Analog Mixers

The following article is a fairly long one, but it will give you a great jumpstart into the world of mixing! It has been edited many times and likely will be edited many times again as I find tidbits that need to be brought up in the context of a basic rundown of analog mixers and mixing principles. For those of you who are either looking to present a training session on mixers or those who would like to have an abridged version of this page, here is a boiled-down version in the form of a handout. Feel free to distribute this information as well, but do not edit the document or use any of this material without my permission and giving credit where credit is due. This material is not technically copyrighted, but I have spent a lot of my time putting it together.


You know those things that you look at and think, “That’s so complicated. I’ll never become proficient at that”? Mixers (a.k.a. “sound boards,” “mixing desks,” or “consoles”) are in that category. Fortunately for you, that is a completely inaccurate statement. Mixers are not difficult, and they’re not even as intimidating as they look. Just a few seconds of explanation by an experienced sound engineer and you’ll understand that mixers are a lot simpler than they look. Before heading over to look at the digital mixers page, read through this one thoroughly. Most of the things you’ll learn here are applicable there, and it’s both better and easier to learn them in the context of analog mixing than in the context of the digital world. Here are a few examples of different sizes of analog mixers: Peavey PV6 — 6 channels Peavey PV6 Mackie 1604 VLZ Pro — 16 channels, 4 subgroups Mackie 1604-VLZ Pro 844x855 Solid State Logic Duality SE — 24, 48, 72, 98-channel options, integration w/ Digital Audio Workstations (DAWs), and much more Solid State Logic Duality SE 4358x2004 There are four major sections to any mixer. They are: 1) Channel I/O, 2) Channel Strips, 3) Master Section, and 4) Master I/O. Now that you know the basic signal flow of a sound board and you’ve seen some of the variations on the analog mixers, let’s dive into the general build of the mixer sections. Note: This article is somewhat lengthy and may need to be digested in chunks. I’ve tried to arrange it so you can cover sections individually as you like or take it in all at once, going from the input to the output of a mixer.

1) Channel I/O

The channel I/O (a.k.a. “inputs/outputs”) are where the signal is received into the board and begins the process of, well, being processed! Think of it this way. A tree as it stands (pardon the pun) is no good for building a house — what you need is lumber before you can sell it to a contractor. The uncut tree is like a guitar, a microphone, etc. — it is still in its raw form. It needs to be made useful as building material, more affectionately known in the audio world as a signal. It is cut roughly (guitar pedals, DI boxes, etc.) and transported to the factory (XLR cables, snake, etc.) where it is received and sorted into the proper areas. These different processing points, or loading bays, can be compared to the channel I/O on a mixer. There are a few different types of connections in this section of a mixer. Here is a picture of the back of a Presonus StudioLive 16.14.2. It is a digital mixer, but it still includes the same inputs as an analog board its size might have:

StudioLive Rear

Note that there are both the channel I/O and the master I/O in this picture, as they are grouped together on all boards regardless of rear-facing, front-facing, or upward-facing I/O connections.

The channel I/O section is the beginning of your control over incoming signals. Everything you do elsewhere on the board is directly affected by what you do here. There are, on boards bigger than those like the PV6, multiple types of input jacks, typically balanced XLR and balanced or unbalanced 1/4″ (accommodating either one), though the 1/4″ inputs are sometimes only unbalanced. Every now and then, you’ll also see RCA inputs, but they will typically be just one or two sets at the end of the board, say on channels twenty-three and twenty-four as a stereo pair. These channel sets are normally combined into one stereo fader, since there is a L (channel 23) and R (channel 24) input for RCA.

   — XLR Input

This is hands-down the most-used connector on the mixer. While other connectors are quite commonly used, none will beat this one for common usage. Snakes in almost every case terminate in an XLR input (though they often include several 1/4″ terminations as well), which is generally the main source for signals moving into the mixer. This connection is also used for most microphones, thus the cables are commonly known as microphone cables. These are identifiable by the ring on the edge and the three holes in an upside-down triangle near the bottom. These three holes accommodate three pins in an XLR cable, which are labeled 1, 2, and 3 (generally ground/shield, hot, and cold wires, respectively). You could think of them as microphone inputs, as that is the signal they handle. This is a low-impedance signal, being low-impedance either because of the nature of the source (microphone) or because a component in the signal flow (DI, low-Z output on an amp, etc.) made it so. Some mixers have a switch or button that indicate which input you are using, whether it be the XLR or the unbalanced 1/4″ input. Not all mixers have this switch, but you should only ever use one or the other, not both.

   — 1/4″ Input

Some mixers move upward, physically, as you go through the I/O list and some go down. Regardless of direction, the next connection to note is the 1/4″ input. It typically accepts both balanced and unbalanced signals, but not always. Check your mixer’s manual if it is not printed next to the connection and you need to know. It is rare that it will be important for your needs, but it is possible. Line-level signals need to be attenuated in order to prevent overloading the circuits in your mixer because they are stronger than mic-level signals, so-called because of the weaker signal received from a typical microphone. There is a pad added to this signal path before it hits the preamp in order to bring those line-level signals under control and keep them from burning out the circuitry by overloading it. Other than this difference, it is usually internally identical to the XLR input. Keep in mind, again, that you should only ever use one input or the other, not both. If your mixer has a switch on it, check this switch’s setting if you are troubleshooting a no-signal situation at your input stage (i.e. no signal on the meter when soloed, the “Signal” LED by the fader is not lighting, etc.)

   — Insert

This connector is used when you wish to connect effects units or processors that are intended to be inserted rather than run as a separate line out. This is often the case when you have a compressor you wish to add to a single vocalist, or if you wish to add a special EQ unit to one channel alone rather than the entire mix. This connector is unique in that it requires only one cable termination in order to push a signal out as well as receive one again. This is attained by utilizing a special TRS (Tip, Ring, Sleeve — a balanced 1/4″ termination) 1/4″ cable, which sends the signal out via the tip and receives the newly processed signal via the ring, with the sleeve functioning as the grounding. The other end of the cable is actually two terminations, which is how one achieves the input and output separately. The termination labeled as the tip is intended to go to the input of your processor, whereas the other termination is intended to go to the output of your processor. In this way, a completed circuit is achieved with the original signal altered to your taste via the external processor while using only the space of one connection on your mixer. Cool, huh?

See the Aux Section further down the page for a bit more discussion on use of the Insert for effects versus the Auxiliary Outs.

Mackie notes that you can use the Insert as a direct out as well, with two types of output available to you based on whether you push the cable all the way in (two clicks) or only partway in (one click). This is possible because the Insert connection is a cutoff switch only when contact is made with the second click, the intended return from your hypothetical processor.

   — Direct Out

Not all mixers have a direct out. These are extremely handy for certain applications, such as running a personal monitor situation. If you’ve never used them before, then check out personal monitor mixers. They’re great, and Sweetwater has options if you want to change your monitor setup (they have awesome customer service). The way this connector works is that it effectively duplicates the signal at the Direct Out’s insertion point in the signal path (whether it is pre-EQ or post-EQ, etc.) in order to send it elsewhere and leave the original untouched, dancing its merry way down through the channel fader and the rest of the mixer. Though it is quite useful for some, it may remain untouched by others running a more simple audio setup. This is one reason why you will not find it as often on smaller mixers. An entire row of connectors that may go unused in a small club or back room somewhere? That’s just wasted space! Open up a mixer, a preamp, or any other unit with in/out jacks and you’ll know what I mean. But for some, it is an indispensable feature. Suffice it to say that Direct Out jacks may or may not be there and simply copy the signal for external uses.

2) Channel Strips

The biggest reason many people look at a sound board and say, “Wow! That’s so complicated!” is because of the channel strips. The bigger the board, the more channel strips, so the more knobs and buttons there are to confuse the average onlooker. Analog mixers, however, are basically the same no matter how big they are. The sections of the channel strip that will be found consistently across the board are the 1) Gain/Trim/HA, 2) Aux section, 3) EQ section, 4) Pan knobs, and 5) Fader section. The channel strip will always be the largest portion of a mixer (there are exceptions in the digital realm, but even those are few at this point in time), so once you’ve learned the channel strip, you have learned most there is to a sound board. Sounds pretty good, huh? Let’s dive in.

   — Gain/Trim/HA

After the signal comes in from the cable through the jack, it hits the familiar Gain (a.k.a. “Trim” or “HA”) knob (a.k.a. “pot”). But wait — why are there three names (gain, trim, and HA) for the little knob at the top of the channel strip? From what I understand, it comes from different ideologies concerning the character of said knob. For some people, its job is to boost a signal, such as when a microphone is plugged in. Microphones usually have a very low output and need to have more gain to work with the signal. An electric guitar, on the other hand, may have a very “hot” (loud) input level, which is where you will need to “trim” the input level so it doesn’t overload the circuitry. Head Amp is a term I first found with a Yamaha M7CL digital console. I would guess that it works with the same basic idea as gain, but I believe that it refers specifically to the preamp driving the channel. There is an article that you may find helpful in explaining further concerning Gain vs. Trim in Sweetwater’s inSync publication. When setting up a new input, you want to make sure the gain is set to an appropriate level. If it is too hot, it will overload the circuitry, as I said before. If it is too low, it will require you to boost the signal elsewhere in the signal path — comparable to having one person lift a piano with his back while another grabs a corner with one hand and calling it good. Each point of signal level alteration must be kept at a similar level, much as a box of sand. There are dips, there are hills, and there are few places with a perfectly flat spot. In the same way, the various gain/cut points (gain, channel fader, master fader, etc.) have leeway to differ but will be overall somewhat similar. This leaves space for creative mixing and contouring of different signals while retaining an equilibrium throughout the signal path. For instructions on setting up a new input, see the Solo Button section further down on this page.

   — Aux Section

If you want to run a separate line from your mains, this is the way to do it. The Auxiliaries are designed to send individual channels to a dedicated one-way output. This section has multiple uses. The obvious one, as I’ve already mentioned, is that you can have a line run out of the mixer that is not linked to the mains. Churches use this to send a signal to the nursery, to a recording unit, and to other places for those not in the service to listen in. One of the benefits (and disadvantages) of the auxiliary feature is that it allows (or requires, depending on how you look at it) a separate mix. This is because the auxiliaries have dedicated volume knobs.

There are two types of auxiliary sends: Prefader and Postfader. They have their individual uses, though I find the prefader to be much more useful. The difference lies in where the send fits into the signal chain. If it is prefader, the send knobs will be the final source of volume change available to you, leaving only the gain knob as the other control. If an auxiliary send is postfader, then the changes you make to the main mix via the faders will alter what goes out through the aux send. Think of when a guitarist brings down the volume pedal for his guitar. The volume coming out of his amp will drop because the signal level coming in is lower. Another way to think of it is to use a water analogy (it’s a common analogy to use in sound in many situations). If you have two handles to control the flow and you cut the first one to half, the water coming out of the end will be a lot less. This analogy sort of breaks down when you look at the boosting aspect of volume control. If that same guitarist adds a booster pedal to his setup right before his pedal, the signal coming out of the amp will be louder. Adding a water pump into a pipeline can only help so much. Anyway . . .

Another use for the auxiliary sends is for inserting time-based effects for your inputs. While there is an “insert” option for each channel on many boards, that does not allow for 1) mixing dry/wet signals (original vs reverb, for example) or 2) using the same effect for multiple channels unless your effect unit is made with multiple inputs. For example, you may have a reverb unit that you like very much. The problem is that certain instruments in your mix do not need any (or much) reverb, such as the bass. Maybe one of your vocalists is already difficult to understand without adding in the intelligibility problems that come with a lot of reverb. If you used a channel insert, you will probably only be able to use the reverb on one channel, since they tend to have stereo inputs at best, let alone four or six inputs. If you inserted your reverb into the main insert, everything would have the same reverb, making certain instruments and/or vocalists muddy. Auxiliary sends to the rescue! Run the auxiliary output to the input of your reverb unit and the output of your reverb unit back into an empty channel of your mixer (preferably in one of the uppermost channels so it stays separated). Your fader in this new channel will be your master reverb level, and your aux sends will be the reverb level for individual channels! As for the dry/wet issue, your faders are the dry signals — straight vocal, no reverb — and your aux send pots are the wet signals — all reverb, no straight vocal — and you can mix accordingly! Another effect that fit well with this method is a delay.

Processors, as distinct from effects, do not belong in aux sends, however. If you have a compressor that you want to use with multiple vocals but it has too few inputs, it won’t work properly to run the vocals through an aux and the compressor back into another channel. Consider the basic idea of a compressor: it pulls back on levels that are too high and boosts levels that are too low. If all three vocalists are singing, they would all need different levels of compression based on their singing styles, so that is a problem. If all three are singing, they might trigger the cut feature of the compressor. If you raise the headroom to eliminate that problem, it won’t work properly when only one person is singing. It’s just a mess to try and combine channels for processing in the same way that you can combine effects.

Some differentiate the category in which reverb belongs from the category in which compressors belong using the terms “parallel processors” and “serial processors.” Mackie offers a brief answer to the question of how to connect these two types of units in the FAQ page of the technical support portion on their web site.

   — Equalization Section

If you’ve ever heard the letters EQ used together concerning music and audio, chances are the person was talking about equalization. Well, actually there’s nothing else they would be talking about, so they were definitely talking about equalization. So what is it? First, let’s look at some principles of sound waves.

Note: I highly recommend not skipping this section. It may sound like the classes from school that you tried to forget, but trust me when I say that it will make you a far better sound engineer if you understand the basic concept. I am no mathematician, but I believe that I have a firm enough grasp to explain it to others, and it has certainly helped me identify problem frequencies for feedback, bad guitar tone, and other issues.

Sound Waves

Do you remember the sine wave from high school? You may remember that it is represented by a curved wave going above and below a horizontal line right in the middle. There are two things that can change in a sine wave: amplitude and frequency. Amplitude measures the strength of the signal, or the loudness in layman’s terms. The frequency (or “wavelength” in subjects such as light)  is how fast the wave passes the center line, measured as Hertz. This is how many times per second the wave completes one full cycle of going above and below. The more times repeated, the tighter the pattern will look on a screen — or, as often seen in the classroom situation, on a blackboard or whiteboard. Click on the gif below to see a sine wave increasing in frequency from 1Hz to 5Hz.

Wave Frequency

Here is a picture comparing several different wavelengths, or pitches.


As the above picture suggests, the more times the up-down cycle is repeated in a set time period, the higher the pitch. Female singers, for example, are using higher frequencies than male singers. The frequency in their case is from their vocal cords, tiny muscles in a person’s throat that vibrate from the air passing over them. Shorter wavelength, or higher frequency, equals a higher pitch. When someone is tuning his guitar, he is increasing and decreasing the frequency at which the string vibrates until it hits a specific frequency. There are many, many frequencies that are not actually used in our western theoretical system of music. If you do a search on the internet for music theory, you should find a plethora of articles on the subject. I find it absolutely exhilarating, but that’s me. Suffice it to say that specific pitches (not notes–a music theory teacher would scold you for confusing those terms) equal specific frequencies.

INTERESTING FACT: The average human ear registers frequencies within the range of 20Hz (cycles per second) and 20KHz (20,000 cycles per second!). That means that if something is vibrating below twenty times per second, most people cannot hear it! Same goes with sources vibrating above 20K per second. I have no idea how high a dog can hear…

So that’s frequency, but what about amplitude? Here’s a picture of a consistent wavelength that drops in amplitude, thus reducing volume output.


As you can see, the frequency remains constant but the amplitude drops off. An example of this can be seen when the guitarist hits a chord or a single note and lets it ring out. What happens? It sounds the same, but it gets quieter. The above picture is essentially what is happening with this ringing pitch. Now that we have at least a simple grasp on how sound waves work, let’s move onto what the EQ section manipulates the sound coming from the source.


Have you heard people say that they want more highs in the vocals or less mids from the bass? The knobs in the EQ section let you manipulate the heck out of your tone, if you so choose. Every pitch coming from a traditional source like a guitar or a vocalist has more than just what is called the Fundamental Frequency. This is where it might get a little confusing. Remember how there is a specific frequency that causes a pitch? Well, any natural sound source in the world gives the frequency required for the pitch plus some extra frequencies. These other frequencies give the sound flavor by their unique combination. The specific extra frequencies included in a dog’s bark, for example, gives it a sound separate and distinct from that of his master’s voice. These extra frequencies are called Overtones. If you want to go down that bunny trail (which I would highly recommend — just not right now), here’s another link to Sweetwater’s inSync publication to get you started.

The overtones are what make different parts of the band sound distinct from one another, and one band’s guitarist unique from another band’s guitarist. This is how the EQ comes in handy. With an EQ, you can change the overtones coming through a particular channel to make it sound one way or another. You can cut lows and add highs to make a sound clearer and more bright, or you can cut the highs and add lows to make the bass more punchy and powerful. You can also add mids while removing the highs and lows to make someone sound like they’re coming from a radio or talking with their hands cupped over their mouth.

The knobs in the EQ section are typically in a 3-band Parametric setup, but sometimes it is extended to four bands. This changes how the frequency spectrum is divided among the various controls. As I said earlier, the average human ear can hear only 20Hz to 20KHz, so that is where the EQ spectrum normally sits. Here is a picture so you can see what I’m talking about with the normal frequency spectrum.

Frequency Chart

These frequencies would be divided into either High, Mid, and Low (thirds on the visual chart, not the mathematical thirds) or High, High-Mid, Low-Mid, and Low (fourths on the visual chart, not the mathematical fourths). In a simple setup, your High and Low will have one knob: Gain. The Mids are where the extra knobs come in. In the simple setup, your Mids will include the Gain knob(s) as well as a Frequency knob. This knob changes the center of the frequency range. To picture this, imagine three blocks of wood. By default, these would be side-by-side, not overlapping or leaving blank spaces between them. The frequency knob lets you slide the middle plank either direction, giving you the same spread of control but over a different area. You would, then, need to make sure you remember that both the Mid and High knob (or Mid and Low) would be controlling one section of the spectrum, so they can both cut or boost the same frequencies at the same time. In a more complex setup, you will also find a knob under the Mid section called Q. This knob changes the width of the frequencies affected by the gain knob. You can take the Q all the way up and narrow the field or bring the Q all the way down to maximize the area of control.

But changing the tone is not the only thing you can do with EQ, oh no.

Eliminating Feedback

I have written a short piece on eliminating feedback using EQ, so head on over to the Troubleshooting page to read up on that.

Carving Spots for Sources

This is a tricky subject that I will choose not to cover in this training because of its depth. If you would like to learn more on this, I would recommend going on forums and buying a few books on using Equalization in live situations. The basics of this principle, though, is that multiple sources overlap each other in the frequencies they utilize. By pulling certain frequencies out of different parts, you can clarify them so they can be heard separately and distinctly from one another. This fixes a mix where it just feels like a wall of sound with no way to tell one vocalist from another or the electric guitar from the bass. That is, of course, a worst-case scenario, but it does happen sometimes if the tech crew do not know their stuff.

The technique of EQing a spot for each part of the band should be used sparingly, however, despite the temptation to carve the heck out of the band. The more you manipulate a signal, the more artificial and stiff it sounds to the human ear. They notice a lot more than you give them credit for, though the brain often has no idea how to process and explain the info coming in.

The best use of this technique, in my opinion, is for clarifying vocals in the mix. Adding some high-mids (just a bit — don’t overdo it) to the vocals can help them come forward in the mix so they can be heard above the instruments. You want the vocals to be the clearest in a church setting because they are there primarily to help the congregation follow the music and the lyrics. If the congregation cannot hear the vocals clearly, they will feel uncomfortable and lost.

Using the stereo field is another option for clarifying one instrument from another, but that is difficult to do in live situations as you would have people on the right side hearing a completely different mix from those on the left. If your room is long and narrow, you have more freedom to use the stereo field for clarification, but this is not usually a readily available option in, well, any venue.

High Pass Filter Button

This button will look somewhat like an upside-down Nike sign, with a flat line and a curve or angle down on the left end. The purpose and application of this button is pretty simple. When depressed, it cuts the frequencies below a set frequency at a set ratio. In other words, the further below the set frequency (often 50Hz), the more it is cut, or trimmed. The general application of this button is any vocal microphone, especially if it is on a stand. The reason for this is that as the platform rumbles or someone touches a microphone or its stand, frequencies are put through the microphone and amplified. These frequencies are garbage, ultimately detracting from the power available to the subs and generally making bothersome noises. Since the human voice does not produce much in this frequency range, there is very little lost with this button. Applying it to certain other channels, however, could greatly reduce the punch your system has. Don’t put it on the bass, for example, if you are trying to pull all the power you can out of the subs.

That is the usual application of this particular button. I heard a story once that reinforced the importance of this button. I was told of a professional sound guy who was looking to hire someone to run the board while he directed the sound crew at large. If anyone came in and did not immediately punch in those buttons for the vocal microphones, the person was gone regardless of how good he or she was at the rest. I don’t know how accurate the story is, but it gets the point across. It is so basic of a principle that there’s not much excuse to miss applying it.

   — Fader Section

The Fader Section is almost the end of the line as far as the signal path in mixers. A typical Fader Section generally consists of a Pan knob, LED indicators (the Signal LED is sometimes replaced with a Decibel Meter in fancier boards), a Mute (a.k.a. “On”) button, Assign buttons, and the Fader itself. On Digital Mixers, the Fader Section will often include a “Select” (Sel) button as well and has the “On” button rather than “Mute” more often than analog boards do. On many Digital Mixers, the Mute/On button is backlit instead of having its own LED indicator. Let’s dig into these components, shall we?

Pan Knob

The main purpose of the pan knobs is to send a signal to either the Right (R) or Left (L) channel in the mains. This could be because you have a keyboardist playing in stereo and you have the luxury of utilizing the stereo field in your system, because you want to bring some representation of the band setup to your mix (guitarist is on the right, so you pan him right, etc.), or one of a number of other reasons. It sounds like a cool thing to do, but you might want to be careful in how much panning you actually do, unless you are using either the stereo instrument example I gave or you are using Sub Groups. More on that in the Assign Buttons section below.

Mute/On Button

The Mute button (or On button, depending on your mixer — they work in the opposite direction) is exactly what it sounds like — it mutes (or turns off) the signal to keep it from going out to the speakers. A red LED indicator usually shows when a channel is muted. Muting a signal can be handy when you are moving from worship into the sermon or another less-noisy part of the service and you have a noisy source. What does that mean? Well, many sources cause some sort of noise even when they are not “supposed to be” putting out sound. Probably the most common example of this is the guitar amp. If you listen closely as you mute and unmute the guitar amp channel, you will probably notice just a little bit of a hiss coming through the system when the amp is live. This is a natural bi-product of the amp’s design. This is easily fixed by, like I said, muting the channel. It can be unmuted after the sermon without changing any of the settings on the mixer, which is amazingly useful! Imagine having to turn the gain down every time you wanted to keep something from putting noise into the system. Each time you did it, you’d have to set the gain back where it is to get that same mix back that you had before. That’s way to annoying to be worth the trouble, so the Mute button make a clean, easy alternative.

A second use of the Mute button is to stop or prevent other noise issues like feedback. If you have a sudden screeching, you don’t want to let it come out until you find the source in the middle of a service (that WILL happen to you at some point, I can almost 100% guarantee it!). No, you want to stop the feedback so it’s not hurting the congregation’s ears and yours, and you figure out your next step in silence. This gives you time to see whether any of the mics are in front of the mains, whether an acoustic guitar is too close to speakers (sometimes the speaker output will vibrate the wood at a particular frequency, go back into the system, and get re-amplified again and again just like a microphone), or some other problem like that.

Your first step when this sudden feedback occurs should be to quickly scan the mic inputs and look for one that has a strong signal as indicated by the Signal LED.If you find it, hit the Mute and see whether that stops it. If it does not, drop the Gain knob back some. The primary concern is to eliminate the disruption rather than maintain that sweet mix you did during practice. If you can’t find the source of the feedback by the indicator lights, look at the stage and think through potential problem sources. Are you using wedge monitors? Is the pastor/guest speaker on the stage or in front of it, possibly getting feedback through the mains? Are any of your mics especially hot, such as for those singers without the proper breath support to give them volume? Are any of your amps mic’d, and are they in close proximity to a wedge or are they hot? Is one of the singers or the pastor/guest speaker pointing a handheld mic down at one of the monitors? That one happens WAY more often than you would expect. I have found that there are more people who have little to no idea how they are supposed to handle a mic than there are those who do. A fallback would be to drop the Master Fader if need be. This could cut some of the level and give you a little more grace from the congregation (though it’s probably already long past their miniscule patience) and time to figure out the problem.

Note: The Mute button mutes only the mains, not the Auxiliaries. If you are running your monitors through the Auxiliaries and that is the source of the feedback, hitting the Mute button will likely do very little, if anything, to fix it. It will probably reduce the feedback because then the mains are not assisting the runaway frequency as well, but it will continue since it has the monitors as a pathway. The options then would be to lower the level on either the Gain or on one of the Auxiliaries, depending on which Auxiliary is the problem.

A third instance when the Mute button is useful is in preventing problems like feedback or other noises before they happen. If the bassist, for example, leaves his bass turned on and turned up, it is a hazard for extra noise. What happens if someone bumps the bass? It will thump through the system. What happens if the stage is hollow and it picks up some low frequencies? It might resonate and pass that on to the bass, thus getting a low frequency “Hum” into the system. These types of things are easily avoided by just muting all the channels that are not or will not be used for some time.

Solo Button

The Solo button is one of your best friends when setting an initial level and when fixing the tone or another quality of a channel during a performance. The Solo button teams up with the Light Meter in the Master Section and gives you indispensable feedback on your source’s level.

The first thing you want to do when setting up a channel with a new guitar, vocal mic, or any other source is to bring the fader down all the way. Yes, I said the bring the fader down all the way as your first step in setting a level. While you’re at it, bring the auxiliaries all the way down as well. This means that nothing will be coming out of the mains or the monitors. See, you can’t properly set a level for people to hear until you have ensured that the signal is matched to the mixer’s capabilities. ignoring this step is comparable to running a sports car on diesel fuel. The car would get screwed up, and the results will disappoint you.

Bring the fader down all the way, and hit the Solo button. Look to the Light Meter in the Master Section. It will probably be in the top-right corner of the Master Section. Have the person play or sing at the loudest he or she will do the entire performance. This may mean using the heaviest distortion on the electric guitar, it might mean wailing away on the snare, or it may mean singing one’s heart out. As the person performs, the Light Meter ought not hit the red.

Hitting the red once every few minutes would be fine, but it is best to set the gain so that the signal peaks below the red. Some engineers will tell you to set it so it hits the top yellow light, some will tell you to set it so it hits between -6 and -12 dB on the meter, and others will have their own input. If you set it so it peaks midway in the yellow at the person’s loudest, you should be fine. The reason you want to do this is so that you do not overload the circuitry in the channel preamp, thus avoiding distortion (not the same as guitar distortion — this is bad distortion).

During a live performance, you can also use the Solo button in conjunction with a pair of headphones (do us all a favor and don’t use ear buds — that has many problems for sound and for your hearing and is just plain sad) to hear just one source in the mix and make necessary adjustments. You can tweak the EQ of the lead vocalist even while everyone is playing, or you can just get a feel for how to listen for a certain person so you know where they sit in the mix. You may not believe how hard it is sometimes to pick a particular vocalist out of a mix even if his or her level is set properly. This brings us to the next part — the Assign buttons.

Assign Buttons

The Assign buttons are used to route a signal from its channel to another location within the sound board. The usual setup is to have a 1/2, a 3/4, and a Master assign button. This tells the board to send the signal to either one of the Subgroups, to the Master Fader, or nowhere at all. Normally you’ll see these all set to Master. In more professional, complicated setups, you will see many of these set to the subgroups rather than the Master output. Those channels routed into the subgroups are typically easily categorized (e.g. guitars and keyboards are all instruments). Inputs such as an iPod or computer audio in don’t fall into these categories as well and are more often routed straight into the mains rather than the subgroups. The advantage to having the channels routed to the Subgroups instead is that you can more easily mix, using categories (vocals, instruments, drums, etc.) rather than individual channels. But more on that will come in the Master Section below.

3) Master Section

You’re almost done! Keep reading!

The Master Section takes the basic pieces from all over the mixer and routes them from one place to another, whether that is internally (Sub Groups to Mains, mix to Control Room, etc.) or externally (effects to Aux Sends, Main Fader to Main L/R output, etc.). The Master Section also controls the levels of different outputs (such as the Main L/R, Control Room, Headphones, Aux Sends, etc.). Included with level control is the Meter, which tells you how hot a signal is — this is your reference when soloing one or more channels, and it generally shows the output of the Main L/R when no solo is engaged. Ready?

   — Auxiliaries

The controls in the Master Section for the Auxiliaries deal with the final destination of the signals from your channels. Recall that you route your signal from the channel to the auxiliary with the auxiliary knobs in each channel. If you look down the horizontal line of knobs in, say, Aux 1, you will see how much of each signal will be mixed together into a new single source. The Send and Return controls decide what happens with this new mixed signal.

Auxiliary Sends

 If you think of the channels and the auxiliaries as functioning as perpendicular channels to each other (think a crossroads), it may help you sort this out a little better. As you feed the different channels into the auxiliary, they go to a sort of different channel strip. This channel’s fader is not a fader but a knob, usually labeled Auxiliary Send. If it is not labeled that, it should be similar. The Auxiliary Send knob will control the overall level going to the external destination, whether it be a stage monitor of some sort or an effects unit. Remember the important difference between parallel and serial effects when choosing whether to send it from the pre-fader auxiliary or to insert it directly into a channel strip. In the case of a serial effect you desire to put into the entire auxiliary, such as a graphic EQ, you will run the Auxiliary Send into the unit and return it via the Auxiliary Return. Simple enough, right? Well, here’s where it can get tricky depending on which mixer you have.

It may or may not help to think of the aux as a horizontal channel as I have set it up. If you find that it is not helpful at all, just strip out my comparison and read this section for the pure application of send/return functionality. Because this may be confusing to some, I am including a link to contact me. Feel free to send me an email asking me to clarify for you. That is what this whole project is for, after all — to help people understand sound systems and their operation.

Monitor Sends and Effects Send and Returns

Midas Venice 320On some boards, the auxiliaries will actually be set up especially to run monitors and effects, as listed separately from the auxiliaries. The Midas Venice 320 is one of these boards. The price for these boards is still (in 2013-2014) around $5,000, and Midas has moved on to other series already. This board has two Effect, two Monitor, and two Auxiliary Sends, with each one optimized for its particular use rather than having six generic auxiliaries like most boards would have.

One of the changes is that the Effects and Monitor channels have faders to control the signal level, though the effects fader controls the return so that you can mix in the effects signal rather than messing up the wet-dry mix. The Auxiliary channel is set up like the normal aux send/return on other boards so that you can use them as plain sends or feed effects into the monitors (with the effects going through the aux, however, NOT the Effects return).

Auxiliary Returns

In generic setups, the Aux Return knob takes everything that has been done in the aux (channel) and then feeds it into a sub group or the main out as you please, assuming it has assign buttons. There may also be solo buttons for pre-fader auxes, which is useful for the same reasons the solo buttons are in channel strips.

The Mackie 1604 VLZ Pro has another take on this generic setup. If you notice in the image (click on the link to the left) auxiliaries 3/5 and 4/6 share knobs, so though there are 6 aux outs, there are only four knobs. They set up this board so that you can duplicate the auxiliary effects to the mains and the monitors, if you so choose. You can also assign Aux Returns 3 and 4 to sub groups or to the mains, so you could even put the effects on a sub group if you wanted. It is a bit confusing to understand their particular setup, but if you spend time reading their manual descriptions and comparing the Stereo P.A. diagram, you will notice there are six aux out jacks and four stereo pair aux returns. Two aux sends for monitors, four aux send/stereo returns to feed where you want. Like I said, it’s difficult and will take time to understand their take on the system, but it works well.

If you find that your particular situation does not match this section and you would like help, I am open to helping you out. Just know that there is no guarantee that I will be able to respond right away. If time is not an issue and you would like to try me out, just contact me with your sound board model, monitor setup, and effects units, and I will try to look up your equipment and help you out.

   — Secondary I/O

Here you may see “Tape In,” “Control Mix,” and the headphones controls. If there is a set of RCA inputs either on the face of the mixer or in the back by the main outs, this is likely a tape in or similar secondary input. There will be a knob In the Master Section, and it will often be accompanied by some assortment of solo/assign/etc. buttons. It may function as a mute/on button, so it can be a kill switch in case something starts playing when it’s not supposed to.

The headphones controls will typically include a volume knob and a pre/post-fader button. Sometimes there are a few extra controls — just consult your manual and your sales associate (Sweetwater’s sales team is great for this) if the particular control confuses you.

If there is a set of secondary outputs such as those for a “Control Room,” there will be some basic controls for said outputs, such as volume/level. There may also be a control for effects level in this secondary output. If you have extra features on your mixer, great! If not, then it’s one less thing for you to worry about.

   — Meters

The meters are a set of lights or some form of display on the mixer that indicate the incoming strength of a signal. They are aligned in a vertical fashion, consisting of a green section as the main portion, a short yellow portion just above that, and a very small red portion at the very top. Fitting with the general approach to colors and their meanings, green lights mean fine, yellow means to be careful, and red means there’s a problem. In the case of the meters, yellow means that the signal is about to “clip” (overload the circuitry) and red means that the signal has clipped. When this happens, the signal will distort, breaking up the signal with something one may call fuzz, crackle, or something similar indicating a degradation. On the flipside, green typically stands for good things. However, the color itself only indicates that the signal level is not too hot — not that the signal is “good.” Here’s where it gets interesting, and a little more hefty of a concept to manage a signal level.

We already know that a signal level left too high can damage the equipment and cause a degradation, resulting in an undesirable sound. Equipment can also be damaged by too low of a signal level, though it happens in a different way. If an incoming signal is too low, it has to be boosted somewhere else in order to keep up with other signals. In one example, the singer with a low level may have to compete with bass, drums, electric guitar, acoustic guitar, piano, and several other vocalists. That’s a lot of stuff to worry about matching, which is where the problem arises. In order to keep up, the signal has to be boosted somewhere. When the level is cut in one place, it has to be boosted in another. This leads to several “choke points,” as it were, one of which is kept small while the other is opened up to compensate.

Think of it this way: when moving a piano with two people, you want them to share a fairly similar load, right? Well, if one of them is using just his pinky finger to move the piano, he’s not really contributing a whole lot to the effort. In order to compensate, the other guy has to lean his entire body into it, perhaps spraining an ankle or pulling a hamstring in the process.

If the first guy had just helped to the extent he should have, the second guy wouldn’t have been injured. It’s the same with the gain structure in a sound system. If you cut back a signal in one place and boost it in another, you are essentially wearing out certain components by requiring them to expend extra energy where the other could just pitch in a bit and alleviate the pressure. Some of the usual choke points to watch are the Gain Knob, the Fader, the Sub Group faders, and the Main Fader.

Again, the green color only indicates that the signal is not clipping, not that the signal is at a good level. It may provide a decent range, making a quick look fairly effective in analyzing the needs of a signal adjustment, but the decibel markings are what provide the better feedback for how low is too low.

Troubleshooting Missing Signals

The meters are some of your biggest friends in building a mix. First off, they tell you if a signal is present. You will generally use this as one of the — if not the very — first tools in troubleshooting a missing signal. If, say, one singer is apparently using the microphone correctly but there is no signal from him or her coming out of the mains while other signals are getting through, you have to figure out where the signal is failing. To begin, engage the Solo button for the singer’s channel. Look to the meters to see whether the signal is coming into the channel. If the meter is bouncing as the singer provides a signal, that means there is no major physical problem between the singer’s lips and the input stage of the mixer. The gain knob is also turned up if the meter is bouncing.

Next, check the fader. This is one of the most likely places for the signal to be lost, since the faders are one of the most common places to change the signal level. Should the fader be set fairly close to the other faders, that is probably not the problem. If that is the case, check the assign buttons to check that the channel is either routed straight to the mains or to the mains via a sub group. If this looks to be good (check the sub group just like you checked the channel if that is the signal routing), then it is possible the channel is bad. Bad channels more often have no signal input rather than having a breakdown after the gain stage, but it is certainly possible. Try plugging into another channel and repeat the process.

If you read through that and were thinking, “But I never had a signal! The meters were never bouncing, or jiggling, or whatever you said!” then check the gain knob. If the gain knob is turned down, then there will be no signal to the channel, and the meters will remain silent as well. If the gain knob is turned up and the meters are still not bouncing, check the connections between the microphone and the mixer.

Here are the typical things to check: Does the microphone have an on/off switch? Is the cable plugged into the microphone all the way? Is the cable plugged into the snake/stage pocket all the way? Is the snake termination plugged all the way into the channel input on the mixer? If you have checked all of these and it is still not coming through, check them again. If it is still not coming through, try another channel. If it works, either you have a bad channel or you have a bad connection/cable somewhere along the line and need to spend some time chasing it down.

All of this started with the meter, as it should every time. That is the first and easiest place to check the signal, because it tells you which half of the system has a problem, whether it is before it comes into the board or whether it is a problem with the signal path. But enough with the input troubleshooting. Let’s move on to decibels! Sound exciting?


What is a decibel? I am not a physicist, nor am I a close follower of any field of study that would make me an authority on the subject. That being said, I can tell you that a decibel is merely a way to measure the signal level coming through a given system. It is used with different sorts of measurements depending on what sort of system you are using, but sound equipment will generally say merely dB or dBu, which measures decibels in voltage in the standard way that voltage is used in audio equipment (specifically, 0dB = 0.775v). If that statement went right over your head, that’s ok. It went over my head too. If you want to read more about decibels, you can check out the Wikipedia entry. Sweetwater also has a short article in their inSync publication that explains a bit further.

But you will often hear people talking about decibels in terms of how loud something is. In this case, you might see someone pull out a decibel meter, which is a clever little device that, you guessed it, measures the decibel level in its current position in a room. More appropriately, this is actually a discussion about Sound Pressure Levels (SPL). This deals with the pressure you may or may not feel in a room as sound waves bounce around inside it. This is important to a sound technician because high SPL will damage people’s ears over time, even if it is not painful in the moment.

As this site says, certain sounds tend to hit a certain decibel level:

Near total silence – 0 dB

A whisper – 15 dB

Normal conversation – 60 dB

A lawnmower – 90 dB

A car horn – 110 dB

A rock concert or a jet engine – 120 dB

A gunshot or firecracker – 140 dB

Yes, a rock concert is approximate to a jet engine. How would you like standing next to a jet engine for three hours straight? Now you can understand why some people prefer to bring earplugs to concerts. In fact, continued exposure to sound at or above 85 dB can cause hearing damage! It doesn’t happen all at once, but it can happen with regular, prolonged exposure. This is why some pastors are anal about the worship being under a certain level. No one wants to be known as the church that caused their congregation to go deaf over the course of a year!

So the two main things that will concern you with decibels are the gain/attenuation on your mixer and the SPL in your worship environment. They are a little bit different, but after working with them for a while, you will be able to intuit the relationship between the two.

Now that we know what a decibel is, let’s think back to the very beginning of this extremely long article. Do you remember when I mentioned earlier that any point at which gain/trim happens, there should be a balance so one is not lifting the proverbial piano while another stands by? Voila! Decibels are a measurement of how hard each bit of your mixer is working! If the gain knob says that it is cutting back to -40 dB while your fader is pushing 40 dB, there’s a problem! It’s not supposed to be an exact balance, as I noted in the gain section that the gain knob’s primary job is to match the incoming signal with a voltage level that the board can handle. This balancing act is more of a general mental guideline that will help you keep from killing your equipment. It will quickly become a background issue that you will think of in different terms as you learn your way around the sound board. You may never think of the balancing act concept again, but it will get you going in the right direction.


So what’s Unity? You may or may not have heard someone mention the term before. Regardless, it is an important concept to understand when discussing gain level and decibels.

So we’ve established (and perhaps read in those other articles? Hint hint…) that decibels in a mixer are a measurement of the boost and cut of a signal’s voltage. Unity is a very special place in the hearts of all mixers, and that is when the mixer is doing virtually no work at all. You see the 0 dB marking by all the gain knobs, EQ knobs, and faders? That little 0 marks the point at which the signal is being neither cut nor boosted. The signal passing through is at its purest level as it moves through the circuitry of your mixer. This means that you have the highest signal-to-noise ratio (aka “S/N”). This principle says that one should have the hottest signal possible without breakup (distortion), because that allows for a hotter signal from the source that can better drown out the little bits of noise that come in from everything else in the sound system (cables, ground issues, etc.).

While I am on the subject of unity levels, I need to address a somewhat common myth in mixing. Have you ever heard of unity mixing? The idea is that you set the faders to unity and actually create your mix with the gain knobs, therefore not coloring the mix with all the adjustments that come in the individual channels. There are some other reasons driving the desire to do this, but it has a couple major issues.

The first issue is that if you are mixing with the gain knobs, they are no longer bringing the signals into a usable level for the mixer. If the signal is extremely hot, it just is what it is. If the signal is extremely quiet, it is what it is. While a hot guitar may quickly be pulled back by the process of unity mixing, it is still subject to audible judgments rather than an objective physical measurement. That leaves some obvious problems, for those of you who read this article from the top to the bottom. If the engineer likes a hot guitar, he could burn out the channel circuitry by leaving it up to be as loud as he wants it. If a source is quiet and he wants it to stay quiet, he doesn’t have a lot of room for equalizing properly because he will be trying to boost the voltage of particular frequencies that are already too quiet, thus introducing noise into the mix. It may be imperceptible to the average listener as noise, but an entire mix like this has a lot of potential to sound terrible! All of this is also related to how well the sound system was set up. If you are unsure of how that is done, see this page for more details.

The second issue is that you are leaving alone a component that the manufacturers have intentionally put into their equipment. If you leave the faders at unity as a rule and do not touch them, you’re basically telling the companies that you know more than them about how the equipment should be run. Obviously the billions of dollars that have been poured into research on how to best build a mixer have been wasted, because they should have just waited and consulted you on the intricacies of proper voltage usage. Just that little bit of simple logic is enough to puzzle me for life as to how someone can justify unity mixing. DON’T DO IT!

In all of this discussion, however, I do feel it necessary to soften my tone concerning the use of this particular method. One of the defenses of this method is that it “sounds better” than using a lot of alteration of the signal via the fader. The idea that I have seen behind this is that the less alteration of the original signal, the better. I can vouch for that concept, as every new item you intriduce into the signal path opens up more possibilities for problems: more cables that may short/cause a buzz, more chance of a ground loop, more places to mess up the gain structure (N/S), etc. But the proper pullaway from this idea of “unity mixing” involves a somewhat new sort of sub-concept within the S/N concept.

We already know that too hot of a signal can overload circuitry and cause distortion in an analog mixer (digital distortion is very different and will be covered in the page on digital mixers). Too low of a signal can introduce noise into the sound by allowing foreign elements like buzzes and such to come alongside the original signal. What if the gain is set at the “proper” level and it is too loud in the mix? Well, you could pull it back at the fader or at the gain pot — but which is better? Well, you would, technically speaking, be introducing the possibility of noise at either place. The question, then, is not which one is “better” but which one would introduce more noise. Generally speaking, downstream reduction in signal will be safer because it allows less room for boosting later on in the chain, therefore introducing more noise. If you have something in the signal path between the gain knob and the fader, then the fader would be the better option. If there is nothing there, the gain would likely have about the same amount of noise introduction as would the fader. *Remember: Every item introduces SOME noise, including everything from a cable to an onboard EQ to an outboard compressor* In the end, however, if you can drop the gain a bit below the ideal level because its source is far too prominent in the mix, then as long as the signal is not being re-boosted downstream, then it is fine. The noise THAT COMES INTO THE MIXER ALONGSIDE THE SOURCE will be reduced just as much as will the source itself. That may spin your head a bit, that I am saying you can reduce the gain a bit, but it’s true. If it is a matter of missing your mix and reducing the gain slightly below ideal, it will not kill your sound. Just follow the general principle first, and follow this new principle second. The gain can be used as a reducer (trim concept, remember?) in the way that unity mixing advocates. That being said, a properly set-up sound system shold not need much reduction below the ideal input level because a good live mix does not generally require many items barely perceptible, and the different sources can mix themselves to some extent, with proper gain structure.

One more thing before I move on. A simple way to think of gain structure primary use is that it is to “gain” at the beginning of the chain and “trim” at the end: it should be basically downstream and, ideally, should never have a boost after a cut anywhere. Now, this applies specifically to the sound engineer. Audio subjects such as electric guitar often break the rules because of the tone they are reaching for. The sound engineer’s job is to accurately and cleanly represent the sound they are receiving from the stage, not tweaking the sound to make it “better.” I hope this section was clear. If not, please contact me with any questions or need for clarification.

   — Sub Groups

As I’ve mentioned previously, the sub groups are intended for grouping channels together. One of the uses is for putting the vocals together, the drums together, and the other instruments together. The advantage to doing this is that you can mix the groups down (e.g. the leader’s vocal channel fader is higher than the background vocals) and then manage the vocals as a whole with just one fader. If you do the same with the drums, you have a master fader for the drum set, so you can set them to relative levels to suit your tastes and then leave them alone. Without the Sub Groups, you would be taking a lot more time adjusting things when you want to drop or raise the overall volume of one group, such as the drums. Where before you would be octopusing the drum channel faders to bring them up the same amount, you just grab a single fader!

The way you set channels to go into sub groups is to depress the appropriate Assign button by the channel fader and rotate the pan knob to throw the channel into the correct side (e.g. Assign 1/2=L/R Pan). If your mixer is big enough to have sub groups, then you may also need to assign them to the main mix. You might have 1/2 buttons and pan knobs, you might have Left buttons and Right buttons, or there may be some other configuration. It all depends on…yup, the type of mixer you have. Just remember that if you are running channels into your sub groups and nothing is coming out of the mains, you may not have the sub groups assigned to the main outs.

   — Master Fader

This fader may be labeled Master Fader, Stereo Fader, or one of several other names. You should be able to identify it as the only fader of a particular color, and it will be set apart from the rest physically to stand out. It may even be a pair of faders, one left and one right, instead of a single stereo fader. This is less common, but it does pop up every now and then.

This is basically the final say in your level before it exits the board. I say basically because you may or may not have a switch or a dial by the main outs that can further cut/boost your signal, but this fader is effectively the last say because you will set any switch/dial you may have and forget about it after that, whereas the Master Fader may change a bit from one week to the next.

Treat this fader much like the other faders on the mixer, because it acts the same way. You need to find your sweet spot for your power amp/powered speaker that works for your room size and equipment so that your Master Fader does not change much from week to week. It could be tweaked, especially if there is a guest band, but it should be fairly stable once you arrive at a norm.

4) Master I/O

The Master I/O section on the rear of your mixer (or top, for some mixers or if you have an adapter) contains all the jacks that go out of the mixer to different sources as well as the special returns and secondary inputs. This could include the Master L/R, Auxiliary Send/Returns, Tape In, Ctl Room Out, Master Inserts, and much more.

   — Inserts

These inserts will change depending on which mixer you have on hand. They will operate in basically the same fashion as the channel inserts, but they are intended for a larger portion of the overall sound. So a sub group insert will apply to every channel going into that insert. Just keep in mind that you will not have control over individual elements at this level, so a sub group insert is probably not the best place to put a reverb, unless you want a simple room correction like a slapback delay (think outdoor concerts). To hear a slapback delay, listen to the vocals in Ricky Nelson’s “Hello Mary Lou” or this random video of a guy playing a rockabilly lick in C Blues. If you wanted all your vocals to have the same delay, such as in this case, this would be a fine way to do it.

Also in the master section inserts will be inserts for the main out. Here you might see a graphic EQ for fixing the room’s unique frequency issues. Basically, these inserts will be just as useful as the channel inserts but in different ways. Keep in mind the limitations of inserts versus aux outs for effects units.

   — Main Outputs

The output of one’s mixer will differ, again, based on the individual mixer. The general output Is XLR, but bigger boards often include 1/4″ outputs. If you are using the 1/4″ outputs, you need to make sure that you are using speaker cables, not instrument cables, on a powered mixer. On an unpowered mixer, you want to make sure that the cables running from the output of your mixer to the power amps have proper shielding. Without proper shielding, you run the risk of picking up a lot of interference in your signal just between the board and the power amp. What’s the difference between instrument and speaker cables, you ask? Well, I’m glad you brought it up.


The basic difference between speaker and instrument cables is whether the cable wants to have more power or more shielding.

Instrument cables have a lower threshold for electrical power they can carry, but they have more shielding to resist interference from fluorescent lighting, power supplies, etc. Since an instrument does not put out the power needed to drive, well, a driver (speakers), they do not need cables that can handle that much power. So good instrument cables have a lot of resistance (impedance), low power transfer, and a lot of shielding.

Speaker cables, on the other hand, need to be able to handle more power. As such, they are thicker, thus providing more space for the power to go and giving the cable lower resistance (impedance). At the same time, that space used for power transfer has ousted the cable shielding, so there is more susceptibility to interference. In a cable carrying a big load of power, though, interference does not make a huge difference because it just becomes a pebble in a pond.

Fender has a decent short article on the exact differences between instrument and speaker cables.

If you have an unpowered mixer and have been hearing a radio station in your sound system, try swapping out the cables — you may just have an unshielded cable in there somewhere. It’s a simple enough fix, but people will treat you with a lot of respect for a few weeks — until the first sign of trouble, of course, because that means you screwed something up again, in their minds.

Powered or Unpowered?

Now for a perhaps more obvious question: How do you know if you have a powered mixer or an unpowered mixer?

A simple answer is, how big is your mixer? If it seems like a small mixer but it has a big ol’ backside, then there’s a good chance it’s powered. That and it probably says so by the name printed on the mixer. See, a powered mixer contains the power amp inside it necessary to power a speaker or set of speakers. Once the mix is made, it has to be pushed by a heavy electrical current in order for the speakers to work. If you don’t have a powered mixer, you might have a power amp or two or three or four. If your mixer isn’t powered and you have no power amps, you probably have powered speakers. You can tell if you have those because you have to plug them into the wall and flip the power switch. Clever method of telling the difference between powered and unpowered speakers, huh? But more on those in the Speakers page.


You made it through the article! You ought to go out and get yourself an ice cream cone for finishing the biggest section in this whole darned set of sound materials! If you’re ready, go ahead and move on to the other articles. That is, unless you’re tired of me by now. I would be.