Handbook of Recording Engineering

HANDBOOK OF RECORDING ENGINEERING FOURTH EDITION

HANDBOOK OF RECORDING ENGINEERING FOURTH EDITION

by

John Eargle JME Consulting Corporation

Springe]

John Eargle JME Consulting Corporation Los Angeles, CA, USA

Eargle, John. Handbook of recording engineering / by John Eargle.~4th ed. p. cm. Includes bibliographical references and index. ISBN 1-4020-7230-9 (alk. paper) 1. Sound—Recording and reproducing. I. Title. TK7881.4.E16 2002 621.389'3-dc21 2002032065

ISBN 0-387-28470-2 (SC) ISBN 978-0387-28470-5 ISBN 1-4020-7230-9 (HC)

e-ISBN 0-387-28471-0

Printed on acid-free paper.

First softcover printing 2006 © 2003 Springer Science+Business Media, Inc. (hardcover edition) All rights reserved. This work may not be translated or copied in whole or in part without the written permission of the pubhsher (Springer Science+Business Media, Inc., 233 Spring Street, New York, NY 10013, USA), except for brief excerpts in connection with reviews or scholarly analysis. Use in connection with any form of information storage and retrieval, electronic adaptation, computer software, or by similar or dissimilar methodology now know or hereafter developed is forbidden. The use in this publication of trade names, trademarks, service marks and similar terms, even if the are not identified as such, is not to be taken as an expression of opinion as to whether or not they are subject to proprietary rights. Printed in the United States of America. 9 8 7 6 5 4 3 2 1 springeronline.com

SPIN 11545002

CONTENTS Preface

vii

SECTION 1. FOUNDATIONS IN ACOUSTICS Chapter 1. Acoustics in the Modem Studio Chapter 2. Psychoacoustics: How We Hear

1 28

SECTION 2. MICROPHONES Chapter 3. Microphones: Basic Principles 46 Chapter 4. Microphones: The Basic Pickup Patterns Chapter 5. Environmental Effects and Departures from Ideal Performance 65 Chapter 6. Microphones: Electronic Performance and the Electrical Interface 74 Chapter 7. Microphone Accessories 84

53

SECTION 3. RECORDING SYSTEMS: ANALYSIS, ARCHITECTURE, AND MONITORING Chapter 8. Basic Audio Signal Analysis 94 Chapter 9. Recording Consoles, Metering, and Audio Transmission Systems 107 Chapter 10. Monitor Loudspeakers 139

SECTION 4, RECORDING TECHNOLOGY Chapter 11. Analog Magnetic Recording and Time Code Chapter 12. Digital Recording 184 Chapter 13. The Digital Postproduction Environment

154 201

vi

Contents

SECTION 5. SIGNAL PROCESSING Chapter Chapter Chapter Chapter

14. Equalizers and Equalization 213 15. Dynamics Control 222 16. Reverberation and Signal Delay 232 17. Special Techniques in Signal Processing

242

SECTION 6. RECORDING OPERATIONS Chapter 18. Fundamentals of Stereo Recording 254 Chapter 19. Studio Recording and Production Techniques Chapter 20. Classical Recording and Production Techniques 311 Chapter 21. Surround Sound Recording Techniques

267 290

SECTION 7. PRODUCTION SUPPORT FUNCTIONS Chapter 22. Mixing and Mastering Procedures 326 338 Chapter 23. Music Editing and Assembly

SECTION 8.CONSUMER MEDIA Chapter 24. Recorded Tape Products for the Consumer 352 359 Chapter 25. Optical Media for the Consumer Chapter 26. The Stereo Long-Playing (LP) Record 371

SECTION 9. STUDIO DESIGN FUNDAMENTALS Chapter 27. Recording Studio Design Fundamentals

Bibliography Index 424

409

394

PREFACE The fourth edition of the Handbook of Recording Engineering follows the same broad subject outline as the third edition and includes new data on the many developments that have taken place in digital technology and surround sound recording techniques during the last six years. The emphasis of the book has shifted slightly toward needs voiced by teachers of recording technology, and students will find this edition easier to read and study than the earlier editions. Sidebars have been introduced in many of the chapters for detailed technical follow-up, leaving the body of the text free for general commentary. The book is divided broadly into nine sections, described below: 1. Foundations in Acoustics. The studio itself becomes the laboratory for our discussion of both acoustics and psychoacoustics. 2. Microphones. The microphone is indeed the central creative tool of our industry and the subject is given five chapters of its own. 3. Recording Systems: Analysis, Architecture, and Monitoring. A new chapter on audio signal analysis pulls together into a single chapter many concepts previously covered in multiple chapters. The modem in-line console is explained in greater depth than in previous editions. 4. Recording Technology. While analog recording retains its pre-eminence in basic tracking activities, the disc-based digital workstation has become the primary digital tool for both multichannel recording and postproduction work. 5. Signal Processing. Major developments in this area include plug-in "modules" for digital workstations that literally duplicate the highly esteemed equalizers and compressors of the past, and new sampling-type reverberation systems that duplicate the acoustics of famous performance venues around the world. 6. Recording Operations. In the last six or so years, surround sound has attained maturity and is now given parity with stereo techniques. 7. Production Support Functions. The techniques of mixing, music editing, and assembly remain much as before and are essential activities in the real world of audio.

viii

Preface

8. Consumer Media, Along with high-performance media such as the DVD audio and the SACD, the half-century-old stereo LP retains its position as the medium of choice for DJ-driven dance music, and as such it deserves its own chapter. 9. Studio Design Fundamentals, While greater numbers of pop and classical music releases are postproduced in the home project environment, the professional studio remains the center of tracking activities for music of all kinds.

John Eargle Los Angeles, 2002

Chapter 1

ACOUSTICS IN THE MODERN STUDIO

INTRODUCTION A basic knowledge of acoustics is essential for all recording engineers, and there is no better place to start than in the studio itself. In this chapter, we will cover the development of both simple and complex waves. We then move on to sound behavior in rooms, developing the concepts of sound transmission, absorption, reflection, and reverberation. Directionality of typical sound sources in the studio will be briefly discussed, and we end the chapter with a discussion of sound behavior in small spaces, such as isolation booths and reverberation chambers.

THE BASICS OF SOUND Sound waves are produced by variations in air pressure above and below its normal static value. For musical signals the time repetition interval for the variation is called its period, which is composed of one cycle of the wave. The number of cycles per second is called ihQ frequency of sound, normally denoted by the term hertz (Hz). The magnitude of the signal is known as its amplitude, and the time relation between two signals of the same frequency is specified as the phase relationship. For convenience, we state that there are 360 degrees in a single cycle of a sound wave, and relative phase relationships are normally stated in degrees. Figure 1-1 shows these basic relationships. For young persons with normal hearing, audible sound covers the frequency range from about 20 Hz up to about 20,000 Hz. The abbreviation kHz stands for 1,000 Hz {k is the abbreviation of the Greek kilo, meaning "one thousand"). We can then relabel 20,000 Hz as 20 kHz. The range of loudness of sound is fairly wide and is shown in Figure 1-2. The solid curve indicates the audible frequency and loudness ranges over which we normally perceive sound. You can identify the frequency scale along the bottom of the graph. The portion of the solid curve at the at the bottom of the graph is known as the threshold of hearing, or minimum audible field (MAF). Any sounds below this range are not normally heard. The part

Chapter 1

Time

Figure 1-1. A sine wave showing period, frequency, phase, and ampHtude.

of the curve at the top is known as the threshold of feeling', any sound in this range or higher is likely to cause a tingling sensation in the ear, or even be painful to the listener. The vertical scale at the left of the graph is stated in decibels (dB). This term is described in Sidebar 1.1, but for now just remember that each 20 dB on the vertical scale represents a 10-to-l sound pressure difference. The total 120 dB range represented on the vertical scale corresponds to an overall pressure difference of a million-to-one between the loudest and softest sounds we can hear. You can also see as you examine Figure 1-2 that the ear is much more sensitive to low-level sounds in the range between 1 kHz and 5 kHz than it is at higher and lower frequencies. This and other hearing phenomena will be discussed in detail in Chapter 2.

The Basics of Sound

50

100

200

500 Ik Frequency (Hz)

2k

Figure 1-2. Total range of hearing (solid curve); normal ranges of music (dashed curve) and speech (dotted curve). Sidebar 1.1: Introduction to the decibel The decibel (dB) is a convenient way to express the ratio of two powers, and that ratio is always expressed by the ternn level. The bei is defined as: Level = log (W^/WQ) bei, where log indicates the logarithm to the base 10. More conveniently, we use the decibel, which is one-tenth bei: Level =

10 log (W^/Wo) decibel

Let our reference power, PQ, be 1 watt; then 2 watts represents a level of 3 dB, relative to 1 watt: Level = 10 log (2/1) = 10(.3) = 3 dB Extending the ratio, 4 watts represents a level of 6 dB, relative to 1 watt: Level = 10 log (4/1) = 10(.6) = 6 dB In a similar manner 10 watts represents a level of 10 dB, relative to 1 watt: Level = 10 log (10/1) = 10(1) = 10 dB

Chapter 1 Figure 1-3 presents a useful nomograph for determining by Inspection the level In dB of various power ratios In watts. Simply locate the two power values along the nomograph and read the level difference In dB between them.

Decibels above and below a one watt reference power (dBW) -30

-20

-10

0

+10

+20

+30

l ' M ^ M I i M i | i M ^ M I / i l i | ' M i i M l i M | i i | i | i | Willi'i|i|Mljili|i'|i|i|l|i|lil I 2 0.001

4 6 l 2 0.01

4 6 1 0.1

2

4 6 «

2 4 6 1 1 10 Power In watts

2

4 6 » 100

2

4 6 » 1000

Figure 1-3. Nomograph for reading power ratios in watts directly in dB.

EXAMPLE: Find the level difference In dB between the maximum output capability of a 20-watt amplifier and a 500-watt amplifier: Above 20 watts read 13; above 500 watts read 27. Then: Level difference = 2 7 - 1 3 = 14 dB You can see that the relative levels between 100 and 10 watts, 60 and 6 watts, 4 and 0.4 watt are all the same: 10 dB. Obviously, the relative level of any 10-to-1 power ratio Is 10 dB. Likewise, the relative level of any 2to-1 power ratio Is 3 dB. Sound power Is proportional the square of sound pressure, and from this we get the relationship: Sound pressure level (SPL) = 10 log (Pi/Po)^ = 20 log (Pi/Po) The reference power for zero dB SPL Is given as the very small pressure value of 20 micropascals. We will not deal directly with pascals, but only with the pressure levels they produce. Here Is an example: What Is the SPL corresponding to a pressure of one pascal: SPL = 20 log (106/20) = 20 log (50,000) = 94 dB

The dynamic ranges typically occupied by music (dashed curve) and speech (dotted curve) are also shown in Figure 1-2. Music in a concert hall is normally perceived over a dynamic range that doesn't exceed about 80 dB, and speech is normally perceived over an even narrower range of about 40 dB. In most aspects of audio engineering, the horizontal scale on a frequency response graph is skewed to make musical intervals such as the octave appear equally spaced. The logarithmic (log) frequency scale preserves this relationship and is used in Figure 1-2. See Sidebar 2.4 for more discussion of the log scale.

The Basics of Sound

Sidebar 2.4: The log frequency scale Figure 1-4A shows a typical grid for presenting audio frequency response data. 500 Hz and Its succeeding octave values are shown at black markers, and you will note that these values are all equally spaced. By comparison, Figure 1-4B shows a grid with a linear frequency scale. As before, the black markers show 500 Hz and its octaves. As we go up in frequency the markers become more widely spaced, and this is counter to the way we actually perceive the octave Intervals.

Log frequency scale 10

m

IJL in Tfl

0

•o

I

^ -10 > o

oc -20

•

"

HJ

Typical response of a I dynamic microphone

jfl IkJ

p

y

1

f

50

%

100

B

1 k 500 Frequency (Hz)

5k

10 k

20 k

Linear frequency scale

10

ff

0

I ^ -10

I oc -20

-30

Lj i

A

A

A

••H m

A

m

0 1 2

3 4 5k

A

10k Frequency (Hz)

Figure 1-4, Typical log frequency scale (A); linear frequency scale (B).

20 k

Chapter 1 SIMPLE SOUND WAVEFORMS Sound travels approximately 1130 feet per second (344 meters per second) at normal temperature. The pressure variation of a low frequency sound of 100 Hz is shown in Figure 1-5A. Since the wave is in motion, the base line of the graph can be measured in time. At 100 Hz, ihQ period, or time taken for the wave to repeat itself, is equal to 1 divided by the frequency: 1/100 = 0.01 seconds. The waveform shown here is that of a pure tone, known as a sine wave. We can also relate the period of the wave to the actual length of the wave as it propagates through air. At a speed of 1130 feet per second, each cycle of the 100-Hz signal will have a wavelength of 1130/100, or 11.3 feet, as shown in Figure 1-5A. The same information is shown for a midrange frequency of 1 kHz in Figure 1-5B. Here, the wavelength of the signal is: 1130/1000 =1.13 feet, or about 13 inches. One period at 100 Hz (11.3 feet)

One period at 1 kHz (1.13 feet) B

One period at 10 kHz (1.35 Inch)

Figure 1-5. Sample waveforms for 100 Hz (A); 1 kHz (B); and 10 kHz (C).

Complex Waveforms

7

At a frequency of 10 kHz the wavelength in air will be: 1130/10,000 = 0.113 feet, or about 1.35 inch (Figure 1-5C). At 20 kHz, the normal upper limit of audible sound, the wavelength is just a little over half an inch. By comparison, the wavelength at 20 Hz is 1130/20, or 56.5 feet, so the entire frequency range of audible sound covers a 1000-to-l ratio of wavelengths. You will often see wavelength expressed as X (Greek letter lambda), The relationships among frequency (/), wavelength (X) and speed of sound (v) are:

f=vlX X = vlf Figure 1-6 shows the frequency ranges of various musical sources as they relate to the keyboard of a piano.

COMPLEX WAVEFORMS Music and speech are largely composed of periodic complex waveforms that repeat at regular intervals. They consist of harmonics of a fundamental sine wave similar to that shown in Figure 1-7. The lowest frequency present,^^, is called ih^ fundamental ox first harmonic. A frequency of 2 times^^ represents the second harmonic, and so forth. The four harmonics in this example combine as shown to produce a complex waveform. Harmonics as high as the tenth or twelfth are common in brass instrument waveforms when those instruments are played loudly. Many conmionplace sounds exist for a very short time, and it is difficult to isolate any periodic behavior in the waveform. Figure 1-8A shows the recorded waveform of a hand clap. The total time occupied by the signal is only about half a second, and most ofthat consists of the acoustical "ringout" of room reflections after the hands have contacted each other. Figure 1-8B shows the waveform of the continuous sound "ah." We can see the periodic nature of the waveform, and we can also see slight cycle-tocycle variations within the waveform. The spoken word "yes" is shown in Figure 1-8C. Note the clear delineation of the waveform for each component of the word. The note "A" below middle C played on a trombone is shown in Figure 1-8D. The harmonic structure of this steady-state waveform is fairly detailed, and the peak value of the waveform is much greater than its average value. (More on this in Chapter 8.)

Chapter 1

00000'9L

00 00001

Figure 1-6. Frequency ranges of instruments and voices compared to piano keyboard.

Sound Behavior in a Large Studio First harmonic

SecorKi harmonic

1

2fn

Third harmonic

13fo

rvwv

Fourth hanrtonic

i

4fn

Frequency

Time

fo

Time

•

2fo

• •

3fo

4fo

Frequency

Figure 1-7. Combining sine waves, first four harmonic waveforms (A); harmonics represented on a frequency scale (B); summed waveform (C); the four contributing frequencies (D).

SOUND BEHAVIOR IN A L A R G E STUDIO Figure 1-9 shows a perspective view of a modem studio used for film scoring as well as for laying down basic tracks for pop recording projects. Both walls and ceiling areas consist of individually adjustable sections so that the studio can be made "live" (reflective) or "dead" (absorptive), as required for the music being recorded. In pop recording, additional movable baffles (goboes)

Chapter 1

10

y||i|iiiiii>iiiiiW)ii|i(Mwi>iii»iiii100 msec)

S - Sound source L - Listener

Effect of a single delay at 45° off-axis of a sound source in front of the listener

B

Image shift

Disturbance

CO

"in c

c-10 (D

-g-15 -20 -25

\40

60

80

Time (milliseconds)

Figure 2-11. Direct, early reflections and reverberation in an auditorium. View of space (A); the subjective effect of a single delay on subjective impression (B).

Variation of reverberation time with frequency The normal tendency in any kind of listening space is for the reverberation time to decrease at high frequencies and increase at low frequencies. This is a result of the normal increase in atmospheric sound absorption at high frequencies and decrease in absorption of most building materials at low frequencies. The average effect of this is shown in Figure 2-13. It is not unusual for a room with a midband reverberation time of 2 seconds to have a reverberation time of 3 seconds at 50 Hz, as indicated by this figure.

43

Hearing Protection 2.2

^

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8 1.8 0)

(D

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125 k 355

250 k 710

500k 1420

IMCUft 2840 m^

VOLUME Figure 2-12. Target reverberation time versus room volume for various activities.

HEARING PROTECTION As the world around us becomes increasingly louder, and as reinforced music performance becomes the norm, recording engineers must be concerned with hearing protection. This is also true in the modem workplace, and both the Occupational Safety and Health Act (OSHA) and the Environmental Protection Agency (EPA) have laid down regulations regarding allowable noise exposure on the job. The OSHA criteria are given below: Sound Pressure Level (A-weighted) 90 92 95 97 100 102 105 110 115

Daily Exposure: (hours) 8 6 4 3 2 1.5 1 0.5 0.25

44

Chapter 2

•1

1.5

\J

1 ^

^ O Co'

rs ^

^

0.5

25

50

100

200

400

800

1600

3150

6300

12.5k 20k

Frequency (Hz)

Figure 2-13. Normal variation of low and high frequency reverberation versus midrange value.

If you have ever left a rock concert, recording session or remix session with even a slight ringing or tingling sensation in your ears, you are at risk for some degree of hearing loss over time.

eartip

stem

end cap

B

0.5

1

2

Frequency (kHz)

Figure 2-14. Hearing protection. Sketch of an earplug suitable for high-level music listening (A); transmission losses for various methods of hearing protection (B). (Data courtesy Etymötic Research)

Hearing Protection

45

Ear defenders range from simple foam plugs to elaborately molded, custom-fitted models which provide fairly uniform attenuation over the frequency range. For most applications here you vs^ill find that models designed to provide a loss of 15 dB uniformly over the frequency range will be quite satisfactory. However, if you are on a firing range, heavy-duty, externally worn ear defenders providing considerably more attenuation will be required for absolute safety. Figure 2-14 shows a sketch of a typical earplug suitable for those in the music business. Note that the transmission loss of this design is fairly uniform at about -20 dB over a large portion of the frequency range.

Chapter 3 MICROPHONES: BASIC PRINCIPLES

INTRODUCTION The microphone was introduced with the earliest telephone systems in the 1870s. Broadcasting and electrical recording came approximately a half century later. The earliest telephone transmission systems did not employ electrical amplification, and the signal from the microphone, or transmitter, was used directly to drive the telephone receiver. By the time electrical recording and broadcasting were introduced, amplification had become an integral part of audio signal transmission, and as a result microphones could be engineered for higher quality rather than for maximum power output. By the time "Hifi" arrived in the late 1940s microphones had reached a high level of performance—so much so that many of the old German and Austrian models of those days are still in use and may sell in the $5,000 range or greater. Today's microphones cost much less than the earlier models and generally have more uniform and extended frequency response as well as a lower noise floor.

PRINCIPLES O F TRANSDUCTION A transducer is a device that converts energy from one domain to another. A microphone is such a device and converts acoustical pressure variation into a corresponding electrical signal. In this chapter we will analyze the primary methods of transduction that have been used in microphone design over the years.

OLDER DESIGN PRINCIPLES Sidebar 3.1 gives details on the operation of two older microphone designs: the carbon microphone and the crystal microphone. The carbon microphone is still used today in telephone applications, where its relative simplicity and ruggedness are its strong points. In operation, variations in sound pressure cause the diaphragm to move, creating variations

Condenser Microphones

47

in the compression of the carbon granules and varying the net resistance in the electrical circuit. The current in the circuit is modulated according to the acoustical signal, causing an acoustical output from the receiver. The crystal microphone was used at one time for paging purposes and is still used today in some very high pressure applications as w^ell as in underwater acoustics. Certain crystalline materials exhibit ?ipiezoelectric (from the GxQQk piezein, "to press") effect; that is, when they are bent or flexed, a proportional voltage is produced across a pair of the crystal's facets. Neither the carbon nor crystal microphone has performance characteristics suitable for recording; however, the piezoelectric effect has been used for contact pickups on guitars and piano sounding boards. Sidebar 3.1: Carbon and crystal microphones Figure 3-1A shows details of the carbon microphone as used in early telephone engineering. Sound impinges on a diaphragm which is connected to a carbon button, a cup containing granules of carbon and a movable electrode. When the diaphragm moves, the carbon granules are alternately compressed an uncompressed, causing a similar variation In the electrical resistance in the circuit. The bypass capacitor provides a low Impedance path for the audio signal around the voltage source, and the varying ac current flowing through the receiver produces an output signal. Figure 3-1B shows details of a typical crystal microphone. In this example, two crystals are cemented together (known as a bimorph) in order to increase the output voltage. When flexed by the motion of the diaphragm a signal voltage appears at the output.

As recording and broadcasting got underway, microphone design shifted to capacitor (condenser) and dynamic principles or transduction because of their inherent greater frequency bandwidth and lower noise floors. Companies such as Western Electric in the United States and Neumann in Germany were among the first to develop high-performance capacitor microphones.

CONDENSER MICROPHONES Capacitor microphones are universally known as condenser microphones, and that is the term we will use throughout this book—even though capacitor is the preferred technical term. The condenser principal is based on the following equation: Q-CE

3.1

Chapter 3

48

A. Carbon microphone system Speech input Pp:::^

Modulated Carbon current ^^^ranules

Speech output

Telephone transmitter Bypass capacitor

Sound input

B. Crystal microphone

o

Electrical output

Figure 3-1. Details of the carbon microphone (A) and crystal microphone (B).

where Q is the electrical charge on the plates, C is the capacitance, and E is the applied voltage. In microphone application, one plate of the condenser, the backplate, is fixed, and the other plate, the diaphragm, is placed very close to it and is free to vibrate when sound strikes it. The combination of backplate and diaphragm in a single structure is generally referred to as a capsule. As the diaphragm moves in and out under the influence of sound waves the capacitance will also vary. As the diaphragm gets closer to the backplate the capacitance will increase, and vice versa. If the charge across the condenser is held constant, the changes in capacitance will result in corresponding changes in the voltage across the condenser. This voltage is the output of the microphone. Sidebar 3.2 gives details on the operation of the condenser microphone.

49

Condenser Microphones

Sidebar 3.2 Operation of the condenser microphone Figure 3-2A shows the basic relationship among fixed charge, capacitance and voltage across the plates of the condenser. The Greek delta (A) indicates a small change or variation in the quantity it is attached to. As we can see in the equation, a small variation in capacitance (AC) will result in a small variation in output voltage (AE). If O remains constant, then voltage and capacitance will vary inversely; that is, when C increases E will decrease. For Q to remain constant, a polarizing DC voltage is applied to it externally through a very high resistance. For small values of delta the variation in output voltage will be a near replica of the variation in capacitance, and the output of the microphone will be linear. A. Voltage (E) across a variable capacitance (C) with fixed charge (Q)

B. Section view of a condenser microphone Diaphragm

Microphone

V ^ Reference

0 decreases; E increases

C increases; E decreases

Backplate

Q = CE AE = Q/AC

C. Externally polarized condenser microphone

Diaphragm

Insulator

Insulating ring

D. Electret polarized condenser microphone

Diaphragm /I II a\j\ I I

Insulator y

Electret coated backplate

Amplifier

Polarizing voltage

Figure 3-2. Details of the condenser microphone. Effect of variations in capacitance (A); cutaway view of a typical condenser microphone capsule (B); operation of the externally polarized condenser microphone (C); operation of the electret condenser microphone (D). A cutaway view of a typical condenser microphone is shown in Figure 3-2B. Figure 3-2C shows the circuit for the standard externally polarized form of the microphone. Here, a battery (or other DC voltage source) is used

50

Chapter 3 to establish the charge on the condenser. The resistor R is in the range of 10 megohms so that the charge on the condenser remains constant. The signal voltage is then amplified directly at the microphone capsule and reduced to a lower impedance so that the microphone can drive the signal over sufficient distance via a microphone cable without loss. Many newer microphone designs make use of prepolarlzed condenser elements known as electrets. An electret is a material that maintains a fixed charge across its front and back surfaces. Such materials have been know for at least a century, but their application to microphone design dates only from the 1960s. The basic design shown at Figure 3-2C can be reconfigured as an electret design as shown at D. Here, there is no polarizing voltage supply and the overall design is simpler. As before, an amplifier must be provided directly at the microphone capsule to produce a low Impedance signal output.

Early electret materials tended to be unstable over time, but those problems have been solved. While the electret has tended to dominate lower cost microphone design, it has also been used in some of the highest quality models of recent years. Most notably, the Bruel & Kjaer company makes use of electret technology in their superb series of studio microphones.

DYNAMIC MICROPHONES Dynamic microphones make use of the principle of magnetic induction^ in which a coil of wire produces a small output voltage as it moves through a magnetic field. It is the inverse of a traditional dynamic loudspeaker, which you are all familiar with. In order to cover the necessary audio frequency range, the voice coil, as it is called, is normally no larger in diameter than about one-half inch. It is attached to a very light diaphragm, normally made of plastic, or in some older designs, thin aluminum. A close relative is the ribbon microphone. Here, the voice coil has been replaced by a thin corrugated ribbon that is suspended in a magnetic field. The ribbon is open on both sides, and its directional response takes advantage of this to produce a "figure-eighf' pickup pattern. Both of these designs are discussed in Sidebar 3.3. Sidebar 3.3: Microphones based on magnetic induction The basic principle of magnetic induction is shown in Figure 3-3A. If the flux (or flow) of a magnetic field is in the direction shown, and if a piece of wire Is moving in the direction shown, a positive voltage will be

Dynamic Microphones

51

induced in the wire in the direction shown. A practical application is shown at B. Here, the wire is in the fornn of a coil and the magnet is shaped to produce a flux path that is circular (or annular) as well. A diaphragm is attached to the coil as shown and moves under the influence of sound pressure, causing a voltage output at the terminal as shown. B. Section view of magnet/coil/ diaphragm assembiy

A. Vector relationships

Diaphragm

Coil Voice coil motion

i

Voltage

Magnetic flux Magnetic flux path

C. Perspective view of the ribbon microphone

Output

D. Response (poiar) pattern of the ribbon microphone

Ribtx)n

180°

-90° Output step-up transfomier

Figure 3-3. Details of the dynamic microphone. Magnetic induction (A); cutaway view of a moving coil dynamic microphone (B); perspective view of a ribbon microphone (C); polar response of a figure-eight microphone (D).

52

Chapter 3 The structure of a typical ribbon microphone is shown in Figure 3-3C. Here, the coil has been replaced by a straight section of corrugated aluminunn known as a ribbon. The magnetic field is likewise straight and cuts through the ribbon over its entire length. The signal output is taken from the ends of the ribbon and is normally stepped up through a small transformer located directly in the microphone case. The figure-eight directional, or polar, pattern is shown at D. It is clear that the output resulting from a sound source in the direction of zero or 180 degrees will be maximum. But for sounds in the directions of plus or minus 90 degrees the response will be zero, since those sound pressures will cancel at the ribbon, resulting In no net motion. Mathematically, the figure-eight shape can be described by the polar equation: p = cos e

3.2

where p (Greek letter rho) is the output signal magnitude and 0 (Greek letter theta) is the angle of sound incidence.

Chapter 4 MICROPHONES: THE BASIC PICKUP PATTERNS

INTRODUCTION Recording engineers have at their disposal a variety of microphone pickup patterns. The two fundamental "building block" patterns are omnidirectional (omni) and bidirectional, or figure-8. The omni is basically uniform in all directions, although at very high frequencies it will show some directionality along its principle axis of pickup. These two patterns are shown in Figure 4-lA and B. By combining these two basic patterns we can produce a cardioid microphone, as shown in Figure 4-2. Today, however, virtually all cardioids are produced using a single diaphragm capsule. In this chapter we will discuss the derivation of the various patterns and introduce the reader to the basics of usage of the patterns.

Figure 4-1. The basic patterns. Omnidirectional (A); bidirectional, or figure-eight (B).

CARDIOID (Unidirectional)

p=1

p = cos e

p = .5 + .5 cos e

Figure 4-2. Producing a cardioid pattern by summing omnidirectional and bidirectional patterns.

Chapter 4

54

PRODUCING A CARDIOID PATTERN The polar equation for the standard cardioid pattern is: p - 0.5(1 + cos e)

4.1

Producing this pattern by combining two separate elements was common at one time, but today we can produce the cardioid pattern more efficiently and accurately as shown in Figure 4-3. The response to a frontal signal (0 degrees) is shown at A, The delay path (At) allows the diaphragm to be actuated since the signal reaching the back of the diaphragm is always delayed by a fixed amount relative to the signal at the front of the diaphragm.

Magnet

ATB

= ATF

Figure 4-3, A single diaphragm dynamic cardioid microphone. Response at 0 degrees (A); response at 180 degrees (B).

55

Producing a Cardioid Pattern

However, for a signal arriving from 180 degrees, the internal delay path and the path around the microphone to the front of the diaphragm are designed to be equal. In this case there will be a cancellation at the diaphragm over a fairly wide frequency range. This design principle applies to dynamic microphones, as shown here, as well as to condenser microphones, as shown in Figure 4-4.




.25^2)

2.5 dB

4.8 dB

5.7 dB

6dB

1.3

1.7

1.9

2

(1) MAXIMUM FRONT TO TOTAL RANDOM EFFICIENCY FOR A FIRST-ORDER CARDIOID. (2) MINIMUM RANDOM EFFICIENCY FOR A FIRST-ORDER CARDIOID.

(Data presentation after Shure Inc.)

Figure 4-9. Data on first-order car<Jioid patterns.

acoustical power arriving along the principal axis of the microphone. RE is related to the directivity index (DI), as discussed in Chapter 1, by the following equation: DI - 10 log RE

4.6

Distance factor (DF) is a measure of the "reach'' of the microphone in a reverberant environment, relative to an omnidirectional microphone. For example, a microphone with a distance factor of 2 can be placed at twice the distance from a sound source in a reverberant field, relative to the position of an omnidirectional microphone, and exhibit the same ratio of direct-to-reverberant sound pickup of the omnidirectional microphone. DF is related to directivity index by the following equation: DI = 20 log DF

4.7

60

Chapter 4

Stated differently, under reverberant conditions, the recording engineer does not think of directional microphones in terms of their rejection of sounds arriving from some back angle; instead, the microphone is judged in terms of its ability to reject all reverberant sound, relative to the on-axis response. Figure 4-10 shows graphically the on-axis pickup properties of microphones that have different values of DF.

e

Sources

Omnidirectional

1.3

e

Subcardloid

Gradient

1.7

Cardloid

1.7

Supercardloid

1.9

Hypercardloid

2.0

Figure 4-10, Distance factors (DF) for the various cardioid patterns.

VARIABLE PATTERN MICROPHONES Outside of the early RCA 77-series ribbon microphones, virtually all of the variable pattern microphones you will encounter in the studio are large format condenser models. Figure 4-11 shows the basic operation of the classic Braummühl-Weber dual diaphragm structure, and a view of a typical studio model is shown in Figure 4-12. Basically, the system acts as a pair of backto-back cardioids which can be either added or subtracted in their effect by the voltage switching around them. Figure 4-13 shows the net result of the various switching combinations.

61

Variable Pattern Microphones

Cardioid 2

Cardioid 1

xO Ö:-:-;:

•:'m ^'^

^>W

4—o

Omni Cardioid

Figure 4-11. Circuit details for Braunmühl-Weber dual diaphragm condenser design.

These microphones generally work best in their normal cardioid configuration, and their omni patterns are probably the least effective. If you need a good omni, we recommend that you choose a fixed pattern omni microphone.

Chapter 4

62

Figure 4-12. Photo of a typical large format, variable pattern condenser microphone. (Courtesy AKG Acoustics)

HIGH DIRECTIONALITY MICROPHONES Engineers working in television and motion pictures very often have a need for a microphone that can be placed 8 to 10 feet away from an actor and still produce good dialog quality. There are a number of so-called "rifle" microphones that will provide the directivity necessary for this application. Figure 4-14 shows a view of such a model. The designation rifle comes from the appearance of the microphone.

63

High Directionality Microphones Cardioid 1

Cardioid 2

Resultant

0

0

Figure 4-13. Producing various patterns by adding and subtracting back-to-back cardioids.

In design, most of these microphones have a hypercardioid capsule as their basis, and an interference tube is placed in front of the capsule. The tube provides a clear path for sounds originating on-axis; for sounds off-axis the presence of the openings along the tube will produce an interference pattern which reduces the net signal reaching the hypercardioid capsule. The effect of this is dominant at high frequencies, and typical polar response of such a microphone is shown in Figure 4-15.

Chapter 4

64

Figure 4-14. Photo of a rifle microphone about 18 inches in length. (Courtesy AKG Acoustics)

180

125 Hz 250 Hz 500 Hz 1000 Hz

180°

2000 Hz 4000 Hz / aOOOHz I 16000 Hz

Figure 4-15. Polar response of the rifle microphone shown in previous figure. (Courtesy AKG Acoustics)

Chapter 5 ENVIRONMENTAL EFFECTS AND DEPARTURES FROM IDEAL PERFORMANCE

INTRODUCTION In this chapter we will discuss some of the aspects of microphone performance in their normal working environments. Topics such as proximity effect and interference effects due to reflections and combined multi-microphone outputs will be discussed. We will also discuss the normal variations in ideal response which all microphones exhibit to some degree.

PROXIMITY EFFECT If you talk closely into any first-order directional microphone you will hear a rise in bass response. This arises from the fact that the directional microphone has both front and back paths to the diaphragm. For close-in sound sources, there will be a significant difference between the sound pressure levels at the two entry points, and this diiference is dominant at low frequencies, causing a rise in response. Omnidirectional microphones do not exhibit this effect, since sound pickup takes place only at the front side of the diaphragm. The following discussion in Sidebar 5.1 presents a more detailed discussion of the proximity effect. Sidebar 5.1: Proximity effect in first-order microphones Figure 5-1A shows the basic cause of proximity effect in a figure-eight microphone. Here, S represents the sound source; D^ is the distance to the front of the microphone and D^ is the distance around the microphone to the back opening. The net force on the diaphragm is shown at B. We can see that there are actually two forces on the diaphragm, a gradient force which rises with frequency and an inverse square force which is constant with frequency. The equation that defines the amount of LF proximity rise is: A/I + (kr)^ Boost (dB) = 20 log -^—-^-^

5.1

Chapter 5

66

Microphone

Net force on gradient element CQ "O)

>

^

/ > ^ Frequency (phase) dependent ^ \ force (6 dB/octave slope)

Log frequency

CD

"05 > 0

Log frequency Figure 5-L Proximity effect. A sound source close to a gradient microphone (A); Net force on the microphone diaphragm (B); net output from the diaphragm (C). The electrical output of the nnicrophone is further nfiodifled by the velocity of the diaphragnn's motion, which causes a 6-dB-per-octave rise at lower frequencies. The net output of the microphone Is shown at C.

Proximity Effect

67

For a figure-eight microphone the proximity effect at several operating distances is shown in Figure 5-2. You can see that for very close operation the LF response rise due to proximity effect is slightly greater than 24 dB at 50 Hz. Even at an operating distance of about 21 inches the rise at 50 Hz is about 7 dB.

36

30

S:-

24

|N^-—H-\-.—H

[-

-j

j

i

J

18

12 I 21 i i ^ ^ 1 10.6irV^ \

4.25irNL

2.1 I r ^ ^

J

!

|

600

Ik

2k

6

0 12.5

25

100

200

5k

Frequency (Hz) Figure 5-2. Proximity effect for various operating distances from a figure-eight microphone.

For a cardioid microphone the degree of proximity effect is less than with a figure-eight since there is a considerable omni component in the derivation of the cardioid pattern. Figure 5-3 shows the proximity rise for several operating distances from a cardioid microphone. Figure 5-4 shows the variation in proximity effect for a fixed operating distance (24 inches) and with varying angles about the microphone. You can see that at 90 degrees there is no proximity effect; this Is because the gradient (cosine) component is zero at that angle. At 180 degrees the proximity effect rises very rapidly at low frequencies. Many so-called "vocal microphones" have a bass-cut switch that compensates for proximity effect, as shown in Figure 5-5. Other vocal microphones are purposely rolled off at low frequencies, with the knowledge that they are going to be used at close distances, as shown in Figure 56.

Chapter 5

68

36

30

^ 24' OQ

i 12

21 In

^^0.5 in

J^^Jn

i ^ ^ •"

I

J

i

•

J

6

0

1

12.5

• ^'''^'''''^^^^*^^ ^"'^'^^^P^'^'^'^H^ 50

25

100

200

i 2k

Ik

500

5k

Frequency (Hz) Figure 5-3. Proximity effect for various operating distances from a cardioid microphone. 6

r^^ 1 : h-

yj

! I

CD

0°

"0

30**

>

H-

60°

h-

CC

-6

90°

U

Vso** 1

20

50

100

200

1 —

500

L-. Ik

2k

Frequency (Hz) Figure 5-4. Proximity effect for various operating angles at a distance of 24 inches (0.6 meter) from a cardioid microphone.

On- and Off-Axis Microphone Performance at High Frequencies

— i

:r|"rT_.: —

-r-j-

69

"^ 1.1: i. .1 - i- - 1 "~i~Tj — —- — __.--—~Z— ZZZL Z l rr: z: ' ~ r ' r \\~ —' iizzz: ___- —-"~:~-^ 2 £r;M: r« [Ü^ Zr^Zi ^:^^^. Vzzz JJ:: m z' rt Tp"- -- EIZ".IE' r j i r "i1 :7->SJ „lIZ J \ 1 'f-i^.-^ -10 ^ — ^ ü^ü ttet ff—p ;Z 4 1 .z:i. ----- ZL E|EJ 4|E:-i 'El-pE/ i."i r.' "7 7:1: ^ -20 L_-Z = I L

\SI

.zz

Z71' 7.: •"Zjl'-f

—

i

fl— — —

T.-.:

I L L Z T Z

j-j;^-'

y\^j^^^

20

50

gJbfeE 100

200

500

1000

2000

5000

10,000

20.000 Hz

Figure 5-5. Effect of a bass rolloff switch on a vocal microphone. (Courtesy Neumann/USA)

oo +10

3 mm (V8IN) 25 mm (1IN) 51 mm (2 IN)

o Q. 00 Ui DC UJ

n ^

> -10 § LU

;

oc

20

. - ' * " •'- v, .--— tv\ ,,^"*. ^ 1

: ^

• • • i

52 .''

1 '1

rl*» ^

1 ^f

fes

0.6 m (2 FT)

50

100

200

500

1,000

10,000 20,000

FREQUENCY IN Hz Figure 5-6. Response of a vocal microphone at several operating distances. (Courtesy Shure Inc.)

ON-AND OFF-AXIS MICROPHONE PERFORMANCE AT HIGH FREQUENCIES: The diffraction effects we studied in Chapter 1 have an important effect on the HF performance of microphones. The data shown in Figure 5-7 indicates the general trend. If a microphone is designed for flat on-axis response, its response to random signals will be as shown a t ^ . The microphone can also be configured for flat response in the random field, and its on-axis response will rise, as shown at B. Each microphone type has its intended uses, and the engineer should always be aware of these HF on- and oif-axis differences. In general usage, microphones that are flat on-axis are most useful when you are operating fairly close-in in the studio environment. If you are operating at a distance (for example, in a concert hall), it may be to your advantage to choose microphones that are flat in the random or diffuse field.

Chapter 5

70

OD •o

\

"

>

10 dB

>

T"

05

I On-axis | B& K DD 0251 grid

\\

0°

\r V

UL

20

40

80100

1 Random r

200

400

8001 k

2k

4k

8 k 10 k 20 k 40 k

Log frequency (Hz) B

1, 1 On-axis |

>

10 dB

a>

1"

>

••-^

(0

DC

20

40

1 1

[Random | \

Bgfk' n n n 0 0 7 nrirl

\^

80100

1 200

400

0"

8001k

2k

4k

8 k 10 k 20 k 40 k

Log frequency (Hz) Figure 5-7. Response of a microphone designed for flat on-axis response (A) and flat random incidence response (B). (Data after Brüel&Kjaer)

INTERFERENCES DUE TO REFLECTION A classic case of microphone interference is shown in Figure 5-8. Here, a microphone is placed at some distance from the sound source; floor reflections interfere with the direct sound from the source, resulting in uneven response as shown. As the microphone is moved closer to the reflecting surface, the disturbances is less. When the microphone is placed directly on the reflecting boundary the effect disappears. Boundary layer microphones are quite useful in picking up sound in the theater as well as on tables, podiums and the like. The microphone model shown at C has been designed for surface mounting and exhibits uniform response from all operating angles. In many studio applications, the severity of floor or other surface reflections can be minimized by using a directional microphone. In some cases, the null angle of the directional microphone can be aimed directly at the source of the reflection, reducing it to inaudibility.

71

Interferences Due to Reflection

Microphone positions

500 Frequency

1k

5k

Figure 5-8. Effect of floor reflections. Positions of talker and microphone (A); responses of microphone (B); photo of a boundary layer microphone (C). (Photo at C courtesy Crown International)

72

Chapter 5

MULTI-MICROPHONE PICKUP PROBLEMS We can get away with many things in stereo recording which may come back to haunt us when the recording is played back in mono. A typical situation here is shown in Figure 5-9, where a piano is recorded in stereo with a pair of spaced microphones. Their distances from the instrument are Z)y and D2> When played in stereo the recording may sound excellent, but if there is a requirement for good stereo-to-mono compatibility, the sum of the two microphones may present problems. Specifically, there will be reinforcements and cancellations in the combined response, as given by the following equation:

/ = ^ ^

5.2

D, • D j

where D2 is the longer distance and c is the speed of sound. There will be signal reinforcements at frequency multiples of 3/2/ 5/2/, 7/2 fand so forth, and cancellations at frequencies intermediate between these values. As you can see, this problem is related to the one shown in Figure 5-8. There is no clear solution to this problem as such; if there is a requirement for good mono compatibility, the engineer and producer should make a mono summation and approve it before moving on with the recording project. This will usually entail moving the microphones somewhat closer together and making sure that their distances from the center of the instrument are minimized. Microphone 1 Microphone 2

Figure 5-9. Interference effects with a single sound source and multiple microphones.

Variations in Microphone Directional Response

73

VARIATIONS IN MICROPHONE DIRECTIONAL RESPONSE Do not make the mistake of assuming that a cardioid microphone has a uniform pickup pattern over the entire frequency range. What you are most likely to see is data such as is shown in Figure 5-10. A set of typical polar plots is shown at A and the corresponding axial measurements at 0, 90, and 180 degrees is shown at B. Study such data carefully if you want to know the frequency range over which the microphone actually has a recognizable cardioid pickup pattern.

270° 16 kHz

180°

125

250

50

1kHz

2 kHz

4 kHz

8 kHz

16 kHz

Figure 5-10. Cardioid microphone directional response aberrations. Polar plots (A); off-axis frequency response curves (B).

Chapter 6 MICROPHONES: ELECTRONIC PERFORMANCE AND THE ELECTRICAL INTERFACE

INTRODUCTION This chapter covers the basic electronic aspects of the microphone and its integration into the audio signal chain. We will discuss the microphone's performance in terms of its basic performance characteristics, such as: output sensitivity, self noise floor, distortion and electrical output impedance. Additional topics will cover powering of condenser microphones, losses in microphone cables and loading effects at the downstream console input. We will also touch briefly on the wireless microphone.

BASIC MICROPHONE ELECTRONIC PERFORMANCE Output sensitivity The output sensitivity of a microphone expresses its signal output for a specified acoustical input. Today we universally use a reference acoustical pressure input of one pascal, which is equivalent to a sound pressure level of 94 dB. The microphone's output when placed in the reference sound field is given in millivolts per pascal (mV/Pa) or as a voltage level per pascal (dBV/Pa). For example, a typical studio condenser microphone may have a rated sensitivity of 20 millivolts per pascal, indicating that the output voltage will be 0.020 volts (or 20 mV) when the microphone is placed in a sound field of 94 dB-SPL. Another way to express this is as a dB rating relative to one volt: Sensitivity = 20 log (0.02 V)/Pa = -36 dB re 1 Pa Typical sensitivity values for various studio microphones are shown in Table 6.1. You will note that the design sensitivity of the microphone is tailored to its application. Microphones designed for normal studio use represent an average of many models. Those microphones intended for close-in use on stage have lower sensitivities, and those that are intended for distant use on-

Basic Microphone Electronic Performance

75

stage or for distant pickup in television or film recording will have higher sensitivity. The aim is to keep the basic microphone output signal fairly uniform, regardless of its primary application. Table 6.1. Microphone Sensitivity Ranges by Use Microphone Usage: Close-in, hand-held Normal studio use Distant pickup

Normal Sensitivity Range: 2-8 mV/Pa 7-20 mV/Pa 10-30 mV/Pa

Microphone noise floor This rating states the electrical output noise level of a microphone relative to the actual environment in which you may be using that microphone. As an example, assume that you are recording in a very quiet concert hall w^ith a noise rating of NC 10 (noise criterion 10). This means that the inherent noise floor in the hall falls below the 10-phon curve as shown in Figure 2-1. If the microphone's self-noise floor, as measured using the A-weighting curve (Figure 2-3), falls just within the 10-phon curve, we state that the microphone's self noise rating is lO-dB(A). You can think of it this way: a microphone with a self-noise rating of 10dB(A) behaves as if it were an ideal, noiseless microphone in a performance environment with an acoustical noise rating of NC-10.

Distortion at high levels There is a limit to the sound pressure level that a microphone can handle before the onset of distortion in the microphone itself. For studio-grade condensers the reference value is 0.5 percent total harmonic distortion (THD). For dynamic microphones normally used on-stage the reference may be either 1 or 3 percent. The microphone's noise floor and its distortion rating define its useful dynamic range, as shown in Figure 6-1. Here we see the dynamic range for a typical studio microphone. Between the noise floor of 10 dB-A and the onset of distortion at 135 dB-SPL there is a useful operating range of 125 dB. This is slightly greater than the dynamic range of a digital recorder operating with 20-bit conversion. Most studio condenser microphones have a built-in switchable pad (output attenuator) that introduces 10 or possibly 12 dB reduction of output level. As you can see in Figure 6-1, the effect of the pad is to shift the entire operating range of the microphone upward, including the microphone's noise floor itself.

Chapter 6

76

150| 0.5% THD i\

140| 130

0.5% THD ii

120 110 100 5

90

2

80 —

1 70 (D

&. 60

T3

i 50 o ^ 40 30 20 10 0

)^ ^ Nor mal

1 ^^ With 10-dB pgId

Figure 6-1. Operating level ranges of a studio condenser microphone with and without integral -10-dB pad engaged,

Microphone output impedance and recommended load impedance Figure 6-2 shows a schematic diagram of a microphone looking into the input of a recording console. Professional microphones, whether condenser or dynamic, are all balanced; that is, the signal is developed between a pair of conductors placed within a shield, as can be seen in the figure. The standard Microphone cable

Console input

Signal is transmitted between pins 2 and 3; pin 2 is "hor and 1 is the ground or shield

For condensers, Zs = 50 - 200 ohms for dynamics, Zs = 200 - 600 ohms

For typical cable: R = 0.025 ohm/foot C = 30 pF/foot

ZL = 3000 - 5000 ohms

Figure 6-2. Illustration of source impedance, cable, and load impedance in a microphone transmission circuit.

Basic Microphone Electronic Performance

11

input/output hardware is the XLR receptacle, with male configuration for outputs and female configuration for inputs. Sidebar 6.1 analyzes in detail the complex relation between microphone output and console input sections. Sidebar 6.1: Microphone signal flow The microphone has an Internal (or source) Impedance that can vary from 50 to 200 ohms for condensers and from 200 to 600 ohms for dynamics. As the microphone output "looks" Into the cable and the console Input downstream, It "sees" a load Impedance. In modern recording system design, the load Impedance Is at least five times that of the microphone's source impedance. In the example shown here, the microphone's source Impedance Is 200 ohms, and it looks at a load of 3000 ohms at the input of the console. The ratio of load-to-source impedance Is 15-to-1.The microphone cable normally consists of two inner conductors surrounded by a shield. The length of the microphone cable may be anywhere between 10 feet (3 meters) and 660 feet (about 200 meters). Typical high quality microphone cable will have a resistance of about 0.025 ohms per foot and Inter-conductor stray capacitance of about 30 picofarads per foot. In a typical studio setting, the microphone cable length will not exceed about 60 feet (20 meters), and cable losses will be negligible. However, for very long runs the stray capacitance may result in HF attenuation as shown in Figure 6-3. If you are using a dynamic microphone with an output impedance of 600 ohms, the loss will be greater than with the condenser microphone. 33 ft (10 m) cable (200 ohm source) -0.5 CD

200 ft (60 m) cable (200 ohm source)

-1

S -''•5 -2 CO

O

200 ft (60 m) cable (600 ohm source)

-2.5 -3 -3.5

JL 50

L 100 200

i 500

1 1 k

I 2k

J

L

5 k 10 k 20 k

Frequency (Hz) Figure 6-3. Microphone cable losses over distance as a function of source impedance.

Chapter 6

78

The stand-alone microphone preamplifier Many recording engineers routinely use dedicated, stand-alone preamplifiers for all recording activities as an alternative to the microphone input sections of recording consoles. While a good console has excellent microphone preamps, a separate preamp may have some very desirable performance features such as variable input impedance, step-type trim controls and higher output capability. Specifically, the variable input impedance allows a better match with the output impedance of the microphone, which may result in smoother frequency response, and the higher output capability may result in better performance using older tube-type studio condenser models with their typically higher output levels. Figure 6-4 shows a photo of a modem stand-alone microphone preamplifier.

0 OUTPUT

' ^^r*

f ^

^

Z our ^

Figure 6-4. Front and rear views of a stand-alone microphone preamplifier. (Data courtesy FM Acoustics)

Powering condenser microphones Modem solid state condenser microphones make use of phantom powering (also known as simplex powering), in which microphone capsule polarization and signal amplification is powered by 48-volts dc across the signal leads and ground or shield through the microphone cable. The basic phantom powering

Basic Microphone Electronic Performance

79 48 V de

R = 6800 Ohms

To preamp

To microphone

1'

5

To preamp

Signal Is transmitted between pins 2 and 3; dc is provided between pins 2-3 and 1

To preamp

Figure 6-5. Details of 48-volt phantom powering.

circuit is shown in Figure 6-5. As you can see, the positive voltage is applied to each signal lead through a 6800-ohm resistor. The powering system is generally referred to as P48. While not widely used, there are two other phantom powering standards, P24 and P12. Table 6.2 details the three phantom powering standards: Table 6.2. Phantom Voltages and Current Limits Supply voltage Supply current Feed resistors

12 ±1 V max. 15 mA 680 ohms

24 ±4 V max. 10 mA 1200 ohms

48 ±4 V max. 10 mA 6800 ohms

Chapter 6

80

T-powering is also used, but to a much more limited extent than phantom powering. Figure 6-6 gives circuit details of this powering system. Here, the dc power source is fed between the two signal leads, and any slight variation in the power supply will be reflected through as signal output from the microphone.

: 180 ohms

To microphone

ü

i3

]C To preamps

Both signal and power transmitted between pins 2 and 3.

? r-L

EI

]C To preamps

12 volts dc Figure 6-6. Details of 12-volt T-powering.

Battery powering Many condenser vocal microphones are powered with a single 9-volt battery placed within the microphone case so that they may be used with older mixing consoles that do not have integral phantom powering. Similar powering is used for wireless, hand-held microphones (See Sidebar 6.2).

WIRELESS MICROPHONES Today, wireless microphones are used throughout the entertainment industry for on-stage and other pickup. The recording engineer will not normally use them in the studio, but live recording will certainly include them. While wire-

Wireless Microphones

81

less microphones have improved over the years, their performance is not as good as wired models. Specifically, w^ireless microphones make use of complementary compression and expansion to attain a workable dynamic range, and this action is sometimes audible. Also, even with the best of care in setup procedures, wireless microphones may present noise problems in dense urban areas where there are many radio frequency (RF) communications channels in operation. See Sidebar 6.2 for technical details concerning wireless microphones. Sidebar 6.2: Details of wireless microphones Wireless microphones operate with a radiated RF power of no more than 10 milliwatts, and their normal operational range can be as high as 300 to 500 feet, if there are no obstacles. Each microphone must have Its own dedicated receiver, although multiple receivers can operate via a common receiving antenna. Within the microphone's transmitter, the signal is compressed in dynamic range by 2 to 1, and the signal given a HF pre-emphasls. At the receiver, a complementary expansion curve and an inverse HF de-emphasis are applied. These processes are shown In Figure 6-7.

Chapter 6

82 A. Action of compressor/expander Pre-emphaslzed

compressor out Transmission

input to transmitter

Input to expander

De-enr)phasi2ed output from receiver

OdB

-20 dB

-

-40 dB

'

-60 dB —

'

/

Noise

^,

^^x

-80 dB Noise

B. Pro-emphasis and de-emphasis curves "1

T

Pre-emphasts (+6 dBADctave)

•

' T

y

T

De-emphasis (-6 dB/octave)

» ./^

'S

5 20

200 2k Frequency (Hz)

20k

20

200 2k Frequency (Hz)

N

20k

C. Principle of diversity reception ^-S,"'-»

"??, "N

\\. pjrec1signal_

W Transmitter

^S'...

V I

_Diirecl_a9naj

B Diversity receiver

I

j

When a diversity receiver is used, two antennas, spaced by atxHit orw-fourth to one wavelength, pick up the signal, and there is a very low likelihood that cancellation will take place at tx>th antennas simultaneously.

Figure 6-7. Wireless microphone details. Companding action (A); pre- and de-emphasis (B); diversity reception (C).

Wireless Microphones In the United States wireless microphones operate over channels ranging as follows: VHF (very high frequency) range: Low band: 49-108 MHz High band: 16^216 MHz UHF (ultra high frequency) range: Low band: 450-806 MHz High band: 900-952 MHz The reception process is further aided by the so-called d/Vers/Yy process, in which there are two receiving antennas for each microphone placed about one-fourth wavelength apart at the transmitting frequency. The stronger of the two received signals is always used, thus ensuring adequate reception at all times. A photograph of wireless transmitters and receiver Is shown in Figure 6-8.

Figure 6-8. Photo of wireless microphone, bodypack, and receiver. (Courtesy AKG Acoustics)

83

Chapter 7 MICROPHONE ACCESSORIES

INTRODUCTION Microphones are rarely used without accessories. Even a simple hand-held microphone will require a foam windscreen, and of course in any studio application, the microphone will have to be positioned by a stand, boom or other mounting method. Accessories fall basically into the following groups: mounting accessories (stands, booms, stand adapters, shock mounts and stereo mounts), environmental protection (wind screens of all types), and electrical accessories (in-line adapters and microphone splitters).

STANDS AND BOOMS Figure 7-1 shows a group of microphone stands and booms as used in studio recording. Stands range from fairly lightweight models that can reach a height of about 12 or 14 feet (3 to 4 m) to more robust models that can fly a large microphone array. A boom is a swivel attachment positioned at the top of a stand and allows the engineer to place a microphone over the heads of the studio performers or to reach inside a drum set. Large booms need to be counterweighted for stability, and some larger models used in scoring sessions may provide a reach into the orchestra of 8 to 10 feet (2.5 to 3 m). A family of hand-held booms is shown in Figure 7-2. These are used throughout the film and video industries for close miking of actors. In normal application the boom operator is required to keep the boom and microphone outside the film or video frame.

Microphone Mounts

85

Figure 7-1. Typical studio microphone stands and booms. (Courtesy AKG Acoustics)

MICROPHONE MOUNTS Every microphone is provided w^ith its own clip, a small attachment that screv^s onto the top of a microphone stand and to which the microphone is snapped, or clipped, in place. These are adequate in many cases, but where there is any possibility of floor-transmitted vibrations a shock mount may be required. Figure 7-3 shows a typical large format variable pattern condenser

Chapter 7

86

Microphone

Boom

Boom operator

Boundary of video or film frame

'^^^^^^m^^^^^^!^m^^^^^p::m^^^^m^m^^m^mm^^m^:^^. Figure 7-2. Hand-held microphone booms and typical usage. (Courtesy K-tel)

microphone mounted on a stand with a shock mount. Mounts of the kind shown here are normally designed for a given model of microphone. For effective performance it is important that the mechanical resonance of the microphone-shock mount combination be well below the audible range.

Stereo Mounts

87

Figure 7-3, Shock-mounted microphones in the studio. (Courtesy Neumann/USA)

STEREO MOUNTS For many studio applications a pair of microphones need to be precisely positioned relative to each other, and there are many stereo mounts to choose from. Figure 7-4 shows a simple stacked arrangement that allows a pair of small format microphones to be closely arrayed. The model shown in Figure 7-5 is more flexible and allows a pair of microphones to be spaced and angled relative to each other.

Chapter 7

Vxgure 7-4. A simple stereo mount for small format condenser microphones. (Courtesy AKG Acoustics)

Figure 7-5. An articulated stereo mount. (Courtesy Audio Engineering Associates)

Hanging Cable Mounts

89

HANGING CABLE MOUNTS For many live concert recordings the engineer has to dispense with stands in favor of hanging microphones. A number of manufacturers provide a mount similar to the one shown in Figure 7-6. A single microphone can be swiveled and tilted as required.

Swivel joint

Tilting microphone clip

Figure 7-6. A typical cable mount allowing adjustment of horizontal and vertical angles.

WIND AND POP SCREENS These accessories are used both outdoors under windy conditions as well as in the studio. For moderate wind problems, relatively small foam screens can be slipped over the microphone to reduce the effects of puffs of wind from the

90

Chapter 7

performer's mouth. Such sounds and "p" and "b" are notorious for causing "pops." It is better to stop these at the source rather than try to reduce them by equahzation during postproduction. A typical example is shown in Figure 7-7.

Figure 7-7. A typical foam windscreen. (Courtesy AKG Acoustics)

For studio vocal recording a Nylon screen is preferred because it is virtually transparent acoustically. A typical application is shown in Figure 7-8. For outdoor use in effects and news gathering a shroud such as is shown in Figure 7-9 may be necessary to provide substantial reduction of wind noise.

IN-LINE ELECTRICAL ACCESSORIES In your work you will come across a variety of plug-in electrical accessories, including filters, polarity switchers, loss pads, matching transformers and the like. These have traditionally been intended for semi-professional public address activities and are not recommended for general use in sound recording. The only device that you may make extensive use of is a microphone splitter—and then only under controlled conditions. A microphone splitter is a device that allows a microphone's output to be fed to two, possibly three, downstream activities: primary recording, sound reinforcement and broadcast feeds. A schematic drawing is shown in Figure 7-10. The model shown here is passive and accommodates the output of a single microphone. Multiple splitters are also available that provide for a direct feed for the primary activity and multiple amplified outputs for other activities.

In-Line Electrical Accessories

Figure 7-8, A Nylon windscreen in typical studio use. (Courtesy Schoeps)

Figure 7-9. A shrouds for high-wind environments. (Courtesy beyerdynamic)

91

Chapter 7

92 Transformer Electrostatic shield -02

n Split

r

output 1

20-t HD3 - 0 1 -» -02

From microphone

3 0 1 0

Split output 2

-03 -01

20-J Direct

-»

jfitlilil-

output

1 0

1^: Ground lifts

Figure 7-10. Circuit details of a passive microphone splitter.

As you can see in the figure, the path from microphone to recording console or preamplifier is straight-through and is thus unaffected by the splitting process. The microphone is fed to a transformer that has two secondary windings, and each of these is used to feed other activities. Note particularly that phantom powering from your console will reach the microphone; the other console destinations are isolated by the transformer. A set of ground-lift switches can be used in case there are hum or buzz problems arising from improper system grounding. There are two important recommendations. If you are in charge of a recording, make sure that your activity is the one that receives the direct microphone output. Also, when everyone else has been connected to the system, make sure that there are no unusual noises or buzzes. You should be completely free and clear from any "hard" connection to the electronic systems the other activities are using. You should also ensure that the transformer secondary circuits look into standard microphone input impedances no lower than about 3,000 ohms.

93

In-Line Electrical Accessories

Figure 7-11 shows a group of in-line accessories. The "tum-arounds" shown at A can be used to straighten out certain miswirings; the polarity inverter shown at B is used to reverse the polarity of a miswired cable; a balanced loss pad is shown at C; a step-up transformer is shown at D; and a lowpass filter is shown at E, Items at A and B may be used with phantom powering, but the others cannot. As you can see, most of the problems that are solved with these in-line accessories are better solved through competent engineering.

B

D

3 2 1

XLR>M

XLR-M

XLR-F

XLR-F

XLR-F

XLR-M

XLR-F

XLR-M

XLR-F

XLR-M

IL 1 1

XnilU

(step-up transformer)

XLR-M

XLR-F E

3 2 1

__ 1 1

T"

_ (high-pass filter)

Figure 7-lL In-line microphone electrical accessories.

Chapter 8 BASIC AUDIO SIGNAL ANALYSIS

INTRODUCTION In this chapter we will take a close look at audio signals of all kinds. These may be speech or music programs in mono, stereo or multichannel, or they may be test signals which are used to diagnose various kinds of transmission problems. Many of the problems you will encounter in recording will be clearly audible, while others may require some kind of test intrumentation or test procedure to be identified precisely. Other problems in signal transmission have to do basically with subjective appraisals, such as loudness and spectral matching, as well as stereophonic judgements regarding spatiality, image specificity, and so forth.

CHARACTERISTICS OF A PROGRAM CHANNEL The term channel is often used to describe an audio program path intended for final delivery to the consumer. It may be a radio signal, TV audio signal, or a signal intended for a home audio playback medium. We can also think of channel groups, such as stereo or surround sound, where a number of channels are intended for simultaneous playback. For now, we'll consider a single, or monophonic, channel. (The term mono is normally used instead of monophonic) A brief "time history" of such a channel can be represented as shown in Figure 8-1. This figure shows the variation in signal level over some period of time. There is an average signal level, a maximum possible signal level and a system noise floor. The maximum level represents the upper limit of signal transmission. For example, in radio broadcasting the maximum level is defined as the degree of signal modulation that will fit into the "broadcast space" allowed by the Federal Communications Commission without interference with a neighboring broadcast station. In an analog recording, it represents the upper signal limit of the recording medium itself before the onset of a stated degree of distortion. In a digital channel, the maximum level is defined as digitalfull-scale, that value beyond which the signal cannot be represented by the digital code.

95

Program and Playback Requirements Program envelope Maximum possible

T — Average level

Time Figure 8-1. Illustration of a program envelope as it varies over time.

The noise floor of the channel is the residual level of the system when there is no signal applied to it. In analog electronic systems the noise results from thermal agitation arising at the molecular level. In a digital system the noise floor arises at the lowest levels of digital signal quantifying (more about this in a later chapter). Analog recording media have their own noise characteristics, and these usually result from granularity in the medium itself. Figure 8-1 also introduces the notions of headroom and signal-to-noise ratio. Headroom is that signal space between average modulation level and maximum possible level. Signal-to-noise (S/N) represents the normal operating range of the transmission channel.

PROGRAM AND PLAYBACK REQUIREMENTS If you are in the broadcast business you are aware of the competitive requirements of keeping your station's signal at the highest possible loudness level relative to other stations on the dial. This means that your average level must be high, and this requires that headroom be minimized. In other words, you will want to "contain," or limit the signal so that its average level might be no less than about 8 or 10 dB below the maximum level available. The same requirements will apply to any kind of commercial recording. Record producers want their product to "sound loud" and catch the immediate attention of the listener, and both producers and artists are keenly aware of the wide variety of conditions under which their product will be heard. The automobile has a very narrow loudness range into which

Chapter 8

96

the signal has to fit. In spite of some of the loud rigs you may hear on the road, the average automobile listener plays music levels no greater than perhaps 80 to 85 dB SPL. Since the road noise level in the average automobile is in the 55 to 60 dB range, this leaves only about a useful 20-dB range for music presentation in that environment. In Figure 1-8, we showed the waveforms for several kinds of audio signals. If we take a continuous, flowing speech signal and compress its time scale into about 20 seconds, it will resemble that shown in Figure 8-2. As you can see, the signal has occasional high peaks, but the bulk of the signal lies at much lower values. This signal has a peak-to-average ratio of about 12 dB, and this means that the average level, which relates directly to program loudness, will be about 12 dB lower than peak levels. If the maximum allowable signal is ±1 volt, then the average signal will be about ±0.25 volts, as shown in this figure. Peak envelope of speech signal over 20-second period

Time

Figure 8-2. A speech waveform over a 20-second time period with a peak-to-average ratio of 12 dB.

A transmission channel carrying this signal will not be very efficient, since it is already "maxed out" with its normal levels at -12 dB. If we compress or limit the signal's amplitude, we can increase its average value while leaving the peak signal values the same as before. For radio use we could easily compress the signal so that the peak-to-average ratio was no more than about 8 dB, as shown in Figure 8-3. For music, we would normally not want to compress the program any more than is shown here.

97

Signal Frequency Spectra Peak envelope of speech signal over 20-second period +1

i l l II llilll 1 lliili 11 III ill II llilll 1 llii 11 IJI

+0.4 V [average

£

|||(|||fi'^p«'fP'|ll||Hf'^P"¥T«l

CD

§ CO

c g) CO

ifiiriiPi^^ l l i l l l 1 ill 1 1 1 r'l 1 1 I I l l i l l l 1 III 1

1 1 >l I I

0.4 V [average

1 li

Time Figure 8-3. A speech waveform over a 20-second time period with a peak-to-average ratio of 8 dB.

SIGNAL FREQUENCY SPECTRA The frequency spectrum of a signal is the envelope normally occupied by the signal averaged over some period of time. Some examples are shown in Figure 8-4. Aflat spectrum, as the name indicates, is uniform across the audible frequency band. Certain test signals have a flat spectrum, and a great deal of instrumental rock music has a spectrum that is fairly flat out to about 8 kHz. A symphony orchestra has a spectrum that begins to roll off above 250 Hz, as shown. Male speech has a spectrum that peaks at about 250 Hz and rolls off above and below that frequency, as shown in Figure 8-5. Since most transmission channels have flat transmission capability over their frequency band, it is fairly obvious that, with a given speech signal, the signal could be easily boosted in the 2-kHz octave band by about 4 to 6 dB, resulting in increased intelligibility. We can do this without modifying the channel to any degree. While we would be affecting the quality of the speech signal by making it somewhat unnatural, there is no question that we would at the same time be increasing intelligibility. In noisy environments, such as transportation terminals, this is a common practice among sound reinforcement engineers.

Chapter 8

98 Typical rock and electronic music spectrum

250

500

Ik

2k

Frequency (Hz) Typical symphonic music spectrum

250

500

1k

2k

Frequency (Hz)

Figure 8-4. Typical octave-band spectra for electronic music and symphonic music.

SIGNAL POLARITY Polarity (or phase) defines a signal in the "positive/negative" sense. Perhaps the best way to explain this is by way of Figure 8-6A. In this recording/playback system (or chain, as it is often called) each element, from microphone to loudspeaker, maintains the same output polarity as shown at the acoustical input of the microphone. The logical way to maintain this condition is to design each element in the entire chain to be non-inverting—^that is, to preserve input polarity at the output of each device. Modem electronic components preserve identical input and output polarity as shown at C. By comparison, an inverting device will operate as shown at D, You can fall into a polarity trap if you aren't careftil. If two of the devices in the audio chain are inverting, the output of the entire chain will still exhibit matched polarity between input and output. But, depending on which devices are

99

Signal Polarity

Long-term Speech Spectrum

1k

500

2k

Frequency (Hz) Figure 8-5. Long-term male speech octave-band spectrum.

involved, you could end up with a polarity problem if the system were to be reconfigured. Such problems happen more often that we'd like to think. In mono transmission there may be no dire consequences; however, if there is a mismatch in a stereo pair of channels you will be in trouble. Referring back to Figure 2-8, the creation of a clear phantom center image requires that the exact signal be fed to both loudspeakers. If one of these signals is inverted, the phasor reconstruction corresponding to a frontal phantom image cannot take place and the resulting sound will be confiising and unnatural. A

Recording chain

1 Acoustical signal input

Preamp

Microphone

• ^

D^O^

Medium

Recorder

Mixer

A=

Playback chain

Preamp

Medium

A= +

a

Non-inverting

Amplifier

Loudspeaker

^

^{>A


o—

1/3-octave real-time analyzer

O O O O

Figure 8-7. A 1/3-octave real-time analyzer fed with a pink noise signal (A); typical measurement application in a control room.

ELECTRICAL SIGNAL SUMMATION When audio signals are added, either directly in the electrical domain or as separate loudspeaker outputs in the acoustical domain, the summations are not necessarily what you might expect. Let's take the electrical case first. As shown in Figure 8-8A, two identical signals will sum directly, producing a value that is twice either one, representing a level increase of 6 dB. At B the two signals have been summed anti-phase, and it is obvious that they will cancel. At C we show the effect of summing two sine waves of the same frequency and level, but differing in their relative phases. The 90 degree shift shown here will resuh in a net output that is 1.4 times the amplitude of either of the input signals and which has a relative phase or 45 degree. At D we show the effect of summing two separate noise sources of the same level and spectral characteristics. The net output has increased by a factor of 1.4, representing a level increase of 3 dB.

Chapter 8

102 A Two equal signals of like polarity (in-phase)

4-

B Two equal signals of opposite polarity (anti-phase)

ZERO

+ 0

C Two sine waves of same frequency & level with 90^ phase shift 4^ = 0° Amplitude = .75 (

\

'0 dB"

1 ^— W

«^ = -90° Amplitude = .75

+1

-+ 0 -1

f

4> = -45'* Amplitude = {.75)x(1.4)

yo dB"

J V -/

V.

Time •

D Two independent noise sources at same level "OdB"

"+3dB"

+° Figure 8-8, Electrical summation of signals. Two equal signals of same polarity (A); two equal signals of reversed polarity (B); two sine waves of equal amplitude at a 90-degree phase angle (C); two independent noise sources of the same level (D).

ACOUSTICAL SIGNAL SUMMATION Let's now present each of these signal pairs in stereo. In the case shown in Figure 8-9A, two identical signals will appear as a phantom center image in stereo, and the level in the listening room will be approximately 3 dB greater than either channel alone. When the anti-phase pair is presented in stereo (B), there is no clear localization of the signal, but the level in the listening room will be approximately 3 dB greater than either channel alone.

Acoustical Signal Summation Identical signals of like polarity O O

o Listener

103 Identical signals in anti-phase

O

O

O Listener

Stereo image: precisely in center

Stereo image: unlocalizable

Level: about 3 dB higher than one channel alone

Level: about 3 dB higher than one channel alone

Identical signals with 90'' phase shift O O

o Listener

Independent noise signals at same level

o a

O Listener

Stereo image: wide center location

Stereo image: wide, natural stereo

Level: akx)ut 3 dB higher than one channel alone

Level: atjout 3 dB higher than one channel alone

Figure 8-9. Stereo image presentation and level of various signals. Identical signals (A); identical signals in anti-phase (B); equal signals shifted 90 degrees (C); independent signals at same level (D).

When the phase-shifted pair is presented in stereo (C), you will hear a "wide" center image, and the level in the listening room will be approximately 3 dB greater than either channel alone. When the two separate noise signals are presented in stereo {D\ the effect is a very broad sound front extending over the entire stereo stage. As with the other cases, the level in the listening room will be 3 dB greater than either channel alone. In all of these cases the stereo level in the listening room was about 3 dB greater than either channel alone. The reason for this is simply that acoustical loudness results primarily from the collection, or ensemble, of reflections in the listening room. In each of these four cases, the contributions from each channel were identical in amplitude, differing only in time domain characteristics between the elements of each signal pair. Since we are summing two

Chapter 8

104

equal signal levels, the acoustical power in the room will be doubled, resulting in a 3-dB increase in perceived level. Note that I have used the term "about 3 dB." Why not exactly 3 dB? The reason here has to do with the spatial distribution of sound energy density in the listening room, which tends toward an average value independent of the instantaneous polarity of the signals. This correlates well with what we hear in the room, and it is what we will read on a sound level meter averaged over the listening area.

STEREO SIGNAL CORRELATION The correlation between a stereo pair of signals is a measure of the commonality of the signals. As an example, two identical (or mono) signals will have a correlation coefficient of unity, or 1. If one of these signals is anti-phase the resulting correlation will be - 1 . If the two signals have no commonality whatever, then their correlation coefficient will be zero. We normally observe these relationships with an oscilloscope, as shown in Figure 8-10. The basic

Vertical input Horizontal / - \ Input

B

Left only

Right only

Stereo (largely uncorrelated)

Left = right

Stereo (with strong LF mono component)

Left = -right

Left and right equal at 90**

Stereo (with strong antiphase LF nrK)rK) component)

Figure 8-10. Oscilloscope patterns. Normal application (A); Lissajous figures for various signal combinations (B to I).

Stereo Signal Correlation

105

use of the oscilloscope is shown diXA, Here, a sine wave is introduced as the vertical signal, and an internal sweep circuit provides the horizontal signal. The resulting display shows the sine wave as a function of time, just as we observed it in Chapter 1. The polarity of the oscilloscope is vertical positive upward and horizontal positive to the right. In observing stereo signals on the scope we introduce the left signal at the vertical input and right signal at the horizontal input. For a left-only signal the display is as shown at B, and a right-only signal is shown at C. Identical signals (left = right) will produce the display shown at Z), and the same pair of inputs with an antiphase relationship is shown at E. If the stereo signals are identical with a 90^ phase shift between them the display is as shown at F. These displays are known as Lissajous figures. A highly uncorrelated stereo signal will appear as shown at G. Such a signal might be generated with a single pair of widely spaced microphones. A normal stereo signal with LF information panned to the center (in-phase) will appear as shown at //, and the same signal with anti-phase LF information is shown at 7. Oscilloscope displays are often tricky to read, but they contain a great deal of useful information. For most stereo recording or remix activities a correlation meter, as shown in Figure 8-11, will be easier to use. The meter operates by shaping both input signals, multiplying them, and then displaying the product of the signals as a fairly slow average value. The signal integration time is in the range of second or so, and the value indicated by the meter represents the short-term average program correlation. A normal stereo program will tend to hover around a zero value with occasional "excursions" into the positive area. Any signal that has exhibits a high degree O-

Wave shaping

Rlght input Q -

Wave shaping

Left Input

Averaging

Normal range for stereo

Figure 8-11. Details of a correlation meter for stereo.

Chapter 8

106

of negative correlation should be carefully analyzed for a possible anti-phase condition. Such a condition may normally be fixed by simply flipping the polarity of one channeL

ADDING ACOUSTICAL SIGNAL LEVELS If we have two acoustical levels, each of 93 dB, and we add them, the sum will be 96 dB. This is simply because the levels are equal and their sum will by definition be 3 dB greater than either one alone. When the values are different we can calculate their sum from the nomograph given in Figure 8-12. 10

2.5

1.5

1

0.9

0.8

0.7

0.6

Figure 8-12. Nomograph for adding signal levels in dB.

Take any two levels, such as 60 dB SPL and 65 dB SPL. Their sum can be determined by taking their difference, 5 dB, and locating that value on the line indicated D in the nomograph. Reading directly below that value you and get 1.8. Then, add 1.8 to the higher of the two original values, or 65 + 1.8 = 66.8 dB SPL. Try summing the values of 50 dB SPL and 60 dB SPL. Your answer should be about 60.4 dB SPL. Remember that when the difference between the levels to be summed is about 10 or greater, the resulting sum is very nearly equal to the higher value. If you are adding a number of individual levels, take them two at a time and sum each pair; continue until you have summed them all.

Chapter 9 RECORDING CONSOLES, METERING, AND AUDIO TRANSMISSION SYSTEMS

INTRODUCTION The console is the control center of any recording activity. It receives all program inputs from the studio, routes them through signal processing devices and assigns them to the various outputs. It provides for monitoring of its own output signals as well as the outputs of recording devices. The majority of consoles you will encounter are of traditional analog design, and they will be the focus of this chapter. Digital consoles will be discussed in Chapter 13. The earliest consoles were not much more than basic summing networks for a group of microphones which were fed to a single output channel. Eventually, engineers needed greater flexibility, and equalizers were added to each input channel. With the advent of tape recording it became necessary to include monitoring switching so that the engineer and producer could audition playback from tape in addition to monitoring the bus output of the console. Later, stereo recording demanded additional output channels, and engineers also required auxiliary outputs for sending signals to external devices such as reverberation (echo) chambers. It is at this point that our discussion in this chapter begins. First, we will cover some important fundamental concepts in signal transmission.

BASIC CONCEPTS Equivalent input noise (EIN) The noise floor of any console or audio transmission system is normally established at the "front-end" of the system and is the result of self-noise in the input resistance of the system. The noise arises from thermal agitation at the molecular level and is fundamental to all audio systems. A detailed discussion is given in Sidebar 9.1.

108

Chapter 9 Sidebar 9.1: Input noise depends on the input resistance of the circuit, the ambient tennperature and the measurement frequency bandwidth. The rms voltage is given by the following equation: -rms

= V4kRTDf

9.1

where k is Boltzmann'c constant (1.38 x lO"^^ joules per kelvin), R is the input resistance (ohms) and Df is the audio bandwidth. Typical values are 7 = 300 degrees K (80 degrees F), R = 200 ohms (typical of current studio quality condenser microphones) and Df = 20,000. These values give a thermal noise of about 0.26 microvolts rms, which is equivalent to -129.6 dBu. Today, studio quality input preamplifiers come within about 2 dB of this theoretical limit. If you look ahead to Figure 9-10 you will note that the microphone noise floor is just at this limiting value. A short-circuited input to the console would result in a noise floor about 10 dB lower, so the dominant noise floor in the audio chain is normally that of the microphone. The measurement of EIN is also shown in Figure 9-1. As we have seen, the noise floor of the microphone is normally expressed in terms of an equivalent acoustical noise level stated in dB(A). Source of EIN

Measurement of EIN

O

Low pass filter

-O

EIN =

e = VikRTAf' T In **Ke(vin k = Boltzmann's constant

Figure 9-1. Origin and measurement of equivalent input noise.

Special circuitry used in console design Figure 9-2 shows details of the operational amplifier (opamp). The opamp is universally used in audio distribution systems because of its flexibility in performing functions of signal addition, subtraction, and combining. The basic amplifier is shown symbolically a t ^ . It has both inverting and noninverting inputs and has a signal amplification of about 100 dB; its input impedance is very high and the output impedance is very low. When combined with external feedback resistors as shown at B and C, the audio bandwidth gain of the amplifier becomes a function of the resistance ratios. The circuit at B is

Basic Concepts

109

inverting while that shown at C is noninverting. The arrangement show at D acts as a combining amplifier, and is used when multiple microphones or other signals are combined into a single output bus. (The term bus is used throughout audio engineering to indicate the various output circuits of a console that are used to distribute signals to their various destinations in the control room, or throughout a broadcast or recording facility.)

rryi^ Rf*-Ri 60 = (e, + 6 2 + 6 3 + . . . . e n ) enCH—W^-*

Microphone

-F\

y

. -^

1 kQ

..

-A—AAA^

y

Output

Figure 9-2. Details of the operational amplifier. Basic element (A); inverting amplifier (B); noninverting amplifier (C); combining amplifier (D); balanced input amplifier (E).

The circuit at E has a balanced input and is often used directly as a microphone preamplifier.

Symbols and conventions in signal flow diagrams As you read various system schematic diagrams you will see elements such as those shown in Figure 9-3. These are defined as follows:

Chapter 9

no A. Line amplifier (gain often stated in dB) B. Variable gain line amplifier C and D. In-line faders (volume controls) E. Ganged faders F. One-in/two-out panpot (panoramic potentiometer) G. Signal processing module (ftmction normally stated) H. In-line termination resistor I. Line-crossing/intersecting conventions J. Line termination conventions K. In-line transformer L. Meter conventions

- ^

EQ

1

I

Either, depending on choice of 12 or 13

-o Non-intersecting

-D

-< Intersecting

L Q)

©

Figure 9-3. Conventions used in audio signal transmission diagrams.

Not all manufacturers use the same conventions, but "translations" among them are generally easy to make.

Basic Concepts

111

Patch bay conventions The patch bay section of a large console makes it possible to introduce external recorders or processors into an audio chain with a minimum of clutter and also to reassign console elements for greater user convenience. The word "jack" indicates either the plug at one of a patch cord or the receptacle into which it is inserted. A jack is shown at Figure 9-4A, as both circuit and symbol. A line termination is shown at B\ when a jack is inserted (as shown at C) the termination is lifted and the signal is sent onward. A normaledjack pair is shown at D, In this configuration there is continuity between the two jacks. An input and output of an external device can be inserted into the two jacks by lifting the normaled connection between them. The configuration shown at E is often called a half normal. A patch cord inserted into the upper jack will lift it, while a patch cord inserted into the lower jack will not. A portion of a typical patch bay is shown at F. The jacks are normally used in pairs, and signal flow is generally from top to bottom. For example, studio microphone receptacles appear at the top and are normaled into the console preamps. If you need to switch microphone positions on the console, it is often easier to re-patch in the control room than to reposition cables in the studio. In the middle set ofjacks, all of the console's insert points are shown. Any piece of line-level outboard gear can be easily patched into the system, as earlier indicated at D. (Line level refers to signals whose normal operating levels are in the range of 0 dBu. By comparison, microphone level signals are in the operating range of-40 dBu.) The bottom row of jack pairs connect multi-track outputs to their normal console inputs. On many occasions you will want to lay out these signal returns in a different order. You will usually find full-size patch bays located in a rack adjacent to the console. Mini-patch bays are normally located on the console's working surface itself.

Chapter 9

112 Schematic

Symbol

Shield

#r-|>-[>^" K>^^^^^M>-I>^^ Mix R Identic^

^ Aux 1 - 6

[^ P>K>ne8

Monitor Select Right Channel Identicat

CR Room •** MonOut Studio Loudspeaker Output

3+

"^=^MF/gwre 9-9. Monitor/master module, split configuration. Views (upper); signal flow (lower). (Courtesy Soundcraft)

Chapter 9

120

LEVEL DIAGRAMS Most console technical literature includes a signal level diagram. Essentially, this shows the normal operating level and noise floor as these values vary from input to output of the console. In the example shown, you can see that the noise floor is set at the input by the microphone preamplifier's noise floor. The microphone signal is amplified by nearly 70 dB at the earliest stages. It is reduced a nominal 10 dB by the input fader in order to give the engineer needed operating range. After that 10 dB has been restored in the following stage, the level varies only slightly from that point onward to the output of the console. Today, virtually all consoles follow the general plan shown here-but it is still possible to get into trouble: for example, when you operate the input fader too low when the input trim has been adjusted too high, you're likely to encounter overload at the microphone preamplifier stage. Thus the rule: try to keep the trims in the "comfort zone" between 10 o'clock to 2 o'clock. +27 dBu maximum Maximum ievel Headroom

Figure 9-10. Level diagram, split configuration console. (Data courtesy Soundcraft).

Setting Proper Gain Structure

121

SETTING PROPER GAIN STRUCTURE There are only three points in the input-to-output chain of the split configuration console where gain can be adjusted: microphone input trim, input fader, and group or bus output fader. We recommend the following procedure for establishing levels through the console: 1. Adjust each microphone input separately, with all others turned off. 2. Set the input fader to its nominal -10-dB position (this is some times labeled 0-dB on the fader scale) 3. Set the group output fader to its full-on position (normally labeled 0-dB). 4. With normal program input at the microphone, adjust the input trim control so that the level at the group output indicates normal modulation on the bus output meter. When you to this, you will find that the marker on the trim control will usually fall within the range from 10 o'clock to 2 o'clock. 5. If this is not the case, then check the feed from the studio. The musician may be playing very loudly, in which case you may need to switch in the microphone's output pad. Or, in extreme cases, you may want to use a less sensitive microphone. 6. If more microphone inputs are added to a given group output, you will have to reduce each one by trimming downward slightly. A good rule is to reduce all levels by 3 dB for each doubling of inputs fed to the same output group. 7. Once you have established a basic mix, feel free to make further gain modifications at the faders.

THE IN-LINE CONFIGURATION CONSOLE Today the in-line console is widely used in multitrack recording. In terms of layout, it does away with the output section of the console, integrating that function with the input in what is called the I/O (input/output) module. As a result of this, the overall console size can be reduced, but the console will be much more complex. The basic idea behind the in-line console is that, in the ultimate case, there will be one microphone or direct pickup for each track on the multitrack recorder. Therefore, why not provide a means of getting from the microphone

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to the recorder as simply and directly as possible, saving for postproduction all decisions regarding equalization, dynamics control, reverberation, and all other aspects of signal processing? During the tracking process, this is exactly what can be done. At the same time it is possible to monitor the recording with all the desired signal processing and make a rough two-track mix. Figure 9-11 shows a generic view and basic functions of an in-line console. Outside of the master section, the entire operating surface consists of I/O modules. The master section itself remains very much the same as for the split configuration design. GENERIC IN-LINE CONSOLE LAYOUT

INPUT AND OUTPUT CHANNELS

1

M A S T E R

INPUT AND OUTPUT CHANNELS

CHANNEL FADERS DIRECT TO TAPE (OR BUS ROUTE) MONITOR FADERS FROM TAPE

CHANNEL FADERS TO TAPE (OR BUS ROUTE) 1

MONITOR FADERS FROM TAPE

INPUT CHANNELS:

ADJUST INPUT GAIN AND SET CHANNEL FADER FOR LEVEL TO TAPE

BUS ROUTING:

DECIDE WHICH TAPE TRACK TO SEND SIGNAL TO - OR DIRECT TO TAPE

MONITOR FADERS:

MIX MULTITRACK CHANNELS TO 2-TRACK MIX

MASTER:

SET MIX LEVELS TO 2-TRACK AND SET CONTROL ROOM MONITORING LEVEL

Figure 9-11. The in-line configuration console, a conceptual view.

A LOOK AT THE I/O MODULE A simplified signal flow diagram for an I/O module is shown in Figure 9-12. For clarity we have omitted a number of functions, primarily the auxiliary send busses, since they are virtually the same here as in a split configuration console. There are two principal paths through the I/O module: the channel path and the monitor path, and they are clearly indicated in Figure 9-12. The module is described in detail:

A Look at the I/O Module

123

Figure 9-12, I/O module, basic flow diagram. (Data after Soundcraft)

Channel path A. Microphone or line inputs from the studio B. Transfer switch shown in normal position C. Signal processing functions; can be switched (via "swap mode") between channel path or monitor path D. Transfer switch shown in normal position

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Chapter 9 E. Fader for setting level R Transfer switch shown in normal position G. Transfer switch shown in normal position H. Output of channel path; normally sent to multichannel recorder

Monitor path J. Return from multichannel recorder; input level adjusted as required K. Transfer switch shown in normal position L. Fader for setting level (Note: AUX sends would normally be located adjacent to this fader; omitted for clarity) M. Panning of signal into stereo (bold lines in diagram indicate a stereo signal pair) N. Routing of panned signal to either stereo (MIX) or to surround (group busses) O. Output of monitor path; normally sent to stereo recorder and studio monitors R Alternate path to stereo mix (recorder and monitors) Q. Output to surround (or other) 8-channel recorder R. Return path from group to multitrack S. Return path from multitrack tape send to monitor return While the foregoing description may make the I/O module seem needlessly complex, remember that paths R and S and transfer switches (J5, D, F, G, K, and N) are normally operated in the positions shown. These "extra" controls allow the engineer to reroute signals for a wide variety of applications, including "bouncing" tracks on the multichannel recorder or overdubbing vocalists or instrumentalists. (Bouncing refers to an operation in which two or more tracks are combined and reassigned to a new track.)

APPLICATIONS The tracking session The best way to gain an appreciation of the in-line console is to observe it in operation. Figure 9-13 shows a single I/O module as it would be set up for a tracking session. In a typical application, there would be as many I/O paths as there were music sources in the studio. There is only one adjustable element in the channel path, and that is the channel fader, which is used to set the proper level going to tape. As many tracks as you care to use at this point in your project are available within the limits of your multitrack capability. If for any reason you preferred to equal-

Applications

125

Figure 9-13. I/O module set up for tracking session.

ize a track (for instance, a low frequency hum from a guitar amp) prior to going to the multitrack recorder, you could swap the assignable filter set to the channel path and correct the problem at this point. In the control room, you are primarily auditioning the outputs of all of the monitor channels. As you can see, the various signal processing modules have been switched to the monitor path, and you can make a "wet" monitor mix, complete with reverberation, equalization and limiting. None of this will be

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reflected in the multichannel feed—only in the stereo monitor mix in the control room. If you wish, you can record this mix for immediate playback and future reference. (Note: A "wef mix is one that includes reverberation, and by extension various equalization adjustments, as opposed to a "dry" mix, which is composed of basic tracks.) In a normal production environment, your experimental stereo mix would be reviewed by artists and producer, and any decisions for overdubbing or adding new tracks would probably be made at this point. Using the switching capability of the I/O modules and the added tracks available on the mutichannel recorder, these changes could be made, along with any new tracks the producer or artist may desire. When all of this has been done the project is ready for a final mix session.

The mixing session The console setup is shown in Figure 9-14. The multichannel recorder outputs are all assigned to the monitor path, and all signal processing modules are likewise assigned to that path. In addition to gain control and panning, equalization and dynamics control can be carried out at this point with those modules assigned to the monitor path of each I/O module. Reverberation sends via the auxiliary send busses can be returned to open faders on the console and can likewise be assigned through the channel return path. With this preparation, a final two-channel mix can be made. Alternatively, a surround sound mix can be made via the group busses.

Applications

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Figure 9-20. Metering. Comparison of types.

cal of stand-alone digital meters. Such meters can be configured to read the instantaneous value of a digital signal, and the exact level of a single digital sample can be observed. The normal steady-state calibration point for digital metering is -20 dBFS (20 dB below^/w// scale modulation). In summary, when the console outputs are calibrated at values of+4 dBu, the digital recorders will be set at -20 dBFS, and this condition requires that the console output sections have 20 dB headroom over the +4 dBu reference point, requiring a 24-dBu output capability. 41^ Mm ^•m'4A_--»m'»m.0»

1. Switch back to the "Gates" tool ("G" at keyboard), then use Zoom to Gates ( X - G ) to view a single segment nrnxe closely

2. Return to the "Fade" tool ("f" at keyboard). Select the leading cut of the segnr>ent you are examining

3. Drag the upper-right "grab-box" to lengthen the fade. Use the space bar to listen to the result Figure 13-6. Executing a musical fade. (Data courtesy Sonic Solutions)

Mixing with automation Anything that has been recorded so far can be mixed into stereo using the automation program. Level changes made during this operation can be stored, auditioned, and updated as desired.

DETAILED EDITING As opposed to pop/rock recording, where musical elements are generally added sequentially one on another and refined through punching in/out or through further overdubbing, classical music and film score editing consists of the assembly of many refinements, each taken from complete studio takes, or takes of sub-sections.

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The workstation environment, with its quick random access to all recorded takes or segments, offers the editor with a large array of options in refining edits: 1. Shape of incoming and outgoing fades. 2. Length of incoming and outgoing fades. 3. Relation of start and end of fades to the nominaledit point. The range of these variables is clear from the view of the Edit Fade Window, as shown in Figure 13-7.

[ Audition ) • l o c k Sound in Place Edit Point Offset: i-00:00:00:00.00{ DBuditlonBoth ^PouK^r I i>i:lc DAHgn Q Ripple Until Black Fade Template Enueiope Duration Ouerlap db douin Alpha [Untitled-Fade | ^Fade-Out Cosine 60.5000% 6.5 1.0942 60.50007« 6.5 1.0942 ]Fade-in Cosine I Delete | [ »^"te ] Nudge: ( Preu ) [ NeKt )

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JM Figure 13-7. View of the Edit Fade Window. (Data courtesy Sonic Solutions)

In classical recording, most edits fall into two categories: 1. Transition to another take: A "better" take of a long or short segment may be inserted as needed to correct a wrong note or any other musical detail the producer determines should be fixed. 2. Removal of very short segments in the recording, possibly a slight noise or a smeared attack. There is a good deal of general "tightening up" that goes on in all editing sessions. These are all possible with the parameter controls shown here.

207

Signal Processing

SIGNAL PROCESSING Audio signals can be subjected to virtually any kind of digital based signal processing in the frequency, dynamics, and time domains, with detailed setting of all parameters. Today, many plug-in options developed by third-party companies can be added to any DAW, extending its flexibility in ways not available in the analog domain.

NOISE REMOVAL Many work stations have sophisticated algorithms for removing noise from old sources as well as various ticks and pops resulting from electrical mishaps in the studio or in live recordings. Figure 13-8 shows an example of a Sonic Solutions program for interpolating waveforms. As shown in the upper panel of Figure 13-8, two clicks in the program have been identified, and markers are set so that the clicks fall between them. The lower panel of the figure shows Moue Zoom Play Mise SndFile Untitled-2

Clicks Gate4 Prior t« InterpoUtlon

Moue

Zoom Piay

Mise

SndFile Desk

Clicks Aft*r li»t*rMl«t«Mi

Figure 13-8. Interpolation of program waveform. Identification of clicks in program (upper panel); removal of clicks (lower panel). (Data courtesy Sonic Solutions)

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the resulting waveform after interpolation. The program waveform has been analyzed before and after the period of disturbances, and a new waveform has been estimated. Such a technique as this is possible over fairly short musical segments, taking advantage of the short-term linear predictability of music.

PROJECT MANAGEMENT A very useful capability of workstations is their ability to handle matters of house keeping. Tabulations of edits and other program changes are easily carried out and can be shown on-screen as in Figure 13-9. Here, the detailed timings and types of crossfades are stored in an editing decision list (EDL), making it easy to identify and alter specific edits. Another feature of modern workstations is the capability of editing and assembling masters for CD, DVD, and SACD manufacture, complete with documentation. :v^5;>cp»e.^cui:j^c^

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F*de-To-Bl*ck 00 00 00 00 00 00 00 02 23 04 F*de-Froi»-Bl»ck ghtmare Att%ck0001 Alien3tt5 2 F*de-To-Bl*ck 00 00 02 12 26 00 00 30 10 52 F«de-Fro«-Bl»ck ghtütre AttackOOOl AlienSHS 2 F*de-To-Bl*ck

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Figure 13-9. View of EDL listing window. (Data courtesy Sonic Solutions)

THE DIGITAL CONSOLE At some point in its expansion of capabilities, the traditional computer user interface reaches a performance limit. (You might think of it as trying to operate a modern vehicle with nothing more than a monitor and a mouse.) There

The Digital Console

209

are human skills that are most effective when a combination of tactile, visual, and auditory sensibilities are employed at the same time, and a digital console can provide an environment for this. Large digital consoles have been available since the early 1990s, and many of them have been little more than virtual replacements for modem inline analog consoles. Their greatest application however has been in the area of postproduction, where they interface directly with existing computers and high-density disc drives. Figure 13-10 shows the operating surface of a Yamaha DM 2000 digital console, which is typical of modem designs that can handle a variety of recording jobs from tracking to mixing. If you compare the console's work surface with the in-line console shown in Figure 13-11 you'll note that it is only about one-fourth the size—^but it has most of the operational capability and flexibility of the larger console.

Figure 13-10. View of a modern digital console. (Photo courtesy Yamaha)

A list of the capabilities of this console include: 1. Capability of processing digital signals at 24-bit/96-k sampling. 2. 24 line/microphone inputs, expandable to three additional layers for a total of 96 inputs.

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Chapter 13 3. Console architecture can be "designed" on-screen, as required for each job. 4. "Soft" (assignable) knobs for all signal processing functions and panning/assignment. 5. Absolute repeatability of all control settings. 6. Accommodation of third-party software plug-in modules. 7. Automation of all functions in remix mode. 8. Small monitor screen for showing patching routes and signalprocessing settings. 9. Accommodation of large-screen detailed graphics via an auxiliary minicomputer program. 10. Storage of multiple scenes for recall later. (Scene here is defined as a global group of console settings and routings) 11. Accommodation of a number of audio monitoring setups including surround sound.

LAYERS AND ERGONOMICS In a typical analog environment, a 96-input console would of course have 96 input strips, all equipped with the same signal processing. A complex mixing session involving this many inputs would have a number of pre-mixed inputs or effects, and the mixing engineer would not normally require immediate access to those faders or other input strip facilities. For effective mixing, the engineer would probably have no more than about 24 active controls in the immediate working area. Thus, the layering of additional inputs, accessible quickly when needed, makes good sense. But it requires a completely different mind set on the part of the mixing engineer. The move to such an operating environment as this will be a giant step for many persons to make, and the secret is to proceed cautiously with simple operations, moving on from there. One thing is clear: A large scoring session would be better done on a traditional in-line console. Another clear difference between analog an digital consoles is the appearance of the input strip. If you refer back to Figure 9-15, you will notice immediately the rich detail in which all functions are shown.

211

Layers and Ergonomics

AD Input Section

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By contrast, an input strip on the digital console, as shown in Figure 13-11, is virtually bare-bones. The analog input section (left panel) contains the following dedicated functions: 1. Phantom power on/off 2. Input pad on/off 3. Variable input gain set 4. Peak signal indicator light 5. Signal presence indicator light 6. Insert point on/off The corresponding channel strip section is shown in the right panel and contains the following generalized functions: 1. ENCODER: Rotary control used to edit Input and Output Channel parameters. Exact operation depends on currently selected encoder mode and layer 2. AUTO button: Used to set automix recording and playback for each channel. Exact operation depends on currently selected layer. 3. SEL button: Used to select input and output channels for editing with the selected channel section. Exact operation depends on the currently selected layer 4. SOLO button: Used to solo the channel 5. ON button: Used to mute input and output channels 6. Channel strip display: Graphic display of the value of the input or output channel parameter currently assigned to the encoder 7. Channel faders: Touch-sensitive motorized 100-mm fader used to set levels of input channels, output channels, aux sends, and matrix sends. Exact operation depends on currently selected fader mode and layer. You can clearly see that any engineer must attain a fairly high level of confidence in the use of a digital console before attempting even a simple mixing session—not to mention a tracking session.

Chapter 14 EQUALIZERS AND EQUALIZATION

INTRODUCTION The term equalizer is taken from early telephone engineering, when HF losses over long distances had to be compensated to "equalize" the sound at the receiver so that it matched that at the transmitter. The name has since been attached to any procedure of altering or adjusting frequency response in an audio chain. You will also encounter the iQrm filter, A filter is a specific type of equalizer that cuts or removes a portion of the audio program in an effort to fix a problem of some kind. The term program equalizer implies a device that is more flexible and that can be used to enhance a given audio program through the boosting or reducing certain portions of the frequency range. Equalizers may also be referred to by the nature of their action. For example, a graphic equalizer has vertical slider controls that can boost or cut specific frequencies, and when these controls are set in given positions the actual plotted frequency response curve will follow those positions. Shelving equalizers provide LF and HF boost or cut, which appears in the plotted frequency response as a shelf below or above the reference line. End-cut filters are used to provide steep cuts in LF and HF response, and a notch filter is used to remove a particular frequency, perhaps hum or HF leakage, from an audio program.

TYPICAL EQUALIZER FAMILIES OF CURVES End-cut filters Figure 14-1 shows a family of LF and HF response curves for a set of end-cut filters. The filter slopes are normally in the range of 18 dB/octave, which is generally steep enough to accomplish the removal of unwanted signals at the frequency extremes. The normal range of LF control may be from 40 Hz to perhaps as high as 160 Hz. The normal range of HF control may be from 5 kHz to 15 kHz. The frequency designation for the filter indicates the specific frequency at which the filter response is -3 dB. In modem console input

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sections, you will normally find a single LF cut filter fixed at 100 Hz. These are very useful in tracking sessions for removing any room rumble, air conditioning noises or other annoying thumps and the like. High-pass

Low-pass

Frequency (Hz) Figure 14-1. Typical end-cut low- and high-pass filter response.

Notch filters Figure 14-2 shows a typical notch filter response plot. A stand-alone set of notch filters may provide two sections, each individually adjustable over a wide range in both frequency and the degree of cut desired. A slight amount of 60-Hz hum may be removed by no more than 12 to 15 dB of cut, while an unwanted 1-kHz tone in an audio program could easily require upwards of 30 dB cut in order to be made inaudible. The major problem with notch filters adjusted for high amounts of attenuation is that they tend to "ring" in the region of the cut frequency. This produces a degree of coloration in the overall sound that may be objectionable. Use no more cut that necessary, and remember to bypass the filter when it is not needed.

Figure 14-2. Typical notch filter response.

Shelving boost and cut equalizers These functions are normally found in console input section equalizers at both LF and HF, and are useful in restoring the frequency extremes in audio programs. They are adjustable both in transition frequency and in amount of

215

Typical Equalizer Families of Curves

boost or cut available These equalizers are very effective in correcting for mild amounts of LF boost due to proximity effect with directional microphones and, at the other end of the spectrum, the differences between on- and off-axis microphone response. It is easy to over-use these equalizers, and you should be very careful making any adjustments when monitoring in a new environment or over an unknown set of monitor loudspeakers. Typical families of curves are shown in Figure 14-3.

Shelving boost and cut -rcxMjty

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40

60 90100

200

400 600 8001k

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Figure 14-3. LF and HF shelving boost and cut response.

Sweepable peak and dip equalizers These functions are found in many console input sections. Typically, there are two such sections, and each may be continuously varied, or swept, over a fairly wide frequency range. They are useful in making balance adjustments in individual tracks, and the maximum level range of such equalizers is about ±15 dB, although such extremes are rarely necessary). These equalizers can be used for purely creative purposes, or for correcting for a basic timbre (tone quality) problem, such as an overly bright or dull track. Typical families of curves are shown in Figure 14-4.

Parametric equalizers The three independent parameters in the setting of an equalizer section are the choice of frequency, the degree of boost or cut, and the sharpness of boost or cut. We have already illustrated the first two of these parameters, but the third

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may be new. Figure 14-5 shows the effect of the sharpness, or Q, of the boost or cut. When the boost or cut is broad the term low-Q is used. Conversely, the sharper response of these equalizer sections is referred to as high-Q. A typical high-end console will have two or three sections of parametric equalizers in each input module, and the frequency ranges of adjacent sections will have considerable overlap. The combination of three such sections, along with LF and HF shelving sections, will give the engineer just about all that is needed in making normal timbral adjustments. Peak and dip (sweepable) -f20clB

-flOdB

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Figure 14-4. Typical sweepable boost and cut response (with fixed Q values).

Parametric (effect of Q control)

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Figure 14-5. Parametric equalizer section, effect of Q control with constant frequency.

Complex Equalizer Response

217

COMPLEX EQUALIZER RESPONSE Recording engineers are normally concerned with the amplitude response aspect of an equalizer inasmuch as this defines the primary audible effect of the equalizen But associated with the amplitude response is a corresponding phase response. Normally, the engineer can ignore the phase aspect since the ear is relatively insensitive to it. Sidebar 14.1 discusses phase and time response of equalizers in greater detail Sidebar 14.1: Most of the equalizers in use today are of the minimum phase type; that is, they introduce the minimum amount of phase shift associated with a given amplitude change. As such, both phase and amplitude are reciprocal, and the "undoing" of a given amount of boost by passing the signal through a complennentary dip will "undo" the phase shift as well. This relationship is shown in Figure 14-6 where both amplitude and phase response are shown for a response peak (A) and a complementary dip (B).

A

B Amplitude Response

Amplitude Response

+4 dB

-4 dB Ptiase Response

Phase Response

+20

-20*»-

Figure 14-6. Phase and amplitude response of both peak and dip sections.

The phase shift of the signal is related to its relative delay by the equation: Relative delay = -d(|)/cl(o

14.1

Relative delay is expressed here as minus the rate of change of phase with respect of frequency; (^ is the phase shift in radians and co is the angular frequency, 271(1). In the example given here, the maximum amount of phase shift for the annplitude boost of 4 dB is 20 degrees. For a 1-kHz signal, 360 degrees represents one period, a time interval of 0.001 seconds. The effect of the phase shift would be to add (20/360)(0.001) seconds, or an additional delay of 5.5 x lO^seconds, to the 1-kHz signal.

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CREATIVE USES OF EQUALIZERS AND FILTERS Today, most applications of equalizers are likely to be creative rather than remedial, and the following list details some of these uses: Fullness may be added by boosting frequencies in the 100-300 range. This will be most effective on normally weak instruments, such as the acoustical guitar, celesta and harp. No more than about 4-6 dB should be necessary. A recessive sound can be made to project more if a broad peak is added in the 800 Hz-2 kHz range. Again, 4-6 dB should be enough. The articulation transients of many instruments may be highlighted by emphasizing the appropriate frequency range. For example, the acoustic bass has fundamental frequencies in the 40-200 Hz range, but its harmonics extend up to about 2 kHz. The sound of the player's fingers on the strings are nonharmonically related to the fundamentals, but in jazz performances they are often very important in defining the musical line. Adding a broad peak in the 1-2 kHz range will emphasize them. Likewise, the same approach can be used with the acoustic guitar by emphasizing the 2-A kHz range. Crispness in percussion instruments can be emphasized by adding an HF shelving boost above 1 or 2 kHz. Bongo and snare drums may also need similar treatment. Some cautions are in order: 1. Boosting and peaking should be done sparingly on metallic transients such as those produced by cymbals, tambourines, triangles and some Latin instruments. The HF output of these instruments is already strong, and adding more may cause problems in postproduction. 2. Never use equalization as a substitute for proper microphone placement. If a microphone needs to be changed or placed closer to an instrument, then by all means make that change. 3. Do not boost too many tracks in a multitrack recording in the same frequency range. Doing this will simply result in an unbalanced spectrum which is musically unsatisfactory. In pop and rock recording the goal should be to attain a fairly uniform overall spectrum from 50 Hz to about 8 kHz during full ensemble passages. More than any other area of signal processing, the creative use of equalization is learned by observing experienced engineers and through oldfashioned apprenticeship. If you listen carefully you will soon learn that the difference between "just right" and "too much" is often no more than a decibel and a half.

Digital Equalizers

219

DIGITAL EQUALIZERS One of the great benefits of digital signal processing (DSP) is the ease with which equahzation and filtering can be synthesized. Normally, you will not encounter stand-alone digital equaHzers, but you will find them nested in digital audio workstations (DAWs) or digital postproduction consoles. Many well known analog equalizer models from the past have been carefully emulated and are available as digital "plug-ins" for use in the postproduction environment. While we are used to knobs and switches to adjust the setting on an equalizer, the graphic user interface (GUI) of a digital realization of an equalizer generally offers a speedier way to arrive at a given response curve. The onscreen view of a four-section parametric equalizer is shown in Figure 14-7. As you can see, there are no knobs. When you want to make a setting change, you click on the parameter blocks at the bottom of the figure and enter the data you wish. An alternate way of data entry is to use conventional pointclick-drag techniques with a computer mouse. Each of the four equalizer sections is represented by a small white "handle" in the figure which can be moved along the frequency axis (for adjusting frequency), and up or down, (for adjusting the amount of peak of dip).

Figure 14-7. Graphic user interface: a 4-section parametric equalizer. (Data courtesy BSS)

In a sense, you are actually drawing the response curve that you want. In this example, filter sections 1 and 4 have been set respectively for low shelf and high shelf action, while the two middle sections have been set for typical MF peak and dip functions, each at different values of Q or width.

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An example of a digitally realized 5-band parametric equalizer with end correction sections is shown in Figure 14-8. The frequency response is shown at the top of the figure, and the on-screen GUI is shown below. Here, the user clicks on a given control in order to make changes. SONY

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Figure 14-8. Graphic user interface: a 5-section parametric equalizer with end sections. (Data courtesy GML and Sony Corporation)

The operational advantages of digital equalizers are: 1. They are space-saving; the equalizer is on-screen only when you wish to make a setting change or to view the response curve. 2. Curves and settings may be paged through rapidly. 3. Multiple settings for a given equalizer can be stored and recalled with precision.

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221

The disadvantages are: 1. Changes made "on the fly" are tricky and may need to be rehearsed. 2. You are "flying blind" much of the time; you can't simply glance over the console and identify how much boost or cut you have applied to a given input channel. In any event, digital equalizers will require a bit of getting used to that will remind you of your early days at the computer. Difficult at first, but speedy in the long run.

Chapter 15 DYNAMICS CONTROL

INTRODUCTION In this chapter we will discuss compressors, limiters, noise gates, and other signal processing devices that perform operations on the dynamic range of audio programs. The need for these devices comes from the fact that speech and music programs often occupy an overall dynamic range that is too great for their intended purposes. For example, live music almost always exhibits a dynamic range too wide for reproduction in the average home environment, and this has led to the general practice of signal compression and limiting during postproduction stages. While an experienced recording engineer can "ride gain" on a program manually, things can get out of hand very quickly. In broadcasting, there are times when no engineer is on duty, and it is to the station's advantage to maintain a uniform broadcast level. There is also the requirement in broadcasting that maximum signal modulation not exceed legal limits. In this chapter we will discuss various means of wide-band audio level control as well specialized tools for operating on specific portions of the audio spectrum.

ANATOMY OF A COMPRESSOR Figure 15-1 shows a block diagram of a compressor. The direct path between input and output is through a voltage controlled amplifier (VCA), whose control voltage is determined through signal processing in the side chain in the bottom portion of the figure. Program level is sensed, and a dc control voltage is produced that lowers the gain of the VCA as the input signal increases. Some compressors have input and output faders, as shown here. The input fader, since it is ahead of the side chain, will determine the amount of signal going to the side chain, thus determining the amount of gain reduction. The output fader acts only as a final gain adjustment for the device. The meter is switchable between the signal output and the side chain so that the engineer can read either the actual output signal level or the amount of gain reduction at a given time (this function is normally calibrated in dB).

Anatomy of a Compressor

223

Voltage controlled amplifier (VGA)

Attack time

Release time

Threshold Figure 15-1. Simplified signal flow diagram for a compressor.

The side chain functions labeled attack time, threshold, and release time determine the speed of the compressor's action and the program level above which that action will take place.

Gain curves Figure 15-2 shows gain curves for a compressor. The diagonal line running from lower left to upper right represents the constant gain of a normal amplifier. For each input signal increase there will be a corresponding output signal increase. Linear in-out gain curve

2-to-1 compression ratio

4-to-1 compression ratio

dB in

Figure 15-2. Typical gain curves for a compressor.

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A compressor operates very much like a linear amplifier at low signal levels, but when a predetermined threshold has been reached the compression action takes over and the overall gain is reduced. The point on the gain curve where the compression action begins is called the threshold of compression. (Some engineers refer to that point on the curve as the "knee.") The compression ratio is related to the slope of the gain curve in the region of compression. Several gain curves are shown in the figure. The twoto-one curves show that for each signal increase of two dB at the input, the output will only increase by one dB. The gain curve for a four-to-one compression ratio indicates that for an increase of 4 dB at the input, the output will increase only 1 dB.

Attack and release action During compression, the actual gain through the compressor is constantly varying, just as though an engineer was constantly manipulating a fader at the console. Such changes should not be made instantaneously, and if a gain setting is suddenly altered, the action will be quite obvious to the ear. The gain changes made by the compressor must be fast enough to catch sudden program peaks (attack time), but slow enough to allow a gentle return to the previous setting when the peak has passed (recovery time). The effects of attack and recovery time are illustrated in Figure 15-3. At A, you see a signal that suddenly increases in level and then later drops back to the previous level. When the input signal switches to a higher level (t|) which is within the range of compression, the gain of the compressor is reduced as shown at B and, after a slight amount of "overshoot," the compressed output signal drops accordingly. When the input signal returns to its original value {\j), the gain of the compressor is restored to its original value. You can see that both attack and release actions are not instantaneous; the attack time may be fairly fast, but the recovery time is relatively slow. The overall shift in compressor gain is shown at C Many compressors have user adjustments for both attack and release time, while other models have both fiinctions fixed internally. Attack times are normally in the range of 100 microseconds to 1 millisecond, while recovery times may vary from 0.5 second to about 2 or 3 seconds. While a very fast attack time would seem to be desirable, it often comes with the penalty that it can be heard as such. Most modem compressors have advanced circuitry that enables nearly instantaneous inaudible gain changes to be made. A zero-crossing detector can be used ensure that the gain change is made at an instant when the audio signal has a value of zero—^thus minimizing the audibility of the gain change as such. For special applications, some compressors delay very slightly the main signal path through the VCA portion of the compressor, while allowing the side chain to operate on the undelayed signal. If this is

225

Anatomy of a Compressor Input signal

Time

Compressed signal

B

^

1

1 1

Attack time

Recovery time

' '

t

1

_y^

V 1

Time

-..,

1

»•

Time

Figure J5-3. Compression action. Input signal (A); gain changes, showing effects of attack and recovery time (B); plot of compressor gain (C).

done carefully, the command from the side chain lowers the VC A gain before the program signal itself reaches the VC A, thus avoiding overshoot when the program peak occurs. Some "smart" compressors provide modified gain control based on the immediate signal history, For example, a sudden pause in the program would normally indicate a return to an uncompressed state in a conventional compressor design. A more sophisticated compressor would wait until the input signal resumed before making such a decision.

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226

THE LIMITER Basically, a limiter is a compressor with a built-in compression ratio often to one or higher, and with relatively fast attack and recovery times. The high compression ratio ensures that the program signal, once the threshold of limiting has been attained, will not increase substantially. Limiters are most often used to prevent accidental overload of transmission channels. For example, a limiter is the last signal processing element in a broadcast system, its chief function being that of preventing inadvertent overmodulation of the transmitter. LP discs are often mastered with a dedicated high-frequency limiter in the circuit for similar reasons.

MULTI-BAND COMPRESSORS The compressors we have discussed so far have been single-band; action takes place uniformly over the entire audio band, and this is ideal for compression of individual tracks. However, for more complex signals a multi-band compressor may be a better choice. As shown in the simplified signal flow diagram of Figure 15-4, the signal is divided into four adjacent frequency bands, and compression action is individually adjusted for each band. The advantage here is that heavy compression action in one band will not influence the gain in the other bands; this allows for greater overall program compression with minimum audibility as such.

Input

Multiple overlapping bands / LF YLMFVHMFV HF \

Output

—O

EH

Multiple threshold and other settings Figure 15-4. Simplified signal flow diagram for a multi-band compressor.

227

The ''De-Esser"

THE "DE-ESSER'' The de-esser is a special HF limiter that is used in vocal and speech recording to reduce the "splattering" effect of loud sibilant ("s" and "sh") sounds. Some singers and announcers have, for whatever reason, strong sibilants that can cause problems in HF overload in some recording chains. The primary frequency range of strong sibilant sounds is between 6 and 8 kHz, and the de-esser is designed to limit signals in that range. The threshold must be adjustable, but the attack and release times are normally fixed. A signal flow diagram is shown in Figure 15-5.

input

Output

o

O

/

t^''^^ \ i

t

Threshold setting Figure 15-5. Simplified signal flow diagram for a "de-esser."

CONTROL FUNCTIONS ON COMPRESSORS AND LIMITERS Some operational suggestions for setting up and using compressors and limiters are given below:

Input level control This control is of limited usefulness since it basically interacts with the threshold control. If more compression is desired, it is best to achieve it by readjusting the threshold control.

Threshold control For a fixed signal input setting, advancing the threshold control will cause the device to go into compression at progressively lower input signal levels. This is a critical adjustment and should be set so that the onset of compression will occur just as the signal is tending to become too loud or prominent in the program.

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Chapter 15

Compression ratio This control determines the departure from natural dynamic relationships existing in the program input signal. Low compression ratios will not materially detract from natural program dynamics; high compression ratios can sound quite unnatural. An experienced ear is required in making this setting.

Attack time The general rule here is to use as short an attack time as possible without having it become audible as such. In fast moving music, short attack times may be more appropriate than in slower music.

Release time This is perhaps the most subjective adjustment of all. It should be set so that there is no "breathing" or "pumping" that become audible due to modulation of the program's noise floor by rapid gain changes.

Output level control This control merely sets the signal level which feeds subsequent devices.

Metering The meter normally has two functions. One of these indicates the output program level, and this is useful in determining the maximum level through the device. The other function lets the engineer know how much gain reduction is employed at any given instant. Good engineering practice, and good taste, dictate that you should no more action than necessary.

Stereo ganging Many compressors can be ganged together to act in unison on stereo program. This ensures that there will be no image shifting due to unequal gain changes between the stereo channels. In this mode, both VCAs are controlled by the same dc voltage. For surround sound applications you will need as many ganged compressors as there channels in the system.

APPLICATIONS OF COMPRESSORS AND LIMITERS In recording, compressors have many uses, including the following:

Variations due to performer movements A performer who tends to move toward and away from the microphone can produce wide variation in level. A properly adjusted compressor can smooth out much of this variation, resulting a recorded track that can be more easily

Applications of Compressors and Limiters

229

processed later. Vocalists are likely to the be most problematic. The compressor should be inserted ahead of the input fader so that the engineer has wide control of overall level. The choice of compression ratio is a matter of taste; in general it should be low as possible, while still accomplishing your desired purpose.

Variations in musical output Variations in the output of an electric bass can be easily smoothed by the application of gentle compression, thus providing an even and solid bass line. If the recovery time is long compared with the natural decay rate of the instrument, then the original character of the instrument will be preserved.

Adjustment of release time In the preceding example, if the recovery time of the compressor is fast compared with the natural decay of the instrument, then the timbre of the bass will be transformed into a sustained, organ-like sound, exhibiting little of the instrument's natural decay characteristic.

Heavy limiting A similar effect can be obtained by applying heavy limiting with as short a recovery time as possible to cymbals. Heavy limiting implies that the input signal is always above the limiting threshold, so that the program will have a very low dynamic range. The effect is bizarre and often sounds like cymbal tracks played backwards.

Voice-over activities Voice-over compression is a method of causing background music or effects to diminish in level when an announcer speaks, allowing the level to return to normal when the announcer stops speaking. This is a special arrangement in which the signal to the side chain is derived not from the signal to be compress, but rather from the announcer's signal. Details of this are shown in Figure 15-6.

Program compression In many broadcast operations there is the need for long-term program compression in which programs from many sources need to be fairly well matched in overall loudness and consistency. This is a tall job for many compressors, in that the signals should ideally be first adjusted to the same reference level before compression takes place. There are special compressors (some including low-signal expansion to duck out noise) which address these specific problems, and they should be considered for these special applications.

Chapter 15

230

VGA

Program input o

Output —o ^

^

Threshokl

W

Level sensing

^ Voice " input Figure 15-6. A ducking circuit for voice-over activities.

NOISE GATES AND EXPANDERS An expander is basically the inverse of a compressor; it is used to increase the dynamic range of an audio program rather than decrease it. The basic form this takes is as downward expansion, an action in which low-level signals are made even lower. The noise gate is as typical example. Operation of the noise gate is shown in Figure 15-7. The gain curve is shown at A. The device acts as a unity gain amplifier at high levels, and this is indicated by the diagonal line with slope of unity. As the input level is lowered, the gating threshold is reached, and the gain of the amplifier is reduced, thus lowering the level of any noise in the input channel. Both the gating threshold and the range of gating are adjustable, as are attack and release times. Some models of noise gates provide for external gating, and this allows one signal to be gated on and off by another for special effects. For example, you could feed a steady signal, such as that of a wind instrument, through the direct path. The gating input could then be fed with a series of bongo drum beats. The output would then be a combination of the two, with the wind instrument being gated on and off with the envelope of the bongo drum.

231

Noise Gates and Expanders dB in

B Program en^

I i

Gating threshold

Time-

c5 O)

I Range of gating (dB)

c c5 O

Time

Figure 15-7. Operation of a noise gate. Output curve (A); setting the gating threshold (B); plot of system gain during gating (C).

Chapter 16 REVERBERATION AND SIGNAL DELAY

INTRODUCTION Acoustical reverberation chambers, often referred to as "echo" chambers, date from the 1930s and were used primarily for special effects in motion picture production. Their earliest uses in the record industry date from the late 1940s. These early systems were monophonic, and natural stereo perspectives were difficult to achieve, even with a pair of mono chambers. During the late 1960s, digital hardware was developed that would ultimately simplify reverberation generation, and electronic signal delay (often called "time" delay) devices became commonplace. Today there are many excellent reverberation and delay devices that take advantage of lower cost hardware as well as advanced internal programming.

A REVIEW OF EARLY DELAY AND REVERBERATION TECHNOLOGY The reverberation chamber was discussed in Chapter 1. While these relatively small rooms could produce a fairly natural decay as such, they did not simulate the normal initial time gap at the listener between direct sound and the onset of reverberation. As pop music relied more and more on reverberation, engineers felt the need for creating the initial delay, and a number of methods were used:

Tape delay A tape recorder running at 30 ips with a record-playback head spacing of about 1.5 inches produces a tape-delayed signal of about 50 milliseconds. The system was clumsy, and tape reels had to be replaced every 15 minutes or so.

Delay tubes For shorter delays, some engineers build delay tube systems. These consisted of pipes approximately 2 inches in diameter with a small loudspeaker driver at one end and a microphone at the other. It was important to include an acoustical termination at the microphone end in order to avoid reflections in

A Review of Early Delay and Reverberation Technology

233

the tube, and a well designed tube about 21 feet long could produce a delay of about 20 milliseconds. Eventually, narrow gauge coiled plastic tubing was used and the devices became relatively small.

Analog "bucket brigade" devices During the 1970s, several manufacturers used a Philips circuit element known as a charged coupled device (CCD) that stored instantaneous signal values. These signal values were sequentially clocked through the CCD, and a delayed signal was produced at the output. The systems were fairly noisy and required pre- and post-equalization to reduce the inherent noise level of the CCDs. These systems disappeared with the coming of digital technology. A number of mechanical "spring-type" reverberation systems were developed during the 1950s. These were virtually useless for critical studio applications, but they were very popular for use with electronic organs and other instruments. The first mechanical system to gain acceptance in the mainstream recording industry was the German EMT reverberation plate. It was introduced in the 1950s, and a stereo version followed shortly. Figure 16-1 shows a perspective view of the EMT 140 stereo model with one side panel removed. A Remote damper control Steel plate

Driving transducer Pickup transducers

Figure 16-L Perspective view of an EMT 140 reverberation plate.

234

Chapter 16

steel plate approximately 1 by 2 meters is mounted in a tubular frame, and its edges are undamped. The plate is driven into transverse vibrational modes, with multiple reflections taking place at its boundaries. When properly tensioned, the plate exhibits high modal density, w^ith especially good high frequency response. A moving coil driving transducer is located toward the middle of the plate, and two piezoelectric pickup transducers are positioned toward each end of the plate. On the back side (not shown in the figure) is a porous damping layer the same size as the steel plate which can be positioned over a range from about one-fourth inch to about 5 inches from the plate. Its purpose is to damp the acoustical field generated by the plate and allow its reverberation time to be adjusted. Figure 16-2A shows top and side views of the suspended plate, and typical reverberation time values for short, medium and long settings of the damping element are shown at B. The EMT units are still in use today, and as you progress in your recording career you will undoubtedly come across them. Regarding spring-based units, the Austrian AKG company introduced the very successful BX-20 model during the late 1960s. The unit consisted of two carefully constructed springs that had been "randomized" in order to diminish the effect of normal torsional modes. Driving and receiving transducers were placed at both ends of each spring. The springs were carefully looped over themselves and were long enough to generate remarkably uniform response with none of the "boinging" effect that had plagued all earlier spring systems.

A Review of Early Delay and Reverberation Technology

235

Side view

7^

^ Drive element

Pickup o

o Pickup o

k-

A Top view Damping membrane

B

Frequency (Hz) Figure 16-2. Details of the EMT plate. Side and top views (A); typical reverberation times for low, medium, and high settings of the damping element.

Chapter 16

236

INTEGRATION OF DELAY SYSTEMS AND BASIC REVERBERATORS The reverberation systems we have discussed thus far all require the application of external signal delay in order to produce the most natural results. Figure 16-3 shows a basic plan for stereo. The intent here is to create a natural impression of ambience for a single-channel signal. Follow the direct path from the top and you will see that the direct signal appears only in the left Direct signal in Q

Direct signal

Listener Figure 16-3. Use of delay values in conjunction with an analog reverberation system.

Integration of Delay Systems and Basic Reverberators

237

stereo loudspeaker. The direct signal is delayed twice (Delays 1 and 2), and these are fed respectively to the left and right stereo loudspeakers. Finally, the signal to the stereo reverberator (R) is delayed (Delay 3), and the outputs of the reverberator are fed to the left and right loudspeakers. If we were actually recording a sound source in a performance space, the signal at one of the microphones would resemble that shown in Figure 16-4A. However, we can create a reasonable facsimile of this, as shown at B. Properly delayed and adjusted in level, a single delay of a direct signal can create an impression of what happens naturally as shown at A. In the range of early reflections up to about 50 milliseconds there is a great deal of temporal masking taking place. We do not hear each reflection as such, and a single delay that produces approximately the same acoustical power as the early sound field will suffice if its relative level and delay are carefully adjusted.

Direct sound

(0 t ^ © m\

2ra

sound field I

I Reverberant sound field

I

50

100

150

Time (msec)

B Direct sound

Artificial reverberation

100

150

Time (msec) Figure J6-4. Direct, early and reverberant fields. Sound picked up by a microphone in a large room (A); a simulation of A using discrete delay and a single-channel reverberator (B).

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Chapter 16

In application, the value of Delays 1 and 2 would be about 15 and 25 milliseconds respectively. The value of Delay 3 would be in the range of about 40 to 60 milliseconds.

DIGITAL REVERBERATION SYSTEMS Figure 16-5 A shows the control surface of a typical digital effects system, and a general signal flow diagram for the reverberation algorithm is shown at B. Digital reverberation units are designed around a number of delay paths that simulate the actual dispersion of sound in real spaces. Various delay settings can be chosen, as can spectral characteristics and a host of other factors. Most models today are stereo; that is, there are two inputs and at least two outputs (some models have four outputs). Even with a single input signal, the stereo models will produce two uncorrelated outputs, similar to a reverberation chamber with one loudspeaker and two microphones for pickup. As you will see, digital systems do not require the use of external delay devices, since those functions are present in the basic digital system.

Fbk3

B

DryL

Lvl3 Dly3 Mix

Analog L ^
^

Reverberator Digital L

PDLY SHAP SPRD

Analog

SIZE XOVR WAND RTIM BASS SPIN LINK TDGY Lvl2

Digital I

•

DryR

^ i ^

'

^-^

[^D—f—KjH^-gU. Fbk4

Figure 16-5. Photograph of a digital effects system (A); signal flow diagram for the system's reverberation algorithm (B). (Data at A courtesy Lexicon)

Digital Reverberation Systems

239

A typical high quality reverberation system today offers the user control over many variables, including the following:

Program choice The user may choose among programs that are specifically modeled on physical spaces, such as concert halls, houses of worship, small reverberation chambers, or even reverberation plates. Within each program there may be other options, such as simulated room size and diffusion.

Predelay This allows the user to delay the onset of reverberation, usually up to a value of 100 milliseconds or so, in order to simulate the early time gap in a physical space.

Early reflections Pre-echo delays and level setting give the user further flexibility in simulating early reflections.

Low and mid-frequency reverberation time These controls enable the user to select different decay rates for low and mid frequencies. The transition frequency between low and mid can also be chosen, giving the user flexibility in simulating spaces with specific absorption characteristics.

High frequency rolloff This option lets the user determine the frequency above which the reverberant decay is quite rapid.

Decay shape Normal decay is exponential, but other options may be useful. For example, the decay can be programmed to build up in level before it begins to decay. Such a program variation might be useful in pop or rock recording as a special effect.

Mode density Some programs are calibrated directly in terms of room size, and increasing room size will increase modal density.

Wet/dry mix Normally, the reverberant signal is fed back into the console and mixed into the program by the engineer. In some applications, the dry (unreverberated) signal can be balanced with the wet (reverberant) signal at the output of the reverberation unit.

240

Chapter 16

SAMPLING REVERBERATION DEVICES The traditional approach in designing a reverberation algorithm is to analyze what physically happens in a given space, and then model that space through delays and various feedback paths that simulate second and higher order reflections. The user has direct access to many of these variables, and a given program can be "fine-tuned" by the user as required. A recent development is the sampling reverberator. In this approach, a room or performance space is actually sampled through the technique shown in Figure 16-6. A wide-range loudspeaker is placed on-stage and a set of spaced microphones are located in the audience seating area. A test signal is fed to the loudspeaker and picked up by each microphone. That data is recorded and converted into what is known as the impulse response of the room. In this case we have four such impulse responses, and, through a mathematical process known as convolution, an incoming dry signal can be reverberated as it would actually liave sounded at each of the microphone locations. The approach has great promise, and typical models may be sold with a library of sampled spaces, including some of the world's great performance venues, both indoors and out. There is no reason to think that sampling technology will replace conventional reverberation algorithms; the two will certainly coexist.

Sampling Reverberation Devices

241

Gathering room impulse response data PC Impulse response data Signal generator Digital recorder

V

Perfomiance space to be sampled

Amplifier

4 spaced microphones

o^

Wide-band loudspeaker

Adding sampled reverberation to a mono channel Mono input signal \j

^

Convolver

^ Four reverberated • outputs

li-npulse respons•e

data Figure 16-6, The sampling reverberator. Method of gathering room reverberation impulse response (upper figure); use of impulse response to create a set of reverberated signals from a single input (lower figure).

Chapter 17 SPECIAL TECHNIQUES IN SIGNAL PROCESSING

INTRODUCTION In this chapter we will discuss a number of signal processing techniques that do not fit neatly into the subject areas of the three preceding chapters. Some of the techniques are rather complex and may not be normally accessible to the engineer; however, the engineer should know how they work and what useful things can be accomplished with them. The following techniques will be discussed: phasing, out-of-band signal generation, pitch and tempo regulation, chorus generation, vocoders, stereo image manipulation and all-pass phase shift networks. The techniques discussed here have all had their basis in analog signal processing, although most of them now take advantage of digital technology.

PHASING (FLANGING) Phasing is a technique dating from the 1960s. Originally, it was done by feeding two tape recorders the same signal and combining their outputs. Any variation in the speed of one machine relative to the other results in a time difference between the two outputs, and the recombined signal exhibits comb filtering, which can be made to vary over a wide range. The basic phasing process is shown in Figure 17-1. At ^, the term T represents the fixed delay of each tape recorder and is the time gap between record and playback heads. The term At represents the difference in delay between the two machines and is the net value of delay that causes the comb filter response. The value of At can be varied electrically by driving one tape recorder with an external ac power source whose frequency can be shifted around 60 Hz. Another method of varying At is for the operator to place a thumb on the flange of the tape feed reel, thus slowing it down. This practice gave rise to the term "flanging" and is synonymous with phasing. The above techniques for phasing are cumbersome and introduce a fixed time delay into the signal path. So-called instant phasing is possible through

Phasing (Flanging)

243

the use of a delay system whose total delay can be varied in small steps over a wide range, or through the use of a variable phase shift network. When direct and delayed outputs are combined, the effect is very similar to that using the two tape recorders. The sound of phasing is hard to describe. It is most effective on broad band program material, such as cymbals and snare drum. It produces a bizarre "swishing" sound as the peaks and dips move up and down the spectrum. On vocal tracks, the effect often has a "disembodied and ghostlike" quality.

Input Tape recorder

Output * ( ^

1 s^ T±At Tape recorder

t Speed variation

B A^^ Af=1/At

HI >

^—n2 Input O

i

T3

Ik2

^^— n4

? ?

^ — ns

?

> ^ — n3

^»-T V \ \ )

Ikd

T4

^ T5

' k5

T1.. 5, values of time delay m . 5, modulating values of low-frequency noise k1.. 5, modulating values of low-frequency noise

Figure 17-7. Circuit for creating chorus effect from a single source.

Outp i^

Chapter 17

250

VOCODERS A vocoder is a VOice CODER, a complex signal processor that analyzes speech into its component parts and synthesizes it again out of these components. In the process of reconstructing speech from its components, substitutions can be made, and speech modulation can be added to a variety of musical or other non-speech sounds. Early work on the vocoder was carried out by Bell Telephone Laboratories as a means of reducing the data transmission rate for speech. The basic information rate of speech is fairly low, and if speech is broken down into its basic components it can be transmitted over narrow bandwidth channels for subsequent reconstruction. The basic layout of the classic vocoder is shown in Figure 17-8. Multipliers

>peech input

Band-pass filter 1 I

Rectifier and low-pass fitter

5?

Band-pass filter 1

Band-pass filter 2

Rectifier and low-pass filter

X

Band-pass filter 2

1 Band-pass filter 3

Rectifier and low-pass filter

X

Band-pass filters

\JL^ Speech

Band-pasd filtern

9^

Rectifier and [_ low-pass filter r

y®

output

Band-pass filtern

T

Excitation analysis

Excitation generator (pitch tracking, pitch offset)

A Other wide-t)and program sources may be introduced here

Figure 17-8. Simplified circuit for a vocoder.

The spectrum filter banks detect the presence of speech signals in their respective bands, and that information is converted into a group oidc control signals. These filter banks are responsible for detecting the formants of

Stereo Image Widening

251

speech, the basic vowel sounds such as "ah," "ee," "oh," and so forth. The pitch/excitation analyzer determines the fundamental frequency of speech and the presence of various "noise" components of speech, such as hisses, buzzes, and plosive sounds (b, p, d, t, etc.). Upon reconstruction, the fundamental is regenerated and the filter banks in the synthesizer portion of the system are gated according to the amount of signal originally present in each band. The excitation function restores the various noise components as required. Depending upon how many band-pass filters there are, and how sophisticated the pitch tracking and excitation functions are, vocoders can synthesize speech very naturally. More recent designs of the vocoder allow a wide variety of effects useful in electronic music and recording. The more sophisticated systems contain many filter banks for more accurate analysis and syntheses of the vocal spectrum. In addition, the pitch/excitation function provides a pitch offset function that allows a man^s voice to be transformed in pitch so that is may sound like a child's voice. Further, the pitch tracking may be replaced by an external source, thus enabling wide-band signal, such as noise or even music, to be speech-modulated! The vocoder is widely used in cinema work, where it seems ideal for special effects in futuristic films.

STEREO IMAGE WIDENING Many times, an engineer will be asked to make a previously recorded stereo program sound more spacious. There are certain conditions that might call for this. A stereo program may have been blended by mixing the two channels together to some degree; a previous engineer may have thought that the program had too much separation. Vocals are sometimes mixed too heavily relative to the instrumental background, and the resulting recording may seem to be center-heavy. Many of these problems can be fixed, or at least alleviated. The basic method is to provide LF cross-feed between the two channels with signals of opposite polarity. The signal flow diagram in Figure 17-9 shows how this can be done, and the operation can be set up on any console that has polarity inversion switching on the input strips. Caution is advised; do not use any more of the negative cross-feed terms than absolutely necessary to produce a mild spreading of the stereo image. As a practical matter, phantom center program can be reduced by no more than about 3 dB, while negative polarity terms will appear in opposite channels down about 10 dB in level. Listen carefully!

Chapter 17

252 Left channel in O

Right channel in

O Lo-pass shelf filter

Lo-pass shelf filter

Hi-pass shelf filter

fiyj^^^-^iNs Amplifier

2Z^ Loudspeaker

Amplifier

Loudspeaker

Figure 17-9. Circuit for producing stereo image widening at lower frequencies.

ALL-PASS PHASE SHIFT NETWORKS A passive all-pass phase shift network is shown in Figure 17-lOA. The lattice network provides flat frequency response but shifts the phase of the signal from 0 degrees at LF to 180 degrees at HF. Such a network can be used to change the crest factor of wide-band signals. For example, the signal shown at B clearly has higher peak values in the positive direction than in the negative direction. When this signal is passed through the network, it is modified as shown at C. The phase shifting does not affect the sound of the signal at all, but the resulting lowering of crest factor can be useful. For example, in broadcasting applications, the natural sound of many male announcers is rich in harmonics and has a fairly high crest factor. The crest factor may in fact be high enough to cause modulation problems which must be fixed. Rather than make use of limiting to contain the peak signal values, the insertion of an all-pass network in the chain may solve the problem far more simply, allowing an overall louder broadcast level for a given average signal level.

253

All-Pass Phase Shift Networks

input

R ^

Output

B

Figure 17-10. All-pass phase shifting. Passive lattice network for producing a phase rotation from 0 degrees at LF to 180 degrees at HF.

Chapter 18 FUNDAMENTALS OF STEREO RECORDING

INTRODUCTION Stereophonic sound, or stereo as it is usually called, refers to any system of recording or sound transmission using multiple microphones and loudspeakers. Signals picked up by the microphones are fed to loudspeakers placed in a geometrical array corresponding to the pickup zones of the microphone array. Many of the spatial aspects of the recording environment are preserved, and the listener can perceive the spatial perspectives of the original performance. Stereo is not limited to two channels. Motion picture systems have included six channels or more, but for home use stereo has traditionally been limited to two channels. Serious studies of stereo were undertaken in both England and the United States during early thirties, but it did not become a commercial success until the advent of multichannel motion pictures during the 1950s. About that same time, two-track stereo tapes for home playback became available, and in 1957 the stereo LP was introduced.

COINCIDENT MICROPHONE ARRAYS FOR STEREO One of the first stereo microphone arrays was developed by Allan Blumlein in the early 1930s. It consisted of a pair of figure-eight microphones positioned one on top of the other and rotated 90 degrees with respect to each other. This so-called ''Blumlein array" is shown in Figure 18-1. Both positive lobes of the crossed microphones are set at 90 degrees, and the arc between the positive lobes is aimed toward the players. For positions 1 and 3, sound localization will clearly be at the left and right loudspeakers since each of the sources lies along the major axis of one microphone and the null plane (at 90 degrees) of the other. A source positioned at 2 will be picked up equally by both microphones and will appear in the center of the stereo array. The pressure ampHtude of the sound picked up by each microphone will be 0.7 (-3 dB) relative to on-axis, and this corresponds exactly to the output produced by a panpot set for center localization (refer to Figure 2-9).

Other Coincident Arrays

255 Source 2

1 Source 1

Sources

Side quadrant

Back quadrant

Figure 18-1. The basic Blumlein crossed figure-8 array.

All sources located within the 90-degree pickup arc of the Blum lein array will appear in stereo playback uniformly positioned according to their location along the pickup arc. In addition to the exact "panning" of sources positioned in the front quadrant, the Blumlein configuration has the following characteristics:

Opposite polarity side quadrant pickup Since the two side quadrants are picked up by both positive and negative lobes of the figure-8 microphones, any sources of sound located at the sides will appear ambiguously in the playback array and should be avoided.

Reverberation pickup Since reverberation may enter the microphone array more or less equally through the back and the two side quadrants, it will contain both in-phase and anti-phase signals. The result is a very natural reverberant pickup with just a hint of localization extending outside the loudspeaker array. In actual use the Blumlein array requires careful judgement in positioning. It must be far enough from the ensemble to fill the front quadrant properly, but it must not be so distant from the ensemble that direct-to-reverberant relationship suffers.

OTHER COINCIDENT ARRAYS Crossed cardioids are often used in a manner similar to the Blumlein array. It is customary to spread the angle between the major axes out to perhaps 120 degrees in order to avoid too much pickup along the central axis of the

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256

array, which would produce a strong center, or monophonic, dominance. Supercardioid and hypercardioid microphones are often used in a similar manner, and these would be the natural choice in spaces that were too live for the Blumlein array. Examples of these alternate arrays are shown in Figure 18-2.

Figure 18-2. Other coincident arrays. Splayed cardioids (A); splayed supercardioids (B).

When cardioid, supercardioid, and hypercardioid microphones are used in a coincident array, it is common to splay them at the angles at which their overlap is -3 dB, relative to on-axis. This information is given in the table below. Pattern

Total angle between microphones for - 3 dB overlap

Bidirectional Cardioid Supercardioid Hypercardioid

90° 131° 115° 105°

STEREO MICROPHONES A number of compound microphones have been designed that include two capsule assemblies that can be individually adjusted in pattern and in their angular orientation. Such a design is shown in Figure 18-3. Stereo microphones of the kind shown here can be used for all coincident stereo pickup arrays. Most models have a control unit that allows pattern changes to be made remotely; capsule orientation is made manually at the microphone.

Mid-Side (MS) Pickup

257

Figure 18-3. Cutaway view of the Neumann SM-69 stereo microphone. (Photo courtesy Neumann USA)

MID-SIDE (MS) PICKUP Coincident microphones are very often manipulated by way of the MS* technique. MS employs a directional microphone (M component) aimed straight ahead, emphasizing the middle of the performing group. A figure-8 microphone {S component) is oriented at 90 degrees so that its two lobes emphasize side pickup.

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258

The outputs of the MS* pair are recorded but are not listened to directly. They are processed through sum and difference circuits to produce left and right signals suitable for normal stereo listening. The basic process is shown in Figure 18-4. You can see a t ^ that the M signal from the forward-oriented cardioid is added in-phase to each of the summing amplifiers. The side-firing S signal is added to the L^^^^ amplifier and subtracted from the R^^^^ amplifier. When the signals are summed in this process, the result is a net stereo pair of microphones as shown at B, one oriented left and the other oriented right. Ensemble

B Left

Right

Figure 18-4. MS recording. Microphone array setup (A); resolution into stereo (B).

When used in MS form, coincident microphones are extremely flexible. By varying the amount of the S component, the apparent width of the array can be altered electrically. Increasing the Mcomponent will narrow the stage, since M represents a mono center phantom image. Increasing the S component will widen the stereo stage, but too much S component will confuse the stereo imaging. A position control allows shifting the array left or right, as

Mid-Side (MS) Pickup

259

desired. These circuit modifications are shown in Figure 18-5, and a typical in-studio application is shown in Figure 18-6. Width control

Position control

Cardlold ( M ) a

OLout

ORout Flgure-8 ( S ) 0

Figure 18-5. Description of width and position control in MS" recording. Ensennble

Figure 18-6. Application of width and position control in MS recording.

Figure 18-6 shows how a typical MS' recording might be made. A main MS pair (1) provides the overall ensemble pickup. A secondary MS pair (2), which would be operated at a lower level and with reduced width, provides accenting of the middle of the ensemble. Using both the stereo width and position controls, MS' pair (3) could be used to highlight that part of the ensemble located at stereo right. When the MS stereo signals are summed for monophonic presentation, the S signal drops out completely, leaving only the M signal, as shown in Figure 18-7. The mono compatibility is excellent, and many broadcast engineers prefer it for this reason. Regardless of the specific pickup method they may use, some recording engineers prefer to take their stereo channels, convert them to MS, and then-

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260

convert them back again to stereo. While in the MS domain, the Mand S components can be separately adjusted so that the final stereo sound stage can be widened or narrowed as desired. This technique, shown in Figure 18-8, can be used for putting the finishing touches on the "spread" of a stereo recording.

Left and right patterns

Summation of left and right channels

Resultant forward-oriented cardioid

m V

(h

Figure 18-7. Mono compatibility of MS recording.

LinO OLout

ORout RinO Figure 18-8. Inserting an MS matrix into a stereo chain.

Figure 18-9 shows the relationship between various MS, or sumdifference, microphone arrays and their equivalent XT, or stereo, forms.

Implementation of MS Recording

261

Sum-djfference form

XY (left-right) form Left

I

Left

Right

Right

Figure 18-9. Various MS arrays and their corresponding stereo {XY) arrays.

IMPLEMENTATION OF MS RECORDING Implementation of MS normally requires a special sum-and-difference arrangement of transformers or amplifiers. Such systems are available as external processors often called "matrix boxes." If you are presented with an MS recording, and you do not have a playback matrix for it, you can improvise one at the console, as shown in Figure 18-10. This arrangement calls for a polarity inversion switch in one input channel, as shown.

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262

M input O Left output O Right output O

S input

Polarity reversal

RL

Figure 18-10. Resolving an MS recording on a console.

NEAR-COINCIDENT STEREO MICROPHONE ARRAYS Near-coincident stereo microphone arrays make use of a pair of directional microphones laterally spaced no more than about 1 foot (30 cm). The purpose of these arrays is to combine some of the excellent imaging of coincident arrays with the added sense of space or ambience that small displacements between microphones can produce. Since the 1970s, a number of specific techniques have been described in detail.

ORTF technique Developed by the French Broadcasting Organization, the ORTF technique employs a pair of cardioid microphones spaced about 7 inches (17 cm) apart and angled at 110 degrees. In detailed listening tests the system has rivaled the Blumlein array localization acuity, while at the same time oflFering immunity to excessive reverberation. The array is shown in Figure 18-11 A.

NOS technique This approach was developed by the Dutch Broadcasting Organization and employs a pair of cardioid microphones spaced about 12 inches (30) cm apart at an angle of 90 degrees. It has many of the advantages of the ORTF array, but with a shade more ambience. The array is shown in Figure 18-1 IB.

Faulkner technique This approach was first described by a British recording engineer during the 1970s. It works best in the recording of smaller groups in fairly live spaces. The figure-eights provide good rejection of excess reverberation, and the primary use of delay cues for localization gives a slightly "soft-edged" quality

Spaced-Apart Microphones for Stereo Pickup

263

17 cm (6.7 In)

B

ORTF Array

30 cm (11.8 in)

NOS Array

20 cm (7.9 in)

Faulkner Array Figure 18-11. Various near coincident arrays. ORTF (A); NOS (B); Faulkner (C).

to the stereo imaging. You may find that widening the spacing between microphones will enhance separation—but in no case should the spacing exceed about 2 feet.

SPACED-APART MICROPHONES FOR STEREO PICKUP The logical starting point for developing spaced-apart stereo microphone placement is shown in Figure 18-12. Here, we have a horizontal line of many microphones in front of a performing group. Each microphone feeds only to its corresponding loudspeaker in the playback space. As the figure clearly shows, there will be accurate wavefront reconstruction in the playback area, and an observer in that area should be able to hear stage events just as if that listener were located in the original recording space.

Chapter 18

264 STAGE SOURCE

— DIRECT SOUND PULSE SCREEN OF MICROPHONES ELECTRICAL CHANNELS VIRTUAL SOURCE SCREEN OF LOUDSPEAKERS INDIVIDUAL POINT-SOURCE SOUND PULSES SINGLE RESULTANT SOUND PULSE AUDITORIUM

OBSERVERS 1 PULSE TO EACH EAR • Figure 18-12. Waveform reconstruction using a horizontal line of microphones.

This is all fine in theory, and 1 have heard one instance where 16 microphones, spaced about a foot apart, were placed in front of a big jazz band. The recording was played back over 16 loudspeakers placed directly below the microphones. While the effect was certainly interesting, it was not all that accurate and hardly justified the expense required to carry it out. (Actually, the problems were more timbral than spatial; you heard instruments where they were located, but the sound quality suffered.) Taking Figure 18-12 as a starting point, the number of channels has been reduced to three, as shown in Figure 18-13. While this is far from ideal, it works much better than you might think. You do not normally hear "three distinct sound pulses" as the figure indicates, because the relative time delays between them are fairly short, well within the range defined by Haas (1954). Haas' experiments measured what had long been referred to as the "law of the first wavefront," which states that localization will tend toward the direction of the earlier arriving sound. Over a range of about 20 to 25 milliseconds, the ears will not normally detect a delay as such, and sounds arriving over that short interval will usually coalesce into a single impression for the listener. As motion pictures adopted stereo during the 1950s, the three-channel approach shown here was used. Three-channel magnetic tracks were adopted by the recording industry at the same, and when the stereo LP was introduced in 1957 the basic three-channel recordings were mixed down to two channels by feeding the center channel equally, and at slightly reduced level, to the left and right channels.

Spaced-Apart Microphones for Stereo Pickup

265

STAGE SOURCE DIRECT SOUND PULSE ^^TX

3 MICROPHONES 3 ELECTRICAL CHANNELS - 3 LOUDSPEAKERS

3 DISTINCT SOUND PULSES

AUDITORIUM

3 PULSES TO EACH EAR Figure 18-13. Waveform reconstruction using three spaced microphones.

This approach pretty much defined stereo recording in both popular and classical fields and has been the basis ofthat art up to present times. The variety of current spaced-apart microphone techniques for stereo is shown in Figure 18-14. The technique shown a t ^ is the direct use of the original threechannel approach we have just discussed. The approaches shown at B and C combine the advantages of spaced and near-coincident techniques. The socalled Decca tree shown at D was developed by the British Decca company in the early days of stereo and remains a popular approach today. In normal application, it specifically calls for five Neumann M-50 microphones, which are omnidirectional at LF and MF, but become increasingly directional at HF. The tree itself consists of three microphones aimed left, center, and right. The middle microphone is center-panned in the stereo mix while the other two are panned hard left and right. The center microphone is normally introduced into the mix at a level some 3 to 4 dB lower. The two outrigger microphones are spaced as best fits the music and are mixed hard left and right at a somewhat lower level than the microphones on the tree. When two spaced omni microphones are used to pick up a large group, there is a strong tendency to spread them too far apart. If they are farther apart than about six feet, then there will be a "hole" in the middle of the stereo stage. This can be easily filled in by using a center microphone of the same pattern and panning it in the center of the stereo stage. You will probably find that it will sound best when it is mixed in about 3 or 4 dB lower than the outrigger microphones.

266

Chapter 18

e

Omni

e

e

Omni

Omni

e

V

©

5

^^

Omni

ORTF

Omni

0.5 meter

5" Ail sutx^ardioids

Decca tree ^.

e

1.5 meter

ez IX V-"-^ 2 meters

e

Figure 18-14. Spaced microphone techniques for stereo. Three spaced omnis (A); two spaced omnis with center ORTF pair (B); two spaced subcardioids with near coincident array (C); Decca tree with outriggers (D).

Chapter 19 STUDIO RECORDING AND PRODUCTION TECHNIQUES

INTRODUCTION Most studio recording activities today are done multitrack and are intended for postproduction for record, video, or film formats. A professional, competitive studio has a variety of working spaces adjacent to the control room, including vocal booths and possibly larger isolation areas. The main room itself will often have variable acoustics, ranging from very dead, or damped, to relatively live. Isolation is a prime requirement in laying down useful tracks, and this can be accomplished through proper microphone placement and the use of isolating elements. In this chapter we will cover specific microphone techniques and procedures that are in everyday use in the industry. The techniques discussed here assume that all basic tracks are stored on multitrack recorders, with signal processing taking place only in the monitor mix and in later postproduction mixing operations. We will also assume that there are headphone monitoring facilities for the players, if required. A very important part of this chapter will be discussions of what tracks to lay down—^the basic decisions of what will be useful and necessary in postproduction.

SOME ISOLATION TECHNIQUES Figure 19-1A shows a wall construction detail you will find in many studios. Individual sections can be reversed as shown to expose either a reflective or absorptive surface, depending on the need. The view at B shows a studio with both live and damped areas. The damped area shown at the upper right would be an ideal location for an acoustical bass or a drum set as part of a small jazz group. A string section might sound better in a more reflective part of the studio, as shown at the left portion of the figure. There are two kinds of isolation requirements: relatively soft performers, such as vocalists, need to be isolated so that unwanted sounds will not enter

Chapter 19

268 A

Reflective surfaces

surfaces

B

^^m^mm^mmmmm

Reflective surfaces

m v////mv/////m^zm Absorptive baffles

Figure 19-1. Isolation in the studio. Wall detail with reflective and absorptive surfaces (A); live and dead areas built into the studio (B).

their microphones; loud performers, such as heavy brass and percussion players, may need to be isolated so that their sound will not interfere with softer performers. Often, both kinds of isolation must be used together. Many groups, such as small jazz combos, are pretty much self-balancing and require no special treatment. On the other hand, a jazz big band will produce levels that will easily swamp out an acoustical bass or vocalist. This is where isolation booths and direct pickups come in very handy, and you will see how these options are chosen as we proceed with this chapter.

Track Logistics

269

TRACK LOGISTICS In preparing for a tracking session, be aware that you are establishing a plan that will follow the project through to the end. Allocate recorder tracks carefully, and combine two or more microphones into stereo pairs when that makes good sense. Make sure that your machine operator is keeping an accurate log entry for every take. You will be making a stereo monitor mix as you go, and control room playback will normally be from that source. Analog tracking will give you a maximum of 24 tracks. Through sync-ing two machines you can reach a maximum of 48 tracks if there are separate time code tracks. This is a complex procedure and is not casually recommended. Digital techniques allow the use of multiple modular units, and track capability is virtually unlimited. Again, as things become more complex, the greater likelihood of logging mistakes and machine glitches. You will not likely be the chief mixer on a tracking session until you have done your apprenticeship as a second engineer. And at that stage you will learn the value of concentration and orderly bookkeeping.

PERCUSSION INSTRUMENTS; RECORDING THE DRUM SET The complex nature of the drum set requires that it be recorded in stereo, often heavily "spotted" with microphones. The modem drum set consists of the following elements, played by one person: 1. Kick drum (played by the right foot). 2. Snare drum (played with either sticks or wire brushes). 3. "High-hat" cymbal set (played with sticks or brushes and left foot). 4. Two "ride" cymbals (played with sticks or brushes). 5. Two or more tom-toms (played with sticks). The arrangement and actual number of individual elements in the set may vary from player to player, but the setup shown in Figure 19-2 is typical. Figure 19-3A shows a basic stereo.setup that would be appropriate for a jazz trio or quartet. There is an overhead spaced stereo pair along with a single kick drum microphone. The overhead pair will normally be small format condenser cardioids, and the kick drum microphone may be either a dynamic or condenser. A number of fairly low cost dynamics have been design specifically for kick drum use, and they are characterized by a slight LF rise in response as well as low distortion at high levels. Try both microphone types

270

Chapter 19 Cymbals

Hl-hat cymbals

Toms Snare drum

Kick drum

Figure 19-2. A basic drum set.

About 7 feet

"P^ Condenser

Dynamic

Figure 19-3. Basic pickup of the drum set (A); a more complex pickup (B).

and, if you have the track capability, record them both. While some engineers prefer a coincident pair for overheads, most use a spaced pair as shown, since it produces a wider stereo image. In larger jazz groups, or where the drummer is to be featured, it will be necessary to add a number of spot microphones, as shown at B, Use any or all that you, the producer, and the player deem necessary. You should also keep

Other Percussion Instruments

271

in mind that recording the drum set can take up many tracks, so lay out a plan for track assignments before you start. In some cases you will be able to combine certain microphones—as long as you keep in mind that, once grouped, they cannot be separated. Internal microphones should be located close to their respective sources, but they should be placed away from the player so that they will not interfere with the player's movements or be struck with a stick. Families of small clipon microphones, as shown in Figure 19-4. are available from many manufacturers and are useful in recording the drum set.

Figure 19-4. A small electret microphone that can be clipped to the rim of a drum. (Courtesy AKG Acoustics)

When closely picked up, spurious resonances in drum sets that would normally not be a problem become a matter of concern. Drummers are well aware of these problems, and they will usually solve them during set-up before the session starts. In fact, drummers are usually the first to arrive at a session so that these details can be worked out.

OTHER PERCUSSION INSTRUMENTS Among the tuned percussion instruments normally encountered are the xylophone, marimba, and vibraphone. Figure 19-5 shows a suggested stereo pickup for these instruments. Many engineers choose hypercardioid microphones

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272

for this application, aiming the major lobes of the microphones at the low and high extremes of the instrument. These instruments should be allocated a pair of tracks, or mixed into a stereo pair along with other instruments if the musical role is secondary.

/

%

Figure 19-5. Suggested pickup of mallet instruments.

Most of the non-tuned percussion instruments used in studio recording are fairly small, and they tend to radiate evenly in all directions. Cardioid microphones placed 2 to 3 feet overhead will usually give excellent results. Very often, a single player will be asked to perform on a number of these instruments, and the player should be made aware of what your particular pickup requirements may be. Many Latin percussion instruments (maracas, claves, gourds) are often picked up at short distances when played at moderate levels. Determine beforehand whether they should be recorded as a stereo pair.

THE PIANO Unless you are doing a classical date, you can assume that the piano will be picked up fairly closely, as shown in Figure 19-6. Details at^ and B show the piano picked up in stereo and recorded to a pair of tracks. For better isolation, position the instrument so that its open side points away from the other players in the studio. If the studio is small, and if ambient music levels are high, it may be necessary to record the instrument on half-stick and covered with a heavy blanket, as shown at C. Most engineers prefer to record the piano using large format condenser cardioids; however, omni condensers may produce a somewhat warmer sound. Experiment with both. You may find that considerations of isolation will favor the cardioids.

Vocalist

273 B

20 Blanket covering opening; piano cover on half-stick

10

•

PRI

Pan 1 left of center and 2 right of center for normal stereo perspective. Both microphones about one foot above strings. Figure J9-6. Recording the piano. Top view (A); front view (B); on half-stick with blanket (C).

VOCALIST If a vocal track is being laid down with a band, it is best to keep the vocalist in the studio—if you can get enough separation. Put the vocalist in a booth only as a last resort, or if the vocalist wishes to be there in the first place. The setup shown in Figure 19-7 is standard, and the microphone height should be set so that it will accommodate the vocalist standing or sitting. You will need a dedicated headphone mix for the singer, and it should include whatever instrumental tracks the singer desires, along with the singer's track—and with reverberation. Do not compress the vocal headphone feed, since it will only confuse the singer. Microphone about 20 to 24 Inches from vocalist

See-through t^ upper sections

Side view

Top view

Figure 19-7. Recording a vocalist. Side view (A); top view (B).

274

Chapter 19

Traditionally, vocalists have gravitated toward large format condensers, and they usually prefer the older tube models. There is a lot of mystique involved here, and you should do everything necessary to satisfy the singer. Professional performers will not attempt to use a handheld microphone, but you can expect just about anything from amateur performers. Some vocalists will prefer a so-called vocal microphone, perhaps one of their favorites used in stage performances. At times like these you'll have to rely on advice from the producer and the artist's manager. Use goboes as shown if you need greater isolation. If the vocalist has any tendency to "pop" b's and/?'s, use a mesh pop screen, which you can attach directly to the microphone. These devices are virtually transparent to sound and they do a very good job of reducing wind noises.

THE BASS (BASS VIOL OR ACOUSTICAL BASS) There are a number of methods of recording the acoustical bass: 1. Microphone placed on a low floor stand. 2. Nesting a microphone between the tailpiece and body of the instrument. 3. Microphone in front of the amplifier-loudspeaker unit. 4. Direct output from an instrument pickup located on the instrument's bridge. Methods 1 through 3 are shown in Figure 19-8. Picking up the bass via a microphone on a floor stand as shown at A is generally preferred because it picks up finger articulation on the string as well as acoustical output from the body of the instrument. In jazz recording, both of these ingredients are important. The method shown at B is useful under certain live performance conditions where the instrument may be moved around to some degree. The fixed position of the microphone relative to the instruments will maintain a fixed pickup level from the instrument. The method shown at C picks up sound from a loudspeaker and may be subject to noise and hum, as well as any distortion generated by the system. A direct line-in to the console from the instrument's built-in pickup is very useful in that it provides complete isolation from any other sound sources near the bass. The sound resembles that of a solid body bass under direct-in conditions. Make sure you have a good active direct box for this purpose. If you have the track capability, I suggest that you use methods 1 and 4. If you are restricted to a single track, record a mix of the two methods which you, the producer, and the performer agree on.

The Acoustic Guitar

275

Fingerboard

Baffle

Baffle Small microphone wedged in place with foam rubber between tailpiece and body

Tailpiece

'///y////////y//77?7////////,

v///yy/////?//////P>////////A Baffle

Guitar amplifier

'///////////TT////////. Figure J9-8. Recording the acoustic bass. Floor microphone (A); microphone mounted on instrument (B); picking up sound from the amplifier-loudspeaker (C).

THE ACOUSTIC GUITAR The acoustic guitar is a relatively soft instrument and may need to be closely baffled to achieve the necessary isolation for microphone pickup. Figure 19-9 shows a typical spaced stereo pickup. In some cases you can achieve a stereo pickup by using one microphone (left channel) and using the instrument's direct-out (right channel). If you opt for this method, place the microphone directly in front of the instrument. As with the bass, use an active direct box.

BRASS INSTRUMENTS Pickup of individual brass instruments is shown in Figure 19-10. When played loudly, all brass instruments produce considerable harmonic development, and the higher harmonics are quite directional along the axis of the bell. A microphone placed about 3 feet in front of a trumpet may pick up an unnaturally bright sound. You may want to roll off the HF a bit, move off-axis or select a microphone whose HF response is rolled off. Older model ribbon microphones are quite popular for this application. The players themselves

276

Chapter 19 Direct out

U

As seen from above

iiiiii'lii'i'liii,i|i|i,'ii

^ ^

Distance: 8-15 Inches Spaced microphones at)out 1 5 - 3 0 inches from guitar txKly Figure 19-9. Recording the acoustical guitar,

often prefer these older microphones, since the recorded sound is more like what they hear while playing. Don't hesitate to experiment with various models and types of microphones. The French horn should never by picked up along the bell axis. As shown in the figure, place a microphone at 90 degrees above the bell, pointing downward. Very high frequencies diminish quickly at 45** off-axis

B

^ 40 In

< - ' Sound on-axis Is very

--^O

bright; consider using ribbon microphone

Baffle no closer than about 40 inches

>^ardloid microphone aimed downward from above. approximately 90'' off-axis ' of the bell. Distance 40 - 80 inches

Figure 19-10. Recording brass instruments. Trumpet (A); trombone (B); French horn (C).

Woodwind Instruments

277

WOODWIND INSTRUMENTS Sound radiation from the woodwinds is fairly complex due to the patterns of tone holes on the bodies of the instruments. Figure 19-11 shows some of the techniques that are used. In studio recording there should be no problems in putting a microphone just where you want it. However, in many live tracking situations it is customary to clip a small wireless microphone onto the bell of the instrument. If this is a requirement, then be prepared for a sound that will require a lot of equalization in postproduction. A

Good balance

B

yP

"Breathy"

'12-20 In

X

Good balance

?

12-20 In

Bass heavy

Good balance V2 - 20 In

Figure J9-11. Recording woodwinds. Clarinet-oboe (A); flute (B); saxophone (C).

STRING GROUPS Because they are relatively soft instruments, strings are almost always used in multiples and a natural stereo pickup is traditional. In popular recording, they are used primarily in large studio orchestras with the following quantities: 6-8 players, 1st violins 6-8 players, 2nd violins 4-6 players, violas 4 players, cellos There is usually one bass, and it is picked up separately from the rest of the ensemble. Figure 19-12 shows details of picking up the string ensemble.

Chapter 19

278 2nd Violins

I St Violins

Cellos

Figure J9-12. Recording a string ensemble.

In arraying the players, allow a space about 5 feet on a side for each stand (pair) of violins and violas. Allow a space about 6 feet on a side for each stand of cellos. The height of the microphones over the instruments is fairly critical. If it is too close, the sound will be edgy. At greater distances, these problems are solved, but then we then face the added problem of excessive leakage into the string microphones from the louder instruments in the studio. Many times the best way to solve this problem is to record the louder instruments on one date, adding the strings later through multitrack overdubbing. Some large studios have the luxury of an adjacent space large enough to isolate the strings from louder instruments, while allowing eye contact through large windows directly into the control room. Headphone monitoring is essential in these cases.

ELECTRONIC AND AMPLIFIED INSTRUMENTS In this section, we will deal with all instruments whose sound is generated or amplified for presentation over a loudspeaker. These include solid body guitars of all types as well as synthesizers. In most cases a feed directly from a pickup on the instrument can be used. It is essential that your direct feed include any signal processing that takes place in the instrument's electronics. If the instrument has stereo outputs, be sure to record both of them. In some older guitar amplifiers the processed output may only be available at the loudspeaker. Try tapping the signal off the voice coil leads; if this works, then use it. If not, you must place a microphone at the loudspeaker, as discussed earlier.

Session Planning and Track Allocation

279

SESSION PLANNING AND TRACK ALLOCATION Remember that the final mix will be made later. New musical ideas may come along in the meantime, and the producer and artist may want to experiment. Plan the session and identify the separate tracks that you know will be needed. Try to avoid "running out" of tracks by using the largest recording format that the budget will allow. The following examples illustrate these points.

SMALL JAZZ GROUPS Assume that a typical jazz trio of piano, bass, and drums is to be recorded for stereo release on compact disc, and possibly in surround as well. Even at this point, we must have some conception of what the stereo stage will be, and both producer and artist will have ideas of their own.

A jazz trio A good plan here would be to place the most prominent instrument in the center, with remaining instruments in flanking positions. A suggested studio layout is shown in Figure 19-13. The piano is picked up as a stereo pair and panned inward as needed so that it occupies the center one-third or one-half of the stereo stage. Drums will picked up with an overhead pair as well as a microphone on the kick drum. The overhead pair can then be panned so that the left signal appears in the center of the stage, with the right signal at far right. The kick drum would best be panned center, for purposes of mono compatibility and FM broadcasting. The bass, picked up by a single microphone or by direct pickup, should be panned slightly in from the left in order to maintain overall musical and stereo stage balance.

n

Drums

Direct out

Use short goboes

Use short goboes

Figure 19-13. Studio setup for a jazz trio.

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The overall intent is to feature the piano, with both drums and bass as musical support functions. Ambience (room) microphones may to be used if the studio is large enough to provide a good reverberant field that can enhance the recording. These microphones should be positioned fairly high and widely spaced in the studio and away from the players. While you may not be able to determine their absolute necessity during a tracking session, the mixing engineer may find them very valuable. In order to cover all postproduction possibilities, the track assignments might be as follows: 1. Piano close (left). 2. Piano close (right). 3. Bass (microphone). 4. Bass (direct in). 5. Drums (overhead, left). 6. Drums (overhead, right). 7. Kick drum. 8. Room pickup (left). 9. Room pickup (right). Obviously we need 16-track capability, and this leaves additional tracks for future overdubbing. Or, you might want to put additional microphones on the snare drum and highhat cymbals. Ask yourself the question: Do we really need a 16-track recorder to make this recording? In many cases the answer may be no. During the early digital era, hundreds of jazz recordings of small groups such as this were made and mastered directly to stereo. In other words, the monitor mix was the final recording, and all balances made at the session were locked in. The rationale for doing it multitrack and remixing later is basically one of musical flexibility. A year after the CD is released, a film producer might want to use a portion of the CD in a film, and the multichannel original would be an advantage. And don't forget that a surround mix may be needed at a future date.

Piano, vocal, and bass Here, the vocalist would be centered, with piano and bass moved out to flanking positions. Stereo reverberation is essential for the vocalist, but it should be subtle and in good perspective. A suggested studio layout is shown in Figure 19-14, and track assignments are: 1. Piano left. 2. Piano right. 3. Vocal (dry). 4. Bass (direct).

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281

5. Bass (microphone). 6. Reverb return left. 7. Reverb return right. Direct out

Use tall see-through

goboes

Use short goboes

Vocal Figure 19-14. Jazz trio with vocal.

MEDIUM-SIZE JAZZ GROUPS A basic rhythm section for a medium size jazz ensemble consists of piano (or organ), bass, drums, and possibly guitar. Against this background, up to three wind instruments are often used in instrumental arrangements, or as backup for a vocalist. There are a number of stereo stage layouts for a group like this, and the selection of tracks must provide for them all. A suggested studio layout is shown in Figure 19-15, and track layouts are: 1. Drums (overhead, left). 2. Drums (overhead, right). 3. Kick drum. 4. Bass (direct). 5. Bass (microphone). 6. Guitar (microphone). 7. Guitar (direct). 8. Piano (or organ) (high). 9. Piano (or organ (low). 10. Vocal. 11. Sax 1. 12. Sax 2. 13. Room ambience (left). 14. Room ambience (right).

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282 Direct out

Drums

Short goboes

1 Short goboes Sax 2

Saxi

Guitar Use tail see-through goboes

Vocal

Short goboes

Figure 19-15, Jazz group with seven musicians.

A larger group might consist of nine musicians: vocal, piano, bass, guitar, trumpet, sax 1, sax 2, guitar, drums, and synthesizer, as shown in Figure 19-16: 1.Vocal. 2. Piano (high). 3. Piano (low). 4. Bass (microphone). 5. Bass (direct). 6. Trumpet. 7. Saxl. 8. Sax 2. 9. Guitar (microphone on body). 10. Guitar (direct). 11. Drums (overhead, left). 12. Drums (overhead, right). 13. Kick drum. 14. Synthesizer (left). 15. Synthesizer (right). 16. Room ambience (left). 17. Room ambience (right).

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283

Bass

Direct *0"*

n

Drums

1

T.

Short goboes Sax 2

Guitar Direct out

Use tail see-through goboes Direct outs

r r>

Vocal

Short goboes

Synthesizer Figure 19-16, Jazz group with nine musicians.

The above plan allows piano, guitar, synthesizer, and drums to be picked up in stereo. This is important because they are all polyphonic sources; that is, they can all produce more than one musical line or sound at a time, making it possible to give them a natural "spread" on the stereo stage. The vocal, trumpet, and saxes are all monophonic sources, producing a single note at a time, and thus only need a single track each. With these track assignments the producer and mixer will have many options to choose from in creating final mixes. This track listing also illustrates a problem that will occasionally arise: with 17 suggested tracks you are required to use a 24-track recorder. If the budget allows it, that's fine. However, if this creates a problem you'll have to remove a track. The obvious choice here to go to a single bass track, creating a mix of microphone and direct pickup that you and the producer will feel comfortable with later on.

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THE JAZZ BIG BAND The usual makeup of this ensemble is: 4 trumpets 4 trombones 5 saxophones (1 baritone, 2 tenors, 2 altos) 1 bass 1 drum Set 1 guitar 1 piano This basic ensemble may be augmented in certain arrangements by added percussion players, French horn, electric organ or synthesizer, and possibly one additional player each in the wind instrument sections. The saxophone players may double on woodwind (clarinet, flute, or oboe), as required. We will focus our attention only on the basic ensemble. A typical studio layout is shown in Figure 19-17, and a suggested track layout is: 1. Drums (overhead) left. 2. Drums (overhead) right. 3. Snare drum. 4. Highhat cymbals. 5. Kick drum. 6. Piano left. 7. Piano right. 8. Bass (microphone). 9. Bass (direct). 10. Guitar (microphone). 11. Guitar (direct). 12. Sax 1. 13. Sax 2. 14. Sax 3. 15. Brass left. 16. Brass right. 17. Solo trumpet. 18. Solo trombone. 19. Ambience left. 20. Ambience right.

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285

? ? 1 Bass

Direct In

J

I Direct in

Guitar {

Figure 19-17, The jazz big band.

At the start of the session, the major challenges to the engineer are: 1. Optimizing levels to the multitrack recorder. 2. Setting up a monitor mix. 3. Providing headphone mixes for musicians. These three tasks must be accomplished quickly. In most cases you will be working with the "home crew" in a studio, and they will be able to estimate settings based on previous sessions. You will very rarely be working "from scratch." Let them make the initial level settings to the primary multitrack recorder. This crew will also have a good idea of headphone routings and levels, and again give them that assignment. The monitor mix is your sole responsibility, and you can start working on it as soon as any musicians arrive before the session begins. As a starting point, you will know beforehand the stereo panning layout, and in most cases

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this will be virtually the same as the left-to-right physical layout of the players in the studio. If you make a mistake or two, there is nothing to worry about. Set levels first, and then consider any reverb feeds that you will need. If there is a vocalist on the date, pay special attention to those requirements. It is imperative that the vocalist hear a near-perfect mix at the first playback break. Many large scoring studios have an auxiliary console located in the studio. This console has parallel feeds from your basic tracks, and an assistant engineer assigned to this job takes the entire responsibility of setting up phone feeds for the players.

THE LARGE STUDIO ORCHESTRA The studio orchestra considered here is the typical ensemble that would be used for recording the score for a TV or motion picture soundtrack. While it appears symphonic in its overall size, the musical demands may vary from jazz performance to massed string writing in a classical context.

COMPOSITION OF THE STUDIO ORCHESTRA A typical studio orchestra setup is shown in Figure 19-18. A group such as this would be used for film or video scoring and may be augmented with additional instruments as well as with a jazz rhythm section and possibly a vocal chorus. Synthesizers are very commonplace today. By symphonic standards, the string section is usually abbreviated to include the following: 8 1st violins 8 2nd violins 6 violas 5 cellos 1 bass viol

Track Assigments

287 Synthesizers

Percussion

X X X X

Timpani

• w

/ Horns

Brass

Tuba

rrrumpets Trombones

•

/

Piano

Woodwinds Clarinets# # Bassoons Flutes # # Oboes

2ndViolins\

Internal strings

^^^^

Conductor

Flank

Flank

Main pair i

Legend

1

• Main microphones # Spot microphones 1 X Direct in

Figure 19-18. The large studio orchestra.

TRACK ASSIGNMENTS Multitrack recording is the norm here, because of the demands for postproduction flexibiHty. Sound tracks are usually mixed with effects and dialog, and individual instruments may need to be emphasized. An important consideration here is how the tracks are to be used in postproduction. The engineer should know which elements will require the most flexibility in rebalancing, and ensure that those elements are on tracks of their own. Conceivably, the entire string section could be recorded on a single stereo pair of tracks, if the postproduction flexibility required only raising them and lowering them as a group. On the other hand, a lead vocal must remain isolated from background vocals, and certain critical rhythmic elements must remain separate. When mixed behind dialog or sound effects, the engineer often has to reach for musical details or thin out textures if they are to be easily heard. In live recording events, a pair of tracks may have be assigned to pick up audience reactions.

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VGA subgrouping, or automated fader subgrouping, will simplify the monitor mixing of a complex session with a studio orchestra. For example, the entire string ensemble, with all stereo and panning assignments, can be grouped under the control of a single fader. The rhythm tracks may be similarly grouped. Within such grouping, an individual microphone inputs may of course be adjusted as required.

ANTICIPATING BALANCE PROBLEMS The single greatest difficulty in recording a large studio orchestra is keeping the sounds of the louder brass instruments from swamping out the softer string instruments. If there is a string isolation area, then you will have few problems; otherwise, you may have to resort to close string microphone placement. Avoid being so close that you run the risk of getting a strident sound; also, use the smoothest microphones you have—and don't hesitate to shelve out some HF above about 6 kHz. Generally, a competent arranger will help you avoid these problems.

KEEPING THE TAPE LOG SHEET A poorly documented session can be a nightmare to sort out later. A log entry should be made just after a machine is put into record mode and should include the following: 1. Title. 2. Take number. 3. Time Code reading at start. 4. Time Code reading at end. 5. Identification of take as a complete take, false start (FS), breakdown (BD), or any other abbreviations the producer and engineer wish to use. 6. Track content. The recorders are not usually stopped after a false start, and the assistant should enter the new time code on the log sheet when the music commences, whether or not a new take number has been assigned. Data to be entered at the top of each log sheet should include: location, date, identification of artists, producer, and engineers, and any project numbers that are pertinent. A copy of the log sheet usually is permanently attached

Keeping the Tape Log Sheet

289

to the tape box and becomes the official record of the session proceedings. Don't forget to label on the tape reel, cartridge or hard disc drive itself. Any backup copies of the session must be correctly labeled as well. The recording engineer may occasionally ask the assistant how much time is remaining on the recording medium. With a knowledge of time code and the medium itself, the assistant should be able to estimate the time remaining to within a few minutes. Multitrack sessions require extra work in that the content of each track must be indicated on the tape log. All details of overdubbing and track combining must be scrupulously documented. The producer may or may not keep detailed notes, and may rely completely on the accuracy of the assistant engineer in this regard.

Chapter 20 CLASSICAL RECORDING AND PRODUCTION TECHNIQUES

INTRODUCTION In this chapter, we will discuss the musical and technical factors involved in producing classical recordings for commercial release. We will discuss the selection of a recording venue, planning the sessions, placing the microphones in order to produce a desired recording perspective, and details of equipment and staffing.

THE COMMERCIAL RECORDING ENVIRONMENT Role of the producer As a professional classical engineer you will almost always be working with a producer. The producer may work directly for the record company, or be an independent agent engaged by the company for a given project. The producer's responsibilities may include any or all of the following: 1. Preparing a budget for the sessions and ensuring adherence to that budget. 2. Working with the artist or conductor in planning how the sessions will run. For example, does the artist or conductor feel comfortable only with long, complete takes, as in actual performance? Or, is the artist willing to work with numerous shorter takes? 3. Determining the sonic aspects of the recording. In practice, this is a joint responsibility of both producer and engineer, and many producers rely heavily on the advice and expertise of engineers they have successfully worked with. Included here are such important points as details of stereo placement of instruments and the balance of direct and reverberant sound pickup. It is essential that the producer and engineer have virtually identical conceptions of how a given recording should sound if the sessions are to be productive. 4. Musical supervision of the sessions. This involves studying the score with the artist and/or conductor well ahead of the sessions so that both will have the same goals. The producer communicates directly

Staffing

291

with the conductor on stage via the talkback system and keeps detailed notes of which parts of the music have been covered during the course of the session. 5. Supervising all musical aspects of editing and post-production. The producer represents the record company in matters involving the musicians union, and the producer often has to function as diplomat as well as drill sergeant during the sessions. Above all, the producer must remain cool and collected—and always in control of things. In many cases, the producer will have the sole authority to call for overtime in order to finish a recording.

The role of the engineer The engineer has the following responsibilities: 1. Checking out and certifying recording venues. Such matters as ambient noise level, acoustical suitability, and physical comfort are covered here. 2. Taking responsibility for the performance of all recording equipment and directing the technical crew. Quick solutions of all technical problems encountered during a session are essential 3. Translating the producer's sonic wishes into technical reality, through choice of microphones and their placement. In this area most producers are happy to leave the matter entirely in the hands of the engineer. 4. Performing all musical balancing functions at the console during the sessions. 5. Working with the producer, as required, in details of postproduction. (In many large companies, editing may be carried out by specialists working from scores previously marked by the producer.) Like the producer, the engineer must be collected and respond quickly when technical problems arise. The engineer must know the equipment inside out and keep detailed setup notes so that a given microphone array and level settings can be accurately duplicated in the same venue at any later date.

STAFFING There are normally three persons in the control room during a professional session: producer, engineer, and assistant engineer. The role of the assistant engineer is generally to keep the recording log, which relates the producer's

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slate numbers with start-stop times on the recording medium, whether it is tape or disc. The assistant engineer should be able to take over for the engineer in case of any emergency.

STUDIOS VERSUS REMOTE RECORDING VENUES Most classical recordings are made in remote recording venues. Orchestras normally prefer to record in their regular performance halls, but these are often unsatisfactory for the recording of large scale works. Specifically, the reverberation times in many halls are not long enough for orchestral recording. Reverberation times in the 2- to 2.5-second range are ideal. In the case of older halls with a proscenium and a deep orchestra shell, there are additional problems. The purpose of the shell is to project sound toward the audience during concerts. This may be a problem in recording, since the acoustical environment is different between the front and back of the stage. For recording, all orchestral players should be in the same acoustical space, and stage extensions are often used to move players to the front of the stage and into the house. Among the spaces used for remote classical recording are churches, ballrooms, and a surprisingly large number of Masonic meeting halls throughout the United States. Most of the good rooms are fairly old, built when lots of concrete and plaster were used for interior surfaces. But many of these older locations are apt to be noisy, cold in winter, and hot in summer. The newer buildings are more apt to be comfortable, but they are likely to acoustically inferior because of excessive acoustical absorption. It is the engineer's responsibility to check out and certify remote venues, and the following are some of the points that should be considered: 1. Is the space acoustically appropriate? If it's too live, can it be partially draped to reduce reverberation time? If the room is too dead, can it be livened? (See Figure 20-13.) 2. If there is a stage, does it project well into the house, or will a stage extension be required? 3. Is the air handling quiet, or must it be turned off* during actual recording? 4. What about various comfort factors and facilities for the musicians? 5. Can a control room be adequately set up close by so that the conductor or artist does not have far to walk? Do not forget to arrange for extensive damping materials to make the control room sufficiently absorptive.

Dynamic Ranges of Musical Instruments and Ensembles

293

6. What about external noise? Traffic around the building should be observed for at least one week, and any unusual patterns noted. Also consider any other activities that may be scheduled in adjacent spaces in the same building. 7. What about electrical service? Is it adequate and free from troublesome transient disturbances? During a recording made in a remote location, it is essential that the assistant engineer keep a sharp ear open to extraneous noises of any kind and note them in the recording log. A private phone link between producer and conductor is essential since it enables sensitive conversation to take place. A video link between stage and control room may be desirable for larger projects A properly designed studio will have few or none of the noise and comfort problems you're likely to find in many remote locations. This leaves us only with the acoustical disadvantages of most studios, but in many cases, it is possible to work around these through the use of modem high quality artificial reverberation. Studios are in fact strongly recommended for smaller musical forms, such as solo instruments and small chamber groups.

DYNAMIC RANGES OF MUSICAL INSTRUMENTS AND ENSEMBLES It usually comes as a surprise to recording engineers that the dynamic ranges of most instruments are as limited as they are. Figure 20-1 gives a clear indication of this. The string instruments have a fairly uniform dynamic capability over their frequency range. By comparison, woodwind and brass instruments shift widely in their dynamic characteristics depending on the range in which they are playing. A string quartet, for example, may normally play with an overall dynamic range that doesn't exceed 30 dB, so there should be no problem in recording that group with a recorder capable of handling a 90-dB signal-to-noise range. The piano produces initial keystroke dynamic ranges not exceeding about 35 or 40 dB. When an orchestra plays loudly, all players are involved, and brass and percussion will predominate. When the orchestra plays very softly, there may be only a few string instruments playing. The acoustical power output of the orchestra may range from 15 to 30 watts for the full ensemble to less than a microwatt for the softest passages, and the resulting dynamic range may be 70 to 75 dB. However, not many home environments have a low enough

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ambient noise level to allow full appreciation of such a wide range program without having the high level portions of the program be extremely loud. Virtually all classical recordings today are adjusted in dynamic range, usually at the mastering stage, to ensure that the recorded product meets the buyer's expectations.

C4

C5 C6 Trumpet (pp to ffsX\ meter)

C3 C4 C5 Horn (ppto /fat 1 meter)

Figure 20-1. Dynamic range of selected musical instruments showing the span in dB between playing very soft (pp) and very loud (ff) over the normal range of the instrument.

Table 20-1 shows some of the published data regarding power output from various instruments and ensembles.

Recording Solo Instruments

295

Table 20-1. Maximum power outputs and levels of various musical sources Source Male speech Female speech Violin Bass viol Flute Clarinet Trumpet Trombone Orchestra

Maximum power output

SPL^ at 10 ft (3.3 m)

0.004 watt 0.002 0.01 0.07 0.3 1.0 2.5 5.0 15.0

73 dB 70 79 88 94 99 106 109 972

Notes: 1. Calculations made assuming Dl = 1. 2. Calculated for a distance of about 30 feet (10 m).

RECORDING SOLO INSTRUMENTS The piano The solo piano is normally recorded with a pair of microphones placed fairly closely to the instrument, as you can see in Figure 20-2. If you choose a coincident cardioid pair, as shown at A and B, the sound stage will be fairly narrow and ambient pickup will be low. This may however be a good choice in a live space. The spaced microphone approach shown at C and D will result in a much broader stereo sound stage. This method, using omni microphones, is preferred by most engineers and producers today since it produces a generally warmer sound. Watch the spacing between the microphones and be careful to avoid a "hole in the middle." You should also be aware that most concert caliber pianos have been voiced to be on the bright side so that they will project well in a concert hall. These instruments usually need to softened or "pulled back" somewhat for recording. A good technician can do this fairly quickly, and it is a good idea to keep the piano technician on stand-by during the recording to ensure that the instrument is in top shape and tune at all times. Regarding the recorded perspective, it is ideal for the piano to be centered and occupying about one-half the total stereo stage width. Ambient program information should be perceived as coming from the entire width of the stereo stage.

The harpsichord Many of the same principles which apply to the piano may be used here. There are several important differences, however. While the modem piano is

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296 2Qr-^--0

D 1 Height: 6.5' Distance: 6.5'

2 8* 10'

1 Height: 6.5' Distance: 6.5' Spacing: 3'

2 8' 10' 5'

Figure 20-2. Recording the solo piano. With coincident or near-coincident microphones (A and B); with spaced-apart microphones (C and D).

a mechanically quiet and smoothly regulated instrument, the harpsichord action is apt to be noisy. If the instrument is recorded too closely this will be problematic. The proximity effect of directional microphones may aggravate the problem, and in that case you may need to use 50- or 80-Hz sharp highpass filters. Since the instrument is basically a "period piece" from the 18th century, many of its musical requirements may call for fairly reverberant spaces. Because of its relatively rich HF content and precise attack, the harpsichord may be presented against a denser reverberant background than would be appropriate for the piano.

The guitar and lute These instruments are small, and they are normally recorded close-in. The apparent stereo width of the instrument should be about one-third the stereo stage, and coincident or near-coincident microphone arrays will do this nicely. Reverberation should convey a feeling of intimacy; that is, it should be fairly short (1 to 1.5 seconds), and there should be enough of it to support the

Recording Solo Instruments

297

relatively thin texture of the instrument. Remember that the lowest string on the guitar is E2 (82 Hz). Figure 20-3 shows two methods for stereo pickup of these instruments. Proximity effect may add an unnatural LF boost if directional microphones are used, and you may have to remove some of this with an LF shelving cut.

Ax= 40to60 in

^^

o

y = 20to40 in

i

o z- 40to60 in

Figure 20-3. Recording the guitar. With coincident microphones (S); with spaced microphones (B).

The guitar or lute may easily be recorded in a relatively dry studio, since good artificial reverberation always works well with these instruments. The HF response of the reverberant signal should be rolled off above about 3 or 4 kHz.

The harp It is almost impossible for a good player to make an ugly sound on a well tuned harp. There are many microphone approaches that work well. Keep in mind that the instrument is not very loud and that room noises can get in the way if you are too far from then instrument. Figure 20-4 shows one approach to placing microphones. Keep the stereo image well centered, and don't hesitate to add some reverberation to give the stereo stage added width.

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^ X

.-"

« . - ^

Top view

Perspective view Figure 20-4. Recording the harp. Distance x about 40 inches.

Many engineers prefer to pick up the harp with a pair of omnidirectional microphones, about 20 inches apart and positioned about 40 inches from the instrument.

The organ Most organs are located in houses of worship, and those spaces often have fairly long reverberation times. The ideal reverberation time for an organ is in the 2.5- to 4-second range. Large cathedrals and European churches may have reverberation times in excess of 6 seconds. Modem organs have borrowed heavily from traditions of eighteenth century North German and French organ design, and most instruments are placed fairly high above the floor. Figure 20-5 shows a typical installation in the rear gallery of a church. ^

^

BJ]

-o

^

Elevation view

Section view

Figure 20-5. Recording the pipe organ. Distance x about 6 to 12 feet, distance j^ about 10 to 20 feet.

Recording Chamber Groups

299

Either a coincident microphone pair or spaced omnidirectional pair (shown in the figure) will usually provide excellent pickup. For spaced microphones, x should be in the range of 6 to 12 feet, and typical distances for y range from 10 to 20 feet from the gallery rail. If the environment is fairly reverberant, then the single microphone pair will pick up enough ambience. In less reverberant spaces, a secondary stereo pair (about 20 feet behind the main pair) will provide the necessary ambience. Microphone height should be at the average height of the instrument. The spaced omnidirectional microphones can create "excellent spatiality, but without image specificity." Many engineers, producers, and organists are willing to sacrifice precise left-right imaging for a greater sense of large-room ambience. Many large organs have low frequencies that reach down to the 20 Hz range. This is another good reason to use omnis, since their LF response is normally quite extended. You need good monitor loudspeakers to ensure that you are actually picking up these frequencies.

RECORDING CHAMBER GROUPS Chamber groups generally range from two to about twelve players. The category includes solo vocal or instrumental with piano, string quartets, and a variety of other instruments with one performer on each part.

Seating the musicians For public performance, musicians normally face the audience. In a recording environment, their positions can be altered as required to make for better sound pickup. While players may initially be reluctant to change their traditional seating positions, they can usually adapt to new ones—especially if there are some real recording advantages. Consider several possibilities for seating and pickup before the recording and discuss them with the producer. Between the two of you, a plan can be worked out that the musicians can adapt to.

PIANO WITH SOLO INSTRUMENT OR VOICE Figure 20-6 is the ideal way for a soloist to maintain good eye contact with the pianist, while allowing the engineer to get the desired balance. In recording solo instruments with piano accompaniment, it is important to keep them both in proper scale, and the method shown here allows the engineer to

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separately adjust both piano and soloist levels. A recommended console setup is shown in the right portion of the figure. Studio setup

Console setup

0 10 ©1 Pan .0. .0.

Reverb send ^

e

_ Soloist

er 3

0

Faders

Microphone

C

0

0

Mi 1

2

3

c 1

1 Reverb return

Figure 20-6. Recording a vocalist or instrumental soloist with piano.

Microphones 1 and 2 can be adjusted to get an optimum pickup of the piano, while microphone 3 can be positioned for optimum solo pickup. For a vocalist the operating distance would be in the range of 2 to 3 feet. In most instances the soloist will need a touch of artificial reverberation, and that is shown in the console setup. A variant of this approach is to use a coincident or ORTF stereo pair on the soloist instead of a single microphone panned to center. The ORTF pair should be panned left and right. This approach may give just a little more feeling of space around the soloist, and allow small sideways movements to be tracked on the stereo stage. Whether you use one or two microphones on the soloist, the microphones should be positioned sHghtly above the source and aimed downward toward the sound source. Most singers can cover an extremely wide dynamic range, and you shouldn't hesitate to adjust balances as you are laying down the basic tracks. The producer will be your best guide here.

RECORDING THE STRING QUARTET The players in a string quartet normally seated as shown in the upper left portion of Figure 20-7. The separation between players is no greater than it has to be, since visual contact is so important to good ensemble playing. Method 1 uses a coincident or near-coincident pair on the group, and the microphones

Recording the String Quartet

301

should be placed overhead looking into the quartet. At a typical operating distance of about 6 feet, the stereo stage may seem a little narrow and the group may sound too distant. Performance setup ^^?"^ violin

vIoHn

Method 1

Cello ^^ ^

^, ^, ,

Viola

^^?"^ violin

Cello ^

^i^jj^

Viola

5 to 6 feet

V Method 2 Second violin

Method 3 ^^„^ ^^"^

^

Second violin

©

U7 First violin

^

0

0

^. , Viola

First ^IQH^

^'Olin ^

r>^u^ ^®"°

0 _ Viola

^

Figure 20-7. Recording the string quartet.

In Method 2, three omni microphones have been placed closer to the players. The sound stage will be considerably wider, and the quartet will have a closer perspective. Left and right microphones are panned hard left and right, while the center microphone is fed into the mix just enough to get a good front-back balance and "anchor" the cello a little right of center-stage. This approach works best with extremely flat microphones at a height of about seven feet, and the overall impression of the recording is that the quartet is performing in your listening room—rather than transporting you into a recital hall. Most quartet recordings you hear today are done in this way. Method 3 in a sense "deconstructs" the quartet, allowing the engineer and producer to reconstruct it in postproduction The microphones would normally be cardioid, placed overhead and aimed at the respective instruments. The approach is often used in live performance recordings. An advantage here is that you can widen or narrow the stereo stage after the fact, and of course you will have excellent immunity from audience noises. Methods 2 and 3 normally require artificial reverberation in order to flesh out the sound.

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OTHER CHAMBER GROUPS The piano-string trio In concert the piano trio must be recorded as shown in Figure 20-8A. In a studio setting the approach shown at C is recommended. As with the vocalist with piano we discussed eariier, the aim here is to put the players in a circle, all looking at each other. The piano pickup would be panned to create a broad center image, and each string player would be panned slightly inboard to give cohesion to the group. The string microphones should be at about the same height as you would use with a string quartet. Omnis would normally be used on the piano, and cardioids on the two strings. Artificial reverberation would normally be used. B )

D ^Violin

)

D

/Horn 7

BassoonX

Cello Oboe>

SCIarlneN

\f

\ /

c

D

)

9

^ . Cello.

Violin

n

_

J fo

p

f

)

^LJ

9 OCIarineO Oboe

Figure 20-8. Recording chamber groups. Trio and quintet in concert setting (A and B); in studio setting (C and D).

The piano-wind quintet A concert setup is shown at B. In the studio the approach shown at D would be used, with the players spread out to give them ample playing and breathing room. The cardioid microphones on the winds should be placed overhead.

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slightly in front of the players, and aimed downward. The horn is the exception here: place the microphone directly over the instrument at about 7 feet. Use artificial reverberation to flesh out the sound.

NOTES ON MICROPHONE LEAKAGE Leakage is the result of pickup commonality between adjacent microphones. In the pop studio we normally try to avoid this, but in classical recording it is an advantage—if it is controlled. Taking another look at Figure 20-8C, you will observe that each omni microphone on the piano will pick up some sound from both cello and violin. This leakage will be several dB lower in level and will not interfere or conflict with the primary violin and cello pickup; in fact, the leakage will add a degree of richness and warmth to the sound, since it simulates nearby, early reflections. There is one very important point you must remember: leakage is most pronounced between adjacent microphones, and those microphone outputs should be panned to adjacent positions on the stereo sound stage. Always seat the players and position their microphones from left to right as you expect them to appear in the stereo stage as heard over the control room monitors. If you need to change the basic positioning of instruments, do it in the studio—not at the console.

NOTES ON ARTIFICIAL REVERBERATION When making a recording in a fairly live room you can place ambience microphones about 15 or 20 feet away from the players. These would normally be cardioids pointed away from the players. In many cases however the room is too dry to support a significant reverberant field, and you will have to use artificial reverberation. Here are some general rules and suggestions: 1. Use the best digital reverberation unit you can get. 2. Use a bass multiplier setting of about 0.8 at a crossover frequency of about 500 Hz. (This will keep the texture clear and avoid any trace of muddiness.) 3. Use a reverberation time setting no greater than about 1.8 seconds. 4. Roll off both the reverberation setting and frequency response above about 3 kHz. 5. Use a predelay no longer than about 20 milliseconds. 6. Disable any randomizing functions that the program offers.

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Some reverberation units provide ambience programs, which generate a set of early reflections with little if any reverberant "ringout." Such programs are very useful and may be better in some situations than an actual reverberation program.

RECORDING LARGE MUSICAL GROUPS Choruses Choruses, large and small, are usually best picked up by using one of the microphone arrays shown in Figure 18-14. The chorus will normally be standing on risers, and the microphones should be about 12 feet above the floor, about 3 or 4 feet in front of the first row of singers. An ambient microphone pair can be used for reverberant pickup, if needed. A soloist in the chorus singing a short passage may sing "in position" and not need a microphone. However, an extended solo with choral background will need to be picked up separately. In that case the singer should stand in front of the group so that there will be minimal leakage into the soloist's microphone. Added reverberation will be required for the soloist, and this is best with an artificial reverberation unit.

The orchestra Figure 20-9 shows a typical seating plan for a modern symphony orchestra. The principal players are the heads of their respective sections, and you can see that they tend to be clustered toward the middle of the orchestra. Typical seating pian for a large symphony orchestra

•

- positions of principal players

Figure 20-9. Seating plan for a modem symphony orchestra.

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Furthermore, the wind and string principals each form a quartet of players. This proximity is important because they often confer during rehearsals, and very often play as isolated quartets. The typical width of an orchestra on-stage is about 55 feet and the depth is about 30 feet. Many modem concert halls have a choral terrace just behind the orchestra, elevated about 10 to 12 feet. By way of terminology, the woodwind instruments are always referred to as "winds." The term "brass" refers primarily to trumpets, trombones, and tuba, and the French horns are always referred to as "horns." We can think of recording the orchestra in layers. The first and most important layer is the frontal microphone array, which we discussed in Chapter 18. A second layer consist of a group of what are called spot, or accent, microphones. These are microphones placed fairly close to certain instruments in order to do one or more of the following: 1. To increase loudness. (Many extended wind and string solo lines need to be increased in volume.) 2. To increase presence. (Many orchestral elements need added presence without resorting to playing louder. (Examples would include harp, celesta, orchestral piano, and various percussion instruments.) 3. Add focus to the recording. (The winds as an ensemble always need more focus than the main microphone array provides. A secondary stereo pair of microphones provides this. Spot microphones will all call for some degree of added reverberation. A third layer is provided by the ambience microphones, and a fourth layer consists of the chorus behind the orchestra, if there is one. A final layer would be any front-stage soloists. It is clear that each layer must be picked up separately, and yet they must all blend into a unified stereo sound stage. It is essential that all spot microphones be panned into the stereo mix at positions matching the player's actual positions on stage.

The session and setting initial balance Prior to the recording session, the producer, conductor, and engineer will have met to discuss details of the recording, including the assignment of spot microphones. A typical list of spot microphones would include: principal first violin (concertmaster), wind pair, first stands of basses, brass overhead, horns overhead, timpani overhead, celesta/piano, and harps. The main microphones would be placed as shown in Figure 20-10. Distance A would usually be about 4 feet from the front row of players, and the height of those microphones would be between 10 and 11 feet. Distance B would be approximately one-third the frontal width of the orchestra. Microphones 5 and 6 would be 9 to 10 feet above the fioor, and they would

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be positioned just in front of the first row of winds, with the two cardioids aimed toward the ends of the second row of winds.

Figure 20-10. Basic microphone plan for an orchestra.

The horn and brass spot microphones should be about 10 feet above the floor and aimed directly downward. Timpani and bass spots should be placed about 8 feet above the floor and aimed downward. Harp, celesta, and piano spots are normally just a few feet away from the instruments. If you are working in a venue for the first time you will need at least one rehearsal to set your basic balances. Begin with the main pair by raising the two faders to their nominal ''zero'' position. Set operating levels using the trims only. Then, raise the two flanking microphones, again to their zero point on the faders, and then trim them so that their contribution to the overall balance is about the same in overall level to the main pair. You will have to bus on and off each pair in order to establish this. Once the four frontal microphones have been balanced to your, and the producer's satisfaction, proceed with the spot microphones. The first is the wind pair. Since most of your microphones will probably have the same output sensitivity, you should be able to zero in fairly quickly on a proper balance using the trims. Next in order are the house microphones, which should be cardioids widely spaced about 20 feet back in the house and facing to the rear. Bring them into the mix using the trims, with the faders at zero. When you are finished with this procedure, all eight of the microphones adjusted so far should be set at their nominal zero positions, and the basic balance will have been made using only the trim controls. Once you have reached this point, you are virtually home free. The remaining spot microphones can be "fine tuned" as the music gets underway.

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As you bring in the remaining spots, you may want to shelve out the LF response by about 3 dB below about 100 Hz. This will give you a little more leeway in manipulating them during the recording.

Overall orchestral level I have never measured an orchestra that produced a level at the main pair greater than about 105 dB SPL, but actual peak levels will depend on the hall and the nature of the music. If you have made your initial settings too high, your first recourse is to pull the overall levels at the group master controls. Conversely, you will have to raise the group levels if you have trimmed too low. In any event, these adjustments can usually be made on a running basis.

Document your settings This means making a chart of the house microphone layout, indicating microphone models, their settings, and their heights. Console trims, fader positions, pan settings, and EQ settings must also be noted. The next time you work in this venue you should be up and running at the first downbeat. Even in other venues, the level settings shouldn't be too different, if you are using the same console and microphones.

LARGE ORCHESTRAL RESOURCES Figure 20-11 shows the floor plan of a large work with chorus and soloists. Many such recordings today are made during live concerts, and you will have to work around some inconveniences. The microphone plan for the orchestra is pretty much as we have already described. The chorus will need three or four microphones that function in the same way as the four frontal microphones on the orchestra. Place them as high as you can without running into a canopy shell, and aim then downward into the chorus. This will minimize your biggest problem, which may be leakage from percussion and brass in the back of the orchestra into the choral microphones. Ideally, each soloist should have a microphone and a track on the recorder so that final balances can be made in postproduction. In many cases there may not be enough track capability to do this, and you will have to balance them as you go, mixing them into a stereo pair. We have all been in these situations, and they are not easy to cope with. It is cases such as these where years of experience will pay off. Take advantage of every opportunity you may have to observe experienced classical recording engineers at work.

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Figure 20-11. Recording orchestra with chorus and soloists.

DELAYING SPOT MICROPHONES Figure 20-12 shows the rationale for delaying spot microphones before they are introduced into the mix. Since these microphones are fairly close to the instruments they are picking up, their introduction into the mix comes before their actual acoustical arrival at the main microphone pair. For example, if a spot microphone is located 20 feet from the main pair, it has a time advantage of 20/1130 seconds, or about 17.7 milliseconds. It is always correct to delay the microphone signal to compensate for the acoustical delay, but it may not be necessary because of certain masking effects. In practice only percussion microphones, which may be as far away from the main pair as 25 to 30 feet, would require delay. It is customary to increase the calculated delay by about 10 milliseconds in order to avoid any tendency for comb filtering to be noticeable.

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Main microphone array Figure 20-12. Implementation of spot microphones.

ALTERING ROOM ACOUSTICS Many newer halls have variable acoustic control built into the structure in the form of heavy drapes that can be deployed as needed to damp the space. Some halls have associated reverberation chambers that can be opened or closed to increase the effective reverberation time of the hall. The usual problem in older halls is that they lack the reverberation that engineers and producers feel the music needs, and there is a technique, shown in Figure 20-13, that can substantially liven a large space. The procedure is to get enough 4 mil (0.004 inch) thick polyethylene plastic sheeting to cover the entire seating area and drape that material loosely over all audience seating. The before and after reverberation time measurements in a concert hall are shown in the figure. The material is available from dealers in construction materials and is normally used for paint drops and covering of carpets and seating when construction is underway. Do not use plastic sheeting thinner than 4 mils. You can hear the difference this treatment provides by comparing bands 2 and 12 on the CD "The Symphonic Sound Stage," released by Delos International D/CD 3504.

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Section View of Hall (crosshatching Indicates areas covered by 18,000 square feet of plastic material)

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Figure 25-10. Signal flow diagram for perceptual coding and decoding.

The net result for stereo music is an advantage of about four-to-one in bit savings as compared standard transmission. The various motion picture low bit rate systems have even greater bit saving ratios due to further simplifications resulting from a joint multichannel analysis of the signal. In some cases the data reduction ratio may approach 10 or 12 to 1, depending on the specific nature of the audio program. MP3, the scheme used for music transmission over the internet, often operates at even higher ratios.

Chapter 26 THE STEREO LONG-PLAYING (LP) RECORD

A BRIEF HISTORY The stereo LP has rapidly declined in sales due to the immense success of the compact disc. Since 1947 to the present day, however, the LP has represented the longest period of compatibility between product and players in the history of consumer audio, exceeding the era of the 78-rpm disc (1895 to 1947). As the 21 St century gets under way, the LP continues to hold its ground in the disk jockey driven world of dance music, where the rapid manual cueing capability of the LP is still a very important factor. Technologically, the disc is an outgrowth of Edison's original cylinder medium which dominated recording during the last quarter of the 19th century. Berliner's disc rapidly overtook the cylinder in the early years of the 20th century, primarily because enormous manufacturing advantages. Until the late 1920s, recording and playback remained an acoustomechanical process. At that time, Maxfield and Harrison developed electrical recording, and the major problems of bandwidth and distortion were solved. In 1947, Peter Goldmark of CBS combined the advantages of a quiet vinyl plastic pressing material with microgroove geometry and 33 1/3 rpm to produce playing times up to about 25 minutes. The stereo LP had been invented conceptually by Blumlein in the early 1930s, when he demonstrated that two independent modulation channels could be cut at ±45 degrees to the surface of the disc. But it wasn't until 1957 that the stereo disc became a commercial success, and during the golden era of the stereo LP (1960 to 1985) many significant improvements were made in the electromechanical aspects of both cutting master discs and playing the pressed discs. As a result, the medium attained audiophile status, compromised only by occasional pressing problems along with ticks and pops. Consumers were inclined to overlook these defects because of the high level of audio quality that was otherwise obtained. The CD and other optical media have been in the ascendancy for about 20 years, but the LP is far from gone. While the great recordings of the past have been reissued in various new formats, there are many other relatively obscure LP recordings which are not likely be reissued in any form. For this reason alone, there will be LP enthusiasts for decades to come.

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As you proceed through this chapter you will develop an appreciation of just how complex and highly engineered the entire disc cutting and playback process has become.

PHYSICAL PROFILE OF THE LP DISC Figure 26-1 shows physical details of the LP. The diameter is 12 inches (301 mm), and the maximum thickness in the center (label) portion is 0.015 inches (3.8 mm). The recorded portion of the disc is thinner than the center and outer diameter; this contouring saves vinyl material and provides some degree of protection for the grooves when the discs are stacked on a record changer. The various starting and stopping diameters of recording are standardized, as are the pitches of lead-in and lead-out grooves. {Pitch here refers to the number of grooves per unit radius, not the frequency of a signal.)

STEREO MICROGROOVE GEOMETRY AND REFERENCE LEVELS Figure 26-2 shows the basic movements of the cutting and playback styli in the plane of the master disc. Lateral motion {Ä) results from identical signals fed to the 45-degree/45-degre cutting coils. Motions at B and C represent right channel only and left channel only, respectively. The motion shown at D results from an anti-phase relationship between the two input signals. Figure 26-3 shows a scanning electronic microscope view of typical stereo modulation. Note that each groove wall is independently modulated. The outer groove wall of the stereo disc is modulated by the right channel and the inner groove wall by the left channel. The cutting stylus is chisel shaped and is made of sapphire or diamond. The nominal width of an unmodulated groove is about 0.0025 inches (0.064 mm). During heavy modulation, the groove width and depth may increase by a factor of about three, while on upward swings of the cutting stylus the width can be as small as 0.001 inch (0.025 mm). In the early days of disc recording, a wax formulation was used as the recording medium. Since the early 1940s, a lacquer formulation on an aluminum substrate has been used. It is customary to use a stylus heated by a small coil to facilitate cutting the lacquer material and to reduce noise that would otherwise be generated in the process.

Stereo Microgroove Geometry and Reference Levels

373

^ 4.76" (121 mm) ' 11.5: (292 mm)12" (305 mm)

Figure 26-1. Physical views of the LR Surface view (A); section view (B).

The normal zero reference level in stereo disc cutting is defined as lateral peak stylus velocity of 7 cm/sec at 1 kHz. On a per-channel basis, this corresponds to peak velocity of 5 cm/sec.

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Figure 26-14. Variable pitch and depth. In the Neumann VMS-70 cutting system, three signals, as shown at (A) are used to determine pitch and depth requirements. Typical action of the system is shown at (B). The right channel modulation in groove 2 requires a decrease in pitch substantially ahead of modulation so that there will be no overcut into groove 1. The decrease in pitch must be maintained one revolution so that groove 3 can be accommodated without overcut. Modulation on the left wall of groove 4 does not require a preview signal for proper pitch decrease; the signal that controls this is the left program input. Again, the decrease in pitch must be maintained one revolution in order to make room for groove 5. (Courtesy Georg Neumann GmbH)

The Cutting Process

389

Vinyl Pressing 0 -5 -10 Master Lacquer 7 kHz 10 kHz 14 kHz

-10 Metal Mother

-10

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10.5

9.5

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7.5

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Figure 26-15. Diameter losses in disc recording.

THE CUTTING PROCESS When a stereo master tape is received at the cutting facility, the lacquer mastering engineer runs the tape down, noting those sections of it that may be extremely loud. Any basic signal processing, such as limiting or equalization, may be noted at this stage. The playing time of the side is carefully noted, since it will influence the actual cutting level on the disc. Banding requirements are also noted, along with any anticipated band-to-band changes in signal processing. An experimental cut is often made at this stage to ensure that all settings are workable. In preparation for mastering, the engineer places a lacquer 14-inch diameter master blank on the lathe's turntable. The outer portion of the blank can be used for a short test cut to ensure that the correct hot stylus current and stylus depth of cut are within standards. The freshly cut groove is examined with a microscope to ensure that everything is working correctly. As the final cut is made, the engineer lowers the cutter head into the rotating lacquer. The "chip" is the portion of the lacquer material that is actually cut from the disc; it must be immediately picked up by the suction tube, otherwise it may become ensnared in the stylus-heating coil assembly. The styli are made of sapphire or diamond material and are good for a number of cuts. If everything goes well, the mastering engineer examines the freshly cut disc

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with the microscope to ensure that there are no cutter lifts or overcut that may cause processing problems in the pressing plant. If the master disc passes this test it is carefiilly packed and sent to the plant. The approximate playing times on an LP side as a fiinction of average cutting pitch are given below: Pitch (lines-per-lnch): 300 250 200

Approximate playing time: 30:20 25:20 20:15

You can easily appreciate that the disc mastering engineer is a person of considerable skill and mechanical sensibilities who has to deal with fussy producers and artists on the one hand and with inspectors at the pressing plant on the other.

RECORD PRESSING The processing of a master lacquer disc through the various metal-to-metal replication operations and finally to the vinyl pressing is a very intricate one involving many disciplines, including metal plating, plastic formulation, and plastic forming. The basic operations in the three-step process are shown in Figure 26-16. The master lacquer is carefully inspected, cleaned, sensitized, and "silvered" by reduction of silver nitrate on its surface. This renders it electrically conductive. It is then preplated at low electrical current density to build up a thin nickel surface, which is a negative representation of the lacquer surface. Then the current density is increased to produce a substantial backing of nickel. The metal negative so produced is called the metal mästen It is further treated so that a metal mother can be grown from it. The mother is a positive and can be played to check for problems in transfer. Minor defects can often be repaired. Finally, the mother is plated and a stamper is produced. This is a metal negative part which is used for final production. The stampers are ground smooth on their backside so that they will fit snugly into the press. The edges are crimped and the parts carefully centered in the press. The pressing cycle begins by placing a charge of hot vinyl plastic between the stampers, along with the labels. Pressure and heat are applied, and the plastic is molded to

391

Record Pressing Master lacquer

Lacquer

Cleaning, sensitizing, silvering Metal master - formed by electroplating nickel on silvered lacquer

Separation of metal master from lacquer; passivation of nickel surface Metal mother - formed by electroplating nickel on metal master

Separation of metal mother from master; passivation of nickel surface Stamper - formed by nickel plating of mother

Stamper

Plastic

Separation of stamper from mother; preparation of stamper for production

Pressing - formed by heating, pressure, and cooling of vinyl plastic

Stamper

Figure 26-16. The three-step disc replication process.

conform with the stampers. When the molding cycle is completed, cold water is run through the channels of the molds, cooling the record so that it can be removed from the press without warping or other deformation. The remaining plastic around the edge of the disc, referred to as "flash," is trimmed and the process is finished.

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DIRECT METAL MASTERING (DMM) Under the trade name Direct Metal Mastering, the Teldec company of Germany introduced a process of cutting master discs directly on freshly plated amorphous copper, eliminating two steps in the replication process. Their efforts have been complemented by those of Georg Neumann GmbH in the areas of lathe and cutter head development. The technology differs from the traditional approach in the following ways: 1. The cutting is done on a copper layer, which directly becomes the metal mother for subsequent production of stampers. Figure 26-17 shows a view of the cutting lathe with a freshly cut master disc on the turntable.

Figure 26-17. Lathe for cutting Direct Metal Mastering (DMM). (Courtesy Georg Neumann GmbH)

2. There is no spring-back effect in the metal, as there is with lacquer, and deformation effects, such as "groove echo," are virtually eliminated.

Direct Metal Mastering (DMM)

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3. The diamond cutting stylus does not require burnishing facets, and HF recorded detail is much greater than with conventional cutting. 4. A new, more powerful, cutting head is required to engrave the signal, and the physical cutting angle is about 5 degrees. This necessitates electronic processing of the stereo signal by delay modulation to produce an effective net cutting angle of 20 degrees. Details of this process are shown in Figure 26-18. plane of QCIUQI cutting stylus motion plane of standard cutting stylus motion non standard cutting angle E

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