend the coverage evenly into the corner. We even took a few minutes to show the effects of adding excess delay (the side effects of the Haas Effect) and watched as the coherence and combined level at the delay location became worse than if the delay speaker had been turned off. This is always an eye-opening moment at my seminars. 

Tuning (and retuning ) the Line Array

Because the class moved along so quickly we had the luxury of extra time to take on the house system. This system is made  available for students to re-design, re-hang, re-angle, re-tune, re-etc…….  This particular config had been specified by a student AMA (against medical advice) so the professors were quite interested to see how it would look under the scope. The answer:  ________________________flat line.

The horizontal orientation was the most severe in-tilt I have ever seen (OK I am pretty new to this but I have seen a FEW systems). It was such an inside job that the Left side of the PA missed most of the…………. left side.  The mix position was in the very rear of the house right side. From a horizontal standpoint the left speaker was pointed at the right wall IN FRONT of the mix position. If you are having trouble visualing this here is a pic to help.

Horror-zontal aim points for the PA

So we measured and found that the left cluster was more than 6 dB louder on the house right than on the left. Obviously the speakers would need to be opened outward. 

Before- ONAX A vs OFF A - Off mic is near top row at the last seat on house Left

We had, however, spent the previous 2-speaker tuning  focused on the horizontal plane interaction between the pair. Here we had 8 boxes in the vertical plane– that is what we wanted to see – and we had 5 mics running from top to bottom in a diagonal line where the speaker was pointed.  As it stood, nobody knew what the current vertical angles of the cluster were. We had the 8 boxes wired in 3 zones 3-3-2 as an ABC array. It was offered to bring down the array and see the angles – then we could play in MAPP and see the response…….NO, NO, NO. Much better to turn it on and see what we have. This way we can learn how to hunt down an array in the wild. We know these 3 subsytems are out there – but where?

I don’t recommend working on systems where you don’t know where the speakers are pointed, but it is important to be able to find where they actually ARE pointed – even if you have a piece of paper (or I-pad) to show you. The learning experience here was the process of finding. Here is a pic to get the idea of where the mics were:

Mic placement strategy

Terminology sidebar – ONAX (On Axis), X (Crossover), VTOP (Vertical top),VBOT (Vertical bottom).

Before we get to any tuning, we dummy checked each mic and speaker to make sure we had everything wired right. In the course of this we set the delay compensation for each mic and they ran from around 50 ms to 13 ms so we are looking at near seats that are around 12 dB closer (a ratio of 4:1) than the rear seats. The array will need to overcome this difference in proximity.   

So we began with just the upper system “A” on (the top 3 boxes). We compared Onax A, VTOP and XAB positions. VTOP (around the mixer ) was a disaster. No HF, no coherence and the far side much louder than the near side.

Original angles - ONAX A vs VTOP

UCI M1D R1 - VTOP A Solo -before EQ

 

Perfect mix area!!!!! XAB was down slightly from ONAX A so now we knew (vertically) where “A” was pointed: Too low.

Before- ONAX A vs XAB

The cluster was already very high so we can’t move it much.  The real answer would need to be getting some up-angle in the array. This would require some real-world rigging and this was not going to happen in our short time frame so it does not seem that we will fully solve the mix position.

Onward. We moved the ONAX A mic up and down a row and found that our original position had the most level – we had found the center of A. We eq’d it and stored it as a reference level.

Next begins the search for B. We looked at the ONAX B mic and moved it up and down until we found its high-water mark. The level at B was stronger by about 3 dB (compared to A). It was also about 3 dB (70%) closer. This made it obvious that the splay angles chosen for this array were wrong. How did we know?  The job of the different splay angles is to create a matched level at different distances. Here we were seeing that as we got closer, it got louder – the expected propeties of getting closer to a symmetric, non-directional source, not one that is creating asymmetry in the vertical plane. We eq’d the B system and reduced its level 3 dB.

Next up was the bottom two boxes, system “C”.  This system covered the front rows REALLY well. It was 7 dB louder than at the back and we were still in the 4th row.  It got even louder up closer but we gave up.

Before- ONAX A vs ONAX C

Conclusion: The cluster system needed to generate around 12 dB of level difference from top to bottom. It actually achieved 3 or 4 dB.  Time for the cluster to come down and redesign the system.

Redesigning and retuning the Line Array

We have no drawings of the room. Not even a napkin sketch. The UCI internet is not getting through to my laptop. We are going to have to go it alone. 

This is what we know (a) the cluster is too low, we have more than enough angle to reach the bottom and we need 12 dB more level at the top than the bottom. This means that the splay angles for the C section need to be at least 4x wider than the A section.

So how do you design a line array with no Manufacturer Official Line Array Calculator, no Mapp On-line, no drawings? We need to know the angle spread from top to bottom, and the difference in range from top to bottom. So we looked at the existing angles and found that the overall angle spread was 40 degrees. We know that was more than wide enough. We know we have a 4:1 distance ratio.

So we need 35 to 36 degrees of spread – we have 8 boxes (7 splay angles) – the average splay angle will be 5 degrees. (5 deg x 7 = 35 deg). We know the widest angle we can get for an M1D is 8 degrees. If we have 8 degrees at the bottom and 2 degrees at the top (a 4:1 ratio) we will approach our 12 dB range ratio. Add ’em up  (2-2)-3-(6-6)-(8-8) = 35 degrees. System A = is 3 boxes at 2 deg (a 6 deg speaker), 3 deg splay to system B (a 12 deg speaker) and then on to C (a 16 deg speaker).  Here is a picture of the design in progress: Yes – that is the AS-BUILT paperwork under my hand.

Calculating the splay angles based on range ratio

The new angles were put into the cluster and up it went – pretty much as high as it could reasonably go (about a foot or two higher than before) and we resumed measuring. This went very quickly now. The center point of each subsytem was easy to find since they each were made up of a symmetric angle set. The center of A was at cabinet #2, the center of B was at # 5 and the center of C was between 7 and 8. Each system was eq’d separately and levels set. The level tapering needed to bring the lower systems into compliance was 1 and 3 dB respectively, a far cry from the 3 and 7 dB previously. The systems were combined – first A & B and then C was added and a very uniform frequency response and level was created over the space. The level from front to back (back being the top row) was now 1 dB. The mix position still sucked – but we knew we could not save that without a rigger.

Reworking the angles

 First we looked at the ONAX A position, and EQ’d it. This will be our level/spectral standard going forward.

The next step was to look at the response at the mix position. We expected that things would not be improved much here since we were not able to aim the array up high enough to hit here……..and we were not disappointed.  Well I mean we were not surprised.

After- A at ONAX A Compared to A at VTOP

After - Response of A solo at ONAX compared to B solo at ONAX B

 The EQ applied is slightly different for A and B respectively. The difference is minor because both “speakers” are comprised of 3 elements. The splay angle is different which creates a different summation gain of 3 dB – the correct amount to compensate for the difference in distance.

After- A at ONAX A Compared to AB at ONAX A

Above – You can see the addition at A that occurs when B is added. The response shows no loss but the gain varies with frequency. As frequency falls, the percentage overlap increases and the addition increases. At 8 kHz the percentage overlap is so low that we see no addition. By contrast, at 125 Hz we see 6 dB addition. All frequencies between show gain values between 0 and +6 dB. This is a great example of 3rd order speaker behavior.

After- AB at ONAX A and ONAX B

After- AB at ONAX A ONAX B and XAB

 

After- HF ZOOM - AB at ONAX A ONAX B and XAB

 Above is a zoomed look at the uniformity of the HF levels.

Combined System A+B  EQ

Once A and B are combined we look at the LF region to see where the coupling was shared in both directions. Frequencies that were boosted in all locations can be equalized by matched filters in the A and B sections. In this case a 160 Hz filter was applied. Below we have a zoomed in comparison of before and after the AB Eq was added.

After- AB at ONAX A -Combined system EQ

 The screen below shows how we have restored the Combined AB response to the same shape as our initial A solo reponse.

After- AB at ONAX A -Combined system EQ compared to solo A EQ

 The AB sytem is now complete

After- AB with combined EQ at ONAX A ONAX B and XAB

Combined System: Adding (AB) + C

Now that AB is complete we turn our focus to C. Speaker C (2 boxes) was EQ’d as a soloist and it’s level set to match the ONAX A standard. The solo EQ response appears below.

After- C at ONAX C EQ and Level
The response below shows the full combined response ABC at ONAX A and C positions, giving us a clear view of the difference between top and bottom (not much!). The distance ratio between these two locations is around 8 dB!

After- ABC at ONAX A and ONAX C - top to bottom compared

 Finally we sell the full system ABC at its 3 main locations.

After- ABC at ONAX A, B and C

Was this the best way to design a system? I would not recommend it, if you have the option of drawings etc. But in the end we still have to test it – and that is where the final design comes from. In this classroom setting we made the tuning process drive the design. What we learned from our data was translated into an updated design and this was then measured. The result was a winner. This process, in a few hours was a distillation of 25 years of work for me. Everything I I ever learned about design came from the process of trying to tune an existing design, and learning from it.

There are additional class photos which will be placed in the “Seminars” Page on the right of this blog page.

and finally………………….

I did manage to bring home some good data from this tuning so I will add those to this post later. Soon… I promise.

6o6

New York Trainings – Updated with Pics

May 19, 2010

I am in NYC this week for two rounds of classes: SIM3 training in Brooklyn and the Broadway Sound Master class on the NYU campus in the East Village. The SIM class is at City Tech in Brooklyn – which is where John Huntington teaches. We have several of his students joining the class which is really nice. This trip marked a first for me – even though I have been coming to NYC quite regularly since 1984 this was the 1st time I ever set fot in Brooklyn. We arelocated over by the legendary brooklyn bridge and I can see it from the school – so hey – I can add another borough to my list……Manhattan, Queens – been 2 places there – La Guardia AND JFK – wow, and now two places in Brooklyn – City Tech AND Peter Luger – the famous steakhouse.

We are halfway through the class and pretty much right on schedule. Looking forward to the BSMC this weekend – always a great learning experience for me

****** John Huntington was kind enough to take some pics of the seminar.

6o6

Macau COD Tuning: Day 7 (Underwater SIM) – Updated with screen dumps

May 16, 2010

Today was scattered around different tasks. We tested the repositioned Sound Beams, further refined the Constellation settings, picked off stragglers in need of verification – or RE-verification, and finally SIM’d the underwater speakers.

Soundbeams repositioned

First order of business was to find out what we had gained from moving the SB-2’s up 2 meters. The answer: 6-8 dB, quite substantial and well worth the effort. The area with the most image distortion from the main arrays is “the beach” , the rows close to the water edge. These stand to benefit the most, vertically, from the SB-2’s contribution. The close seats have the lowest risk of hearing the unit directly behind them . By contrast, the  high ground seats near the back need the SB-2’s the least, but carry the highest risk of  experiencing distracting localization and pre-echo from the SB-2’s.  And yet there is another layer at work once we get the full circle of crossed swords into play (all 8 SB-2’s). The rear seats have the highest amount of isolation – being in the pattern of only 2 of the SB-2’s (the one facing them and the one behind).  Also in their favor is the front-back asymmetry of the human hearing system, which favors the front by virtue of our pinna.

The lowest seats have the most interactivity between the 8 crossing SB-2 patterns, being much closer to the the angular coverage edge of the 6 remaining speakers. The multiple extra arrivals come from the sides, which are much easier to localize.  What we have is a very complex weave of multiple arrivals out of time and differing in level. It can’t be solved with delay unless you are MC Escher – because it’s a circle. The only solution is careful monitoring of the SB-2 level with respect to the other main systems. Fortunately you can get a lot of localization effect with very little level, so the SB-2’s will be able to add an image steering with minimal detection   

Curtain Time

Sound is invisible. Highly directional devices are REALLY challenging to aim at a spot 130 ms away – even on a clear day. But there is NEVER a clear day in the shower right?  Well our SB-2’s are behind a shower curtain so we could not do simple tasks like putting a laser on top and seeing where it goes. It goes a foot in front of the speaker. Great. So we have to measure it and find it, in order to verify that the 4 SB’s on the opoosite match the near side. This went ok for the most part – but #3 did not match its brother across the ocean. Different in 3 positions, and NOT making any sense in terms of angled down – or to the left etc. The culprit as it turned out: folds in the curtain. All but #3 and 4 had no folds in front of them. #3 abd 6 did – and they did not match. Where is the interior designer when you need him!  

Underwater SIM

Having completed our must-do list we could not resist the chance to measure the underwater monitor speakers. These 36 speakers are used to communicate to the divers, the swimmers and to play music to keep the swimmers on cue.  The large quantity is due to the fact that the pool is a highly variable environment with lifts going up and down, bubbles, fountains, leaking oil wells – oh wait – no that is the Gulf of mexico – anyway LOTS OF STUFF GOING ON. So we put in a DPA hydrophone, plugged it into SIM and got a transfer function of some of the speakers in the water. VERY AMAZING! I have some plots which I will post when I get a chance. I have NEVER seen so many strong reflections – the impulse response looked like a 3 minute earthquake. The freq response looked terrible – but it looked AMAZINGLY like the spec sheet for the product. We eq’d a bunch of different ones and even found one that was reverse polarity!

When we were all done the next some folks got in the pool to listen. Quite cool. We have renamed it the SIMMING pool.

**** Update*****

Here is some data from the underwater measurement and tuning. The impulse response is quite fascinating because the reflections were so much closer than I expected. I was thinking only of the reflections off the sidewalls and not so much of the floor and ceiling. The CEILING?  Well in waterworld the H2O/Air interface is like a wall. Not much sound energy moves across that barrier and a LOT comes back down. This was pointed out to me by Dr. Roger Schwenke and the up/down bounce is very much in evidence in the impulse response. The density of the reflections is greater than I have ever seen for a set of speakers and their wall reflections. They are closely spaced roughly 1.1 ms – which is the same distance as 5ms in air,  a wavelength of about 1.7m (5.5 ft).

Above – impulse response of underwater speakers. First arrival is at 11.25 ms.

Impulse response: Underwater speaker closeup

Above. You can see the approx 1.1ms spacing

Here then is a freq response of the speaker(s). We did an initial eq at 890 Hz, the worst offender. You can see a before and after eq screen shot here. I turned off the coherence blanking to help see the extent of the damage these reflections threw at us. THe phase response looks so mangled you might thing we set the delayfinder wrong – but no. It it is right………….for the 1st arrival at least, just not so right for the 2nd, 3rd, 4th, 5th,……….  Interesting, though was the fact that, in spite of all the reflections, the basic tonal shape was quite consistent. Consistently bad, but consistently similar to the manufacturer’s data sheet. Concistent means we can EQ and get some positive effect.

 
1st Eq (before and after) of underwater speakers

As we looked at more and more of these we saw the basic shape come through and refined the EQ. The screens here shows before the EQ was applied to the system and after. Flat as a ruler eh????????????? LOL

Before EQ

After EQ

Before and after EQ

Macau COD Tuning: Day 6 – Constellation

May 11, 2010

Constellation Tuning

Most of today was dedicated to Constellation tuning. Steve Ellison programmed up the menu of user-settable presets that will become the painter’s pallette for the system designers, Francois Bergeron and Vikram Kirby. The pallette gives them easily understood parameters such as the the reverb Time, gain, etc, that will allow them to tailor the response of the sound system/room to the music and spectacle as the creative process unfolds. 

The beauty of electroacoustic architecture is that the acoustics can be reshaped from song to song, gradually so that the audience has no conscious awareness or the opposite: a dramatic moment to create a strong conscious effect. The settings can be made completely plausible for the shape of the space that you see around you, or can be dryer and more intimate than you might expect or, of course, much larger and more reflective. Once the lights go down, the mind loses sight of the scale of the performance space, and creative minds can begin to operate on rescaling the room to most appropriately contain the soundscape.

Imagine yourself having the ability to pull down a wall of thick curtains in a small room and reveal the walls of the Notre Dame Cathedral behind them. This is the level of capability now in the hands of the system designers. This is NOTHING like having a Lexicon at FOH. I use this simple analogy: A dry room with house reverb puts the singer in the shower but leaves the audience watching from the desert. (Who the singer is that you imagine in the shower I will leave up to you). All the reveberation is in front of the listeners, and the room acoustics clues of spatiality are missing. A room with electro-acoustic architecture puts us ALL (audience and performer) in the shower, desert, or something in-between TOGETHER. The spatial clues are there – precisely because they are ACTUALLY there. An audience member’s clap will reverberate from the “walls” just as the performers do – this absolutely will NOT happen with FOH reverb.

Yes it can happen with actual hard walls – but walls only have one setting. Yes it can happen with variable acoustics (moving panels, drapes etc.) such as we see in some modern concert halls. But the Constellation system does not require a four hour labor call to open chamber doors, drop in curtains etc, to move a hall from highly reverberant (symphony) to less reverberant (chamber), or to extremely reverberant (organ). Constellation can move in seconds, with a single click (or cue) from dry enough to feel a tight, pulsing, fast-paced drum beat all the way to cathedral chanting (and very importantly, the land between).

It is no coincidence that this capability was designed into this system. Francois Bergeron has been an expert in complex spatial sound systems for all the years I have known him. After all he is the guy who programmed “The Little Mermaid” show for me at Tokyo Disney Sea, where an entire orchestra is swirls around and goes down the drain. It has been running there every 20 minutes since 2001.

Hopefully you get the idea. What Steve, Pierre and I will leave Vikram and Francois with with will be a simple web page with programmable presets which can be logged in as cues in LCS. Then the fun begins, integrating this into the production.

SIM Tuning – Leftovers

In addition to Constellation tuning there were a few leftovers from our previous tuning work. We had to check the 28 Melodie boxes of the 4 opposite side clusters. The fact that we waited this long for this step lots about the quality of the install by Solotech. The very small number of wiring/install issues gave us high confidence that these clusters would be in good shape. 28 speakers checked: 28 speakers good.  

The fact that we did not have to reposition any of the 8 clusters or modify any of the inter-box angles says lots about the quality level of the Thinkwell design team. Anyone reading this who has been on a job site where I did the SIM tuning knows what the odds are that speakers are going to SUBSTANTIALLY moved is: very high.  In this install there were 56 Melodies (Mains),  23 CQ-2, 32 UpJunior, 43 UPM-1P, 32 MM4-XP (Surrounds), 10 600 HP (subs) and 24 UPJ-1Ps (Constellation) that required NO REPOSITIONING OR ANGLE CHANGE.  The cardioid configuration of subwoofers was re-angled to make best use of its cardioid steering (a very simple job for the riggers). Only the Soundbeams had to be moved (two meters rise in level and angularly adjusted) – again a minor change in the big scheme of things.  Hats off to the Thinkwell team for excellent design work and to the installers for putting it in like the plans.

Sound beam discovery

The design goals of the SB-2 are quite challenging: to cover the opposite side of the circle -without disturbing the near side. The intent was not COVERAGE in the traditional sense – as in – sound or NO sound, but foremost for vertical image control.  This is where scenic design adds a challenge: a shower curtain. Yes a thin plastic sheet around the room perimeter to obscure technical areas, catwalks etc. The SB-2s are behind it. Acoustically transparent, of course….. kind of.  We measured with the curtain in place – and pulled back – 6 dB loss from 2k Hz on up. Result: we have to drive the HF harder to get to the other side. Result: splash and spill on the near side is stronger than desired. Result:  Adjusted the tilt angle up to reduce the level on the near side. Result: Better but still not optimal. Decision: Riggers will move the SB-2s up 2 meters during the daytime tomorrow and we will reset the angle down and try again.

More Verification: Opposite side line subwoofers

The verification proofing technique makes use of symmetry of the room. We placed mics at opposite sides and stored the individual responses of box 1 to 5 with its mirror opposite. 1 polarity reversal found.

Still More Verification: Underbalcony speakers 8-32

Level and EQ were verified as matched to each of the remaining UB speakers. Delays were adjusted individually for each because the geometry of the Mains and Ubalcs, as constructed do not make for an absolutely concentric pair of circles. The differences overall ranged about 3ms over the 70ms of approximate range. To have a system designed to able to be tweaked to this level of detail is something you just don’t see every day………. or pretty much ANY other day.  Wow!

Macau COD Tuning: Day 5 (Cardioid sub day)

May 8, 2010

Looking for Mr. Cardioid

During the daytime (remember that we start at 11pm) the crew set up 2 measurement mics for us high in the grid. The mics were placed as a front/back pair in the near field of the coupled subwoofer array. The goal was to measure the responses in front and behind on the center axis at symmetric distances. This would allow us to see the cardioid action right there AT the array and help to clarify the mysteries of the day before.

At 11pm we started on it right away. We measured the response of each speaker in front and in back. The mics were not quite equidistant – 16ms (front mic) and 23 ms (rear mic). This translated to around 3 dB so we prorated the data with that in mind.  We 1st observed the 3 front-firing drivers, individually and in combination. We found a 1 dB of front/back ratio at 30 Hz but 6 or more by 80 hz. The rear facing drivers did the same – in reverse and we felt ready to put in the parameters we had developed in MAPP (pol reverse, 4ms, -2.5 dB) on-line and ess what would happen. We did. We measured. 1 dB of cardioid action – LESS than just having the boxes all face forward. ????????????

Now we were spooked. We had seen no evidence of cardioid steering in the far field in the room during the day before – now we had none in the near-field. What gives?  So we decided to simplify the 5-box rig to the center 3 boxes. Now it became 2 backward and 1 forward (pol rev @ 4 ms).  This would steer the OPPOSITE of our design – but MAYBE we could get a measured result that related to the predictions…..pretty please.

Before we measured the combination Vikram abd I looked at the individual parts – amplitude and phase, with all the parameters put in. In phase in front of the two boxes, 180 degrees out in front of the single box. Combined we got the 20 dB ratio we were looking forward. Perfect cardioid steering………….. into the flyspace.  Wrong direction, but it was what we expected – that was definite progress. Now we went back to 3-2 ratio and adjusted the level until it was equal in level in the back (2.5 dB down for the rear firing pair – the ORIGINAL predicted number).  Now it worked perfectly.

I could not stand to NOT know why it had NOT worked before – so we ran the human error scenarios until we found it. We had previously put the pol reverse on the correct speakers, but the delay on the wrong ones. If you ever want a REALLY REALLY OMNI sub array – let me know, I have the recipe.  So NOW it works up in the grid. Maybe our human error was the reason it had not worked in the house the previous day. Back to the far field.

We compared the mic 70 ms away directly in front of the array, with the one at the audience area MOST behind the array. It was TWICE the distance – so it should be 6 db down just by distance. Result: So close to the same level as to be insignicant. Maybe 1-2 dB. On the sides it was about 3 dB down. It was also 3 dB further. The impulse response at the rear revealed something VERY interesting. The second set of arrivals back there were WAY stronger than the 1st. The direct sound wave (earliest arrivals) had been reduced, but the steeing had concentrated the front lobe on a concave GIGANTIC wall of glass where the control booths are – pretty much the only major league reflector in the building. So at the end of the day, we have a cardioid array that reduced the first arrival, but could not stop the second. Our analyzer has a 640ms time window down there so it saw the strong 120ms reflection as integrated with the direct sound. It will be interesting to see how we would perceive the difference between front and back – same level, but much higher direct/reverb ratio in the front.  Today the crew will angle the coupled sub array further downward, to reduce the direct sound on the window wall. This will put the cancel lobe more into the flyspace (which can only be a good thing) rather than trying cancel toward the rear of the house.  We will test that tomorrow.

The next task was merging the coupled directional configuration with the uncoupled steered subwoofer line array. The levels and delay were set to merge them in the corner (yes I know circles don’t have corners) – the corner was where two parallel lines of 5 subs meet the coupled array. Altogether the subs make the shape of an “n”, with the uncoupled lines on the sides and coupled array at the top.

Verification

The latter part of the session was spent in verification mode: checking speakers to make sure they were like the other speakers of the same type. We had already completed the first 1/3 of the circle, now we had to do the remaining 2/3rds.  This amounted to: 14 CQ-2’s, 30 UMP-1Ps, and 24 UPJ-1Ps. Each was checked for frequncy response, delay, level, polarity and level.  This took about 3-4 hours. We found about 4 poarity reversals, one speaker with a mysterious extra 4 dB of level, a CQ with a lighting instrument partially blocking it, and lots of well matched speakers. This allowed us to turn the system over to Steve and Pierreto try out the Constellation settings they had programmed in.

Tomorrow we have a few mop-up operations: verification of the opposite side main clusters and subwoofers, and 25 little MM4-XP under balcony delays.  We will fit these in during times when Constellation tuning does not require the system to be in use. We have been trading time between the teams, prioritizing the SIM tuning to get the Constellation team the data it needed to keep moving forward while we SIM’d other subsystems.