Measuring Sound: Part 2

In an interview with System Sound’s Julian Spink, we explore PA measurement techniques, test mics, the problems with line array and A/B PA systems.

30 July 2007

Text: Hugh Covill

Last issue I wrote a short article that attempted to peel away some of the mystery surrounding measuring sound. To follow up some of those themes I thought it might be good to talk with someone who’s using these measurement applications routinely, in order to better understand the relationship between a sound system and its acoustic space.

Julian Spink has engineered live sound for a host of local and international artists including Bachelor Girl, John Farnham, Christopher Cross, Warren Zevon, and has toured extensively with many bands including Split Enz, Crowded House, Little River Band, Icehouse, and The Divinyls.

In 1993 he joined System Sound as Head of Sound on the musicals Les Miserables, Sunset Boulevard, and West Side Story. His sound system design credits include Boy from Oz, the Millennium New Year’s Eve Spectacular (Melbourne), the 2000 Olympics Arts Festival – Mahler’s 8th Symphony and the Australian productions of Man of la Mancha, Footloose, Hair, Leader of the Pack, Chicago, and Cabaret.

I recently caught up with Julian at The Melbourne Arts Centre’s State Theatre, where he was hard at work doing rehearsals for ‘The Production Company’ musical Thoroughly Modern Millie. With 30-odd channels of radio microphones and a fully miked on-stage orchestra, all managed by an impressive BSS Soundweb rack which feeds a A/B Meyer PA system… we could have spoken at some length just about the gear rider. But I was keen to get Julian to share his thoughts on the use of measurement tools and hoped our chat might prove a little more engaging than a session of trophy gear envy. I wasn’t disappointed. I’ve always been pretty interested in the path people take to arrive at a particular job. So that’s where we kicked off.

EXPLANATION OF A/B PA SYSTEMS

In a musical theatre scenario, you’ll often have the situation where the lead characters need to sing together in close proximity. When their vocals are captured at approximately the same level in each other’s microphones and then combined electrically in a common mix bus it can produce a phasing effect. Otherwise termed ‘comb-filter distortion’, it’s the result of a single source mixing with itself but delayed a slight amount (that’s the path difference between the two microphones). The severity of the effect varies with the distance between the microphones along with number of microphones. This effect is most audible when two or more singers are physically close together during a song where the vocal is the driving element.

There are several sound system design solutions to this problem. By far the best (but most expensive) solution is to employ what is termed an ‘A/B’ system. Simply put, it involves utilising two completely discrete signal paths. Each vocal will have its own mix bus, amp channel and speaker. Much of the design requirements double and you therefore only typically see this technique being used for big budget music theatre. It is, however, the only solution that completely removes the problem.

Yet another solution is an automatic matrix mixer that supports variable delay at the matrix cross-point. This solution is based around adjusting the amount of delay applied to each channel in real time relative to where each performer might be on stage.

And here is a ‘roll your own’ solution that I was once taught: Y-split your mixer stereo outputs to become a single mono out, then insert a five millisecond delay onto one of the outputs (say left-out), five milliseconds is akin to moving the mics a metre or so away from each other. Now if you hear any phasing you can perform a correction using your pan pot. As you pan from right to left the pan pot’s facility to provide an equal energy crossfade means there will be no change in level just the ability to provide each mic with a smooth real-time remedy for the problem. It’s not as perfect a solution as the A/B, but it’s cheap!

THE PROBLEM

The singers with mics positioned close to each other. Their mics are combined using one or more of a mixer’s output buses. The mix is then sent to an amp and ultimately to a speaker. The mix bus will be combining the sound from Mic 1 with that same sound as captured by Mic 2 at the same level but delayed slightly, creating a distinct phasing sound. Not exactly sure what you’re listening for? Try this: stack two nearfield loudspeakers one on top of the other. Send them a mono mix of some music you know well, and listen. Now move (or have moved) the top speaker slowly away from you. What you’re hearing is comb filter distortion.

THE A/B SOLUTION

An A/B speaker system provides a totally discrete path from each mic to a dedicated speaker. This includes a discrete mix bus which then feeds a dedicated amp channel and speaker or array of loudspeakers. This method completely removes the problem of comb filter distortion, however, in a large multi-channel PA system it greatly increases system complexity and the requirement for sophisticated automatic matrix mixers.

PURE THEATRE

Hugh Covill: System Sound’s particular forte is theatrical sound. When did you make that transition and how did you enjoy that compared to working with bands?

Julian Spink: System Sound offered me a gig mixing Five Guys Named Moe, which was a little short-run three-month production at The Athenaeum. It was really interesting and I was fascinated by what went on: automated consoles, multiple speaker systems, all delayed back to each other. There were a lot of technical things going on that I felt I could really get my teeth into, and eventually I was offered a permanent job. Managing lots of speakers is my main interest now: dealing with delay times, locating sound to a specific place, the Hass Effect [panning sound via the use of delay], exploring psychoacoustics, that sort of thing.

HC: Which segues nicely into our discussion on measuring sound. How do you approach the task of sound system measurement, what’s the goal?

JS: Essentially the goal is to make the system sound flat – certainly for musical theatre and orchestral purposes this is definitely the case. But I tend to approach it from the angle that there’s no right or wrong way of getting to that point – you have to make your mind up based on what you’re dealing with.

Theatre productions use a lot of A/B systems [see box for more]. With a system like this, decisions about what you’ll tune first – whether you’re going to EQ it, delay it, and level match it – are inter-dependant questions to a certain extent. You have to make your mind up about what you’re going to start off with based on the system design. And each one is different; the number of speakers on a truss in the centre, the number of speakers on each side, whether or not you’ve got a separate band system, and so on. I remember in the old days I used to tune PAs by simply singing ‘yeah, yeah, yeah’ into an SM58 and just cranking up and down equaliser bands to try and find which ones appeared to be hotter than others – quite a bit different to what we do now!

HC: Indeed! When we met last night you were seated in front of a Meyer SIM2 measurement system. You’re proficient in a number of different measurement platforms. Can you share some of your thoughts on their differences?

JS: Sure. My experience with measurement platforms began around the same time I started working for System Sound, when I built my own measurement system from a kit in an electronics magazine – IMP Impulse Measurement Program (or something like that). It was pretty much the same time as I was seeing John Scandrett [from System Sound] using MLSSA, which, in those days, was state-of-the-art. System Sound still has two MLSSA systems; it’s pretty old hat nowadays but I was using that up until quite recently. It’s a particularly good lab instrument but a bit fiddly to set up in the field. You have to manually set a time window for an impulse to the actual size that you want to FFT [Fast Fourier Transform – see last issue for more on FFT] and then you perform an FFT on it. There isn’t great resolution below about 250Hz but it’s great for finding delay times. I think it’s a better tool than, say, Smaart in that respect. Smaart’s really good from the point of view that you don’t have to go out and buy a really expensive piece of hardware – and it’s quite an amazing tool. Its shortcoming is that it’s not great with the delay time finding function. You can’t readily see the impulse trace where there are reflections off wall surfaces – sometimes that’s really handy to see in great detail – and I’ve never really seen that with Smaart. I feel like you’re flying blind a bit compared to ‘Melissa’ [MLSSA] or, say a SIM System.

I remember in the old days I used to tune PAs by simply singing ‘yeah, yeah, yeah’ into an SM58 – quite a bit different to what we do now!

ON LINE ARRAYS

HC: I noted that all the speakers you’re utilising in this current design are constant Q boxes. Can you comment generally with regards to the ubiquitous use of line arrays?

JS: I feel like we’ve been spending the last number of years being brow beaten over how great line array speaker systems are. Yet in terms of controlling sound, as a designer and operator you probably have less input with line array systems and less control. There’s certainly less pattern control in the horizontal plane, which is probably 50% of the whole equation really. In theatre, where we don’t want to splash the walls with sound we really need to consider elements that have a better pattern control in a horizontal plane than what line arrays offer. Unfortunately, everybody’s been sold the idea – producers, engineers, rental companies – that line array is the magic bullet.

HC: I think a lot of folks are starting to question why we’ve moved from a philosophy of constant Q boxes (which was about putting sound on the people very specifically) to a line source model which, as you point out, has a very wide horizontal dispersion pattern. That’s not to say line arrays aren’t a fabulous tool for the right application, of course. As a centre cluster they make a lot of sense. That said, the number of musicals that pass through town with large line arrays left and right, hard against the box booms continues to dismay me. These sound systems are essentially splashing an enormous amount of energy onto the sidewalls…

JS: And at the back wall as well, and that’s one thing that I have issues with. Depending on where you’re seated within the room, it can be a situation whereby you’re almost getting equal or more energy off the back wall than you are into your ears directly from the line array. All you need to do is shift your head one way or the other and it’s like there’s a party going on behind you. So the whole directionality of the system, i.e., where the sound is coming from, is thrown into question. Musical theatre is all about providing a natural sound experience, and to that end having control of the location of sound is really what it’s all about. With big fat line arrays, it’s like using a fire hose to fill your glass of water!

HC: Can you share your thoughts and experiences using Meyer Sound’s SIM system?

JS: Well as far as I’m aware, SIM 1 was never available for anybody other than the Meyer folk, and SIM 2 was an exceptionally expensive system. SIM 3’s much more realistically priced, and much easier to use. The advantage of SIM is the whole preamplifier arrangement. Everything’s calibrated throughout, so the readings you make on one SIM system compared to a hired SIM system in another country will be identical. Meyer uses SIM2 as a tool for calibrating all the speakers and electronics. It’s a common tool. It’s also part and parcel of Meyer’s MAPP online, which I’ve used an incredible amount for placing speakers in a venue and just having a look at what happens with arrays; it’s a fantastic program. It’s not as well developed or detailed as some other architectural sound install programs, but it’s really accurate for simulating positioning Meyer speakers in a space.

HC: System Sound guards a number of matched B&K reference microphones that you use when you’re measuring systems. How important do you think microphones are to the measurement process? Can you get away with a less expensive mic?

JS: Well personally I think you can get away with a less expensive one. My first measurement microphone was a $9.95 Panasonic. With SIM we use B&K 4006s mostly, and we have a 4007, which has a lower sensitivity but is also calibrated for SIM. Sometimes, instead of using an omnidirectional microphone, it might be pertinent to use a cardioid condenser if the space you’re measuring is really awful and reverberant. We use Sennhesier MKH40s, Schoeps MK4s or 21s sometimes and the differences between the mics appear to be mainly the phase response in the top end.

We’ve had a look at a couple of different microphones with a reference source just the other week. There are two basic aspects to note: one is finding out what’s ‘flat’ and what’s not and the other thing is making relative measurements. I tend to work more in the relative scope of things. That is, I tend to say: ‘okay, if this sounds like that, how will that affect this other sound?’ Some people get very anal about measurements and delay times, whereas I think once you start putting program through a system, there’s nothing to say you can’t play with it and change it: change the delay times, slow things down, speed things up, see what changing these ingredients does. You might find that things actually work better.

HC: So you can measure until the cows come home, but at some point you’ve got to decide…

JS: … whether the process that you’ve gone through has made it sound good or not!

HC: There are a lot of different philosophies with microphones and placement. There’s quite a movement in The States where people are laying out big panels over the seating and lying the microphone on the panel and measuring that way. Some people like to get their microphone up very high in the air away from any other boundaries. Do you have a preference about where you like to place your microphone when you’re measuring?

JS: One factor is how far away the microphone is from a hard surface. I tend to place a mic at around a standing ear level. I’m not as close to an acoustic seat material and hopefully I’m far enough away from whatever reflection the floor surface is likely to produce that I can see it in the measurement.

To tune a system with multiple speaker systems, I think there’s a number of mic positions you have to select, and you also have to accept that where you put a microphone sometimes might not work. You’ve got to be prepared to reject the data if you think what you’re seeing is rubbish. And it might be that you’re out by an inch or two, which can make a difference to what you’re seeing.

HC: What are you looking at when measuring multiple speaker systems?

JS: What I tend to look at is what kind of energy each speaker is going to be delivering into each area, and finding an equal energy position between the horn patterns. I’ll often find an equal energy position, set the delay time between the boxes, and then tune. I might even shift microphone positions for tuning, so you might be on-axis and not necessarily between the axes of the two horn patterns. Then ultimately you might find that one of the speakers is too hot so you might turn it down and that will affect the delay time. In that circumstance you’ve got to go back and find the equal energy spot again. Then I’ll re-check the delay time and make sure it’s still in order, so it’s not always a refined process.

HC: Finally, I’d like to ask whether measurement has influenced your practice more generally?

JS: The whole process of tuning a PA does feed back into the design again the next time around. You discover things that didn’t work, you discover things that did. The problem with it is that sometimes the next show doesn’t require the same approach at all so you’ve got to learn what’s appropriate and what’s not. You have to constantly ask yourself: ‘is this just a trip I’m on? Is it really warranted to employ this kind of method now?’ So, as everybody in sound knows, as soon as you think you know everything, it’s probably time to become a bus driver!

Measuring Sound: Part 1

Most of us couldn’t give an FFT about measuring sound, but how many of us know how it actually works?

Read Part 1 Here

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