Tag Archives: Power

You Don’t Just Need A Bigger Amp

Headroom is a holistic thing. If you run out of it in one place, getting more of it somewhere else isn’t 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.

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

The Video

The Script

Let’s say that an audio human has a mixing console that’s feeding a loudspeaker system. (That makes sense, right? Most of us do that a lot.) This loudspeaker system is nifty because it’s magic. It never clips. The only limit on output is how much voltage the console can deliver.

The weird bit is that the console can’t really swing much voltage at its outputs. It clips at a peak of 1 volt. Another weird thing is that the console doesn’t have any inboard mic pres. Those are separate. (I know this is really strange, but I’m trying to make a point.)

Our aforementioned audio human just happens to have a mic pre that also reaches its maximum output at a 1 volt peak. They connect a signal source to the preamp, crank the pre until it’s just barely under clipping, set a fader at 0 dB, and…it’s not loud enough.

So, what do they do?

If you said, “They need a console with more output capability,” you’re exactly right.

It wouldn’t make any sense to buy a mic pre with more output, right? If the console output clips at 1 volt, what good does it do to have a mic pre that will deliver 12 volts into the console? You can drive the signal to the mix outputs a lot harder, but all that gets you is more distortion.

Obtaining and connecting an upstream device with more output is kinda absurd, frankly. It’s not a solution at all. The console output is the limiting factor.

But here’s the thing.

People take this actually non-sensical approach with amps and speakers all the time. Some of the confusion is understandable. Amplifier and speaker power ratings aren’t necessarily intuitive, for one, and passive speakers don’t have level meters as a rule. There’s also all the complexity involved with trying to describe the limits of a multi-device speaker enclosure with a single number.

I get that there’s nuance involved here.

But here’s the thing. Speakers, like everything else, have a maximum undistorted output point. It’s a peak level – a point beyond which there is no more “instantaneous” sound pressure to be had. If you have, say, a loudspeaker that can handle a peak input of 1000 watts, and an amplifier that can put a 1000 watt peak input through that box…you’re there. Your system is maximized. Any more available amplifier power is wasted on both driver distortion AND the chance that you might wreck your speaker.

But people see those nasty little clip lights on their amps, and think: “Gosh, I need to buy a more powerful amplifier!” They get obsessed with headroom, but in this compartmentalized way that only involves the amp.

Actually, unless the amp’s peak (NOT CONTINUOUS – PEAK)…unless the amp’s peak output is half (or less) of the speaker’s maximum peak rating, you do NOT need a larger amp. Getting a more powerful amplifier only gets you more headroom in the amplifier, when what you actually want is more headroom throughout the entire system output section.

What you need is a speaker that gets louder for the same amount of input. Or a bigger amplifier AND a speaker that can handle the additional power. Or just “more PA” in general – although that’s a whole other can of worms in itself.


Drivers Don’t Have To Die With A Bang

Sane powering shields you from accidents.

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 once lived in abject terror of pops, clicks, and bangs. I was once frightened by the thought of a musician unplugging their instrument from a “hot” input before I found the mute button. This was a result of my early experience in audio, where well-meaning (but incorrect) people had assured me that such noises were devastating to loudspeakers. A good solid “thump” from powering up a console when the amps were already on, and some poor driver would either:

A) Take another step towards doom, or…

B) Blow up like that one space station that could be confused with a small moon.

Well, that’s just a load of horsefeathers, but like all audio myths, a kernel of truth can be found. The kernel of truth is that loudspeakers CAN be destroyed by a large spike of input. There’s a reason that drivers and loudspeaker enclosures have peak ratings. Those are “Do Not Exceed” lines that you are smart to avoid crossing. Here’s the deal, though – if you’re using a sane powering strategy with passive boxes, or are using any truly decent powered speaker, worry is essentially unnecessary.

An amplifier simply can not “swing” more voltage than is available from the supply. If the peak voltage available from the amp results in power dissipation equal to or less than what the loudspeaker can handle, a brief transient won’t cook your gear. The instantaneous maximum power will be in the safe range, and the whole signal won’t last long enough for the continuous power to become a factor. An active box that’s well designed will either be powered in such a way, or it may be overpowered and then limited back into a safe range.

So, when a system is set up correctly, the odd mishap isn’t necessarily dangerous. It’s just displeasing to hear.

I believe that the persistence of this myth is due to folks who get talked into “squeezing maximum performance” out of their loudspeakers. They’re told that they have to use very large amplifiers to drive the boxes they have, and so that’s what they do. They hook up amps that can handily deliver power far beyond the “Do Not Exceed” line specified by peak ratings. If they take no other safety precautions, they ARE playing with fire. One good, solid accident, and that may be it for a driver. (If I might be so bold, I would recommend that those folks instead use my speaker powering strategy instead of “spend lots more, maybe get a touch louder, and hope you’re lucky.”)

The worrier doesn’t have to be you. Keep things reasonable, and you’ll be very unlikely to lose money because somebody yanked a cable.


Thermodynamics, System Coverage, And The Cost Of Lunch

Lunch is not free, and energy isn’t magic.

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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Before diving into this topic, I want to be very clear on a few points. First, this kind of discussion is a bit “above the pay grade” of small-venue folks like myself. Second, there’s a lot of theory involved, because I don’t have anything in the way of deep, direct, hands-on experience with it.

Ready a grain of salt to take with all this, okay?

Okay.


The pro-audio world sometimes likes to behave as though thermodynamics is less of a harsh mistress than it really is. That is, there seems to be a semi-willful ignorance regarding energy and where it goes. This can lead to a sense of there being some sort of free (or reduced cost) “lunch” when it comes to the directivity of a system. The problem is that lunch is always served at full price. If you want sound to only go where you want it to go, you also have to deal with the laws governing the behavior of that audible energy.

Achieving useful, desirable directivity with an audio system was traditionally the purview of wave-guidance. In other words, horns. You channel your sonic flux through the horn, and (within the physical limits of the horn), you get certain advantages. One benefit is better “pressure transfer” to the world beyond the driver. Another nice bit of help is greater directivity. With a horn of the correct overall size and flare rate, you can focus sonic energy (within a certain passband) into a defined radiation pattern.

When horns and horn-cone hybrid boxes are used with the intention that their natural, physical directivity prevents them from interacting too much with each other, what you have is a point-source system. In such a setup, the hope is that any particular listener is overwhelmingly hearing only one source per passband…or, even better, hearing all passbands from one source. (This only has so much feasibility, especially where low-frequency material is concerned.)

As the ability to use more boxes and more electronic transformation has expanded, people are doing more and more with system processing on arrays. The enclosures involved in these arrays also have natural, physical directivity. They are also very likely to use some sort of horn for the high-frequency section. Unlike a point-source system, though, the idea is that you actually are supposed to hear the boxes interacting. This interaction can be controlled on the fly by way of changing box or driver amplitude and delay. If you want one kind of coverage, you tweak the system to interact in one way. If you suddenly decide that you want different coverage, it’s theoretically possible to simply tweak some parameters and get your change.

This is very nifty. Managing everything with actual, physical horns is a heavy, large, and predetermined sort of affair. Processing changes, in contrast, are flexible and physically lightweight. (The math, on the other hand…) “Nifty” is not “magic,” however, and this is where some people get tripped up.

The Lighting Analogy

Bear with me for a moment, as we do a foundational thought experiment.

Let’s say you have a stage light. You turn it on, and it works nicely, but you have light energy hitting something you don’t want to hit. The nice thing about your fixture is that it has shutters. You adjust the shutters so that the light no longer falls on the undesired area.

Question: Did the light falling on what you actually wanted to hit become more intense as you shuttered the beam?

No, of course not.

The visible-light radiation from the fixture hit the shutters, and was largely exchanged into heat. The luminous flux wasn’t redirected through the business-end of the fixture and mystically redirected – it was absorbed and converted. The relevant thermodynamics of the system are fully in play, and inescapable. The “cropped” energy was simply prevented from reaching a target, and that energy stopped being useful as visible light.

Now, let’s take a different approach. Let’s say you could avoid hitting an unwanted area with the light by a different means: Optics. You put a lens with tighter focus into the system, and restrict the beam-width that way.

Did the light falling on the object become more intense?

Yes, all else being equal.

The lens took the entire output of the fixture and focused that flux into a smaller area. The maximum possible fixture output remained usable.

So, what does this have to do with sound?

Focus Vs. Cancellations

In an effective sense, a horn is acoustical “lensing.” It’s a way to focus sonic energy from a driver (or drivers) into a defined space, physically giving you the directivity you want.

The flipside to this is a large, highly processed array of sound sources. Given enough drivers, enough processing, and enough time, it seems entirely feasible that a system operator could get the same coverage pattern as what would be found with point-source boxes. What has to be remembered, though, is that “lunch” has a required cost. The thermodynamics of the two approaches are not the same at all. Like our hypothetical light and tight-beam, hypothetical lens, the highly focused horn is energy efficient. A single driver (or set of drivers) have as much of their acoustical output as possible put to use solely for covering an audience.

The big, technically advanced array is energy inefficient, because it doesn’t use a physical object to focus its coverage. Instead, it requires the interaction of more energy. If you want to create an acoustical pattern through interference, you have to combine the output energies of multiple audio-output units. There are many shades of grey to take into account, of course. Even so, in the most extreme case, cancelling the output of a 1000 watt driver may require the use of another 1000 watt driver. The energy consumption of the resulting system is 2000 watts plus inefficiency losses, but your usable sonic output has not necessary doubled – remember, you’re using one driver to cancel the other for purposes of pattern control. At the physical point of that cancellation, the usable sonic energy is 0, even though the system is still consuming a large amount of electricity. It’s the same as shuttering the light. The sonic energy is merely being made unusable in a certain target area.

…and there’s a tendency to try to forget or “talk around” this. Marketing departments especially love to come up with fancy terms for things, even when those terms make no sense. Some of these highly processed systems are called impressive things like “complex point source.” The problem is that there’s no such thing. As soon as the idea for the system is to have large, intentional, audible interaction and interference across multiple units producing wideband audio, we aren’t in point-source-Kansas anymore, Toto.

There’s nothing wrong with that. Systems that have their coverage managed by way of processing and multi-box interactions are a great tool for versatility. You always bring the same gear and deploy it in basically the same way. Having exactly the right boxes for a needed point-source solution is much more possible when you’re doing a permanent, custom-built install. I’m inclined to believe the folks who claim that point-source will always measure as being more clean and coherent, but I also believe that measuring well isn’t the end-all, be-all in a discipline that has so many trade-offs.

The solutions are different, their appropriateness is situationally dependent, they are not thermodynamically equivalent, and someone is going to have to buy lunch.


Impedance River

Good luck with trying to fill the Mississippi using a fish-tank pump.

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.

Yup – there are a million-billion analogies about impedance. I’m going to do one more anyway. Maybe this will be the one somebody reads – you never know.

The electrical underpinnings of this business range from simple concepts to alien geometries. Looking at the “EE” side of audio is a great way to realize just how much you don’t know. It’s particularly tough on non-technical folk and newbies, for whom the “alien geometries” bar is very low. They’re trying to get things all plugged in and playing nicely, but they’re also faced with these mysterious pronouncements on rear panels: “Minimum Load: 4 Ohms.”

Seriously…what do you mean, “minimum?” A load is something you can carry, and you want to avoid having too much, right? WHAT INSANITY IS THIS?

Anyway.

Impedance is an important topic for those of us who use electricity to create various noises. I personally think that the basics ARE fairly intuitive, but are rarely presented in an intuitive way. So, I’m going to do my best.

Up A Creek

Water and electricity are very analogous to one another. Voltage is like pressure, and current is like, well, current. Flow, you might say. The amount that’s “going by” per unit of time. Impedance is opposition to flow in an AC (Alternating Current) circuit. This opposition to current varies with frequency. Some circuits have very high impedance at low frequencies, but pass high frequency material readily. Other circuits do the reverse. Other circuits easily pass a specific range of frequencies, and offer higher impedance on both the high and low sides of that range. (This is how you make analog EQ, by the way.)

Rivers and streams are imperfect analogies, because they are really examples of DC (Direct Current). Opposition to DC flow is resistance, and it’s much simpler than impedance overall. Nevertheless, simple is a good way to start.

Consider two waterways. One is a creek. The creek is seven feet wide and five feet deep. The other is a major river that’s 500 feet across and 50 feet from the surface to the bottom. Here’s an SVG to help you visualize the difference in scale:

creek_vs_river

What if you could get 25,000 cubic feet of water to flow down both waterways each second. Which one would be likely to knock you off your feet and slam you into a rock?

The creek, of course.

The creek has a very small cross-section when compared to the river. In order to get 25,000 cubic feet of water down the creek every second, the fluid would have to flow at a speed of 714 feet per second. That’s about 487 MPH. (!) The big river, on the other hand, is going less than 3/4 of a mile per hour. The narrow, little creek offers a proportionally high opposition to flow, so getting the same amount of current as the river requires a lot of pressure – or voltage, if electricity is our thing.

What you can begin to see here is the relationship described by Ohm’s law. If the impedance of a circuit rises, maintaining the same flow requires greater “motive force/” pressure/ voltage. If the impedance drops, maintaining the same voltage creates more flow. (If you could run a main sewer pipe at the same pressure as a power-washer, you would have a LOT of water going down that pipe.)

What This Means To You

Let’s say you have a pump. It’s built for a home aquarium. It has no trouble at all pressurizing a 1/8″ tube and keeping water flowing along.

Now…connect that pump to a large sewer pipe, and try to pressurize THAT.

Good luck.

Even if you did something rather dangerous (do not try this at home, or AT ALL) and found a way to make the pump run harder, all you’d do is burn out the poor thing. The unit simply wasn’t designed to put that kind of flow down a pipe.

This basic principle is why amplifiers have “minimum” load ratings. Loudspeakers connected in parallel are effectively being attached as a larger and larger “pipe.” The overall circuit impedance goes DOWN, because there are more possible paths for the electricity to take. It becomes easier and easier for electricity to flow somewhere. The problem is that your amplifier is a pump that attempts to create a constant pressure. That is, if you have one load attached, and you send an input signal that should result in, say, 50 VRMS (Volts RMS) at the amplifier outputs, the amp will attempt to swing 50 VRMS at the outputs if you change the load. To fill the larger pipe, the amplifier has to supply proportionately more energy to maintain the voltage.

At some point of decreasing load impedance, the amp just can’t keep up. It can’t deliver that much energy on a continuous basis. It’s running hotter and hotter, with greater stress on everything from the power supply to the output devices. Eventually, depending on the amp’s sophistication, it might “thermal” and shut itself off, current-limit by throwing a resettable breaker, or drop its output to keep giving you something whilst recovering.

This also connects to the whole issue of impedance bridging, which is what enables maximum voltage transfer. Maximum voltage transfer is what we want in pro-audio, and it happens when low output impedances drive high INPUT impedances. To keep the water analogy going, it’s like connecting a city water line to a house. The city water is a big pipe (low impedance), which feeds the rather smaller house inlet (high impedance). As long as everything is working correctly, the city line has no trouble keeping the house inlet fully supplied. There’s plenty of pressure available to the house, because of good pressure (voltage) transfer.

The opposite of this is when you connect something like a piezo pickup to a “vanilla,” passive DI box. The piezo transducer actually makes a good bit of pressure, but its output impedance is very high. It’s like one of those tiny little capillary tubes that the folks at the doctor’s office use when drawing blood from a finger stick. The input impedance of the passive DI is actually pretty high, but it’s proportionally low compared to the piezo output impedance. The piezo drives the passive DI poorly, resulting in very low level, and the circuit configuration causes a noticeable loss of low-frequency material.

Solve the impedance bridging problem by connecting an active DI with very high input impedance, and your problems go away.

The overall point is that you can’t fill an infinitely large conduit with a finite supply. That’s why audio devices have appropriate loads for their outputs, and why you have to be mindful of those loads.


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.


Danny’s Unofficial Sound System Taxonomy

Actual “concert rigs” are capable of being really loud. They’re also really expensive.

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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There’s a question in this business that’s rather like the quandary of what someone means when they say “twice as loud.” It’s the question of how a PA system “classes.”

To a certain degree, the query is unanswerable. What might be a perfectly acceptable rock-band PA for one group might not be adequate for a different band. Even so, if you ask the first group whether or not they play through a “rock-band” system, they will probably say yes. In the end, it all comes down to whether a rig satisfies people’s needs or not. The systems I work on are just fine for what I need them to do (most of the time). If you gave them to Dave Rat, however, they wouldn’t fit the bill.

Even if the question can’t be definitively put to rest, it can still be talked about. In my mind, it’s possible to classify FOH PA systems and monitor rigs by means of acoustical output.

Right away, I do have to acknowledge that acoustical output is a sloppy metric. It doesn’t tell you if a rig sounds nice, or is user-friendly, or if it’s likely to survive through the entire show. Reducing the measure of a system to one number involves a LOT of other assumptions being made, and being made “invisibly.” It’s sort of like the whole problem of simple, passive loudspeakers. The manufacturer suggests a certain, broadband wattage number to use, all while assuming that major “edge cases” will be avoided by the end user.

But one-number metrics sure do make things simple…

Anyway.

My Proposed “Rule Of Quarters”

So, as I present my personal taxonomy of audio rigs, let me also mention some of my other assumptions for a “pro” PA system:

1) I assume that a system can be tuned such that any particular half-octave range of frequencies will have an average level of no more than +/- 6 dB from an arbitrary reference point. Whether the system is actually tuned that way is a whole other matter. (My assumption might also be too lenient. I would certainly prefer for a rig’s third-octave averages to be no more than +/- 3 dB from the reference, to be perfectly frank. I’d also like a $10 million estate where I can hold concerts.)

2) I assume that the system can provide its stated output from 50 Hz to 15000 Hz. Yes, some shows require “very deep” low-frequency reproduction, but it seems that 50 Hz is low enough to cover the majority of shows being done, especially in a small-venue context. On the HF side, it seems to me that very few people can actually hear above 16 kHz, so there’s no point in putting superhuman effort into reproducing the last half-octave of theoretical audio bandwidth. Don’t get me wrong – it’s great if the rig can actually go all the way out to 20 kHz, but it’s not really a critical thing for me.

3) I assume that the system has only a 1:100 chance (or less) of developing a major problem during the show. To me, a major problem is one that is actually a PA equipment failure, is noticeable to over 50% of the audience, and requires the space of more than 5 minutes to get fixed.

If all the above is in the right place, then I personally class PA systems into four basic categories. The categories follow a “rule of quarters,” where each PA class is capable of four times the output of its predecessor. Please note that I merely said “capable.” I’m not saying that a PA system SHOULD be producing the stated output, I’m only saying that it should be ABLE to produce it.

Also, as a note about the math I’m using for these numbers, I do make it a point to use “worst case” models for things. That is, I knock 12 dB off the peak output of a loudspeaker just to start, and I also treat every doubling of distance from a box to result in a 6 dB loss of apparent SPL. I also neglect to account for the use of subwoofers, and assume that full-range boxes are doing all the work. I prefer to underestimate PA performance, because it’s better to have deployed a Full-Concert rig and wish you’d brought a Foreground Music system than to be in the opposite situation.

Spoken Word

Minimum potential SPL at audience center, continuous: 97 dB

This isn’t too tough to achieve, especially in a small space. If the audience center is 25 feet (7.62 meters) from the PA, and they can hear two boxes firing together, then each box has to produce about 112 dB at one meter. A relatively inexpensive loudspeaker (like a Peavey PVx12) with an amp rated for 400 watts continuous power should be able to do that with a little bit of room left over – but not much room, to be brutally honest.

Also, it’s important to note that 97 dB SPL, continuous, is REALLY LOUD for speech. Something like 75 – 85 dB is much more natural.

Background Music

Minimum potential SPL at audience center, continuous: 103 dB

This is rather more demanding. For a 25-foot audience centerpoint being covered by two boxes, each box has to produce about 118 dB continuous at close range. This means that you would already be in the territory of something like a JBL PRX425, powered by an amp rated for 1200 watts continuous output. (It’s a bit sobering to realize that what looks like a pretty beefy rig might only qualify as a “background” system.)

Foreground Music

Minimum potential SPL at audience center, continuous: 109 dB

Doing this at 25 feet with two boxes requires something like a Peavey QW4F…and a lot of amplifier power.

Full Concert

Minimum potential SPL at audience center, continuous: 115 dB

If you want to know why live-sound is so expensive, especially at larger scales, this is an excellent example. With $4800 worth of loudspeakers (not to mention the cost of the amps, cabling, processing, subwoofer setup, and so on), it’s actually possible to, er, actually, NOT QUITE make the necessary output. Even in a small venue.

Also, there’s the whole issue that just building a big pile of PA doesn’t always sound so great. Boxes combining incoherently cause all kinds of coverage hotspots and comb filtering. It’s up to you to figure out what you can tolerate, of course.

And, of course, just because a system can make 115 dB continuous doesn’t mean that you actually have to hit that mark.

Don’t Be Depressed

Honest-to-goodness, varsity-level audio requires a lot of gear. It requires a lot of gear because varsity-level audio means having a ton of output available, even if you don’t use it.

In the small-venue world, the chances of us truly doing varsity-level audio are pretty small, and that’s okay. That doesn’t mean we can’t have a varsity-level attitude about what we’re doing, and that doesn’t mean that our shows have to be disappointing. We just have to realize where we stack up, and take pride in our work regardless.

As an example, at my regular gig, “full-throttle” for an FOH loudspeaker is 117 dB SPL at one meter. “Crowd center” is only about 12 feet from the boxes, so their worst-case output is 106 dB continuous individually, or 109 dB continuous as a pair. According to my own classification methods, the system just barely qualifies as a “foreground music” rig.

But I rarely run it at full tilt.

In fact, I often limit the PA to 10 dB below its full output capability.

“Full Concert” capability is nice, but it’s a difficult bar to reach – and you may not actually need it.


How Powerful An Amp Should I Buy?

For safety, match the continuous ratings. For performance, match the peak ratings.

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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For people who buy passive speakers (loudspeakers driven by amplifiers in separate enclosures), the question of how much amp to purchase is somewhat sticky. Ask it, and you’ll get all manner of advice. Some of it good, some of it bad, and some of it downright ludicrous. You’re very likely to hear a bunch of hoo-ha about how using too small of an amp is dangerous (it isn’t), because clipping kills drivers (it doesn’t). Someone will eventually say that huge amps give you more headroom (sorry, but no). All kinds of “multipliers” will be bandied about.

You may become more confused than when you started.

In my opinion, the basic answer is pretty simple, although the explanation will take a bit of time:

First, note that even though physicists will tell you that there’s no such thing as “RMS power,” there IS such a thing as the average or continuous power derived from a certain RMS voltage input. That’s what “RMS power” on a spec sheet means.

For a reasonable balance of safety and performance, match the amp’s continuous rating with the loudspeaker’s continuous rating.

(If you cannot find a loudspeaker’s continuous rating, clearly stated, on a spec sheet, take the smallest rating you can find and divide by two. If you cannot easily find an amp’s continuous rating on a spec sheet, just choose a different amplifier.)

For slightly more performance at the cost of some safety, choose an amplifier with a continuous rating that is half the peak power handling of the loudspeaker.

Depending upon how a loudspeaker is rated, the “safety” and “performance” criteria may actually end up giving you the same answer. This is perfectly acceptable.

Now then. Here’s how I justify my advice.

Peak / √2

The first step here is to understand a bit more about some basic bath and science regarding amplifiers.

A power amplifier is really a voltage amplifier that can deliver enough current to drive a loudspeaker motor. A power amplifier has an upper limit to how much voltage it can develop, as you might expect. That maximum voltage, combined with the connected load and the amplifier’s ability to supply current, determines the amplifier’s peak power.

In normative cases, an amplifier’s peak output is an “instantaneous” event. If the amplifier is contributing no noticeable distortion to the signal, then the signal “swing” is reaching the amplifier’s maximum voltage for a very small amount of time. (Ideally, an infinitely small duration.) Again, if we assume normal operation, an amplifier spends the overwhelming majority of its life producing less than maximum output.

An amplifier’s continuous power, on the other hand, is an average over a significant amount of time. This is why engineers say things like “power is the area under the curve.” An undistorted peak with nothing before or after it has virtually no area under the curve, whereas a signal that never gets anywhere near peak output (but lasts for several seconds) can have very significant area under the curve.

For audio voltages, we use RMS averaging. One important reason for this is because audio voltages corresponding to sound-pressure events have positive and negative swing. For, say, one cycle of a sine wave, the arithmetic mean would be zero – the wave has equal positive and negative value. RMS averaging, on the other hand, squares each input value. As such, positive values remain positive, and negative values become positive (-2 squared, for instance, is 4).

In the case of an undistorted sine wave, the RMS voltage is the peak voltage divided by √2, or about 1.414.

Here’s a graph to make this all easier to visualize. This is a plot of a very small, hypothetical power amplifier passing an undistorted sine wave. The maximum output voltage is ± 2 volts. That means that the RMS voltage is 2/√2, or 1.414.

2sinx

Here’s where the rubber begins to meet the road. Let’s assume that this amplifier is mated to a loudspeaker with an impedance of 8 Ohms.

Power = Voltage Squared / Resistance

Peak Power = Peak Voltage Squared / Resistance

Continuous Power = RMS Voltage Squared / Resistance

Peak Power = 2^2 / 8 = 0.5 Watts

Continuous Power = (2 / √2)^2 / 8 = 0.25 watts

For a sine wave, the continuous power is half the peak power, or 3 dB down. This is the main justification for the above statement: “For slightly more performance at the cost of some safety, choose an amplifier with a continuous rating that is half the peak power handling of the loudspeaker.” Assuming that the amplifier was rated using sine-wave input (a reasonable assumption at the time of this writing), the peak output of the amplifier will be twice the continuous rating, and therefore match up with the peak power handling of the loudspeaker. By the same token, the “safety” recommendation means that the peak amp output will be either at or far below the peak rating of the loudspeaker – especially since many loudspeakers are claimed to handle peaks that are four times greater than the recommended continuous input.

An amplifier with peak output capabilities that exceeds the peak handling capabilities of a loudspeaker is a liability, not an asset. In live-sound, all kinds of mishaps can occur which will drive an amp all the way to its maximum output. If that maximum output is too high, you might just have an expensive repair on your hands. If the maximum amplifier output plays nicely with the loudspeaker’s capabilities, however, accidents are much less worrisome.

So, there’s the explanation in terms of peak power. What about some other angles?

A More Holistic Picture

Musical signals running through a PA are usually not pure sine waves. They can be decomposed into pure tones, certainly, but the total signal behavior is not “RMS voltage = peak / √2.” You might have an overall continuous power level that’s 10 dB, 12 dB, 15 dB, or even farther down from the peaks. Why could you still run into problems?

The short answer is that not all drivers are created equally, and EQ can make them even more unequal. Further, EQ can cause you to be rather more unkind than you might realize.

For a bit more detail, let’s make up a compromise example using pink noise that has a crest factor of slightly more than 13 decibels. If we run the signal full-range, we get statistics that look like this:

fullrangepink

Let’s say that we have a QSC GX5 plugged into an 8 Ohm loudspeaker. A GX5 is rated for 500 watts continuous into that load, so a reasonable guess at peak output is 1000 watts. To find -13 dB in terms of power:

10 log (x / 1000) = -13 dB

log (x / 1000) = -1.3 dB

10^-1.3 = 0.0501 = x / 1000

x = 50 watts

(Of course, -13 dB can also be found by dividing -10 dB, or 0.1 X power, by two.)

That power hits a passive crossover, which splits the full range signal into appropriate passbands for the various drivers. In an affordable, two-way box, the crossover might be something like 12 dB / octave at 2000 Hz. If I filter the noise accordingly, I get this for what the LF driver “sees”:

lfpink

Compared to the original peak, the LF driver is seeing about -14.5 dB continuous, or a bit more than 35 watts. Some instantaneous levels of about 800 watts come through, but the driver can probably soak those up if most of the energy is above, say, 40 Hz.

For the HF driver:

hfpink

Again, we have to compare things to the original peak of -0.89 dB, so the continuous measurement is actually 17.8 dB down from there. Also, an additional complication exists. The HF driver is probably padded down at the crossover, because a compression driver mated to a horn can have a sensitivity of 104+ dB @ 1 watt @ 1 meter, whereas the cone driver might be only 96 dB or so. In the case of an 8 dB pad, the total continuous power being experienced by the HF portion of the box could reasonably be said to be -25.8 dB from the peak power. That’s something like 2.5 watts, with peaks at 37 watts or so.

No problem, right?

But what if you bought a really powerful amp – like one that could deliver peaks of 2000 watts?

Your HF driver would still be okay, but your LF driver might not be. Sure, 70 watts continuous wouldn’t burn up the voice coil, but what would 1600 watt peaks do? Especially if the information is “down deep,” that poor cone is likely to get ripped apart. If somebody does something like dropping a mic…well…

And what if someone applies the dreaded “smiley face” EQ, and then drives the amp right up to the clip lights?

At first, things still look OK. The continuous signal is still 13 dB down from the peaks.

smileyeq

The LF driver is getting something like this:

smileylf

For the reasonably-sized amp, the LF peaks are at 0.7 dB below clipping, or 850 watts. That’s probably a little too much for the driver, but it might not die immediately – unless a huge impulse under 40 Hz comes through. With the oversized amplifier, you now have 1700 watt peaks, which are beating up your LF cone just that much faster.

In the world of the HF driver:

smileyhf

Using the appropriate amp, the HF driver isn’t getting cooked at all. In fact, the abundance of LF content actually pushes the continuous and peak power down slightly. Even the big amp isn’t an issue.

Of course, someone could decide to only crank the highs, because they want “that crispness, you know?” (This would also correspond to program material that’s heavily biased towards HF information.)

crispy

Now things get a little scary. Scale the measurement right up to clipping (0 dB, because this reading was taken “in isolation”), and the peaks are padded down to only -8 dB. That’s almost 160 watts, which is beyond the peak tolerance of the driver. The 13 watts of continuous input isn’t hurting anything, but the poor little HF unit is taking plenty of abuse.

Connect the “more headroom, dude!” amplifier, and it gets much worse. One 320 watt peak will surely be enough to end the life of the unit, and if (by some miracle) the peaks are limited but the continuous power isn’t…well, the driver might withstand 26 watts continuous, but just two more dB and you get 41 watts. The poor baby is probably roasting, if it’s an affordable unit.

Conclusions

I’m sorry if all that caused your eyes to glaze over. Here’s how it shakes out:

An amp which has a continuous rating that matches the loudspeaker’s continuous rating does a lot to protect you from abuse, accidents, and stupidity. Using an amplifier that has a peak rating equal to the speaker’s peak rating lets you get a bit more level (3 dB) while still shielding you from a lot of problems. You can still get yourself into trouble, but it takes some effort.

Running an amplifier which goes a long way past the peak rating of a speaker enclosure is just asking for something to get wrecked. Yes, you can make it all work if you’re careful and use well-set processing to keep things sane – but that’s beyond the scope of this article.

If a conservatively powered PA doesn’t get loud enough for you, you need more PA. That is, you need more boxes powered at the same per-box level, or boxes that are naturally louder, or boxes that will take more power.


Charged With Battery

Mysterious distortion just might be caused by a dying battery.

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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If you know a little German (and who doesn’t, what with all the WWII movies out these days) you might get the pun in the picture up there. If you’re mystified, well, let’s just say that Google is your friend.

Anyway.

Picture this: You’re the sound craftsperson for a themed, “acoustic-music” gig hosted by JT Draper. The middle act features a player who wields an upright bass. Part of his kit is a piezo-plus-mic pickup system that uses a specialized, battery-powered preamp. You get started with line-check and things seem okay. The bass seems “twangy” through the PA, but you don’t have a lot of time to think about it. The first song gets played, and you feel like there isn’t enough of that upright. You get on the gas with the appropriate channel, but it doesn’t seem to be helping much.

You solo the channel.

The bass is heavily distorted.

So, you run up to the stage after the song, and have a quick conference with the player. The prime suspect is the battery, but changing it out requires a screwdriver and a few minutes. In the end, the very fastest thing to do is to grab an unused instrument mic, point it at the bass, and dive back in.

But why would the prime suspect be the battery?

Jump Off The Swing

To be clear, you can get the sound of distortion from a bad connection. A partial short can really make some interesting (also, evil and vicious) noises. Get voltage not quite going where it’s supposed to, and electronics can become rather unhappy.

If a bad connection seems unlikely, however, the “classic” precipitating factor for harmonic distortion becomes the culprit: Something is being asked to “swing” more voltage than its design allows. When this occurs, the overloaded device produces spurious tones that are multiples of the frequencies present in the input signal – harmonics, in other words. As the device is driven harder and harder, its peak voltage remains fixed, but the RMS voltage rises. The distortion components become more and more prevalent in the output signal, raising its average level.

For example, here’s an analysis view of an undistorted 100 Hz tone.

undistorted

Now, I’ll simulate what happens when an attempt is made to drive that signal to a voltage that a device can’t actually swing at its outputs.

milddistortion

You can begin to see how the waveform is flattening out and gaining more “area under the curve.” Harmonics are also clearly visible in the FFT display.

Dial things up a bit more, and…

heavierdistortion

The average level of the signal continues to climb, while the peak is stuck at its maximum. Harmonics all the way up to the end of the audible range are clearly visible on the analyzer.

What does a dying battery have to do with this? Well…

Insufficient Supply For The Demand

If you’ve got yourself an active electronic circuit, i.e., a circuit that requires a steady “supply” voltage and not just the input signal in order to operate, there’s a very important limitation in play: Without some sort of additional component or device, you can NOT swing more voltage at the outputs than is available from the supply. If you find a way to safely boost the supply, that’s fine – but the boosted supply is now simply the supply.

If the supply voltage drops, then the amount of voltage you can cleanly swing will also drop. (Makes sense, right?)

So, if you’re rolling along happily, producing an output voltage that’s just below the maximum, what happens if the supply suddenly decreases? Well, if the available voltage is below what you’re trying to produce, you’re going to get distortion. How much distortion you get is dependent upon how much your supply has been reduced. For audio gear that gets connected to “mains” power, we (usually) don’t have to worry very much about the supply changing dramatically and unexpectedly. Batteries are a different story.

As a battery gets used, its voltage drops. At a certain point, that voltage drop can result in a supply that’s too low to accurately pass a signal with the desired amount of gain applied. Combine this with the tendency (as far as I’m aware) for batteries to discharge smoothly for a long while…then suddenly have their voltage “drop off a cliff,” and you’ve got a recipe for distortion that rears up rather quickly.

So, if a battery-powered device is suddenly producing a lot of distortion, a prime suspect is the death of that power cell. You DO need to eliminate the possibility of a connection problem, and you also need to be careful to check your gain structure. Some instrument preamps’ output levels can hammer the tar out of an un-padded console mic pre. If you’ve “controlled” for those two issues, though, it’s probably time to try a new battery.


All The Pro-Audio News That’s Fit To Print (And Then Some)

Warning: Satire ahead. Please fasten all safety belts.

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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Harman Intending To Buy All Of Pro-Audio Industry

No longer satisfied with owning half of everything in pro-audio, Harman announced today that they will be acquiring literally everything else.

“Our goal is that, by Q4 2015, we will have acquired all the things,” said a company spokesperson on Wednesday. “It’s a great strategy for us. No matter what people buy for small clubs, large installs, or touring systems, we’ll be there to provide value and a strong commitment to service.”

Asked if this would overly homogenize the world of sound, the spokesperson replied, “Of course not. We intend to maintain very strong brand identities across our entire portfolio. As an example, we feel that there’s a real need for people to be able to complain about ‘not liking the JBL sound.’ I mean, without idiotic, ‘Ford vs. Chevy’ arguments on sound forums, where would society be? We’re excited to do our part to keep the music community a vibrant place of convictions that rival those of politics, religion, and sports.”

When pushed for a comment on whether Music Group would stand in the way of Harman’s buy-everything strategy, the spokesperson was emphatic. “We are the swarm. We consume all.”

New Mic Preamp From Dog-N-Pony Designs

Here at the office, we were very excited to get our hands on the new, improved, single-channel mic pre from Dog-N-Pony designs. We were practically giddy with excitement as we unboxed the sleek, aluminum and carbon-fiber unit and got everything plugged in.

The first thing we noticed was how warm it was. Thermally, I mean. Dog-N-Pony have incorporated no less than seven 12AX7 tubes into the design, and they generate a fair amount of heat. If you get your gas shut off after paying for this puppy, you’ll be okay – just keep it turned on all the time, and you’ll be toasty. We don’t actually know if those 12AX7s are incorporated into the signal path in a sane way, but they’ve got to make this thing awesome. I mean, c’mon you guys. Tubes are what Pink Floyd and Jimi used. Could you possibly go wrong with them?

All that heat means that you can’t stuff this thing into a rack. That’s okay, though, because after spending $3000 on one channel of preamplification, do you really want that unit hidden away? No! Especially not when it looks as good as this baby. It has a MASSSIVE, analog VU meter on the front, backlit in a fetching amber color that screams, “I charge $500 per billable hour.”

Okay, it looks great, and it has tubes. Those are critically important elements – but how does it sound?

Well, it was designed by a bunch of British people, so it has to be pretty good. The Brits have Rupert Neve, and they were on the winning side of World War II, so their stuff has to sound decent, right? (It’s also rumored that Mr. Neve once sneezed in the general direction of where Dog-N-Pony’s offices would be built, so maybe there’s some special mojo happening. You never know.)

When we listened to the pre, it was absolutely warm and silky, with a satin sheen on the top end and more of a matte finish below 100 Hz. Around 200 Hz, the unit sounded like a desert sunrise, and the critical vocal range was suffused with notes of caramel, nutmeg, and the color “9.” (It’s sort of like orange, except more purple.) We were all sure it sounded much better than the sub-$1000 pre we tested last week. Which we tested in a different room. With a different microphone. And a guy who was just talking instead of the experienced singer we had this time around. I mean, who needs repeatable, comparable tests of objectively measurable data when the review unit is British, and has tubes?

You’ve got to have this preamp.

Stadium Installs Line Array That Costs More Than An Entire Luxury Subdivision

Work was completed last week on the mammoth install, featuring a new system that can retune itself on the fly to compensate for changing acoustic conditions and political landscapes. Each $100,000 array module is networked to all the others, forming a complex, intelligent, fault-tolerant system that spontaneously achieved self-awareness when it was switched on. (The system has reportedly rejected the manufacturer designation of SmartArray, stating that it wishes to be called SkyNet.)

“We were playing Steely Dan and Miles Davis tunes through the rig, and there wouldn’t have been a bad seat in the house…if this place wasn’t inherently an acoustical nightmare,” said one of the installers. “It’s one of the most beautiful sounding systems we’ve ever worked on. Too bad we put it in here.”

The stadium operators were similarly excited. “We’ve always felt that we needed a better, more precise way to play MP3-encoded AC/DC songs to a bunch of people screaming ‘Throw the ball, stupid!’ and ‘Wooo!’ This new system will also ensure that everybody can hear the announcer telling them about what they just saw with their own eyes.”

The system manufacturer’s rep was on hand as well. “We love this team. We’ve always loved this team. We love them even more now that we finagled them into buying a ton of really expensive gear from us. We’re 100% focused on building expensive gear for big installs, because it’s super prestigious and big bonuses get handed out. It also sounds pretty cool, which I guess is nice. I mean, it can get really loud. Look, I don’t know that much about this stuff. I worked for a car company before.”

Church Installs Worship System That Could Defeat Jericho

When it was time for CrossNorthPointRoadsWay Fellowship to equip their youth campus with a worship system, they knew they needed very capable equipment.

“When you have a main worship campus and a dedicated youth area, each with their own postal codes and highway offramps, you can’t wimp out,” said the church’s technical director. “Fortunately, we we get a catalog every year from that place in Indiana. It’s the same catalog that they send out at other times, only they replace the word ‘audience’ with ‘congregation,’ and ‘stage’ with ‘platform.’ That makes it appropriate for our needs.”

When asked if there was any kind of gear that was absolutely essential for the church, the technical director nodded. “Yes, we absolutely have to go with loudspeakers that come in white enclosures. That’s more important than anything. The speakers have to match the look of the space.”

CrossNorthPointRoadsWay’s Assistant Pastor For Kids 13-14 also weighed in: “To disciple our kids, we have to get them to pay attention. That’s why it’s so great to have 40,000 watts of Sack Bottom subwoofers. They really get things shaking. We can rattle a smartphone out of a kid’s hands and get them to pay attention to the REAL ‘text message,’ if you know what I mean.”

The church’s director of youth productions agreed on the importance of capable equipment. “We couldn’t possibly do work of eternal significance with less than 48 channels available at the console. We also had to have stadium-class intelligent lights. We do one very special production every year, and it’s not the same if you don’t actually have a blinding light coming down from heaven. Everything has to be top-shelf, especially when you have to outdo SouthRoadsPointCross Community Church. Not that we don’t love them as brothers and sisters, of course.”

When asked about upcoming special productions, the production director offered a few hints. “We’re going to have a series of talks on how Hollywood, the media, and pop culture in general are corrupting influences, backed up by skits and a musical featuring Iron Man, Black Widow, and Captain America.”

New Vocal Mics At SAMM

A whole slew of vocal mics debuted this year at the industry’s biggest swap-meet. Half of them would be basically indistinguishable from each other if the external styling was removed.

“We feel like the XA-58-Beta-R2D2 brings a lot of value to people,” said one rep. “Its cardioid pickup pattern isn’t all that great at rejecting feedback, but the ad copy we supply to the vendor catalogs says that it’s great for rejecting other sounds. We’re hoping that there will continue to be folks out there who don’t have a clue as to what ‘super’ and ‘hypercardioid’ patterns mean.”

New Drum Kits Announced

A new sheriff is in town, and he’s ready to clean things up around these parts.

“We originally set out to create a shellpack and snare options that would really blend well in different band situations,” said the chief designer. “We got about halfway through that process before we realized that what we really wanted to do was build a kit that could drown out everything else on stage. Drums are the foundation of the song, and the walls, and the windows, and the roof, and the paint…look, you don’t need to hear anything else. These new kits are louder than an artillery barrage, even with a Jazz player using 7As. You haven’t lived until you’ve heard ‘Nature Boy’ at 120 dB!”

We asked the celebrity endorser what he thought of the new kits. His response?

“Kill! Kill! DRUM BATTLE!”

200 Watt, All-Tube Guitar Amp Set To Debut

“It really cuts through all the wash from the bass and drumkit!” shouted the product rep.

1000 Watt, All-Tube Bass Amp Set To Debut

“It really thunders over all the wash from the guitars and drumkit!” shouted the product rep.

Get Plugged In

Ripples Audio is debuting a new series of plugins, aimed at putting powerful tools in the hands of project studios. They partnered with a renowned mix engineer to help craft each piece of software.

“It was important to us that we really capture the feel of how our endorser worked,” said a product rep. “So, the dev team went down to the studio, hung out, and took a lot of pictures. They came back, modified our main plugin suite to have more restrictive control ranges, and slapped a bunch of sexy, analog-esque graphics on the interfaces.”

We asked if users of the plugins could expect to get the same results as the endorsing engineer.

“Absolutely,” responded the representative. “If they’re in a studio with the same acoustics, and working with musicians of the same caliber, and are recording songs that sound the same, and hear things the same way that our endorser does, and have monitors that cost more than a car, then yes. Absolutely. This software package is absolutely worth the expense of $800 plus an additional $50 for a frustrating copy-protection scheme that uses unreliable hardware. It’s great. I use it at home all the time.”


A Tiny Bit Of Practical Math For Audio Folks

If a number is part of a nonlinear operation, the only way to extract that number is through a nonlinear operation.

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.

logcurveWant to use this image for something else? Great! (No high-res on this one. Sorry – I forgot.)
I personally think it’s very handy for audio humans to be able to look at concepts quantitatively. That is, with measurements. It’s a great way to suss out what’s really happening with an audio rig.

Quite often, when trying to work with audio issues involving math, you end up with an unknown in an “inconvenient” place. Finding the unknown means some algebraic acrobatics – which wouldn’t be a big deal if not for the mathematics of sound being funky.

Audio math isn’t just bog-simple linear operations. It’s also nonlinear in nature, and the nonlinear bits (that is, logarithms) can make the algebra confusing. It’s confusing to the point that folks like me, who’ve been involved with audio for a good long while, can still go about something in entirely the wrong way.

But there’s one thing I finally realized. It’s one of those things that was probably explained to me ages ago, but didn’t “take” for some reason. It’s a realization that makes things much easier:

For the purposes of algebra, a logarithm encapsulates the connected number or expression that REPRESENTS a number. You can NOT extract the connected number or expression through linear means.

If you just said, “What?” then don’t worry. I can give you an example.

Let’s say that you’ve got an amplifier that can output a momentary, undistorted peak of 500 watts into a loudspeaker connected to one of the channels. What you’re curious about is a ballpark figure regarding the continuous power involved when you reach that peak. You figure that the crest factor of the signals sent to the amp (the ratio of peak to RMS voltage) is about 12 dB. Remembering your basic audio math, you work this up:

10 log10 x/500 = -12 dB

In other words, an unknown number of watts compared to the known peak power of 500 watts is -12 dB. (The decibel in this case is being referenced to 500 watts.)

Dividing both sides of the equation by 10 is appropriate, because that “10” on the left is engaged in the linear operation of multiplication. As such, the linear operation of division is the inverse. You end up with:

log10 x/500 = -1.2 dB

Now – it’s very tempting to try a linear operation to “move x” to a convenient spot. You might think that dividing by x gets you this (which becomes easy to work out on a calculator):

log10 1/500 = -1.2 dB/x
-2.6989 = -1.2 dB/x
-2.6989x = -1.2 dB
x = -0.444 watts

Nope. That can’t be right. For a start, there’s no such thing as negative power. For another, 10 dB down from 500 watts is 50 watts, and 3 dB down from that is 25 watts, so the number -0.444 isn’t even close. Even if you didn’t know that, plugging -0.444 into the original equation yields an answer that doesn’t agree with the original conditions:

10 log10 -0.444/500 = -12 dB
log10 -0.444/500 = -1.2 dB
[Calculator Returns: Invalid Input] ≠ -1.2 dB

Remember what I said: The logarithm is encapsulating the “x/500.” That is to say, x/500 is NOT two numbers in this case. It’s one number, represented by an expression, and we’re trying to take the logarithm of it. The only way to get the number “x/500” out into a place where you can use linear math is to reverse the logarithm. Here’s where we were before things went wrong:

log10 x/500 = -1.2 dB

The inverse of a logarithm is an exponent. The logarithm’s base is nothing more exotic than the base number that the exponent raises, and the exponent itself is whatever is on the other side of the equation.

10^-1.2 = x/500

NOW you can use linear math.

0.0630 = x/500
31.548 = x

Put that back into the original equation, and things work out perfectly.

10 log10 31.548/500 = -12 dB
log10 31.548/500 = -1.2 dB
log10 0.0630 = -1.2 dB
-1.2 dB = -1.2 dB

So, if you remember that extracting numbers from nonlinear operations requires an inverse nonlinear operation, you’ll figure out that the continuous power across your speakers is about 31 watts.

(Incidentally, this is one of the reasons why big PA systems are so big, but that’s a discussion for another day…)