1.5.1 Overview

There are three things you may want to set up in order to work with this book. It’s possible that you’ll need only one of the first two, depending on your focus.  Everyone will probably need the third to work with the suggested exercises in this book.

  • A digital audio workstation
  • A live sound reinforcement system
  • Software on your computer to do hands-on exercises

First, we assume most readers will want their own digital audio workstation (DAW), consisting of a computer and the associated hardware and software for at-home or professional sound production (Figure 1.1).  Suggestions for particular components or component types are given in Section 1.5.2.

Figure 1.1 Basic setup and signal flow of a digital audio workstation
Figure 1.1 Basic setup and signal flow of a digital audio workstation

Secondly, it’s possible that you’ll also be using equipment for live performances.  A live performance setup is pictured in Figure 1.2.  Much of the equipment and connectivity is the same as or similar to equipment in a DAW.

Figure 1.2 A simple live sound reinforcement system
Figure 1.2 A simple live sound reinforcement system

Thirdly, to use this book most effectively you’ll need to gather some additional software so that you can view the book’s learning supplements, complete some of the exercises, and even do your own experiments.  The learning supplements include:

  • Flash interactive tutorials, accessible at our website and viewable within a standard web browser with the Flash plug-in installed (generally included and enabled by default).
  • Max demo patchers, which can be viewed with the Max run-time environment, freely downloadable from the Cycling ’74 website.  (If you wish to do the Max programming exercises you’ll need to purchase Max, or use the free alternative, Pure Data.)
  • MATLAB exercises (with Octave as a freeware alternative).
  • Audio and MIDI processing worksheets that can be done in Logic, Cakewalk Sonar, Reason, Audition, Audacity, or some other digital audio or MIDI processing program.
  • C and Java programs, for which you’ll need C and/or Java compilers and IDEs if you wish to complete these assignments.

We don’t expect that you’ll want to go through all the learning supplements or do all the exercises.  You should choose the types of learning supplements that are useful to you and gather the necessary software accordingly.  We give more information about the software for the learning supplements in 1.5.3.

In the sections that follow, we use a number of technical terms with only brief, if any, definitions, assuming that you have a basic computer vocabulary with regard to RAM, hard drives, sound cards, and so forth.  Even if you don’t fully understand all the terminology, when you’re buying hardware and software to equip your DAW, you can refer your sales rep to this information to help you with your purchases.  All terminology will be defined more completely as the book progresses.

1.5.2 Hardware for Digital Audio and MIDI Processing

1.5.2.1 Computer System Requirements

Table 1.1 gives our recommendations for the components of an affordable DAW as well as equipment such as loudspeakers needed for live performances.  Of course technology changes very quickly, so make sure to do your own research on the particular models of the components when you’re ready to buy.  The components listed in the table are a good starting point.  Each category of components is explained in the sections that follow. We’ve omitted optional devices from the table but include them in the discussion below.

[listtable width=”50%”]

  • Computer
    • Desktop or laptop with a fast processor, Mac or Windows operating system.
    • RAM – at least 2 GB.
    • Hard drive – a fast hard drive (separate and in addition to the operating system hard drive) dedicated to audio storage, at least 7200 RPM.
  • Audio interface (i.e., sound card)
    • External audio interface with XLR connections. The audio interface may also serve as a MIDI interface.
  • Microphone
    • Dynamic microphone with XLR connection.
    • Possibly a condenser microphone as well.
  • Cables and connectors
    • XLR cables for microphones, others as needed for peripheral devices.
  • MIDI controller
    • A MIDI piano keyboard, may or may not include additional buttons and knobs. May have USB connectivity or require a MIDI interface. Possible all-in-one devices include both a keyboard and basic audio interface.
  • Monitoring loudspeakers
    • Monitors with flat frequency response (so you hear an unaltered representation of the audio).
  • Studio headphones
    • Closed-back headphones (for better isolation).
  • Mixing Console
    • Analog or digital mixer, as needed.
  • Loudspeakers
    • Loudspeakers with amplifiers and directional/frequency responses appropriate for the listening space.

[/listtable]

Table 1.1 Basic hardware components for a DAW and live performance setups

A desktop or even a laptop computer with a fast processor is sufficient as the starting point for your DAW.   Audio and MIDI processing make heavy demands on your computer’s RAM (random-access memory) – the dynamic memory of a computer that holds data and programs while they’re running.   When you edit or play digital audio, a part of RAM called a buffer is set aside to hold the portion of audio data that you’re going to need next.  If your computer had to go all the way to the hard disk drive each time it needed to get the data, it wouldn’t be able to play the audio in real-time.  Buffering is a process of pulling data off permanent storage – the hard drive – and holding them in RAM so that the sound is immediately available to be played or processed.  Audio is divided into streams, and often multiple audio streams are active at once, which implies that your computer has to set aside multiple buffers.  MIDI instruments and samplers also make heavy demands on RAM.  When a sampler is used, MIDI creates the sound of a chosen musical instrument by means of short audio clips called samples that are stored on the computer.  All of these audio samples have to be loaded into RAM so they can be instantly accessible to the MIDI keyboard.  For these reasons, you’ll probably need to upgrade the RAM capacity on your computer. A good place to begin is with 2 GB of RAM.  RAM is easily upgradeable and can be increased later if needed.  You can check the system requirements of your audio software for the specific RAM requirements of each application program.

[aside]Early digital audio workstations utilized SCSI hard drives. These drives could be chained together in a combination of internal and external drives. Each hard drive could only hold enough data to accommodate a few tracks of audio, so the multitrack audio software at the time would perform a round-robin strategy of assigning audio data from different tracks to different SCSI hard drives in the chain. These SCSI hard drives, while small in size, provided impressive speed and performance and to this day, no external hard drive system can completely match the speed, performance, and reliability of external SCSI hard drives when used in digital audio.[/aside]

You also need memory for permanent storage of your audio data – a large capacity hard disk drive. Most hard drives found in the standard configuration for desktop and laptop computers are not fast enough to keep up with real-time processing of digital audio. Your RAM buffers the audio playback streams to maintain the flow of data to your sound card, but your hard drive also needs to be fast enough to keep that buffer full of data. Digital audio processing requires at least a 7200-RPM hard drive hard that is dedicated to holding your audio files. That is, the hard drive needs to be a secondary one, in addition to your system hard drive. If you have a desktop computer, you might be able to install this second hard drive internally, but if you have a laptop or would simply like the ability to take your data with you, you’ll need an external hard drive. The capacity of this hard drive should be as large as you can afford. At CD quality, digital audio files consume around ten megabytes per minute of sound. One minute of sound can easily consume one gigabyte of space on your hard drive. This is because you often work simultaneously with multiple tracks – sometimes even ten or more. In addition to these tracks, there are backup copies of the audio that are automatically created as you work.

New technologies are emerging that have the potential for eliminating the hard drive bottleneck. Mac computers now offer the Thunderbolt interface with bi-directional data transfer and a data rate of up to 10 Gb/s. Solid state hard drives (SSDs) – distinguished by the fact that they have no moving parts – are fast and reliable. As these become more affordable, they may be the disk drives of choice for audio.

Before the advent of Thunderbolt and SSDs, the choice of external hard drives was between FireWire (IEEE 1394), USB interfaces, and eSATA. An advantage of FireWire over USB hard drives is that FireWire is not host-based. A host-based system like a USB drive does not get its own hardware address in the computer system. This means that the CPU has to manage how the data move around on the USB bus. The data being transferred must first go through the CPU, which slows down the CPU by taking its attention away from its other tasks. FireWire devices, on the other hand, can transmit without running the data through the CPU first. FireWire also provides true bi-directional data transfers — simultaneously sending and receiving data. USB devices must alternate between sending and receiving. For Mac computers, FireWire drives are preferable to USB for simultaneous real-time recording and playback of multiple digital audio streams. FireWire speeds of 400 or 800 are fine. These numbers refer to approximate Mb/s half-duplex maximum data transfer rates. However, mixing 400 and 800 devices on the same bus is not a good idea. It’s best just to pick one of the two speeds and make sure all your FireWire devices run at that speed.

The most important factor in choosing an external FireWire hard drive is the FireWire bridge chipset. This is the circuit that interfaces the IDE or SATA hard drive sitting in the box to the FireWire bus. There are a few chipsets out there, but the only chipsets that are reliable for digital audio are the Oxford FireWire chipsets. Make sure to confirm that the external FireWire hard drive you want to purchase uses an Oxford chipset.

Unfortunately, recent Windows operating systems have proven somewhat buggy for FireWire, so many Windows-based DAWs use USB interfaces, despite their shortcomings. Alternatively, Windows computers could use eSATA hard drives, which perform just like internal SATA drives.

1.5.2.2 Digital Audio Interface

In order to work with digital sound, you need a device that can convert physical sound waves captured by microphones or other inputs into digital data for processing, and then convert the digital data back into analog form for your loudspeakers to reproduce as audible sound. Audio interfaces (or sound cards) provide this functionality.

Your computer probably came with a simple built-in sound card. This is suitable for basic playback or audio output, but to do recording with a high level of quality and control you need a more sophisticated, dedicated audio interface. There are many solutions out there. Leading manufacturers include AVID, M-Audio, MOTU, and Presonus. Things to look for when choosing an interface include how the box interfaces with the computer (USB, FireWire, PCI) and the number of inputs and outputs. You should have at least one low-impedance microphone input that uses an XLR connector. Some interfaces also come with instrument inputs that allow you to connect the output of an electric guitar attached with joyo pedal to record all the metal notes directly into your computer. Figure 1.3 and Figure 1.4 show examples of appropriate audio interfaces.

Figure 1.3 Presonus AudoBox USB audio interface
Figure 1.3 Presonus AudoBox USB audio interface
Figure 1.4 MOTU UltraLite mk3 FireWire audio interface
Figure 1.4 MOTU UltraLite mk3 FireWire audio interface

1.5.2.3 Drivers

A driver is a program that allows a peripheral device such as a printer or sound interface to communicate with your computer.  When you attach an external sound interface to your computer, you have to be sure that the appropriate driver is installed.  Generally you’re given a driver installation disk with the sound interface, but it’s better to go to the manufacturer’s website and download the latest version of the driver.  Be sure to download the version appropriate for your operating system.  Drivers are pretty easy to install.  You can look for instructions at the manufacturer’s website and follow the steps in the windows that pop up as you do the installation.  Remember that if you upgrade to a new operating system, you’ll probably need to upgrade your driver as well.  Some interfaces come with additional interface-related software that allows access to internal settings, controls, and DSP provided by the interface.  This extra software may be packaged with the driver or it may be optional, but either way it is usually quite handy to install as well.

1.5.2.4 MIDI Keyboard

A MIDI keyboard is required to input MIDI performance data into your computer. A MIDI keyboard itself makes no instrument sounds. It simply sends the MIDI data to the computer communicating the keys pressed and other performance data collected, and the software handles the playback of instruments and sounds. There exist MIDI keyboards that are a combination MIDI input device and audio interface.  These are called audio interface keyboards.  Consolidating the MIDI keyboard and the audio interface into one component is convenient because it’s easier to transport. The downside is that features and functionality may be more limited, and all the functionality is tied into one device, so if that one device breaks or becomes outdated, you lose both tools. Standalone MIDI controller keyboards connect either to your computer directly using USB, or to the MIDI input and output of a separate external audio interface. MIDI keyboards come in several sizes. Your choice of size depends on how many keys you need. Figure 1.5 and Figure 1.6 show examples of USB MIDI keyboard controllers.

Figure 1.5 M-Audio Oxygen25 25-key MIDI keyboard controller
Figure 1.5 M-Audio Oxygen25 25-key MIDI keyboard controller
Figure 1.6 AKAI MPK49 49-key MIDI keyboard controller
Figure 1.6 AKAI MPK49 49-key MIDI keyboard controller

1.5.2.5 Recording Devices

Recording is one of the fundamental activities in working with sound.  So what type of recording devices do you need?  One possibility is to connect a microphone to your computer and use software on your computer as the recording interface.  A computer based digital audio workstation offers multiple channels of recording along with editing, mixing, and processing all in the same system.  However, these workstations are not very portable or rugged, so they’re often found in fixed recording studio setups.

Sometimes you may need to get out into the world to do your recording.  Small portable recorders like the one shown in Figure 1.7 are available for field recordings.  A disadvantage of such a device is that the number of inputs is usually limited to two to four channels. These recorders often have one or two built-in microphones with the added option of connecting external microphones as well.

Dedicated multitrack hardware recorders as shown in Figure 1.8 and Figure 1.9 are available for situations where portability and high channel counts are desirable. These recorders are generally very reliable but offer little opportunity for editing, mixing, and processing the recording. The recording needs to be transferred to another system afterwards for those tasks.

Figure 1.7 Zoom H4n small portable audio recorder with built-in microphones
Figure 1.7 Zoom H4n small portable audio recorder with built-in microphones
Figure 1.8 Sound Devices 788t 12-channel multi-track recorder
Figure 1.8 Sound Devices 788t 12-channel multi-track recorder
Figure 1.9 Tascam X-48 mkII dedicated 48-channel multi-track recorder
Figure 1.9 Tascam X-48 mkII dedicated 48-channel multi-track recorder

1.5.2.6 Microphones

Your computer may have come with a microphone suitable for gaming, voice recognition, or audio/video conferencing. However, that’s not a suitable recording microphone. You need something that gives better quality and a wider frequency response. The audio interfaces we recommend in Section 1.5.2.2 include professional microphone inputs, and you need a professional microphone that’s compatible with these inputs. Let’s look at the basic types of microphones that you have to choose from.

The technology used inside a microphone has an impact on the quality of the sound it can capture. One common microphone technology uses a coil that moves inside a magnet, which happens to also be the reverse of how a loudspeaker works. These are called dynamic microphones. The coil is attached to a diaphragm that responds to the changing air pressure of a sound wave, and as the coil moves inside the magnet, an alternating current is generated on the microphone cable that is an electrical representation of the sound. Dynamic microphones are very durable and can be used reliably in any situation since they are passive devices, meaning that they require no external power source. Most dynamic microphones tend to come in a handheld size and are fairly inexpensive. In addition to being durable, they’re not as sensitive as other types of microphones. This lower sensitivity can be very effective in noisy environments when you’re trying to capture isolated sounds. However, dynamic microphones are not very good at picking up transient sounds – quick loud bursts like drum hits. They also may not pick up high frequencies as well as capacitance microphones do, which may compromise the clarity of certain kinds of sounds you’ll want to record. In general, a dynamic microphone may come in handy during a high-energy live performance situation, yet it may not provide the same quality and fidelity as other types of microphones when used in a quiet, controlled recording environment.

Another type of microphone is a capacitance or condenser microphone. This type of microphone uses an electronic component called a capacitor as the transducer. The capacitor is made of two parallel conductive plates, physically separated by an air space. One of the plates requires a polarizing electrical charge, so condenser microphones require an external power supply. This is typically from a 48-volt DC power source called phantom power, but can sometimes be provided by a battery. The conductive plates are very thin, and when sound waves push against them, the distance between the plates changes, varying the charge accordingly and creating an electrical representation of the sound. Condenser microphones are much more sensitive than dynamic microphones. Consequently, they pick up much more detail in the sound, and even barely perceptible background sounds may end up being quite audible in the recording. This extra sensitivity results in a much better transient response and a much more uniform frequency response, reaching into very high frequencies. Because the transducers in condenser microphones are simple capacitors and don’t require a weighty magnet, condenser microphones can be made quite large without becoming too heavy.  They also can be made quite small, allowing them to be easily  concealed. The smaller size also allows them to pick up high frequencies coming from various angles in a more uniform manner. A disadvantage of the capacitor microphone is that it requires external power, although this is often easily handled by most interfaces and mixing consoles. Also, capacitor elements can be quite delicate, and are much more easily damaged by excessive force or moisture. The features of a condenser microphone often result in a much higher quality signal, but this comes at a higher price. Top-of-the-line condenser microphones can cost thousands of dollars.

Electret condenser microphones are a type of condenser microphone in which the back plate of the capacitor is permanently charged at the factory. This means the microphone does not require a power supply to function, but it often requires an extra powered preamplifier to boost the signal to a sufficient voltage. Easy to manufacture and often miniature in size, electret condenser microphones are used for the vast majority of built-in microphones in phone, computer, and portable device technologies. While easy and economical to produce, electret microphones aren’t necessarily of lower quality. In the field of professional audio they can be found in lavaliere microphones attached to clothing or concealed for live performance. In these cases, the small microphones are typically connected to a wireless transmitter with a battery that powers the preamplifier as well as the RF transmitter circuitry.

Generally speaking, you want to get the microphone as close as possible to the sound source you want to capture. This improves your signal-to-noise ratio. When getting the microphone close to the source is not practical – such as when you’re recording a large choir, performing group, or conference meeting – a type of microphone called a pressure zone microphone (PZM) can be useful. A PZM, also called a boundary microphone, is usually made of a small electret condenser microphone attached to a metal plate with the microphone pointed at the plate rather than the source itself. These microphones work best when attached to a large reflective surface such as a hard stage floor or large conference table. The operating principle of the pressure zone is that as a sound wave encounters a large reflective surface, the pressure at the surface is much higher because it’s a combination of the direct and reflected energy. Essentially this extra pressure results in a captured amplitude boost, a benefit normally available only by getting the microphone much closer to the sound source. With a PZM, you can capture a sound at a sufficiently high volume even from a significant distance. This can be quite useful for video teleconferencing when a large group of people must be at a greater distance to the microphone, as well as in live performance where microphones are placed at the edge of the stage. The downside to a PZM is that the physical coupling to the boundary surface means that other sounds such as foot noise, paper movement, and table bumps are picked up just as well as the sound you’re trying to capture. As a result, signal-to-noise ratio tends to be fairly low. In a live sound reinforcement situation you can also have acoustic gain problems if you aren’t careful about the physical relationship between the microphone and the loudspeaker. Since the microphone is capturing the performer from a great distance, the loudspeakers directly over the stage could easily be the same distance or less distance from the microphone as the performer, resulting in the sound from the loudspeaker arriving at the PZM at the same level or higher than the sound from the performer, a perfect recipe for feedback. Feedback and acoustic gain are covered in more detail in Chapter 4.

As part of a newer trend in this digital age, the prevalence of USB digital microphones is on the rise. Many manufacturers are offering a USB version of their popular microphones, both condenser and dynamic. These microphones output a digital audio stream and are intended for direct recording into a computer software program, without the need for any additional preamplifier or audio interface equipment. You could even think of them as microphone-interface hybrids, essentially performing the duties of both. The benefits of these new digital microphones are of course simplicity, portability, and perhaps even cost if you consider not having to purchase the additional equipment and digital audio interface. However, while these USB microphones may be studio quality, there are some limitations that may influence your choice. Where traditional XLR cables can easily run over a hundred feet, USB cables have a maximum operable length of only 10 to 15 feet, which means you’re pretty tied down to your computer workstation. Additionally, having only a USB connection means you won’t be able to use the microphone in a live situation, or plug it into an analog mixing console, portable recorder, or any other piece of audio gear. Finally, a dedicated audio interface allows you to plug in multiple microphones and instruments, provides a multitude of output connections, and provides onboard DSP and mixing tools to help you get the most out of your audio setup and workflow. Since you’ll probably want to have a dedicated audio interface for these reasons anyway, you may be better off with a traditional microphone that interfaces with it, and is more flexible overall. That being said, a USB microphone could certainly be a handy addition to your everyday audio setup, particularly for situations when you’re travelling and need a self-contained, portable solution.

If you buy only one microphone, it should be a dynamic one. The most popular professional dynamic microphone is the Shure SM58. Everyone working with sound should have at least one of these microphones. They sound good, they’re inexpensive, and they’re virtually indestructible. Figure 1.10 is a photo of an SM58. If you want to purchase a good-quality studio condenser microphone and you have a recording environment where you can control the noise floor, consider one like the AKG C414 microphone. This is a classic microphone with an impressive sound quality. However, it has a tendency to pick up more than you want it to, so you need to use it in a controlled recording room where it isn’t going to pick up fan sounds, the hum from fluorescent lights, and the mosquitoes in the corner flapping their wings. Figure 1.11 is a photo of a C-414 microphone.

Figure 1.10 Shure SM58 dynamic microphone
Figure 1.10 Shure SM58 dynamic microphone

AKG C-414 condenser microphone
AKG C-414 condenser microphone

Another way to classify microphones is by their directionality.  The directionality of a microphone is its sensitivity to the range of audible frequencies coming from various angles, which can be depicted in a polar plot (also called a polar pattern).  The three main categories of microphone directionality are directional, bidirectional, and omnidirectional.

You can think of the polar pattern essentially as a top-down view of the microphone.  Around the edge circle are numbers in degrees, representing the direction at which sound is approaching the microphone.  0 degrees at the top of the circle is where the front of the microphone is pointing – often referred to as on-axis – and 180 degrees at the bottom of the circle is directly behind the microphone.  The concentric rings with decreasing numbers are the sound levels in decibels, abbreviated dB, with the outer ring representing 0 dB, or no loss in level. The blue line shows the decibel level at various angles.

We don’t explain decibels in detail until Chapter 4, but for now it’s sufficient to know that the more negative the dB value (closer to the center), the less the sound is picked up by the microphone at that angle.  This may seem a bit counterintuitive, but remember the polar plot has nothing to with distance, so getting closer to the center doesn’t mean getting closer to the microphone itself.  The polar pattern for an omnidirectional microphone is given in Figure 1.12.  As its name suggests, an omnidirectional microphone picks up sound equally from all directions.  You can see that reflected in the polar pattern, where the sound level remains at 0 dB as you move around the circle regardless of the angle, as indicated by the blue boldface outline.

Figure 1.12 Polar plot for an omnidirectional microphone
Figure 1.12 Polar plot for an omnidirectional microphone

A bidirectional microphone is often referred to as a figure-eight microphone.  It picks up sound with equal sensitivity at its front and back, but not at the sides.  You can see this in Figure 1.13, where the sound level decreases as you move around the microphone away from the front (0°) or rear (180°), and at either side (90° and 270°) the sound picked up by the microphone is essentially none.

Figure 1.13 Polar plot for a bidirectional microphone
Figure 1.13 Polar plot for a bidirectional microphone

Directional microphones can have a cardioid (Figure 1.14) a supercardioid (Figure 1.15), or a hypercardioid (Figure 1.16) pattern.  You can see why they’re called directional, as the cardioid microphone picks up sound in front but not behind the microphone.  The super and hypercardiod microphones behave similarly, offering a tighter frontal response with extra sound rejection at the sides. (The lobe of extra sound pickup at the rear of these patterns is simply an unintended side-effect of their focused design, but usually isn’t a big issue in practical situations.)

Figure 1.14 Polar plot for a cardioid microphone
Figure 1.14 Polar plot for a cardioid microphone
Figure 1.15 Polar plot for a supercardioid microphone
Figure 1.15 Polar plot for a supercardioid microphone
Figure 1.16 Polar plot for a hypercardioid microphone
Figure 1.16 Polar plot for a hypercardioid microphone

A special category of microphone called a shotgun microphone can be even more directional, depending on the length and design of the microphone (Figure 1.17). Shotgun microphones can be very useful in trying to pick up a specific sound from a noisy environment, often at a greater than the typical distance away from the source, without picking up the surrounding noise.

Figure 1.17 Polar plot for a shotgun microphone
Figure 1.17 Polar plot for a shotgun microphone

Some microphones offer the option of multiple, selectable polar patterns.  This is true of the condenser microphone shown back in Figure 1.11.  You can see five symbols on the front of the microphone representing the polar patterns from which you can choose, depending on the needs of what you’re recording.

Polar plots can be even more detailed than the ones above, showing different patterns depending on the frequency.  This is because microphones don’t pick up all frequencies equally from all directions. The plots in Figure 1.18 show the pickup patterns of a particular cardioid microphone for individual frequencies from 125 Hz up to 16000 Hz.  You’ll notice the polar pattern isn’t as clean as consistent as you might expect.  Even for a directional microphone, lower frequencies may often exhibit a more omnidirectional pattern, where higher frequencies can become even more directional.

Figure 1.18 Polar plot of a dynamic cardioid microphone, showing pickup patters for various frequencies
Figure 1.18 Polar plot of a dynamic cardioid microphone, showing pickup patters for various frequencies

[aside]Shure hosts an interactive tool on their website called the Shure Microphone Listening Lab where you can audition all the various microphones in their catalog. You can try it out yourself at http://www.shure.com/americas/support/tools/mic-listening-lab[/aside]

The sensitivity that a microphone has to sounds at different frequencies is called its frequency response (a term also used to describe the behavior of filters in later chapters).  If a microphone picks up all frequencies equally, it has a flat frequency response.  However, a perfectly flat frequency response is not always desirable. The Shure SM58 microphone’s popularity, for example, can be attributed in part to increased sensitivity at higher frequencies, which can make the human voice more clear and intelligible.  Of course, you could achieve this same frequency response using an EQ (i.e., an equalization process that adjusts frequencies), but if you can get a microphone that naturally sounds good for the sound you’re trying to capture, it can save you time, effort, and money.

Figure 1.19 On-axis frequency response of the Shure SM58 microphone
Figure 1.19 On-axis frequency response of the Shure SM58 microphone

Some microphones may have a very flat frequency response on-axis but due to the directional characteristics, that frequency response can become very uneven when off-axis. This is important to keep in mind when choosing a microphone. If the sound you’re trying to record is stationary and you can get the microphone pointed directly at the sound, then a directional microphone can be very effective at capturing the sound you want without capturing the sounds you don’t want. If the sound moves around or if you can’t get the microphone pointed directly on-axis with the sound, you may need to use an omnidirectional microphone in order to keep the frequency response consistent. However, an omnidirectional microphone is very ineffective at rejecting other sounds in the environment.  Of course, that’s not always a bad thing, as with measuring and analyzing sounds in a room when you want to make sure you’re picking up everything that’s happening in the environment, and as accurately and transparently as possible.  In that case, an omnidirectional microphone with a flat frequency response is ideal.

Figure 1.20 A small-diaphragm omnidirectional microphone specialized for measurement use
Figure 1.20 A small-diaphragm omnidirectional microphone specialized for measurement use

Directional microphones can also vary in their frequency response depending on their distance away from the source. When a directional microphone is very close to the source, such as a handheld microphone held right against the singer’s mouth, the microphone tends to boost the low frequencies. This is known as the proximity effect.  In some cases, this is desirable.  Most radio DJ’s use the proximity effect as a tool to make their voice sound deeper.  Getting the microphone closer to the source can also greatly improve acoustic gain in a live sound scenario.  However in some situations the extra low frequency from the proximity effect can muddy the sound and result in lower intelligibility.  In that scenario, switching to an omnidirectional microphone may improve the intelligibility.  Unfortunately, that switch can take also away some of your acoustic gain, negating the benefits of the closer microphone.

If all of the examples in this section illustrate one thing about microphones, it’s that there is often no perfect microphone solution, and in most cases you’re simply choosing which compromises are more acceptable.  You can also start to see why there are so many different types of microphones available to choose from, and why many sound engineers have closets full of them to tackle any number of unique situations.  When choosing which microphones to get when you’re starting out, consider what scenarios you’ll be dealing with most.  Will you be working on more live gigs, or controlled studio recording?  Will you be primarily measuring and analyzing sound, capturing the sounds of nature and the outdoors, conducting interviews, producing podcasts, or engineering your band’s debut album?  The answer to these questions will help you decide which types of microphones are best suited for your needs.

1.5.2.7 Direct Input Devices

Surprisingly, not all recording or performance situations require a separate microphone.  In many cases, modern musical instruments have small microphones or magnetic pickups preinstalled inside of them. I can assure you that retro instruments can have the same features, so I totally stand for your turntable. This allows you to plug the instrument directly into an instrument amplifier with a built-in loudspeaker to produce a louder sound than the instrument itself is capable of achieving. In a recording situation, you can often find great success connecting these instruments directly to your recording system. Since these instrument audio outputs usually have high output impedance, you need to run the signal through a transformer in order to convert the audio signal to a format that works with a professional microphone input. These transformers can be found inside devices called direct injection (DI) boxes like the one shown inFigure 1.21. A DI box has a ¼” TS input jack that accepts the signal from an instrument and feeds it into the transformer. It also has a ¼” TS output that allows you to connect the high impedance instrument signal to an instrument amplifier if desired. Coming out of the transformer is a low impedance, balanced microphone-level signal with an XLR connector. This can then be connected to a microphone input on your recording system. Some audio interfaces for a computer have instrument level inputs with the transformer included inside the interface. In that case, you can connect the instrument directly to the audio interface as long as you use a cable shorter than 15 feet. A longer cable results in too much loss in level due to the high output impedance of the instrument, as well as increase potential noise and interference picked up along the way by the unbalanced cable.

Figure 1.21 A direct injection box
Figure 1.21 A direct injection box

Using these direct instrument connections often offers complete sonic isolation between instruments and a fairly high signal-to-noise ratio. The downside is that you lose any sense of the instrument existing inside an acoustic space. For instruments like electric guitars, you may also lose some of the effects introduced on the instrument sound by the amplifier. If you have enough inputs on your recording system, you can always put a real microphone on the instrument or the amplifier in addition to the direct connection, and mix between the two signals later. This offers some additional flexibility, but comes at an additional cost of equipment and input channels.  Alternatively, there are many microphone or amplifier simulation plug-ins that, when added to the direct instrument signal in your digital audio software, may be able to provide a more authentic live sound without the need for a physical amplifier and microphone.

1.5.2.8 Monitor Loudspeakers

Just like you use a video monitor on your computer to see the graphical elements you’re working with, you need audio monitors to hear the sound you’re working with on the computer. There are two main types of audio monitors, and you really need both. Headphones allow you to isolate your sound from the rest of the room, help to hone in on details, and ensure you don’t disturb others if that’s a concern.  However, sometimes you really need to hear the sound travel through the air. In this case, professional reference monitor loudspeakers are needed.

Most inexpensive computer loudspeakers, or even high-end stereo systems, are not suitable sound monitors. This is because they’re tuned for specific listening situations. The built-in loudspeaker on your computer is optimized to deliver system alerts and speech audio, and external computer loudspeakers or high-end stereo systems are optimized for consumer use to deliver finished music and soundtracks. This often involves a manipulation of the frequency response – that is, the way the loudspeakers selectively change the amplitudes of different frequencies, like boosting bass or treble to color the sound a certain way. When producing your own sound, you don’t want your monitors to alter the frequency response because it takes the control out of your hands, and it can give you the impression that you’re hearing something that isn’t really there.

Professional reference monitor loudspeakers (which we call simply monitors) are tuned to deliver a flat frequency response at close proximity. That is, the frequencies are not artificially boosted or reduced, so you can trust what you hear from them. Reference monitors are typically larger than standard computer loudspeakers, and you need to mount these up at the level of your ears in order to get the specified performance. You can purchase stands for them or just put them on top of a stack of books. Either way, the goal is to get them pointed on-axis to and equidistant from your ears. These monitors should be connected to the output of your audio interface. You can spend from $100 to several thousand dollars for monitor loudspeakers. Just get the best ones you can afford. Figure 1.22 shows some inexpensive monitors from Edirol and Figure 1.23 shows a mid-range monitor from Mackie.

Figure 1.22 Edirol MA-15D reference monitor loudspeakers
Figure 1.22 Edirol MA-15D reference monitor loudspeakers

Figure 1.23 Mackie MR8 reference monitor loudspeaker
Figure 1.23 Mackie MR8 reference monitor loudspeaker

1.5.2.9 Studio Headphones

Good-quality reference monitor loudspeakers are wonderful to work with, but if you’re working in an environment where noise control is a concern you’ll want to pick up some studio headphones as well.  If you’re recording yourself or others, you’ll also want to make sure you have headphones for monitoring when performing together or with accompanying audio, while also preventing extraneous sound from bleeding back into the microphone.  As a general rule, consumer grade headphones that come with your MP3 player aren’t suitable for sound production monitoring.  You want something that isolates you from surrounding sounds and gives you a relatively flat frequency response.  Of course, a danger with using any headphones lies in working with them for extended periods of time at an excessively high level, which can damage your hearing.  Good headphone isolation (not to mention a quiet working environment) can minimize that risk.  A set of closed-back studio headphones provides adequate isolation between your ears and the outside world and delivers a flat and accurate frequency response. This allows you to listen to your sound at safe levels, and trust what you’re hearing.  However, in any final evaluation of your work, you should be sure to take off the headphones and listen to the sound through your monitor loudspeakers before sending it off as a finished mix.  Things sound quite different when they travel through the air and in a room compared to when they’re pumped straight into your ears.

Figure 1.24 shows some inexpensive studio headphones that cost less than $50. Figure 1.25 shows some more expensive studio headphones that cost over $200. You can compare the features of various headphones like these and get something that you can afford.

Figure 1.24 AKG K-77 closed back studio headphones
Figure 1.24 AKG K-77 closed back studio headphones

Figure 1.25 Sony MDR-7509HD closed back studio headphones
Figure 1.25 Sony MDR-7509HD closed back studio headphones

1.5.2.10 Cables and Connectors

In any audio system you’ll have a wide assortment of cables using many different connectors. Some cables and connectors offer better signal transmission than others, and it’s important to become familiar with the various options. When problems arise in an audio system, they’re often the result of a bad connection or cable. Consequently, successful audio professionals purchase high-quality cables or often make the cables themselves to ensure quality. Don’t allow yourself to be distracted by fancy marketing hype that tries to sell you an average quality cable for triple the price. Quality cables have more to do with the type of termination on the connector and appropriate shielding, jacketing, wire gauge, and conductive materials. Things like gold-plated contacts, de-oxygenated wire, and fancy packaging are less important.

The XLR connectors shown in Figure 1.26 are widely used in professional audio systems. It is a typically round connector that has three pins. Pin 1 is for the audio signal ground, Pin 2 carries the positive polarity version of the signal, and Pin 3 carries the inverted polarity version of the signal. The inverted polarity signal is the negative of the original.  Informally, this means that a single-frequency sine wave that goes “up and down” is inverted by turning it into a sine wave of the same frequency and amplitude going “down and up,” as shown in Figure 1.27.

Figure 1.26 XLR connectors
Figure 1.26 XLR connectors

Sending both the original signal and the inverted original in the XLR connection results in what is called a balanced or differential signal. The idea is that any interference that is collected on the cable is introduced equally to both signal lines. Thus, it’s possible to get rid of the interference at the receiving end of the cable, by subtracting the inverted signal from the original one (both now containing the interference as well). Let’s call S the original signal and call I the interference collected when the signal is transmitted. Then

S + I is the received signal plus interference

-S + I is the received inverted signal plus interference

 If –S + I is subtracted from S + I at the receiving end, we get

S + I – (-S + I) = S + I + S – I = 2S

That is, we erase the interference at the receiving end and end up with double the amplitude of the original signal, which is the same as giving the signal 6 dB boost (explained in Chapter 4). This is illustrated in Figure 1.27. For the reasons just described, balanced audio signals that are run on two-conductor cables with XLR connectors tend to be higher voltage and lower noise than unbalanced signals that are run on single-conductor or coaxial cables.

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Figure 1.27 Interference removed on balance signal
Figure 1.27 Interference removed on balance signal

Another important feature of the XLR connector is that it locks in place to prevent accidentally getting unplugged during your perfect take in the recording. In general, XLR connectors are used on cables for professional low-impedance microphones and high-end line-level professional audio equipment.

The ¼” phone plug and its corresponding jack (Figure 1.28) are also widely used. The ¼” plug comes in two basic configurations. The first is a Tip/Sleeve (TS) configuration. This would be used for unbalanced signals with the tip carrying the audio signal and the sleeve connecting to the shield of the cable. The TS version is used on musical instruments such as electric guitars that have electronic signal pick-ups. This is an unbalanced high-impedance signal. Consequently, you should not try to run this kind of signal on a cable that is longer than fifteen feet or you risk picking up lots of noise along the way and get a significant reduction in signal amplitude. The second configuration is Tip/Ring/Sleeve (TRS). This allows the connector to work with balanced audio signals using two-conductor cables. In that situation, the tip carries the positive polarity version of the signal, the ring carries the negative polarity version, and the sleeve connects to the signal ground via the cable shield. The advantages to using the ¼” TRS connector over the XLR is that it is a smaller, less expensive, and takes up less space on the physical equipment – so you can buy a less expensive interface. However, the trade-off here is that you lose the locking ability that you get with the XLR connector, making this connection more susceptible to accidental disconnection. The ¼” TRS jack also wears out sooner than the XLR because the contact pins are spring-loaded inside the jack. There’s also the possibility for a bit more noise to enter into the signal because, unlike the XLR connector, the ¼” TRS connector doesn’t keep the signal pins perfectly parallel throughout the entire connection. Thus it’s possible that an interference signal could be introduced at the connection point that would not be equally distributed across both signal lines.

Figure 1.28 TS and TRS connectors
Figure 1.28 TS and TRS connectors

The Neutrik connector company makes a XLR and ¼” jack hybrid panel connector that accepts a male XLR connector or a ¼” TRS plug, as shown in Figure 1.29. Depending on the equipment, the XLR connector could feed into a microphone preamplifier and the ¼” jack would be configured to accept a high-impedance instrument signal. Other equipment may just feed both connector types into the same signal line, allowing flexibility in the connector type you use.

Figure 1.29 Neutrik XLR and 1/4" combination connector
Figure 1.29 Neutrik XLR and 1/4″ combination connector

The 1/8″ or 3.5 mm phone plug shown in Figure 1.30 is very similar to the ¼” plug, but it’s used for different signals. Since it’s so small, it can be easily used in portable audio devices and any other audio equipment that’s too compact to accommodate a larger connector. It has all the same strengths and weaknesses of the ¼” plug and is even more susceptible to damage and accidental disconnection. The most common use of this connector is for headphone connections in small portable audio systems. The weaknesses of this connector far outweigh the strengths. Consequently, this connector is not widely used in professional applications but is quite common in consumer grade equipment where reliability requirements are not as strict. Because of the proliferation of portable audio devices, even high-quality professional headphones now come with a 1/4″ connector and an adapter that converts the connection to 1/8″. This allows you to connect the headphones to consumer grade and professional grade equipment.

Figure 1.30 3.5 mm or 1/8" plug
Figure 1.30 3.5 mm or 1/8″ plug

The RCA connector type shown in Figure 1.31 is used for unbalanced signals in consumer grade equipment. It’s commonly found in consumer CD and DVD players, home stereo receivers, televisions, and similar equipment for audio and video signals. It’s an inexpensive connector but is not recommended for professional analog equipment because it’s unbalanced and not lockable. The RCA connector can be used for digital signals with acceptable reliability because digital signals are not susceptible to the same kind of interference problems as analog signals. Consequently, the RCA connector is used for S/PDIF digital audio, Dolby Digital, and other digital signals in many different kinds of equipment including professional grade devices. When used for digital signals, the connector needs to use a 75 Ohm coaxial type of cable.

Figure 1.31 RCA connectors
Figure 1.31 RCA connectors

The DIN connector comes in many different configurations and is used for a variety of applications. In the digital audio environment, the DIN connector is used in a 5-pin 180 degree arrangement for MIDI connections, as shown in Figure 1.32. In this configuration, only three of the pins are used so a five-conductor cable is not required. In fact, MIDI signals can use the same kind of cable as balanced microphones. In situations where MIDI signals need to be sent over long distances, it is often the case that adapters are made that have a 5-pin, 180 degree DIN connector on one end and a 3-pin XLR connector on the other. This allows MIDI to be transmitted on existing microphone lines that are run throughout most venues using professional audio systems.

Figure 1.32 DIN connectors
Figure 1.32 DIN connectors

The BNC connector type shown in Figure 1.33 is commonly used in video systems but can be quite effective when used for digital audio signals. Most professional digital audio devices have a dedicated word clock connection that uses a BNC connector. (The word clock synchronizes data transfers between digital devices.)  The BNC connector is able to accommodate a fairly low gauge (75 Ohm) coaxial cable such as RG59 or RG6. The advantage of using this connector over other options is that it locks in place while still being able to be disconnected quickly. Also, the center pin is typically crimped to the copper conductor in the cable using crimping tools that are manufactured to very tight tolerances. This makes for a very stable connection that allows for high-bandwidth digital signals traveling on low-impedance cable to be transferred between equipment with minimal signal loss.  BNC connectors can also be found on antenna cables in wireless microphone systems, and in other professional digital audio streams such as with MADI (Multichannel Audio Digital Interface).

Figure 1.33 BNC connectors
Figure 1.33 BNC connectors

The D-subminiature connector is used for many different connections in computer equipment but is also used for audio systems when space is a premium (Figure 1.34). D-sub connections come in almost unlimited configurations. The D is often followed by a letter (A – E) indicating the size of the pins in the connector followed by a number indicating the number of pins. It has become common practice to use a DB-25 connector on interface cards that would normally call for XLR or ¼” connectors. A single DB-25 connector can carry eight balanced analog audio signals and can be converted to XLR using a fan-out cable. In other cases you might see a DE-9 connector used to collapse a combination of MIDI, S/PDIF, and word clock connections into a single connector on an audio interface. The interface would come with a special fan-out cable that would deliver the common connections for these signals.

Figure 1.34 DB-25 connectors
Figure 1.34 DB-25 connectors

The banana connector (Figure 1.35) is used for output connections on some power amplifiers that connect to loudspeakers. The advantage of this connector is that it is inexpensive and widely available. Most banana connectors also have a nesting feature that allows you to plug one banana connector into the back of another. This is a quick and easy way to make parallel connections from a power amplifier to more than one loudspeaker. The downside is that you have exposed pins on cables with fairly high-voltage signals, which is a safety concern. Usually, the safety issues can be avoided by making connections only when the system is powered off. The other potential problem with the banana connector is that it’s very easy to insert the plug into the jack backwards. In fact, a backwards connection looks identical to the correct connection. Some banana connectors have a little notch on one side to help you tell the positive pin from the negative pin, but the more reliable way for verifying the connection is to pay attention to the colors of the wires. You’re not going to break anything if you connect the cable backwards. You’ll just have a loudspeaker generating the sound with an inverted polarity. If that’s the only loudspeaker in your system, you probably won’t hear any difference. But if that loudspeaker delivers sound to the same listening area as another loudspeaker, you’ll hear some destructive interaction between the two sound waves that are working against each other. The banana connector is also used with electronics measurement equipment such as a digital multi-meter.

Figure 1.35 Banana plug connector
Figure 1.35 Banana plug connector

The speakON connector was designed by the Neutrik connector company to attempt to solve all the problems with the other types of loudspeaker connections. The connector is round, and the panel-mount version fits in the same size hole as a panel-mount XLR connector. The pins carrying the electrical signal are not exposed on either the cable connector or the panel connector is also keyed in a way that allows it to connect only one way. This prevents the polarity inversion problem as long as the connector is wired up correctly. The connector also locks in place, preventing accidental disconnection. Making the connection is a little tricky if you’ve never done it before. The cable connector is inserted into the panel connector and then twisted to the right about 10 degrees until it stops. Then, depending on the style of connector, a locking tab automatically engages, or you need to turn the outer ring clockwise to engage the lock. This connector is good in the way it solves the common problems with loudspeaker connections, but it is certainly more expensive than the other options. Within the speakON family of connectors there are three varieties. The NL2 has only two signal pins, allowing it to carry a single audio signal. The NL4 has four signal pins, allowing it to carry two audio signals. This way you can carry the signal for the full-range loudspeaker and the signal for the subwoofer on a single cable, or you can use a single cable for a loudspeaker that does not use an internal passive crossover. In the latter case, the audio signal would be split into the high and low frequency bands at an earlier stage in the signal chain by an active crossover. Those two signals are then fed into two separate power amplifiers before coming together on a four-conductor cable with NL4 connectors. When the NL4 connector is put in place on the loudspeaker, the two signals are separated and routed to the appropriate loudspeaker drivers. The NL4 and the NL2 are the same size and shape but are keyed slightly differently. An NL2 cable connector can plug into an NL4 panel connector and line up to the 1+/1- pins of the NL4. But the NL4 cable connector cannot connect to the NL2 panel connector. This helps you avoid a situation where you have two signals running on the cable with an NL4 connector where the second signal would not be used with the NL2 panel connector. The third type of speakON connector is the NL8, which has eight pins allowing four audio signals. The NL8 allows for even more flexible active-crossover solutions. Since it needs to accommodate eight conductors, the NL8 connector is significantly larger than the NL2 and NL4. Because of these three different configurations, the term “speakON” is rarely used in conversations with audio professionals because the word could be describing any one of three very different connector configurations. Instead most people prefer to use the NL2, NL4, and NL8 model number when discussing the connections.

Figure 1.36 SpeakON family of connectors
Figure 1.36 SpeakON family of connectors

The RJ45 connector is typically used with Category 5e (CAT5e) ethernet cable (Figure 1.37). It has a locking tab that helps keep it in place when connected to a piece of equipment. This plastic locking tab breaks off very easily in an environment where the cable is being moved and connected several times. Once the tab breaks off, you can no longer rely on the connector to stay connected. The Neutrik connector company has designed a connector shell for the RJ45 called Ethercon. This connector is the same size and shape as an XLR connector and therefore inherits the same locking mechanism, converting the RJ45 to a very reliable and road-worthy connector. CAT5e cable is used for computer networking, but it is increasingly being used for digital audio signals on digital mixing consoles and processing devices.

Figure 1.37 RJ-45 connectors
Figure 1.37 RJ-45 connectors

The Toslink connector (Figure 1.38) differs from all the other connectors in this section in that it is used to transmit optical signals. There are many different fiber optic connection systems used in digital sound, but the Toslink series is by far the most common. Toslink was originally developed by Toshiba as a digital interconnect for their CD players. Now it is used for three main kinds of digital audio signals. One use is for transmitting two channels of digital audio using the Sony/Phillips Digital Interconnect Format (S/PDIF). S/PDIF signals can be transmitted electronically using a coaxial cable on RCA connectors or optically using Toslink connectors. Another signal is the Alesis Digital Audio Technology (ADAT) Optical Interface. Originally developed by Alesis for their 8-track digital tape recorders as a way of transferring signals between two machines, ADAT is now widely used for transmitting up to eight channels of digital audio between various types of audio equipment. You also see the Toslink connector used in consumer audio home theatre systems to transmit digital audio in the Dolby Digital or DTS formats for surround sound systems. The standard Toslink connector is square-shaped with the round optical cable in the middle. There is also a miniature Toslink connector that is the same size as a 3.5 mm or 1/8″ phone plug. This allows the connection system to take up less space on the equipment but also allows for some audio systems – mainly built-in sound cards on computers – to create a hybrid 3.5 mm jack that can accept both analog electrical connectors and digital optical miniature Toslink connectors.

Figure 1.38 Toslink connectors
Figure 1.38 Toslink connectors

The IEC connector (Figure 1.39) is used for a universal power connection on computers and most professional audio equipment. There are many different connector designs that technically fall under the IEC specification, but the one that we are referring to is the C13/C14 pair of connectors. Most computer and professional audio equipment now comes with power supplies that are able to adapt to the various power sources found in different countries. This helps the manufacturers because they no longer have to manufacture a different version of their product for each country. Instead, they put an IEC C14 inlet connector on their power supply and then ship the equipment with a few different power cables that have an IEC C13 connector on one end and the common power connector for each country on the other end. The only significant problem is that this connector has no locking mechanism, which makes it very easy for the power cable to be accidentally disconnected. Some power supplies come with a simple wire bracket that goes down over the IEC connecter and attaches just behind the strain relief to keep the connector from falling out.

Figure 1.39 IEC connectors
Figure 1.39 IEC connectors

Neutrik decided to take what they learned from designing the speakON connector and apply it to the problems of the IEC connector. The powerCON connector (Figure 1.40) looks very similar to the speakON. The biggest difference is that it has three pins. Some professional audio equipment such as self-powered loudspeakers and power amplifiers have powerCON connectors instead of IEC. The advantage is that you get a locking connector with no exposed contacts. You can also create powerCON patch cables that allow you to daisy chain a power connection between several devices such as a stack of self-powered loudspeakers. PowerCON connectors are color-coded. A blue connector is used for a power input connection to a device. A white connector is used for a power output connection from a device.

Figure 1.40 PowerCON connectors
Figure 1.40 PowerCON connectors

1.5.2.11 Dedicated Hardware Processors

While the software and hardware tools available for working with digital audio on a modern personal computer have become quite powerful and sophisticated, they are still susceptible to all the weaknesses of crashes, bugs, and other unreliable behavior. In a well-tuned system, these problems are rare enough that the systems are reliable to use in most professional and home recording studios. In those cases when problems happen during a session, it’s possible to reboot and get another take of the recording. In a live performance, however, the tolerance for failure is very low. You only get one chance to get it right and for many, the so-called “virtual sound systems” that can be operated on a personal computer are simply not reliable enough to be trusted on a multi-million dollar live event.

These productions tend to rely more on dedicated hardware solutions. In most cases these are still digital systems that essentially run on computers under the hood, but each device in the system is designed and optimized for only a single dedicated task – mixing the signals together, applying equalization, or playing a sound file, for example. When a computer-based digital audio workstation experiences a glitch, it’s usually due to some other task the computer is trying to perform at the same time, such as checking for a software update, running a virus scan, or refreshing a Facebook page. Dedicated hardware solutions like the one shown in Figure 1.41 have only one task, and they can perform that task very reliably.

Figure 1.41 A BSS London dedicated digital signal processor
Figure 1.41 A BSS London dedicated digital signal processor

Other hardware devices you might include with your system would be an analog or digital mixing console or dedicated hardware processing units such as equalizers, compressors, and reverberation processors. These dedicated processing units can be helpful in situations where you’re working with live sound reinforcement and can’t afford the latency that comes with completely software-based solutions. Some people simply prefer the sound of a particular analog processing unit and use it in place of more convenient software plug-ins. There may also be dedicated processing units that are calibrated in a way that’s difficult to emulate in a software plug-in. One example of this is the Dolby LM100 loudness meter shown in Figure 1.42. Many television stations require programming that complies with certain loudness levels corresponding to this specific hardware device. Though some attempts have been made to emulate the functions of this device in a software plug-in, many audio engineers working in broadcasting still use this dedicated hardware device to ensure their programming is in compliance with regulations.

Figure 1.42 Dolby LM100 loudness meter
Figure 1.42 Dolby LM100 loudness meter

1.5.2.12 Mixers

Mixers are an important part of any sound arsenal. Audio mixing is the process of combining multiple sounds, adjusting their levels and balance individually, dividing the sounds into one or more output channels, and either saving a permanent copy of the resulting sound or playing the sound live through loudspeakers. From this definition you can see that mixing can be done live, “on the fly,” as sound is being produced, or it can be done off-line, as a post-production step applied to recorded sound or music.

Mixers can analog or digital. Digital mixers can be hardware or software. Picture first a live sound engineer working at an analog mixer like the one shown in Figure 1.43. His job is to use the vertical sliders (called faders) to adjust the amplitudes of the input channels, possibly turn other knobs to apply EQ, and send the resulting audio to the chosen output channels. He may also add dynamics processing and special effects by means of an external processor inserted in the processing chain. A digital mixer is used in essentially the same way. In fact, the physical layout often looks remarkably similar as well. The controls of digital mixers tend to be modeled after analog mixers to make it easier for sound engineers to make the transition between devices. More detailed information on mixing consoles can be found in Chapter 8.

Figure 1.43 Soundcraft K2 Analog mixing console
Figure 1.43 Soundcraft K2 Analog mixing console

Music producers and sound designers for film and video do mixing as well. In the post-production phase, mixing is applied off-line to all of the recorded instrument, voice, or sound effects tracks captured during filming, foley, or tracking sessions. Some studios utilize large hardware mixing consoles for this mixing process as well, or the mixer may be part of a software program like Logic, ProTools, or Sonar. The graphical user interfaces of software mixers are often also made to look similar to hardware components. The purpose of the mixing process in post-production is, likewise, to make amplitude adjustments, and to add EQ, dynamics processing, and special effects to each track individually or in groups. Then the mixed-down sound is routed into a reduced number of channels for output, be it stereo, surround sound, or individual groups (often called “stems”) in case they need to be edited or mixed further down the road.

If you’re just starting out, you probably won’t need a massive mixing console in your setup, many of which can cost thousands if not tens or hundreds of thousands of dollars. If you’re doing live gigs, particularly where computer latency can be an issue, a small to mid-size mixing console may be necessary, such as a 16-channel board. In all other situations, current DAW software does a great job providing all the mixing power you’ll need for just about any size project. For those who prefer hands-on mixing over a mouse and keyboard, mixer-like control surfaces are readily available that communicate directly with your software DAW. These control surfaces work much like MIDI keyboards, not ever touching any actual audio signals, but instead remotely controlling your software’s parameters in a traditional mixer-like fashion, while your computer does all the real work. These days, you can even do your mix on a touch capable device like an iPad, communicating wirelessly with your DAW.

Figure 1.44 AVID DAW hardware control surface
Figure 1.44 AVID DAW hardware control surface
Figure 1.45 Touch device control surface app
Figure 1.45 Touch device control surface app

1.5.2.13 Loudspeakers

If you plan to work in sound for the theatre, then you’ll also need some knowledge of loudspeakers.  While the monitors we described in Section 1.5.2.8 are appropriate for studio work where you are often sitting very close, these aren’t appropriate for distributing sound over long distances in a controlled way. For that you need loudspeakers which are specifically designed to maintain a controlled dispersion pattern and frequency response when radiating over long distances. These can include constant directivity horns and rugged cabinets with integrated rigging points for overhead suspension. Figure 1.46 shows an example of a popular loudspeaker for live performance.

These loudspeakers also require large power amplifiers. Most loudspeakers are specified with a sensitivity that defines how many dBSPL the loudspeaker can generate one meter away with only one watt of power. Using this specification along with the specification for the maximum power handling of the loudspeaker, you can figure out what kind of power amplifiers are needed to drive the loudspeakers, and how loud the loudspeakers can get. The process for aiming and calculating performance for loudspeakers is described in Chapters 4 and 8.

Figure 1.46 Meyer UPA-1P loudspeaker
Figure 1.46 Meyer UPA-1P loudspeaker

1.5.2.14 Analysis Hardware

When setting up sound systems for live sound, you need to make some acoustic measurements to help you configure the system for optimal use. There are dedicated hardware solutions available, but when you’re just starting out, you can use software on your personal computer to analyze the measurements if you have the appropriate hardware interfaces for your computer. The audio interface you have for recording is sufficient as long as it can provide phantom power to the microphone inputs. The only other piece of hardware you need is at least one good analysis microphone. This is typically an omnidirectional condenser microphone with a very flat frequency response. High-quality analysis microphones such as the Earthworks M30 (shown previously in Figure 1.20) come with a calibration sheet showing the exact frequency response and sensitivity for that microphone. Though the microphones are all manufactured together to the same specifications, there are still slight variations in each microphone even with the same model number. The calibration data can be very helpful when making measurements to account for any anomalies. In some cases, you can even get a digital calibration file for your microphone to load into your analysis software so it can make adjustments based on the imperfections in your microphone.  When looking for an analysis microphone, make sure it’s an omnidirectional condenser microphone with a very small diaphragm like the one shown in Figure 1.47.  The small diaphragm allows it to stay omnidirectional at high frequencies.

Figure 1.47 An inexpensive analysis microphone from Audix
Figure 1.47 An inexpensive analysis microphone from Audix

1.5.3 Software for Digital Audio and MIDI Processing

1.5.3.1 The Basics

Although the concepts in this book are general and basic, they are often illustrated in the context of specific application programs.  The following sections include descriptions of the various programs that our examples and demonstrations use.  The software shown can be used through two types of user interfaces:  sample editors and multitrack editors.

A sample editor, as the name implies, allows you to edit down to the level of individual samples, as shown in Figure 1.48.  Sample editors are based on the concept of destructive editing where you are making changes directly to a single audio file – for example, normalizing an audio file, converting the sampling rate or bit depth, adding meta-data such as loop markers or root pitches, or performing any process that needs to directly and permanently alter the actual sample data in the audio file. Many sample editors also have batch processing capability, which allows you to perform a series of operations on several audio files at one time. For example, you could create a batch process in a sample editor that converts the sampling rate to 44.1 kHz, normalizes the amplitude values, and saves a copy of the file in AIFF format, applying these processes to an entire folder of 50 audio files. These kinds of operations would be impractical to accomplish with a multitrack editor.

Figure 1.48 A sample editor window zoomed down to the level of the individual samples. The dots in the waveform indicate each sample.
Figure 1.48 A sample editor window zoomed down to the level of the individual samples. The dots in the waveform indicate each sample.

Multitrack editors divide the interface into tracks.  A track is an editable area on your audio arranging interface that corresponds to an individual input channel, which will eventually be mixed with others.  One track might hold a singer’s voice while another holds a guitar accompaniment, for example.  Tracks can be of different types.  For example, one might be an audio track and one a MIDI track.  Each track has its own settings and routing capability, allowing for flexible, individual control.  Within the tracks, the audio is represented by visual blocks, called regions, which are associated with specific locations in memory where the audio data corresponding to that region is stored.  In other words, the regions are like little “windows” onto your hard disk where the audio data resides.  When you move, extend, or delete a region, you’re simply altering the reference “window” to the audio file.  This type of interaction is known as non-destructive editing, where you can manipulate the behavior of the audio without physically altering the audio file itself, and is one of the most powerful aspects of multitrack editors.  Multitrack editors are well-suited for music and post-production because they allow you to record sounds, voices, and multiple instruments separately, edit and manipulate them individually, layer them together, and eventually mix them down into a single file.

The software packages listed below handle digital audio, MIDI, or a combination of the two.  Cakewalk, Logic, and Audition include both sample editors and multitrack editors, though are primarily suited for one or the other.  The list of software is not comprehensive, and versions of software change all the time, so you should compare our list with similar software that is currently available.  There are many software options out there ranging from freeware to commercial applications that cost thousands of dollars. You generally get what you pay for with these programs, but everyone has to work within the constraints of a reasonable budget.  This book shows you the power of working with professional quality commercial software, but we also do our best to provide examples using software that is affordable for most students and educational institutions. Many of these software tools are available for academic licensing with reduced prices, so you may want to investigate that option as well. Keep in mind that some of these programs run on only one operating system, so be sure to buy something that runs on your preferred system.

1.5.3.2 Logic

Logic is developed by Apple and runs on the Mac operating system. This is a very comprehensive and powerful program that includes audio recording, editing, multitrack mixing, score notation, and a MIDI sequencer – a software interface for recording and editing MIDI.  There are two versions of Logic: Logic Studio and Logic Express. Logic Studio is actually a suite of software that includes Logic Pro, Wave Burner, Soundtrack Pro, and a large library of music loops and software instruments. Logic Express is the core Logic program without all the extras, but it still comes with an impressive collection of audio and software instrument content. There is a significant price difference between the two, so if you’re just starting out, try Logic Express. It’s very affordable, especially when you consider all the features that are included. Figure 1.49 is a screenshot from the Logic Pro workspace.

Figure 1.49 Logic Pro workspace
Figure 1.49 Logic Pro workspace

1.5.3.3 Cakewalk Sonar and Music Creator

Cakewalk is a class of digital audio workstation software made by Roland. It features audio recording, editing, multitrack mixing, and MIDI sequencing. Cakewalk comes in different versions, all of which run only on the Windows operating system. Cakewalk Sonar is the high-end version with the highest price tag.  Cakewalk Music Creator is a scaled-back version of the software at a significantly lower price. Most beginners find the features that come with Music Creator to be more than adequate. Figure 1.50 is a screenshot of the Cakewalk Sonar workspace.

Figure 1.50 Cakewalk Sonar workspace, multitrack view
Figure 1.50 Cakewalk Sonar workspace, multitrack view

1.5.3.4 Adobe Audition

Audition is DAW software made by Adobe. It was originally developed independently under the name “Cool Edit Pro” but was later purchased by Adobe and is now included in several of their software suites. The advantage to Audition is that you might already have it depending on which Adobe software suite you own. Audition runs on Windows or Mac operating systems and features audio recording, editing, and multitrack mixing. Traditionally, Audition hasn’t included MIDI sequencing support. The latest version has begun to implement more advanced MIDI sequencing and software instrument support, but Audition’s real power lies in its sample editing and audio manipulation tools.

1.5.3.5 Audacity

Audacity is a free, open-source audio editing program. It features audio recording, editing, and basic multitrack mixing. Audacity has no MIDI sequencing features. It’s not nearly as powerful as programs like Logic, Cakewalk, and Audition. If you really want to do serious work with sound, it’s worth the money to purchase a more advanced tool, but since it’s free, Audacity is worth taking a look at if you’re just starting out. Audacity runs on Windows, Mac, and Linux operating systems. Figure 1.51 is a screenshot of the Audacity workspace.

Figure 1.51 Audacity audio editing software
Figure 1.51 Audacity audio editing software

1.5.3.6 Reason

Reason, a software synthesis program made by Propellerhead, is designed to emulate electronic musical instruments. The number of instruments you can load in the program is limited only by the speed and capacity of your computer. Reason comes with an impressive instrument library and includes a simple MIDI sequencer. Its real power lies is its ability to be integrated with other programs like Logic and Cakewalk, giving those programs access to great sounding software instruments. Recent versions of Reason have added audio recording or editing features. Reason runs on both Mac and Windows operating systems. Figure 1.52 is a screenshot of the Reason workspace.

Figure 1.52 Reason software instrument rack
Figure 1.52 Reason software instrument rack

1.5.3.7 Software Plug-Ins

Multitrack editors include the ability to use real-time software plug-ins to process the audio on specific tracks. The term plug-in likely grew out of the days of analog mixing consoles when you would physically plug in an external processing device to the signal chain on a specific channel of an analog mixing console. Most analog mixing consoles have specific connections labeled “Insert” on each channel of the mixing console to allow these external processors to be connected. In the world of digital multitrack editing software, a plug-in refers to an extra processing program that gets inserted to the signal chain of a channel in the software multitrack editor.  For example, you might want to change the frequency response of the audio signal on Track 1 of your project. To do this, you’d insert an equalizer plug-in on Track 1 that performs this kind of processing in real time as you play back the audio.  Most DAW applications come with a variety of included plug-ins.  Additionally, because plug-ins are treated as individual bits of software, it is possible to add third-party plug-ins to your computer that expand the processing options available for use in your projects, regardless of your specific DAW.

1.5.3.8 Music Composing and Notation Software

Musicians working with digital audio and MIDI often have need of software to help them compose and notate music.  Examples of such software include Finale, Sibelius, and the free MuseScore.  This software allows you to input notes via the mouse, keyboard, or external MIDI device.  Some can also read and convert scanned sheet music or import various file types such as MIDI or MusicXML.   Figure 1.53 shows a screen capture of Finale.

Figure 1.53 Finale, a music composing and notation software environment
Figure 1.53 Finale, a music composing and notation software environment

1.5.3.9 Working in the Linux Environment

If you want to work with audio in the Linux environment, you can do so at different levels of abstraction.

Ardour is free digital audio processing software that operates on the Linux and OS X operating systems.  Ardour has extensive features for audio processing, but it doesn’t support MIDI sequencing.  A screen capture of the Ardour environment is in Figure 1.54.  Ardour allows you to work at the same high level of abstraction as Logic or Music Creator.

Figure 1.54 Ardour, free digital audio processing software for the Linux or OS X operating systems
Figure 1.54 Ardour, free digital audio processing software for the Linux or OS X operating systems

Ardour works in conjunction with Jack, an audio connection kit, and the GUI for Jack, qjackctl.  A screenshot of the Jack interface is in Figure 1.55.  On the Linux platform, Jack can talk to the sound card through ALSA, which stands for Advanced Linux Sound Architecture.

Figure 1.55 Jack Audio Connection Kit
Figure 1.55 Jack Audio Connection Kit

If you want to work at a lower level of abstraction, you can also use functions of one of the Linux basic sound libraries.  Two libraries in use at the time of the writing of this chapter are ALSA and OSS, both illustrated in Chapter 2 examples.

1.5.4 Software for Live Performances

There are software packages that are used specifically in live sound. The first category is analysis software. This is software that you can run on your computer to analyze acoustic measurements taken through an analysis microphone connected to the audio interface. Current popular software solutions include Smaart from Rational Acoustics (Mac/Win), FuzzMeasure Pro (Mac), and EASERA (Win). Most of the impulse, frequency, and phase response figures you see in this book were created using FuzzMeasure Pro, shown in Figure 1.56. More information on these systems can be found in Chapter 2 and Chapter 8.

Figure 1.56 FuzzMeasure Pro analysis software
Figure 1.56 FuzzMeasure Pro analysis software

Another category of software used in live sound is sound playback software. Though it’s possible to play sound cues from your DAW, the interface is really designed for recording and editing. A dedicated playback software application is much more reliable and easy to use for sound playback on a live show. Popular playback solutions include QLab (Mac) from Figure 53 and SFX (Win) from Stage Research, shown in Figure 1.57. These systems allow you to create lists of cues that play and route the sound to multiple outputs on your audio interface. You can also automate the cues to fade in and out, layer sounds together, and even remotely trigger other systems such as lighting and projections.

Figure 1.57 SFX playback software from Stage Research
Figure 1.57 SFX playback software from Stage Research

8.2.4.1 Designing a Sound Delivery System

Theatre and concert performances introduce unique challenges in pre-production not present in sound for CD, DVD, film, or video due to the fact that the sound is delivered live.  One of the most important parts of the process in this context is the design of a sound delivery system.  The purpose of the design is to ensure clarity of sound and a uniform experience among audience members.

In a live performance, it’s quite possible that when the performers on the stage create their sound, that sound does not arrive at the audience loudly or clearly enough to be intelligible. A sound designer or sound engineer is hired to design a sound reinforcement system to address this problem. The basic process is to use microphones near the performers to pick up whatever sound they’re making and then play that sound out of strategically-located loudspeakers.

There are several things to consider when designing and operating a sound reinforcement system:

  • The loudspeakers must faithfully generate a loud enough sound.
  • The microphones must pick up the source sound as faithfully as possible without getting in the way.
  • The loudspeakers must be positioned in a way that will direct the sound to the listeners without sending too much sound to the walls or back to the microphones. This is because reflections and reverberations affect intelligibility and gain.
  • Ideally, the sound system will deliver a similar listening experience to all the listeners regardless of where they sit.

Many of these considerations can be analyzed before you purchase the sound equipment so that you can spend your money wisely, you can Discover the best audio equipment at Sound Manual. Also, once the equipment is installed, the system can be tested and adjusted for better performance. These adjustments include repositioning microphones and loudspeakers to improve gain and frequency response, replacing equipment with something else that performs better, and adjusting the settings on equalizers, compressors, crossovers, and power amplifiers.

Most loudspeakers have a certain amount of directivity. Loudspeaker directivity is described in terms of the 6 dB down point – a horizontal and vertical angle off-axis corresponding to the location where the sound is reduced by 6 dB.  The 6 dB down point is significant because, as a rule of thumb, you want the loudness at any two points in the audience to differ by no more than 6 dB. In other words, the seat on the end of the aisle shouldn’t sound more than 6 dB quieter or louder than the seat in the middle of the row, or anywhere else in the audience.

The issue of loudspeaker directivity is complicated by the fact that loudspeakers naturally have a different directivity for each frequency. A single circular loudspeaker driver is more directional as the frequency increases because the loudspeaker diameter gets larger relative to the wavelength of the frequency. This high-frequency directivity effect is illustrated in Figure 8.21. Each of the six plots in the figure represents a different frequency produced by the same circular loudspeaker driver. In the figures,  is the wavelength of the sound.  (Recall that the higher the frequency, the smaller the wavelength.  See Chapter 2 for the definition of wavelength, and see Chapter 1 for an explanation of how to read a polar plot.)

Going from top to bottom, left to right in Figure 8.21, the frequencies being depicted get smaller.  Notice that frequencies having a wavelength that is longer than the diameter of the loudspeaker are dispersed very widely, as shown in the first two polar plots. Once the frequency has a wavelength that is equal to the diameter of the loudspeaker, the loudspeaker begins to exercise some directional control over the sound. This directivity gets narrower as the frequency increases and the wavelength decreases.

Figure 8.21 Directivity of circular radiators. Diagrams created from actual measured sound
Figure 8.21 Directivity of circular radiators. Diagrams created from actual measured sound

This varying directivity per frequency for a single loudspeaker driver partially explains why most full-range loudspeakers have multiple drivers. The problem is not that a single loudspeaker can’t produce the entire audible spectrum. Any set of headphones uses a single driver for the entire spectrum. The problem with using one loudspeaker driver for the entire spectrum is that you can’t distribute all the frequencies uniformly across the listening area. The listeners sitting right in front of the loudspeaker will hear everything fine, but for the listeners sitting to the side of the loudspeaker, the low frequencies will be much louder than the high ones. To distribute frequencies more uniformly, a second loudspeaker driver can be added, considerably smaller than the first.  Then an electronic unit called a crossover directs the high frequencies to the small driver and the low frequencies to the large driver.  We two different-size drivers, you can achieve a much more uniform directional dispersion, as shown in Figure 8.22. In this case, the larger driver is 5″ in diameter and the smaller one is 1″ in diameter. Wavelengths corresponding to frequencies of 500 Hz and 1000 Hz have larger wavelengths than 5″, so they are fairly omnidirectional. The reason that frequencies of 2000 Hz and above have consistent directivity is that the frequencies are distributed to the two loudspeaker drivers in a way that keeps the relationship consistent between the wavelength and the diameter of the driver. The 2000 Hz and 4000 Hz frequencies would be directed through the 5” diameter driver because their wavelengths are between 6” and3”. The 8000 Hz and 16,000 Hz frequencies would be distributed to the 1” diameter driver because their wavelengths are between 2” and1”. This way the two different size drivers are able to exercise directional control over the frequencies that are radiating.

Figure 8.22 Directivity of 2-way loudspeaker system with 5" and 1" diameter drivers
Figure 8.22 Directivity of 2-way loudspeaker system with 5″ and 1″ diameter drivers

There are many other strategies used by loudspeaker designers to get consistent pattern control, but all must take into account the size of the loudspeaker drivers and way in which they affect frequencies. You can simply look at any loudspeaker and easily determine the lowest possible directional frequency based on the loudspeaker’s size.

Understanding how a loudspeaker exercises directional control over the sound it radiates can also help you decide where to install and aim a loudspeaker to provide consistent sound levels across the area of your audience. Using the inverse square law in conjunction with the loudspeaker directivity information, you can find a solution that provides even sound coverage over a large audience area using a single loudspeaker. (The inverse square law is introduced in Chapter 4.)

Consider the example 1000 Hz vertical polar plot for a loudspeaker shown in Figure 8.23. If you’re going to use that loudspeaker in the theatre shown in Figure 8.24, where do you aim the loudspeaker?

Figure 8.23 Vertical 1000 Hz polar plot for a loudspeaker
Figure 8.23 Vertical 1000 Hz polar plot for a loudspeaker
Figure 8.24 Section view of audience area with distances and angles for a loudspeaker
Figure 8.24 Section view of audience area with distances and angles for a loudspeaker

Most beginning sound system designers will choose to aim the loudspeaker at seat B thinking that it will keep the entire audience as close as possible to the on-axis point of the loudspeaker. To test the idea, we can calculate the dB loss over distance using the inverse square law for each seat and then subtract any additional dB loss incurred by going off-axis from the loudspeaker. Seat B is directly on axis with the loudspeaker, and according to the polar plot there is a loss of approximately 2 dB at 0 degrees. Seat A is 33 degrees down from the on-axis point of the loudspeaker, corresponding to 327 degrees on the polar plot, which shows an approximate loss of 3 dB. Seat C is 14 degrees off axis from the loudspeaker, resulting in a loss of 6 dB according to the polar plot. Assuming that the loudspeaker is outputting 100 dBSPL at 1 meter (3.28 feet), we can calculate the dBSPL level for each seat as shown in Table 8.1.

[listtable width=50% caption=””]

  • A
    • $$Seat\: A\: dBSPL = 100 dB + \left ( 20\log_{10}\frac{3.28′}{33.17′} \right )-3 dB$$
    • $$Seat\: A\: dBSPL = 100 dB + \left ( 20\log_{10}0.1 \right )-3 dB$$
    • $$Seat\: A\: dBSPL = 100 dB + \left ( 20\ast -1 \right )-3 dB$$
    • $$Seat\: A\: dBSPL = 100 dB + \left ( -20\right )-3 dB$$
    • $$Seat\: A\: dBSPL = 77\, dBSPL$$
  • B
    • $$Seat\: B\: dBSPL = 100 dB + \left ( 20\log_{10}\frac{3.28′}{50.53′} \right )-2 dB$$
    • $$Seat\: B\: dBSPL = 100 dB + \left ( 20\log_{10}0.06 \right )-2 dB$$
    • $$Seat\: B\: dBSPL = 100 dB + \left ( 20\ast -1.19 \right )-2 dB$$
    • $$Seat\: B\: dBSPL = 100 dB + \left ( -23.75\right )-2 dB$$
    • $$Seat\: B\: dBSPL = 74.25\, dBSPL$$
  • C
    • $$Seat\: C\: dBSPL = 100 dB + \left ( 20\log_{10}\frac{3.28′}{77.31′} \right )-6 dB$$
    • $$Seat\: C\: dBSPL = 100 dB + \left ( 20\log_{10}0.04 \right )-6 dB$$
    • $$Seat\: C\: dBSPL = 100 dB + \left ( 20\ast -1.37 \right )-6 dB$$
    • $$Seat\: C\: dBSPL = 100 dB + \left ( -27.45\right )-6 dB$$
    • $$Seat\: C\: dBSPL = 66.55\, dBSPL$$

[/listtable]

Table 8.1 Calculating dBSPL of a given loudspeaker aimed on-axis with seat B

In this case the loudest seat is seat A at 77 dBSPL, and seat C is the quietest at 66.55 dBSPL, with a 10.45 dB difference. As discussed, we want all the audience locations to be within a 6 dB range. But before we throw this loudspeaker away and try to find one that works better, let’s take a moment to examine the reasons why we have such a poor result. The reason seat C is so much quieter than the other seats is that it is the farthest away from the loudspeaker and is receiving the largest reduction due to directivity. By comparison, A is the closest to the loudspeaker, resulting in the lowest loss over distance and only a 3 dB reduction due to directivity. To even this out let’s try having the farthest seat away be the seat with the least directivity loss, and the closest seat to the loudspeaker have the most directivity loss.

The angle with the least directivity loss is around 350 degrees, so if we aim the loudspeaker so that seat C lines up with that 350 degree point, that seat will have no directivity loss. With that aim point, seat B will then have a directivity loss of 3 dB, and seat A will have a directivity loss of 10 dB. Now we can recalculate the dBSPL for each seat as shown in Table 8.2.

[listtable width=50% caption=””]

  • A
    • $$Seat\: A\: dBSPL = 100 dB + \left ( 20\log_{10}\frac{3.28′}{33.17′} \right )-10 dB$$
    • $$Seat\: A\: dBSPL = 100 dB + \left ( 20\log_{10}0.1 \right )-10 dB$$
    • $$Seat\: A\: dBSPL = 100 dB + \left ( 20\ast -1 \right )-10 dB$$
    • $$Seat\: A\: dBSPL = 100 dB + \left ( -20\right )-10 dB$$
    • $$Seat\: A\: dBSPL = 70\, dBSPL$$
  • B
    • $$Seat\: B\: dBSPL = 100 dB + \left ( 20\log_{10}\frac{3.28′}{50.53′} \right )-3 dB$$
    • $$Seat\: B\: dBSPL = 100 dB + \left ( 20\log_{10}0.06 \right )-3 dB$$
    • $$Seat\: B\: dBSPL = 100 dB + \left ( 20\ast -1.19 \right )-3 dB$$
    • $$Seat\: B\: dBSPL = 100 dB + \left ( -23.75\right )-3 dB$$
    • $$Seat\: B\: dBSPL = 73.25\, dBSPL$$
  • C
    • $$Seat\: C\: dBSPL = 100 dB + \left ( 20\log_{10}\frac{3.28′}{77.31′} \right )-0 dB$$
    • $$Seat\: C\: dBSPL = 100 dB + \left ( 20\log_{10}0.04 \right )-0 dB$$
    • $$Seat\: C\: dBSPL = 100 dB + \left ( 20\ast -1.37 \right )-0 dB$$
    • $$Seat\: C\: dBSPL = 100 dB + \left ( -27.45\right )-0 dB$$
    • $$Seat\: C\: dBSPL = 72.55\, dBSPL$$

[/listtable]

Table 8.2 Calculating dBSPL of a given loudspeaker aimed on-axis with seat C

In this case our loudest seat is seat B at 73.25 dB SPL, and our quietest seat is seat A at 70 dBSPL, for a difference of 3.25 dB. Compared with the previous difference of 10.55 dB, we now have a much more even distribution of sound to the point where most listeners will hardly notice the difference. Before we fully commit to this plan, we have to test these angles at several different frequencies, but this example serves to illustrate an important rule of thumb when aiming loudspeakers. In most cases, the best course of action is to aim the loudspeaker at the farthest seat, and have the closest seat be the farthest off-axis to the loudspeaker. This way, as you move from the closest seat to the farthest seat, while you’re losing dB over the extra distance you’re also gaining dB by moving more directly on-axis with the loudspeaker.

[aside]EASE was developed by German engineers ADA (Acoustic Design Ahnert) in 1990 and introduced at the 88th AES Convention.  That’s also the same year that Microsoft announced Windows 3.0.[/aside]

Fortunately there are software tools that can help you determine the best loudspeakers to use and the best way to deploy them in your space. These tools range in price from free solutions such as MAPP Online Pro from Meyer Sound shown in Figure 8.25 to relatively expensive commercial products like EASE from the Ahnert Feistel Media Group, shown in Figure 8.26. These programs allow you to create a 2D or 3D drawing of the room and place virtual loudspeakers in the drawing to see how they disperse the sound. The virtual loudspeaker files come in several formats. The most common is the EASE format. EASE is the most expensive and comprehensive solution out there, and fortunately most other programs have the ability to import EASE loudspeaker files. Another format is the Common Loudspeaker Format (CLF). CLF files use an open format, and many manufacturers are starting to publish their loudspeaker data in CLF. Information on loudspeaker modeling software that uses CLF can be found at the website for the Common Loudspeaker Format Group http://www.clfgroup.org.

Figure 8.25 MAPP Online Pro software from Meyer Sound
Figure 8.25 MAPP Online Pro software from Meyer Sound
Figure 8.26 EASE software
Figure 8.26 EASE software

8.2.4.2 System Documentation

Once you’ve decided on a loudspeaker system that distributes the sound the way you want, you need to begin the process of designing the systems that capture the sound of the performance and feed it into the loudspeaker system. Typically this involves creating a set of drawings that give you the opportunity to think through the entire sound system and explain to others – installers, contractors, or operators, for example – how the system will function.

[aside]You can read the entire USITT document on System Diagram guidelines by visiting the USITT website.[/aside]

The first diagram to create is the System Diagram. This is similar in function to an electrical circuit diagram, showing you which parts are used and how they’re wired up.  The sound system diagram shows how all the components of a sound system connect together in the audio signal chain, starting from the microphones and other input devices all the way through to the loudspeakers that reproduce that sound. These diagrams can be created digitally with vector drawing programs such as AutoCAD and VectorWorks or diagramming programs such as Visio and OmniGraffle.

The United States Institute for Theatre Technology has published some guidelines for creating system diagrams. The most common symbol or block used in system diagrams is the generic device block shown in Figure 8.27. The EQUIPMENT TYPE label should be replaced with a descriptive term such a CD PLAYER or MIXING CONSOLE. You can also specify the exact make and model of the equipment in the label above the block.

Figure 8.27 A generic device block for system diagrams
Figure 8.27 A generic device block for system diagrams

There are also symbols to represent microphones, power amplifiers, and loudspeakers. You can connect all the various symbols to represent an entire sound system. Figure 8.28 shows a very small sound system, and Figure 8.29 shows a full system diagram for a small musical theatre production.

Figure 8.28 A small system diagram
Figure 8.28 A small system diagram
Figure 8.29 System diagram for a full sound system
Figure 8.29 System diagram for a full sound system

While the system diagram shows the basic signal flow for the entire sound system, there is a lot of detail missing about the specific interconnections between devices. This is where a patch plot can be helpful. A patch plot is essentially a spreadsheet that shows every connection point in the sound system. You should be able to use the patch plot to determine which and how many cables you’ll need for the sound system.  It can also be a useful tool in troubleshooting a sound system that isn’t behaving properly. The majority of the time when things go wrong with your sound system or something isn’t working, it’s because it isn’t connected properly or one of the cables has been damaged. A good patch plot can help you find the problem by showing you where all the connections are located in the signal path. There is no industry standard for creating a patch plot, but the rule of thumb is to err on the side of too much information. You want every possible detail about every audio connection made in the sound system. Sometimes color coding can help make the patch plot easier to understand. Figure 8.30 shows an example patch plot for the sound system in Figure 8.28.

Figure 8.30 Patch plot for a simple sound system
Figure 8.30 Patch plot for a simple sound system

8.2.4.3 Sound Analysis Systems

[aside]Acoustic systems are systems in which the sounds produced depend on the shape and material of the sound-producing instruments. Electroacoustic systems produce sound through electronic technology such as amplifiers and loudspeakers.[/aside]

Section 8.2.4.1 discussed mathematical methods and tools that help you to determine were loudspeakers should be placed to maximize clarity and minimize the differences in what is heard in different locations in an auditorium.  However, even with good loudspeaker placement, you’ll find there are differences between the original sound signal and how it sounds when it arrives as the listener.  Different frequency components respond differently to their environment, and frequency components interact with each other as sounds from multiple sources combine in the air.  The question is, how are these frequencies heard by the audience once they pass through loudspeakers and travel through space encountering obstructions, varying air temperatures, comb filtering, and so forth? Is each frequency arriving at the audience’s ears at the desired amplitude? Are certain frequencies too loud or too quiet?  If the high frequencies are too quiet, you could sacrifice the brightness or clarity in the sound.  Low frequencies that are too quiet could result in muffled voices.  There are no clear guidelines on what the “right” frequency response is because it usually boils down to personal preference, artistic considerations, performance styles, and so forth.  In any case, before you can decide if you have a problem, the first step is to analyze the frequency response in your environment. With practice you can hear and identify frequencies, but sometimes being able to see the frequencies can help you to diagnose and solve problems. This is especially true when you’re setting up the sound system for a live performance in a theatre.

A sound analysis system is one of the fundamental tools for ensuring that frequencies are being received at proper levels. The system consists of a computer running the analysis software, an audio interface with inputs and outputs, and a special analysis microphone.  An analysis microphone is different from a traditional recording microphone. Most recording microphones have a varying response or sensitivity at different frequencies across the spectrum. This is often a desired result of their manufacturing and design, and part of what gives each microphone its unique sound. For analyzing acoustic or electroacoustic systems, you need a microphone that measures all frequencies equally.  This is often referred to as having a flat response.  In addition, most microphones are directional. They pick up sound better in the front than in the back. A good analysis microphone should be omnidirectional so it can pick up the sound coming at it from all directions. Figure 8.31 shows a popular analysis microphone from Earthworks.

Figure 8.31 Earthworks M30 analysis microphone
Figure 8.31 Earthworks M30 analysis microphone

There are many choices for analysis software, but they all fall into two main categories: signal dependent and signal independent.  Signal dependent sound analysis systems rely on a known stimulus signal that the software generates – e.g., a sine wave sweep.  A sine wave sweep is a sound that begins at a low frequency sine wave and smoothly moves up in frequency to some given high frequency limit.  The sweep, lasting a few seconds or less, is sent by a direct cable connection to the loudspeaker. You then place your analysis microphone at the listening location you want to analyze. The microphone picks up the sound radiated by the loudspeaker so that you can compare what the microphone picks up with what was actually sent out.

The analysis software records and stores the information in a file called an impulse response.  The impulse response is a graph of the sound wave with time on the x-axis and the amplitude of the sound wave on the y-axis.  This same information can be displayed in a frequency response graph, which has frequencies on the x-axis and the amplitude of each frequency on the y-axis.  (In Chapter 7, we’ll explain the mathematics that transforms the impulse response graph to the frequency response graph, and vice versa.) Figure 8.32 shows an example frequency response graph created by the procedure just described.

Figure 8.32 Frequency response graph created from a signal dependent sound analysis system
Figure 8.32 Frequency response graph created from a signal dependent sound analysis system

Figure 8.33 shows a screenshot from FuzzMeasure Pro, a signal dependent analysis program that runs on the Mac operating system.  The frequency response is on the top, and the impulse response is at the bottom.  As you recall from Chapter 2, the frequency response has frequencies on the horizontal axis and amplitudes of these frequency components on the vertical axis.  It should how the frequencies “responded” to their environment as they moved from the loudspeaker to the microphone.  We know that the sine wave emitted had frequencies distributed evenly across the audible spectrum, so if the sound was not affected in passage, the frequency response graph should be flat.  But notice in the graph that the frequencies between 30 Hz and 500 Hz are 6 to 10 dB louder than the rest, which is their response to the environment.

Figure 8.33 FuzzMeasure Pro sound analysis software
Figure 8.33 FuzzMeasure Pro sound analysis software

When you look at an analysis such as this, it’s up to you to decide if you’ve identified a problem that you want to solve. Keep in mind that the goal isn’t necessarily to make the frequency response graph be a straight line, indicating all frequencies are of equal amplitude. The goal is to make the right kind of sound. Before you can decide what to do, you need to determine why the frequency response sounds like this. There are many possible reasons.  It could be that you’re too far off-axis from the loudspeaker generating the sound. That’s not a problem you can really solve when you’re analyzing a listening space for a large audience, since not everyone can sit in the prime location. You could move the analysis microphone so that you’re on-axis with the loudspeaker, but you can’t fix the off-axis frequency response for the loudspeaker itself.  In the example shown in Figure 8.34 the loudspeaker system that is generating the sound uses two sets of sound radiators. One set of loudspeakers generates the frequencies above 500 Hz. The other set generates the frequencies below 500 Hz. Given that information, you could conclude that the low-frequency loudspeakers are simply louder than the high frequency ones. If this is causing a sound that you don’t want, you could fix it by reducing the level of the low-frequency loudspeakers.

Figure 8.34 Frequency response graph showing a low frequency boost
Figure 8.34 Frequency response graph showing a low frequency boost

Figure 8.35 shows the result of after this correction. The grey line shows the original frequency response and the black line shows the frequency response after reducing the amplitude of the low-frequency loudspeakers by 6 dB.

Figure 8.34 Frequency response graph showing a low frequency boost
Figure 8.34 Frequency response graph showing a low frequency boost

The previous example gives you a sketch of how a sound analysis system might be used. You place yourself in a chosen position in a room where sound is to be performed or played, generate sound that is played through loudspeakers, and then measure the sound as it is received at your chosen position. The frequencies that are actually detected may not be precisely the frequency components of the original sound that was generated or played.   By looking at the difference between what you played and what you are able to measure, you can analyze the frequency response of your loudspeakers, the acoustics of your room, or a combination of the two. The frequencies that are measured by the sound analysis system are dependent not only on the sound originally produced, but also on the loudspeakers’ types and positions, the location of the listener in the room, and the acoustics of the room. Thus, in addition to measuring the frequency response of your loudspeakers, the sound analysis system can help you to determine if different locations in the room vary significantly in their frequency response, leaving it to you to decide if this is a problem and what factor might be the source.

The advantage to a signal dependent system is that it’s easy to use, and with it you can get a good general picture of how frequencies will sound in a given acoustic space with certain loudspeakers. You also can save the frequency response graphs to refer to and analyze later. The disadvantage to a signal dependent analysis system is that it uses only artificially-generated signals like sine sweeps, not real music or performances.

If you want to analyze actual music or performances, you need to use a signal independent analysis system. These systems allow you to analyze the frequency response recorded music, voice, sound effects, or even live performances as they sound in your acoustic space. In contrast to systems like FuzzMeasure, which know the precise sweep of frequencies they’re generating, signal independent systems must be given a direct copy of the sound being played so that the original sound can be compared with the sound that passes through the air and is received by the analysis microphone. This is accomplished by taking the original sound and sending one copy of it to the loudspeakers while a second copy is sent directly, via cable, to the sound analysis software. The software presumably is running on a computer that has a sound card attached with two sound inputs. One of the inputs is the analysis microphone and one is a direct feed from the sound source. The software compares the two signals in real time – as the music or sound is played – and tells you what is different about them.

The advantage of the signal independent system is that it can analyze “real” sound as it is being played or performed. However, real sound has frequency components that constantly change, as we can tell from the constantly changing pitches that we hear. Thus, there isn’t one fixed frequency response graph that gives you a picture of how your loudspeakers and room are dealing with the frequencies of the sound. The graph changes dynamically over the entire time that the sound is played. For this reason, you can’t simply save one graph and carry it off with you for analysis. Instead, your analysis consists of observing the constantly-changing frequency response graph in real time, as the sound is played. If you wanted to save a single frequency response graph, you’d have to do what we did to generate Figure 8.36 – that is, get a “screen capture” of the frequency response graph at a specific moment in time – and the information you have is about only that moment. Another disadvantage of signal independent systems is that they analyze the noise in the environment along with the desired sound.

Figure 8.36 was produced from a popular signal independent analysis program called Smaart Live, which runs on Windows and Mac operating systems. The graph shows the difference, in decibels, between the amplitudes of the frequencies played vs. those received by the analysis microphone. Because this is only a snapshot in time, coupled with the fact that noise is measured as well, it isn’t very informative to look at just one graph like this. Being able to glean useful information from a signal independent sound analysis system comes from experience in working with real sound – learning how to compare what you want, what you see, what you understand is going on mathematically, and – most importantly – what you hear.

Figure 8.36 Smaart Live sound analysis software
Figure 8.36 Smaart Live sound analysis software

8.2.4.4 System Optimization

Once you have the sound system installed and everything is functioning, the system needs to be optimized. System optimization is a process of tuning and adjusting the various components of the sound system so that

  • they’re operating at the proper volume levels,
  • the frequency response of the sound system is consistent and desirable,
  • destructive interactions between system components and the acoustical environment have been minimized, and
  • the timing of the various system components has been adjusted so the audience hears the sounds at the right time.

The first optimization you should perform applied to the gain structure of the sound system. When working with sound systems in either a live performance or recording situation, gain structure is a big concern. In a live performance situation, the goal is to amplify sound. In order to achieve the highest potential for loudness, you need to get each device in your system operating at the highest level possible so you don’t lose any volume as the sound travels through the system. In a recording situation, you’re primarily concerned with signal-to-noise ratio. In both of these cases, good gain structure is the solution.

In order to understand gain structure, you first need to understand that all sound equipment makes noise. All sound devices also contains amplifiers. What you want to do is amplify the sound without amplifying the noise. In a sound system with good gain structure, every device is receiving and sending sound at the highest level possible without clipping. Lining up the gain for each device involves lining up the clip points. You can do this by starting with the first device in your signal chain – typically a microphone or some sort of playback device. It’s easier to set up gain structure using a playback source because you can control the output volume. Start by playing something on the CD, synthesizer, computer, iPod or whatever your playback device is in a way that outputs the highest volume possible. This is usually done with either normalized pink noise or a normalized sine wave. Turn up the gain preamplifier on the mixing console or sound card input so that the level coming from the playback source clips the input. Then back off the gain until that sound is just below clipping. If you’re recording this sound, your gain structure is now complete. Just repeat this process for each input. If it’s a live performer on a microphone, ask him to perform at the highest volume they expect to generate and adjust the input gain accordingly.

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If you’re in a live situation, the mixing console will likely feed its sound into another device such as a processor or power amplifier. With the normalized audio from your playback source still running, adjust the output level of the mixing console so it’s also just below clipping. Then adjust the input level of the next device in the signal chain so that it’s receiving this signal at just below its clipping point. Repeat this process until you’ve adjusted every input and output in your sound system. At this point, everything should clip at the same time. If you increase the level of the playback source or input preamplifier on the mixing console, you should see every meter in your system register a clipped signal. If you’ve done this correctly, you should now have plenty of sound coming from your sound system without any hiss or other noise. If the sound system is too loud, simply turn down the last device in the signal chain. Usually this is the power amplifier.

Setting up proper gain structure in a sound system is fairly simple once you’re familiar with the process. The Max demo on gain structure associated with this section gives you an opportunity to practice the technique. Then you should be ready to line up the gain for your own systems.

Once you have the gain structure optimized, the next thing you need to do is try to minimize destructive interactions between loudspeakers. One reason that loudspeaker directivity is important is due to the potential for multiple loudspeakers to interact destructively if their coverage overlaps in physical space. Most loudspeakers can exercise some directional control over frequencies higher than 1 kHz, but frequencies lower than 1 kHz tend to be fairly omnidirectional, which means they will more easily run into each other in the air. The basic strategy to avoid destructive interactions is to adjust the angle between two loudspeakers so their coverage zone intersects at the same dBSPL, and at the point in the coverage pattern where they are 6 dB quieter than the on-axis level, as shown in Figure 8.37. This overlap point is the only place where the two loudspeakers combine at the same level. If you can pull that off, you can then adjust the timing of the loudspeakers so they’re perfectly in phase at that overlap point. Destructive interaction is eliminated because the waves reinforce each other, creating a 6 dB boost that eliminates the dip in sound level at high frequencies.   The result is that there is even sound across the covered area. The small number of listeners who happen to be sitting in an area of overlap between two loudspeakers will effectively be covered by a virtual coherent loudspeaker.

When you move away from that perfect overlap point, one loudspeaker gets louder as you move closer to it, while the other gets quieter as you move farther away. This is handy for two reasons. First, the overall combined level should remain pretty consistent at any angle as you move through the perfect overlap point. Second, for any angle outside of that perfect overlap point, while the timing relationship between the two loudspeaker arrivals begins to differ, the loudspeakers also differ more and more in level. As pure comb filtering requires both of the interacting signals to be at the same amplitude, the level difference greatly reduces the effect of the comb filtering introduced by the shift in timing. The place where the sound from the two loudspeakers arrives at the same amplitude and comb filters the most is at center of the overlap, but this is the place where we aligned the timing perfectly to prevent comb filtering in the first place. With this technique, not only do you get the wider coverage that comes with multiple loudspeakers, but you also get to avoid the comb filtering!

Figure 8.37 Minimizing comb filtering between two loudspeakers
Figure 8.37 Minimizing comb filtering between two loudspeakers

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What about the low frequencies in this example? Well, they’re going to run into each other at similar amplitudes all around the room because they’re more omnidirectional than the high frequencies. However, they also have longer wavelengths, which means they require much larger offsets in time to cause destructive interaction. Consequently, they largely reinforce each other, giving an overall low frequency boost. Sometimes this free bass boost sounds good. If not, you can easily fix it with a system EQ adjustment by adding a low shelf filter that reduces the low frequencies by a certain amount to flatten out the frequency response of the system. This process is demonstrated in our video on loudspeaker interaction.

You should work with your loudspeakers in smaller groups, sometimes called systems. A center cluster of loudspeakers being used to cover the entire listening area from a single point source would be considered a system. You need to work with all the loudspeakers in that cluster to ensure they are working well together. A row of front fill loudspeakers at the edge of the stage being used to cover the front few rows will also need to be optimized as an individual system.

Once you have each loudspeaker system optimized, you need to work with all the systems together to ensure they don’t destructively interact with each other. This typically involves manipulating the timing of each system. There are two main strategies for time aligning loudspeaker systems. You can line the system up for coherence, or you can line the system up for precedence imaging. The coherence strategy involves working with each loudspeaker system to ensure that their coverage areas are as isolated as possible. This process is very similar to the process we described above for aligning the splay angles of two loudspeakers. In this case, you’re doing the same thing for two loudspeaker systems. If you can line up two different systems so that the 6 dB down point of each system lands in the same point in space, you can then apply delay to the system arriving first so that both systems arrive at the same time, causing a perfect reinforcement. If you can pull this off for the entire sound system and the entire listening area, the listeners will effectively be listening to a single, giant loudspeaker with optimal coherence.

The natural propagation of sound in an acoustic space is inherently not very coherent due to the reflection and absorption of sound, resulting in destructive and constructive interactions that vary across the listening area. This lack of natural coherence is often the reason that a sound reinforcement system is installed in the first place. A sound system that has been optimized for coherence has the characteristic of sounding very clear and very consistent across the listening area. These can be very desirable qualities in a sound system where clarity and intelligibility are important. The downside to this optimization strategy is that it sometimes does not sound very natural. This is because with coherence optimized sound systems, the direct sound from the original source (i.e. a singer/performer on stage) has typically little to no impact on the audience, and so the audience perceives the sound as coming directly from the loudspeakers. If you’re close enough to the stage and the singer, and the loudspeakers are way off to the side or far overhead, it can be strange to see the actual source yet hear the sound come from somewhere else. In an arena or stadium setting, or at a rock concert where you likely wouldn’t hear much direct sound in the first place, this isn’t as big a problem. Sound designers are sometimes willing to accept a slightly unnatural sound if it means that they can solve the clarity and intelligibility problems that occur in the acoustic space.

[aside]While your loudspeakers might sit still for the whole show, the performers usually don’t.  Out Board’s TiMax tracker and soundhub delay matrix system use radar technology to track actors and performers around a stage in three dimensions, automating and adjusting the delay times to maintain precedence and deliver natural, realistic sound throughout the performance.[/aside]

Optimizing the sound system for precedence imaging is completely opposite to the coherence strategy. In this case, the goal is to increase the clarity and loudness of the sound system while maintaining a natural sound as much as possible. In other words, you want the audience to be able to hear and understand everything in the performance but you want them to think that what they are hearing is coming naturally from the performer instead of coming from loudspeakers in a sound system. In a precedence imaging sound system, each loudspeaker system behaves like an early reflection in an acoustic space. For this strategy to work, you want to maximize the overlap between the various loudspeaker systems. Each listener should be able to hear two or three loudspeaker systems from a single seat. The danger here is that these overlapping loudspeaker systems can easily comb filter in a way that will make the sound unpleasant or completely unintelligible. Using the precedence effect described in Chapter 4, you can manipulate the delay of each loudspeaker system so they arrive at the listener at least five milliseconds apart but no more than 30 milliseconds apart. The signals still comb filter, but in a way that our hearing system naturally compensates for. Once all of the loudspeakers are lined up, you’ll also want to delay the entire sound system back to the performer position on stage. As long as the natural sound from the performer arrives first, followed by a succession of similar sounds from the various loudspeaker systems each within this precedence timing window, you can get an increased volume and clarity as perceived by the listener while still maintaining the effect of a natural acoustic sound. If that natural sound is a priority, you can achieve acceptable results with this method, but you will sacrifice some of the additional clarity and intelligibility that comes with a coherent sound system.

Both of these optimization strategies are valid, and you’ll need to evaluate your situation in each case to decide which kind of optimized system best addresses the priorities of your situation. In either case, you need some sort of system processor to perform the EQ and delay functions for the loudspeaker systems. These processors usually take the form of a dedicated digital signal-processing unit with multiple audio inputs and outputs. These system processors typically require a separate computer for programming, but once the system has been programmed, the units perform quite reliably without any external control. Figure 8.38 shows an example of a programming interface for a system processor.

Figure 8.38 Programming interface for a digital system processor
Figure 8.38 Programming interface for a digital system processor

8.2.4.5 Multi-Channel Playback

Mid-Side can also be effective as a playback technique for delivering stereo sound to a large listening area. One of the limitations to stereo sound is that the effect relies on having the listener perfectly centered between the two loudspeakers. This is usually not a problem for a single person listening in a small living room. If you have more than one listener, such as in a public performance space, it can be difficult if not impossible to get all the listeners perfectly centered between the two loudspeakers. The listeners who are positioned to the left or right of the center line will not hear a stereo effect. Instead they will perceive most of the sound to be coming from whichever loudspeaker they are closest to. A more effective strategy would be to set up three loudspeakers. One would be your Mid loudspeaker and would be positioned in front of the listeners. The other two loudspeakers would be positioned directly on either side of the listeners as shown in Figure 8.39.

Figure 8.39 Mid Side loudspeaker setup
Figure 8.39 Mid Side loudspeaker setup

If you have an existing audio track that has been mixed in stereo, you can create a reverse Mid-Side matrix to convert the stereo information to a Mid-Side format. The Mid loudspeaker gets a L+R audio signal equivalent to summing the two stereo tracks to a single mono signal. The Side+ loudspeaker gets a L-R audio signal, equivalent to inverting the right channel polarity and summing the two channels to a mono signal. This will cancel out anything that is equal in the two channels essentially, removing all the Mid information. The Side- loudspeaker gets a R-L audio signal. Inverting the left channel polarity and summing to mono or simply inverting the Side+ signal can achieve this effect. The listeners in this scenario will all hear something similar to a stereo effect. The right channel stereo audio will cancel out in the air between the Mid and Side+ loudspeakers and the left channel stereo audio will cancel out in the air between the Mid and Side- loudspeakers. Because the Side+/- loudspeakers are directly to the side of the listeners, they will all hear this stereo effect regardless of whether they are directly in front of the MID loudspeaker. Just like Mid Side recording, the stereo image can be widened or narrowed as the balance between the Mid loudspeaker and Side loudspeakers is adjusted.

You don’t need to stop at just three loudspeakers. As long as you have more outputs on your playback system you can continue to add loudspeakers to your system to help you create more interesting soundscapes. The concept of Mid-Side playback illustrates an important concept. Having multiple loudspeakers doesn’t mean you have surround sound. If you play the same sound out of each loudspeaker, the precedence effect takes over and each listener will source the sound to the closest loudspeaker. To create surround sound effects, you need to have different sounds in each loudspeaker. The concept of Mid-Side playback demonstrates how you can modify a single sound to have different properties in three loudspeakers, but you could also have completely different sounds playing from each loudspeaker. For example, instead of having a single track of raindrops playing out of ten loudspeakers, you could have ten different recordings of water dripping onto various surfaces. This will create a much more realistic and immersive rain effect. You can also mimic acoustic effects using multiple loudspeakers. You could have the dry sound of a recorded musical instrument playing out of the loudspeakers closest to the stage and then play various reverberant or wet versions of the recording out of the loudspeakers near the walls. With multiple playback channels and multiple loudspeakers you can also create the effect of a sound moving around the room by automating volume changes over time.

8.2.4.6 Playback and Control

Sound playback has evolved greatly in the past decades, and it’s safe to say tape decks with multiple operators and reel changes are a thing of history.  While some small productions may still use CD players, MiniDiscs, or even MP3 players to playback their sound, it’s also safe to say that computer-based playback is the system of choice, especially in any professional production.  Already an integral part of the digital audio workflow, computers offer flexibility, scalability, predictability, and unprecedented control over audio playback.  Being able to consistently run a performance and reduce operator error is a huge advantage that computer playback provides.  Yet as simple as it may be to operate on the surface, the potential complexity behind a single click of a button can be enormous.

Popular computer sound playback software systems include SFX by Stage Research for Windows operating systems, and QLab by Figure 56 on a Mac.  These playback tools allow for many methods of control and automation, including sending and receiving MIDI commands, scripting, telnet, and more, allowing them to communicate with almost any other application or device.  These playback systems also allow you to use multiple audio outputs, sending sound out anywhere you want, be it a few specific locations, or the entire sound system. This is essential for creating immersive and dynamic surround effects. You’ll need a separate physical output channel from your computer audio interface for each loudspeaker location (or group of loudspeakers, depending on your routing) in your system that you want to control individually.

Controlling these systems can be as simple as using the mouse pointer on your computer to click a GO button.  Yet that single click could trigger layers and layers of sound and control cues, with specifically timed sequences that execute an entire automated scene change or special effect.  Theme parks use these kind of playback systems to automatically control an entire show or environment, including sound playback, lighting effects, mechanical automation, and any other special effects.   In these cases, sometimes the simple GO isn’t even triggered by a human operator, but by a timed script, making the entire playback and control a consistent and self-reliant process.  Using MIDI or Open Sound Control you can get into very complex control systems.  Other possible examples include using sensors built into scenery or costumes for actor control, as well as synchronizing sound, lighting, and projection systems to keep precisely timed sequences operating together and exactly on cue, such as a simulated lighting strike.  Outside of an actual performance, these control systems can benefit you as a designer by providing a means of wireless remote control from a laptop or tablet, allowing you to make changes to cues while listening from various locations in the theatre.

Using tools such as Max or PD, you can capture input from all kinds of sources such as cameras, mobile devices, or even video game controllers, and use that control data to generate MIDI commands to control sound playback.  You’ll always learn more actually doing it than simply reading about it, so included in this section are several exercises to get you going making your own custom control and sound playback systems.

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