Category Archives: Gear for Sound and Lighting

Reviews and opinions regarding audio and lighting equipment.

The Turbosound Milan M12

A nice box, but flawed.

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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When I was adding onto my system last year, I chose Turbosound Milan as the product line for FOH. Since putting those boxes into service, my feelings have been mixed. The most mixed of those feelings have been reserved for the mid-highs I chose, designated as “M12” by the manufacturer.

I do like the compact nature of the package. Other powered 12s that I carry are similar in weight, but inefficient in their use of bulk. The Milans chew up less space, and yes, they have a monitor-angle on both sides. You can properly book-match a pair of the little darlings, which is something I appreciate.

I also like the overall fit and finish. Yes, they’re plastic boxes, but it’s the kind of plastic that can take some wear gracefully. The controls and connection points seem to be reasonably well-engineered, with slide switches that clearly indicate where they’re set. (Push-toggles are fine if they unambiguously show their state, but plenty of them don’t – so, kudos to Turbosound on this front.) I often work with other boxes that really are just fine…but feel “cheap” when it comes to XLR connectors and back-panel interaction. The Milans are a definite upgrade there.

M12s do seem to be tuned pleasingly at the factory, which is a big help for throw-n-go gigs where you have to make things work out tonally without a lot of prep time. Your mileage may vary, of course, especially since just about anything can be whipped into shape these days.

Also, let’s be honest: My anti-establishment nature has a special place for brands that are less common. Everybody knows JBL, Peavey, EV, Yamaha, and so on, but Turbosound is a loudspeaker marque that’s a little less trafficked in small-format circles. (Turbo’s big-boy boxes are more well known to the folks who work at that level.)

What do I not like? Well…

Milan M12s are a (tiny) bit expensive for what you get – both in money and weight. When JBL marked their Eon 612s down, they really threw the gauntlet at Turbo. Spend $50 less, get a box that has essentially the same performance, and save about 12 lbs.

…and Turbo, geeze, can we please have a real “thru” on the back? Sometimes I just want to chain two boxes together, and I don’t want to have to volume-match them by ear. Especially if I’ve forgotten to do so before the speakers are eight feet in the air already.

But that’s not the biggest thing.

What really put me off with the M12s was how they will audibly distort before they illuminate their clip indicators. It’s not a horribly nasty sound, but its “too obvious” and a little embarrassing. When somebody addresses the crowd at concert level, using a mic that has some low-mid dialed into it, there’s no reason that a loudspeaker of this type should suddenly give the impression of being underpowered. Sure, these units travel with the crowd that peaks under 130 dB SPL @ 1 meter, but so do my Eons and they don’t seem to misbehave when still running “in the green.” I was so unsettled by this quirk of the Turbos that I retired them to moderate-volume-only use – which they are great at, I should mention.

Someone might point out that the Turbosounds could simply dislike my gain structure. I often run powered loudspeakers with the input controls at full-throttle (when it’s practicable), because full-throttle is an easily repeatable setting. Also, I know I can get maximum SPL at around -20 dBFS on my console outputs. I can’t discount the possibility that the M12s fail to handle that kind of use gracefully at the input side, which means that my dislike is user-error. At the same time, though, I have to go back to my JBL Eons; They tolerate being run wide-open without any marked complaint, which is what I expect from a loudspeaker in this price-range.

Milan M12s are good, but they don’t seem to be good enough to spend “more money” on.


What’s Next?

I don’t know, but we’re probably not going to blow the lid off of audio in general.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

I get extremely suspicious when somebody claims to have solved a fundamental problem in audio. Mostly, this is because all the basic gremlins have been thoroughly killed and dried. It’s also because sonic physics is a system of laws that tolerate zero BS. (When somebody claims that they have a breakthrough technology that sounds great by way of spraying sound like a leaky hose, I know they are full of something brown and stinky.)

Modern audio is what I would definitely call a mature technology. In mature technologies, the bedrock science of the technology’s behavior is very well understood. The apparent breakthroughs, then, come when another technology allows a top-shelf behavior to be made available to the masses, or when it creates an opportunity to make a theoretical idea a reality.

A great example is the two-way, fullrange loudspeaker. They’re better than they have ever been. Anyone who remembers wrestling Peavey SP2 TI boxes is almost tearfully grateful to have small, light, loud enclosures available for a rock-bottom price. Obviously, there have been advances. We’ve figured out how to make loudspeaker drivers more efficient and more reliable. Commercially viable neodymium magnets give us the same field strength for less mass. The constant-directivity horn (and its refined derivatives) have delivered improved pattern control.

These are important developments!

Yet, the unit, as an overall object, would be entirely recognizable to someone magically transported to us from three decades in the past. The rules are the same. You’ve got a cone driver in a box, and a compression driver mated to a horn. The cone driver has certain characteristics which the main box has to be built around. It’s not as though we’ve evolved to exotic, crystalline sound-emitters that work by magic.

The palpable improvements aren’t really to do with audio, in a direct sense. They have to do with miniaturization, computerization, and commoditization. An active loudspeaker in the 21st century is likely to sound better than a 1980s or 1990s unit, not because it’s a completely different technology, but because the manufacturer can design, test, tune, and package the product as a bundle of known (and very carefully controlled) quantities. When a manufacturer ships a passive loudspeaker, there’s a lot that they just can’t know – and can’t even do. Stuff everything into the enclosure, and the situation changes dramatically. You know exactly what the amplifier and the driver are going to do to each other. You know just exactly how much excursion that LF driver will endure, and you can limit the amplifier at exactly the point to get maximum performance without damage. You can use steeper crossover slopes to (maybe) cross that HF driver a little lower, improving linearity in the intelligibility zone. You can precisely line up the drivers in time. You can EQ the whole business within an inch of its life.

Again, that’s not because the basic idea got better. It’s because we can put high-speed computation and high-powered amplification in a small space, for (relatively) cheap. Everything I’ve described above has been possible to do for a long time. It’s just that it wasn’t feasible to package it for the masses. You either had to do it externally and expensively, shipping a large, complicated product package to an educated end user…or just let the customer do whatever, and hope for the best.

I can’t say that I have an angle on what the next big jump will be for audio. I’m even skeptical on whether there will be another major leap. I’m excited for more features to become more affordable, though, so I’ll keep looking for those gear catalogs to arrive in the mail.


The Lessons Of El Ridiculoso

Loudspeaker experiments are very educational.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

El Ridiculoso is an idea that’s been bumping around in my head – conceptualized in various morphologies – for years. With the help of the extravagantly cool Mario Caliguiri, who does custom woodworking out here in the high desert, the idea is now incarnated.

Inwoodnated.

Inwoodnated is a real word, because I made it up. All words are made up.

Anyway…

El Ridiculoso is a quad-amped monstrosity meant to go “pretty loud” (but not insanely loud) with 2300 watts of peak input creating about 131 dB of peak, 1 meter SPL. It is very definitely NOT meant to play down low. The conveniently-sized, sealed box for the 15″ driver starts rolling off somewhere around 75 Hz, and really, El Ridiculoso is supposed to be used with subwoofers carrying everything up to 100 Hz anyway. (Sealed boxes are easier to build, and generally pretty forgiving. You can “fudge” the internal volume a bit and still have the whole driver-and-box system work pretty well.)

A few days ago, I got to hook amplifiers up to the boxes and hear them make noise. I found the experience to be rather educational in a few areas.

If You Tune It By Ear You Will Probably Get It Wrong

I set up an X32 mixing console to act as a four-way crossover: You downmix two channels to the main bus, and then send the main bus to matrices 1-4. (The matrices have crossover filters available to them if you have the right firmware upgrade in place.) Because I wouldn’t be working with subwoofers for the test run, I started off by putting the 15’s high-pass at 75 Hz, with the low-pass at 400 Hz. The 12 handled 400 – 1600, the big horn did 1600 – 6400, and the smaller horn took everything above that.

And, of course, I started out by playing music and pushing the different bandpass levels around.

I ended up with an overall sound that was reasonably pleasing, but somewhat tubby (or resonant) at certain bass frequencies. I wondered if the 15’s box was booming for some reason – maybe it was acting like a drum?

In any case, I decided it was time to do some measuring for a real, honest-to-goodness magnitude line-up of the boxes. As I started running sweeps and making adjustments, one thing became VERY clear: Tuning the system by ear had sent me way off course. In some cases, 10+ dB off course. (!)

A Basic Bandpass Magnitude Alignment Fixes A Lot

When you’ve missed the mark as far as I had, information that should blend nicely with other information…doesn’t. You get things like overpowering bass notes, because the crucial midrange just isn’t there to balance it all out. I was actually pretty stunned at just how much better the stack sounded with all the boxes in basically the right place, volume-wise. The music I was playing suddenly started to have the tonal characteristics I’d grown used to from listening at home.

This was without any corrective EQ, which is what I worked on next.

Going through and getting a fine-detail equalization solution certainly changed things, but the difference was not nearly as pronounced as what had happened before. This surprised me as well. I had expected that applying the “make-em-really-flat” solution would result in a massive change in clarity, but really, we were most of the way there already.

Large Horns Make Large Noise

I discovered rather quickly that sitting with my head right up against the 2″ driver-exit horn was unpleasant. The amount of noise that thing can make is impressive. The matrix feed to that bandpass ended up being 12 dB down from everything else, and I still preferred being across the room. I’ve known for years – at an academic level – that 2″ exit compression drivers are used when you need to tear faces off, but this was the first time that I even got a whiff of what they’re really capable of.

Awesome But Impractical

Playing with El Ridiculoso is a great treat, but I can’t imagine getting three more built for regular gigs. For a start, they’re relatively complicated to set up, because all the bandpasses are in separate enclosures…and there are four bandpasses per speaker system. Big-boy loudspeakers might have three bandpasses, but they package them all into a single cabinet. Plus, you usually get one Speakon connector which you can use to mate all your power channels to all your drivers in one click. El Ridiculoso needs four separate connections to work.

Add to that the need for subwoofers in many cases, and now you’ve got a five-way system. Then you have to add all the amplifiers necessary, and all the crossovers/ system management, which results in a pretty hefty drive rack or two. Then you have to add all the speaker cable. You end up spending a lot of money, and a lot of weight, just to make the things work.

And, the only way to get them up in the air is scaffolding, or stacking them on a big pile of subs.

In the end, a compact, ultra-engineered box from a major manufacturer really has the advantage. El Ridiculoso sure does have a lot of “cool factor” as an exotic idea, but a good, solid, self-powered biamp unit will go just about as loud and require far less care and feeding to be day-to-day useful.

This doesn’t mean I’m sad about the experiment. I knew from the beginning that I wasn’t going to design a better mousetrap than every speaker manufacturer on the planet. What I wanted is what I got: A different implementation that I could use to get more hands-on understanding of how these things work.


The Pro-Audio Guide For People Who Know Nothing About Pro-Audio, Part 2

The series continues with a discussion on cable.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

From the article:

“The simplest and most robust connection possible is a single cable carrying analog electrical signals. Analog cabling is subject to many problems, of course, including noise induced by electromagnetic interference. However, its simplicity reduces the number of ways that an outright failure can occur, and the connection tends to degrade “gracefully.””

Read the whole thing for free by visiting Schwilly Family Musicians.


Console Questions

A few simple queries can get you going on just about any console.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

Back when I was in school, we were introduced to “The Four Console Questions.” The idea behind the questions was that, if you walked up to a strange mixer, you could get answers to the questions and be able to get work done. Mixing desks come in many varieties, but there aren’t very many truly different ways to build them that make sense. In any case, all the basic concepts have to essentially stay the same. If a console can’t take some number “a” of audio inputs, and route those inputs to some number “o” of outputs, you don’t have a mixing console anyway.

With the growing commonality of digital mix systems, I feel that the essential “console questions” need some expansion and tweaking. As such, here’s my take on the material that was presented to me over a decade and a half (GEEZE!) ago.

1. Do I Know What I Want To Do?

You might say that this isn’t a console question at all, but in truth, it’s THE most important one. If you don’t know what you want to do with the console, then knowing a bunch of information about the console’s operation won’t help you one iota. The unfortunate reality is that many people try to engage in this whole exercise backwards; They don’t know what they want to accomplish, but they figure that learning the mixer’s whys and wherefores will help them figure it out.

Certainly, learning about a new feature that you haven’t had access to previously can lead you to new techniques. However, at a bedrock level, you have to have some preconceived notion of what you want to accomplish with the tool. Do you want to get a vocal into the FOH PA? Do you want to get three electric guitars, a kazoo, and a capybara playing Tibetan singing-bowls into 12 different monitor mixes?

You have to know your application.

2. How Do I Correctly Clock The Console?

For an analog console, the answer to this is always: “No clock is required.”

For a digital rig, though, it’s very important. I just recently befuddled myself for an agonizing minute with why a digital console wasn’t showing any input. Whoops! It was because I had set it to receive external clock from a master console a few weeks before, and hadn’t returned it to internal clocking now that it was on its own.

You need to know how to indicate to the console which clock source and sample rate is appropriate for the current situation.

3. How Do I Choose What Inputs Are Available To The Channels?

This is particularly important with consoles that support both on-board input and remote stageboxes. You will very likely have to pick and choose which of those options is available to an individual channel or group of channels. What you need to discover is how those selections are accomplished.

4. How Do I Connect A Particular Input To A Particular Channel?

You might think this was covered in the previous question, but it wasn’t. Your global input options aren’t the end of the story. Many consoles will let you do per-channel “soft-patching,” which is the connection of a certain available signal to a certain channel without having to change a physical connection. Whether on a remote stagebox or directly at the desk, input 1 may NOT necessarily be appearing on channel 1. You have to find out how those connections are chosen.

5. How Do I Insert Channel Processing?

In some situations, this means a physical insert connection that may be automatically enabled…or not. In other cases, this means the enabling and disabling of per-channel dynamics and/ or EQ, and maybe even other DSP processing available onboard in some way. You will need to know how that takes place, and with all the possible variations that might have to do with your particular application, it is CRITICAL that you know what you want to do.

6. How Do I Route A Channel To An Auxiliary, Mix Bus, Or The Main Bus?

Sometimes, this is dead-simple and “locked in.” You might have four auxiliaries and four submix buses implemented in hardware, such that they can only be auxiliary or mix buses, with the same knobs always pushing the same aux and a routing matrix with pan-based bus selection. On the other hand, you might have a pool of buses that can behave in various ways depending on global configuration, per-channel configuration, or both.

So, you’ll need to figure out what you’ve got, and how to connect a given channel to a given bus so that you get the results you want.

7. How Do I Insert Bus Processing?

This might be just like question 5, or wildly different. You will need to sort out which reality is currently in play.

8. How Do I Connect A Given Signal To A Physical Output?

Just because you have a signal running to a bus, there’s no guarantee that the bus is actually going to transfer signal to any other piece of equipment. Especially in the digital world, there may be another layer of patching to assign signals to either digital or analog outputs. Bus 1 might be on output 7, because six matrices might be connected to the first six outputs. Maybe output 16 is a pre-fader direct out from channel 4.

You’ll have to figure out where all that gets specified.

Obviously, there’s more to being a whiz at any particular console than eight basic questions. However, if you can get a given signal into the desk, through some processing, combined with other signals you want to combine, and then off to the next destination, you can at least make some real noise in the room.


The Pro-Audio Guide For People Who Know Nothing About Pro-Audio, Part 1

A series I’m starting on Schwilly Family Musicians.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

From the article:

“The fundamental key to all audio production is that we MUST have sound information in the form of electricity. Certain instruments, like synthesizers and sample players don’t produce any actual sound at all; They go straight to producing electricity.

For actual sound, though, we have to perform a conversion, or “transduction.” Transduction, especially input transduction, is THE most important part of audio production. If the conversion from sound to electricity is poor, nothing happening down the line will be able to fully compensate.”

Read the whole thing here, for free!


Retort Report

Responses, and responses to those responses.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

Right after I posted my last article, somebody (that somebody being Jason Knoell of H2Audio) decided to REALLY kick the hornets’ nest and share the link in Stagehand Humor. Of course, I could not help myself: I had to read through the commentary and experience the reactions. The general themes, as well as some particular thoughts make for some excellent, extended discussion of the “Analog VS Digital” topic.

Please note that I’m not identifying individual users here. I don’t feel that it’s necessary, as this site’s main purpose isn’t to host community discussion anyway. (Such discussion is welcome and encouraged, but I don’t host the software to run it.) If you really want to know who said what, the discussion thread is here.

Also, I am 100% aware that I’m not going to change anyone’s opinion. That doesn’t bother me in the least. I’m writing this out so that people who haven’t yet formed their opinion can examine some viewpoints and decide whether or not they think I have a valid take on things.

I will try not to be too snarky, but I can’t make iron-clad promises.

Here we go.

“Have fun when your digital board crashes its operating system and you don’t have the flash drive on the truck.”

People say things like this as though analog consoles have never failed at terrible moments. Big, heavy, hot, expensive, rackmounted power supplies have been known to quit – sometimes spectacularly. Ribbon cables can get unseated. Channel modules can fail with very loud, showstopping results.

Hardware isn’t eternal, no matter its design principle. So…have fun when your analog console has the equivalent of a crash and you can’t fix it with anything as simple and convenient as a flash drive, plus you mightn’t have any spares (especially because analog is big, heavy, and expensive). Me? I effectively have two spare consoles that I carry with me, and the rig still costs less and weighs less than an analog counterpart.

“Say what you will but I can punch, spit, spill a drink, and blow smoke for 16 hours a day on a Mackie 1604 and still have a reliable board!”

An analog console MAY have an advantage in that damage to one section of the unit may not prevent the rest of the device from working. There may also be a period of time where component degradation due to repeated abuse isn’t immediately audible. “Integrated” digital setups tend to either work 100% or cease operating. That difference in failure behavior isn’t enough for me to take a technological leap backwards, however.

And being reasonably nice to a piece of equipment (rather than abusing it) is not as hard as some people might think. Be nice to your digital gear and it will last, as long as there aren’t any manufacturing flaws. Analog follows exactly the same rules, by the way.

“Forgot to address harmonic distortion, saturation of tone, or any acknowledgement of different consoles and their own signature coloration of tonality.”

I didn’t, actually, but I also didn’t go into much detail. I didn’t dig deep because I see running after that kind of thing as a giant waste of limited money and time. I’m not saying that it isn’t nifty when it’s there, if it’s working in your favor. The problem that I have is that the necessary premium to get it is vastly out of proportion to its utility. A console that’s just automatically magic with bass guitars and snare drums is a cool thing. A console that forgives being run hot, potentially in a way that’s even helpful at times, is also pretty rad.

I still maintain, though, that “baked in” signal coloration is basically a design limitation that happens to be fun. I personally prefer a console designed to be flat and clinical, where coloration of all kinds (possibly including distortion, if you’re into that) can be added with explicit intention by the individual operator. If I find I’m missing some sort of magical boost in the low mids – something that very rarely happens, but even so – I can always dial it up with the parametric across my main outputs.

And know EXACTLY what just happened.

You may not prefer that. You may be able to bear the direct and indirect premiums necessary to have an analog signal path that imparts a desired flavor to inputs automatically. That’s great! Don’t let me or anyone else get you down. All I’m saying is that gear which fits your workflow and not mine does not necessarily represent an inherent improvement in technology.

“Guy’s never heard an early 70s concert in a real theater with a real band.”

While I’ve never heard an early 70’s concert in a real theater, I HAVE heard real bands both in and out of various venues that I also consider pretty real.

And I wonder if it’s just possible that the sound of a great band, in a beautiful acoustical environment, playing to an appreciative audience, might represent sonic and experiential factors that are orders upon orders of magnitude more important than any inherent tonality imparted (or not) by the mix rig?

“Bad thing is you spent 20 grand on a console and by the time you were done figuring out all the routing and fx it was obsolete. That’s my biggest problem with digital.”

No, digital consoles are not obsolete the minute you get them. A new model may be waiting in the wings because development cycles are so fast anymore, but it’s not like the console that just got delivered won’t mix bands anymore, or is fatally flawed.

If you want to talk about a long-term ecosystem of support, spares, rental-stock, and add-ons, I can see where you’re coming from – but in all cases, that kind of thing only comes about for the mix rigs that have gotten picked as favorites by the industry at large. Consoles are like pop-stars and rock bands. We remember those that stood the test of time, and conveniently forget that lots of analog and digital offerings didn’t manage to spark, and thus never generated that kind of sustaining ecosystem.

By way of example, I have a pair of Tascam DM24 consoles that sound just fine, and work just fine. I mixed on them for years. They never had the following that the Yamaha 01V series had, though, so I was basically on my own in terms of support and ancillaries. They were “obsolete” even when I got them, in the sense that Yamaha had handily passed them by. So what? They were still powerful tools.

“When analogue peaks, you get “warmth” or natural distortion. When digital peaks, you get clipping and digital breakup.”

Both events being described are an overload. Both are distortion/ clipping. The phenomena on display are not fundamentally different, though the specific tonalities of the events do differ.

My question is: Driving your console’s main bus into clipping isn’t a best practice. Why are you doing it so much that the console’s ability to forgive your gain structure is a main factor in your purchasing decisions?

“Small venues can’t afford digital that doesn’t have latency issues.”

I once built a digital mix system that had a roundtrip latency of about 9 ms, as I recall. That’s really not the best situation…but I used that system for years at Fats, and nobody every complained about it. Mostly, they raved about how great the shows there sounded, both on and off the deck.

My new, non-homebrew rig has a stated latency of about 1 ms. Nobody’s complaining about that either. Latency is a convenient audio boogeyman that gets blamed for all kinds of problems that seem vague or unexplained. It really isn’t as huge a factor as it’s made out to be, and it certainly does not account for all the ills that some folks love to attribute to digital.

“We had a mid level digital board that when pushed just broke up and sounded terrible. We had to boost the processing on the amplifiers and run the mixer as low as possible.
It sounds like the guy writing this article is new to sound and maybe has not used pro equipment!”

If you’re “pushing” any console, analog or digital, in order to drive the PA to full power, your system gain structure is wrong. It’s especially wrong if you’re pushing the console into clip. On the dBFS scale, the region around -20 is the equivalent of “nominal” level on an analog console.

Digital systems, as a rule, do sound horrible when clipped. So, don’t clip them. It’s really not hard to get yourself into the mindset.

I know this stuff because I’m NOT new to audio. I know this stuff because I’ve had hands-on time with consoles that cost everywhere from $50 – tens upon tens of thousands when they were new. I’ve never been bothered by the sound of any of them. I’ve never had a religious experience because of the sound of any of them either. It’s because I’ve used real equipment on real shows (gigs that play to a couple-hundred patrons are VERY real, by the way), and have had to make real purchasing decisions with real money – that is, my own money – that I’ve come to my conclusions.

A case in point is a story that I’ve told several times, in several forms. My schooling was when I had my major, hands-on experience with spendy, large-frame analog desks. Next to the big, premiere, “A Room” was the new “D Room” with a pair of Tascam digital consoles. The two Tascams together were about $6000. The A Room SSL was the high-rent behemoth. Material in both rooms sounded plenty nice. The consoles in the D Room, though, did nearly everything that the SSL could do. They did it more easily, and faster, and for maybe 10% of the cost (if the percentage was even that high).

In real life, convenience, features, and affordability are vastly more important to a console than “It seems to sound super nice under certain circumstances which may not really be the direct result of its technology base.”

“I could make a very good mix on the analogue board in a large venue where not as much fine tuning and adjustment is necessary as you’re not battling stage spill and close proximity to PA as much…The analogue does sound warmer, the mix sounded slightly fuller and for just an acoustic act or a simple band I can still make a mix sound awesome!

Once again – the factors being described here as making a huge difference have nothing to do with the console’s technology base. They are environmental and circumstantial, which hold vastly more sway over the sound of the show.

Also: If you didn’t do exactly the same mix, at exactly the same SPL, of exactly the same band, in exactly the same room, with exactly the same audience, you can’t seriously claim that the analog console was the primary reason it sounded “better.”

“Simple fact– you cannot digitize EVERY bit of sound.”

Simple fact: Yes, you can, and we have been for a long time. Even 44.1 kHz systems reliably capture the entire audible spectrum, and 24-bit converters have dynamic range that (to my knowledge) continues to outperform the analog input stages they are necessarily mated to.

“But it’s better to teach in analog, it requires you actually to listen not just look at a screen.”

Consoles had labeling and useful meters, and signal analysis devices did exist before digital audio was “a thing.” Besides, “listening only” can trick you. Combining your ears and your eyes – and making sure they agree – is a powerful tool for doing better work as an audio human.

“Sacrificing quality for convenience is all it is. If analog is so bad, why are there so many Midas consoles on the road still? Hell, Bonnie Raitt was out with a Gamble console. She sounded better then any digital board could get her to sound.”

No, actually, it’s choosing to have quality, convenience, and features at a great price-point over getting quirks at high expense.

And I’m not saying that analog is bad. I’m saying that it’s not better. There’s a difference.

There are a billion Midas desks on the road because they ARE good consoles. They are consoles that people know how to use, and associate with good sound. I never said they weren’t. I’m also saying that the cost of obtaining, maintaining, and transporting them is hard to justify – unless you’re a touring company or rental house, of course, and everybody who calls you wants one.

And I will close by saying that Bonnie Raitt is a master performer with a killer group of musicians at her side. That matters far more than the console ever could, and I don’t see any way to practically back up the (“handwaved” at best) assertion that her performances would sound any less than brilliant through a digital desk.


No, Analog Isn’t Better

Analog gear does look cool, though.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

Although the fight isn’t nearly so pitched as it once was, some folks might still ask: “Is analog better than digital?”

Analog audio gear does indeed have one major advantage over its number-crunching counterparts. Especially with the right lighting, it often looks a lot cooler on Instagram. Other than that, I’ll take digital over analog any day of the week, and twice on Sunday.

Everyone’s got their own opinion, of course, and I can respect that. I believe that I can back mine up pretty convincingly.

“Back in the day,” you could make a case that analog sounded better. I maintain that this was because both analog and digital grunged up signals to about the same degree, but that digital grunge is generally perceived as being less pleasing. We’re in the 21st Century now, though, and those problems were fixed a good while back. Today’s digital is clear, hyper-accurate, and pristine, even with all manner of gain-changes piled on and low-level signals being passed. Along with that, digital gear is compact, lightweight, flexible, cheap, and feature rich.

Analog, on the other hand, is large, heavy, inflexible, expensive, and feature-limited. It also does not sound “better.”

What do I mean?

Let’s take the example of a modern, digital console, like an X32 Core. Such a console is the ultimate expression of digital’s strengths:

First of all, the setup is tiny. With six rack-spaces handy, you can have 32 X 16 I/O, plus a separate console for FOH and monitor world. Of course, the system has no control surface, so you’ll need a laptop or tablet to act as a “steering wheel.” Even so, the whole shebang could fit in the trunk of a small car. A similar analog setup would necessitate a good-sized SUV, truck, or van for transport.

This also factors into the lightweight aspect. I don’t know exactly how much the above system weighs, but I know it’s a LOT less than two, 32 input analog boards. Even with no other accoutrements, the old-school solution will put you into the 80-pound range at a minimum. Add in a traditional multicore and stagebox splitters, and…well…it’s a lot to carry.

The flexibility argument comes next. Although everything has a design limit, gear that runs on code can have updates applied easily. As long as any new functionality falls within what the hardware and basic software platform can manage, that new functionality can be added – through a simple software update – for as long as the manufacturer cares to work on the system. Front-end control is just as malleable, if not more. If it turns out that the software portion of the interface could do things better, an update gets written and that’s that. Equipment that operates on physical circuits either has no path for similar changes, or if it does, accomplishing the changes is a task that’s profoundly difficult in comparison.

Cost and feature-set dovetail into one another. At the very bare minimum, you can purchase the mixers for a dual-console analog system for about $2800. That’s not too bad in the grand scheme of things, until you realize that a similar investment in the digital world can also get you the stagebox and snake. Also, the digital system will have tons of processing muscle that the analog setup won’t be able to touch. Twelve monitor mixes, fully-configurable channel-per-channel dynamics, four-band parametric EQ, a sweepable filter, EQ and dynamics on every output, plus eight additional processing units? Good luck finding that in an integrated analog package. Such a thing doesn’t even exist as far as I know, and anything even remotely comparable won’t be found for less than tens of thousands of dollars.

So, what about my last point? That analog doesn’t actually sound better?

It doesn’t. No, really. It may sound different. You may like that it sounds different. I can’t argue with personal taste. The reality, though, is that the different sound (especially “warmth” or “fatness” or “depth”) is the product of the gear not passing a clean signal. Maybe the circuitry imparts a nice, low-frequency bump somewhere. Maybe it rolls off in the highs. Maybe there’s just a touch of even-harmonic distortion that creeps in at your preferred gain structure. That’s nifty, but in any objective sense it’s either a circuit that’s inflexibly pre-equalized or is forgiving when being run hard. That may be what some people want, but it’s not what I want, and I’m not going to label it as “better” when a pleasing result is precipitated by a design limitation. (Or only appears when the gain is set just-so.)

Analog isn’t dead, and it isn’t going to die. Our digital systems require well-designed analog stages on the input and output sides to function in real life. At the same time, there are good reasons to make as much of the signal chain digital as is possible. Digital sounds great, and holds too many practical advantages for it to lose out in an objective comparison.


Hitting The Far Seats

A few solutions to the “even coverage” problem, as it relates to distance.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

This article, like the one before it, isn’t really “small venue” in nature. However, I think it’s good to spend time on audio concepts which small-venue folk might still run across. I’m certainly not “big-time,” but I still do the occasional show that involves more people and space. I (like you) really don’t need to get engaged with a detailed discussion regarding an enormous system that I probably won’t ever get my hands on, but the fundamentals of covering the people sitting in the back are still valuable tools.

This article is also very much a follow up to the piece linked above. Via that lens, you can view it as a discussion of what the viable options are for solving the difficulties I ran into.

So…

The way that you get “throw” to the farthest audience members is dependent upon the overall PA deployment strategy you’re using. Deployment strategies are dependent upon the gear in question being appropriate for that strategy, of course; You can’t choose to deploy a bunch of point-source boxes as a line-array and have it work out very well. (Some have tried. Some have thought it was okay. I don’t feel comfortable recommending it.)

Option 1: Single Arrival, “Point Source” Flavor

You can build a tall stack or hang an array with built-in, non-changeable angles, but both cases use the same idea: Any given audience member should really only hear one box (per side) at a time. Getting the kind of directivity necessary for that to be strictly true is quite a challenge at lower frequencies, so the ideal tends to not be reached. Nevertheless, this method remains viable.

I’ve termed this deployment flavor as “single arrival” because all sound essentially originates at the same distance from any given audience member. In other words, all the PA loudspeakers for each “side” are clustered as closely as is practical. The boxes meant to be heard up close are run at a significantly lower level than the boxes meant to cover the far-field. A person standing 50 feet from the stage might be hearing a loudspeaker making 120 dB SPL at 3 feet, whereas the patrons sitting 150 feet away would be hearing a different box – possibly stacked atop the first speaker – making 130 dB SPL at 3 feet. As such, the close-range listener is getting about 96 dB SPL, and the far-field audience member also hears a show at roughly 96 dB SPL.

This solution is relatively simple in some respects, though it requires the capability of “zone” tuning, as well as loudspeakers capable of high-output and high directivity. (You don’t want the up-close audience to get cooked by the loudspeaker that’s making a ton of noise for the long-distance people.)

Option 2: Single Arrival, Line-Array Flavor

As in the point source flavor, you have one array deployed “per side,” with each individual box as close to the other boxes as is achievable. The difference is that an honest-to-goodness line-array is meant to work by the audible combination of multiple loudspeakers. At very close distances, it may be possible to only truly hear a small part of the line, and this does help in keeping the nearby listeners from having their faces ripped off. However, the overall idea is to create a radiation pattern that resembles a section of a cylinder. (Perfect achievement of such a pattern isn’t really feasible.) This is in contrast to point-source systems, where the pattern tends towards a section of a sphere.

As is the case in many areas of life, everything comes down to surface area. A sphere’s surface area is 4*pi*radius^2, whereas the lateral surface area of a cylinder is 2*pi*radius*height. The perceived intensity of sound is the audible radiation spread across the surface area of the radiation geometry. More surface area means less intensity.

To keep the calculations manageable, I’ll have to simplify from sections of shapes to entire shapes. Even so, some comparisons can be made: At a distance of 150 feet, the sound power radiating in a spherical pattern is spread over a surface area of 282,743 square feet. For a 10-foot high cylinder, the surface area is 9424 square feet.

For the sphere, 4 watts of sound power (NOT electrical power!) means that a listener at the 150 foot radius gets a show that’s about 71 dB. For the cylinder, the listener at 100 feet should be getting about 86 dB. At the close-range distance of 50 feet, the cylindrical radiation pattern results in a sound level of 91 dB, whereas a spherical pattern gets 81 dB.

Putting aside for the moment that I’m assuming ideal and mathematically easy conditions, the line-array has a clear advantage in terms of consistency (level difference in the near and far fields) without a lot of work at tuning individual boxes. At the same time, it might not be quite as easily customizable as some point-source configurations, and a real line-source capable of rock-n-roll volume involves a good number of relatively expensive elements. Plus, a real line has to be flown, and with generous trim height as well.

Option 3: Multiple Arrival, Any Flavor

This is otherwise known as “delays.” At some convenient point away from the main PA system, a supplementary PA is set. The signal to that supplementary PA is made to be late, such that the far system aligns pleasingly with the sound from the main system. The hope is that most people will overwhelmingly hear one system over the other.

The point with this solution is to run everything more quietly and more evenly by making sure that no audience member is truly in the deep distance. If each PA only has to cover a distance of 75 feet, then an SPL of 90 dB at that distance requires 118 dB at 3 feet.

The upside to this approach is that the systems don’t have to individually be as powerful, nor do they strictly need to have high-directivity (although it’s quite helpful in keeping the two PA systems separate for the listeners behind the delays). The downside is that it requires more space and more rigging – whether actual rigging or just loudspeakers raised on poles, stacks, or platforms. Additionally, you have to deal with more signal and/ or power runs, possibly in difficult or high-traffic areas. It also requires careful tuning of the delay time to work properly, and even then, being behind or to the side of the delays causes the solution to be invalid. In such a condition where both systems are quite audible, the coherence of the reproduced audio suffers tremendously.

If I end up trying the Gallivan show again, I think I’ll go with delays. I don’t have the logistical resources to handle big, high-output point-source boxes or a real array. I can, on the other hand, find a way to boxes up on sticks with delay applied. I can’t say that I’m happy about the potential coherence issues, but everything in audio is a compromise in some way.


How Could 10 Watts Be Too Loud?

We think audiences want volume, but I’m not sure that’s really true.

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.

Want to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.

I’m not just hammering on players here. The context for this is very much “pro-sound.”

I used to have this regular gig that I loved dearly. Fats Grill is now a hole in the ground, but just a couple of years ago we had live-music every weekend. The PA in the downstairs venue was anything but huge, and yet it was very, very adequate for the space. The mid-highs were mated to an amplifier capable of putting 1000-watt peaks into each box. That works out to a theoretical 127 dB SPL peak for each enclosure – if only at close range (1 meter).

If you were in the middle of the room, you were about 4 meters (or 13-ish feet) away. We’ll say that makes for a practical peak of 115 dB SPL per mid-high, although the room being tightly enclosed would make the real number around 118. Put the two boxes together, and you had a system that could deliver a 121 dB peak in the midrange, plus whatever the subs could do.

Now then.

In pro-audio terms, a 121 dB peak isn’t considered “really loud.” It’s especially not considered loud when you realize that the continuous level, or what humans hear readily, was about 10 dB below that.

But here’s the thing: My experience suggests to me strongly that most folks don’t really want their live-music as loud as “music people” might think. Even for those that love their Rock and/ or Roll, 111 dB continuous can be considered bombardment. This is especially true for the 100 Hz – 15 kHz range. (Subwoofer material is far more easily tolerated, generally speaking.)

At Fats, I very regularly had the system limited so that the top boxes hit a brick wall at their amplifier’s -10 dB point. That’s a peak output of 111 dB in the middle of the audience area, with only about 101 dB of continuous level. That still felt loud for some people. It felt loud for me at times. I wore my earplugs religiously.

To be fair, the PA wasn’t the only thing making noise in the room. The monitor rig and the band’s instrumentation could easily give the total acoustical output a shove that got you into the upper reaches of the 100 dB decade. But even so, you have to realize that 101 dB of continuous system output at room-center resulted from only about 10 watts of continuous input. Remember that I said the limiter for FOH stopped the peaks at 10 dB down. So, that 1000-peak-watt amp was now really only 100 watts maximum, with the continuous power available being 10 dB down from that.

What I’m NOT saying here is that we should all downsize our audio rigs to run on hamster wheels. Headroom (holistic headroom, that is) continues to be a very good idea. There are situations where very large peak-to-continuous ratios have to be handled. What I am saying on balance, though, is that dumping a ton of resources into system capacity that’s actually excess isn’t something I can advise. I just can’t escape this ever-building perception that what a good number of live-music audiences really want are balanced mixes which stay well under an A-weighted level of 100 dB SPL continuous. Add the subwoofer information and you might get to 100 dB or more on another weighting, but that’s a different story.

(And, of course, we have to do what we have to do. Keeping up with a band that’s running hot is a necessity. There were plenty of Fats gigs where I started opening the limiters a little. There was one night where I had to adjust my threshold up to the point where the main amp would show clipping – and then drive hard into that limiting point.)

But there are plenty of gigs that aren’t a slugging match. In those cases, 10 watts of continuous input power might be all that’s actually used. Maybe even less than that. Ten watts can be “too loud” sometimes. I’ve gotten complained at during acoustic shows that people could easily talk over, for goodness sake. I did a few nights at a place with a very nice install that you could barely use in any meaningful way; You would just start pushing some clarity past the monitor wash, and somebody would comment that the music was too loud.

A lot of us aspire to “the big rig,” and I don’t think there’s anything wrong with that on the surface. I simply urge caution. A huge system can be hard to get people to pay for, requires a lot of logistical work, and may be a tremendous amount of excess capacity that never gets leveraged.