Tag Archives: Experimentation

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 Grand Experiment

A plan for an objective comparison of the SM58 to various other “live sound” microphones.

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.

Purpose And Explanation

Ever since The Small Venue Survivalist became a reality, I have wanted to do a big experiment. I’ve been itching to round up a bunch of microphones that can be purchased for either below, or slightly above the price point of the SM58, and then to objectively compare them to an SM58. (The Shure SM58 continues to be an industry standard microphone that is recognized and accepted everywhere as a sound-reinforcement tool.)

The key word above is “objectively.” Finding subjective microphone comparisons isn’t too hard. Sweetwater just put together (in 2017) a massive studio-mic shootout, and it was subjective. That is, the measurement data is audio files that you must listen to. This isn’t a bad thing, and it makes sense for studio mics – what matters most is how the mic sounds to you. Listening tests are everywhere, and they have their place.

In live audio, though, the mic’s sound is only one factor amongst many important variables. Further, these variables can be quantified. Resistance to mechanically-induced noise can be expressed as a decibel number. So can resistance to wind noise. So can feedback rejection. Knowing how different transducers stack up to one another is critical for making good purchasing decisions, and yet this kind of quantitative information just doesn’t seem to be available.

So, it seems that some attempt at compiling such measurements might be helpful.

Planned Experimental Procedure

Measure Proximity Effect

1) Generate a 100Hz tone through a loudspeaker at a repeatable SPL.

2) Place the microphone such that it is pointed directly at the center of the driver producing the tone. The front of the grill should be 6 inches from the loudspeaker baffle.

3) Establish an input level from the microphone, and note the value.

4) Without changing the orientation of the microphone relative to the driver, move the microphone to a point where the front of the grill is 1 inch from the loudspeaker baffle.

5) Note the difference in the input level, relative to the level obtained in step 3.

Assumptions: Microphones with greater resistance to proximity effect will exhibit a smaller level differential. Greater proximity effect resistance is considered desirable.

Establish “Equivalent Gain” For Further Testing

1) Place a monitor loudspeaker on the floor, and position the microphone on a tripod stand. The stand leg nearest the monitor should be at a repeatable distance, at least 1 foot from the monitor enclosure.

2) Set the height of the microphone stand to a repeatable position that would be appropriate for an average-height performer.

3) Changing the height of the microphone as little as possible, point the microphone directly at the center of the monitor.

4) Generate pink-noise through the monitor at a repeatable SPL.

5) Using a meter capable of RMS averaging, establish a -40 dBFS RMS input level.

Measure Mechanical Noise Susceptibility

1) Set the microphone such that it is parallel to the floor.

2) Directly above the point where the microphone grill meets the body, hold a solid, semi-rigid object (like an eraser, or small rubber ball) at a repeatable distance at least 1 inch over the mic.

3) Allow the object to fall and strike the microphone.

4) Note the peak input level created by the strike.

Assumptions: Microphones with greater resistance to mechanically induced noise will exhibit a lower input level. Greater resistance to mechanically induced noise is considered desirable.

Measure Wind Noise Susceptibility

1) Position the microphone on the stand such that it is parallel to the floor.

2) Place a small fan (or other source of airflow which has repeatable windspeed and air displacement volume) 6 inches from the mic’s grill.

3) Activate the fan for 10 seconds. Note the peak input level created.

Assumptions: Microphones with greater resistance to wind noise will exhibit a lower input level. Greater resistance to wind noise is considered desirable.

Measure Feedback Resistance

1) Set the microphone in a working position. For cardioid mics, the rear of the microphone should be pointed directly at the monitor. For supercardioid and hypercardioid mics, the the microphone should be parallel with the floor.

2a) SM58 ONLY: Set a send level to the monitor that is just below noticeable ringing/ feedback.

2b) Use the send level determined in 2a to create loop-gain for the microphone.

3) Set a delay of 1000ms to the monitor.

4) Begin a recording of the mic’s output.

5) Generate a 500ms burst of pink-noise through the monitor. Allow the delayed feedback loop to sound several times.

6) Stop the recording, and make note of the peak level of the first repeat of the loop.

Assumptions: Microphones with greater feedback resistance will exhibit a lower input level on the first repeat. Greater feedback resistance is considered desirable.

Measure Cupping Resistance

1) Mute the send from the microphone to the monitor.

2) Obtain a frequency magnitude measurement of the microphone in the working position, using the monitor as the test audio source.

3) Place a hand around as much of the mic’s windscreen as is possible.

4) Re-run the frequency magnitude measurement.

5) On the “cupped” measurement, note the difference between the highest response peak, and that frequency’s level on the normal measurement.

Assumptions: Microphones with greater cupping resistance will exhibit a smaller level differential between the highest peak of the cupped response and that frequency’s magnitude on the normal trace. Greater cupping resistance is considered desirable.


THD Troubleshooting

I might have discovered something, or I might not.

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.

Over the last little while, I’ve done some shows where I could swear that something strange was going on. Under certain conditions, like with a loud, rich vocal that had nothing else around it, I was sure that I could hear something in FOH distort.

So, I tried soloing up the vocal channel in my phones. Clean as a whistle.

I soloed up the the main mix. That seemed okay.

Well – crap. That meant that the problem was somewhere after the console. Maybe it was the stagebox output, but that seemed unlikely. No…the most likely problem was with a loudspeaker’s drive electronics or transducers. The boxes weren’t being driven into their limiters, though. Maybe a voice coil was just a tiny bit out of true, and rubbing?

Yeesh.

Of course, the very best testing is done “In Situ.” You get exactly the same signal to go through exactly the same gear in exactly the same place. If you’re going to reproduce a problem, that’s your top-shelf bet. Unfortunately, that’s hard to do right in the middle of a show. It’s also hard to do after a show, when Priority One is “get out in a hurry so they can lock the facility behind you.”

Failing that – or, perhaps, in parallel with it – I’m becoming a stronger and stronger believer in objective testing: Experiments where we use sensory equipment other than our ears and brains. Don’t get me wrong! I think ears and brains are powerful tools. They sometimes miss things, however, and don’t natively handle observations in an analytical way. Translating something you hear onto a graph is difficult. Translating a graph into an imagined sonic event tends to be easier. (Sometimes. Maybe. I think.)

This is why I do things like measure the off-axis response of a cupped microphone.

In this case, though, a simple magnitude measurement wasn’t going to do the job. What I really needed was distortion-per-frequency. Room EQ Wizard will do that, so I fired up my software, plugged in my Turbos (one at a time), and ran some trials. I did a set of measurements at a lower volume, which I discarded in favor of traces captured at a higher SPL. If something was going to go wrong, I wanted to give it a fighting chance of going wrong.

Here’s what I got out of the software, which plotted the magnitude curve and the THD curve for each loudspeaker unit:

I expected to see at least one box exhibit a bit of misbehavior which would dramatically affect the graph, but that’s not what I got. What I can say is that the first measurement’s overall distortion curve is different, lacking the THD “dip” at 200 Hz that the other boxes exhibit, significantly more distortion in the “ultra-deep” LF range, and with the “hump” shifted downwards. (The three more similar boxes center that bump in distortion at 1.2 kHz. The odd one out seems to put the center at about 800 Hz.)

So, maybe the box that’s a little different is my culprit. That’s my strong suspicion, anyway.

Or maybe it’s just fine.

Hmmmmm…


How Much Light For Your Dollar?

Measurements and observations regarding a handful of relatively inexpensive LED PARs.

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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I’m in the process of getting ready for a pretty special show. The album “Clarity” by Sons Of Nothing is turning 10, and a number of us are trying to put together one smasher of a party.

Of course, that means video.

And our master of all things videographic is concerned about having enough light. We can’t have anybody in the band who’s permanently stuck in “shadow.” You only get one chance to shoot a 10th anniversary concert, and we want to get it right.

As such, I’m looking at how to beef up my available lighting instruments. It’s been a long while since I’ve truly gone shopping for that old mainstay of small-venue lighting, the LED wash PAR, but I do take a look around every so often. There’s a lot to see, and most of it isn’t very well documented. Lighting manufacturers love to tell you how many diodes are in a luminaire, and they also like to tell you how much power the thing consumes, but there appears to be something of an allergy to coughing up output numbers.

Lux, that is. Lumens per square meter. The actual effectiveness of a light at…you know…LIGHTING things.

So, I thought to myself, “Self, wouldn’t it be interesting to buy some inexpensive lights and make an attempt at some objective measurement?”

I agreed with myself. I especially agreed because Android 4.4 devices can run a cool little Google App called “Science Journal.” The software translates the output from a phone’s ambient light sensor into units of lux. For free (plus the cost of the phone, of course). Neat!

I got onto Amazon, found myself a lighting brand (GBGS) that had numerous fixtures available for fulfillment by Amazon, and spent a few dollars. The reason for choosing fulfillment from Amazon basically comes down to this: I wanted to avoid dealing with an unknown in terms of shipping time. Small vendors can sometimes take a while to pack and ship an order. Amazon, on the other hand, is fast.

The Experiment

Step 1: Find a hallway that can be made as dark as possible – ideally, dark enough that a light meter registers 0 lux.

Step 2: At one end, put the light meter on a stand. (A mic stand with a friction clip is actually pretty good at holding a smartphone, by the way.)

Step 3: At the other end, situate a lighting stand with the “fixture under test” clamped firmly to that stand.

Step 4: Measure the distance from the lighting stand to the light meter position. (In my case, the distance was 19 feet.)

Step 5: Darken the hallway.

Step 6: Set the fixture under test to maximum output using a DMX controller.

Step 7: Allow the fixture to operate at full power for roughly 10 minutes, in case light output is reduced as the fixture’s heat increases.

Step 8: Ensure the fixture under test is aimed directly at the light meter.

Step 9: Note the value indicated by the meter.

Important Notes

A relatively long distance between the light and the meter is recommended. This is so that any positioning variance introduced by placing and replacing either the lights or the meter has a reduced effect. At close range, a small variance in distance can skew a measurement noticeably. At longer distances, that same variance value has almost no effect. A four-inch length difference at 19 feet is about a 2% error, whereas that same length differential at 3 feet is an 11% error.

It’s important to note that the hallway used for the measurement had white walls. This may have pushed the readings higher, as – similarly to audio – energy that would otherwise be lost to absorption is re-emitted and potentially measurable.

It was somewhat difficult to get a “steady” measurement using the phone as a meter. As such, I have estimated lux readings that are slightly lower than the peak numbers I observed.

These fixtures may or may not be suitable for your application. These tests cannot meaningfully speak to durability, reliability, acceptability in a given setting, and so on.

The calculation for 1 meter lux was as follows:

19′ = 5.7912 m

5.7912 = 2^2.53 (2.53 doublings of distance from 1m)

Assumed inverse square law for intensity; For each doubling of distance, intensity quadruples.

Multiply 19′ lux by 4^2.53 (33.53)

Calculated 1 meter lux values are just that – calculated. LED PAR lights are not a point-source of light, and so do not behave like one. It requires a certain distance from the fixture for all the emitters to combine and appear as though they are a single source of light.

The Data

The data display requires Javascript to work. I’m sorry about that – I personally dislike it when sites can’t display content without Javascript. However, for the moment I’m backed into a corner by the way that WordPress works with PHP, so Javascript it is.


Ascending sort by:


Knowledge VS Wisdom

You can know the terminology and not know what you’re doing.

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.

knowledge-and-wisdomWant to use this image for something else? Great! Click it for the link to a high-res or resolution-independent version.
Regarding the above picture, I got the Kanji for knowledge and wisdom from Google. At least, I think that’s what I got. The above might actually read, “Your mother is a recalcitrant platypus,” but I wouldn’t know.

Anyway…

I’m pretty confident that any average, adult human can be taught in less than one hour everything necessary to operate a parametric EQ.

I did not say that they would operate it well, or appropriately. Throwing them onto a stage-side monitor console to run rock-n-roll wedges in real time would probably be a very poor move. The process of you getting fired would be legendary.

The problem isn’t knowledge, especially now that many of us carry devices capable of accessing vast reserves of information by way of wireless data. The problem is wisdom born of experience. The way that you get good at wielding all manner of EQ implementations against all manner of audio goblins is by wielding EQ against audio goblins. There’s no substitute for it. Encountering problems, making changes, and hearing the results of those changes immediately is how learning takes place.

I do urge people to learn the vocabulary and the concepts. I wouldn’t spend so much time pushing math, science, and applied audio nerdery on this site if I felt differently. Knowing the words and the numbers allows you to do (at least) two things: First, you can put names and values on both problems and solutions, and second, with that ability you can then ask better questions. The heart of all engineering – that is, the application of mathematical, scientific, and logical processes to the solving of puzzles – is the asking and answering of a series of questions. The questions can be abstract or concrete. They can be theoretical, or relating to something happening in the here and now. The circumstances hardly matter; Better questions return better answers.

But if all you do is memorize the words and ideas without getting your “fubs” on them in real life, your application of the concepts will be stunted. It’s a bit like this gem from Tom Roche, written down on Pro Sound Web’s LAB Basement Forum: “As I understand it, knowledge is knowing that a tomato is a fruit. Wisdom is not putting it in a fruit salad.”

There are lots of guys and gals out there who know everything there is to know about metaphorical tomatoes, and yet make terrible, metaphorical fruit salads. Folks who can name every knob and switch on a console, yet seemingly can’t hear that their mix is all drums and barely any vocal. People who talk about audio concepts using words that you definitely recognize, yet it’s all strung together in ways that are nonsensical. Craftspersons who memorized a workflow without knowing why it worked in the first place, and who get completely wrecked when the situation calls for something different.

More knowledge is good, but there’s a point where you can no longer read your way through problem solving. At some point, the book has to be set down and some knobs turned. That’s where most of the fun is, anyway.


Double Hung Discussion

It’s not magic, and it may not be for you. It works for me, 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.

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

On the heels of my last article, it came to may attention that some folks were – shall we say – perplexed about my whole “double hung” PA deployment. As can be the case, I didn’t really go into any nuance about why I did it, or what I expected to get out of it. This lead some folks to feel that it was a really bizarre way to go about things, especially when a simpler solution might have been a better option.

The observations I became aware of are appropriate and astute, so I think it’s worth talking about them.

Why Do It At All?

First, we can start with that logistics thing again.

When I put my current system together, I had to consider what I was wanting to do. My desire was to have a compact, modular, and flexible rig that could “degrade gracefully” in the event of a problem. I also had no desire to compete with the varsity-level concert systems around town. To do so would have required an enormous investment in both gear and transport, one that I was unwilling (and unable) to make.

What I’ve ended up with, then, is a number of smaller boxes. If I need more raw output, I can arrange them so that they’re all hitting the same general area. I also have the option of deploying for a much wider area, but with reduced total output capability. I wouldn’t have that same set of options with a small number of larger, louder enclosures.

That’s the basic force behind why I have the rig that I have. Next come the more direct and immediate issues.

The first thing is just a practical consideration: Because my transport vehicle isn’t particularly large, I don’t really have the necessary packing options required to “leave gear on the truck.” If I’m getting the rig out, I might as well get all of it out. This leads to a situation where I figure that I might as well find a way to deploy everything all the time. The gear is meant to make noise, not sit around. “Double hung” lets me do that in a way that makes theoretical sense (I’ll say more on why in a bit).

The second reason is less practical. I have a bit of a penchant for the unconventional and off-the-wall. I sometimes enjoy experiments for the sake of doing them, and running a double hung system is just that kind of thing. I like doing it to find out what it’s like to do it.

Running double hung is NOT, by any means, more practical than other deployments. Especially if you’re new to this whole noise-louderization job, going with this setup is NOT some sort of magical band-aid that is going to fix your sound problems. Also, if you’re getting good results with a much simpler way of doing things, going to the extra trouble very well may not be worth it.

At the same time, though, the reality of making this kind of deployment happen is not really all that complicated. You can do it very easily by connecting one pair to the left side of your main mix, and the other pair to the right side. Then, you just pan to one side or the other as you desire.

System Output And Response

Up above, I mentioned that running my system as a double hung made sense in terms of audio theory. Here’s the explanation as to why. It’s a bit involved, but stick with me.

I haven’t actually measured the maximum output of my FOH mid-highs, but Turbosound claims that they’ll each make a 128 dB SPL peak. I’m assuming that’s at 1 meter, and an instantaneous value. As such, my best guess at their maximum continuous performance, run hard into their limiters, would be 118 dB SPL at 1 meter.

If I run them all together as one large rig, most people will probably NOT hear the various boxes sum coherently. So, the incoherent SPL addition formula is what’s necessary: 10 Log10[10^(dB SPL/ 10) + 10^(dB SPL/ 10)…]. What I put into Wolfram Alpha is 10 Log10[10^11.8 + 10^11.8 + 10^11.8 + 10^11.8].

What I get out is a theoretical, total continuous system output of 124 dB SPL at 1 meter, ignoring any contribution from the subwoofers.

At this point, you would be quite right to say that I can supposedly get to that number in one of two ways. The first, simple way, is to just put everything into all four boxes. The second, not simple way is to put some things in some boxes and not in others. Either way, the total summed sound pressure should be basically the same. The math doesn’t care about the per-box content. So, why not just do it simply?

Because there’s more to life than just simply getting to the maximum system output level.

By necessity of there being physical space required for the speakers to occupy, the outer pair of enclosures simply can’t create a signal that arrives at precisely the same moment as the signal from the inner pair, as far as the majority of the audience can perceive. Placed close together, the path-length differential between an inner box and an outer box is about 0.0762 meters, or 3 inches.

That doesn’t seem so bad. The speed of sound is about 343 meters/ second in air, so 0.0762 meters is 0.22 ms of delay. That also doesn’t seem so bad…

…until you realize that 0.22 ms is the 1/2 cycle time of 2272 Hz. With the outer boxes being 1/2 cycle late, 2272 Hz would null (as would other frequencies with the same phase relationship). If everything started as measuring perfectly flat, introducing that timing difference into a rig with multiple boxes producing the same material would result in this transfer function:

combfiltering0.22ms

Of course, everything does NOT start out as being perfectly flat, so that craziness is added onto whatever other craziness is already occurring. For most of the audience, plenty of phase weirdness is going on from any PA deployed as two, spaced “stacks” anyway. To put it succinctly, running everything everywhere results in even more giant holes being dug into the critical-for-intelligibility range than were there before.

Running double hung, where the different pairs of boxes produce different sounds, prevents the above problem from happening.

So, when I said that I was running double hung for “clarity,” I was not doing it to fix an existing clarity problem. I was preventing a clarity problem from manifesting itself.

Running absolutely everything into every mid-high, and then having all those mid-highs combine is a simple way to make a system’s mid-highs louder. It’s also a recipe for all kinds of weird phase interactions. These interactions can be used intelligently (in an honest-to-goodness line-array, for instance), but for most of us, they actually make life more difficult. Louder is not necessarily better.

More On Output – Enough Rig For The Gig?

For some folks reading my previous installment, there was real concern that I hadn’t brought enough PA. They took a gander at the compactness of the rig, and said, “There’s no way that’s going to get big-time sound throughout that entire park.”

The people with that concern are entirely correct.

But “rock and roll level everywhere” was not at all what I was trying to do.

The Raw Numbers

What I’ve found is that many people do NOT actually want everything to be “rock and roll” loud over every square inch of an event area. What a good number of events actually want is a comfortable volume up close, with an ability to get away from the noise for the folks who aren’t 100% interested. With this being the case, investing in a system that can be clearly heard at a distance of one mile really isn’t worthwhile for me. (Like I said, I’m not trying to compete with a varsity-level sound company.)

Instead, what I do is to deploy a rig that’s in close proximity to the folks who do want to listen, while less interested people are at a distance. Because the folks who want more volume are closer to the PA, the PA doesn’t have to have crushing output overall. For me, the 110 dB SPL neighborhood is plenty loud, and I can do that for the folks nearby – by virtue of them being nearby.

Big systems that have to cover large areas often have the opposite situation to deal with: The distance differential between the front row and the back row can actually be smaller, although the front row is farther away from the stacks in an absolute sense. With my rig, the people up close are probably about three meters from the PA. The folks far away (who, again, aren’t really interested) might be 50 meters away. That’s more than a 16-fold difference. At a bigger show, there might be a barricade that’s 10 meters from the PA, with the main audience extending out to 100 meters. That’s a much bigger potential audience, but the difference in path lengths to the PA is only 10-fold.

Assuming that the apparent level of the show drops 6 dB for every doubling of distance, my small show loses about 24 decibels from the front row to the folks milling around at 50 meters. The big show, on the other hand, loses about 20 dB. (But they have to “start” much louder.)

That is, where the rubber hits the road is how much output each rig needs at 1 meter. At the big show, they might want to put 120 dB SPL into the front seats. To do that, the level at 1 meter has to be 140 dB. That takes a big, powerful PA. The folks in the back are getting 100 dB, assuming that delays aren’t coming into the picture.

For me to do a show that’s 110 dB for the front row, my PA has to produce about 119 dB at 1 meter. That’s right about what I would expect my compact setup to be able to do, with a small sliver of headroom. At 50 meters, my show has decayed to a still audible (but not “rock show loud”) 86 dB SPL.

That’s what I can do, and I’ve decided to be happy with it – because the folks I work with are likely to be just as happy with that as I am. People don’t hire me to cover stadiums or have chest-collapsing bass. They hire me because they know I’ll do everything in my power to get a balanced mix at “just enough” volume.

The Specifics Of The Show

Ultimately, the real brass tacks are to be found in what the show actually needed.

The show did not need 110 dB SPL anywhere. It needed a PA that sounded decent at a moderate volume.

The genre was folksy, indie material. A 110 dB level would have been thoroughly inappropriate overkill. At FOH control, the show was about 80 – 90 dB, and that was plenty. There were a few times where I was concerned that I might have been a touch too loud for what was going on. In that sense, I had far more than enough PA for raw output. I could have run a single pair of boxes and been just fine, but I didn’t want to get all the speakers out of the van and not use them. As I said before, I chose “double hung” to use all my boxes, and to use them in the way that would be nicest for people’s ears.


If you’re curious about running a double hung setup, I do encourage you to experiment with it. Curiosity is what keeps this industry moving. At the same time, you shouldn’t expect it to completely knock you off your feet. If you have a good-sounding system that runs everything through one pair of mains, adding another pair just to split out some sources is unlikely to cause a cloud-parting, ligh-ray-beaming experience of religious proportions. Somewhat like aux-fed subwoofers, going double hung is a taste-dependent route to accomplishing reinforcement for a live event. For me, it solves a particular problem that is mostly logistical in nature, and it sounds decent doing it.


Buzzkill

Ridding yourself of hum and buzz is like all other troubleshooting: You have to isolate the problem to fix 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.

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Not all hums and buzzes are equally bad. Honeybees hum and buzz, but they’re super-helpful creatures that are generally interested in being left alone and making honey. Wasps, like the one pictured above, are aggressive jerks.

Of course, this site isn’t about insects. It’s about audio, where hum and buzz mean problems. Unwanted noise. Blech.

I recently got an email from a friend who wanted to know how to de-buzzify (I just made that word up) a powered mixer. When you mercilessly distill what I told him, you come up with a basic truth that covers all of troubleshooting:

The probability of an effective fix for a problem is directly proportional to your ability to isolate the problem.

Solitude

The importance of finding the exact location of a fault is something that I don’t believe I can overemphasize. It’s the key to all the problem-solving I’ve ever had to do. It doesn’t matter if the problem is related to audio signal flow, car trouble, or computer programming; if you can actually nail down the location of the problem, you’ve got a real shot at an effective (and elegant) fix.

The reverse is also true. The less able you are to pinpoint your conundrum’s place of residence, the more likely you are to end up doing surgery with a sledgehammer. If you can’t zero-in on a root cause, you end up “fixing” a certain amount of things that aren’t actually being troublesome. The good news is that you can usually take an iterative approach. All problems begin with “this system isn’t working as I expected,” which is a completely non-specific view – but they don’t have to end there. The key is to progressively determine whether each interrelated part of the system is contributing to the issue or not. There are lots of ways to do this, but all the possible methods are essentially an expression of one question:

“Is the output of this part of the system what I expect it to be?”

So…here’s a way to apply this to buzz and hum problems.

Desperately Seeking Silence

Talking in depth about the exact electrical whys and wherefores surrounding strange and unwanted noises is a little bit beyond my experience. At a general level, though, the terminology of “ground loop” provides a major clue. Voltage that should be taking a direct path to ground is instead taking a “looping” or “circuitous” path. A common cause of this is equipment receiving mains (“wall”) power from two different circuits, where each path to mains ground has a significantly different impedance. There is now a voltage potential between the two pieces of gear.

Bzzzzzzzz….

You can also have a situation where two device’s audio grounds are interconnected such that there is a potential between the two devices.

Hmmmmmmzzzzzzzz…

Anyway.

The first thing to do is to decide what piece of equipment you’re testing against. Maybe it’s a mixing console. Maybe it’s an amplifier. Whatever it is, you are asking the question from before:

“Is the output of this part of the system what I expect it to be?”

Or, more specifically…

“I expect this device’s output to be quiet, unless an audio signal is present. Is that the case?”

To answer that question, you need isolation.


WARNING: At NO point should you do anything to disconnect the mains-power/ safety grounds from your equipment. It’s there to prevent you from dying if the equipment chassis should become energized. In fact, as a start, try to verify that the mains-power sockets you are using actually DO provide a connection to “earth.” If they don’t, stop using them until they’re fixed. You may even find that your noise problem goes away.


To get isolation, start by disconnecting as much as you possibly can from the DUT (the Device Under Test). Of course, you’ve got to have some kind of way to monitor the output, so that might mean that you can’t disconnect everything. As much as possible, try to ensure that all mains-power grounds offer the same impedance – if it must stay connected, and it requires mains power, get all the power to connect to the same socket. A multi-outlet power tap can come in handy for this.

Is the output what you expect?

If yes, then something which was connected to your DUT’s input has a good chance of being the problem. At this point, if possible, treat each potential culprit as a secondary DUT in turn. If feasible, connect each suspect directly to your monitoring solution. If the ground loop manifests itself, and the suspect device requires mains power, try getting power from the same tap that the primary DUT is on. If the loop goes away, you’ve established that the two devices in play were likely having an “unequal impedance to ground” problem. If the loop stays in effect, you can jump back up to the beginning of this process and try again, but with the gear you had just plugged in as the new, primary DUT. You can keep doing this, “moving up the stack” of things to test until you finally isolate the piece of gear that’s being evil. (IMPORTANT: Any piece of the chain could be your problem source. This includes cables. You may need to pack a lunch if you have a lot of potential loop-causers to go through.)

If you can’t get the buzz to manifest when adding things back one at a time, then you might have a multi-device interaction. If possible, work through every possible combination of input connections until you get your noise to happen.

But what if the output on the original DUT was NOT what you expected, even with everything pulled off the output side?

At that point, you know that an input device isn’t the source of your trouble with this particular DUT. This is good – your problem is becoming isolated to a smaller and smaller pool of possibilities.

Try to find an alternate way to connect to your monitoring solution, like a different cable. If the problem goes away, that locates the cable as the menace. If you’re switching the connection, and the noise remains with no audio path, then the monitoring system has the problem and you need to restart with a new DUT. (If you’ve got a mixer connected to an amp and a speaker, and a ground loop stays audible when the mixer-to-amp connection is broken, then the amp is your noise source.)

If you’ve tried all that and you still have the buzz, it’s time to try a different circuit. Get as far away from the original mains-power socket as you can, and reproduce the minimal setup. If the ground-loop goes away, then you may have a site-wiring issue that’s local to the original socket(s). If the problem doesn’t go away, it’s time to take a field-trip to another building. It’s possible to have a site-wide electrical problem.

If the loop still won’t resolve, it’s very likely that your DUT has an internal fault that needs attention. Whether that means repair or replace is an exercise left to the reader.

Hopefully, you don’t get to that point – but you won’t figure out if you ARE at that point unless you can isolate your problem.


The Sublime Beauty Of Cheap, Old, Dinged-Up Gear

Some things can be used, and used hard, without worry.

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 really do think that classy gear is a good idea in the general case. I think it sends a very important signal when a band walks into a room, and their overwhelming impression is that of equipment which is well-maintained and worth a couple of dollars. When a room is filled with boxes and bits that all look like they’re about to fail, the gigs in that room stand a good chance of being trouble-filled. In that case, musician anxiety is completely justified.

In the past, I have made updates to gear almost purely for the sake of “politics.” I don’t regret it.

At the same time, though, “new n’ shiny” equipment isn’t a guarantee of success. I’ve had new gear that developed problems very quickly, but more than that, new and spendy gear tends to make you ginger (in the timid sense). You can end up being so worried about something getting scratched up or de-spec’d that you forget the purpose of the device: It’s there to be used.

And that’s where the sublime beauty of inexpensive, well-worn equipment comes in. You’ve found a hidden gem, used it successfully in the past, will probably keep using it successfully in the future, and you can even abuse it a bit in the name of experimentation.

Case Study: Regular Kick Mics Are Boring

I’ve used spendy kick mics, and I’ve used cheap kick mics. They’ve all sounded pretty okay. The spendy ones are pre-tuned to sound more impressive, and that’s cool enough.

…but, you know, I find the whole “kick mic” thing to be kinda boring. It’s all just a bunch of iteration or imitation on making a large-diaphragm dynamic. Different mics do, of course, exhibit different flavors, but there’s a point where it all seems pretty generic. It doesn’t help that folks are so “conditioned” by that generic-ness – that is, if it doesn’t LOOK like a kick mic, it can’t be any good. (And, if it doesn’t COST like a kick mic, it can’t be any good.)

I once had a player inquire after a transducer I used on his bass drum. He seemed pretty interested in it based on how it worked during the show, and wanted to know how expensive it was. I told him, and he was totally turned OFF…by the mic NOT costing $200. He stated, “I’m only interested in expensive mics,” and in my head, I’m going, “Why? This one did a good enough job that you started asking questions about it. Doesn’t that tell you something?”

Anyway, the homogeneity of contemporary kick mic-ery is just getting dull for me. It’s like how modern car manufacturers are terrified to “color outside the lines” with any consumer model.

To get un-bored, I’ve started doing things that expose the greatness of “cheap, old, and dinged up.” In the past, I tried (and generally enjoyed) using a Behringer ECM8000 for bass drum duty. Mine was from back when they were only $40, had been used quite a bit, and had been dropped a few times. This was not a pristine, hardwood-cased, ultra-precision measurement mic that would be a real bear to replace. It was a knock-around unit that I had gotten my money out of, so if my experiment killed it I would not be enduring a tragedy.

And it really worked. Its small diameter made it easy to maneuver inside kick ports, and its long body made it easy to get a good ways inside those same kick ports. The omni pattern had its downsides, certainly. Getting the drum to the point of being “stupid loud” in FOH or the drumfill wasn’t going to happen, but that’s pretty rare for me. At an academic level, I’m sure the tiny diaphragm had no trouble reacting quickly to transients, although it’s not like I noticed anything dramatic. Mostly, the mic “sounded like a drum to me” without having to be exactly like every other bass-drum mic you’re likely to find. The point was to see if it could work, and it definitely did.

My current “thing” bears a certain similarity, only on the other end of the condenser spectrum. I have an old, very beat-up MXL 990 LDC, which I got when they were $20 cheaper. I thought to myself, “I wonder what happens if I get a bar-towel and toss this in a kick drum?” What I found out is that it works very nicely. The mic does seem to lightly distort, but the distortion is sorta nifty. I’m also freed from being required to use a stand. The 990 might die from this someday, but it’s held up well so far. Plus, again, it was cheap, already well used, and definitely not in pristine condition. I don’t have to worry about it.

Inoculation Against Worry Makes You Nicer

Obviously, an unworried relationship with your gear is good for you, but it’s also good in a political sense. Consternation over having a precious and unblemished item potentially damaged can make you jumpy and unpleasant to be around. There are folks who are so touchy about their rigs that you wonder how they can get any work done.

Of course, an overall attitude of “this stuff is meant to be used” is needed. Live-audio is a rough and tumble affair, and some things that you’ve invested in just aren’t going to make it out alive. Knowing this about everything, from the really expensive bits to the $20 mic that’s surprisingly brilliant, helps you to maintain perspective and calmness.

The thing with affordable equipment (that you’ve managed to hold on to and really use) is that it feeds this attitude. You don’t have to panic about it being scuffed up, dropped, misplaced, or finally going out with a bang. As such, you can be calm with people. You don’t have to jump down someone’s throat if they’re careless, or if there’s a genuine accident. It’s easy to see that the stuff is just stuff, and while recklessness isn’t a great idea, everything that has a beginning also has an end. If you got your money out of a piece of equipment, you can just shrug and say that it had a good life.

Have some nice gear around, especially for the purpose of public-relations, but don’t forget to keep some toys that you can “leave out in the rain.” Those can be the most fun.


Why I Think Steam Machines Are Cool

My audio-human mind races when thinking of high-performance, compact, affordable machines.

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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“Wait,” you’re thinking, “I thought this site was about live shows. Steam Machines are gaming devices.”

You’re right about that. What you have to remember (or just become aware of), is that I have a strange sort of DIY streak. It’s why I assembled my own live-audio console from “off the shelf” products. I really, really, REALLY like the idea of doing powerful things with concert sound via unorthodox means. An unorthodox idea that keeps bubbling up in my head is that of a hyper-customizable, hyper-expandable audio mix rig. It could be pretty much any size a user wanted, using pretty much whatever audio hardware a user wanted, and grow as needed. Also, it wouldn’t be too expensive. (About $900 per 16X16 channel “block.”)

When I look at the basic idea of the Valve Steam Machine, I see a device that has the potential to be a core part of the implementation.

But let’s be careful: I’m not saying that Steam Machines can do what I want right now. I’m not saying that there aren’t major pitfalls, or even dealbreakers to be encountered. I fully expect that there are enormous problems to solve. Just the question of how each machine’s audio processing could be conveniently user-controlled is definitely non-trivial. I’m just saying that a possibility is there.

Why is that possibility there?

The Box Is Prebuilt

The thing with prebuilt devices is that it’s easier for them to be small. A manufacturer building a large number of units can get custom parts that support a compact form factor, put it all together, and then ship it to you.

Of course, when it comes to PCs, you can certainly assemble a small-box rig by hand. However, when we’re talking about using multiple machines, the appeal of hand-building multiple boxes drops rapidly. So, it’s a pretty nice idea that a compact but high(er) performance computing device can be gotten for little effort.

The System Is Meant For Gaming

Gaming might seem like mere frivolity, but these days, it’s a high-performance activity. We normally think of that high-performance as being located primarily in the graphics subsystem – and for good reason. However, I also think a game-capable system could be great for audio. I have this notion because games are so reliant on audio behaving well.

Take a game like a modern shooter. A lot of stuff is going on: Enemy AI, calculation of where bullets should go, tracking of who’s shooting at who, collision detection, input management, the knowing of where all the players are and where they’re going, and so on. Along with that, the sound has to work correctly. When anybody pulls a trigger, a sound with appropriate gain and filtering has to play. That sound also has to play at exactly the right time. It’s not enough for it to just happen arbitrarily after the “calling” event occurs. Well-timed sounds have to play for almost anything that happens. A player walks around, or a projectile strikes an object, or a vehicle moves, or a player contacts some phsyics-enabled entity, or…

You get the idea.

My notion is that, if the hardware and OS of a Steam Machine are already geared specifically to make this kind of thing happen, then getting pro-audio to work similarly isn’t a totally alien application. It might not be directly supported, of course, but at least the basic device itself isn’t in the way.

The System Is Customizable

My understanding of Steam Machines is that they’re meant to be pretty open and “user hackable.” This excites me because of the potential for re-purposing. Maybe an off-the-shelf Steam Machine doesn’t play nicely with pro-audio hardware? Okay…maybe there’s a way to take the box’s good foundation and rebuild the upper layers. In theory, a whole other OS could be runnable on one of these computers, and a troublesome piece of hardware might be replaceable (or just plain removable).


I acknowledge that all of this is off in the “weird and theoretical” range. My wider goal in pointing it out is to say that, sometimes, you can grab a thing that was intended for a different application and put it to work on an interesting task. The most necessary component seems to be imagination.


Where’s Your Data?

I don’t think audio-humans are skeptical enough.

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.

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

If I’m going to editorialize on this, I first need to be clear about one thing: I’m not against certain things being taken on faith. There are plenty of assumptions in my life that can’t be empirically tested. I don’t have a problem with that in any way. I subscribe quite strongly to that old saw:

You ARE entitled to your opinion. You ARE NOT entitled to your own set of “facts.”

But, of course, that means that I subscribe to both sides of it. As I’ve gotten farther and farther along in the show-production craft, especially the audio part, I’ve gotten more and more dismayed with how opinion is used in place of fact. I’ve found myself getting more and more “riled” with discussions where all kinds of assertions are used as conversational currency, unbacked by any visible, objective defense. People claim something, and I want to shout, “Where’s your data, dude? Back that up. Defend your answer!”

I would say that part of the problem lies in how we describe the job. We have (or at least had) the tendency to say, “It’s a mix of art and science.” Unfortunately, my impression is that this has come to be a sort of handwaving of the science part. “Oh…the nuts and bolts of how things work aren’t all that important. If you’re pleased with the results, then you’re okay.” While this is a fair statement on the grounds of having reached a workable endpoint through unorthodox or uneducated means, I worry about the disservice it does to the craft when it’s overapplied.

To be brutally frank, I wish the “mix of art and science” thing would go away. I would replace it with, “What we’re doing is science in the service of art.”

Everything that an audio human does or encounters is precipitated by physics – and not “exotic” physics, either. We’re talking about Newtonian interactions and well-understood electronics here, not quantum entanglement, subatomic particles, and speeds approaching that of light. The processes that cause sound stuff to happen are entirely understandable, wieldable, and measurable by ordinary humans – and this means that audio is not any sort of arcane magic. A show’s audio coming off well or poorly always has a logical explanation, even if that explanation is obscure at the time.

I Should Be Able To Measure It

Here’s where the rubber truly meets the road on all this.

There seems to be a very small number of audio humans who are willing to do any actual science. That is to say, investigating something in such a way as to get objective, quantitative data. This causes huge problems with troubleshooting, consulting, and system building. All manner of rabbit trails may be followed while trying to fix something, and all manner of moneys are spent in the process, but the problem stays un-fixed. Our enormous pool of myth, legend, and hearsay seems to be great for swatting at symptoms, but it’s not so hot for tracking down the root cause of what’s ailing us.

Part of our problem – I include myself because I AM susceptible – is that listening is easy and measuring is hard. Or, rather, scientific measuring is hard.

Listening tests of all kinds are ubiquitous in this business. They’re easy to do, because they aren’t demanding in terms of setup or parameter control. You try to get your levels matched, setup some fast signal switching, maybe (if you’re very lucky) make it all double-blind so that nobody knows what switch setting corresponds to a particular signal, and go for it.

Direct observation via the senses has been used in science for a long time. It’s not that it’s completely invalid. It’s just that it has problems. The biggest problem is that our senses are interpreted through our brains, an organ which develops strong biases and filters information so that we don’t die. The next problem is that the experimental parameter control actually tends to be quite shoddy. In the worst cases, you get people claiming that, say, console A has a better sound than console B. But…they heard console A in one place, with one band, and console B in a totally different place with a totally different band. There’s no meaningful comparison, because the devices under test AND the test signals were different.

As a result, listening tests produce all kinds of impressions that aren’t actually helpful. Heck, we don’t even know what “sounds better” means. For this person over here, it means lots of high-frequency information. For some other person, it means a slight bass boost. This guy wants a touch of distortion that emphasizes the even-numbered harmonics. That gal wants a device that resembles a “straight wire” as much as possible. Nobody can even agree on what they like! You can’t actually get a rigorous comparison out of that sort of thing.

The flipside is, if we can actually hear it, we should be able to measure it. If a given input signal actually sounds different when listened to through different signal paths, then those signal paths MUST have different transfer functions. A measurement transducer that meets or exceeds the bandwidth and transient response of a human ear should be able to detect that output signal reliably. (A measurement mic that, at the very least, significantly exceeds the bandwidth of human hearing is only about $700.)

As I said, measuring – real measuring – is hard. If the analysis rig is setup incorrectly, we get unusable results, and it’s frighteningly easy to screw up an experimental procedure. Also, we have to be very, very defined about what we’re trying to measure. We have to start with an input signal that is EXACTLY the same for all measurements. None of this “we’ll set up the drums in this room, play them, then tear them down and set them up in this other room,” can be tolerated as valid. Then, we have to make every other parameter agree for each device being tested. No fair running one preamp closer to clipping than the other! (For example.)

Question Everything

So…what to do now?

If I had to propose an initial solution to the problems I see (which may not be seen by others, because this is my own opinion – oh, the IRONY), I would NOT say that the solution is for everyone to graph everything. I don’t see that as being necessary. What I DO see as being necessary is for more production craftspersons to embrace their inner skeptic. The lesser amount of coherent explanation that’s attached to an assertion, the more we should doubt that assertion. We can even develop a “hierarchy of dubiousness.”

If something can be backed up with an actual experiment that produces quantitative data, that something is probably true until disproved by someone else running the same experiment. Failure to disclose the experimental procedure makes the measurement suspect however – how exactly did they arrive at the conclusion that the loudspeaker will tolerate 1 kW of continuous input? No details? Hmmm…

If a statement is made and backed up with an accepted scientific model, the statement is probably true…but should be examined to make sure the model was applied correctly. There are lots of people who know audio words, but not what those words really mean. Also, the model might change, though that’s unlikely in basic physics.

Experience and anecdotes (“I heard this thing, and I liked it better”) are individually valid, but only in the very limited context of the person relating them. A large set of similar experiences across a diverse range of people expands the validity of the declaration, however.

You get the idea.

The point is that a growing lack of desire to just accept any old statement about audio will, hopefully, start to weed out some of the mythological monsters that periodically stomp through the production-tech village. If the myths can’t propagate, they stand a chance of dying off. Maybe. A guy can hope.

So, question your peers. Question yourself. Especially if there’s a problem, and the proposed fix involves a significant amount of money, question the fix.

A group of us were once troubleshooting an issue. A producer wasn’t liking the sound quality he was getting from his mic. The discussion quickly turned to preamps, and whether he should save up to buy a whole new audio interface for his computer. It finally dawned on me that we hadn’t bothered to ask anything about how he was using the mic, and when I did ask, he stated that he was standing several feet from the unit. If that’s not a recipe for sound that can be described as “thin,” I don’t know what is. His problem had everything to do with the acoustic physics of using a microphone, and nothing substantial AT ALL to do with the preamp he was using.

A little bit of critical thinking can save you a good pile of cash, it would seem.

(By the way, I am biased like MAD against the the crowd that craves expensive mic pres, so be aware of that when I’m making assertions. Just to be fair. Question everything. Question EVERYTHING. Ask where the data is. Verify.)