Audio Studio Engineering
~ Basics and Techniques
Anthony D.
Beardslee
without written permission from the copyright holder.
TABLE OF
CONTENTS
CHAPTER ONE
- THE MIXING CONSOLE
1.1 Introduction
1.2 Terms and Parts
1.3 The Input Module
1.4 Main Output Bus
1.5 Group Output Busses
1.6 Auxiliary Busses
1.7 Master Module
1.8 Monitor Module
CHAPTER TWO
- USING THE CONSOLE
2.1 Signal Types
2.2 Gain Structure
2.3 Levels
2.4 Solo Functions
2.5 Connections
CHAPTER
THREE - STUDIO SETUP AND SAFETY
3.1 Microphone Setup
3.2 In the Control Room
CHAPTER ONE – THE MIXING CONSOLE
If you asked any experienced engineer what the most important part of operating a console is, you may be surprised by their answer. It is not being competent on the most state of the art console. It is understanding signal flow; understanding how a console - any console - operates.
Consider this analogy: whether you drive a compact hatchback or a full-size luxury car, they both have one primary purpose – to provide transportation. The difference (other than the price) is the features available by each vehicle. But they both have the same basic elements – a steering wheel, a gas pedal, a brake, a switch to turn on the lights, a switch for the wipers, and the list goes on. When you step into a new car, you may not know exactly where the light switch is, but you know that somewhere there is a light switch. It’s just a matter of finding it. Mixing consoles are very much the same. When you sit in front of a new console, you know, for example, that there should be some way to provide phantom power to a condenser microphone.
Every engineer should be equipped with an understanding of the fundamental elements of ALL mixing consoles. Suppose someone taught you how to operate a mixing console. They showed you where to plug in the mic, what buttons to push, and viola – the microphone works. But they neglected to tell you what those buttons mean. Why is that a problem? You only know how to operate THAT console. What happens when you start working on a different one? The phantom power, and many other features, are not always in the same location for all consoles.
You should not just learn what buttons to push, you need to know WHY you are pushing them. This is the difference between a professional engineer – and a trained chimp.* This chapter will provide you with a fundamental base of knowledge that can be applied to any console – compact or luxury.
*Our good friend Bobo the chimp is featured in many examples in this book.
Terms introduced in this
section:
Signal: Acoustic energy that has been converted into
electricity.
Bus: An electronic device that combines any number
of separate signals into one signal.
The purpose of a mixing console is very much the same as the above definition of a bus – to combine signals together. If you want to record eight microphones onto a CD, something has to come in-between the mics and the CD recorder. After all, the recorder only has two inputs. The mixing console allows you to combine the signals created by those microphones into one signal (or two for stereo) that can be plugged into the CD recorder.
Many people misunderstand the purpose of the mixing console. A novice will look at all those buttons and lights and assume that it is the console that makes the “magic”. Not true. The console is the main interface, the hub between the source signals and the destination medium. The “magic” comes from you – the engineer, and how you use your tools.
A side note about “magic.” Timothy Miller has long used this analogy to illustrate the importance of the proper use of your tools: If you were to give a master woodworker a block of pine, a screwdriver and a hammer with the instructions to carve a statuette, he would most likely produce a decent product. On the other hand, if you handed Bobo the chimp a block of cedar and a set of the finest carving tools available, what would you get? Most likely, a bloody Bobo. The point is - a quality product is not just produced by the tools, but the proper use of the tools. It is important to understand that an engineer’s best tool is the one located between your ears – your brain.
MODULAR DESIGN
The part that makes mixing consoles intimidating to the beginner is the sheer size of them. You look at an SSL console in the magazines and you wonder, “how could anyone possibly know what all those buttons do?” But there is no reason to be intimidated. The console is made up several modules, most of which are exactly the same. So if you know the functions of the modules, you know what the whole console does…whether it is an eight-channel mixer or a forty-eight-channel mixer.
Here are the main parts you will find in most consoles. In the following sections, we will discuss each of these parts in greater detail.
Input modules
Where a sound source (such as a microphone) comes into the mixer
Main output bus
The primary output bus that is normally fed to the end product.
Group output busses
These busses have two main purposes. They can be used as secondary output busses or for mix grouping.
Auxiliary busses
These busses can also be used as secondary outputs, but can offer more flexibility than group outputs.
Master module
This is the module that usually contains the main bus, but also handles features that affect the rest of the console.
Monitor module
Usually found on larger scale consoles, the monitor module controls what you hear in your control room monitors or headphones. Sometimes the monitor module features are integrated into the master module, depending on the size and capability of the console.
Each of these modules and parts could, in theory,
exist on
their own. This is why the parts are
called “modular.” If you were to remove
one module from a console and properly connect the power supply and
audio
outputs, it could exist all on its own as a single channel. That’s not a common practice, since the back
of the module is completely exposed and the lack of wiring to the power
supply
could be a potential electrical hazard.
But single modules do, in fact, exist in the professional
market, such as the Massenburg
GML2020 Integrated Input Channel.
Knowing this, it is easier to understand a mixing console if you do not think of it as one big device. Instead, think of it as several individual devices all connected together for a common purpose. The buttons you press will change how these devices interact with each other.
SECTION
1.3 – THE INPUT MODULE
There are numerous ways to describe a mixing console. But the main method is in the number of input modules, or channels available on the console. The input module is the device in a console that receives a sound source, whether a microphone or a direct line from a keyboard. The terms, sound source and input module, are sometimes used interchangeably. An engineer will oftentimes ask, “how many inputs are there in this recording session?” What the engineer wants to know is how many individual sound sources there are, thus telling him how many input modules are necessary to handle the recording session. Some sound sources, like a guitar, may only need one microphone, therefore only needing one input module. Other sound sources, like a drum set, may require numerous microphones – all of which will need their own input modules.
Remember that a single input module is a device all by itself. We will first examine this module as an independent device. Some input modules are very basic, while others are loaded with features and extras. But most input modules will have six basic stages, some of them fixed, some of them adjustable. They are input, gain, equalization, auxiliary, assignment and fader.
1.3.1 - Input
This is where the signal from a sound source enters the input module. This usually in the form of a connector, usually a female XLR, or a female ¼” phone jack. The XLR connector is normally for inputting microphones while the phone jack is typically for line level signals. The difference between mic and line level signals is the amount of voltage being produced by the source. Microphones produce a voltage in the millivolt range, while line level signal can be as high as 7 volts.
1.3.2 - Gain
The gain stage, or pre-amplifier, is used to control how much of the input signal is being allowed into the module. The gain stage is sometimes referred to as trim. The purpose of the pre-amp is to amplify, or boost, the input signal to a useable level for the rest of the console. Depending on the level being produced by the input source, you may not have to turn it up at all. Or in the case of some microphones, you may have to turn it up a lot.
In either event, care must be taken to watch how far the signal gets turned up. Allowing too much signal to enter the module can result in distortion. If distortion occurs at the gain stage, the signal will sound that way for the rest of the console. How do you avoid this from happening? Most consoles are equipped with visual indicators to tell you when distortion occurs. If these light up, something is wrong with your gain structure. Gain structure will be discussed in detail later in this book.
1.3.3 -
Equalization (EQ)
Not all consoles have EQ’s included in their input modules. Some are very basic, others elaborate. An equalizer modifies the electric signal to achieve an acoustic result. The treble and bass controls on your stereo are examples of equalizers. Adjusting these will give you an acoustic result, such as more bass. The EQ modifies the signal electrically, the result of which you will hear coming through the speakers.
1.3.4 -
Auxiliary (Aux)
The aux section has many variations of its name. It can be known as send, effects (EFX), monitor send, foldback, or others. Whichever name is used, they all serve the same purpose. The aux section is used to send the input module’s signal to some alternative, or auxiliary, location. Examples of these auxiliary locations may include reverb or effects devices, or a monitor speaker. The use of this send has no effect on the input module itself.
1.3.5 - Assignment
This is the part of the input module that is the most overlooked, which is strange since it is arguably the most important part. You may recall reading about the busses that are used to combine the individual signals together. The assignment stage directs the input module signal to the busses.
The assignment stage is usually a row of buttons on the input module with the labels MIX or L-R, 1-2, 3-4, etc. The busses on a console are usually grouped in pairs, since most work in audio is produced in stereo. When you press the MIX button on the input module, you are directing the signal to the main bus, with an equal amount of that signal going to the Left and Right busses. Since the output of the Left bus usually ends up in the left speaker, and the output of the Right bus ends up in the right speaker, assigning a signal to both busses will result in that signal being heard in the center; between the two speakers.
Here is where the pan control comes in. The pan control is part of the assignment stage. When you assign a signal to the MIX bus, and the pan control is set at center, (pointing straight forward) the signal is being distributed equally to the Left and Right busses, as described above. When the pan control is turned all the way to the left, the signal is only assigned to the Left bus. When the pan control is turned all the way to the right, the signal is only assigned to the Right bus. The use of the pan control allows us to create stereo signals from the console.
The pan control is used to separate a signal from any adjacent bus, so that the busses can be used individually. If you assign a signal to the 1-2 group bus and pan left, the signal is only going to bus 1. Pan to the right side and the signal is only going to bus 2. Leave the pan control at center and the signal will be going to both 1 and 2.
A side note about stereo. There is much confusion out there about the difference between mono and stereo. Many amateurs think that the presence of two speakers automatically means stereo, and that two microphones mean stereo as well. Well, here’s the real scoop: Two microphones summed into one destination are mono. Likewise, one microphone assigned equally to two speakers is also mono. The meaning of mono is often interpreted as “one speaker.” That is true, but mono really means “one signal.” In order for something to be stereo, there must be at least two discrete, unconnected signals.
1.3.6 - Fader
If you can think of the gain stage as the input control to the module, the fader is the output control. After you have assigned the signal to a bus, the fader controls how much of that signal is actually getting there.
When you assign an input module signal to MIX or L-R, it is going to the main output bus, (also known as mix, master bus or program bus). As the input module is an input device, the mix bus is an output device. All signals that are assigned to the mix bus are combined there and then sent out of the console.
So where does it go? The main output bus is normally fed to the end product, whatever that happens to be. In the studio, the end product is the master stereo tape. In this case the mix bus output is connected to the input of the stereo tape machine or CD burner. In live sound, the product is what the audience hears, so the mix bus if fed to the main PA speakers. In a broadcast situation, the product is what hits the airwaves, so the mix bus feeds the FM transmitter.
There are usually two faders associated with the master bus. These control how much of the mixed signal (or bussed signal), leaves the console. Some consoles have only one master fader, but this does not mean that it is a mono bus. Electrically, they are two separate faders, but they are mechanically connected so that they move in perfect sync with each other.
In most cases, the mix bus consists of two discrete busses, which are intended to be used as a stereo pair. But they can be used for completely separate things. For example, the left bus could be used to feed a set of PA speakers, while the right bus feeds a tape machine. By keeping everything in the input modules panned center, the speakers and the tape machine will be getting the same mono signal.
And before you say, “eewww, icky…mono sucks,” let me just say that there is more happening in mono than you may realize. Mono doesn’t suck. There are many times when going mono is more appropriate than stereo. Still not convinced? If you were recording a voice-over track using only one microphone into a computer based recording system, recording in stereo would be using twice the hard drive space for no good reason.
SECTION
1.5 – GROUP OUTPUT BUSSES
You may have heard people talk about 8-bus consoles. When they say, “8-bus” they are talking about these – the group or output busses. This can get confusing since you know that the master and auxiliaries are busses as well. A console with eight group busses, a master bus and ten auxiliaries is, in reality, a twenty-bus console – but no one refers to it that way. What makes the group busses so special is that they can be used for many purposes beyond that of the main bus.
In terms of function, the group busses are essentially the same as the main bus. They combine signals together, and the group fader controls how much of that bussed signal leaves the group module. There are many possible uses for the group busses. The following are a few examples.
1.5.1 - Multi-track
Recording
In many cases, the outputs of the group modules are connected to the inputs of the multi-track tape machine. For example, group output 1 would be connected to multi-track input 1, group output 2 would be connected to multi-track input 2, and so on. So if you had a microphone coming into an input module, and you wanted to record it onto track 2 of the multi-track tape, you would assign that input module to the 1-2 bus and pan right.
Some recording consoles will be described as 8 or 12-bus consoles, but don’t have any group faders. This is because the busses are intended for recording alone, so the bussing occurs internally, and the output shows up at a patch point on the console.
1.5.2 - Multiple
Output Sources
The group busses can also be used to feed other 2-track sources. Suppose you were running sound for a live show with an 8-bus console, but you also needed to record the show for reference, and the local TV station wants a stereo feed from you so the show can be broadcast. “Oh, and by the way, our camera operators need to hear a feed in their intercom headsets.”
No problem. The mix bus feeds the main PA speakers. Groups 1 and 2 will feed the tape machine for recording, and groups 3 and 4 will go to the TV station’s remote truck. Then you could connect group 5 or 6 to the main intercom box. (It’s most likely a mono input, so you need only connect one of the outputs). As far as assignment, you would assign ALL input modules to MIX (or L-R), 1-2, 3-4, and 5-6. Every destination gets the same mix as the audience.
But wait…not everyone is happy! The camera operators don’t want to hear the vocals in their headsets because it interferes with the director’s instructions. Simple fix – just unassign the input modules that carry a vocalist’s microphone from group 5-6. That will have no effect on the speakers, TV broadcast or recording because those vocal mics are still going to MIX, 1-2 and 3-4.
One point of caution…when you are using a group bus as an alternative output, make sure that bus is NOT assigned to the mix bus.
1.5.3 - Mix Grouping
Group busses can make mixing more manageable. If you have eight channels of drum mics, for example, you can assign each of those modules to a group bus. This will allow you to control the overall level of the drum mix with one fader. To make this work properly, you must unassign the drum mic input modules from the mix bus, then assign the group bus to the mix bus. Also, if you are grouping a stereo mix, you need to assign the modules to two group busses and pan the bus modules left and right to maintain the stereo image.
Most mixing consoles have a number of auxiliary busses, which can be used in many of the examples described in the group module section. But the aux busses, in most cases provide a little more flexibility and control than group busses.
Aux busses have several different names – effects sends, echo, foldback, etc. In any case, they accomplish the same purpose – to allow the user to send the signal from one or several input modules to some auxiliary location. And this can be done without affecting any other part of the console.
Each input module will have a set of aux sends, and somewhere on the console, you will find the auxiliary masters. The masters are the busses – where the signals are combined. If you want to send a signal from an input module to the aux bus, you simply turn up the aux send on that input module. The aux send works like a water valve, with the input module as the source. The signal “flows” to the aux master, which can then be sent to the final destination.
There are two types of auxiliary sends – pre-fade and post-fade. The short explanation is that the source input module fader will not affect a pre-fade aux send, the post-fade send will be affected by the fader.
If that didn’t make any sense at all, consider this plumbing analogy: imagine that water pipe A is our source input module. Valve A is the aux send, and valve B is the input module fader. In figure 1, the aux valve is tapped before the input module fader. If valve B were closed, water would still flow to the aux valve. This is a description of a pre-fade send. Whether valve B is open or closed, the water still flows to the aux valve.
In figure 2, the aux valve is tapped after
the fader
valve. So if the fader valve is closed,
no water can flow to the aux valve. This
is a description of post-fade. The aux
send level is dependent upon the level of the source input module fader. In contrast, a pre-fade aux send is independent
of the input module fader.
The master module is going to vary from console to console. But the main thing you will find is the master, or mix, bus. You may also find the auxiliary masters as well. Also found here may be the talkback section, test oscillator, solo functions, or any other feature that affects the console as a whole. The variations are just too numerous to list.
This module, like the master, will vary depending on your console’s capability. This module will control anything that has to do with your monitoring – that is, what you actually hear sitting at your console.
*It is important to note that, in this case, monitoring refers to the listening environment in a control room and studio environment, NOT the wedge-shaped speakers on a stage that the performers are always screaming about.
1.8.1 - Control Room
Level
The control room level is where you adjust the volume of your control room speakers. It is important to know that adjusting this control will not have any effect on the output busses. In other words, you can turn your control room level all the way down, and the output busses will still be driving signal into their destinations without interruption.
It’s also important to remember that you should not adjust the mix bus when it’s “too loud” in the control room. Get in the habit of changing the control room level – never the mix bus. Only adjust the mix bus when you intend to make a change to the output, (i.e. fading out the song).
1.8.2 - Studio Level
The studio level should not be confused with the control room level. Control rooms are often mistakenly called studios as a generic term. The studio level controls the volume of speakers that you might have in the studio. That is, the room where the talent performs when you are recording them. It is a good idea to have a set of studio speakers, especially if you are recording large groups. This way you can play a recording back to them without having to jam twenty people into your control room.
1.8.3 - Monitor
Source Selection
This part of the console allows you to (as the name suggests) select which 2-track source you are hearing in your control room speakers. Common 2-track sources are CD players, DAT machines, or the mix bus itself. When you’re mixing, you are usually monitoring the mix bus. After you record the mix to a 2-track recorder (like DAT) you would change your monitor source selection to monitor the DAT machine outputs.
The number of 2-track sources will vary from place to place, and is really a matter of personal needs. You may only need to monitor one 2-track recorder. Some control rooms will have several 2-track sources. Some broadcasting studios will even use a commercial receiver as a 2-track source. This allows them to listen to how their mix is actually sounding as it hits the airwaves.
Like the control room level, changing the monitor source selection will have no effect on the output busses. It simply changes what you are hearing in the control room. The studio level will often have its own monitor source selection, so you can be listening to the mix while the studio is listening to something else. But not all consoles have this feature. On some consoles, the studio level will share the monitor source selection of the control room level. Some consoles don’t have a studio level at all.
1.8.4 - Solo Level
In addition to the control room level, some consoles will have an independent solo level. Some engineers like to have the level of a track they are “solo-ing” to be higher or lower than the average mix level. The solo level allows you to do this. A complete description of the solo function will be discussed later in this book.
CHAPTER TWO –
USING THE CONSOLE
SECTION 2.1 – SIGNAL TYPES
There are basically two types of signal: line signal and mic signal. The difference between these types is the amount of voltage present. A line signal can be anywhere from 0.1 to 7 volts, while mic signal lies in the millivolt range. Why is this important? Because input modules (and other input devices) operate at either mic level or line level. If you try to drive a line level (higher voltage) signal into an input that is expecting a mic level signal, you will cause distortion. Conversely, if you try to drive a mic level signal into a line input, the module will barely recognize the signal.
For example, if you plugged your guitar into a mic level input, you would cause a great deal of distortion in the module – no matter how much you turned down the trim. And if you tried to plug a mic into a line mixer, like the Behringer RX1602, you could turn the level up all the way and still have a small signal. This mixer was designed for only line level signals such as keyboards, guitars, and sub-mixers. In order to support microphones, the mixer must be equipped with microphone preamplifiers. Most mixing consoles have a certain number of “mic pres” built into the input modules, but you can also find devices that are stand-alone microphone preamps. Many professionals use these stand-alone pres for their superior quality, and then send their line level outputs directly into the multi-track recorder, bypassing any circuitry in-between.
2.1.1 - Level Pads
“So, what if the level is too hot to begin with?” Suppose you bring a microphone into an input module and, even with the input gain turned all the way down, you are still distorting the input. This is where a pad can come in handy. A pad will attenuate, or decrease the level of the input signal as it is entering the gain stage of the module. Most consoles have a pad switch built right into the input module. If your console does not have a built-in pad, you can insert an inline pad between the mic cable and the console input, such as the Audio-Technica AT8202.
2.1.2 - Mic Pads
In some cases, you may find that you have done everything right on the console to keep the level under control, but it still sounds distorted. This may be because the microphone itself is distorting. This can happen when the sound pressure being created is too high for the microphone to handle. This can sometimes be solved by using a pad built onto the microphone. This will prevent the microphone’s internal preamplifier from distorting, giving you a clean signal going into the console.
If that doesn’t work, you have three choices: back
the mic away from the sound source, choose a different microphone, or
get the musician to play/sing/jam quieter...and good luck with THAT!
2.1.3 - D.I. (Direct
Input) Boxes
When using line-level devices, such as keyboards or guitars, it may be practical to use a DI box, such as the Rolls DB25. This device allows you to plug a line level output into a mic input. This is especially helpful if your console has no line level inputs.
Remember that each part of the console is a separate device. So even though your output meters may say that you have proper levels, you can still introduce distortion. If you turn up the mic gain too much, the input module will be distorted, and no matter how much you turn down the fader, it will still sound bad. So you must pay attention to the amount of signal that you feed into each part of the console. This is called gain structure.
Think of it like painting a wall in several coats. If a fly lands on the wall while you’re painting the first coat and you paint over him, you’ll never be able to “cover it up” by applying more coats of paint. Unless you remove the fly during the first coat, it will not go away. The same is true for gain structure. If you distort your signal as it enters the input module, the distortion will pass onto the main bus.
Using the solo and PFL features on your console will help this process. Most consoles will directly reflect the signal you are soloing in the meters. For example, suppose you are bringing a mic into a module, assigning it to a group module, and then to the master module. You can solo the input module first to see if you are allowing too much signal into the module. Then you can solo the group module. The meter will now show you the signal in the group module alone. If that’s good, then you can look at the main meters to monitor the mix bus.
*Some consoles have meters all over the place for easy gain structure monitoring, while others only have a few meters. Since metering is unique in every console, make sure you read the manual and understand exactly what the meters are telling you.
2.2.1 - Distortion
I’ve mentioned distortion a few times, and before we continue, we should discuss this in greater detail. Distortion is ANY deviation from the original sound, whether intended or not. For example, you may have noticed how a song will have slight differences in sound when played through different speaker systems. This is because no speaker is perfect. All sound systems, no matter how expensive, will distort the original sound to some extent.
The type of distortion that most people are
familiar with is
called clipping. It is called clipping
because of the “clipped” appearance of the sine wave when viewed with
an
oscilloscope, (see figure 3). The
sound clipping makes is very recognizable and can be described as a
“crunched”
sound.
FIGURE 3:
There are several types of distortion, such as total harmonic distortion (THD), crossover distortion, intermodulation, etc. Hereafter in this book when we discuss distortion, we will be referring to clipping distortion.
There are many, many things that make a good mix. But I would consider two of these things to be the most important. The first is the obvious one – relative levels. That’s what mixing is all about – setting the level of each individual signal relative to all the others. The second is not so obvious – proper levels.
2.3.1 - “Where Do You
Put It???”
Most consoles are equipped with VU (Volume Units)
meters. A steady-state signal,
(unchanging in terms of frequency and amplitude, or a sine wave tone)
when
generated to the point where the VU needles are sitting exactly on
zero, (see
figure 4) will produce 1.228 volts AC on the output of a line
level
bus. When you do the math, 1.228 volts
usually reflects a +4 dB output, or the standard reference output for
line
level devices. Now, why in the world do
we need to know all of that gibberish???
FIGURE 4:
2.3.2 - Unity Gain
Proper levels – that’s why. (And by the way, calling it gibberish is unprofessional. If you are serious about being an engineer, prepare yourself to walk the walk, and talk the talk).
When you are mixing with the intention of dumping
the mix to
a 2-track master, you must set unity gain between the console (source)
and the
destination tape machine (see figure 5).
You can do this using a steady state tone set at 0 VU. Put the tape machine into input mode and
adjust the input level until it’s at 0 VU as well.
This will ensure that, as long as you keep
your mix on the console within appropriate levels, your tape machine
will be
receiving the proper levels. This is extremely important. If you
were to set the tape machine levels
too low, (see figure 6A) your mix would record too low on the
tape. If the levels were set high, (see
figure 6B)
your mix would record too hot, causing distortion.
FIGURE 5:
FIGURE 6 - Unity Gain Errors:
2.3.3 - Unity Gain
Using Digital Devices
Digital machines, such as DAT, minidisc recorders or CD burners are a slightly different animal when it comes to level setting. The goal with digital recorders is to get as much signal into the machine without going “over.” In the case of a studio mix, you simply cue the mix to the loudest section. Play this section over and over while adjusting the input level of the digital 2-track recorder. Make sure the “over” lights on the digital machine never light up. An occasional peak is fine with analog machines, but digital distortion is not gradual like analog. When a digital machine peaks, believe me, you’ll know it!
In the case of a live mix, you have no idea what
the loudest
section is going to be, so you will need to set unity gain between the
console
and the digital tape machine. For
digital machines, it is generally accepted that a 0 VU signal from the
console
should read at -20 dB on the digital 2-track, (see figure 7). This is to mirror the fact that consoles
generally have 20 dB of headroom above 0 VU before they go into
distortion. So again, as long as you
keep your mix on the console within appropriate levels, your tape
machine will
be receiving the proper levels.
FIGURE
7:
2.3.4 - “So...What IS
the Right Level During the Mix?”
I have generally categorized program material into two main areas – music and speech. Music, at its highest level, (that is, the loudest part of the song) should average on the VU’s between -3 and 0. Speech, since it generally has lots of peaks and dips, should average between -7 and -3.
Whether mixing speech or music, you should avoid peaking. Most VU meters have an LED that lights up when the signal is too hot. When this happens, you might wonder, “why does the peak light come one, but the needle isn’t anywhere near the top?” The reason is because VU meters are slow to react. When a sudden peak occurs, the needle physically cannot move fast enough to display that peak. For this reason, many people refer to VU’s as average meters.
2.3.5 - “Well, Who
Cares Anyway?”
“If my mix level is too low, I can just turn up the control room right?” Wrong. And again, let’s not be unprofessional. If it didn’t matter, it wouldn’t be in the book.
By keeping your levels in the appropriate range, you are balancing the medium between distortion and a poor signal-to-noise ratio. Distortion we’ve discussed, and it’s usually obvious when you’re talking about clipping-style distortion. A poor signal-to-noise ratio can lead to a “hissy” sounding mix. Here’s why:
All electronic devices have a certain amount of “self noise.” This is a slight, unavoidable hiss that is inherent in microphones, consoles, tape machines, and everything else electronic, whether passive or active. Some have less self noise than others, and those usually cost more.
Signal-to-noise ratio refers to the difference in level between the audio signal and the device’s self noise. The greater your input signal, the greater the ratio in relation to the noise. In other words, if your input signal is low, you become forced to turn up other circuits – thereby amplifying the noise.
Most consoles have solo functions that allow you to monitor a single module or any of the busses. But you need to understand what is happening when you hit a solo button. If you’re not careful, you can create unwanted results when you hit the solo button.
There are two basic kinds of solo functions: destructive and non-destructive. The word “destructive” is relative to the bus assignments. In other words, hitting the solo button while in a destructive solo mode could detrimentally affect the output of your busses.
In destructive mode, soloing a module will instruct all other modules to mute, leaving only the module that you’ve soloed. This is usually called “in-place” soloing, since the level and panning of the module are unaffected, showing you how that module is actually “placed” in the mix in terms of level and spatial imaging. The potential “destruction” is due to the fact that all other modules are muted, and therefore are no longer being routed to their destination busses. So if you hit the solo button while you are recording your mix to 2-track, you will ruin the mix. And if you’re mixing live to 2-track or broadcast, you’ve not only ruined the mix, you will have the producer jumping down your throat faster than you can say “oops!”
In-place soloing is not always a bad thing. In some mix situations, you can use it to your advantage. Suppose you are mixing a song with a quick breakdown right before the guitar solo. You decide that you want everything to suddenly mute except for the lead guitar. Hitting the solo on the guitar track takes care of it. So even though it is considered destructive, that does not necessarily mean “bad.”
*Yet another example of the importance of understanding the function…not just the process.
SECTION 2.5 –
CONNECTIONS
Now that we know most of the basic functions of the console,
we must discuss how to hook everything up in the control room. This is a little difficult to discuss,
because there are numerous ways to connect a control room.
It is going to depend on your needs.
It is important to know that ANY output can be fed
into ANY
input. However, consoles are designed
with certain functions in mind to serve the practicality of those
functions. For example, you could
connect your control room monitors
from the main output, but then you wouldn’t be able to hear a PFL, or
non-destructive solo. Also, you would
not be able to control your monitor level with the “control room level”
control.
In this section, we will discuss some of the connection “standards.”
Issues of balancing, unbalancing, and the combinations of connection types will be discussed later in this book. For now, let’s concentrate of pure signal flow.
2.5.1 - Monitors
Again, we’re not talking about stage monitors. These are control room and studio monitors.
Most consoles have a dedicated control room output that is intended to feed a pair of speakers. In fact, some consoles have more than one set of monitoring outputs for far-field speakers, near-field speakers and studio monitors.
To connect these speakers, you first need to know what kind of speakers you are using – passive or active. Passive, or non-powered speakers, will need an amplifier between the console and the speakers. In this case, connect the control room output into the amplifier. Then connect the amplified, speaker output into the speakers.
Active, or powered speakers, do not need an amplifier. In fact, if you did use an amplifier with a powered speaker, you could potentially damage the speakers or start a fire, (no kidding). So, for powered speakers, simply connect the control room output into the speaker input.
For some smaller consoles that are not equipped with a dedicated monitor output, you may need to use the headphone output to feed a set of speakers
*Make sure you follow the manufacturer’s instructions when connecting these devices.
2.5.2 - 2-Track Tape
Machines
As mentioned earlier, the main output of a console is typically connected into the main product. In a recording studio, the main product is the mix, which is recorded onto a 2-track tape machine, hard-disk recorder, or DAW (Digital Audio Workstation). Simply connect both of the main outputs of the console into the inputs of the 2-track recorder.
If you have more than one 2-track recorder, you can make a “Y” cable to connect them from the same output. However, you should not “Y” into more than three devices. Soldering techniques for various cables and adaptors will be discussed later in this book.
2.5.3 - Multi-Track
Recorders
Connection of multi-track recorders will vary depending on the console you are using. Some consoles are simply not designed for multi-track recording, and others, while they were not specifically designed for multi-tracking, can easily be configured to do so.
The typical way to connect a console to a multi-track recorder is using the group bus outputs. For example, if you have an eight-bus console (that is, eight sub-groups), you would connect the output of group one into input “1” of the multi-track recorder. Then you would connect group out “2” to multi-track input “2”, then 3 to 3, 4 to 4, etc., (see figure xxx). Now, when you want to record a microphone (or other sound source) to multi-track, you would assign that mic from the input module to the group bus that corresponds with the track you want to record onto. Use the group bus fader to control how much signal is getting into the multi-track recorder.
Now, once you have recorded something, how do you listen to it from the multi-track recorder? This is where it is best to have a console specifically designed to handle multi-tracking. In other words, you need a console with tape returns. Tape returns are like mini-input modules designed to accept the outputs of the multi-track recorders. This is so you can create a mix from multi-track without having to re-configure your input modules.
The console of choice for multi-tracking is called an in-line console. This is a console that has the input module and the tape return located in the same module.. These are typically referred to as the channel path and mix path, respectively.
2.5.4 - Effects
Devices / Parallel Processing
Effects devices are those units that add a special effect to your sound, such as reverb, delay, pitch shift, chorusing, and many others.
There are a number of ways in which you can connect effect devices, but in general, you should use parallel processing. This means returning the output of the effect unit to a separate input module or effect return, as opposed to using it as an insert device, (see next section). Getting signal into the device is accomplished by using the auxiliary busses or the direct outputs from the source module.
There is a disadvantage, however, to using a direct output to drive the input of an effects unit. If you only have one effect unit, using a direct out will hog the device with that single input. If you decide later that you want the same effect on an additional sound source, you will have to reconfigure the connection.
By using an auxiliary bus, you can send all of your sound sources in the console to a single effect unit. Also, using the auxiliaries allows you the flexibility of deciding how much of the effect should be applied to each sound source. This is still considered parallel processing, since you will be returning the output of the effect unit to a separate input module.
2.5.5 – Insert
Devices / Serial Processing
The opposite of parallel processing is serial processing. Many engineers refer to this process as “inserting.” This is accomplished by using the insert section of the input module, group busses or master bus. On the console, this may be labeled in a number of different ways: insert, send and return, or channel access. In any event, the concept is the same. “Inserting” is sending a signal from a module or bus into a device and, rather than returning to a separate channel, returning the output of the device back into the path of the same module or bus.
For example, suppose you have a microphone in module 1 of the console, and you want to apply a parametric EQ to that signal. You would begin by connecting the “send” from that module into the EQ – that gets the signal into the equalizer. Then, connect the output of the EQ into the “return” of module 1.
As illustrated in figure xxx, the new equalized signal from the EQ interrupts the un-equalized signal that appeared before the send point. In effect, what you have done is made that equalizer a part of that module. By placing it in the serial path of the module, a signal that enters that module must pass through the EQ before it can leave the module.
2.5.6 – “When Should
You Insert?”
This question often comes up when talking about the difference between parallel and serial processing. The truth is, there is no right answer. There are lots of different ways to wire, configure, mix, etc. However, there are a few so-called “accepted standards” when it comes to this issue.
Parallel processing is normally used for effects – things that are added to a sound to enhance the existing sound. This includes reverb, delay, etc.
Serial processing is used for devices that modify the sound. For example, if you used an equalizer in parallel processing, you would be defeating your purpose. By returning the output of the EQ to a separate module, you would still be hearing the un-equalized sound in the mix. That wouldn’t make much sense, would it?
Serial processing is typically used for equalizers
and
dynamics processors (compressors, gates, limiters and expanders).
2.5.7 – Patch Bays
A patch bay is certainly not necessary, but it can
simplify
life in a control room – if it is set up properly.
Improperly wired patch bays, or a poor
understanding of patch bays, can turn into your worst nightmare. So, before you incorporate a patch bay into
your setup, you’d better know what you’re doing.
CHAPTER THREE
CHAPTER THREE – STUDIO SETUP & SAFETY
There are various reasons for understanding the proper way to set up your studio, but the most important are equipment safety and liability. Yes, liability. If a client trips and ruins his or her nose job due to your negligence, you may have a liability lawsuit on your hands. This chapter will give you some guidelines that may help you avoid such a calamity. But be warned – even if you do everything possible to avoid an accident, your client may still find a way to trip, bust their face, and blame it on you.
Now…on that happy note…
SECTION 3.1 – MICROPHONE
SETUP
Microphones are probably the most vulnerable piece of equipment in the studio. This is because they move around a lot…being set up on a stand, taken down, moved, swiveled, etc. Some microphones can be dropped from the ceiling without a single sign of damage, except maybe a dented windscreen. Other mics can be destroyed by being dropped six inches. The problem is, you never know which ones will survive a drop until you drop them.
The value of your microphones could far exceed the sticker price, particularly if you get your hands on some vintage mics that have been out of production for fifty years. The original price may have been $500, but today’s auction price could exceed $20,000. (That’s not a typo!) I personally know engineers that would consider donating a kidney to get their hands on a vintage Telefunken.
The point is – treat ALL microphones with extreme care. If you find yourself in an assistant engineer position, remember that those mics are someone’s prized possessions. Treat them as if they were YOUR kidneys!
The most crucial time for TLC with a microphone is during setup and teardown – while it’s going on or off of the stand.
Step One – Put the cable on the stand first. If you put the mic on the stand first, there will be some time in which the mic could fall from the clip and plummet to its death. Clips are sometimes precarious, and should not be trusted. By putting the cable on the stand first, you can connect the cable to the mic right away, giving it a measure of security in case the clip fails. If it does, the cable (hopefully) will stop the mic’s fall to the floor.
Step Two – Wrap the cable around the stand and use cable clips, (figure 10). This step ties into the first step described above. Wrapping the cable on the stand gives the microphone another level of security should it fall from the clip.
If the mic is being used as a vocal mic, and the vocalist plans to take the mic off of the stand, it would be better NOT to wrap it. If the vocalist has to unwrap the cable, it will make them upset, and they will most likely break something in their attempt to unwrap it. In this case, simply drape the cable over the key on the elbow.
Step Three – Connecting the clip. Get in the habit of turning the stand into the clip, not threading the clip onto the stand. Some mics have the clips permanently installed onto them, and others are kept in shock-mounts that practically require an M.D. to remove. A common mistake, in this case, is for an engineer to turn the clip - mic and all - onto the stand. By doing this, there is a good chance that you will drop the mic or cross thread the clip.
Loosen the arm on the end of the stand, (if possible), and hold the clip still while turning the arm. Again, get into the habit of using this technique, whether or not the mic is attached.
Step Four – Leave
slack on both ends. Leaving a nice
coil of cable next to the mic will help to prevent accidents. With no slack, (figure 11) someone
tripping on the mic cable will immediately result in a toppled stand. With two or three coils next to the stand,
(figure 10) the clumsy one will (hopefully) notice that they
are about
to cause an accident before it happens.
The same rule applies to the other end of the cable. Failure to leave slack can result in an accident and a damaged cable or input panel.
Step Five – Use the path of least traffic. Running the cable from the mic to the panel should be preceded by a little investigation. Where will people be walking? Is this cable out of their path? What is the safest path for this cable? You need to use some common sense in this instance, but in figure xxx, you will find a few suggestions on how to approach some common situations.
Step Six – Wrap your cables properly. Microphone cables have two conductors and a shield inside them, and twisting the cable can put tension on the internal conductors to the point where they could break. If this happens, the only way to repair the cable is to cut it where it broke and solder on a new connector. However, how do find the break if it’s covered by a rubber jacket?
You can avoid this mess by properly wrapping your cables. This also saves you the trouble of unwrapping a big ball of frustration!
Cables should be wrapped in nice, neat coils about the size of a steering wheel. NEVER, NEVER, NEVER, NEVER, NEVER wrap cables around your elbow and forearm! This causes a lot of strain on the internal conductors. It also results in a big mess once you pull your arm out.
SECTION
3.2 – IN THE CONTRL ROOM
When you begin setting up for a session, you should follow certain steps to ensure a smooth operation.
Step One – Proper turn-on sequence. Remember this rule of thumb: amplifier on last, off first. This is to prevent you from “popping” your speakers.
When you turn on anything in your equipment chain, the first half-cycle (1/120th of a second) of the AC that powers that unit is actually DC until the cycle resolves. This also occurs when you turn things off. If the amplifier is on while something connected to it is turned on or off, it will amplify that sudden surge of DC. Incidentally, speakers don’t like DC. In fact, continued exposure to DC in this manner can lead to damaged speakers.
The other danger is to your ears. If the amplifier is on, and turned up full-blast, that pop of DC could cause permanent hearing loss.
By the way, when you leave a rock concert and your ears are ringing, that sound you hear is actually nerve cells dying a painful death. They’re singing goodbye – for the last time. That’s right – permanent hearing loss. So, if you haven’t figured this out already, your most valuable asset as an audio engineer is your ears. Take care of them.
Step Two – Establish monitoring first. This means making sure you have sound coming out of your speakers before you ever begin dialing in mics or mixing. Following this step takes a level of troubleshooting out of the equation when you find a problem.
For example, if you begin dialing in a mic and find that it’s not working, you will most likely look for things like a bad cable, phantom power not being on, etc. But if you fail to notice that you forgot to turn up your control room level, you will be chasing your tail all day!
The quickest way to establish monitoring is to run a known signal into an input module and go through a checklist: gain up, assign mic, fader up, bus master up, monitor source selected, control room level up. If you see the meters moving, but still no sound, the monitor amp is probably turned off.
Step Three – Don’t turn up that fader yet! When you start dialing in a mic, one of the last things you should do is turn up the input module fader. This is to avoid any popping or other unexpected loud sounds. Keeping the fader down until you need it is simply a protection. Also, turn the fader down (or mute the input module) when you need to make a change, such as replacing a cable.