Tag Archives: Science

The Unterminated Line

If nothing’s connected and there’s still a lot of noise, you might want to call the repair shop.

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 thought we fixed the noise on the drum-brain inputs?” I mused aloud, as one of the channels in question hummed like hymenoptera in flight. I had come in to help with another rehearsal for the band called SALT, and I was perplexed. We had previously chased down a bit of noise that was due to a ground loop; Getting everything connected to a common earthing conductor seemed to have helped.

Yet here we were, channel two stubbornly buzzing away.

Another change to the power distribution scheme didn’t help.

Then, I disconnected the cables from the drum-brain. Suddenly – the noise continued, unchanged. Curious. I pulled the connections at the mixer side. Abruptly, nothing happened. Or rather, the noise continued to happen. Oh, dear.


When chasing unwanted noise, disconnecting things is one of your most powerful tools. As you move along a signal chain, you can break the connection at successive places. When you open the circuit and the noise stops, you know that the supplier of your spurious signal is upstream of the break.

Disconnecting the cable to the mixer input should have resulted in relative silence. An unterminated line, that is, an input that is NOT connected to upstream electronics, should be very quiet in this day and age. If something unexplained is driving a console input hard enough to show up on an input meter, yanking out the patch should yield a big drop in the visible and audible level. When that didn’t happen, logic dictated an uncomfortable reality:

1) The problem was still audible, and sounded the same.

3) The input meter was unchanged, continuing to show electrical activity.

4) Muting the input stopped the noise.

5) The problem was, therefore, post the signal cable and pre the channel mute.

In a digital console, this strongly indicates that something to do with the analog input has suffered some sort of failure. Maybe the jack’s internals weren’t quite up to spec. Maybe a solder joint was just good enough to make it through Quality Control, but then let go after some time passed.

In any case, we didn’t have a problem we could fix directly. Luckily, we had some spare channels at the other end of the input count, so we moved the drum-brain connections there. The result was a pair of inputs that were free of the annoying hum, which was nice.

But if you looked at the meter for channel two, there it still was: A surprisingly large amount of input on an unterminated line.


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 3 feet 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 -20 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) 6 inches 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 four times.

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

Assumptions: Microphones with greater feedback resistance will exhibit a lower input level on the fourth 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…


Measuring A Cupped Mic

What you might think would happen isn’t what happens.

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 most popular article on this site to date is the one where I talk about why cupping a vocal mic is generally a “bad things category” sort of experience. In that piece, I explain some general issues with wrapping one’s hand around a microphone grill, but there’s something I didn’t do:

I didn’t measure anything.

That reality finally dawned on me, so I decided to do a quck-n-dirty experiment on how a microphone’s transfer function changes when cupping comes into play. Different mics will do different things, so any measurement is only valid for one mic in one situation. However, even if the results can’t truly be generalized, they are illuminating.

In the following picture, the red trace is a mic pointing away from a speaker, as you would want to happen in monitor-world. The black trace is the mic in the same position, except with my hand covering a large portion of the windscreen mesh.

You would think that covering a large part of the mic’s business-end would kill off a lot of midrange and high-frequency information, but the measurement says otherwise. The high-mid and HF information is actually rather hotter, with large peaks at 1800 Hz, 3900 Hz, and 9000 Hz. The low frequency response below 200 Hz is also given a small kick in the pants. Overall, the microphone transfer function is “wild,” with more pronounced differences between peaks and dips.

The upshot? The transducer’s feedback characteristics get harder to manage, and the sonic characteristics of the unit begin to favor the most annoying parts of the audible spectrum.

Like I said, this experiment is only valid for one mic (a Sennheiser e822s that I had handy). At the same time, my experience is that other mics have “cupping behavior” which is not entirely dissimilar.


The Difference Between The Record And The Show

Why is it that the live mix and the album mix end up being done differently?

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.

Jason Knoell runs H2Audio in Utah, and he recently sent me a question that essentially boils down to this: If you have a band with some recordings, you can play those recordings over the same PA in the same room as the upcoming show. Why is it that the live mix of that band, in that room, with that PA might not come together in the same way as the recording? The recording that you just played over the rig? Why would you NOT end up having the same relationship between the drums and guitars, or the guitars and the vocals, or [insert another sonic relationship here].

This is one of those questions where trying to address every tiny little detail isn’t practical. I will, however, try to get into the major factors I can readily identify. Please note that I’m ignoring room acoustics, as those are a common factor between a recording and a live performance being played into the same space.

Magnitude

It’s very likely that the recording you just pumped out over FOH (Front Of House) had a very large amount of separation between the various sources. Sure, the band might have recorded the songs in such a way as to all be together in one room, but even then, the “bleed” factor is very likely to be much smaller than what you get in a live environment. For instance, a band that’s in a single-room recording environment can be set up with gobos (go-betweens) screening the amps and drums. The players can also be physically arranged so that any particular mic has everything else approaching the element from off-axis.

They also probably recorded using headphones for monitors, and overdubbed the “keeper” vocals. They may also have gone for extreme separation and overdubbed EVERYTHING after putting down some basics.

Contrast this with a typical stage, where we’re blasting away with wedge loudspeakers, we have no gobos to speak of, and all the backline is pointed at the sensitive angles of the vocal mics. Effectively, everything is getting into everything else. Even if we oversimplify and look only at the relative magnitudes between sounds, it’s possible to recognize that there’s a much smaller degree of source-to-source distinctiveness. The band’s signals have been smashed together, and even if we “get on the gas” with the vocals, we might also be effectively pushing up part of the drumkit, or the guitars.

Time

Along with magnitude, we also have a time problem. With as much bleed as is likely in play, the oh-so-critical transients that help create vocal and musical intelligibility are very, very smeared. We might have a piece of backline, or a vocal, “arriving” at the listener several times over in quick succession. The recording, on the other hand, has far more sharply defined “timing information.” This can very likely lead to a requirement that vocals and lead parts be mixed rather hotter live than they would be otherwise. That is, I’m convinced that a “conservation of factors” situation exists: If we lose separation cues that come from timing, the only way to make up the deficit is through volume separation.

A factor that can make the timing problems even worse is those wedge monitors we’re using, combined with the PA handling reproduction out front. Not only are all the different sources getting into each other at different times, sources being run at high gain are arriving at their own mics several times significantly (until the loop decay becomes large enough to render the arrivals inaudible). This further “blurs” the timing information we’re working with.

Processing Limits

Because live audio happens in a loop that is partially closed, we can be rather more constrained in what we can do to a signal. For instance, it may be that the optimal choice for vocal separation would simply be a +3 dB, one-octave wide filter at 1 kHz. Unfortunately, that may also be the portion of the loop’s bandwidth that is on the verge of spiraling out of control like a jet with a meth-addicted Pomeranian at the controls. So, again, we can’t get exactly the same mix with the same factors. We might have to actually cut 1 kHz and just give the rest of the signal a big push.

Also, the acoustical contribution of the band limits the effectiveness of our processing. On the recording, a certain amount of compression on the snare might be very effective; All we hear is the playback with that exact dynamics solution applied. With everything live in the room, however, we hear two things: The reproduction with compression, and the original, acoustic sound without any compression at all. In every situation where the in-room sound is a significant factor, what we’re really doing is parallel compression/ EQ/ gating/ etc. Even our mutes are parallel – the band doesn’t simply drop into silence if we close all the channels.


Try as we might, live-sound humans can rarely exert the same amount of control over audio reproduction that a studio engineer has. In general, we are far more at the mercy of our environment. It’s very often impractical for us to simply duplicate the album mix and receive the same result (only louder).

But that’s just part of the fun, if you think about it.


Case Study: Creating A Virtual Guitar Rig In An Emergency

Distortion + filtering = something that can pass as a guitar amplifier in an emergency.

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

Imagine the scene: You’re setting up a band that has exactly one player with an electric guitar. They get to the gig, and suddenly discover a problem: The power supply for their setup has been left at home. Nobody has a spare, because it’s a specialized power supply – and nobody else plays an electric guitar anyway. The musician in question has no way to get a guitar sound without their rig.

At all.

As in, what they have that you can work with is a guitar and a cable. That’s it.

So, what do you do?

Well, in the worst-case scenario, you just find a direct box, run the guitar completely dry, and limp through it all as best you can.

But that’s not your only option. If you’re willing to get a little creative, you can do better than just having everybody grit their teeth and suffer. To get creative, you need to be able to take their guitar rig apart and put it back together again.

Metaphorically, I mean. You can put the screwdriver away.

What I’m getting at is this question: If you break the guitar rig into signal-processing blocks, what does each block do?

When it comes right down to it, a super-simple guitar amp amounts to three things: Some amount of distortion (including no distortion at all), tone controls, and an output filter stack.
The first two parts might make sense, but what’s that third bit?

The output filtering is either an actual loudspeaker, or something that simulates a loudspeaker for a direct feed. If you remove a speaker’s conversion of electricity to sound pressure waves, what’s left over is essentially a non-adjustable equalizer. Take a look at this frequency-response plot for a 12″ guitar speaker by Eminence: It’s basically a 100 Hz to 5 kHz bandpass filter with some extra bumps and dips.

It’s a fair point to note that different guitar amps and amp sims may have these different blocks happening in different orders. Some might forget about the tone-control block entirely. Some might have additional processing available.

Now then.

The first thing to do is to find an active DI, if you can. Active DI boxes have very high input impedances, which (in short) means that just about any guitar pickup will drive that input without a problem.

Next, if you’re as lucky as I am, you have at your disposal a digital console with a guitar-amp simulation effect. The simulator puts all the processing I talked about into a handy package that gets inserted into a channel.

What if you’re not so lucky, though?

The first component is distortion. If you can’t get distortion that’s basically agreeable, you should skip it entirely. If you must generate your own clipping, your best bet is to find some analog device that you can drive hard. Overloading a digital device almost always sounds terrible, unless that digital device is meant to simulate some other type of circuit.
For instance, if you can dig up an analog mini-mixer, you can drive the snot out of both the input and output sides to get a good bit of crunch. (You can also use far less gain on either or both ends, if you prefer.)

Of course, the result of that sounds pretty terrible. The distortion products are unfiltered, so there’s a huge amount of information up in the high reaches of the audible spectrum. To fix that, let’s put some guitar-speaker-esque filtering across the whole business. A high and low-pass filter, plus a parametric boost in the high mids will help us recreate what a 12″ driver might do.
Now that we’ve done that, we can add another parametric filter to act as our tone control.

And there we go! It may not be the greatest guitar sound ever created, but this is an emergency and it’s better than nothing.

There is one more wrinkle, though, and that’s monitoring. Under normal circumstances, our personal monitoring network gets its signals just after each channel’s head amp. Usually that’s great, because nothing I do with a channel that’s post the mic pre ends up directly affecting the monitors. In this case, however, it was important for me to switch the “monitor pick point” on the guitar channel to a spot that was post all my channel processing – but still pre-fader.

In your case, this may not be a problem at all.

But what if it is, and you don’t have very much flexibility in picking where your monitor sends come from?

If you’re in a real bind, you could switch the monitor send on the guitar channel to be post-fader. Set the fader at a point you can live with, and then assign the channel output to an otherwise unused subgroup. Put the subgroup through the main mix, and use the subgroup fader as your main-mix level control for the guitar. You’ll still be able to tweak the level of the guitar in the mix, but the monitor mixes won’t be directly affected if you do.


Monitor World – Is “More” Better?

Often, the answer is “nope.”

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.

Monitor world is a PA system, just like FOH is a PA system. The only difference is that monitor world handles a few very small audiences, and FOH usually deals with one comparatively large audience. All the helpful AND problematic physics considerations are the same.

This being the case, the stage is yet another place where simply piling up more and more boxes (all doing the same thing) to get “more” can be counterproductive. A vocalist wants more vocal, but their monitor is already doing everything it can, so you add another box. Does it look impressive? Yes! Is it louder? Yes! Is it better?

Yea- er…well…wait a second…

What you very well might end up with is a different set of issues. If the singer isn’t precisely situated between the wedges, the wedge outputs arrive at different times. This means that all kinds of destructive phase weirdness might be happening, and that can lead to intelligibility issues. The vocal range is very easy to louse up with time-arrival differences, and a sensation of “garble” can lead to a player wanting even MORE monitor level in compensation. In that instance, you haven’t actually gotten anywhere; Monitor world is louder, but it’s not any easier to hear in the information-processing sense. You also have greater effective loop-gain with that extra volume rocketing around, which destabilizes your system.

Plus, the low-frequency information still does combine well, which can lead to a troublesome buildup of mud. This goes double for everybody who’s off-axis (and that’s probably just about everybody who isn’t the intended audience of those wedges). That makes them want their own mixes to be hotter, which compounds all your problems even more.

And, of course, there’s even more bleed into FOH.

The brutal reality is that, for any single sound that a given player needs to hear, that signal will always sound better coming from a single box that “can get loud enough.” More wedges (all producing the same output) can only combine less and less coherently as you add more of them.

“But, Danny,” you protest, “you’ve done dual wedges for people. You’ve even rolled out some really excessive deployments, like the one in the article picture. Who are you to tell folks not to do that kind of thing?”

Fair point! In response:

1) It’s because I’ve tried some strange monitor solutions that I can say they weren’t necessarily improvements over simpler approaches.

2) Sometimes you do things that look cool, accepting that you’ll have to deal with some sonic downsides as a result.

3) Just because you’ve piled up a bunch of wedges, it doesn’t require you to put the exact same thing through each enclosure. Somebody might have two boxes in front of them, but one might be for vocals only and the other for instruments only.

With some bands, especially those who are naturally well balanced and don’t need a ton of monitor gain, the extra fun-factor and volume bump can trade off favorably with the coherence foibles. As the rest of this article indicates, yes, I am in the camp that says that a single box will always “measure better.” However, there’s more to life than just “measuring better.” If you have some room to compromise, you can be a little weird without hurting anything too badly.

Audio is an exercise in compromise. If you know what the compromise factors are, you can make an informed judgement. If you know that throwing a bunch of boxes at a problem might cause you other problems, then you’ve got more knowledge available to help you make the right decision for a fix.


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.


The Decibel…And You

Logarithmic scales are groovy.

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:

About the music playing underneath the narration:

Frost Waltz by Kevin MacLeod is licensed under a Creative Commons Attribution license (https://creativecommons.org/licenses/by/4.0/)
Source: http://incompetech.com/music/royalty-free/index.html?isrc=USUAN1100516
Artist: http://incompetech.com/

Here’s the narration script, if you like:

The decibel – what is it?

The decibel is a nonlinear unit of measure created by Bell Telephone Laboratories. In telecommunications systems and professional audio applications, it is often necessary to compare large differences in measured power. This can be inconvenient with linear units.

The decibel solves this problem using a logarithmic scale. No, no, not phat beats being produced by striking a piece of wood at regular intervals. The logarithm: The inverse of an exponent. Logarithmic scales compact large, linear ranges of values into a much more manageable form. The logarithm used by the decibel is concerned with powers of 10, hence it is a base-10 logarithm. Be sure that any decibel calculations you perform use a base-10 logarithm; Some mathematics systems default to the natural logarithm instead.

The decibel is a unit that describes a power ratio. As such, you should be aware of three main rules for the use of this unit: First, that the decibel has no meaning unless a reference point is designated. Second, this reference point is the denominator for the ratio, and thus, must not be zero. Third, logarithms are only valid for ratios with a positive value. A decibel value can be negative, but the input ratio must not be.

All sorts of reference points for decibels exist. There is dBW, which references one watt of power. There is dBu, which references 0.775 volts RMS, un-terminated. There is dBSPL, which references 20 micro Pascals, the threshold of human hearing at 1 Khz.

For a power ratio, the decibel value is the 10 times the base-10 logarithm of the ratio. A ratio of one – that is, the reference point itself, is always zero decibels. Ratios greater than one give positive decibel values, whereas ratios less than one give negative results.

But wait, you say! Professional audio is often concerned with voltage, yet the decibel is concerned with power. How can we square that circle?

Remember that voltage can be related to power in various ways. One such form is this: Power equals voltage squared over resistance. Because we are concerned with the ratio of voltages, and not the actual power value, we can set the resistances as being equal to one. This leaves us with voltage squared over voltage squared. This may seem clumsy to calculate, but never fear! The same result may be obtained by multiplying the base-10 logarithm of the simple voltage ratio by 20 instead of 10. Isn’t that swell?

The decibel is a versatile unit of measure that can be adapted to many needs in the professional audio world. Know it, and use it well.


Why Are Faders Labeled Like That?

Gain multipliers are hard to read.

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