Tag Archives: SPL

Transient Impact

Music that hits hard requires careful management of the parts that don’t hit hard.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

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A few weeks ago, I had the unexpected pleasure of working with a band called “Outside Infinity.” I say that the pleasure was unexpected because I had some major concerns going into the show. Metal, as a genre, can be pretty challenging in a small space. The sheer volume can be tough (or even impossible) to work with, and the arrangements are often quite dense – which compounds the volume problem. Several instruments banging away at full-blast can make for lots of challenges when trying to differentiate each part of a mix.

Outside Infinity had none of those problems. In fact, they were a prime example of how heavy metal – or any type of music that you want to “hit hard” – actually achieves that goal. (They were so much fun to listen to that I’m pretty sure I had a stupid grin on my face for large portions of the night.) I was really impressed by the sound that they had crafted, and I started to think about it.

Why were they so much fun?

Why did they capture what I’ve loved about heavy metal in the past?

Why did their sound have what so many rock and metal bands want, but so often fail to achieve?

I think that the generalized answer to all of those questions is this: Transient impact.

The Stopping Is As Important As The Starting

There are a number of necessary elements to a really great song performed live in a really great way. The lyrics have to be interesting, of course, and a memorable melody (or overall musical theme) is required. Skipping those steps will efficiently torpedo a tune’s ability to grab and hold an audience. There’s more, though: The overall sound of the song has to keep the listener interested. It’s analogous to eating a meal that leaves you remembering the food for years. Every bite is delicious, yes, but certain bites contain an extra explosion of flavor that plays on the mouth and tongue…and then dissipates. That “taste transient” pokes out from the “steady state deliciousness” of the rest of the meal, creating an ebb and flow of special delight, anticipation, and reward.

But if that burst of flavor just continued unabated, with no steady-state to contrast it against, then the “burst” wouldn’t be attention-getting anymore. It would BE the steady-state, and would quickly become unremarkable.

Sound behaves in a way that’s fundamentally the same. We perceive it differently, and the time-scales involved are sometimes much shorter, but the transient content is still the basis of what holds attention. Transient content is the determining factor behind the (ironically) nebulous idea of music that’s “really defined.” In music that aims to convey power and force, sounds that hit above the steady-state, and then swiftly decay are what cause the individual parts to “slam into you.” Everything just banging away at full throttle, continuously, for several minutes, has no impact. No spark of flavor. The brain starts to have trouble distinguishing the music from noise, because of the lack of anything to lock on to.

The mastery of stopping notes at the right time is what creates epic riffs. The mastery of creating a pleasing steady-state, which is then punctuated by sharp, sonic flavors, is the essence of the “thunderous” rock show.

…and because transients are all about proportionality, it is entirely possible to create a pile-driving artillery barrage of a show within the confines of a small venue. More on that later. First:

Dynamics And Articulation

Music, especially rock and metal, has a long history of breaking rules and pushing boundaries. This is what drives innovation, and it’s a good thing. However, there are certain rules that can’t truly be broken successfully. Those rules are the ones that are based in fundamentals of the physical universe and human perception.

One such rule is that, for a particular musical part to seem “big,” the other parts around it must be proportionally small. There are different ways of achieving this, but it all pretty much boils down to volume. The “small” part must either be quieter across the entire audible spectrum, or quieter across the most important part of the spectrum occupied by the “big” part. Especially in the small-venue context, plenty of bands shoot themselves in the foot with this. I’ve heard too many groups that interpret the instrumental breaks of their songs as “there’s no vocal, so now all the instruments should play as loudly as they can, occupy every frequency possible, and we’ll just hope that the audio-human can crank the actual solo above all that.”

(The best bands avoid this problem by interpreting the solo instrument as being “the new vocal,” and thus they keep all the other instruments in a supporting role until it’s their turn to be in front.)

Anyway.

In music, there are lots of broad-brush ways to accomplish this necessary contrast. There are the overall dynamics of individual parts across a number of beats, and there are also the rests – where a part is silent for a time. Whether formally or informally, these contrasts can be reliably notated. It’s pretty easy to explicitly define the necessary negative space, whether by a symbol for a rest, a “pp” for being very quiet, or a scribbled note saying, “For the fingerpicked guitar part, no drums at all and everybody else turns way down.”

There’s something else, though, that’s required for mastery. It’s hard to explicitly notate. It’s articulation.

Articulation (as I see it) is the manner in which notes and chords are played. It’s a crucial part of getting transients to contrast with the rest of the music, because it involves dynamics and rests that are too short and frequent to write down…and yet have a massive effect on how other parts sound. Playing a power chord with a “micro rest” at the end can be key to getting a kick-hit to punch through. Making that kick-hit decay into silence quickly can make room for a note from the bass. Going through a run of notes where each tone is connected, but there’s a very slight volume drop just before the next sound, can make for a clean and precise solo line. The singer hitting a big note and then backing off means that they can help support that solo line without a miniature volume war erupting.

The very best bands have a reliable handle on making this all work – even if they’re not explicitly aware of what they’re doing. Their riffs are powerful and defined because the individual notes have space around them. Their drum hits are forceful and satisfying because there’s space for them to stick up above everything else – and yet the drums don’t overpower the tonal instruments, because the individual hits decay into the “steady state volume” before the tonals hit THEIR next transient.

This leads me into that promised bit about how this is possible in small venues.

The SPL Difference Is The Key, Not The Absolute SPL Magnitude

A common mistake in trying to reproduce big-show impact in a small room is trying to replicate the big-show’s absolute SPL (Sound Pressure Level). It’s very easy to think that “so and so sounded huge, and they were making about 115 dBC in the center of the crowd, so that’s what we should do.” What tends to happen, though, is that reaching that kind of level chews up all the power available in a small-venue audio rig. The result is a show that doesn’t have those oh-so-cool transient hits, because there’s just no room for them to assert themselves.

Instead of defeating yourself with excessive volume, what you have to think about is WHY the big-show PA was making 115 dBC in the center of that huge crowd. It’s proportionality. Several thousand humans having a big party can make a surprising amount of noise – and so, to be clearly audible, the audio rig has to make even more noise. If a giant crowd is hollering at 105 dBC, then the audio-human running the system up to 115 dBC is understandable…if maybe a bit excessive. (Or not. It depends.)

From that previous paragraph, you can see that the proportionality between the steady-state volume of the crowd and the steady-state volume of the band was 10 dB. In certain kinds of small venues, that might be a little bit too much. A window of +6 to +9 dB of continuous level above the crowd is worth trying for in most contexts – in my opinion. Note that the “trying” part is most likely going to be in the downward direction. Getting loud is surprisingly easy, but holding your level in check to a point where the crowd is still pretty-darned audible is HARD. It’s hard for bands, and hard for audio-humans, but it’s worth trying for.

The point of holding your continuous level down, beyond just being nice to your crowd, is that it creates space for your show’s transients. Especially if you’re a metal band, and you want that big, thunderous kick, your best chance is to be had by giving the PA lots of room. If the audio-human has to run the system at full tilt just to keep up, then there probably won’t be enough power available to put those chest-thumping transients where you want them to go. On the other hand, keeping the show’s continuous level at a manageable point means that there’s reserve power – reserve power that has to be available to create large, proportional differences for things that need accenting.

Running the audio for Outside Infinity was fun, because they had an instinctive handle on negative space and transient impact. There was plenty of power available for the musical peaks, because the continuous level of the band was appropriate and comfortable. They knew how to articulate their notes so that the music was sharp and defined. I was really impressed.

And, if you take the time to think about your music’s transients, you’ll probably also have a good shot at being that impressive.


Dirty Secrets About Power

The amount of power actually being delivered to your loudspeakers might not be what you think. What power IS getting delivered might not be doing what you think.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

I’m pretty sure that power – that is, energy delivered to loudspeaker drivers – is one of the most misunderstood topics in live-audio. It’s an area of the art that’s often presented in a simplified way for the sake of convenience. Convenience is hardly a bad thing, but simplifying a complex and mission-critical set of concepts can be troublesome. For one, misinformation (or just misinterpretation) starts to be viewed as fact. Going hand-in-hand with that is the phenomenon of folks who mean well, but make bad decisions. These bad decisions lead to the death of loudspeakers, over and under spending on amps and speakers, seemingly reckless system operation…the list goes on.

So, with all the potential problems that can be caused by the oversimplification of the topic “Powering Loudspeakers,” why does “reduction for the sake of convenience” continue to occur?

I think the answer to that is ironically simple: The proper powering of loudspeakers is, in truth, maddeningly complex. There are lots of “microfactors” involved that are quite simple, but when they all get stuck together…things get hairy. At some point, educators with limited time, equipment manufacturers with limited space in instruction manuals, and established pros with limited patience have to decide on what to gloss over. (I’ve done it myself. Certain parts of my article on clipping let some intricacies go without complete explanation.)

With that being the case, this article can’t possibly cover every little counter-intuitive detail. What it can do, however, is give you some idea of how many more particulars are actually out there, while also giving you some insight into a few of those particulars.

So, in no particular order…

Dirty Secret #1: Amp And Speaker Manufacturers Assume A Lot

You may have heard the phrase “Assume Nothing.” That saying does NOT apply to the people who build mass-produced loudspeakers and amplifiers. It doesn’t apply because it CAN NOT apply – otherwise, they’d never get anything built, or their instruction manuals would be gigantic.

Amplifier manufacturers, on their part, assume that you’re going to use their product with mostly “musical” signals. They also assume that you can put together a sane system with the “how to make this thing work” information they provide in their documentation. Further, they make suggestions about using amplifiers with continuous power ratings that are greater than the continuous power ratings of your speakers, because they assume that you’re not going to drive the amp up to its clip lights all the time.

Loudspeaker manufacturers also assume that you’re going to drive their boxes with music. They also ship products with the assumption that you’ll use the speaker in accordance with the instructions. They publish power ratings that are contingent on you being sane, especially with your system equalizers.

The upshot of it all is that the folks who make your gear also make VERY powerful assumptions about your ability to use their products within the design limits. They do this (and disclaim a lot of responsibility), because a ton of factors related to actual system use have traditionally been outside their control. Anytime you read an instruction manual – especially the specifications page – take care to remember that the numbers you see are simplifications and averages that reflect a mountain of assumptions.

Dirty Secret #2: Musical Signals Don’t Get You Your Continuous Power Rating

The reason that technical folks distinguish between signals like sine waves, pink noise, and “music” is because they have very different power densities. Sine waves, for instance, have a continuous level that’s 3 dB below their peak level. Pink noise often has to have an accompanying specification of “crest factor” (the ratio between the peak and average level), because different noise generators can give you different results. Some pink noise generators give you a signal with 6 dB between the peak and average levels. Others might give you 12 dB.

Music is all over the map.

Some music signals have peaks that are 20+ dB above the average power. Of course, in our current age of “compress and limit everything,” it’s common to see ratios that are much smaller. I myself use rather aggressive limiting, because I need to keep a pretty tight rein on how loud the PA system can go. Even so, my peak levels tend to be about 10 dB above the average level.

So if you’ve got an amp that’s rated for “x” continuous watts, and you drive the unit all the way to its undistorted peak, music is probably giving you x/10 watts…or less. In my case, the brickwall limit that I set is usually 10 dB below clip, which means that my actual continuous power is something like 5 watts per channel. This calculation is pretty consistent with what I think the speakers are actually doing, because they get about 96 dB @ 1 watt @ 1 meter. Five watts continuous would mean about 103 dB SPL per full-range box, and there are two full-range boxes in the PA, so that’s 106 dB total…yup, that seems about right.

Yeah, so, your system? If you’re driving it with actual music that isn’t insanely limited, you can go ahead and divide your amp’s continuous power rating by about 10. Don’t get overconfident, though, because you can still wreck your drivers. It’s all because…

Dirty Secret #3: Power Isn’t Always Evenly Distributed

Remember that bit up there about manufacturers making assumptions? Think about this sentence: “They publish power ratings that are contingent on you being sane, especially with your system equalizers.”

Dirty secret #2 may have you feeling pretty safe. In fact, you may be thinking that secret #2 directly contravenes some of the things that I said about cooking your loudspeakers with an amp that’s too big.

Hold up there, chum!

When a loudspeaker builder says that the system will handle, say, 500 watts, what they actually mean is: “This system will survive 500 watts of continuous input, as long as the input is distributed with roughly equal power per octave.” Not everything in the box will take 500 watts without dying. In particular, the HF driver may be rated for a tenth – or less – of what the total system is advertised to do. Now, if you combine that with a system operator who just loves to emphasize high-frequency material (“I love that top-end snap and sizzle, dude!”), you may just be delivering a LOT of juice to a rather fragile component…

…especially if the operator uses a huge amp, because they’re under the false impression that amp headroom = safety. A 1000 watt amplifier, combined with a tech who drives hard, scoops the mids, and has boxes with passive crossovers, is plenty capable of beating a 50-watt-rated HF driver into the ground.

On the flipside, a system without protective filtering on the low-frequency side can get killed in a similar way. Some audio-humans just HAVE to “gun”the low-frequency bands on their system EQ, because “boom and thump are what get the girls dancing, dude!” Well, that’s all fine and good, but most live-sound speakers that are reasonably affordable can’t handle deep bass at high power. Heck, the box that the drivers are in often acts as a filter for material that’s below about 40 Hz.

Of course, there may not be an electronic filter to keep 40 Hz and below out of the amplifier, or out of the LF driver. Thus, our system operator might just be dumping a huge amount of energy into a woofer without actually being able to hear it. The power doesn’t just disappear, of course, which means that “driver failure because of too much power at too low a frequency” might be just around the corner.

Dirty Secret #4: Accidents Aren’t Usually Musical Signals

Building on what I’ve said above, I should be clear that folks do get away with using overpowered amps (for a time) because of feeding them “music.” They end up keeping the peaks at a reasonable level, and so the continuous power stays in a safe place as well.

Then, something goes wrong.

Maybe some feedback gets really out of control. Maybe somebody drops a microphone. All of a sudden, you might have a high-frequency sine-wave with peaks – and continuous level – that’s far beyond what a horn driver can live with. In the blink of an eye, you might have a low-frequency peak that can rip a subwoofer cone.

Ouch.

Dirty Secret #5: Squeezing Every Drop Of Performance From Something Is For Either Amateurs Or Rich People

This secret connects pretty directly with #3 and #4. Lots of folks worry about getting every single dollar’s worth of output from a live-audio rig. It’s very understandable, and also very unhealthy. To extract every possible ounce of output from a loudspeaker system requires powerful, expensive amplifiers that have the capability to flat-out murder the speakers. For this reason, “performance enthusiasts” are either people who can’t afford to buy both more power AND more speakers, or they’re people who can afford to buy (and fix, and fix, and fix again) a lot of gear that’s run very hard.

The moral of the story is that your expectation needs to be that – in line with secret #3 – getting continuous output consistent with about 1/10th of a rig’s rated power is actually getting your money’s worth. If you don’t have enough acoustical output at that level, then you either need to upgrade to a system that gets louder with the same number of boxes, or you need to buy more loudspeakers and more amps to expand your system.

Dirty Secret #6: More Power Means More Than Just Buying More Amps

This follows along with secret #5. If you want more power, then you need more gear. That seems simple enough, but I’m convinced that linear PA growth is accompanied by geometric “support” growth.

What I mean by this is that getting ahold of a more powerful PA is more than just getting the amps and speakers together. More power means heavier and more expensive amp racks, or more (and more expensive because of quantity) amp racks. It may mean that you have to construct patch panels to keep everything organized. More PA power also means that you need more AC power “from the wall” in the venue. Past a certain point, you have to start thinking about an actual power distro system – and that can be a major project with huge pitfalls in and of itself. You need more space for storage. You need a bigger vehicle, if you’re going to transport it all.

Getting more power doesn’t just mean more of the “core” gear that creates and uses that power. It means more of everything that’s connected to that gear.

Dirty Secret #7: The Point Of Diminishing Returns Occurs Very Quickly. Immediately, In Fact.

The last secret is also, in some ways, the biggest bummer. Audio is a logarithmic affair, which means that the gains you get from spending more money and providing more power to a system begin decreasing as soon as you even get started. I’m dead serious.

For example, let’s say you’ve got a loudspeaker that averages about 95 dB SPL @ 1 watt @ 1 meter. You put one continuous watt – one measly watt – across the box, and stand roughly three feet away. That 95 dB SPL seems pretty good. Now, you go up to two watts. Did you get 95 dB more? Nope – that would mean that you could get “space shuttle takeoff” levels out of one loudspeaker. Not gonna happen.

So…did you get 20 dB more?

No.

10 dB?

Nope.

You doubled the power, and got three decibels more level out of the speaker. That’s just enough of a difference to definitively notice that things have gotten louder. If you want three more dB, you’ll have to double the power again. So far we’re only at four watts, but I think you can see just how fast the battle for more output starts to go against you. If your system is running at full tilt, and you want more output, you’re going to have to find a way to “double” the system – and even when you do, you’ll only get a little more out of it. If you want to get 10 times as loud, you need 10 times as much total PA.

The vast majority of a PA system’s output comes from the first watt going into each box. It’s a fact that’s in plain sight, but it (and its ramifications) often aren’t talked about very much.

That makes it one of the dirtiest secrets of all.


The Peavey PVXp-12

As usual, Peavey delivers a competent product with only a few downsides.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

I’m sure that Peavey encompasses many adjectives that start with “P,” like “proficient” and “pugilistic.” (They’re feisty.) My favorite Peavey adjective, though, is “predictable.”

Now, don’t get me wrong! I love innovation and “cool new stuff,” but I also love being able to get cool new stuff that I know is made well. That’s where Peavey delivers: They make affordable gear that delivers usable performance and holds up under the rigors of live-audio. You know what to expect when you order a box with the Peavey badge, and that is tremendously valuable for live-sound humans.

When it comes to speaker enclosures, the big “P” has never let me down. Even when a box has suffered some sort of problem, the issue was either too subtle for most people to notice, or correctable with a few minutes of work. Almost every Peavey loudspeaker that I’ve ever owned is either still in service somewhere, or was traded up for the next box. I had some cheap subs that I overpowered (because I was young and dumb), and they endured the punishment that I was dishing out for gig after gig after gig. The voice coils did get pushed a bit out of true, but the drivers never entirely quit – in fact, the only component to actually fail was the crossover on one of the boxes. A quick bypass operation later, and I had a working sub again.

It’s fitting, then, that my monitor-wedge woes would be brought to an end by a bevy of Peavey units. After some disappointing misadventures with offerings from Avid/ M-Audio and Seismic Audio, a sextet of PVXp-12s has put the smile back on my face.

I Don’t Have Lots Of Numbers, Because I Don’t Need Them

When I did my review of the monitor wedges I procured from Seismic Audio, there was a fair bit of testing involved. Numbers…you know, quantitative analysis.

I haven’t done anything like that for the PVXp-12s. They might be able to do what Peavey claims they can do, or they might not.

But I don’t care.

Why?

Because, whatever the PVXp boxes do, they do enough of it to satisfy my needs as a small-venue audio human. What’s more, they do what they do in a seemingly effortless way.

You might not think that says much, but it actually says a lot – and loudly. I measure when a piece of gear is giving me a reason to be skeptical. If I have no reason to “pick at” a manufacturer’s claims, then I don’t. Peavey claims that PVXp-12s can produce a peak of 127 dB SPL with music. Of course, every time a manufacturer says “peak,” you can subtract 3 – 6 dB to get an idea of what the box will actually do in real life. My guess is that a strong vocal input through these units has a fighting chance of doing 120 dB SPL continuous at a listener’s position. That guess is backed up by the fact that, over a good number of shows, I have never been able to observe the DDT™ (Peavey’s proprietary limiting system) indication on the units that I have. In contrast, other monitor wedges that I’ve had in service would either light their limiting indicators regularly, or be in audible distortion.

The bottom line is that I don’t have to nitpick the PVXp-12s. I don’t care if they can actually reach the claimed 325/ 75 watts continuous into the LF (Low Frequency) and HF drivers, because whatever wattage is actually being dissipated is plenty. I commonly “double up” two units, which gives a theoretical “maximum continuous vocal output” of 123 dB SPL.

Quite frankly, if you need more than that on stage, your show doesn’t belong in a venue that seats 200 people or fewer. Either that, or somebody is playing WAY too loud and needs to be fired.

I’ll also mention that, at one show, the lead singer asked for a pretty good amount of kick in the wedges. A box loaded with a 12″ LF driver can’t be asked to deliver crushing “boom,” but for that show (which was of about average overall volume), the PVXps delivered enough thump that I didn’t need any kick in the FOH (Front Of House) PA. Not bad for a box that retails at $350 – at least, in my opinion.

As far as sound-quality goes, I don’t really know what to say. PVXp-12s “sound like music to me,” which is to say that they seemed to be tuned in a pretty sane fashion. No, you’re probably not going to have a spiritual experience when you listen to these boxes, but that’s not what they’re for. The primary purpose of a sound-reinforcement box is to deliver sufficient output, cleanly, with a smooth response across the critical frequencies for music (about 100 Hz to 12 kHz, or a little more depending on the application). That’s what these Peavey’s seem to do.

If your experience is similar to mine, you may actually need to apply a 3 to 6 dB, 1 – 2 octave wide boost at around 2 kHz, along with a less pronounced, 1-ish octave wide boost at 8 kHz to make the boxes “flat.” It all depends on what you want, though.

Again, there just isn’t much to say. As monitor wedges, my PVXp enclosures pass signals and don’t make me struggle. That’s all I want, and judging by the number of compliments I get regarding the sound on deck, that’s all that most bands seem to be looking for. I know there are better sounding boxes out there because there is ALWAYS a better sounding box out there, but everything beyond the basics is gravy…and gravy is pretty expensive.

The Quibbles

Another piece of Peavey’s predictability – at least for me – is that they always seem to make some kind of design decision that causes me to scratch my head. It’s a different thing for every product line, but I swear, it isn’t Peavey unless I want to write a letter to them that reads: “In regards to this design aspect of this product…REALLY?”

The PVXp-12 is no exception in this regard.

To start with, the XLR input on the boxes is connected to circuitry with much higher gain than the TRS input. On one hand, this makes some sense. It allows people to plug a microphone directly into the box and get results without having to hit a mic/ line switch. On the other hand, not having a switch to select mic/ line gain means that using the XLR jack for line-level input requires that the input potentiometer be set quite low, in its “finicky” range. Even there, I have to trim my monitor send masters down about 6 dB to keep my on-channel sends in an operational area that’s consistent with other things.

Now, this isn’t a huge deal. It’s certainly a “first world problem,” which can be corrected with just a bit of doing. I can acknowledge that. Still, I’m a little surprised at Peavey apparently thinking that a robust, multipin connector shouldn’t be the first choice for line-level AND mic-level audio.

There’s also the issue of how the input plate is located. For some cables, you may find that a monitor placement causes a certain amount of shearing (sideways) force on your cable’s strain relief. This may or may not be enough to cause a problem – it’ll depend on your usage patterns, though.

Another oddity is that the Peavey design department apparently lives in a world where only one side of a box needs to be angled for monitor usage. This means that, whether you want it or not, a PVXp-12 doing monitor duty will have the HF horn on the stage-right side. If you want to “bookmatch” a pair of these boxes when doubling them up, you’re out of luck. It’s hardly a critical issue, but I swear, even manufacturers who build questionable boxes have figured out how to let you lay the enclosure on either side.

Going back to the level potentiometer, I’ve found myself wishing that it would be easier to get a “repeatable” setting for the knob. If you’re using the XLR input for line-level signals, it’s impossible to accurately see where the knob is if the box is in a monitor placement. In fact, to accurately set the knob, the box has to be rotated onto its face. Further (and this isn’t just a Peavy thing), the knob is of a “continuous sweep” variety. I just don’t understand why – on a piece of gear that is probably going to be used in multiples – level controls aren’t given clickstops for easy and accurate repeatability.

All of this is just nitpicking, though. Sure, you can spend more on a speaker enclosure. Sure, there are other boxes which may be more or less “your taste.” Still, my opinion is that the PVXp-12 is a great example of how far we’ve come in terms of affordable gear. Think about it: These boxes are biamped, with all kinds of nifty processing that’s been set at the factory, and it’s all been stuffed into a pretty compact package. I got started in pro-audio during the ’90s, and the functionality in a PVXp-12 wasn’t even something we were dreaming about then.

Maybe it’s just me, but there seems to be a lot of “bang” in these Peaveys for the bucks you’ll pay for them. The boxes aren’t flashy, and there’s no hype surrounding them…

…and there’s no need for any of that, because these units just go to work, get to work, and consistently deliver.

Well, they do for me, anyway.


Inconsistent Distance

In small rooms, audience proximity to loudspeakers can mean a wildly different mix.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

I love doing “advanced application” stuff where people don’t expect it. It’s not that I’m into complexity for complexity’s sake, but I do like to exceed expectations when possible. So, when The Floyd Show wanted to take things to the next level by having quadrophonic sound available, I was pretty thrilled.

I was so thrilled that, the second time we did a show that way, I went a little too crazy. I put a bunch of channels through reverb and delay, and pushed all that through the rear speakers. Loud. I wanted to hear it!

About a third of the way through the show, one of the club’s security humans came up to me.

“Dude, you’ve REALLY got to turn those down.”

Oops.

What went wrong? Was I tearing people’s heads off?

No, as I found out later. What was happening was that some people were getting an overpowering “FX to dry” ratio – and it was all because they were really close to the rear loudspeakers.

The Correct Solution Over Here Is Wrong Over There

In small-venue sound, there’s a bit of truth that’s hugely relevant…and yet rarely discussed:

A mix “solution” that is the result of both acoustical sources and PA reinforcement is spatially dependent. A listener at a different point in space is not necessarily receiving a solution of the same validity as people at other points in space.

In other words, what sounds perfect in one spot may not sound all that perfect when you’re on the other side of the room, especially if a listener gets (proportionally) very close to part of the PA.

Why?

SPL (Sound Pressure Level) increases as distance to a source decreases. Not everybody knows the math involved for modeling this reality with physics, but I’m pretty sure that almost everybody has an intuitive grasp of the idea. (The math actually isn’t that hard, by the way.) The reason this matters so much for small-venue audio humans is as follows:

If the sound reinforcement system is only responsible for a portion of a mix “solution,” a listener that is in close proximity to the system is likely to be experiencing a mix which is overbalanced in favor of the PA.

(Yes, this is essentially a restatement of the first point.)

A Common Example

To look at this in familiar terms, let’s consider a PA system that’s only reproducing vocals. The PA is located just in front of the band, with about twenty feet between the stacks. Everything else in the room is coming from the band’s instruments on stage. An audio human, situated 30 feet from the stage, in the center of the audience area, creates a mix solution that they like. This mix solution is, of course, a blend of the PA plus everything else. The validity of the solution depends on the blend’s proportionality remaining the same.

For many points in the room, the proportionality does indeed remain relatively stable. It remains stable because the DIFFERENCE in distance from the listener to either the PA or the band doesn’t change too wildly. In fact, as listeners get farther away, the proportion between the distance to the PA versus the distance to the band is reduced – that is, the proportion gets closer and closer to being 1:1. If you’re somewhere behind the sound operator, your chances of getting basically the same mix solution are pretty good – even if you’re off to one side.

(Of course, that mix solution may be highly colored by room reflections – that is, reverb – but the fact remains that what you’re hearing is the “correct” solution plus reverb. Then again, to be fair, very strong and/ or unpleasant reverberation can result in a total acoustical sound that’s utterly terrible…)

Anyway.

Where problems start to happen is in the area in front of the FOH (Front Of House) engineer. The closer that a listener gets to the front of the room, the more the proportionality between the sound sources diverges from 1:1. In this particular example, a person standing dead center, four feet from the stage is almost three times closer to the stage than they are to the PA. It’s quite likely that, for that listener, the stagewash is overpowering the vocals-only PA to some degree. (The issue is probably compounded by the listener being out of the throw patterns of the PA speakers, although that’s beyond the scope of this article.) On the other hand, a person that’s down front and off to one side could be getting four times as much PA as stagewash, if not more. For them, the vocals might be a bit too “hot.”

Some Things Can Be Fixed. Other Things…

The bottom line here is that if the PA (or even just some part of the PA) isn’t the whole mix, then you have to be mindful of where and how your mix solution can change. In my case, what I failed to consider was that the people in the back of the room were getting an overdose of FX from the surrounds. I pulled the rear speakers down, and everybody was a lot happier. The folks in front probably weren’t getting much from the rear boxes anyway, so it wasn’t a big loss to them.

You can’t fix everything, though.

In small venues, people can listen from all kinds of places, and you probably won’t have the gear available to cover all of those places. Big shows can fix their problem areas with fills and delay stacks of all kinds, but little shows just have to “shoot for the average.” In my personal opinion, if 75% or more of the audience seems to be getting basically the same mix, then you’ve done your duty. Of course, trying for 100% is usually praiseworthy, but completely overcoming the problem of inconsistent distance in a small venue is expensive, time consuming, and chews up a lot of space. In fact, 100% coverage might not even be what you want – in small rooms, it can be very nice for people to be able to “get away” from the full-blast of the show.

(You also have to consider other psychology that’s involved. For some folks, an “off” mix is a tiny price to pay for being able to be nose to nose with their favorite band. A happy audience is a happy audience, any way you slice it.)


The Acoustic Crossover

If you don’t need it, don’t spend power (or volume) on it.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

For loudspeakers, a crossover is used to separate full-range audio into multiple “passbands,” with each passband being appropriate for a certain enclosure or driver. For instance, there’s no need to send a whole bunch of high-frequency information to a large-diameter speaker if you’ve also got a handy device that’s better for top-end. On the flipside, failing to filter low-frequency information is a good way to wreck a “meant for HF” output transducer.

A beautifully implemented crossover creates a smooth transition from box to box and driver to driver. Crossovers can also help with getting the maximum performance out of an amplifier/ loudspeaker chain – again, because pushing material to a driver that can’t reproduce it is a waste of power.

Most of the time, we think of a crossover as an electrical device. Whether the filter network is a bunch of passive components at the end of a speaker cable, or a DSP sitting in front of the amplifiers, the mental image of a crossover is that of a signal processor.

…but remember how I’ve talked about acoustical resonant circuits? The reality of the pro-audio life, especially in small rooms, is that the behaviors of electrical devices show up in acoustical form all the time. In the past few years, I’ve found that creating acoustical crossovers between the stage wash and the FOH (Front of House) PA can be incredibly useful.

Why This Matters In Small Rooms

In a small venue, you don’t always have a lot of power to spare. It’s rarely practical to deploy a PA system that can operate at “nothing more than a brisk walk” for most of the show. Instead, you’re probably using a LOT of the audio rig’s capability at any given time.

Even if you have a good deal of power to spare, you often don’t have very much volume to spare. A small venue gets loud in a big hurry – not only because of acoustics, but because the average audience member is “pretty dang close” to the stage and PA.

Taken together, these issues present hat-explodingly good reasons to avoid chewing up your power and/ or SPL budget with audio that you just don’t need. Traditionally, dealing with this has taken the form of not reinforcing entire sources or channels. (This can oftentimes, and unfortunately, be appropriate. I’ve done several shows where one person was so loud that everyone EXCEPT them was in the PA.) An “all-or-nothing per channel” approach is sometimes a bit too much, though. What can be better is to use powerful and dramatic, yet judiciously applied subtractive EQ.

Aggressive Filtration

A good way to illustrate what I mean by “powerful and dramatic, yet judiciously applied subtractive EQ” is to show you some analysis traces. For instance, here’s my starting point for a vocal HPF (High Pass Filter):

vocalfilter

The filter frequency is 500 Hz. Effectively, I’m chucking out everything at or below about 250 Hz.

“But, doesn’t that sound really thin,” you ask?

Indeed, it does sound a bit thin at times. If I don’t have a lot of monitor wash, or the singer doesn’t have a voice that’s rich in low-mid, or if they just don’t want to get right up on the mic, then I need to roll my filter down. On the other hand, in situations where the monitors were loud, the vocalists had strong voices, and they had their lips stuck to the mics, I’ve had HPF filters up as high as 1 kHz or more.

The point is that the stage-wash often gives me everything I need for low-mid in the vocals, so why duplicate that energy in the FOH PA? If I create a nice transition between the PA and what’s already in the room, I only have to spend power on what I need for clarity.

Now, here’s a trace for a guitar amp:

guitarfilter

Of course, you don’t necessarily need something as extreme as this all the time. What’s great about filtering a guitar like this, though, is that you’ve thrown away everything except the “soul” of the instrument – 400 Hz to 2 kHz. Especially with “overly scooped” guitar sounds, what you need for the guitar to actually sit in the live mix is more midrange than what you’re getting. Of course, you could turn up the ENTIRE guitar to get what you need – but why? You’ll be killing the audience. It’s much better to “just turn up the mids” without turning up anything else.

…and even if the guitar is only really in the PA during solos, this kind of filter can still be a good thing to implement. If you have to REALLY get on the gas for a lead part, you can avoid tearing people’s heads off with piercing high end – as well as avoid stomping all over the rhythm player and the bassist.

By combining a highly filtered sound with the stage volume, you effectively get to EQ the guitar without having to completely overwhelm the natural sound from the amp. (This is just an acoustical version of what multiband equalizers do anyway. You select a frequency range to work on, and everything else is left alone. Whether this happens purely with electrical signals or in combination with acoustic events is relevant, but ultimately a secondary issue.)

Now, how about a kick drum?

kickfilter

Again, this kind of thing isn’t appropriate in all contexts. You wouldn’t do this for a jazz gig…but in a LOT of other situations, what you need from the kick drum is “thump” and an appropriately placed “pop” or “click.”

And that’s it.

In a small venue, reproducing much of a rock or pop kick’s midrange is unhelpful. All you do is run over everything else, which makes you turn up everything else, which makes your whole mix REALLY LOUD.

Instead, you can create an acoustical crossover to sweeten the kick “just enough,” without getting any louder than necessary.

All Wet

Saving power and volume also applies for situations where you want effects to come from the PA. It’s very easy to get too loud when you want to put reverb, delay, or even chorus on something. The reason for this is because these effects have a “dry” (unprocessed) component, that has to be blended properly with the “wet” sound. What can happen, then, is that you end up pushing the entire sound up too far – because you want to hear the effects. The “dry” sound in the signal combines with the “dry” sound in the room, which makes for an acoustical result that isn’t as “wet” as you wanted…so, you push the volume until the “dry” sound through the PA overwhelms the sound in the room.

That can be pretty loud.

Instead of brute force, though, you can just tilt the “wet” ratio much further in favor of the effect.

In fact, I’ve been in some situations where, say, a snare drum was in exactly the right place without any help from the PA. In that case, I set up my routing so that the snare reverb was 100% wet – no “dry” signal at all. I already had all the “dry” sound I needed from the snare in the room, and so I just turned up the “all wet” reverb until the total, acoustical result was what I wanted.

The bottom line with all this is that, in a small space, you can get pretty darn decent sound without a screaming-loud PA. You just have to use the sound that you already have, and very selectively add the bits that need a little help. The more fine-grained you can be with the creation of this acoustic crossover, the more you can bend the total acoustical result to your will…within reason, of course.


The Curious Case Of The Miced Acoustic That Fed Back

Putting a mic in front of an acoustic guitar does NOT allow the laws of physics to be overcome.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

Every so often, I’ll work on a set (or even a whole show) where I struggle. It’s why I try to remember to say, “I hope it’s not my night to suck.” I think it’s important to be honest about not being able to work miracles.

Anyway.

Not too long ago, I did a show where the opening act brought an acoustic guitar. Please note my exact words: “Acoustic Guitar.” Not electro-acoustic!

Acoustic guitar. No pickup, that is.

Luckily for me, the opening act’s set was pretty short. This was lucky because I had more feedback problems in that one set than I usually have in two-months-worth of shows. Weird rings. Phantom squeals. High-pitched ghosts that bared their teeth and then disappeared. It was embarrassing, and un-fun.

My mistake primarily lay in trying harder to make the performer happier than the laws of physics would allow. I should have gotten on the talkback and said, “I’m sorry, but I think that’s all we can get out of this setup tonight,” but I didn’t. I tried to fight my way through, and I think the end result was worse for it.

…but everything seemed okay during soundcheck. What went wrong?

The Changing Environment

Your gear isn’t the only thing with a noisefloor. (The noisefloor is the voltage or sound pressure level where non-musical information can be found. It usually sounds like hiss, or rumble, or hum, or a combination of all three.) A venue also has a noisefloor, and unlike a well-maintained piece of equipment, a venue’s noisefloor can change wildly and quickly.

In the case of the problematic set, we were fine at soundcheck. The performer was happy with the onstage blend between his voice and his guitar, and we all liked how things sounded out front.

The venue noisefloor was also about 50 – 60 dB SPLC (Sound Pressure Level, C weighted).

Between soundcheck and the actual show, a rather dramatic thing happened: A whole bunch of college-age humans arrived. Unsurprisingly, most of them were talking to each other. If I had my guess, the new noisefloor was probably between 75 and 85 dB SPLC. In “linear” terms, that’s a magnitude difference with a factor between about 30 and 300.

I’m not joking. An 85 dB SPLC noisefloor is just a bit more than 300 times louder when compared to a 60 dB SPL noisefloor. Logarithmic math is a heck of an eye-opener, I tell ya.

For a performer who’s perception of the “correct” level for their sound was formed in an empty, relatively quiet space, the addition of the crowd certainly had a HUGE effect. What’s more, I’m guessing that the total level on stage was only slightly higher (3 – 6 dB) than the level of the crowd’s conversations. Even worse, the “roar” was probably right in the critical ranges for both the guitar and the vocals.

So, of course, the performer wanted more level from the monitors. He couldn’t hear himself properly anymore – he even said so, outright.

I got on the gas with both the guitar mic and the vocal mic, and that’s when the fight start – I mean, that’s when my feedback issues took hold.

I Had A Problem, So I Added A Mic. Then, I Had Two Problems

Another issue that worked against me was that I had two mics contributing to one “loop.” There was a mic for vocals, and one for the guitar. The mics were in relatively close proximity, and being put through the same monitor.

At high gain.

See where this is going?

Essentially, the two microphones combined into a single, extremely high-gain device that was in a partially closed loop with the wedges. Of course the system was unstable. Of course it was a battle. The gain was so high that, if one of the “so much vocal power that my usual head-amp preset would be driven into hard clip” singers around town had grabbed a mic at that setting, they would have launched a monitor’s LF driver through the grill and into their face.

But here’s the thing:

Gain is proportionally related to acoustic output, but gain is NOT absolutely related to acoustic output.

That is to say, more gain will produce more volume compared to lower gain on the same signal, but the measured, acoustic sound pressure level for any particular gain setting will not always be the same. The entire acoustical and electrical signal chain is ultimately responsible for that.

So, we were running at “super hot” gain levels, but we weren’t all that loud. Unfortunately:

Undamped feedback in a loop is a product of gain, not volume. The only limiting factor that volume represents is that the system must be able to produce enough level to be audible over the noisefloor.

The performer could barely hear himself, but when the system “took off,” all of us could hear THAT just fine.

Reflection and Resonance

There are a couple of other factors that contribute to acoustic guitar feedback issues, especially when monitor wedges are involved.

The first factor is resonance. An acoustic guitar works as an acoustic guitar because of the big, vibrating box that the strings are attached to. The box works because it vibrates in response to external stimuli. The problem is that the box can’t tell the difference between the stimulus presented by the strings, and the stimulus presented by a sufficiently-loud monitor wedge. Get the wedge loud enough at the right frequency, and the resonant acoustic circuit you’ve just unleashed will ring until you do something to stop it.

In the case of the show I’ve been referencing, I don’t think we got the monitors loud enough for wedge-to-body resonance to be a real factor. What may have been a factor, though, is reflection.

Onstage feedback happens when the audio captured by a mic is output through a loudspeaker, and then re-enters the same mic. It doesn’t really matter how the audio returns to the mic – it just matters that it does. So, what do you think happens when a mic is pointed at an acoustic guitar body, which is big, and flat, and not completely absorptive, and which is also right in the path of the audio coming out of the monitors?

Yup.

The monitor audio hits the guitar body and reflects back into the mic. Sure, the lower frequencies might diffract around the guitar, or just pass through the thin walls of the body, but the high frequencies are a different story.

SQUEEEAALLL!

And, of course, the squeal comes and goes, because the guitar player is probably moving around a bit. A lot of the time, you might just barely be okay, and then the guitarist gets everything in just the right alignment…

SQUEEEAALLL!

The Upshot

At this point, the question becomes: “What can we take away from this?”

I think the main takeaway – and it applies to everybody, performers and techs alike – is that a purely acoustic guitar really can’t be expected to be dramatically louder than it already is. Perhaps even more correctly, a purely acoustic guitar can’t be expected to be dramatically louder than it is, as experienced by the microphone capsule.

As a result, if an acoustic guitar needs to be at 90 dB SPL in order to compete with a rowdy crowd, then it really needs to be making at least 87 dB SPL without any help from the PA. If, for some reason, the guitar needs to be a great deal (10 dB or more) louder than it is naturally, then we must have some way of “partially opening the loop” that includes the guitar, the mic, and the audio rig. Either that, or we have to make the guitar much louder – from the mic’s perspective – than the wedges and main PA.

The most practical way to do this is with an internal pickup, optionally coupled with a soundhole cover. The internal pickup gains some isolation by virtue of being inside the guitar body (or outside, but directly coupled to some part of the guitar), and the mic also “perceives” the guitar as being quite loud.

Because it’s, you know, inside or directly attached to the guitar. Life is pretty loud right there, just like it’s really loud inside a piano.

The soundhole cover helps by providing even more isolation from external sounds, and also by changing the resonant frequency of the guitar body. The size and shape of the soundhole is a major component in determining what an acoustic guitar sounds like, and closing the hole may just shift the body resonance to a non-problematic area.

In the end, we all need to know our abilities, and the abilities of our tools, and be aware of when we’re asking too much of ourselves or our gear. We also need to be able to look back at our problems with an analytical eye, and figure out exactly what went wrong.

Of course, I’ll probably end up trying to break the laws of physics again in six months, because I have a short memory for situations I don’t encounter every week…


It’s Called “A Band”

A Small Venue Survivalist Saturday Suggestion

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

This thing that you’re involved in is called “a band.”

It is not called “the sound of your amp’s power tubes saturating while a few other people hang out on stage for no discernible reason.”


The SA-15MT-PW

You won’t be let down if you’re not asking too much.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

I have a weakness for “upstarts.”

I love to find little companies that are doing their own thing and trying to have some kind of fresh take on this business. When I ran across Seismic Audio, I got a little infatuated with ’em. Here were some audio-gear humans that were doing something other than selling everybody else’s boxes. Here were some folks who had some interesting, hard to find, “nifties.”

Seriously – maybe I just wasn’t looking hard enough, but Seismic had a reel mounted, 8 send/ 4 return snake with 50′ of trunk when nobody else did. And it was the perfect thing for what I needed at that moment. Neat!

Anyway.

Of course, when you’ve been at this kind of thing for a while, you develop a sort of BS detector for sales. You look at a page that’s trying to get you to buy equipment, and you instinctively throw out all the numbers and claims that you think are “best case scenarios.”

What other people call, “horrific, crippling cynicism that threatens to poison all joy and blot out the sun” is what I call “being realistic about what the sales guys are saying.” (Welcome to the music business, boys and girls!) With a company like Seismic, my thought process goes like this: “These guys have some neat offerings, and the prices are quite low. In order to do that, they must be saving money somewhere. I should expect that some part of these products is going to be demonstrably ‘cheap,’ and not be disappointed when I discover that part.”

So – when it was time to buy some new monitor wedges (because half of the current set had committed full or partial suicide), I decided to take a chance on Seismic. My goal was to get some boxes that would be “enough” for a small stage, and not to worry about having varsity-level build quality or performance. The gear wouldn’t be moving around a lot, and the bands I work with are good about not abusing monitor wedges, so I was confident that a “middle of the road” unit from Seismic would work out.

Trying Not To Ask Too Much

As I looked around Seismic’s site, I found myself gravitating towards the SA-15MT-PW monitors. Because of their configuration, which puts the HF (high frequency) driver above the LF cone, they would be naturally “bookmatched” (vertically symmetrical) when operating in pairs.

In the absence of a specific claim one way or another, and at the box’s price point, I was very sure that the 15MTs were neither biamped nor equipped with fancy processing. I would essentially be receiving passive monitors that had an amplifier and input plate added. I was perfectly okay with this.

I hoped that the 15″ driver would provide an overall output advantage over a 12″ unit, especially since I just decided to not care about anything below 75 – 100 Hz. In an environment like the one where I work, using the onstage monitoring rig to create thundering bass and kick is not a priority. The monitors are mostly there for the vocalists, with instrument midrange as an added bonus.

Seismic’s site claims that the box will play down to 35 Hz, but let’s be real. Making 35 Hz at “rock show volume” is hard enough with a properly built “mid-pro” sub, and 15MTs are affordable drivers in a small box. I’m sure the LF cone will vibrate at 35 Hz, but you’re not going to get much usable output down there. See? Being cynic – ah – REALISTIC is good for you.

I’ve also become very good at ignoring “peak” numbers in power ratings. Whenever you see “peak,” you should mentally change the wording to “power that’s not actually available to do anything useful.” What you really want to find is the lowest number with “continuous” or “RMS” next to it, and assume that’s what you’re going to get. SA claims that their powered 15MT units have an amplifier that’s capable of 350 watts continuous. I think this may be an increase from when I was shopping, as I remember something more along the lines of 250 watts.

What I was most prepared to accept was the published sensitivity rating and maximum SPL. With one watt (actually, 2.83 VRMS, but that’s a whole other discussion), the boxes were supposed to be capable of 97 dB SPL, continuous. The maximum SPL was listed as 117 dB SPL.

This seemed pretty reasonable to me, and here’s why: With a 97 dB sensitivity, 250 watts of continuous input should get you into a calculated maximum SPL range of 121 dB. That they published a maximum number LOWER than that (117 dB) made me feel more comfortable. Seeing a number that suggested that the driver could only really make use of half the available continuous power reassured me. This isn’t actually strange – when you see a number that is below the theoretical maximum, it gives the impression that the boxes were actually measured as being able to hit that number.

I thought to myself: “If I put two boxes together, I can get 120 dB SPL continuous out of the pair. That’s enough for our stage. If somebody wants more than that, then they’re playing the wrong venue. I’m going to order these.”

I got free shipping and a discount. That felt pretty good.

A Short Honeymoon

When the 15MTs arrived, I was encouraged. They weren’t too heavy, and you could actually lift and carry them in a not-awkward way by using the handles. I wasn’t wild about the way that an XLR connector would stick out of the side-mounted control panel when everything was connected, but I’m pretty much convinced that getting a powered speaker’s input panel to be perfect is an impossibility. I tried the boxes with a bit of music, and the sonics seemed fine. The playback was clean, and the overly present (too much high-mid) timbre of the boxes was easily tamed with the onboard EQ – which is a good sign, because if simplified, broad-brush EQ can fix an issue, then the problem isn’t really bad.

The units were put to work immediately in a festival setting. The venue was hosting an annual showcase of local talent, and this seemed like an ideal way to put the monitors through their paces. They’d need to perform at both quiet and moderately loud volumes, and do a lot of different acts without a hiccup.

The first day went by without a problem.

The second day wasn’t so good.

The high-energy point in the night was being provided by a band with strong roots in both country and rock. Even in an acoustic setting, this part of the show was meant to be pretty “full tilt” in terms of feel and volume. I got the lead singer’s vocal switched up to the necessary level – and that’s when I noticed the problem.

The 15MTs were distorting, and distorting so audibly that the “crunch” could clearly be heard through the cover that FOH (Front Of House) was providing. I was very, very embarrassed, and made it a point to approach the lead singer with an apology. I assured him that I knew the sound was unacceptable, and that I wasn’t just ignoring it.

He was very gracious about the whole thing, but I was not happy.

Oversold On The Power

With what seem to be updated specs on the Seismic Audio site, I can’t be sure that the SA-15MT-PW units currently being sold are the same as the 15MTs that I was shipped. What I can say, though, is that the units I was shipped do not appear to meet the minimum spec that was advertised at the time.

Not the hyped specifications – like I said, I ignore those as a rule – the MINIMUM spec was not achieved.

I’m about 90% sure that I’m right about this. The 10% uncertainty comes from the fact that I don’t exactly have NASA-grade measurement systems available to me. I can’t discount the issue that significant experimental error is probably involved here. At the same time, I don’t think that my tests were so far off as to be invalid.

After empirically verifying that sufficient volume with a vocal mic caused audible distortion (and at a volume that I felt was “pretty low”), I decided to dig deeper.

I chose a unit out of the group, removed the grill, and pulled the LF driver out of the enclosure. I then ran a 1 kHz tone to the unit, and increased the level of that tone until I could clearly hear harmonic distortion. The reason for doing this is that, for a single-amped box equipped with a 1″-exit HF driver, 2+ kHz distortion components will be audible in the HF section without being incredibly loud. The small HF driver is probably crossed over at 2 kHz or above, so the amp going full-bore with a 1 kHz input is unlikely to cook the driver.

Having found the “distortion happens here” point, I then rolled the test tone down to 60 Hz. My multimeter is designed to test electrical outlets in the US, and that means that its voltage reading is only likely to be accurate at the US mains electrical frequency of 60 Hz. I connected my multimeter’s probes to the LF hookup wires, and got this:

As the image states, 32.9 volts squared over 8 ohms (the advertised nominal impedance of the drivers in the unit) works out to be 135 watts…and that is with VERY significant distortion. In general, when an audio manufacturer claims a continuous or RMS power number, what that means is the unit can supply that power without audible distortion. In this case, however, that didn’t hold.

To be brutally frank, if this measurement is correct then there is NO PHYSICAL WAY that the units I was shipped have 250 watts of continuous power available, with the peaks exceeding that number. You might be able to get 250 watts continuous out of the amp if you drove it as far as possible into square-wave territory, but you wouldn’t want to listen to anything coming out of the monitors at that point.

Oversold On The SPL

After reconnecting and reseating the LF driver, I ran pink noise into the unit at the same level as the test tones. I then got out my SPL meter, and guesstimated what 1 meter from the drivers would be. Here’s what I got:

If the claimed maximum SPL was 117 dB, then this is actually a pretty consistent reading with the box only being able to do about half of what was advertised. Even so, this number was generated by a unit that was beyond its “clean” output capability.

Now – again – let’s be very clear. I can only speak about the specific units that were shipped to me. It would be wrong to say that these results can be categorically expected across all of Seismic’s product lines. It would also be wrong to say that the 15MT-PWs being shipped today are definitely the same design as what was shipped months ago. Products can change very fast these days, and it may be that Seismic has upgraded these units.

I also want to be clear that, dangit, I’m rooting for Seismic Audio. I love scrappy, underdog-ish businesses that carve out a space for themselves. Heck, maybe a parts manufacturer sent them the wrong amplifier plates. Maybe the passive versions of these boxes (and every other speaker they sell) meet the minimum spec. The only way to know would be to buy and test a sample of everything.

Still, in this particular case, the products I was shipped did not live up to what they were advertised to do. They’re not useless by any means, but they can’t quite cover the bases that I need them to cover. I’m planning on finding them a good home with people who can actually put some miles on ’em.

…and, Seismic folks, if you read this: I don’t want my money back, or I would have asked when I still could. I do want you guys to keep doing your thing, and I also want you to be vigilant that the products you ship actually meet the minimum spec that you list on your site.


The Tipping Point

“Loud” can become a vicious cycle, and FAST.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

I’ve mixed my fair share of bands in small, “live” rooms. What I mean by “live” is that the space has a lot of hard surfaces. For any sound introduced into the space, a good deal of the pressure wave travels directly to the audience. At the same time, a large portion of the same wave encounters an acoustically reflective boundary, and re-emits into the room. The re-emitted sound has a pretty good chance of also reaching the audience, even if the overall intensity has dropped somewhat.

This is why audio humans, music folks, and audience persons say things like “this is a loud room.” It’s not their imagination, or just a trick of perception. It’s physics, and you might not realize it, but physics impacts your ability to build a fan base and get paid decently for your efforts as a band.

I’m serious.

I’m sure that you’ve heard all the old (and not old) saws about excessive volume driving people away. I’ve written about that on this site. There have been plenty of discussions about volume wars. What I haven’t heard much talk about, though, is how excessive volume not only drives people away, but also how people being driven away creates even more excessive volume.

The Cycle Of Increasing Loud

The hingepoint for all this is that humans are pretty decent at being acoustical absorbers. Pack a room with a bunch of water-filled creatures that drape themselves in fabric, and a fair amount of sonic energy is prevented from bouncing around.

If you remove people from the room, then the “human absorption” factor drops. As the absorption drops, the amount of sound that can travel to the remaining listeners goes up, as does the amount of sound that can travel to them indirectly after hitting a boundary.

So, here’s what happens.

A band that produces a good deal of volume gets on stage in a small room with lively acoustics. They’re still building their following (or it’s just an off night), and so there aren’t a ton of humans available for absorption. The indirect sound from room reflections combines with their already “just a bit hot” volume to create “kinda uncomfortable” volume. The folks who aren’t really into it decide to head for a different part of the bar, or go outside. Now, there’s even less absorption in the room, especially for the people who were hanging out next to the folks who just left. As a result, the “just slightly too loud” show is now “just slightly more just slightly too loud.”

A couple more people decide to take a break from the onslaught.

The room is now a bit less absorptive.

The show gets a bit louder.

Do you see where this is going?

The “loud” ends up building on itself, progressively clearing the room, until some equilibrium point is reached. The band is playing to fewer people, which makes the show less fun. If the venue is a bar, the show is less profitable for the room. On top of that, “walk up” traffic from people in the neighborhood is pushed away, because the show is uncomfortably loud while simultaneously feeling “dead.”

The band’s playing, but they’re not generating much momentum for their career.

For Your Own Sake, Stay Away From The Tipping Point

The key to not being bitten by a vicious cycle is to avoid feeding the cycle. If you’re in a band that needs to grow its fanbase, you will do yourself a huge favor by figuring out how to be elite at being quiet.

Seriously, you should become freaking NINJAS at putting on a show that generates less than 100 dB SPL C from the stage.

Although…maybe that’s not really ninjitsu. Maybe that’s more like “normal” difficulty. Get down to 97 dBC for a “heroic” rating, and 94 for “legendary” status. If you want to be a mythical figure, then you should be able to pull things back even farther when you notice that you’re in an exceptionally challenging room. Yes, your guitar tone might not “sing” in exactly the way it does when you get on the metaphorical gas in your practice space. Yes, the bass might not feel like it can move continents. Yes, the snare might not have that supermassive “crack-boom” that you always wanted.

But the thing is that nobody in the room cares about any of that, except for maybe those two dudes from the big-box music store who are totally opinionated about “creaminess,” and “breakup,” and “punch,” and “really taking advantage of those EL84 tubes,” and “rumble,” and “pop.” Ask yourself: Are they actually helping you write your ticket? Are they really the people you need to impress?

(The answer is probably “no,” by the way.)

Stay away from the tipping point, and you’ll have a much better chance of having more people in the room who love hearing your songs. That means a better show overall, more money for the venue, more money for you, and ultimately, a more viable career path.

…and isn’t that what this is all about?


Get Out Of The Effing Way!

Do what’s actually helpful, and then stop “do-ing.” The show will be just fine.

Please Remember:

The opinions expressed are mine only. These opinions do not necessarily reflect anybody else’s opinions. I do not own, operate, manage, or represent any band, venue, or company that I talk about, unless explicitly noted.

I’m told that live-audio is a thankless job. This is news to me, because I’m fortunate enough to be thanked – often – for the work I do.

The problem is that when I receive the most effusive praise, I feel like a fraud. When somebody says, “you made it sound SO GOOD,” I’m pretty sure that credit isn’t really going where it’s due. I can’t clearly remember the last time I took something that sounded bad, and through force of will, turned it into an amazing sonic experience. Mostly, I take bands that already sound good, and then help the couple of things that can’t help themselves to be in quite the right place.

The more time I spend working in small rooms, the more it seems to me that at least 50% – 75% of my job is staying out of the way of the band. Functionally, what this means is that a lot of my faders sit at either the actual “negative infinity” point, or at a functional negative infinity (that is, the fader is set at a level where the contribution from the PA is not readily perceptible).

Because of the popular perception of what an audio human is supposed to do, the desirability of having a number of channels “doing nothing” is counter-intuitive. I’m pretty comfortable with it as a concept, and even I will go through periods where I get out of discipline.

But it works beautifully.

If You Don’t Need To Be On Deck, Don’t Be On Deck

That heading up there is one of the phrases that stage crews learn. It’s also pretty easy to grasp at a physical level. At some point, it’s easy to see that chewing up space on stage while contributing nothing to the show is a bad idea. You’re very likely to impede the progress of someone who actually needs to, you know, get stuff done. If you’re impeding the progress of someone with a job to do, you’re very likely to hear the words “Get out of the !@#$ing way!”

The issue for the audio human standing behind the console is that the line between “helpful contribution” and “just taking up space” isn’t clearly marked. Audio is a really subjective sort of business, and a lot of what audio humans do involves end results that aren’t easily measured in an objective way. The overall problem is exacerbated because of the mistaken belief that the most important work in audio engineering for entertainment is done with mixing consoles and rack gear. It isn’t – but that’s probably a whole other article.

With work at show control being the object of worship, the audio human can feel a lot of pressure to “make magic happen” with the tools that are seen as being most important. Especially because live-audio is an additive affair – where the sound of the PA combines with the sound from the stage – this need to appear productive can lead to three, generalized, “bad sound” scenarios:

1) The band’s sound is tossed out in favor of the engineer getting his or her sound, with questionable results.

2) The show is very, very loud, with questionable tolerability.

3) A combination of both.

Trying To Fix Everything

There’s a hilarious Xtranormal video floating around that illustrates scenario #1. In it, a jazz drummer is up against an audio dude. In complete deadpan, dialogue in this overall vein is uttered:

“We need to cut a hole in your kick drum.”

“This is a jazz kit. It uses special, very expensive heads. We are not cutting a hole in anything.”

“But, dude, how will I get my sound if we don’t cut a hole in the front of the kick?”

The third line is the crux of the whole thing. Getting what is perceived to be the most impressive kick sound – from the tech’s perspective – has become such a priority that the audio human is blind to what they’re actually dealing with. The need to be looked upon as a wizard with a console and outboard processing is so great that the tech is ready to turn a jazz act into a rock band, and to do so by wrecking an instrument.

When this need to fix everything generalizes into a whole-band situation, you can very quickly cross into the territory where the band no longer sounds like itself. Instead, you get the tech’s best effort at a huge snare, massive kick, elephantine toms, roaringly thick guitars, thundering bass, and “radio announcer” vocals…all blended into a result that sounds like a completely different band playing another band’s songs. Usually, the overall sound is that of the audio human’s favorite genre, and so the tech gets away with this behavior as long as the band is in that general area of music. When the band is significantly different, though, people walk away saying, “It just didn’t sound right,” without necessarily knowing why. If something about the band prevents the tech from achieving “the right sound,” the audio human is apt to complain that the band was hard to work with.

Luckily, the medicine for this condition is cheap, and easily available: Get out of the effing way! It’s amazing how adopting the attitude of “it’s about getting the band’s sound in the room, and not my own sound” can simplify your life. It’s amazing how much less agonizing you have to do when you don’t need to do microsurgery on every input’s frequency response and dynamic range. It’s amazing how fluid, simple, and enjoyable a show can be when every second of it doesn’t have to be managed.

It’s also amazing how, when you stop trying to fix everything, you don’t have to throw so much money at more and different gear.

The Show Sounds Huge, But Everybody Left

The #2 and #3 scenarios often result from trying to fix everything. As I mentioned earlier, this is because sound reinforcement is an additive exercise.

Live-sound engineer and gear retailer Mark Hellinger really nailed it when he stated a particular belief of his: Audio techs don’t feel like they’re really in control of the show until the PA is 10 dB ahead of everything else.

This anecdotally supported belief dovetails nicely with quantitative observations of SPL (Sound Pressure Level) addition. If you add the SPLs of two sound sources, where one source is observed to be 10 dB more intense than the other, the result will be the SPL of the louder source plus about 0.4 dB. The loud source pretty much wipes out the quieter sound.

So…

As the “I have to fix everything!” audio tech goes about getting their sound, they have to overcome the sound of the band in the room. The amount of SPL that a band can produce, even without a PA, can be rather surprising. (If you ever want a quick and unmerciful education in this, work with a band that switches from an acoustic drumkit to an electronic drumkit. You will be shocked at just how hard you have to drive the PA to make the e-kit fit in the same SPL “box” as the acoustic drums. I speak from experience.) A full-tilt band – even a fairly reasonable one – in a small venue can be a pretty loud experience. If the audio human just has to override the band’s sound with his or her own sound, “pretty loud” has probably just had anywhere from 6 – 12 dB of continuous SPL added on.

In a small space, this can mean that the engineer’s mad rush to fix everything creates a mad rush for the exits. A tech’s incorrect prioritization can clear a room just as much as a musician’s myopia can do the same thing.

As with trying to fix everything, getting out of the effing way can keep more people in the venue. When the goal becomes working WITH the sound of the band in the room, instead of against it, you have a much better shot at keeping within an audience’s reasonable SPL range. (No guarantees, of course. This is a subjective business.)

Going against the flow of a band’s sound is difficult, loud, and requires a ton of work. Getting out of the way and swimming with the current is much easier, much quieter, and a lot less tiring. It can be hard to discipline yourself to work this way (I still get things very wrong in certain situations), especially since not getting “your sound” fails to feed your ego (please refer to the previous parenthetical statement).

But I’ll be doggone if getting out of the effing way hasn’t proven itself to be very effective.