Our last “episode” was all about transferring electric signals to a mixing console or recording device. Now, it’s time to talk about the input section of a console.

Jack Be Nimble

When connecting to a console input, it’s important to choose the correct connection point. Some consoles make this very easy by offering only one physical input, usually a simple female XLR – or, a female XLR/ female TRS “combo” jack. In other cases, you might have separate connectors for microphone-level signals and line-level signals.

When the choice is necessary, the major deciding factor is primarily based on the amount of gain (that is, voltage increase) necessary for the signal to work well with the console. We’ll talk more about this later.

Another kind of jack you might encounter in a console’s input section is the insert. Inserts are actually a combination input/ output point that, electronically, comes after the preamp or trim. An insert is meant for pulling a line level signal out of a console entirely, processing the signal, and then putting the signal back into the console so it can be mixed as normal. This is often accomplished by using a cable with a TRS male on one end, and two TS males on the other. One TS male connects to the tip and ground of the TRS, whereas the other connects to sleeve and ground. One TS, then, is the “send” to the external processing, and the other is the “return.”

PAD

Some consoles, as well as some DI boxes, come equipped with a PAD switch. “Padding” an input means running the signal through electronics designed to reduce voltage before other devices are encountered. PADs are used to avoid overloads from a following gain stage. For instance, in the case of a console with only one input point for each channel, the input stage itself might not be able to handle certain voltage levels. With a PAD engaged, that incoming voltage can be reduced to something more manageable.

Preamps…Or Just A Trim?

Many audio processing devices work best when the audio level presented to them is between roughly 1 volt and 10 volts, also known as “line-level.” Signals from microphones and DI boxes can often be well below this general area, perhaps down in the tenths or hundredths of a volt. These lower-voltage signals are commonly categorized as “mic-level.” Signals at mic-level may require large, positive gain changes to correctly drive downstream electronics, and so a jack that can be connected to a microphone preamp is needed in that case. Mic pres are important and specially designed, as they have a difficult job: They make very large changes to input signal voltages while being as noiseless as possible. They are the “highest” gain stage in a console; Nothing else can change the ratio of input to output voltage like they can.

Signals already at (or very near) line-level may need only a small gain adjustment, if any, and so they can be connected to a “trim.” A trim is a gain control with a more limited range…and, often, the ability to reduce a signal’s level.

If only one input point is available, it’s very likely connected to a gain control that can trim down line-level signals, while also having enough positive gain available to work with mic-level inputs. (Or, a mic pre with an available PAD).

Digits

As a quick aside, let’s discuss the difference between analog and digital systems. In an analog console, signal voltage is passed directly from the input stage to everything else. In a digital system, the signal is passed to a converter, which then sends data along. In either case, an appropriate drive level from the input stage is necessary – and although the “technology bases” are different, the general behavior of the console regarding signal levels is unchanged. A good, healthy, line-level signal is necessary, whether that signal will be passed as analog voltage or data that represents a voltage.

Leveling Off

So, when setting up your level from the preamp or trim, what should you look for?

A basic rule of thumb is to shoot for 10 – 20 dB below overload. This gives you room for the input to get louder without clipping, while avoiding being too low. This also gives you some room to make changes with processing and other level controls later. It is NOT necessary to “get as loud as possible without clipping,” especially because trying for that tends to lead to levels that are too hot. On the contrary, in most cases you’ll be just fine if the signal is reaching the middle of whatever metering you have available. If all you have is a “Signal Present” light, you’re probably in decent shape if the light is continuously illuminated during louder passages.

Please do read up on the manufacturer’s specifics for metering on your console. Knowing what all the lights and numbers mean is very helpful for proper operation of your equipment. Also, be aware that digital consoles often use a different dB scale than analog mixers.

The previous installment of this series dealt with how sound must be represented as electricity for us to work with it in the modern, pro-audio context. Mostly, we talked about the transducers you’re likely to encounter – microphones, that is.

The next step is to get that electricity passed along to the input stage of a console or other device.

“Wire” You Looking At Me Like That?

The simplest and most robust connection possible is a single cable carrying analog electrical signals. Analog cabling is subject to many problems, of course, including noise induced by electromagnetic interference. However, its simplicity reduces the number of ways that an outright failure can occur, and the connection tends to degrade “gracefully.” In other words, the cable will continue to pass some kind of signal unless it’s completely unable to function.

A cable is a bundle of conductors that have different roles in a successful connection. Audio connections require a minimum of two conductors: Signal (Hot, +) and Ground. It is, of course, entirely possible to bundle more conductors into a single cable, with the extra conductors having different roles. A three-conductor system might be used for a balanced connection, with Signal (Hot, +), Return (Neutral, -) and Ground…but this isn’t the only possible configuration! A three-conductor cable might also be unbalanced stereo, with two “hots” and a shared ground.

Cables do not have brains, and don’t “know” which application they are meant for – although certain cable constructions are better for different situations. The application, or interpretation of the carried signals is up to the manufacturers of input and output devices.

Balancing Act

In pro-audio, there is a definite preference for “balanced” connections. As stated above, a balanced connection requires Signal and Return conductors for each audio signal being carried (along with a proper Ground conductor scheme as needed).

Balanced connections are helpful because of their ability to suppress induced noise. Interference gets “into” a balanced cable just like any other cable. All cables function as antennae, and the longer the cable the more functional it is. With a balanced connection, though, the induced noise is the same on both conductors, whereas the actual signal is inverted on one conductor. Balanced input stages are meant to accept only what is DIFFERENT between the two conductors, while – to some degree – canceling anything that is common. If the signal on one conductor is inverted, but the noise is the same on both, the signal should be preserved while the noise largely disappears.

Because of what I said above (that cables are unaware of how they’re supposed to be used), you can theoretically put connectors on any cable with the appropriate number of conductors, and use that cable in a balanced situation. However, cable made specifically for balanced connections will use “twisted pairs,” so that the noise exposure for any given conductor is the same along the length of the cable. Cables built for unbalanced use may not have any twist, which can cause one conductor to be more susceptible to noise. If the noise on one conductor is greater than on the other, the differential input stage will happily amplify the difference and pass the noise along.

I’m A Terminator

In a pro-audio context, we have a strong tendency to name cables by referring to the connectors attached to the ends. (We could connect everything via bare wire, but that wouldn’t be quick or convenient.) It’s important to note that cable termination does not necessarily guarantee that the cable will function a certain way.

For example, I can terminate an unbalanced cable with a connector generally meant for balanced operation. The cable will never actually be capable of carrying a properly balanced signal, even though the connector is capable of doing so.

Some common termination types – and thus, cable names – are:

— 3-Pin XLR, commonly referred to as XLR, or “mic-cable.” XLR connectors offer robust construction, with relatively large electrical contact points arranged in a triangle, and the ability to latch built in at both ends. Please note that XLR is actually a reference to an overall connector form-factor. An “XLR cable” might have connectors with 5 pins instead of three. 3-Pin XLR is most often used for mono, balanced audio, but there’s nothing to prevent the connector from being utilized in other applications. (Some older equipment even used 3-Pin XLR for loudspeaker connections.) XLR cables are most often wired with a male end and female end.

— TS (Tip-sleeve), often called a phone cable or guitar cable. TS connectors and their variants are available in various sizes, with 1/4″ and 1/8″ diameters being the most common. TS and other similar connectors are not quite as heavy-duty as XLR, because their “all in a line” construction can be more easily damaged by sideways force, and also because TS connectors are not often built with latching capabilities. The tip being larger than the sleeve offers some protection against accidental disconnect, though, and some manufacturers have also created latching jacks. TS cables are often seen with two male ends. TS cable finds applications in mono, unbalanced audio applications, along with certain switching operations.

— TRS (Tip-ring-sleeve), also called a phone cable (confusingly with TS), or stereo phone cable. TRS can be used for stereo audio, balanced, mono audio, and switching applications where two switches are to be addressed with one connection.

— TRRS (Tip-ring-ring-sleeve), which ALSO may be called a phone cable (causing even more confusion), or headset cable. TRRS, due to having four connection points, commonly finds use where stereo, unbalanced audio travels to a receiving end, and a mono, unbalanced signal travels from a sending end located on the same device – such as a headset with a built-in microphone.

Multiple TS variants can be physically inserted into a mismatching jack, with varying results. Cables with “too few” connection points will often seem to work normally when plugged into a jack with more connections. The opposite, however, is much less certain. TRRS-equipped headphones, for instance, can’t be counted on to pass audio as expected if mated to a TRS headphone output.

— RCA, also called a pin-jack. Any single RCA connector only has the electrical capabilities of a TS connector, limiting it to unbalanced, mono audio or single-point switching. Two or more cables with RCA termination are often combined by the manufacturer for convenience. RCA connectivity is even less robust than other types, especially as simple friction is the only barrier to a cable being inserted or removed. Many RCA cables are wired with male connectors at both ends.

A cabling “super-type” is that of the snake, or multicore. These are effectively a cable of cables, meant to help collect and run a number of connections in a single, physical bundle. Snakes come in many varieties, with many combining cables with different termination types. Some include a stagebox for one end, which may be removable if the snake is a high-end model.

Conversion

At some point, you will very likely encounter a situation where a device producing an audio signal is incompatible with the receiving device, such as a mixing console. In certain cases, this is a purely physical problem, such as a balanced output on TRS faced with a balanced input on XLR.

In such a situation, the only conversion necessary is a simple adapter. There’s no problem to be solved with the signal itself. Rather, the connection for the signal needs to be reconfigured. A simple adapter has no electronics, but only the necessary wiring required to pass the audio straight through to a different connector. You must be careful that the wiring is as you expect, however. Some TRS to XLR adapters don’t actually include all three conductors!

Simple adaption is also all that’s necessary in many situations where you are reducing electrical complexity, such as mating a balanced output to an unbalanced input.

The problem gets a little more complex when you need to convert an unbalanced signal to a balanced one, and/ or where the signal producing device can’t effectively drive the console input. (The latter case is an impedance problem. A real discussion of impedance is beyond the scope of this series, but a tell-tale sign of an impedance issue is a connection where the signal seems surprisingly weak and sounds VERY poor.)

When you need to improve/ upgrade the ELECTRICAL part of the equation, a simple adapter is insufficient. You will need intervening electronics in the form of a DI box or similar device.

Some of these converters are very bare-bones, stuffing an electrical transformer into the body of an adapter and doing nothing else. Others may be a bit more full-featured, operating as an actual DI box with, perhaps, a pad for high-level signals, a pass-through, and a ground-lift on the output. In both cases, though, the internal transformer does all the real work. The signal is converted to a balanced output, and a reasonably broad range of impedances can be successfully handled.

These simple devices are especially easy to use because they are passive, requiring only the input signal to be present in order to function. They can’t handle every possible input situation, though, because their transformers have inherent limitations regarding input impedance.

Active direct boxes are more flexible at the cost of complexity. The core of an active DI is a circuit involving an op-amp which requires some sort of steady power supply to operate. The op-amp can be effectively driven by almost any input, though, making active DIs compatible with pretty much anything you’re likely to encounter. A reasonable rule of thumb, then, is to use an active DI box whenever you’re in doubt about what kind of DI is appropriate.


So…

You’ve done some playing and singing. You’ve written some songs. You’ve got this whole “music” thing down to some degree, and now you’re thinking about gigging or recording.

But you’re bewildered. You don’t know how to get started with the maddening, intimidating, even terrifying pile of hardware and software that gets used in modern production. This series is for you. It should help you understand a little more about what’s going on, so you’re not as mystified.

We’re Going In!

Obviously, what we’re working with is sound – a vibration in something physical that we can hear. Any real dive into the physics of sound is beyond the scope of this series, but you should be aware that all sound:

1) Has an intensity, or amplitude.

2) Has a rate of vibration, or frequency.

Sound has other properties as well, but these two will be the most important for a basic understanding.

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

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

Mr. Microphone

Transducers come in various forms, but the most commonly recognized sound-to-electricity transducers are microphones.

Microphones come in a large array of sizes, shapes, and behaviors. They all derive from one of two basic flavors, though:

1) Dynamics, which use wire coils and magnetism to generate current.

2) Condensers, which create a “variable capacitor” to produce current.

You should be aware that there are sub-categories for each basic flavor, such as moving-coil dynamics, dynamic ribbons, “active” dynamics, electret condensers, tube-amplified condensers, and whatever else the industry can cook up. However, in the most common scenarios, what you can keep in mind as a baseline is that dynamic mics don’t fundamentally require a steady supply of electricity to work, whereas condensers do.

Another generalization that can be made is the overall character of the microphone flavors. Although all microphones react quickly by human standards, dynamic microphones have moving parts which tend to be “heavy.” The moving portion of a condenser microphone can have far less mass, which makes for a vibration sensor that can start and stop moving very easily. Condenser mics are a common choice for the transduction of quiet, “delicate,” or “complex” sounds, and condensers can more easily be extremely accurate – but this does not necessarily mean that condensers are correct for what you need to do. There are plenty of dynamic mics which sound very pleasing on a tremendous variety of sound sources, and they tend to be more resistant to accidents and mishandling (although dynamic ribbons can be very fragile indeed).

Microphones also differ from one another in terms of their directionality, or the relative sensitivity of the microphone at different angles around the microphone element. This is also referred to as the “polar pattern,” in reference to how this directionality is commonly plotted on specification sheets. In terms of the basic microphone types, any directionality is possible. There are omnidirectional dynamics and ultra-selective condensers, and the opposite is also true. A list of common polar responses includes:

1) Omnidirectional, which has essentially the same sensitivity at all angles around the element.

2) Figure-Eight, which is sensitive to the front and rear, and tends to reject sound from the sides.

3) Cardioid, which is highly sensitive to the front, somewhat less so at the sides, and has a point of very low sensitivity at the back.

4) Super-Cardioid, which is highly sensitive to the front, less sensitive than a cardioid at the sides (with a particular side angle which is very low sensitivity), and has some sensitivity at the back.

5) Hyper-Cardioid, which is like super-cardioid, but narrower and with a more pronounced “sensitivity bump” for sounds coming from behind.

In many applications, mics with strong directionality are often preferred and even necessary. However, omnidirectional transducers see quite a bit of utilization as well, especially when accuracy is needed or tonal consistency at varying distances is required.

Contact

To close this installment, it’s worth talking about another kind of transducer, the “contact mic.” Contact transducers aren’t really microphones at all, in the sense that they are not designed to work well with sounds in air. Rather, they are intended to be fixed to a vibrating surface, which causes the element to deform or flex and thus create an electrical current. This is a piezoelectric effect, and so these pickups are often referred to as piezos.

Contact transducers generally sound rather artificial when compared with microphones, but most microphones aren’t in direct physical contact with a sound source. At the same time, piezo pickups can be very handy for dealing with certain problems, like instruments which need to be made disproportionately loud with minimal feedback.

Back in the heyday of the Grateful Dead, a special sub-scene emerged: The Tapers. Not to confused with tapirs, an exotic animal, Tapers would record the live shows to share with other people later.

Does that sound familiar?

I would argue that, in many ways, livestreaming your show is a new form of taping. It’s an attempt to capture part of the experience so as to give something to your current audience, and hopefully reach some new enthusiasts as well.

The thing with taping or livestreaming is that the physics and logistics have not really changed. Sure, the delivery endpoints are different, especially with livestreaming being a whole bunch of intangible data being fired over the Internet, but how you get usable material is still the same. As such, here are some hints from the production-staff side for maximum effectiveness, at least as far as the sound is concerned.

1) Directional microphones are your friend.

While it might seem like a good idea to grab a wide, or even 360 degree soundfield, you will generally get a better result overall by being selective. Especially if you’re streaming from a bar or club, it’s really not a great idea to capture all the conversations, room reflections, and general disruption happening around you. A full-on shotgun mic probably isn’t necessary; Just find a decent cardioid or super-cardioid and point it at what you want to hear.

2) Keep your gear out of the way. Super out of the way, actually.

Audiences have an incredible ability to walk into, stand on, swat, and otherwise mess with your recording setup, often without even trying. Endeavor to find a spot where your streaming goodies are protected from the general public. The audio human’s spot can be pretty good for this. Just remember to ask politely first.

3) Run your own gear as much as possible.

As a sound operator, I am (as a rule), happy to help by pressing record on your device. However, it’s important to understand that the start of a show can be a bit like getting an airliner off the ground: A lot is going on that requires my close attention. I may end up forgetting to hit the little red button. If you can do it yourself, that’s much better.

Also, if there is any complexity at all to getting things rolling (beyond just pressing the aforementioned button), you really should take care of it yourself. It’s THE way to ensure success.

4) A direct feed might sound better, but…

…remember that many direct feeds are just a split from some output, often the main bus. There are many rooms and situations where the main bus is carrying a ton of vocals and just a touch of a few other things. Unless the PA is truly doing all the heavy lifting, you may find that a line-level feed isn’t musically balanced.

I like clean audio as much as anybody – maybe even more – but I can also recognize when “clean” isn’t necessarily the best capture of the show as a whole.

(There are some ways around this conundrum, but they are beyond the scope of this article.)

5) If you want a feed, please do your advance work.

Find out the day before, or earlier, what kind of connections and signals might be available to you. Sometimes, it’s easy for a sound tech to get something sorted out for you…and sometimes, it’s nearly impossible. The difficulty generally rises as the amount of time before the show decreases.

And please, please, educate yourself on the different kinds of audio connections that you might run into, and have your own adapters. Again, when speaking for myself I can say that I’m happy to help out in whatever way I can – but it’s always best when YOU are “Johnny On The Spot” in terms of having what you need to make your own gear play nicely with everything else.

“Hey, Man. It’s a touch loud in the house. Can I trade you some amp volume for monitor gain?”

“But my amp’s only on, like, two!”

Have you been part of a conversation like this? I have. It rarely ends well, because somebody is always frustrated or disappointed at the end. Oftentimes, there are at least two somebodies: The audio human and the amplifier user.

The sticking point in the debate is an idea that “low knob position = acceptable volume.” Unfortunately, this notion is anything but watertight. The reality is that acceptable volume = acceptable volume, with the position of any relevant control being nearly immaterial.

To put it another way, the position of the knob is the cause, and the resulting audio output is the effect. In the end, the effect is what matters. If the effect is causing a problem for the band, then the “state” of the cause isn’t a valid argument that the overall result is okay.

Nobody has ever fought a speeding ticket by claiming that the car’s accelerator was only a third of the way down.

The same reasoning also applies when the disagreement ventures into drive percentages. Somebody might say, “I’m only using about 10% of the amp, and for it to sound right I need at least 40%.” That’s fair enough in some respects, but it points to an issue of bringing an artillery piece to a neighborhood cap-gun game. If the amplifier doesn’t sound good until most people think it’s too loud to sound good, then the amplifier doesn’t actually sound good.

It’s the wrong tool. And the wrong tool at the right price, or with the right look, or with the right capabilities for some other job is still the wrong tool.

If two is too loud for the band, then two is too loud. If you’re finding yourself in this kind of situation, it may be time to do some horse-trading. Find yourself a rig that’s just a little too hot for the band when it’s wound up all the way, and you’ll have lots more room to actually use the front-panel settings for creative control.

You might even end up with something easier to carry, as a bonus. (Maybe.)

Dear Musicians,

Over the years that I’ve worked with you, many things have become apparent. One of those concepts is that, quite often, you need me to make some sort of change in the middle of a show. Often, that change is necessary to make your life on stage more comfortable, such that you can create the best possible experience for your audience.

At times, it may have been hard to get that change made for you. Such difficulties commonly arise due to communication problems. As such, I am writing this letter to help you transmit your needs and wants to the audio humans you will inevitably encounter.

First and foremost: Please use your words.

I understand that there is a stubborn stigma attached to “talking through” an issue in the middle of a show. However, any aesthetic problems this can cause are quite minor, especially when you consider that not getting a need met may cause real problems with your performance.

When it comes to a complex topic, especially in a pressure situation, the ability of spoken language to convey nuance and relay information unambiguously is a huge bit of leverage. By speaking over the PA, you can make it very clear, say, that “I think my vocal is starting to feed back in the highs.” There’s actually a lot of information in that sentence, yet it comes across quickly and elegantly when turned into speech.

On the flipside, I’m not sure how that concept would be effectively transmitted by way of hand signals – unless there was a lot of rehearsal time with the engineer involved.

Also, concerts are full of distractions to the eye. A sound operator may have their visual attention elsewhere, while still devoting their ears to the music. As such, addressing them over the PA is generally a sure method for getting their full attentiveness returned to you.

My second point is in regards to visual signals: Think big, think simply, and think patiently.

When you don’t have the opportunity to verbalize a request, visual communication is a necessity. However, as I’ve alluded to already, it has limitations. You have to restrict yourself to basic concepts that have a small number of interpretations, and require no rehearsal to understand.

(Many years ago, I had a musician attempt to take me through a large number of hand signs that would convey things like “The stage-left guitar needs more midrange in the monitors” and “Less reverb on my vocals for this next tune.” I can’t say that it worked out very well.)

Simplicity and “largeness” go together. Remember that the audio engineer may be quite a distance from you, causing detailed motions to become lost. Ad-hoc sign language at shows must be “big” so that it can be seen, and only so many ideas should be signaled in a short period of time.

I highly recommend the approach of “Who, What Instrument, Where, and Up/Down.” For instance: Point at the guitar player, mime the guitar playing, point at your monitor, and then make an up or down motion until the guitar level is where you want it. It’s compact, relatively unambiguous, and the involved motions are easy to see.

As to patience, please do remember that it takes time to interpret your signals, figure out how to get you what you want, and then start to make it all happen. Several seconds may have to elapse before you hear any change, and some “iteration” may have to take place before you’ve gotten exactly what you want. This is simply an inherent hazard of doing things on the fly, but when taken in stride it’s not too hard to handle.

Hopefully this all makes sense. Effective communication is important for a good show, and a little bit of forethought about how to go about it can make a huge difference.

Thank you for taking these thoughts into consideration.

Your friend,

Danny (An Audio Human)

I seem to be on a bit of a theme lately.

The last time around, I talked about how most bands don’t need more or better gear to solve their problems. Mostly, they need to work as a team.

That idea closely ties in with equipment used to reproduce the sound of the band and it’s gear. You know – PA systems. There’s a myth about sound-reinforcement gear which can be voiced in many different ways, but usually boils down to this: “This problem will get better when we’re on a big stage, with lots of monitors and a big FOH system for the audience to listen to, all with enough power to melt somebody’s face off.”

You know what I’m going to say, of course. The above is not true.

Bigger and better reinforcement rigs are sort of like fortune or wealth, as understood by Marie-Jeanne Riccoboni. She said: “Fortune does not change men; it only unmasks them.” In the same vein, I can tell you that more and better PA rarely solves a problem with a band. Rather, it confirms the problem, or makes it more obvious.

I’ve been in more than one situation where the monitor system was far, far better than what a band was using in rehearsal. We had much more power, better initial tuning, and a ton of EQ available. Do you think the poor singer could finally hear themselves?

Not really. All that the extra toys did was confirm that the rest of the band wouldn’t give the vocalist any room to work. They were convinced that pro-audio could make up the difference in their teamwork (or lack thereof). Unfortunately, the difference was too great to be mended. There wasn’t enough gain-before-feedback to undo their steamrolling.

On the other hand, a PA becomes a powerful tool when used with an act that sounds balanced and beautiful right out of the gate. In that case, the system’s reserves can be used to optimally translate the group into whatever space they happen to be in that day. Tasteful sweetening can be applied, just as one might season a bit of carefully prepared food; Good ingredients can be enhanced, but bad ingredients will stay bad.

There are limits to these metaphors, of course. In some cases, an engineer can use a powerful system to blast over a problem. Depending on the situation, this might result in a tolerable sound. It might also be so loud that half the audience leaves. Even so, the need to take drastic measures is an unmasking: It tells you that something is very wrong somewhere.

A great PA with an experienced operator won’t fix inherent flaws with your music or performance. What it will do is make them obvious, because everything that can be improved will be improved. The unsolvable problems, then, will remain…unmasked.

I don’t think I’ve ever been a fan of any “Battle of the Bands” setup, but I’ve been a judge for a couple of them. People asked, and it was something to do.

After one such outing, a band that didn’t win was curious as to what had prevented them from reaching the top of the podium. Having conferred with one another, they had identified at least one potential “deal breaking” problem – and they asked about it:

“Do we need better equipment?”

The answer that day was “no.” The answer for most bands on most days is “no.”

What they had failed to do was to play as a team, and that made their perfectly adequate gear SEEM like a problem area. (To be specific, you couldn’t hear anything the fiddle player was doing, because nobody would give the poor guy any space.) So, of course, the answer is to spend money on a bigger, fancier amp for the fiddle player, along with some extra doodads and geegaws to fight the inevitable feedback that results from trying to make a fiddle SCREAMING LOUD…

…Right?

People, please.

Their gear wasn’t fancy, but it was adequate and working. The only upgrade they needed was teamwork.

Now, yes, there’s a point where instruments, amplifiers, and their associated accoutrements just can’t do the job. However, that point is best identified as an “absolute:” The setup just sounds terrible, or it’s constantly breaking down, or it’s too hard to use. If that isn’t the case, though, then it’s very likely you’re facing some sort of issue with working together properly.

If your band doesn’t sound right, but everything seems to be working decently for everyone individually, you most likely need to put your wallet away. Before you spend any money on stuff, spend time on becoming a team.