Understanding Audio: Getting the Most Out of Your Project or Professional Recording Studio

thought.

Chapter 1 The Recording Studio

A Brief History and Overview

To better understand audio in the context of recording and the recording studio, we must understand the process first. What is it that we are trying to accomplish? To appreciate this fully, it is beneficial to look at how we have gotten to the point at which we are today. Technology and recording has always been a two-way street, from the point of view of development. The emergence of new technologies, such as multitrack recording, MIDI, and digital audio workstations (DAWs), not only radically change the way we do things but also open up new creative possibilities previously unimagined. At the same time, the drive for new creative directions and for easier, faster ways to do what we need to do often inspire and spawn new technologies. Let’s take a brief look at how recording has evolved over the last century.

Early Recording

Recording Through the 1920s

At the beginning of the twentieth century, recordings were all made direct to disc. The storage/playback media were either wax cylinders or shellac discs that were cut live, one at a time. These discs were then played back on one form or another of phonograph (predecessors to the modern turntable). The recording studio setup consisted of a room in which musicians were arranged around a horn. This horn gathered sound and fed it acoustically to a vibrating diaphragm and cutting stylus (figure 1.1). As the musicians played, the disc or cylinder rotated and a pattern was cut into the wax or shellac corresponding to the acoustic pressure changes of the original signal. The cylinder could then be loaded onto a phonograph with a lighter stylus (needle) and the process reversed. The pattern on the cylinder caused the needle and diaphragm assembly to vibrate, and the resulting air pressure changes were amplified by the horn. From beginning to end, this was a fully acoustic process, with no electronics involved.

Fig. 1.1. Audio recording setup through the 1920s. By the turn of the century, the flat disc coexisted with, and then eventually replaced, the cylinder.

Making multiple copies consisted of having several horn-loaded cutting machines lined up and run simultaneously, as well as having the musicians play the piece multiple times (each time resulting in a slightly different performance, of course). Thus, one could say that the recording, mixing, mastering, and manufacturing processes were all rolled into one; it all happened simultaneously at the initial recording session. Mixing simply consisted of arranging the musicians and instruments at varying distances (and heights) from the main recording horn(s). Further development of the Emile Berliner’s flat disc as well as Thomas Edison’s cylinder did allow for the manufacture of multiple copies from the one master. The flat disc eventually won out commercially in the 1910s.

Mid 1920s to 1950

With the development of the vacuum tube amplifier and the condenser microphone in the 1920s came a new setup, shifting away from a purely acoustical recording process to an electrical one. The microphone could transduce (convert one form of energy into another) the acoustical vibrations of the source into an alternating electrical current. This current would then feed a drive amp and a cutting stylus (figure 1.2). The development of the moving coil loudspeaker allowed for the playback process to also be electrified. Before long, working in the electrical realm would allow for the possibility of having a setup that included multiple micro-phones, each accenting a different portion of the ensemble, each feeding its own dedicated preamplifier and associated circuitry, and collectively feeding the drive amp and cutting stylus. This development in turn gave rise to the need for, or usefulness of, one device or platform that might group together all level controls and switches—namely, the mixer or recording console (figure 1.3). It also gave rise to the development of a two-room studio setup—the studio where the musicians and mics are set up, and the control room where the engineer can monitor the performance through the console and through speakers under more critical conditions. Note that in the late 1940s, magnetic tape recording took hold in the United States and began to be used initially as a safety backup to direct-to-disc recording, and as the standard for prerecorded radio broadcast.

Fig. 1.2. Audio recording setup through the 1940s. Mics feed individual preamps and level controls, which collectively feed a drive amp and cutting stylus.

Still direct-to-disc and mono.

Fig. 1.3. Audio recording setup through the early 1950s. Mics feed preamps and amps through a passive console with level controls (stepped resistor networks) and cutting lathe direct to disc (and/or tape after 1946).

1950 to 1960

Up through the early 1950s, the console consisted of a black box the size of a large book, with four large rotary knobs (with level markings 1 to 10) for respective input levels, one larger knob for overall level, and a few switches. The tube amplifiers for each channel were located in racks accessible via a patch bay. Equalization (EQ) originally consisted of self-contained plug-in cassettes made up of passive resistors tailored to specific microphones. Rather than being used for creative purposes, they were meant to flatten out the peaks and roll-offs inherent in the sonic characteristics of specific microphones. These equalizations would result in a signal more equal to the original sound source being captured. Eventually, EQ would make it into the console as a series of stepped switches that could be manipulated as desired. Magnetic tape recording had arrived after the war (WWII) and coexisted with direct-to-disc recording for about a decade, sometimes playing the role of backup for the main disc master. The great advantage of tape, of course, was that it could be rerecorded as well as edited. Thus, the best segments of several performances could be cut together and presented as a single performance. This practice remains with us to this day, even for classical music, which is often thought to be presented as a live unedited performance. This is, in fact, rarely the case.

Artificial reverb was generally added to the final tape rather than to individual signals, and was in the form of an acoustical echo chamber (both echo and chamber are words that linger today on some consoles and patch bays to designate reverb). The signal from the original tape was sent to the chamber via speaker lines, allowed to reverberate in the chamber, and the result was captured using mics, and recorded onto the final tape or disc. Before long, because of its increased fidelity, decreased surface noise, and ease of editing (not to mention rerecord-ability), magnetic tape recording replaced direct-to-disc altogether.

Late 1950s to 1980

With the advent of stereo recording in the 1950s came the need for ganged (stereo) faders and equalizers, and of course, stepped panpots, which direct a signal towards the left or right channel. These last—consisting of two levels controls (resistors) ganged in inverse proportion (as one level is increased the other decreases proportionally) —were employed mainly to direct individual spot mics, placed to enhance instrument groups within orchestral ensembles, to coincide with their physical placement within the stereo field. Before the end of the decade, legendary musician and audio pioneer Les Paul had conceived of recording using multiple tracks, giving rise to the practice of overdubbing—recording new parts to coincide with and enhance previously recorded tracks (as heard in Les Paul’s classic recordings with Mary Ford). This new technology marks the beginnings of the modern recording studio. With the widespread adoption of 4-track recording in the 1960s (with 8-tracks soon to follow), a whole new approach to music production was born, as is evinced in the Beatles classic Sergeant Pepper’s Lonely Hearts Club Band, a seminal album whose intricate production was astoundingly all done using 4-track recording. Monitoring was typically accomplished using four speakers, tracks 3 and 4 being sent to the inner pair of speakers. The possibility of sending more than one input signal to a given track necessitated the use of combining networks, or busses, which allowed the operator to combine input signals and assign them to a given destination track. Busses were also now used to send signals from each channel via individual level controls to the reverb chamber, the output of which returned to the console and could be mixed in with the final 2-track (stereo) mix.

The emergence of the transistor as a much smaller alternative to the tube for amplification made it easier for the console to include all level or gain stages internally, first in cassette plug-in form, and ultimately in either discrete or integrated circuit (IC) chip form. (Few consoles exist with all tube rather than transistor stages.)

Because of overdubbing, it became necessary for the musician in the studio to hear what had previously been recorded so as to know when and what to play. This necessitated the inclusion of a fold-back or cue system, which generally consisted of an on-off switch on each channel (including the reverb return channel). This switch allowed that channel’s signal to be sent back into the studio for the musician (s) to hear. At this point, level controls were also gradually moving away from stepped rotary pots (circular knobs) and towards linear faders and (continuously) variable-resistor rotary pots.

With the advent of 8-track recording, and given the implausibility of using eight speakers, it was found that virtually any position could be reproduced using just two speakers through phantom imaging.¹ It is really at this point that the modern studio setup and recording console were born in earnest (figure 1.4). We see the emergence of the monitor mix path for the return of tape track outputs. Here, every level control, mute, and solo of the record path is duplicated in a path independent of the recording, for the sole purpose of creating a preview mix for the producer or engineer. This development allows for significant experimental manipulation during the recording session without disturbing the actual recording to multitrack. At this point, equalization also became available in both the record and the monitor path, as did reverb. Foldback switches became cue mix rotary controls, and as tracks multiplied so did the complexity of the cue mix system. And just like that, glossing over a few developmental details along the way, we arrive at the modern recording studio.

Fig. 1.4. The emergence of the modern studio in the 1960s

The Modern Recording Studio

Figure 1.5 shows what a standard multitrack recording session setup might look like in a modern-era studio. The console or desk is the heart of the studio. Through it, all signals pass to be properly balanced, processed, and routed to the appropriate destination. (In a more modest setup such as a home or small project studio, a smaller-format mixer or control surface might replace the console.) It also provides a means of communication between the studio and the control room. The engineer communicates with the musicians in the studio via a talkback mic or engineer’s mic on the console. This mic is routed either through the musicians’ cues (headphones) or to the studio speakers. A communications mic is also set up in the studio and routed through the console to the control room speakers to allow the musicians to talk to the engineer or producer.

Fig. 1.5. Bird’s-eye view of a typical multitrack recording sessions layout in a modern-era recording studio

Instruments can be acoustically isolated from one another using movable barriers called baffles or gobos. Microphones positioned on individual instruments in the studio are patched into the mic input patch panel, which is connected by cables running through the wall, to the mic inputs on the console. Within the console, each low-level mic signal is boosted to a usable line level by a mic preamplifier. The signals can then be processed using equalizers or EQs to adjust tone or timbre, compressors for dynamic level control and punch, noise gates to eliminate unwanted sounds, and faders and panpots, respectively used for level balancing and stereo (or surround) placement or imaging. These effects can be part of the console or can be accessed as outboard gear, along with artificial reverberation, delay, and other effects, via a patch bay, or as computer-based software plug-ins. The destination for signals can be either individual tracks of the multitrack machine, or the 2-track stereo mixdown machine (or, more recently, multiple channels of the multitrack surround mixdown machine for 5.1 surround mixes). These machines can take the form of analog reel-to-reel tape machines (as pictured), digital reel-to-reel (DASH) or cassette-based modular digital multitracks (MDMs, such as ADATs or DA88s), or stand-alone or computer-based hard-disk recorder systems (such as Pro Tools). In addition, the main output signal from the console or mixer feeds power amplifiers that boost the signal level enough to drive the control room speakers or monitors.

Basic Recording Studio Signal Flow

A simplified global studio signal flow is shown in figure 1.6. Input signals are grouped and routed to the multitrack via the track busses, where bus 1 out is normalled to track 1 in, bus 2 to track 2, etc. A bus is a signal path where audio signals can be combined and are jointly routed to a particular destination. A normal is a connection that has been set up between an audio source and destination and does not require repeated patching. The outputs of the multitrack are normalled to the line-level inputs of the console. The main stereo output of the console is normalled to the 2-track machine and to the control room outputs (speakers). The specifics of the signal flow will depend on the type of session occurring. Sessions break down into four general categories: basics, overdubs, mixdown, and live-to-2 (excluding preproduction, postproduction, or mastering).

Fig. 1.6. Basic modern-era recording studio flow. Inputs are routed to the multitrack via the track busses, track outputs are normalled (appear automatically without having to be patched) to the line inputs of the console, and the main stereo bus is normalled to the inputs of the 2-track and feeds control room outputs to the speakers.

Fig. 1.7. Live-to-2 session signal flow. Mic is the source, 2-track (and monitors) the destination.

The most straightforward of these sessions is the live-to-2 (figure 1.7). This type of session is reminiscent of pre-multitrack productions of the ’50s and ’60s. Essentially, all musicians are in the studio at the same time, microphones are routed directly to the main stereo mix bus, and the music is recorded to the 2-track stereo master recorder live, as it happens (hence the term live-to-2). All level adjustments, effects, and other production decisions are made in real-time. Figure 1.7 shows the basic flow for a live-to-2 session. The idea is to make the flow as direct as possible from source to 2-track, as if it were a mixdown session; the difference is that the source signals are from live microphones rather than prerecorded tracks.

The advantage of this type of session is that it tends to be very time-efficient, has a definite immediacy, and captures the natural and spontaneous interaction between the musicians that is sometimes lost in the course of lengthy isolated overdubs. For this reason, it is probably the most common recording situation for jazz as well as classical music. The downside is that decisions about sounds, effects, and levels, once made, cannot easily be changed. A common alternative, live-to-multitrack, overcomes this limitation.

The basics session (figure 1.8) is the initial recording session in a multitrack production project where the basic rhythm section (drums, bass, and perhaps guitar or piano) is often recorded. In this case, our source is still the microphone, but our destination is now the multitrack (as well as the control-room speakers, so that we can hear what we are doing). Individual microphones are generally routed to individual tracks or subgrouped to individual tracks or pairs of tracks. Outboard effects at this point are not generally recorded, but rather included in the monitor mix only, as a preview.

Fig. 1.8. Basics session signal flow. Source is mic; destination, multitrack (and monitors).

The overdub session(s) occurs once the basics session is completed. Tracks are added one by one, in isolation, to fill out and complete the production. In this case, we have two different sources. On the one hand, we have the live mic (or alternatively, a line input) for the signal currently being recorded; on the other hand, we have the previously recorded tracks, which must be monitored and performed to. We also have two different destinations: the live mic is routed to the multitrack to be recorded (and control room monitors to be heard), while the previously recorded tracks are arranged in a rough mix to be sent to the control room monitors (as well as headphones for the musician). The principal flow for an overdub session is shown in figure 1.9.

The mixdown session occurs once all material has been recorded (hopefully). The source is the multitrack, the final destination is the 2-track machine, whether it is a 1/2-inch reel-to-reel or, increasingly, a computer-based or stand-alone hard disk destination. At this point, final effects are added and will be recorded as part of the final mix to the stereo master. Several passes may be performed with minor alterations, such as vocal slightly up (louder) and vocal slightly down, or an instrumental version with no vocals. Additional editing may follow to create a composite or comp mix using favorite sections from the various passes, as well as a shorter radio edit version, etc. The principal mixdown session flow is shown in figure 1.10.

The mastering session is usually done in a studio specializing in this type of work. It consists of taking all of the final 2-track stereo mixes for the entire project (or multichannel mixes, in the case of surround-sound masters), and making global sonic refinements, including global EQ, compression, level matching, and song sequencing.

Fig. 1.9. Overdub session signal flow. Source is mic and previously recorded tracks, destination is multitrack for the mic only (and monitors) as well as headphone mix (cues).

Fig. 1.10. Mixdown session signal flow. Source is multitrack, destination is 2-track master recorder (generally 1/2-inch reel-to-reel analog tape, DAW, or CD-R) as well as control room monitors. Outboard effects are finally recorded along with the mix.

While DATS (digital audio tapes) are sometimes used as masters, the more common professional 2-track stereo master format is still ½-pinch 2-track analog tape. A common format for multi-channel surround masters is the digital tape recording system (DTRS) or Hi-8 8-mm tape used in 8-track digital multitracks such as the Tascam DA-78. Other options include mixing down to two tracks (or multiple, for surround tracks) of a hard disk recorder, or mixing directly to CD or DVD. Mastering can be done in the analog realm, but more commonly, it is done on a digital audio workstation (DAW, figure 1.11). Any analog master tape is transferred onto hard disk through a hardware analog-to-digital (A/D) converter, sometimes preceded by choice analog processing equipment such as vintage EQs and compressors. It is then manipulated entirely in the digital domain, digitally signal-processed, edited, and finally burned directly to a CD or DVD master by means of a CD or DVD recorder.

Fig. 1.11. Mastering session signal flow. Source is 2-track, destination is generally CD or DVD master, by way of a digital audio workstation (DAW).

While the technology of recording changes at a sometimes furious pace, particularly in recent years, the basic underlying principles of session and signal flow, signal level management, acoustics, mic placement technique, and problem solving remain relatively unchanged. [It is interesting to note that as far as we have come, we still often refer to recording as cutting tracks, to reverb as chamber or echo, and to the multitrack recorder as "tape."] It is only through the development and mastery of these fundamental skills that we are able to adapt to the rapid changes in technology. These principles form the basis of all good past and future recordings (as do creativity, experience, experimentation, and love of music); they are the basis of this book. The following chapters will explore each of these topics in great detail, and hopefully lead the reader to a better understanding and a greater ability to make the best recordings under the available conditions.

Chapter 2 The Modern Studio

Recording Studio Basics

As an overview, let’s start by taking a quick look at a typical scenario encountered in the recording studio. We will touch on the elements used in most modern recordings, as they appear in typical order in the recording chain, from source to destination. In every case, the scenario can be broken down into three general elements: a sound source or signal to be recorded, a series of stages to modify that signal in various ways, and a number of possible destinations for the modified signal. Each of these elements will be discussed in much greater detail throughout the book. The concept of source and destination is fundamental to understanding audio and the recording process, and to becoming a functional participant in that process. We will return to it often.

This chapter will provide a foundation for the reader who may be relatively new to the process. The advanced reader with previous recording session experience may still find this a helpful source for review, as well as the starting point for subsequent topics discussed.

Sound Source

The first step is to look at the sound source to be recorded. A typical basics session for rock or pop may consist of drums, bass, keyboard, and scratch vocal. (A scratch vocal is simply a vocal track that is recorded along with the rhythm section, ultimately to be replaced during a later overdub session by the final vocal performance; it is there merely as a guide for the rhythm section, so that the players are better able to respond to the subtleties of the song’s melody and phrasing.) For simplicity’s sake, let’s look at a single element such as the vocal. Most sound sources, musical or otherwise, contain a vibrating or rotating element, such as the string on a guitar or the head on a drum. This element, when struck, vibrates, generating fluctuations in pressure in the air around it. In the case of our vocal, it is the vocal cords that vibrate to generate pressure fluctuations in air.

We hear this as sound when our ear drum responds to these pressure changes and vibrates sympathetically. Our hearing mechanism transduces the vibrations into electrical impulses that are sent back and forth between the brain and the inner ear, and are eventually interpreted as sound. A transducer is any element that converts energy from one form into another, in this case from mechanical energy (vibrating eardrum) to electrical energy (impulses sent between the inner ear and brain).

In the recording process, we need a mechanical, magnetic and/or electrical transducer to take the place of the ear in order to be able to store these fluctuations. The two most common types of transducers used are the microphone and the pickup. Thus, our process begins with placing the sound source in a room and locating a transducer (mic or pickup) near the source in such a way as to best capture the instrument’s sound.

Fig. 2.1. Recording vocals

Microphone

As part of the setup process, one of the first choices encountered by the engineer in the signal chain is selecting the proper microphone to best capture the sound source. All microphones contain a moving element called a diaphragm, which picks up fluctuations in air pressure around it, not unlike the eardrum. When our vocalist sings into the microphone (figure 2.1), the diaphragm is set into motion and its movements are converted into an electrical waveform. This electrical waveform or current is an analog, or copy, of the original sound wave. It travels down the microphone cable and can now be recorded in various ways. Microphone choices include dynamic (moving coil), ribbon, and condenser (capacitor).

Dynamic: Moving Coil

Perhaps the most common microphone is the so-called dynamic microphone. A dynamic mic requires no external power to work. Most dynamic mics contain a plastic diaphragm attached to a coil of wire (figure 2.2). This type of mic is called a moving-coil microphone. The coil sits inside a magnet. When the diaphragm moves, the coil also moves within the magnetic field. This motion creates an electrical current within the coil proportional to the original acoustical waveform. The current is fed to the output of the microphone via wire leads. This process of converting mechanical and magnetic energy to electrical energy is called electromagnetic induction.

Fig. 2.2. Simplified details of a moving-coil microphone

Moving-coil microphones are known for being ruggedly built. They can handle high sound-pressure levels (SPL) as are generated by electric guitar amps or kick and snare drums. They tend to exhibit a roll-off (attenuation) in the higher frequencies (above 10 kHz) as well as a boost in attack range frequencies (2 to 5 kHz, sometimes called presence peak). For this reason, they are often used on drums and are popular for live sound and touring because of their dependability and ability to take abuse, as well as their low handling noise. Some of the most popular dynamic moving coil mics include the Shure SM57 (figure 2.3) and SM58, as well as the Electrovoice (EV) RE20 and Sennheiser MD421. In the studio, moving-coil microphones are simply referred to as dynamic mics. If we choose a moving-coil microphone for our vocal, it will tend to be fairly aggressive, somewhat sibilant, and not overly detailed or shimmery sounding. While moving-coil mics are often used for live vocals on stage, they are not generally a first choice for recording vocals, unless a particularly aggressive sound is desired, or the vocalist insists on holding the mic while recording.

Dynamic: Ribbon

A more particular type of dynamic mic is the ribbon microphone. Instead of a coil, the ribbon mic uses a very thin piece of corrugated metal suspended within a magnetic field (figure 2.4). This ribbon acts as both the diaphragm to capture sound energy and vibrate sympathetically, and the transducer itself, taking the place of the coil. Otherwise, the function is exactly the same as that of the moving coil. However, ribbons are less sensitive than moving coils and therefore will need more gain from the external mic preamplifier. Sensitivity is a measure of output voltage generated given a reference input sound-pressure level.

Fig. 2.3. Shure SM57 dynamic (moving coil) microphone (© 2003 Shure Incorporated. Used by permission.)

Fig. 2.4. Simplified details of a ribbon microphone

Because of its tiny mass, the ribbon is extremely susceptible to damage and tearing. Thus, special care must be taken never to drop or hit ribbon mics at all. For this reason, they are rarely used for live sound reinforcement applications. It is also generally not a good idea to place a ribbon in front of a loud sound source that generates high SPLs, such as an electric guitar amp or a kick drum, although newer ribbon mics are more rugged and better able to handle high SPLs. In addition, because of their construction, ribbon mics must NOT receive any external power. The thin ribbon element is likely to overheat and burn up. While moving-coil mics also do not need external power, under most circumstances, they will be unaffected by it.

At the same time, ribbon mics tend to exhibit a warm low end, and gentle roll-off of high frequencies. For this reason, they are especially pleasing when used on certain bright-sounding sources such as horns, piano, and certain vocals. Some classic ribbon mics include the RCA 44 BX and 77-DX (figure 2.5), as well as mics by Coles and Royer. In the studio, ribbon microphones are typically not referred to as dynamic but simply as ribbon mics. A ribbon mic could be a good choice for our vocal, although these mics are somewhat less common and less likely to be encountered in every studio situation.

Fig. 2.5. RCA 77-DX ribbon microphone (Courtesy Rob Jaczko)

Condenser/Capacitor

Unlike dynamic microphones, condenser microphones require some form of external power in order to work, generally in the form of phantom power (+48 V DC) from the console or mixer. Rather than having an element (coil or ribbon) suspended in a magnetic field, condensers use an electrical element called a capacitor. A capacitor is made up of two metal plates that hold a charge. In a condenser microphone, these take the form of a metallic diaphragm (often metal foil) and fixed backplate (figure 2.6). Phantom power supplies the polarizing voltage to the element and powers the internal preamp in the microphone. The capacitance (a form of electrical resistance) of the element is determined by the distance between the two plates and the voltage across them. When our vocalist sings into the microphone, the diaphragm vibrates, moving alternately closer then further from the backplate. This motion causes a commensurate fluctuation in capacitance, which in turn yields a variation in electrical current. The varying current is an electrical analog of the original acoustical waveform.

Fig. 2.6. Simplified details of a capacitor microphone capsule

Tube condenser mics, which use vacuum tubes instead of transistors for power, come with their own proprietary power supply and should not be fed with the console’s phantom power, unless it is expressly permitted in the mic’s technical documentation.

Condenser microphones are the most sensitive of microphones (highest output voltage for same reference pressure), and also tend to exhibit a much truer sonic characteristic than dynamic mics, extending well into both the low and high ends of the spectrum. For this reason, they are often used to capture detailed and nuanced