Sound Engineer's Pocket Book, Second Edition - PDF Free Download (2023)

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Pocket book of the sound engineer

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Sound Engineer's Pocket Book Second Edition Bewerkt door Michael Talbot-Smith

Focal Press OXFORD AUCKLAND BOSTON JOHANNESBURG MELBOURNE NEW DELHI

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Focal Press An imprint of Butterworth-Heinemann Linacre House, Jordan Hill, Oxford OX2 8DP 225 Wildwood Avenue, Woburn, MA 01801–2041 A division of Reed Educational and Professional Publishing Ltd A member of the Reed Elsevier plc group

First published 1995 Reprinted 1998 Second edition 2000 © Reed Educational and Professional Publishing Ltd 2000 All rights reserved. No part of this publication may be reproduced in any material form (including photocopying or storage in any medium by electronic means and whether or not temporarily or incidentally for any other use of this publication) without the written permission of the holder of the copyright, except in accordance with the provisions of the Copyright, Designs and Patents Act 1988 or under the terms of a license issued by the Copyright Licensing Agency Ltd, 90 Tottenham Court Road, London, England W1P 0LP. Requests for written permission from the copyright holder to reproduce any part of this publication should be addressed to the publisher British Library Cataloging in Publication Data Sound engineer's pocket book. – 2nd edition. 1. Acoustic Engineering – Handbooks, Manuals, etc. 2. Sound – Recording and Reproducing – Handbooks, Manuals, etc. I. Talbot-Smith, Michael 621.3'828 Library of Congress Cataloging Publication Information A catalog record for this book is available from the Library of Congress ISBN 0 240 51612 5 Typeset by Florence Production Ltd, Stoodleigh, Devon Printed and bound in Great Britain

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Contents

Foreword to second edition Introduction

live live

basic principles

The physics of sound waves

The hearing process

Acoustic noise and its measurement

Typical noise levels

Electromechanical analogies

Digital principles

Acoustics

Sound insulation

Microphones

Radio microphone frequencies

Loudspeakers

Stereo

Analog equipment for mixing sound

Signal processing

104

Analog recording and playback

112

Analog noise reduction

122

Compact disc

131

Digital audio tape

141

audio measurements

154

Contents

Digital equipment

159

Midi

166

Studio airconditioning

173

Distribution of audio signals

177

Radio propagation

182

Digital interfacing and synchronization

186

Ultrasonic

191

Radio studio facilities

195

Connectors

202

Public address data

205

Useful literature

207

Table of contents

211

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Foreword to the second edition

I have had many satisfactory comments about this Pocket Book. It's good to know that it fills a need, at least for some professionals. The second edition has been carefully reviewed and has led to many changes and (hopefully) improvements. I am especially grateful to Keith Spencer-Allen for critically reviewing the first edition and also for providing most of the material for the section on Digital Audio Tape. I am also grateful to Vivian Weeks for his important contribution to the section on audio measurements. Michael Talbot Smith

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Introduction

Some pocketbooks seem to be miniature textbooks. I believe that the place for textbooks is back in the office for occasional reference! Pocket books, I think, should contain the kind of information that may often be needed; that such a book has an important place in the pocket and is as essential there as one's wallet or car keys. Of course, it is unlikely that everyone will need all the information in such a book, but I hope that this Pocket Book will be of interest to most sound engineers. For more detailed information on each topic, the reader is referred to the Audio Engineer's Reference Book (Focal Press, Second Edition, 1999) from which most, but not all, of the data has been extracted. Some of the material is from my own work and some is from the excellent little Audio System Designer handbook published a few years ago by Messrs. Klark-Technik. I gratefully acknowledge their permission to quote from it. The ideal target, as I said, would have been reached if a practicing engineer suffered a panic attack upon discovering that he or she had left behind the relevant notebook. I doubt that will ever quite be the case, but perhaps this audio engineer's pocketbook comes close to that ideal. Michael Talbot Smith

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1 Basics

Useful relations Speed of light in free space 299 792,458 km/s Surface area of a sphere 4r2 4 Volume of a sphere r3 3 Peak value of sine wave 1,414 r.m.s. e base to natural logarithms 2.718 28 SI Prefixes and multipliers Multiplier

1 000 000 000 000 1 000 000 000 1 000 000 1 000 1/1000 1/1 000 000 1/1 000 000 000 1/1 000 000 000 000

1012 109 106 103 103 106 109 1012 1015 1018

Prefix

Symbol

tera giga mega kilo milli micro nano pico femto act

T G M k m n p f a

Definitions Electric current (I). The unit is the ampere (A). One ampere is the current which, when flowing through two conductors of infinite length, one meter apart and of negligible cross-section, produces a force between them of 2107 newtons per meter of length.

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Force (F). Unit the newton (N). A force of one newton causes a mass of one kg to accelerate by one m/s/s.

basic principles

Work or energy (W). Unit is the joule (J). One joule of work is done when one newton of force moves through one meter in the direction of the force. Power (P). The unit is watts (W). Power is the rate at which work is done: one joule of energy expended in one second requires one watt of power. (The non-preferred unit of horsepower equals 746 watts.) Charge (Q). Unit is the coulomb (C). It is the amount of electricity that passes through a conductor when one ampere is flowing for one second. Potential difference (V). The unit is the volt (V). A potential difference of one volt exists between two points if one joule of work is done in transferring one coulomb of charge between them. Resistance (R). Unit of the ohms (). One ohm is that resistance at which a current of one ampere generates energy at a rate of one joule per second. Capacity (C). Unit is the farad (F). A capacitor with a capacitance of one farad will increase the potential between its plates by one volt when the charge on it is increased by one coulomb. Magnetic flux (⌽). Unit is the weber (Wb). If the current experienced by a conductor changes at the rate of one weber per second, a potential difference of one volt is created across the ends of the conductor. Flux density (B). The unit is the tesla (T) (Wb/m2). Flux density is the flux per square meter perpendicular to the field. Inductance (L). Unit is the henry (H). A closed circuit has an inductance of one henry when a potential of one volt is produced by a current rate of change of one ampere per second. Press (p). The unit is the pascal (Pa). This corresponds to one newton per square meter (N/m2).

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Frequency (f). The unit is the hertz (Hz). The number of complete waves passing a fixed point in one second. (The units of cycles/second, c/s or c.p.s, are not currently approved.) Wavelength (␭). The distance between corresponding points on successive waves. Permeability (␮). 1. The permeability of free space (0) is the ratio of the flux density (B) to the magnetizing force (H) in a non-magnetic material (space). 0 B/H4 107 (H/m) 2. The relative permeability (r) of a material is the factor by which the flux density increases for the same magnetizing force.

Foundations of mathematics Base 10 ("decimal" or "denary") e.g. 39610 (3 102) (9 101) (6 100) 30010 9010 610 39610 Base 2 (binary) E.g. 10112 (1 23) (0 22)(1 21) (1 20) 810 0 210 110 11 Base 16 (hexadecimal) i.e. 0,1,2,3,4,5,6,7,8,9,A, B,C,D,E,F Hexadecimal numbers are usually preceded by '&'.

Basic principles e.g. &B2F16

(11 162) (2 161) (15 160) 281610 3210 1510 286310

The decibel In principle, the decibel is a unit of comparison of two powers. If the powers are P1 and P2, then the gain or loss of one over the other is dB 10 log10(P1/P2) (Note that the subscript indicating that the logarithm to base 10 is normally is omitted.) Since Power ∝ (voltage)2 and also to (current)2 when used with voltages or currents (or pressures in sound), the expression becomes dB 20 log (V1/V2) or

dB 20 log (I1 /I2)

dB 20 log (p1 /p2)

where p1 and p2 are sound pressures

Acoustic resonances Membrane or panel.

f0 =

√Md

where M is the mass/area unit of the membrane or panel in kg/m2 and d is the depth of the air space in metres. Helmholtz resonator

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(Helmholtz resonator, continued)

f0 =

kr 2

current

2 V(2Ir)

where c is the speed of sound, r is the radius of the neck, l is the length of the neck, and V is the volume. Decibel values for power, current and sound pressure level (SPL) ratios Power ratio 1 1.5 2.0 2.5 3.0 4.0 5.0 10.0 20.0 30.0 50.0 100.0 500.0 1 000.0 10 000.0

Electrical Formulas Ohm's Law:

V = IS I =

V i

Voltage, current or SPL ratio 1 1.23 1.41 1.58 1.73 2.00 2.24 3.16 4.47 5.48 7.07 10.00 22.36 31.6 100.0

dB 0 1,8 3,0 4,0 4,8 6,0 7,0 10,0 13,0 14,8 17,0 20,0 27,0 30,0 40,0

basic principles

Current:

P = IV = I 2R =

V2 Watt R

Resonance:

f0 =

1 2 √LC

in series LC circuits and in parallel circuits where the resistance is small and can be neglected.

Color Coded Resistance Band

Bands 1 and 2: first and second digit. Band 3: multiplier. The color code is the same as for bands 1 and 2, but indicates the number of zeros after the first two digits. Band 4: percentage tolerance (if any). Band 5: stability. Pink, if present, indicates high stability. Bands 1 and 2:

Band 4 (tolerance):

Black Brown Red Orange Yellow Green Blue Violet Gray White No band

0 1 2 3 4 5 6 7 8 9 ± 20%

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Silver Gold Red

± 10% ± 5% ± 2%

Amplitudemodulatie

Figure 1 Modulation index is given by m B/A

Frequency modulation

Figure 2 Frequency modulation (continued on page 11)

Basic Principles Decibel Values for Power Ratio Power Ratio

1 1,5 2,0 2,5 3,0

0 1,76 3,01 3,98 4,77

4,0 5,0 6,0 7,0 8,0

6,02 6,99 7,78 8,45 9,03

9,0 10,0 20,0 30,0 40,0

9,54 10,00 13,01 14,77 16,02

50,0 60,0 70,0 80,0 90,0 100,0

16,99 17,78 18,45 19,03 19,54 20,00

200,0 300,0 400,0 500,0 600,0

23.01 24.77 26.02 26.99 27.78

700,0 800,0 900,0 1 000,0 5 000,0

28,45 29,03 29,54 30,00 36,99

10 000,0 100 000,0 1 000 000,0

40,00 50,00 60,00

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Decibel values for voltage, current and sound pressure level (SPL) Voltage, current or SPL ratio

1 1,5 2,0 2,5 3,0

0 3,52 6,02 7,96 9,54

4,0 5,0 6,0 7,0 8,0

12.04 13.98 15.56 16.90 18.06

9,0 10,0 20,0 30,0 40,0

19.08 20.0 26.02 29.54 32.04

50,0 60,0 70,0 80,0 90,0 100,0

33,98 35,56 36,90 38,06 39,08 40,0

200,0 300,0 400,0 500,0 600,0

46,02 49,54 52,04 53,98 55,56

700,0 800,0 900,0 1 000,0 5 000,0

56,90 58,06 59,08 60,00 73,97

10 000,0 100 000,0 1 000 000,0

80,00 100,00 120,00

Basics Voltage, current or SPL ratios for decibel values dBs

Voltage, current or SPL ratios

0 1,5 2,0 2,5 3,0

1,00 1,19 1,26 1,33 1,41

3,5 4,0 5,0 6,0 7,0

1,50 1,58 1,78 2,00 2,24

8,0 9,0 10,0 15,0 20,0

2,51 2,82 3,16 5,62 10,00

25,0 30,0 35,0 40,0 45,0

17,78 31,62 56,23 100,00 177,82

50,0 55,0 60,0 65,0 70,0

316,23 562,34 1 000,0 1 778 3 162

75,0 80,0 85,0 90,0 95,0

5 623 10 000 17 782 31 622 56 234

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(Frequency Modulation, continued from page 7) The modulation index for an f.m. system is given by

carrier frequency deviation fD modulating frequency fm

2 The physics of sound waves

Units of Pressure: Pascal (Pa)

1 N/m2 (also equal to 10 dyne/cm2)

bar: 1 bar

105 Pa

Torr: 1 torr 133.22 Pa Sound wave pressures fall in the range between 0.00002 Pa or 20 Pa), roughly corresponding to the average ear threshold around 3–4 kHz, to about 200 Pa, which is generally considered to be about the level of pain .

Intensity Watt/m2 (W/m2), or, more practical in terms of sound, W/m2.

Speed of sound, typical values See adjacent table. A general expression for the speed of sound in gases is given by: c ( P/ )2 where is the ratio of the specific heat of the gas (1.414 for air), P is the pressure and the density of the gas.

Frequency (f ) and wavelength (␭) These are related by the expression c f Note that this formula applies to all waves. In the case of electromagnetic waves (radio, light, etc.) c is about 300 000 km/s (3 108 m/s) versus about 340 m/s for airborne sound waves. The formula can only be used with caution for surface waves on water, as the speed of the waves can vary with amplitude.

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Speed of sound, typical values Substance

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Air, 0°C* Hydrogen, 0°C Oxygen, 0°C Carbon monoxide, 0°C Carbon dioxide, 18°C Water, 25°C Seawater, 20°C Glass Aluminum Brass Copper Iron (wrought) Iron (cast ) Concrete Steel Wood, pine, long grain oak pine

c (m/s) 331,3 1284 316 337 266 1498 1540 ~5000 5100 3500 3800 5000 4300 3400 5000–6000 5000 4000–4400 3300

* The speed of sound in air increases with temperature by about 2/3 m/s per °C rise in temperature. More precisely: c 331 0.6t where t is the temperature in °C.

Sound wavelengths in air at 20°C

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Frequency (Hz)

Golf length (m)

16 20 30 50 100 200 500 1 000 5 000 10 000 16 000

21,43 17,15 11,43 6,86 3,43 1,72 0,69 0,34 0,069 0,034 0,021

The physics of sound waves

The inverse square law Intensity (I) decreases with distance (d) according to:

1 d2

Pressure, on the other hand, follows the law:

1 d

The Doppler effect Assuming a stationary medium, if the source is moving towards the observer with velocity versus the apparent frequency, fa, is given by

fa =

fc (c - vs)

If the observer moves towards the source with speed vo, then

fa =

f (c vo) c

The scale The adjacent table shows the equal temperament scale, where the ratio of the frequency of one note to the next above is 12⎯ √2 or 1.059 463 1.

Frequencies of vibrations in pipes and strings Pipe open on one side:

nc 4 (to)

where n is 1,2,3 etc. l is the length of the pipe and a is the final correction. For a pipe with a significant flange, a is about 0.8r, where r is the radius. (Continued on page 17)

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De toonladder Noot A A# B C C# D E E# F F# G G# A A# B C′

Figure 3 Part of a keyboard

Frequency (Hz) 220.00 233.08 246.94 261.63 277.18 293.66 311.13 329.63 349.23 369.99 392.00 415.30 440.00 466.16 493.88 523 .25

The physics of sound waves

Figure 4 Frequency ranges of some typical sounds

Figure 5 Standing golf cartridge

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(Vibration Frequencies in Pipes and Strings, continued from page 14) In the case of an unflanged pipe, a is approximately 0.6r. Pipe open at both ends:

nc 2 (l 2a)

The speed of a transverse wave in a series of voltage T and mass m per unit length:

Tsukasa Tm

1/2

The lowest vibration frequency in such a string is then

Tsukasa

1 T 2l m

1/2

Enkele Italiaanse termen uit de muziek fff (Molto fortissimo) ff (fortissimo) f (forte) mf (mezzo forte) mp (mezzo piano) p (piano) pp (pianissimo) ppp (erg pianissimo) minus plus staccando crescendo diminuendo Grave Lento Largo Larghetto Adagio Andante Moderato

very loud very loud loud fairly loud fairly quiet quiet very quiet extremely quiet less more a very short sound getting louder very slowly slowly broad quite broad in a quiet way at a moderate pace at a moderate speed

De fysica van geluidsgolven Allegretto Allegro Vivace Presto Prestissimo pizzicato* arco* semper

fairly fast fast lively very fast as fast as possible the strings are plucked played with the bow (*string instruments only) 'always' or 'continue' - as a reminder that a previous instruction to play a certain way is still in effect.

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3 The hearing process

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Figure 6 The structure of the ear (after Kessel and Kardon (1979))

The response of the ear to different frequencies This is best represented graphically, as shown in Figure 8. Each line is an equal loudness curve. See below for an explanation from Phons. The reference zero point, a sound pressure level (SPL) of 0 dB, is taken as a pressure of 20 Pa (0.000 02 N/m2).

Frequency discrimination

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The smallest change in frequency that can be detected is known as the frequency difference limen, dl. At sound levels 20 dB above threshold, and for frequencies below 1–2 kHz, and for duration

The hearing process

Figure 7 Cross-section through one coil of the cochlea

greater than 0.25 s dl is fairly constant at 1 Hz. Above 2 kHz, dl increases approximately proportionally to frequency.

Loudness Loudness is a subjective property related to sound intensity. It is generally assumed that with continuous tones a level change of 1 dB is just perceptible. In speech and music, 3dB is the minimum level change that is normally detectable. In general, a 10 dB change in level causes a doubling or halving of perceived loudness. The sone. This is an attempt to have unity proportional to loudness. It is defined by one sound being the loudness at 40 dB above threshold, with 1 sound added for every 10 dB increase in level (or halved for every 10 dB below 40 dB). s 2( p40)/10 where s is the number of sones for a sound pressure of p dB above the threshold.

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Figure 8 Equal loudness curves (from Robinson and Dadson (1956))

The telephone. Phon is defined by the SPL in dB at 1 kHz and the Equal Loudness Curves as shown in Figure 7. So, for example, if we take the 80 phon curve, at 63 Hz the SPL must be about 90 dB to be as loud as 80 dB. sound. at 1 kHz.

Noise Masking The property of a louder signal to make a quieter sound inaudible. This effect is greatest when the two sounds are within the same critical band and when the mask signal is lower than the masked signal. Figure 9 shows the effect of a 90 Hz wide noise band centered at 410 Hz on sounds of other frequencies. When the level of the mask signal is 60 dB, it provides approximately 30 dB of masking for frequencies between approximately 350 Hz and 600 Hz. At a masking noise level of 80 dB, there is approximately 20 dB masking between 260 Hz and 1500 Hz. (Courtesy of Egan and Hale (1950).)

The hearing process

Figure 9 Noise masking

Beats With two tones whose frequencies are close to each other, the signals will alternately go in and out of phase. One of the three sensations is then possible. up to about 10 Hz difference, the low-frequency level changes are observed; between frequency differences of about 10 Hz and about 50-500 Hz (according to the average frequency) there is a feeling of 'roughness'; at higher frequencies, the two frequencies are audible separately. These beats are not difference tones and do not imply non-linearity in the hearing process. However, difference tones can be heard when the two frequencies are in a ratio of 1.1:1 or 1.2:1 to 1.5:1.

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4 Acoustic noise and its measurement Definitions Sound level meters, unless they are very cheap, have mainly two settings, giving measurements in dBA and LIN (or dBC). dBA or dB(A) - both seem acceptable in the UK at the moment. This is a measurement that gives a reasonable approximation of loudness as perceived by the average person. The 'A' weighting network gives the instrument a response that takes into account the ear's lack of sensitivity to low frequencies. See figure 10. dB(C). Some sound level meters, particularly in the US, have a setting marked 'C'. (Older meters may have a 'B' setting. This represents a weighting network that lies between the 'A' and 'C' attributes. This has fallen into disuse.) LIN. Flat response.

Figure 10 A, C and LIN frequency weightings

Acoustic noise and its measurement

Leq,T – equivalent sound level. This is the noise level which, integrated over a time period T, is equivalent to a constant SPL.

Injustice T1

Leq,T = log10

p ~ decibel

p 2(t) dt

If, as usual, measurements are taken with the A weighing net, this is indicated by the caption A: LAeq, . LAeq,T is widely used in assessing the likely effect of industrial and other noise sources, such as roads, on residential areas. For more information, please refer to British Standard BS 4142:1990 ('Assessment of industrial noise affecting mixed residential and industrial areas'). See note on page 27.) Other terms that appear in various aspects of noise measurement include: Ln where n is a number, often 10, 50, or 90, meaning that the SPL indicated by L is exceeded n percent of the time. So if L90 were 60 dBA, the value of 60 dBA would be exceeded for 90 percent of the time considered. LeP,d is very similar to LAeq,T in that essentially the same type of equipment is used to measure it. However, it represents the personal daily 'dose' of noise. In fact, LAeq,T applies to locations, LeP,d to people.

Hearing Impairment The sound levels that put a person's hearing at risk can be a subject of controversy, especially as it appears that different individuals may have different sensitivities. However, it is generally accepted in the UK and much of Europe that there is little risk of hearing damage if the ears are exposed to an LAeq of 90 dB for 8 hours a day. Above 90 dB, the allowable exposure time is halved for every 3 dB increase. See table opposite.

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Permissible exposure times LAeq

Allowed exposure time

90 93 96 99 102 105 108 111

8 4 2 1 30 15 8 4

hours hours hours hours min. min. min

Current health and safety regulations in the UK can be summarized by stating that there are two levels of action: 1. If the LeP,d for the worker(s) is greater than 85 dBA but less than 90 dBA then the employer must inform the employee(s) and ensure that suitable hearing protection is available to anyone who requests it. 2. Above 90 dBA or when the peak sound pressure exceeds 200 Pa (140 dBA), the employer must minimize the noise level by means other than hearing protectors and mark all zones with a noise level above 90 dBA. The above is just an outline and the HSE literature should be consulted for more information.

Sound Level Meters There are four defined classes: Grade 0: The very highest standard and mainly used for laboratory work; Class 1: for precision measurements; Grade 2: for general use; Grade 3: For noise examinations.

Acoustic noise and its measurement

Most meters, except maybe Grade 3, can measure: 'Impulse' or 'Peak' values 'Fast', which is averaged over 1⁄8 s 'Slow', which is averaged over 1 s. Sound measurements in the field should normally be made with at least 1.2m of free space (including the ground) around the microphone. The measured noise must be at least 10 dB above the background noise. If not, a correction should be made as shown in the table below. Sound level measurement, LAeq,T minus background, LA90,T dB

Correction: subtract from sound level measurement dB

6 to 9 4 to 5

1 2

Difficult to correct

Summation of sound levels Sound levels in dB cannot be meaningfully added together because they are logarithms and the result will be a multiplication. Antilogs of the readings should be used as shown below. Note that the noise value must be divided by 10 before finding the antilog, as antilogs of 100 or more will result in an ERROR indication on normal calculators. To take an example: Find the noise level resulting from 92 dBA 98 dBA 96 dBA 68 dBA Noise measurement

Divide by 10

antilog

92 98 96 68

9,2 9,8 9,6 6,8

1.585 109 6.310 109 3.981 109 6.310 106 –––––––––– 1.188 1010

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Log (1,188 1010) 10,075 Result 10 10,075 100.75 dBA (Note that the contribution of the 68 dBA measurement is insignificant, about 30 dB lower than the other measurements.) The above method, modified to account for time duration, can are used to find a LeP,d from a series of measurements of LA eq .

National and International Standards In the list below, the British standards relating to acoustic noise and their measurements are given with equivalent standards in bold. The following abbreviations are used: ANSI

US National Standards Institute

Brits Standards Institute

IEC

International Electrotechnical Commission

ISO

International Standardization Organization

BS 3539: 1986 Sound level meters for measuring noise from motor vehicles *BS 4142: 1990 Method for assessing industrial noise in mixed residential and industrial areas BS 5969: 1981 (IEC 651: 1979) Specification for sound level meters BS 6402: 1994 (IEC 1252: 1993) Personal sound exposure meters BS 6698: 1986 (IEC 804: 1985) Integrating sound meters – averaging BS 7580: 1992 (IEC 645–1: 1992) Specification for verification of sound level meters ANSI S1.4 – 1983 Specification for sound level meters ANSI S1. 25 – 1991 Personal Dosimeter Specification * Most of BS 4142: 1990 is relevant to any outdoor measurement. It can be considered a safe procedure for use in many applications other than industrial noise, such as traffic, entertainment, and animal sounds, such as kennels.

5 Typical noise levels

The measurements below should not be taken as definitive as there can be significant variations between different conditions: for example, traffic noise can vary with the type and speed of vehicles, wear of bearings in industrial equipment can change the noise emission. Nevertheless, the figures given are indicative of the estimated noise levels that may be likely under the stated conditions. Unless stated otherwise, the measurements were made at 1 m from the sound source.

Bron

dBA

Notes

Transport Jet aircraft Jet aircraft, small

140 120

At 30 m, start at 150 m

Train, freight train Train, freight train Train, high-speed train interior

99 96 95 75

Wagons at 12 m Locomotive at 12 m At 12 m from the track Medium-sized train at 120 km/h

Road, moderately busy Road, moderately busy Road, moderately busy Motorway

80 52 49 48

Along the road At 150 m At 300 m 3 km away. Little wind.

Industrial machines Air line Angle grinder Automatic saw Belt sander, joinery Hard soldering

104 105 99 87 94

1 m from nozzle On 4 mm steel 1 m from saw blade On soft wood For the ears of the user

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Bron

dBA

Notes

Gas welding torch Guillotine Hammer, 2 lb Hammer, 11⁄2 lb Hammer, 2 lb Hand grinder Hand grinder Induction furnace Industrial vacuum cleaner Large fan Lathe Milling machine Polishing wheel Electric saws Electric presses

100 85–90 120 116 116 98 110 85

To the ears of the operator On 6 mm sheet steel, on anvil On sheet metal on anvil On steel tube on anvil On box steel On 9 mm sheet steel On 3 m

94 112 85 80 72 100–115 95–110

Agriculture Chainsaw Diesel Rotavator Excavator Excavator Dump Truck, 1 ton

108 97 87 93 89

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Dump truck, 1 ton forklift truck

100 85

Gasoline Rotavator Tractors Tractors Domestic Electric drill Jigsaw, electric Loud radio or TV Normal speech

88 80-90 70-80

100 97 70 60-65

Used when annealing glass 3000 rpm on brass

Fast, unloaded At operator's ear At operator's ear, PTO speed 1 m to side of engine Operator's position, normal speed 1 m at rear, normal speed Operator's position, working speed At operator's ears Inside cab, medium speed Inside the cab, engine running

1 ⁄2″ masonry drill bit in brick Operator's ears, large piece of wood, 30 mm thick

Typical noise levels

Bron

dBA

Notes

Rotary lawnmower with petrol engine. Smoke detector

90 105

In the ears of the operator

Miscellaneous Orchestra, fortissimo Church Bells Sea

100 72 68

Rural area at night

<30

At 5m 35m from base of tower 20m from surf, force 3 off-shore wind No roads nearby

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6 Electromechanical analogies

Summary of analogous quantities, units, symbols and relations The table below shows the general units and dimensions: Mechanical

Acoustic

Electric

Mass, M (kg)

Mass, mother, inertia

Force, F (N) Displacement, x (m) Velocity, x˙ (m/s)

Sound pressure, p - Volume velocity

Energy, work (J) Power (W, J/s) Stiffness, F/x compliance, x/F Frequency, f (l/s,Hz) Resistance, Rm Temperature (K)

joules (J) watts (W) – Conformity, Ca – Sound resistance, Ra –

Length, L

–

Inductance, H, (Henry) Volt (V) Charge Q (C) Current (Q/s,A),i Joule (J) Watt (W) – Capacitance, C – Resistance R Magnetic flux density, B –

Impedance relationships. Mechanical impedance, Zm F/x . Specific acoustic impedance, Zs p/x . Analog acoustic impedance p/x A, where A is the area of the element and p is the sound pressure Admittance 1/Z . Transformations: F Bli; v Blx .. Newton's law: FMx .. Kinetic energy 1⁄2 Mx 2 Potential energy 1⁄2 sx2

Electromechanical analogies

The following table is more complete than the one on page 31 and includes magnetic units. Analog quantity

Symbol

Energy

Units in which it is expressed joules ⬅ newton meter ⬅ coulomb volt watt joule second ampere volt coulomb meter kilogram newton second 2/meter volt ampere coulombs/second

Current

Lading Massa

Potential Current

V i

Transduction coefficient

newton/ampere volt seconds/meter

Magnetic flux Magnetic flux density Magnetic field vector Magnetization Magnetomotive force Magnetic reluctance

B H M F R

Induction

Sensitivity

webers volt seconds webers/meter amps/meter amps/meter amps turns coulombs/s amps turns/weber coulombs/volt seconds2 henries webers/ampere volts second2/coulomb

Electrical impedance Electrical resistance Electrical reactance

Z R X

ohms volts/amps volts seconds/coulomb

power

Busy

Density Stiffness

newton joule/meter coulomb volt/meter pascal newton/meter2 joule/meter3 kilogram/meter3 Ns2/m4 newton/meter coulomb volt/meter

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Analog quantity

Symbol

Compliance

Specific acoustic impedance

Mechanical impedance

Acoustic impedance

FOR

Permittivity of free space Electric field Electric displacement Electric polarization Capacitance

0 E

Units in which it is expressed meter/newton meter2/coulomb volt kilogram/meter4 second coulomb volt/meter4 newton second/meter3 newton second/meter kilogram/second newton second/meter5 kilogram/meter4 second ohm (acoustic) 8.854 1012 (1/36 ) 109 farad/m volt/meter coulombs/meter2 coulombs/meter2 farad coulombs/volt

Electromechanical analogies

Figure 11 shows the acoustic elements and their electrically equivalent circuits. Flow rate profile

SI units

Similar track

128ηl πd 4

16ρl 3π d 2

I) f < f 0

(a hole

(

= 7.58×10–4l d4

((

l = 2,04 d2

(

II) f > f 0 128ηl πd 4

64η πρd 2 3,16 × 10–4 f0 = d2

(

f0 =

(d < b )

(b) Spleen

f f0

7.58×10–4l

4rl pd 2

f f0

((

l = 1,53 d2

(

I) f < f 0 12ηl bd 3

6ρl 5bd

(

= 2,23×10–4l bd 3

((

= 1,44 l bd

(

d II) f > f 0

f0 =

12ηl bd 3

36e prd 2

= 2,23×10–4l f f0 bd 3

f 0 = 1,78 × 10–4/d 2

pl bd

f f0

(

((

= 1,2 l bd

(

V ρc 02

(= 7.04×10–6V)

V = content

(d) Piston diaphragm

S = area m = mass f r = vacuum resonance approx

(e) Radiating piston

S = area

⬃0

m S2

ρc 0 S

0,8 cent pS

( ( = 413 z

S2 4π2 mf r2

( ( = 0,54 s

Figure 11 The acoustic elements, electrically equivalent circuits and R, L and C expressed in terms of the most easily measurable parameters. Viscosity coefficient 1.86 105 kg/(m/s), air density 1.2 kg/m3, speed of sound c0 340 m/s (Poldy 1988)

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Figure 12 Systems with two degrees of freedom

Electromechanical analogies

Figure 13 Vibration isolation analysis. (a) damped mechanical system; (b) equivalent electrical circuit; (c) Insulation factor (ip) plotted against f/f 0 for different degrees of attenuation

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7 Digital Principles

Sampling rate The number of times per second that the signal is 'measured'. A figure of 2.2 times the highest audio frequency is considered practical. The sampling rate for standard digital audio such as compact discs is 44.1, which gives the highest audio frequency that can be recorded and played back as 44.1/2.2 or just over 20 kHz. 48 kHz is an alternative 'professional' frequency. Bit rate is the number of 1s and 0s transmitted or recorded per second and is the product of the number of bits in the sample and the sample rate. So with a 16-bit system and a sampling frequency of 44.1 kHz, the number of bits per second is 705.6 103. The number of bits/sample is closely related to the number of quantization levels – the 'resolution' of the system as shown in the following table. Quantization levels

Number of bits

1 024 2 048 4 096 8 192 16 384 32 768 65 536 131 072 262 144 524 288 1 048 576 2 097 152 4 194 304 8 388 608 16 777 216

10 11 12 13 14 15 16 17 18 19 20 21 22 23 24

Digital principles

Each additional bit doubles the number of quantization levels, halving the noise component. This means there is a 6 dB improvement in the signal-to-noise ratio for each added bit. A fairly good approximation of the signal-to-noise ratio in terms of the number of bits is given by

S = 6N 1.75 dB N The following table of the S/N ratio for different numbers of bits uses the formula above, rounded to the nearest dB. Number of bits

Signal to noise ratio dB

10 11 12 13 14 15 16 17 18 19 20 21 22 23 24

62 68 74 80 86 92 98 104 110 116 122 128 134 140 146

Definitions 1. Dither. The addition of a low level random noise signal to reduce quantization noise. 2. Two's complement. A way to handle positive and negative numbers without resorting to ampersands. In a digital system, the most significant bit (MSB) represents the sign, being 1 for a negative number and 0 for a positive number. Using 10-bit numbers as an example, the positive

Digital Principles 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

range is from 0000000000 to 0111111111 while negative range is from 1111111111 to 1000000000. When a signal crosses the zero line, all bits change from 1s to 0s and vice versa. 3. Aliasing. The production of spurious frequencies resulting from beats formed by the sampling rate and out-of-band frequencies. These can be eliminated by filtering, although very steep 'brick wall' filters may be required (see 'Oversampling' below.) 4. Oversampling. The data is read out two, four, eight, etc. times the sampling frequency. This has the effect of increasing aliasing frequencies by one, two, three, etc. octaves, simplifying filter design.

Digital Signal Processing Mixing – the digital equivalent is the addition of the numbers representing the samples. To deserve. In the digital domain, this means multiplying the numbers. So a gain of 6 dB is the equivalent of multiplying by 2. Dividing by 2 would mean a gain reduction of 6 dB.

Error detection and correction The simplest method of error detection is called 'parity'. The number of ones in the sample is rounded to an even number if necessary by adding an extra 1. (This is 'even' parity: 'odd' parity is sometimes used and then the number of ones is rounded to an odd number.) Parity fails if there are more than two errors, but in a well-designed system this should be a very rare occurrence. When an error is detected, the following options are possible: 1. correct the error; 2. hide the error by, for example, repeating the previous example; 3. mute the error. More complex parity methods make it possible to see which bit(s) are wrong and correct them accordingly.

Digital principles

To illustrate such a process: Original word (12-bit for simplicity): b 1 b 2 b 3 b 4 b 5 b 6 b 7 b 8 b 9 b 10 b 11 b 12

Rearranged: b1

b 10 b 11 b 12

The parity bits P1 etc. can check each column and each row. So if errors are shown by P2 and P4, it's clear that b5 is faulty - if it shows up as a 1, it changes to a 0, and so on.

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8 Acoustics

Resonant frequencies in rooms Rayleigh's formula for the frequencies of modes in a rectangular room:

fm=

c 2√(nL/L) (nW /W)2 (nH/H)2 2

where nL, nW and nH are positive integers including 0, L, W and H are the room dimensions and c is the speed of sound. Modal density. If the number of resonance modes N below the frequency f is reasonably high, then a reasonable approximation is given by

N ≈ 4 f 3 true

V S fL f 2 2 3c3 4 8c

V is the volume, S is the total area, 2(LW WH HL) and L is the sum of the edge lengths 4(L W H).

Reverberation time (RT, r.t. or T60) This is defined as the time it takes for the reverberation sound level to decrease by 60 dB. An approximate expression, called the Sabine formula, valid for rooms with small amounts of absorption is:

T60 = 0,16

In Sa

where V is the volume in m3, S is the total surface in m2 and a¯ is the average absorption coefficient of the surfaces. A more accurate formula, applicable in areas with large amounts of absorption, is due to Eyring and Norris:

T60 =

0.16V Sun (1 a)

Acoustics

Typical absorption coefficients Absorption coefficient, a, is the fraction of the incident sound energy that is absorbed. For complete, 100 percent absorption a 1; for no absorption a 0. Frequency Hz Material

500

1 k

225mm masonry

0,05

0.04

0.01

0.0

113mm masonry

0,10

0,05

0.0

Windblok van 75 mm

0.09

0.16

0.0

Wooden panels of 12 mm on slats of 25 mm

0,33

0,10

0,12

glass (> 6 mm thick)

0.03

3 mm hardboard on 25 mm slats

0.30

0,43

0.07

0.11

Brick (surface)

0.02

0.04

0.07

Raw concrete

0.01

0.02

0.06

0,10

Happy concrete

0.01

0.02

0,05

Smooth plaster (painted)

0.01

0.02

Hout

0,05

0.07

0,10

0,12

Lino

0.02

0.03

0.04

Rubber floors

0.01

0.03

0.04

0.02

Haircord carpet on underfelt

0,05

0.17

0,29

0.30

Wilton carpet on underfelt

0.04

0,22

0,64

0,71

Typical carpet tiles

0.01

0.11

0,39

0,55

Curtains, velvet, draped

0,05

0,31

0,80

0,65

Lightweight fabric with over 50mm air space

0.0

0,10

0,50

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Frequency Hz Material

500

1 k

25 mm mineral wool, 5% perforated hardboard cover

0.03

0,47

0,90

0,31

As above but with 25mm air space

0.04

0,65

0,90

0,31

As above, but with 175mm of air space

0,35

0,89

1.02

0,44

50mm mineral wool, 5% perforated cover

0,10

1.10

0,90

0,31

50mm mineral wool over 150mm airspace

0,60

0,95

0,81

0,85

Audience (units/person)

0,15

0,40

0,45

Orchestra (units/person including instruments)

0,20

0,85

1.39

1.20

Typical recommended reverberation times Activity

T60 (s)

Comments

speech (1)

0,6–1,2

e.g. Lecture Halls, Council Rooms Conference Rooms

speech (2)

1,0–1,4

Theaters

reproduced sound

0,8–1,2

Cinemas

Multifunctional use

1,0–1,5

School halls, multi-purpose halls, community halls

Acoustics

Recommended reverberation times for broadcasts

Figure 14 Sound studio reverberation times (BBC)

Figure 15 Television studio reverberation times (BBC)

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9 Sound insulation

Sound design criteria Single digit values in dBA cannot be considered reliable for design purposes, as they do not give an indication of the frequency spectrum. The list below may nevertheless be useful to provide an approximate indication of loudness (dBA) and also approximate NC and NR (see Figures 16 and 17). Environment

NC/NR-index

Equivalent dBA

Entertainment: Concert hall, theater Lecture hall, cinema

20–25 25–30

30–35 35–40

Hospital: Operating room Multi-bed ward

30–35 35

40–45 45

Hotel: Individual room, suite Banquet hall Corridor

20–30 30–35 35–40

30–40 40–45 45–50

Retail, restaurants: Restaurant, department store Café, cafeteria Shop

35–40 40–45 40–45

45–50 50–55 50–55

Industry: Light engineering workshop Heavy engineering workshop

45-55 50-65

55-65 60-75

Sound insulation

Environment

NC/NR-index

Equivalent dBA

25 30

35 40

Offices: Open office Drawing room Boardroom Boardroom

35 35–45 25–30 30–35

45 45–55 35–40 40–45

Public buildings: Court Library, bank, museum Aula Sportarena Church

25–30 30–35 25–35 40–50 25–30

35–40 40–45 35–45 50–60 35–40

Educational: Classroom, lecture hall Laboratory

25–35 35–40

35–45 45–50

Domestic: private house, bedroom private house, living room

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Sound classification curves

Figure 16 ISO noise classification (NR) curves

Sound insulation

Figure 17 Sound criteria (NC) curves

Criteria for broadcast studios

Figure 18 Background noise criteria for BBC studios (i) relaxed criterion; (ii) normal criterion; (iii) radio drama studios

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Airborne sound insulation Formula for transmission loss introduced by a partition wall or barrier with surface mass m/m2 at frequency f:

pr 0 c = ps fm where pr and ps are the sound pressure levels at the receiving and transmitting sides of the barrier and 0c is the characteristic impedance of the air. Alternatively, the noise reduction index R is given by R 20 log (fm) 47 dB

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Influence of reverberation time on SRI When sound is transferred from one room to an adjacent room, the effective SRI, R, is affected by the reverberation time in the receiving room. Than

R = Ps Pr 10 log

ST 0.163V

where Ps and Pr are the sound pressure levels in the source room and the receiving room respectively, S is the area of the wall separating the two rooms, T and V are the reverberation time and the volume of the receiving room.

Sound Insulation Performance for Building Materials (Average 100-3 kHz) The figures in the following tables (pages 50-52) assume source and receiver are both rooms. If either is in the open air, subtract 5 dB from the specified value as an estimated correction.

Sound insulation

Material WALLS Light unplastered blockwork Light plastered blockwork Solid 100 mm unplastered brick Solid 100 mm brick with 12 mm plaster Dense dense concrete, 150 mm

approx. sound level reduction (dB) 35 40 42 45 47

Solid unplastered masonry, 230 mm Solid plastered masonry, 230 mm Dense concrete, sealed and plastered, 200 mm Plastered brick, 450 m

48 49 50

225mm double sided brick wall with 50mm cavity

WOOD FRAME CONSTRUCTIONS 50mm frame with 12mm plasterboard on each side 50mm frame filled with quilt and 2 12mm plasterboard on each side As above but 75mm frame FLOORING 21mm planks or 19mm chipboard 110mm concrete and screed 21mm planks or 19mm chipboard on plasterboard and 50 mm sand putty 200 mm reinforced concrete and 50 mm screed 150 mm concrete on special floating raft

33 41 45

35 42 45

47 55-60

Sound insulation 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Material

approx. sound level reduction (dB)

DOORS Hollow core, properly fitted panel door, no seals As above with seals Solid core, properly fitted door without seals As above, with seals or cut close to carpet Solid core door, 60mm with good seals

15 20 15 25 30

Typical noise reduction indices Material

Octave band center frequency (Hz) kg/m

4 mm glass 6 mm glass 6.4 mm glass laminated 12 mm glass

125

250

500

1 k

10 15

20 18

22 25

28 31

34 36

34 30

22 26

24 30

30 35

36 34

33 39

21 20 26 26

20 19 26 34

22 29 34 40

29 38 40 42

35 36 39 40

Double glazing, closed units: glass/airspace/glass 3/12/3 6/12/6 6/12/10 6/20/12 Single-leaf brick, plastered on both sides 240 Cavity masonry with tension bands 480 Double-leaf masonry, plastered on both sides 480

Sound insulation

Material

Octave band center frequency (Hz) kg/m

Three-leaf masonry plastered on both sides

125

250

500

1 k

22 50

24 27

29 35

31 41

32 39

30 39

720

9mm plasterboard on 50 studs of 100mm at 400mm centers 13mm plasterboard on studs as above 13mm plasterboard as above with 25mm mineral wool between studs 9mm ply on frame 25mm T&G wood boards Two plies Gypsum board of 13 mm 6 mm steel plate 43 mm recessed hollow door, normal hanging 43 mm solid core door, normal hanging 50 mm steel door with good seals

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10 microphones

Basic Requirements Before purchasing microphones for high quality applications such as recording or broadcasting, the following checklist should be considered.

Sound quality 1. Is the frequency response satisfactory? Ideally, this should be ±2 dB over the 20 Hz to 20 kHz range, but tighter tolerances may be acceptable at the high and low frequencies. For some specialized purposes, a non-flat response can have advantages: for example, a lower bass response can be an advantage in public address situations. 2. Temporary response. This is not easy to judge except through listening tests, which should be done with a wide variety of sound sources. 3. Sensitivity. Is the electrical output sufficient taking into account the situations in which the microphone will be used? 4. What is the self-generated sound? This is sometimes quoted as the equivalent acoustic noise in dBA. With digital recording work, the sound level of the microphone should of course be as low as possible. 5. What is the maximum sound pressure level that can be handled without distortion? (Note that the limit is generally determined in the first stage of amplification, which is in the microphone itself in the case of electrostatic microphones.) 6. How much variation is there between different examples of microphones of the same type? In low-cost devices, this can be significant.

Microphones

Directional response 7. What is the directional pattern (polar diagram)? 8. How much does this vary with frequency and does it depend on the orientation of the microphone: i.e. is it different in the vertical plane than in the horizontal plane? 9. Is the directionality fixed or variable? As variable is the controls on the body of the microphone or is there a remote control? What patterns are there and how good are they?

Physical 10. Physical dimensions. 11. Weight. 12. Appearance, including color and finish: e.g. matte or glossy.

Reliability 13. Can trouble-free service be expected? This can probably only be determined by referring to other users with experience with this microphone. 14. Does it seem robust? 15. Are there reasons to believe that deterioration could occur with age and use? 16. How likely is it that maintenance can be performed on site, how available are spare parts, what repair service, including lead time, can the manufacturers provide?

Vulnerability 17. Is it susceptible to wind noise, 'pops', etc.? 18. Is there a windshield? If so, how effective is it (are they)? 19. To what extent is the microphone affected by humidity and temperature?

Microphones 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

20. Is it reasonably immune to external magnetic and electrostatic fields, including r.f. to retrieve? 21. How sensitive is it to the effects of vibration, handling noise, rustling of cables, etc.?

Electrical 22. What is the optimum electrical load? 23. What power supplies does it need, if any? 24. If the battery can run on batteries, what is the battery life and how readily are the batteries available? 25. Connectors. Are these standard? 26. Is there built-in frequency correction, e.g. a bass suppression switch?* 27. Is there a built-in, switchable attenuator?* (*The absence of these features does not necessarily mean an inferior microphone, as it is quite possible that there will never be a need for them.)

Miscellaneous 28. What types of mounting are there? 29. What are the costs? 30. What is the reputation of the manufacturer? 31. If made abroad, are there any good agents in this country?

Microphone sensitivities in common units See table on next page.

Microphone transducers Moving coil The sensitivity of the average moving coil microphone is such that an e.m.f. of the order of 0.5 to 1 mV is produced with normal speech at a distance of 0.5 m.

Microphones

Microphone sensitivities in commonly used units dB relative to 1 V/Pa

mV/bar

mV/10bar

9.5

25 30 35 40 45 50 55 60 65

5,5 3,0 1,8 1,0 0,55 0,30 0,18 0,10 0,055

55 30 18 10 5,5 3,0 1,8 1,0 0,55

Figure 19 Moving coil microphone (simplified)

Notes Typical electrostatic microphones

Typical moving coil Some ribbon microphones.

Microphones 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Typical Characteristics: 1. Common directional patterns are cardioid, omnidirectional, or gun. Figure-eight answers require two back-to-back units and have rarely been satisfactory. 2. Fairly immune to humidity and temperature. 3. Moderately robust. 4. They have the advantage of not needing any food.

Ribbon

Figure 20 Baseband microphone

The ribbon impedance is usually of the order of 1 and a built-in transformer is used to raise the impedance at the output to a much higher value. This has the added benefit of increasing the output voltage, yet it is small, being less than 1 mV for normal speech at a distance of 0.5 m. Typical Characteristics: 1. The most common directional patterns are figure eight or hypercardioid. 2. Fragile. The ribbon must provide effective shielding against air blasts in the microphone housing. They are therefore not suitable for outdoor use.

Microphones 3. Sensitivity is almost always low. 4. A special use is the 'lip ribbon' microphone used for commentary in noisy environments (see later).

The combination of low sensitivity, relative fragility and a limited range of directional patterns means that ribbon microphones are rarely used, the lip ribbon being an exception.

Electrostatic In simple form, the basic circuit for an electrostatic (sometimes called "condenser" or "capacitor") microphone is shown in Figure 21.

Figure 21 Essential circuit for an electrostatic microphone

The distance between the diaphragm and the back plate is typically about 0.02 mm, creating a capacitor with a value of about 20 pF. the d.c. supply can be in the range of 50 V to 100 V, where R has a very high value of several hundred. The effect is that the CR combination acquires a relatively long time constant. Then, with a variation of C in the capacitance, there is a change in voltage given by VQ C Electrets - materials with a permanent electrostatic charge are often used for the backplate or diaphragm and this makes a polarizing voltage unnecessary. r.v. electrostatic – the capsule is part of an LC circuit which in turn is part of an r.f. discernment. The output of an oscillator, typically about 8 MHz, is fed into the discriminator. The output of

Microphones 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

the latter is an audio signal representing the variations of the capsule's capacitance. The advantage of this type of transducer is that it is largely unaffected by humidity, although the cost is likely to be high (see Figure 22).

Figure 22 Simplified r.f. electrostatic microphone

Typical characteristics of electrostatic microphones. 1. Good frequency response due to the lightness of the diaphragm. The high-frequency response is generally very good because the diaphragm is under tension, resulting in a high resonant frequency. 2. High sensitivity. 3. All kinds of directional patterns (polar diagram) can be produced. 4. Susceptible to moisture problems, although careful drying in a warm environment can generally restore normal operation within 30 to 60 minutes.

Directional Patterns (Polar Diagrams) Omnidirectional Essentially the result of pressure acting on the diaphragm – i.e. the force on the diaphragm depends only on the sound wave pressure and not a derivative thereof.

Microphones

'Omni' microphones are only truly omnidirectional if the wavelengths of the incident sound are large compared to the microphone diameter.

Figure 23 Directional patterns for a typical 'omni' microphone

A useful approximate calculation to find the frequency to which the microphone is omnidirectional results from finding the frequency corresponding to the diameter of the microphone. The mic will be fairly omnidirectional up to frequencies between two and three octaves below that. For example, a microphone has a diameter of 2 cm. The frequency corresponding to a wavelength of 2 cm is 17 kHz. An octave below that is about 8 kHz; two octaves, 4 kHz; three octaves, 2 kHz. So this microphone will be fairly omnidirectional up to about 3 kHz. Phase cancellation, i.e. a reduction in sensitivity to sounds coming from the side, can also occur when there is a full wavelength, or about so, across the aperture, as shown in Figure 24.

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Figure 24 Phase Cancellation

Figure 25 shows the frequency response under different angles of sound for a typical omni microphone.

Figure 25 Typical frequency response graphs of an omni microphone

Microphones

General characteristics of omni microphones are: 1. Usually less noisy and noisy than other types. 2. Do not show proximity effects (see below image of eight microphones). 3. Although they offer no advantage in suppressing sounds from certain angles, their relatively constant sensitivity, except at high frequencies, can sometimes be useful.

Figure-eight microphones The force on the diaphragm is due to the pressure gradient (pressure difference) between the two sides. A figure-eight pattern can be represented by rcos, where r is the effective sensitivity at angle.

Figure 26 Figure-of-eight polar diagram

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The pressure gradient can be calculated from

FG2 = Ff2 Ft2 2Ff Ft cos where FG is the pressure gradient, Ff and Fr are the forces on the front and back of the diaphragm respectively, and is the angle found from

Tsukasa dfc360

d is the acoustic distance between the front and back of the diaphragm, f is the frequency, and c is the speed of sound. Figure 27 illustrates the above.

Figure 27 Method for determining the pressure gradient

If the distance from the sound source is large compared to d so that effects of the inverse square law can be neglected, the expression for FG simplifies to

FG2 = 2Ff2(1 cos ) The variation in pressure gradient with frequency is illustrated in Figure 28.

Proximity effect (“Bass tip-up”) This occurs when the sound source is close to the microphone, so Ff and Fr are not equal due to inverse-square law effects and the frequency f is low, so it is small. The result is that the microphone's output becomes exaggerated for low-frequency sounds (see Figure 29).

Microphones

Figure 28 The variation in pressure gradient with frequency

Figure 29 Proximity effect

The effect is used to good advantage in certain microphones used by out-of-sight radio or television commentators – the so-called 'lip ribbon' microphone. Bass cut restores the voice to an approximately normal response, but reduces the level of distant l.f. sounds. The table below gives the increase in microphone output over the output at 10 kHz for a typical microphone of this type. General characteristics of figure-eight microphones: 1. Very sensitive to rumble and vibration.

Microphones 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Proximity effect for a pressure gradient microphone (for a point source, 50 mm from the daphragm) Frequency (Hz)

Relative output (dB)

50 100 200 500 1k 2k 10k

26 20 14 7 3 1 0

2. Show proximity effect (see above). 3. The pattern of the eight is generally quite well preserved in terms of frequency, although the nulls at 90° to the axis can be ill-defined at some frequencies.

Cardioid

Figure 30 Cardioid pattern

Microphones The cardioid pattern in Figure 30 is a graph of r 1 cos

In practice, this is rarely achieved with a "cardioid" microphone. The frequency response graph of a typical high performance unit is shown in Figure 31 showing that the 180° response, far from being zero, cannot be more than about 10 dB below the axial response at low frequencies, and possibly even worse at high frequencies.

Figure 31 Typical cardioid frequency response graphs

General characteristics of cardioid microphones: 1. Usually exhibit a proximity effect. 2. Tends to be sensitive to rumbling and vibration effects.

Hypercardioid The term "hypercardioid" is often used for any microphone with "blind" spots in the rear quadrants, but the most common usage is for blind spots at 45° from the 180° axis. (See Figure 32.) The equation is in that case

√2

cos

Microphones 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 32 Hypercardioid pattern

Hypercardioid microphones can be thought of as halfway between figure-eight and cardioid microphones, and their general characteristics reflect this. General characteristics of hypercardioid microphones: 1. Proximity effect is usually less than for cardioid but more than eights. 2. More prone to rumble effects.

Variable Directivity Microphones The capsule of a variable directivity microphone typically consists of two back-to-back electrostatic cardioid microphones, with the potential at the diaphragm of the rear unit variable, as shown in Figure 33.

Highly directional microphones Interference tube microphones ('gun') (see Figure 34). Typical polar diagrams are given in Figure 35. Interference tube microphones about 250 mm in length often have polar patterns that are almost hypercardioid in shape.

Microphones

Figure 33 Simplified circuitry for a microphone with variable directivity pattern

Figure 34 Simplified interference tube

Microphones 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 35 Typical directional characteristics for an interference tube microphone approximately 500 mm in length

Phantom Power Systems 1. 48 V standard systems (sometimes indicated with a "P" in the type number).

Figure 36 48 V phantom power – the power side

Microphones

The resistors R, in figure 36, are for current limiting in case of short circuit. A common value is 6.8k.

Figure 37 48 V phantom power – microphone side

2. A-B power supply

Figure 38 A–B power supply

Microphones 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

A-B power is generally a low voltage method - typically 7-9 V - used with a few microphones, while the 48 V system is usually applied to all microphones in a studio. It has the advantage that only two wires are needed. The downside is that a phase reversal will result in the microphone not working until the reversal is corrected.

11 Radio Microphone Frequencies VHF Frequencies News Frequencies

Shared Frequencies

Coordinated Frequencies 173,800 174,100 174,500 174,800* 175,000

176.800

175.250 175.525 176.600

Deregulated frequencies 173.800 174.100 174.500 174.800* 175.000

176.400 177.000

184.600 184.800 185.000 191.700 192.100 192.600

191.900 192.800 193.000

199.900

199.700 200.300 200.600

200.800 201.000 207.900 208.800 216.300 217.000

208.300 208.600 209.000 216.100 216.600 216.800

192.300

200.100

207.700 208.100

*This frequency is not fully compatible with the other coordinated/deregulated frequencies and may result in intermodulation.

Radio microphone frequencies 73 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

The information in this chapter is reproduced with permission from John Willets, MIBS, of Sennheiser UK and the Institute of Broadcast Sound. Radio microphone frequencies are changed from time to time and the reader is advised to contact JFMG (the licensing authority) whose address is given at the end of this section. In the table on page 72, news frequencies are what the term implies - they can be used by anyone engaged in newsgathering. Shared frequencies are for general use all over the UK, but before using a radio microphone, make sure no one is nearby using the same frequencies. Coordinated frequencies are for use in predetermined locations and must be pre-approved with JFMG. Deregulated frequencies can be used by anyone with type-approved equipment. Shared Frequencies:TV Channel 69: Frequency 854.900 855.275 855.900 860.400 860.900 861.750 856.175 856.575 857.625 857.950* 858.200 858.650 861.200 8 61.55 0*

Remarks For general indoor or outdoor use

冧

Users should be careful not to interfere with adjacent TV services

* While these frequencies are legal, they can cause problems with some equipment because they are close to adjacent channels.

Radio microphone frequencies

Coordinated Frequencies for the London Area This table lists only the channels assigned to radio microphones. The full allocation list is available from JFMG, as are similar lists for many other parts of the country. TV channel

Notes

36 38 39 42 43

Shared with Radar Shared with Radio Astronomy May be available with channel 38 3rd choice for new assignments 3rd choice for new assignments

44 45 46 47 48

3e 3e 3e 3e 3e

56 57 58 59 60

2nd choice for new assignments 2nd choice for new assignments 2nd choice for new assignments Sufficient for continued use Sufficient for continued use

61 66 67 68

Sufficient for continuous use 1st choice for new jobs 1st choice for new jobs 1st choice for new jobs

choice choice choice choice choice

before before before before before

new new new new new

commands commands commands commands commands

Radiomicrofoonfrequenties 75 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

License It is a requirement that a station is licensed for use and that the frequency is cleared with JFMG Limited 72 Upper Ground London SW1 9LT Tel: 020 7261 3797 Fax: 020 7737 8499 Email:[emailprotected]Website: www.jfmg.co.uk

12 speakers

Moving coil drive units

Figure 39 Basic elements of a moving coil unit

The magnitude of the force on the coil is given by F BIl newton, where B is the flux density (Wb/m2), I is the current (A) and l is the coil length (m). The speed of the coil/cone is found from v F/Z m/s where F is the force (N) and Z is the mechanical impedance (mech. ).

Loudspeakers 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 40 Basic magnet configurations: (a) conventional center 'pot' magnet; (b) flat shape used with ceramic magnets

To avoid distortion in large amplitude signals, and thus large coil movements, it is necessary that the length of the coil is effectively constant within the magnetic field. Figure 41 shows typical setups. Peak excursion v frequency. Figure 42 shows the relationship for different cone diameters radiating 1 W.

Ribbon loudspeaker Similar in basic design to the ribbon microphone (see there). It is difficult to make them efficient at low frequencies, so they are mostly used as tweeters.

Piezoelectric Loudspeaker Again used for high frequency units, these exploit the property of certain materials to undergo physical deformation when voltage is applied to them.

Multiple Driver Units Because it is nearly impossible to satisfactorily cover the entire audio range with one drive unit, it is common practice to use two or more units, each designed to cover a specific frequency range. 'Crossover units' act as band-pass filters to ensure that each unit is fed only the correct frequencies. (Continued on page 80)

Loudspeakers

Figure 41 Typical coil/gap setups. (c) is acceptable in small stroke systems such as mid-range units and tweeters

Loudspeakers 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 42 Peak coil excursions against frequency for different coil diameters to radiate 1 W

Figure 43 Crossover networks: (a) high level or passive; (b) low level or active

Loudspeakers

(Multiple drivers, continued on page 77) Broadly speaking, there are two ways to power such systems: "high level" or passively, where there is one power amplifier but each filter circuit must handle a high power signal, Figure 43 (A) ; 'low level', or active, where the filters operate at low level and there is an amplifier for each drive unit, Figure 43(b). This has the advantage that relatively small and cheap components are used in the filters and this can outweigh the costs of the extra amplifiers.

Figure 44(a) First-order crossover network with a slope of 6 dB/octave; (b) second-order crossover network with a slope of 12 dB/octave; (c) Third-order crossover network with a slope of 18 dB/octave

Loudspeakers 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Types of Crossover Networks Figure 44 (a), (b), and (c) (page 80) show the basic circuits for networks with roll-off slopes of 6, 12, and 18 dB/octave. Corresponding response curves are given in Figure 45. (Note that the circuits shown in Figures 44(a), (b) and (c) assume that the nominal impedance is the same for all drivers and constant at all frequencies - i.e. resistive .)

Figure 45 Response curves for first, second and third order crossover networks

Baffles and enclosures 1.

Flat shot

Cheap and basic but poor bass response. The approximate lower limit of the frequency is given by L 2 where L is the shortest dimension of the shot and the wavelength. For a square shot of side 1 m this gives a lower frequency limit of about 700 Hz.

Closed box (also called "infinite baffle")

The bass response is not as good as expected, because the trapped air acts like a spring, the stiffness of which raises the bass resonance frequency of the system. Nevertheless, with careful design, highly effective small loudspeakers are produced with closed enclosures.

Loudspeakers

Ventilated housing (‘bass reflex’)

The entire housing contains a vent, creating a Helmholtz resonator tuned so that the air in the vent is in phase with the cone of the main drive unit. An auxiliary bass resonator (ABR) is sometimes used instead of a vent. This is a passive device, flat or cone-shaped.

Transmission line (‘labyrinth’)

Back-radiation from the cone passes through a long pipe lined with sound-absorbing material. The design must be such that the sound emerges at the end in phase with the cone's radiation. Theoretically, the line should be a quarter wavelength long at the bass resonance frequency, i.e. for the lower frequency limit of 30 Hz, the line should be about 3 m long. Careful damping and tapering are necessary to prevent the effects of standing waves.

Figure 46 Speaker housings. (a) closed box; (b) ventilated housing; (c) ventilated housing using ABR; (d) transmission line

Load horn

A properly flared horn can greatly increase a loudspeaker's efficiency – up to 5 percent compared to possibly only 1 to 2 percent possible with other systems. Dimensions and flame speed are critical and the horn itself should not vibrate (as can happen with the metal horns often found on racetracks, for example). For good bass reproduction, the horn must be several meters long.

Speaker Impedance and Frequency This varies greatly with frequency, as shown in Figure 47. The stated impedance, often 8 , is usually the value measured at 400 Hz. This is often about 20 percent higher than the d.c. coil resistance.

Loudspeakers 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 47 Impedance versus frequency for a typical closed-loop loudspeaker

The most commonly quoted and most useful indication of a loudspeaker's sensitivity is the sound pressure level (SPL) at 1 m distance in an open field for a given input signal level. The latter can be 2.83 V, which corresponds to 1 W at a nominal impedance of 8 . Alternatively, the peak SPL at 1 m before significant distortion occurs can be given. For professional monitor speakers, 120 dBA is a typical rating.

13 Stereo

Time-of-arrival (TOA) difference This is the difference in arrival time of a sound at the listener's two ears. It is probably the most important factor in determining the direction of a sound. It is zero for sounds coming directly in front of the listener and is a maximum of about 0.6 ms (depending on head size) for sounds arriving at an angle of 90° to the frontal axis. In general, the TOA difference is proportional to sin in Figure 48.

Figure 48 Difference in arrival time, t is given by t sin /c where c is the speed of sound in air

Stereo 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Pure tones are generally very difficult or even impossible to locate and there must be a low-frequency component. This may take the form of a modulation of an otherwise stable tone.

The Haas Effect If the same sound is emitted from two sources, as would be the case, for example, if a mono signal were fed into two similar loudspeakers, the brain appears to fuse the two signals together, provided the time difference between them is less than approximately 50 ms. There is a discernible time interval if the difference is greater than approximately 50 ms. With differences of less than 50 ms, not only do the two sources appear as one, but the apparent single source tends to the origin of the first arrival. In order for the delayed source to appear as loud as the real time source, it must be higher in level by an amount that depends on the time delay. The relationship is shown in Figure 49.

Figure 49 The 'Haas effect' curve regarding time delay and level difference for two sound sources to appear to have the same loudness

Stereo

Sound image positions with two speakers TOA differences, while very important for the location of sounds in real life, are not applicable in the case of two-speaker stereo reproduction. It can be shown that the position of the sound image depends on the 'difference between channels', i.e. the number of dB's difference in level between the left and right signals. The relationship is shown in Figure 50.

Figure 50 Relationship between difference between channels and image position

Figure 50 was originally created from speech sources, but seems to apply to other types of sound. The shaded area indicates the disagreement that can arise between different listeners. Interestingly, if the signals to the speakers are swapped so that the left signal goes to the right speaker and the right signal goes to the left speaker, the shape of the curve is different for many listeners. That is, it is not a mirror image of the original.

Stereo 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Stereo Terminology Channel Application

Links

To the right

White (or no color)

Rood

Indication in broadcast equipment

Name used in some stereo microphones

Color in broadcast equipment*

Rood

Vegetables

Designation in home equipment Color in home equipment (including headphones)

* Note that the broadcast colors follow the color convention used for marine navigation lights.

M and S Signals The M signal is essentially the sum of the signals in the two channels; the S signal is the difference: MAB

SAB

In practice, a correction is applied. The addition of A and B, if they are identical as in a central stereo image, then M is 6 dB higher than A or B. The correction applied depends on the circumstances and sometimes on the broadcaster – it could be 3 dB or 6 dB . Hence M A B 3 dB

M A B 6 dB

SAB 6 dB

Evenzo S A B 3 dB

Stereo

Derivation of M/S signals from A/B and vice versa Two ways to do this are shown in Figure 51.

M An electrical method using transformers wired in-phase for M and out-of-phase for S

A summing amplifier

M Electronic method with two summing amplifiers and an inverter (phase reversal)

B Inverting amplifier

Summary amplifier

Figure 51 Conversion of stereo signals from A/B to M/S (or M/S to A/B).

Microphones for stereo 1 Coinciding pair of microphones, A/B (or XY) configuration Both microphones have, as far as is practical, identically similar characteristics and their diaphragms are mounted as close together as possible (ideally they should coincide in space). As shown in Figures 52 and 53, for two figure-eight microphones, conversion from an A/B system to an M/S system can be accomplished by rotating the microphone 45°! The out-of-phase regions may or may not be significant. Individual sound sources in these areas will appear in the stereo image as difficult to locate. The reverberation, for example in a concert hall, is diffused enough not to adversely affect the components of the reverberation entering the microphones from the out-of-phase corners.

Stereo 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 52 Polar diagrams for a pair of coinciding microphones in the figure of eight. Note the out-of-phase quadrants

Figure 53 The equivalent M/S charts for Figure 52

Approximate Stereo Imaging Positions The following tables assume well-matched coincident-pair microphones positioned at 90° angles to each other. The positions of the sound images are based on well-matched speakers and good listening

Stereo

conditions. They can only be approximate because listeners may differ in the placement of such images (see Figure 50). The angle of incidence of the sound is relative to the centerline between the microphones.

1. Figure-eight microphones Sound angle (°)

Approximate image position

45 30 15 0 15 30 45

Full Left 3/4 L 1/2 L Middle 1/2 R 3/4 R Full Right

The level of the M signal is approximately 3 dB at 45° relative to the level at 0°.

2. Hypercardioid Note that the polar diagrams for most hypercardioid microphones vary considerably with frequency. The figures in the table below are nevertheless probably reasonable indications for most speech and music. Angle of sound incidence (°)

Approximate image position

75 60 45 20 0 20 45 60 75

Full Left 3/4 L 1/2 L 1/4 L Center 1/4 R 1/2 R 3/4 R Full Right

Stereo 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

The level of the M signal is about –3 dB at about 60° to 70° compared to the level at 0°.

3. Cardioids Note that like hypercardioids, the polar diagrams for most cardioid microphones vary considerably with frequency. The numbers in the table are, again, probably fair indications for most speech and music. Angle of sound incidence (°)

Approximate image position

90 60 30 0 30 60 90

Full Left 1/2 L 1/4 L Middle 1/4 R 1/2 R Full Right

The level of the M signal is about 3 dB at about 75° compared to the level of 0°.

4. M/S Microphones In principle, any M/S microphone should consist of a side-to-side figure-eight microphone with a front-firing microphone that can have any pattern in principle. A common system consists of a forward-firing cardioid with the inevitable side-eight (Figure 54). An M/S signal must sooner or later be converted into an A/B signal, especially for auditory monitoring. Methods to do this are illustrated in Figure 55. The width of the stereo image depends on the amplitude of the S-signal – if it is zero, there is no stereo component and the result is mono. A method of obtaining an A/B signal from M/S using three channels on a mixer is shown in Figure 55, where a fader in the third channel controls the S signal and is thus a width control.

Stereo

(Video) How To Win Friends And Influence People Audiobook

Figure 54 (a) M/S microphone, forward facing cardioid; (b) the A/B equivalent - two cardioids at 90°

5. Panpot Systems The output of each microphone is placed in the stereo image by a 'panpot' which is essentially a potential divider that splits the signal between the A and B channels according to the position of the wiper. It is of course possible to use coincident pair microphones and pan-pot microphones at the same time.

Stereo 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 55 Derivation of A/B from M/S using three mixer channels, with the fader on the 'spare' channel providing width control

Figure 56 Panpot's law

14 Analog sound mixing equipment Levels and dynamic range 1. Low level signals. These are generally accepted as the levels of microphones operating under average conditions. This can range from 70 dBu (taking the unit dBu as a reference 'zero level' or 0.775 V) to about 50 dBu. 2. High level or 'line level' signals. Signals in the range of approximately 15 dBu to 20 dBu. The terms "low level" and "high level" are not intended to be precisely defined. 3. Dynamic range. The range in dB between the lowest and highest program level. The lowest acoustic level likely to be encountered is about 20 dBA; the highest can be 110 dBA or more. The acoustic dynamic range is therefore 90 dB. Digital recording systems such as compact discs can handle dynamic ranges of about 90 dB; a.m. radio, about 45 dB, while a.m. radio is limited to 30 dB or even less.

Balanced and Unbalanced Circuits The difference is illustrated in Figure 57. The important point of balanced circuits is that the two legs are identically equal electrically, so that induced interference will ideally produce exactly equal voltages in each leg. These are in opposition and will therefore 'cancel'. Perfect balancing is almost impossible, if only because the two conductors in the cable can never have exactly the same distance from the sources of interference. Twisting the conductors improves matters and one type of microphone cable ('star quad') uses four twisted conductors, with opposite conductors used for each leg. Such an arrangement gives very good immunity to the effects of interference, yet 100 percent protection can never be guaranteed.

Analog sound mixing equipment 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Import

Exit

Chassis ground

(A)

Figure 57 (a) unbalanced; (b) balanced circuits

Input Impedances Microphone: Channel input impedances are normally about 1k. Channel: Input impedances are normally 10k.

Jack plugs The jacks most commonly used in audio work are 6.35 mm (1⁄4 inch) in diameter. There are two commonly used types: A track. These can be two-pole (unbalanced) or three-pole (balanced). The latter can be used for balanced microphones or unbalanced stereo. Its use is limited to domestic and non-professional applications. B size (IEC 60268–12). These are used in professional applications, are always in the three-prong configuration and are made of solid brass. Note that the tip and ring are smaller in diameter than the sheath. They are therefore not interchangeable with A-gauge jacks.

Analog equipment for mixing sound

Figure 58 Jack plugs and sockets

Modes of Interconnection 1. 'Normalize'. Connections are semi-permanent, but may need to be interrupted. 'Full normalling' requires jacks to be plugged into both connections. 'half-normalling' only needs one jack to break the circuit. See Figure 59. 2. A 'listen' socket allows a plug to be inserted without breaking the circuit.

Analog sound mixing equipment 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 59 (a) Half normalization; (b) full normalization

Channel Facilities Figure 60 shows a typical mono channel. This does not mean that the mixer cannot process stereo signals, but that a mono channel, for example, could be used for half of a stereo source, so two channels would effectively be one stereo channel. In such use, the panpots for the two channels would likely (but not necessarily inevitably) be set to full left and full right. The diagram is largely self-explanatory. However, the following notes may be helpful. 1. The Dynamics unit is probably a limiter/compressor device with possibly a noise gate on the more advanced mixers.

Figure 60 The facilities of a typical mono channel

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9. Tools, inserts, PFL etc. omitted for clarity

100 Analog sound mixing equipment 2. Additional outboard equipment can be connected to the Insert point. A graphic equalizer is one such option. 3. Pre-fade listen (often referred to as PFL) allows an operator to monitor, either audibly or visually, or both, a source before fading. This is particularly useful in live broadcasts where a contribution from an outside source needs to be checked before being faded. 4. Auxiliary outputs can be used for various purposes. While feeds to 'echo', PA (public address) etc. can be dedicated outputs on many mixers, on some installations where flexibility is required auxiliary outputs can provide the same functions. 5. In Figure 60, one auxiliary output is provided with pre-fader and post-fader switching - i.e. the output can be derived before or after the fader. In other words, in Prefader mode, changes to the fader have no effect on the level of the auxiliary output; in post-fader operation, the aux output level varies with the fader setting. In some echo and PA situations it can be beneficial to have the choice between the two modes of operation. 6. Note that the filters, dynamic unit, etc. can be bypassed. This is desirable for several reasons: the unit can be shut down quickly if a fault occurs; when setting up the mixer, the effect of the unit can often best be judged by turning it on and off; after the controls are set on the device, it may not be needed until a later stage of the recording or broadcast, in which case it will be disabled until needed. Figure 61 shows a typical stereo channel for line input (e.g., for stereo tape machines, CDs, or other stereo sources). The balance control provides a limited (approximately 5 dB) amount of total shift left or right.

Additional Terminology AFL. Listen after fade. A feed available for operator monitoring and taken after the fader, as opposed to pre-fader as in PFL. One use is to check for suspected distortion in a source.

Analog sound mixing equipment 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

101

Solo. All other channels are muted. 'Solo-in-place' leaves the image of the transmitter unchanged in the stereo scene. Clean feed (‘Mix minus’). An output from a mixer that is missing one or more of the mixer input signals. The US term 'Mix minus' is more descriptive than 'Clean feed' which is the normal term in the UK. An example of a broadcast is a program broadcast to different countries: music and sound effects, etc. can be sent to broadcasters of other countries without the local commentary/introductions. The program material without the speech would be a clean feed. Groups. For example, by selectively combining two or more channel outputs into one group, the group can have a group fader that is effectively a sub-master fader. On a large mixer, the group modules can provide similar features such as equalization, echo distraction, and PA as the channel modules.

Monitoring 1. Auditory monitoring (eg by loudspeakers). The following are typical controls: Stereo/mono. Mono on both (useful for checking the balance between L and R speakers). Mono on one LS, usually the L speaker. (This can give some measure of check of the compatibility of a stereo signal with what is heard on mono equipment. On some installations there may be a small 'domestic grade' speaker as a further aid in assessing compatibility .) Phase reversal - often in the feed to the L speaker. Dim - a switch provided 12 dB or more attenuation. Balance – adjustment of the relative gain of the feeds to the two speakers. Volume control. 2. Visual. Typical instruments are VU meters (Volume Unit). These are essentially just voltmeters.

Figure 62 A typical monitoring system

Analog sound mixing equipment 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

103

PPM (“Peak Program Meter”). Quasi-peak indicating instruments with a fast pointer rise time and slow fall back. The scale has white on black markings designed for easy reading with minimal eye strain. The PPM is better than the VU meter to indicate incipient overloads. It's much more expensive. See also section 20. Stereo PPMs have concentric pointer spindles. Two stereo PPMs are often mounted side by side, and the hands are color coded as follows: PPM 1: White M signal and equivalent to a mono PPM Yellow S signal PPM 2: Red L (A) signal Green R (B) signal A typical monitoring system is shown in Figure 62.

15 Signal processing

Basic 'tone control' features boost bass

increase the treble

reduce bass

lower treble

(A)

increase bass

increase the treble

reduce bass

lower treble

(B)

Figure 63 (a) bass; (b) triplicate

'Tilt' control, where a single knob varies the relative levels of bass and treble (see figures 65 and 66).

Compressors Definitions: 1. A compressor is a device that reduces the dynamic range of an audio signal by a controllable amount without significant waveform distortion. 2. Threshold. The output level of a compressor is the same as the input level up to the threshold. 3. Compression ratio. This is defined as the ratio:

increase the input level above the threshold increase the output level

Signal processing 105 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 64 Baxendall tone control circuit and response

106 Signal Processing

Figure 65 Circuit for 'tilt' control

4. Restrict. A very high (ideally infinite) compression ratio. Compression ratios greater than approximately 20:1 are considered limiting in practice. On some devices, the threshold is increased by a fixed amount, typically 8 dB, when "restrictive" is selected. Figure 67 illustrates the operation of what is called a hard-knee device. Operators sometimes prefer a less abrupt start of compression. Soft-knee compression is shown in Figure 68.

Compressor Control Methods The control ('side-chain') signal can be used in either a 'reverse' or a 'feed-forwards' mode – or possibly both, as illustrated in Figures 69 and 70.

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111 Figure 66 Tilt Control Frequency Response

108 Signal Processing

Figure 67 Input/output characteristics of a compressor

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Figure 68 Soft-knee compression

Figure 69 (a) Feedback operation; (b) Feed-forwards operation

110 Signal Processing

Figure 70 Compressor/limiter with two side chains

Expanders These increase the dynamic range. Figure 71 shows the basic block diagram and Figure 72 shows a typical input/output characteristic.

Figure 71 Basic diagram of an expander

Figure 72 Input/output characteristic of an expander

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The expansion ratio is defined as

change in input level in dB below the threshold change in output level in dB

Noise gates This term is applied to expanders when the expansion ratio is high – 20:1 or more.

16 Analog Recording and Playback Formulas Magnetomotive force, m.m.f is given by m.m.f. NI where N is the number of turns of the coil and I is the current. The units of m.m.f are amp turns/meter (At/m). The intensity or field strength is the total force acting per unit length I of the magnetic circuit. This is called the magnetizing force H. H mm.m.f./l NI/l(AT/m) The magnetizing force causes a flux in the magnetic circuit. The flux density B is the flux per unit area. The unit is the tesla (webers/m2). m.m.f./S where S is the reluctance of the circuit. Figure 73 shows a typical BH loop (hysteresis loop).

Magnetic tape - typical characteristics Base material - Mylar Base thickness - 25 to 40 m Coating, usually iron oxide (Fe2O3) in the form of acicular particles about 0.6 to 1 m long and less than one tenth that in diameter. Layer thickness – 10 to 15 m.

Band Transfer Characteristic Figure 74 shows the distortion that would result if corrective measures (bias) were not applied.

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113

Figure 73 Hysteresis loop

Tape head data Typical head openings for 38 cm/s (15 in/s) tape speeds: Erase head:

~ 100 meter

Recording height: ~ 20 m Playback height: ~ 5–10 m or less

Recorded Wavelengths The recorded wavelength is the length on tape of one cycle of the recorded signal.

r = true

v meter f

r recorded wavelength v tape speed f recorded frequency

114 Analog recording and playback

Figure 74 Band transfer characteristic

Standard tape widths Tape width (mm)

(in.)

3,8 6,25 12,5 25 50

5/32 1/4 1/2 1 2

Typical use

Cassettes Mono full-track, 1 of 2 track, stereo Stereo mastering, 4 track 4-track mastering, 8 track 16 track, 24 track

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115

The recorded wavelengths are shown in cm in the table below. Tape speed (cm/s) / (in/s) f (Hz)

4,75 17⁄8

9,5 33⁄4

19 71⁄2

38 15

30 60 125 250 500 1k 2k 4k 8k 16 k

0,16 0,08 0,04 0,02 9,5 103 4,8 103 2,4 103 1,2 103 0,6 103 0,3 103

0,32 0,16 0,08 0,04 0,02 9,5 103 4,8 103 2,4 103 1,2 103 0,6 103

0,63 0,32 0,15 0,08 0,04 0,02 9,5 103 4,8 103 2,4 103 1,2 103

1,27 0,63 0,32 0,15 0,08 0,04 0,02 0,01 5 103 2,3 103

Comparison of hysteresis loops for tape stock and head cores

Figure 75 Hysteresis loops for (a) 'hard' magnetic material - tape oxide; (b) 'soft' - head core material

116 Analog recording and playback

Bias Bias Frequency This should be about 4-6 times the highest recorded signal, i.e. about 100 to 150 kHz. Lower bias frequencies can result in unwanted intermodulation products. For example, if the bias frequency was 30 kHz and the highest audio frequency was 18 kHz, then there are sum and difference intermodulation frequencies of 48 kHz and 12 kHz, the latter being within the audible range.

Optimal Bias Setting This is shown graphically in Figure 76.

Figure 76 Illustrating the optimal bias setting

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117

Replay Amplifier Equalization Used to compensate for losses caused by the length of the gap, the thickness of the tape coating and in the pole pieces, etc.

Figure 77 (a) Equivalent circuit of a standard repeat reaction; (b) replay playhead and amplifier response for an ideal head; (c) real playhead and amplifier responses

118 Analog recording and playback

Standard equalizations in IEC and NAB standards

These are given as the relevant time constants, where

1 2RC

Band speed

IEC1

IEC2 a NAB

9,5 cm/s; 3/75 inch/sec

50Hz; 1800Hz 3150s; 90s (NAB only)

19cm/sec; 7,5 inch/sec

0 Hz; 2240 Hz (∞, 70 s)

50 Hz; 3150 Hz (3150 s; 50 s)

38cm/sec; 15 inch/sec

0 Hz; 4500 Hz (∞, 35 s)

50 Hz; 3150 Hz (3150 s; 50 s)

76 cm/s; 30 inch/sec

(IEC1 not used) 0 Hz; 4500Hz (∞, 35s)

AES 1971 0Hz; 9000 Hz (∞ S; 17,5 s)

Head track formats See Figure 78.

Cassettes Track layout. See figure 79.

Mechanical Adjustments of a Tape Head See Figure 80.

Level The tone portion of a tape is usually recorded at 700 Hz or 1 kHz. The exact frequency is usually not critical. The exact level is important. It is taken as the 0 dB reference, but if it is not the required reference flux, the example on page 121 shows how to make a correction.

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Figure 78 Headtrack formats

119

120 Analog recording and playback

Figure 79 Audio cassette tape

Figure 80 Illustration of azimuth and zenith adjustments

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121

Suppose the machine is to be aligned on the NAB characteristic at 320 nWb/m for 0 VU 4 dBm. The reference flux on the band is 200 nWb/m. The reference ﬂuxivity difference is calculated from

Difference (dB) = 20 log10

desired reference fluxivity reference fluxivity on tape

In the example given above:

Difference (dB) = 20 log10

≈ 4 dB m 320 200 °C

17 Analog noise reduction

Sound masking (see also section 3 and figure 8) Sound masking is an essential part of good sound reduction systems. Figure 81 illustrates sine wave masking at 65 dBA SPL.

Figure 81 Sinus masking at 65 dBA SPL (courtesy of Ehmer (1959))

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Elementary compander

Figure 82 (a) Basic compander, (b–d) the dynamic response of the compressor

123

124 Analog noise reduction

Constant Slope Devices Characteristics are shown in Figure 83.

Figure 83 Characteristics of companders for constant slope noise reduction

These devices are independent of the input level. The result is: 1. The dynamic properties must also be level independent. 2. Compression continues until there is no signal. 3. All frequencies in the input signal are processed equally.

Bilinear Devices These have constant gain at low levels, constant but different gain at high levels, with an intermediate range in which gain changes occur. With bilinear devices it is important that the decoder receives the correct level signal.

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125

Figure 84 Bilinear compression and expansion

Practical noise reduction systems Dolby A system This gives approximately 10 dB noise reduction. High-level signals far above the noise are not processed. The audio spectrum is split into four bands: Band 1: low-pass at 80 Hz; Band 2: band pass 80 Hz to 3 kHz; Band 3: high-pass at 3 kHz; Band 4: high pass at 9 kHz.

126 Analog noise reduction

Figure 85 Block diagram of Dolby A noise reduction

Figure 86 Dolby A differential network. This is the same for encoding and decoding

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127

Dolby B system Widely used to reduce the effects of tape noise on cassettes. This gives about 10 dB of noise reduction in the higher frequencies where band noise is most annoying.

Figure 87 Block diagram of Dolby B noise reduction

The system operates on one frequency band, but it has a variable range and can be thought of as a fixed magnitude variable pre-emphasis. See figure 88.

Figure 88 Sliding Band Responses of Dolby B Noise Reduction

128 Analog noise reduction

Dolby C system This uses two sliding bands, similar to Dolby B, but two octaves lower. Approximately 20 dB noise reduction is achieved.

Dolby SR (spectral recording) system With a good analog tape recorder, a dynamic range of more than 90 dB can be achieved.

Dolby S-Type System A development of the SR system, this uses two high frequency stages giving 12 dB noise reduction above 400 Hz and a single low frequency band providing 10 dB NR below 200 Hz. It can provide up to 24 dB noise reduction on cassette tape, with a dynamic range of approximately 85 dB.

dbx systems These are constant slope broadband companders. Type I and II have a compression ratio of 2:1, type 321 has a ratio of 3:1.

Figure 89 Block diagram of dbx noise reduction systems dbx I. The VCA is preceded by a fixed pre-emphasis in the form of a 12 dB shelf rising from 400 to 1600 Hz.

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129

dbx II. Intended for home recording, this system is similar to Type I, but includes a band-pass filter to reduce response to signals outside the 30 Hz to 10 kHz range. dbx 321. This is a professional system designed for use with analogue satellite connections only. It mainly differs from dbx I in that it has a constant compression ratio of 3:1. phone c4. This uses a constant slope of 1.5:1 and four frequency bands as follows: Band 1: 35 to 215 Hz; Band 2: 215 to 1450 Hz; Band 3: 1450 to 4800 Hz; Band 4: 4800 to 16 000 Hz.

Figure 90 Features of the telcom c4 noise reduction system

130 Analog noise reduction

Figure 91 telcom c4 system:- (a) compressor and (b) expander

VHS Hi-Fi A conventional 2:1 constant-slope compander is used with preemphasis in the signal path and frequency weighting in the control path.

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18 Compactdisc

CD parameters Disc diameter Rotational speed Playing time (max.) Number of tracks Track spacing Lead-in diameter Lead-out diameter Total track length Linear speed

120 mm 568–228 tpm (bij 1,4 m/s) 486–196 tpm (bij 1,2 m/s) 74 min 20 625 1,6 m 46 mm 116 mm 5300 m 1,2 of 1,4 m/s

The CD track See Figure 92.

Stages of 'cutting' a CD See Figure 93. 1. The glass plate is polished for maximum smoothness. 2. Photoresist applied. 3. The coating is exposed to a modulated laser beam. 4. Coating is developed. 5. Surface is silver plated to protect 'pits'. 6. Surface is nickel plated to master metal. 7. Metal master is used to make 'mother' plates.

Figure 92 Track and pit dimensions

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133

Figure 93 Stages in the production of CD masters and masters

Error compensation: detection, correction and concealment Detection: find out if there is an error; Correction: It may be possible to completely correct the error; Concealment: If correction is not possible, the effects of the error can be made inaudible, for example by repeating the previous correct sample. These functions are illustrated in Figure 94.

CRC Error Detection (Cyclic Redundancy Check) A bitstream of n bits can be represented as a polynomial with n terms. 11010101 can be written as: M(x) 1x7 1x6 0x5 1x4 0x3 1x2 0x1 1x0 x7 x6 x4 x2 1 Another polynomial G(x) is chosen and in the decoder M(x) is divided by G(x) to get a quotient Q(x) and a remainder R(x).

134 Compactdisc

Figure 94 Error handling

Figure 95 The CRC check principle

A new message U(x) is then generated so that U(x) can always be divided by G(x) to produce a quotient with no remainder. If there is a remainder, an error has occurred. On a 16-bit system, the detection probability is 99.9985 percent.

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135

Hide error. Three types are illustrated in Figure 96. The defective samples can be: (a) muted; (b) the previous Sample(s) may be retained and repeated; or

Figure 96 Methods for Hiding Errors

Figure 97 The timing of the P and Q subcodes

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138 Compactdisc

'Burst' errors For example, caused by holes in the reflective surface of the CD (or by 'drop-outs' in digital tape recording).

CIRC – Cross Interleave Reed-Solomon Code The principle is essentially that samples are encoded to a certain code. Thus, the effects of a burst error after decryption are diffused and can be addressed by standard error detection and correction methods.

The P and Q Subcodes The P subcode is a separator flag for music tracks. It is normally 0 during music and the lead-in track, but is 1 at the beginning of each selection. It can be used for simple search systems. In the lead-out track, it switches between 0 and 1 in a 2 Hz rhythm to indicate the end of the disc. The Q subcode is used for more advanced control purposes; it contains data such as track number and time. The timing of the P and Q subcodes is shown in Figure 97.

EFM encoding (eight-to-fourteen modulation) This converts each 8-bit symbol to a 14-bit symbol and reduces the required bandwidth, reduces the d.c. of the signal. content and adds additional synchronization information (see Figure 98).

CD optical system See figures 99, 100.

Follow Three possible conditions are shown in Figure 101.

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Figure 99 Three-beam optical pickup

Figure 100 Characteristics of a typical injection laser diode

139

140 Compactdisc

Figure 101 Three possible tracking conditions

Figure 102 Signal decoding in a CD player

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19 Digital audio tape

Introduction Since the first commercially available digital audio recording systems appeared in 1979, there has been a proliferation of digital tape formats. Many of the early formats are now obsolete, but their recorded tapes still exist and need to be recognized. Most formats are proprietary developments of manufacturers and are recognized as standards as more manufacturers agree to support the format. Reel-to-reel digital recording has been dominated by the DASH (Digital Audio Stationary Head) and PD (ProDigi) formats, the latter now unsupported. Both accommodated a family of subformats, many of which never materialized as commercial products. The most widely accepted format is R-DAT, or DAT as it is more commonly known. While most DAT equipment manufacturers have adhered closely to the standard, commercial and user pressures have created variants to expand applications and technical specifications. Some changes, such as timecode, are included in the DAT specification, most deviations are not, so compatibility is not always certain. In some recent formats, where proprietary rights have been more strictly enforced, manufacturers supporting the same format have added incompatible functionality while maintaining full compatibility at the level of digital audio tracks, making defining the characteristics of the digital format manufacturer and machine dependent is. It is also necessary to be aware of third party processors used in conjunction with standard digital multitrack recorders to enhance basic digital audio specifications. This normally requires using two standard 16-bit digital audio tracks to store a word length of 20 or 24 bits within the tape format, reducing the capacity of the audio track and removing practical format compatibility. This process is most common within the digital DTRS format.

142 Digital audio tape

Format Notes DAT: Consumer DAT recorders only record at 48 kHz sampling rate, while professional machines record at 44.1 and 48 kHz. Both machines play 44.1 and 48 kHz. The other optional sampling rates are now less common. Deviations from the DAT format by the manufacturer, achieved by increasing the transport rate: – Pioneer 96 kHz sampling rate option – Tascam 24-bit word option – Sonosax 2 or 4 tracks at 44.1/48/88.2/96 kHz sampling rate DASH HR. DASH multitrack machines are compatible on a track-to-track basis, i.e. a 24-track tape replays on a 48-track, while a 24-track replays 24 tracks of a 48-track tape. DASH HR (High Rate) machines are not similarly compatible and, because different techniques are used, they are also not compatible between manufacturers' machines. PD. The ProDigi format is not supported, but many machines are still in use. There are three variants in the 2-track standard that support changed specifications, namely 20-bit or 96 kHz. There is also a 1⁄2-inch, 16-track version of the multitrack format. ADAT types I and II. Type I contains time code recorded in the subcode area, can record and play back 20-bit digital audio, can record at 44.1 kHz, and has an analog track for cue/edit. Type II machines can handle all formats, but backward compatibility is limited by the bit rate used. DTRS. The DTRS HR version has been announced to handle 24-bit digital signals. HR format machines will also play standard DTRS format tapes. Nagra D. Machine will record as programmed within the Nagra D format. Can process up to 24-bit word on tape, but choice of A/D converters limits analog input. Programming option for operation of two or four audio tracks.

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111 Summary of specifications of currently used tape formats Format

Kind of tape

The audio channel

Sampling rate

word length

Notes

THAT

dat tape

32/44.056/44.1/48kHz

16-bits

High spec deviations in size

DASH DASH DASH HR

⁄2″ open haspel ⁄2″ open haspel 1 ⁄2″ open haspel

24 48 24/48

44,1/48 kHz 44,056/44,1/48 kHz 44,1/48 kHz

16-bits 16-bits 24-bits

⁄4″ open haspel

44,1/48(96)kHz

16/20-bits

PD ADAT I

1″ S-VHS with open coil

32 8

44,1/48 kHz 48 kHz

16-bits 16-bits

ADAT II

S-VHS

44,1/48 kHz

16/20-bits

DTRS

Hi8 cassette

44,1/48 kHz

16-bits

Nagra D Sony 1630

U-matic cassette with open reel

2/4 2

32/44,1/48 kHz 44,056/44,1 kHz

18/24-bits 16-bits

Different DASH HR standards No model can handle all digital specifications Sampling rate of 44.1 kHz fails in varispeed range Analog track included for editing DTRS HR 24-bit just announced Spiral scan heads CD mastering processor – still in use

144 Digital audio tape Sony PCM1630. The latest processor developed for use with the U-matic VCR in CD mastering applications. Still in use, although declining.

Main Azimuth System See Figure 103.

DAT cassette distinctive holes See Figure 107 and the table on page 147.

Bandpad See figures 108–110.

Figure 103 DAT head azimuth system

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Figure 104 DAT track format

Figure 105 DAT drum position relative to the track

145

146 Digital audio tape

Figure 106 The DAT cassette

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Figure 107 Distinctive holes

Settings of the distinctive holes in DAT cassettes Hole 1

Gat 2

Gat 3

Metal tape or equivalent/tape thickness 13 m Metal tape or similar thin tape Track spacing 1.5 /tape thickness 13 m Track spacing 1.5 /thin tape

Gat 4 1 0

Pre-recorded music tape Non-prerecorded music tape (standard tape for recording)

Gat 5 1 0

Recording not possible Recording possible

147

148 Digital audio tape

Figure 108 DAT band pad: threaded (–––––); without thread (– – – –)

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Figure 109 The positions of the DAT motors

Figure 110 The DAT control arm assembly

149

150 digital audio tape

Main DAT Specifications (a) Recording/Playback Mode Item Channel No. (CH) Sampling frequency (kHz) Quantization bit number (bit)

Standard Option 1 Option 2 Option 3 2

12 12 (non-linear)

16 (linear)

Transfer Rate (MB/s) Modulation System Correction System Tape Size (mm) Recording Time (min) Tape Type Tape Thickness (m) Tape Speed (mm/s) Track Spacing (m) Track Angle Drum Rotational Speed (r.p.m.) Relative Speed (m/s) Main Azimuth Angle (°)

2.46

2,46 1,23 2,46 8-10 conversie Double Reed-Solomon code 73 54 10,5

120

120 240 Metal powder 13 ± 1 8.15 4.075 13.591 6 ° 22′ 59.5″

8.15

2000 3.133

1000 1.567 ± 20

(b) Prerecorded tape (playback only) Item Channel no. (CH) Sampling frequency (kHz) Quantization bit number (bit) Transfer rate (MB/s) Modulation system

Normal track

Wide track

4 44.1

16 2.46 8–10 conversion

120

8.15

2000 3.133

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151

(b) Prerecorded tape (playback only) continued Item Correction system Cassette size (mm) Recording time (min) Tape type Tape thickness (m) Tape speed (mm/s) Track spacing (m) Track angle Drum rotational speed ( r.p.m.) Relative speed (m/s) Main azimuth angle (° )

Normal track

Wide track

Double reed Solomon code 73 54 10.5 120

80 Oxidetape 13 ± 1

8.15 13.591

12.225 20.41 6° 23′ 29.4″ 2000

3.133

3.129 ± 20

MiniDisc© This system is becoming increasingly popular in the professional field, where its compactness and ease of use make it well suited for sound recordings of speech on location. There are two types of discs: 1. pre-recorded, basically small CDs that play the same way; 2. writable, which are magnetic. When recording, the laser heats a very small area of the disc to a temperature above the Curie point (about 185°C), allowing a magnetic head to record the digital data on the disc. In replay mode, a much lower laser power is used. The polarization of the reflected laser beam is rotated by the magnetic field of the recorded signal and this rotation is detected by a suitable optical process.

Figure 111 Track Format. ATF: automatic track finding: IBG, inter-band gap

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153

256K RAM 64K RAM

64K RAM

CXD2601Q DRUM RF SIGNAL

RF-VERSTERKER CXA 1045Q

ATF CXA 1046Q

DRUM MOTOR

FG, PG

CXD 1008Q

KAPSTAN SERVO 20084

T F3 S FG

D-1/O CXD 1148Q

DIGITAL INPUT/OUTPUT RX DAC

ADC

Count filter

DRUM SERVO CXD 1052Q

KAPSTAN ENGINE

MOTOR MOTOR

CXD 1009Q

4 bit MAN MICROCOMPUTER

4 bit REEL MICROCOMPUTER

ANALOG INPUT

4 bit KEY DISPLAY PROCESSING MICROCOMPUTER

4 bit MECHA MICROCOMPUTER

CXP80524 MECHA-SENSOR

Figure 112 Block diagram of a DAT system

Data: Disc diameter Disc thickness (Cartridge size Track spacing Scan speed Playback and recording time Sampling frequency Encoding Laser wavelength Laser power: Record Replay

64 mm 1,2 mm 72 68 5 mm) 1,6 m 1,2–1,4 m/s Max. 74 minuten (148 minuten in mono) 44,1 kHz ATRAC (Adaptive Transform Coding) 780 nm 2,5–5 mW Ongeveer 0,5 mW

20 audiometingen

Weighting curves The French CCIR 468–4 is universally used for noise measurements in audio systems. It is shown in Figure 113. The 'unweighted' response curve is necessary to avoid the effects of inaudible components that may be present. The A-weighting curve with similar scales is shown in Figure 114 for comparison. The tables provide precise values and tolerances.

The use of special dB is sufficient in audio engineering Note: A suffix applied to the expression dB is only enclosed in brackets where it indicates a reference quantity, e.g. dB (mW) expresses a power measurement relative to a reference quantity of 1 mW. dBu – colloquial abbreviation of dB (0.775 V) dBm – colloquial abbreviation for dB (mW) dBA – weighted sound pressure level in accordance with BS EN 60651: 1994 (SLOW dynamic) dBq – audio system sound level (unweighted) with quasi-peak equipment per CCIR 468 –4 dBqp – audio system sound level (weighted) with quasi-peak equipment per CCIR 468–4

Sound Measurements of Audio Systems Quasi-peak cross measurement equipment that conforms to CCIR-4 (see list of standards) is now commonly used for measuring noise in audio systems. Weighted measurements are performed via the curve shown in Figure 113, while the bandpass response for unweighted measurements is also included to eliminate the effect of inaudible components. For expressing measured values, using the quasi-peak dynamic response means that neither dBu nor dBm is appropriate. The standard therefore specifies the use of dBq

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111 Figure 113 CCIR 468–4 noise weighting curves

Figure 114 A weighting curve (IEC179)

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157

and dBqp for unweighted and weighted measurements, respectively. On professional equipment, noise measurements are made at points where the audio signal peaks to the equivalent of 8 dBu, while the normal line-up tone level of 0 dBu at 1 kHz gives a reading of 0 dBq/0 dBqp on the sound level meter.

Weighting Curves Program Level Measurements A standard program level meter is normally located at the output of professional mixing equipment; this can be one of two patterns: (a) VU meter (volume unit). Although this is really just a moving coil meter equipped with a bridge rectifier, the ballistics of the meter are very precisely specified and only instruments that comply with the IEC/ASA standard (see list of standards) may be used. Proper use of the VU meter requires special training and experience. (b) PPM (Peak Program Meter). The basic instrument consists of a moving coil meter with very precisely specified ballistics combined with electronic processing to give a fast pointer rise time and slow fall back. There are two instruments, standardized by the IEC (see list of standards) and with significantly different characteristics. The 'type 1' (DIN) version has a longer scale and an integration time of 5 ms, while the 'type 2a' (UK) version has a scale length of 24 dB and an integration time of 10 ms. The 'Type 2a' specification additionally requires a 'preferred display meter', which includes electrical, dynamic and scale marking functions; it is generally used on broadcasting equipment across the UK. (The type 2b PPM only differs with respect to the EBU scale.)

158 Audiometingen

Audio measurement standards Program level measurement IEC 60268–10 (2nd edition): 1991/BS 6840: Part 10: 1991 Sound system equipment Part 10: Methods for specifying and measuring the characteristics of peak program level meters (Definitive information on PPMs – Type 1 (DIN) and Type 2a/b (UK/EBU).) IEC 60268–17: 1990/BS 6840: Part 17: 1991 Sound system equipment, Part 17: Methods for specifying and measuring the characteristics of standard volume indicators ( Supersedes ANSI/ASA C16.5:1954 – original VU meter specification.) IEC 60268–18 (1st edition): 1995/BS 6840: Part 18: 1996 sound system equipment, Part 18: Level meters for peak programs: Guide to digital audio level meter ITU/R Report BS 292 (was CCIR 292–2) Use of IEC type 1 PPM ITU/R report BS 820–1: 1994 (was CCIR 820–1) Comparison of readings using VU meter and PPM Audio system noise measurement ITU/R recommendation BS 468–4: 1994, Measurement of audio frequency noise voltage level in audio broadcasts (formerly CCIR REC 468–4: 1990) Note: With the exception of the original edition (CCIR REC 468.1970 which retained the RMS noise measurement) earlier editions had slightly different tolerances but were in essentially the same. AES Presentation, CCIR/ARM, A Practical Noise Measurement Method, Audio Engineering Society, 60th Convention, Los Angeles (1978).

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21 Digital equipment

Artificial reverberation The three main elements are: 1. all-pass filter; 2. comb filter; 3. finite impulse response filter. See figures 115–118.

Digital Mixing and Filtering See Figure 119.

Figure 115 All pass filter

160 Digital equipment

Figure 116 Comb filter

Digital equipment 161 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 117 Finite impulse response (FIR) filter.

162 Digital equipment

Figure 118 Reverberator with FIR, comb, and all-pass filters

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111 Figure 119 Simplified diagram of a digital mixer

164 Digital equipment

Basic components of a digital signal processor

Figure 120 Basic components

Figure 121 shows a simple two-channel digital mixer. The fader settings are converted into coefficients by the analog-to-digital converters. The digital channel samples are multiplied by these coefficients, this process corresponds to a change in the channel sample level. The resulting bitstream can be up to 32 bits wide and care should be taken to avoid low level distortion when converted back to 16 bits.

Digital equipment 165 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Image 121 Two-channel digital mixer

Digital filters These have the advantage over analog filters that steep slopes can be achieved accurately and with time stability (analog filters with slopes greater than about 18 dB/octave require critical component values), and linear phase filters are relatively easy to design.

Picture 122 Simple digital filter

22 MIDI (Musical Instrument Digital Interface) This is a serial interface with start and stop bits at the beginning and end of each byte transmitted. The MIDI standard specifies a baud rate of 31.25 k. (The baud rate is the maximum number of bits per second.) A tolerance of ±1 percent is specified for the clock frequency.

Figure 123 The format of a MIDI message byte. LSB least significant bit; MSB most significant bit

MIDI Hardware Interface In Figure 124, UART stands for universal asynchronous receiver/transmitter. The IN connector receives data from other devices, the OUT connector transfers data generated by the device, and the THRU connector is a relay of the data present on the IN connector. The THRU connector allows multiple MIDI devices to be daisy chained together. Figure 125 illustrates daisy chaining of MIDI devices.

MIDI message data format The status byte contains information about the channel number to which the message applies. It indicates which recipient the message came from

MIDI 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 124 The MIDI hardware interface

Figure 125 A MIDI daisy chain

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Figure 126 MIDI message date format

167

MIDI data messages Message

State

Data 1

Data 2

NOTE OFF NOTE ON POLYPHONIC AFTERTOUCH CONTROL CHANGE

&8n &9n &An &Bn

Note number Note number Note number Controller no.

Speed Speed pressure data

14-bit controllers MSByte (examples)

&Bn

01 02 04 05 06 07

Data Data Data Data Data

&Bn

21 (mod wheel etc)

Facts

&Bn

40 41 42 43

}00–3F(off) }40–7F(on) }40–7F(on) }40–7F(on)

Channel specific messages

(modulation wheel) (breath controller (foot controller) (portamento time) (data entry slider) (master controller)

14-bits controllers LSByte

7-bit controllers/switches (examples)

(sustainpedaal) (portamento) (sostenutopedaal) (softpedaal)

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111 60 (data increment) 61 (data increment)

7F 7F LSByte

62 (unregulated parameter control) 63 (unregulated parameter control) 64 (regulated parameter control) 65 (regulated parameter control) 79 (reset all controllers)

&Bn

7A (local) 7B (all notes off) 7C (omni off) 7D (omni on) 7E (monaural) (Number of channels) 7F (poly)

00 off/7Fon 00 00 00 00 00–ON 00

&Cn &Dn &In

Program No. Print LSByte

– – MSByte

MSByte LSByte MSByte

Channel Modes

PROGRAM CHANGE CHANNEL AFTERTOUCH PITCH WHEEL

MIDI data messages (continued) Message

State

Data 1

Data 2

System messages System exclusive: SYSTEM EXCLUSIVE START END SYSTEM EXCLUSIVE

&F0 &7F

Manufacturer ID –

Facts …

System Common SONG POINTER SONG SELECT TUNE REQUEST

&F2 &F3 &F6

LSByte Song Number –

MSByte –

MIDI-tijdcode QUARTER-FRAME

&F1

Facts

–

System real time TIMING CLOCK START CONTINUE STOP ACTIVE SENSING RESET

&F8 &FA &FB &FC &FE &FF

MIDI 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

171

is intended for and also what type of message should follow - e.g. a NOTE OFF message. The status byte is given in hexadecimal form, e.g. &9n, where n is the channel number in hexadecimal form. The tables on pages 168–170 provide MIDI data messages.

Sequencers A sequencer stores MIDI information coming from one or more MIDI inputs with the option of transmitting it at a later time from one or more MIDI outputs. During the retention time, edits and other data manipulations can take place. See figure 127.

Figure 127 An example of a sequenced MIDI system

172 MIDI Figure 128 shows a more complex system in which time code from a tape machine is used for synchronization.

Figure 128 A complex MIDI system that uses tape machine timecode for synchronization

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23 Studio airconditioning

Air quality Potential pollutants: 1. reduced oxygen and increased carbon dioxide from humans; 2. water vapor, cigarette smoke; 3. formaldehyde from studio equipment.

Ventilation Requirements To maintain carbon dioxide at the maximum recommended level of 0.5% by volume, the minimum ventilation requirement is 0.8 L/s for seated persons. Dilution of body odors to acceptable levels requires a ventilation rate of 8 l/s for seated non-smoking occupants. For heavy smokers, the rate should be at least 32 l/s.

Thermal comfort Temperature A sitting and resting person produces approximately 115 W of heat (75 percent through radiation and convection, 25 percent through evaporation). A person's comfort depends on: 1. the temperature of the ambient air; 2. the average radiation temperature; 3. the average airspeed; 4. humidity; 5. worn clothes; 6. activity undertaken.

174 Studio Air Conditioning The most commonly used thermal index in the UK is denoted by the resulting temperature, tres, and is given by

three =

tr (oor 10v) 1 10v

where tai is the indoor air temperature (°C), tr is the average radiant temperature (°C) and v is the average air velocity (m/s). The recommended resulting temperature range for sedentary occupations, such as studios, is 19–23°C, when the relative humidity is between 40 percent and 70 percent and the air velocity is less than 0.1 m/s. When the airspeed is less than 0.1 m/s the above expression is simplified to tres 1⁄2tr 1⁄2tai

Humidity Relative humidity, RH is defined as

RV =

actual vapor pressure 100% saturation vapor pressure

See table opposite. For most studios, the RH should be between 40 and 70 percent. If the RH is less than about 40 percent, static electric charges can build up.

Studio airconditioning 175 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

RH based on wet and dry bulb thermometer readings Wet bulb compression (°C) 14

Dry bulb temperature (°C) 16

0,5 1,0

95 90

95 91

96 91

96 92

96 93

1,5 2,0

85 79

85 81

86 82

87 83

88 84

88 85

89 85

3,0 3,5

70 65

71 67

73 69

74 70

76 72

77 73

78 74

78 78

4,0 4,5

60 56

63 58

65 61

66 63

68 64

69 66

71 67

72 69

5,0 5,5

51 47

54 50

57 53

59 55

61 57

62 59

64 61

65 62

6,0 6,5

42 38

46 42

49 45

51 48

54 50

56 53

58 54

59 56

7,0 7,5

34 30

38 34

41 38

44 41

47 44

49 46

51 49

53 51

8,0 8,5

26 22

30 26

34 30

37 34

40 37

43 40

46 43

48 45

9,0 9,5

18 14

23 19

27 23

31 28

34 31

37 34

40 37

42 40

10.0

Figure 129 Scheme of a typical air conditioning system

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24 Audio Signal Distribution Electromagnetic Telephones Basic Features and Functions: 1. Hybrid. The telephone is a four-wire device, but must be converted to two-wire for connection to the exchange. 2. Sidetone. Part of the transmitted signal is fed into the receiver. 3. Power supply. The transmitter must be powered with d.c. but the receiver must be isolated from it. 4. Impedance matching. The AC. The phone's impedance must be matched to the line's impedance. The phone must be able to handle a wide variation in line impedances. 5. Protection – against strange voltages, voltage peaks, lightning strikes.

Figure 130 Telephone voice circuit

178 Distribution of audio signals The speech circuit. Figure 130 shows the use of a hybrid transformer for use with electromagnetic telephones. The current from the receiver is split into two components, I1 and I2. These produce currents I3 and I4 respectively. These almost cancel each other out - the residual current provides the sidetone.

Basic signaling This is shown schematically in Figure 131.

Figure 131 Basic signaling for electromagnetic telephones

Electronic telephones The essential requirements are similar to those for electromagnetic telephones: 1. provide two-wire to four-wire conversion; 2. draw power to operate the transducers and integrated circuits; 3. set loudness gains for sending and receiving; 4. provide sidetone; 5. match impedances. Two-wire to four-wire conversion is illustrated in the bridge shown in Figure 132.

Distribution of audio signals 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

179

Figure 132 Bridge circuit for conversion from two to four wires

For a perfect balance of the bridge in figure 132

Zlijn R1 = Zbal R2

Transducers in electronic telephones 1. Microphone. Generally of the electret type. The electret material carries a permanent charge equal to a d.c. potential of 100 V. 2. Receivers. Either a moving coil or a rocking arm. See figure 134.

180 Distribution of audio signals

Figure 133 Electret microphone

Figure 134 Moving coil receiver (top), rocking armature (bottom)

Distribution of audio signals 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

181

Dual-tone multifrequency signaling (DTMF) Sixteen audio frequencies are arranged in a four-by-four matrix. Numbers are represented by two of eight frequency combinations. For example, the number 6 is formed by 770 Hz plus 1447 Hz.

Low frequencies (Hz) 697 770 852 941

High Frequencies (Hz) 1209

1336

1447

1633

1 4 7 *

2 5 8 0

3 6 9 #

A B C D

The tones usually last 50 ms with 40 ms intervals between tones.

ISDN (Integrated Services Digital Network) The aim is to provide voice and non-voice services in the same network. Two basic delivery methods are defined for ISDN. 1. A base rate service running at 144 Kbits/s, with 2 × 64 Kbits/s channels and 16 Kbits/s signaling channel. 2. A primary speed service running at 2.048 Mbits/s and providing 30 × 64 Kbits/s channels for signaling and synchronization.

25 Radio propagation

The transmission frequency range Medium wave (MF): 526.5–1606.5 kHz Very high frequency (VHF band 2): nominally 88–108 MHz Short wave (SW) or high frequency (HF): 3900–4000 kHz (75 m- band) 5950–6200 kHz (49 m band) 7100–7300 kHz (41 m band) 9500–9775 kHz (31 m band) 11 700–11 975 kHz (25 m band) 15 100–15 450 kHz (19 m band) 17 700–17 900 kHz (16 m band) 21 450–21 750 kHz (13 m band) 25 600–26 100 kHz (11 m band)

Effective Isotropic Radiated Power (EIRP) This is the product of the power applied to the actual antenna multiplied by the antenna gain relative to an isotropic antenna radiating uniformly in all directions. Figure 135 shows how power density varies with distance for a 1 kW EIRP.

Reference Field Strength This is 1 V/m. Other fields are usually referred to as dB with respect to this reference. Figure 136 shows the relationship between dB V/m and the field in V/m.

The ionosphere This extends from about 50 km above the earth. It is the result of UV and X-rays, as well as high-energy charged particles from the sun. For convenience, the ionosphere is divided into D, E, and F layers, with the latter sometimes split into F1 and F2 layers. Figure 137 shows in a simplified way the heights of the layers and also the nature of the height variations during the day.

Radio propagation 183 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Figure 135 Power density variation with distance for a 1 kW EIRP. The reference (0 dB) at 1 km is approximately 7.96 105 W/m

Figure 136 Relationship between field strength and decibel field

184 Radio propagation Median field strength for f.m. broadcasting in band-2 VHF Field strength (dB) Area Rural Urban Large cities

Monophonic

Stereophonic

48 60 70

54 66 74

Figure 137 Ionospheric Layer Heights with Diurnal Variations (Simplified)

Critical frequency (Fcrit) is the maximum frequency that is vertically reflected back to earth by a given layer. Typical values are: FcritE 4 MHz FcritF1 6 MHz FcritF2 15 MHz

Properties of the layers D layer. About 50 to 90 km above the earth. During the day, MF waves are almost completely absorbed by this layer. At sunset, the absorption decreases rapidly, so that air wave reflection from the E layer occurs. E layer. About 110 km above the Earth. MF signals are reflected from it at night. Lower frequency HF waves can be as well

Radio propagation 185 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

reflected in the day. Maximum ionization occurs around noon in the summer. Sporadic E (Es) occurs when clouds of intense ionization form. Abnormal reproduction over great distances can then take place. F low. More than 150 km. It splits into F1 and F2 during the day. F1 is located about 200 km above the Earth and has greater ionization than the E layer, resulting in higher critical frequencies. FcritF1 peaks in summer. F2 is about 300 km above the Earth. Ionization is greater than in any of the other layers. FcritF2 is greater in winter than in summer.

26 Digital interface and synchronization The AES/EBU interface (AES Audio Engineering Society; EBU European Broadcasting Union). This format allows two digital audio channels to be serially transferred over one balanced interface using RS422 drivers and receivers. Figure 138 shows the hardware. This allows the two channels to be transmitted over distances of up to 100 m.

Figure 138 AES/EBU hardware interface

The format of the 'subframe' is shown in Figure 139. The 'Sync' bits can be one of three patterns that identify which of the two channels the sample represents or mark the beginning of a new channel status block. 'Aux' contains additional data. See also Figure 140.

The Sony-Philips Digital Interface (SPDIF) This is very similar to the AES/EBU interface with only subtle differences.

Sony digital interface (SDIF) The common version is the SDIF-2 which is mainly used for audio data transfer of SONY professional digital audio

Digital interface and synchronization 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

187

Figure 139 AES/EBU subframe format

Figure 140 AES/EBU sync patterns. A is the start of the A channel subframe, B is the start of the B channel subframe, and C is the start of a new channel status block

Figure 141 SDIF-2 data format

188 Digital interface and synchronization equipment, especially the PCM-1610 and 1630, as well as some semi-professional equipment (see Figure 141).

Mitsubishi interfaces Mitsubishi ProDigi format machines use an interface similar to SDIF, but not compatible with it. Separate connections are used for each audio channel. 'Dub A' and 'Dub B' are 16-channel interfaces found on multi-track machines; Dub A with tracks 1–16 and Dub B with tracks 17–32. The pin assignments are shown below. Dub A: Pen 1.18 2.19 3.20 4.21 5.22 6.23 7.24 8.25 9.26 10.27 11.28 12.29 13.30 14.31 15.32 16 .33 34.35 36.37 38.39 40.41 17.50

Functie Ch.1(/) Ch.2(/) Ch.3(/) Ch.4(/) Ch.5(/) Ch.6(/) Ch.7(/) Ch.8(/) Ch .9(/) Ch.10(/) Ch.11(/) Ch.12(/) Ch.13(/) Ch.14(/) Ch.15(/) Ch.16(/) Bitklok( /) WCLK(/) Rec A(/) Rec B(/) GND

Dub B: Pin 1,18 2,19 3,20

Function Ch.17(/) Ch.18(/) Ch.19(/)

Digital interface and synchronization 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

4,21 5,22 6,23 7,24 8,25 9,26 10,27 11,28 12,29 13,30 14,31 15,32 16,33 17,50 Dub C-pen 1,14 2, 15 5,18 6,19 7,20 12,25

189

Ch.20(/) Ch.21(/) Ch.22(/) Ch.23(/) Ch.24(/) Ch.25(/) Ch.26(/) Ch.27(/) Ch. 28(/) Ch.29(/) Ch.30(/) Ch.31(/) Ch.32(/) GND

Function Left(/) Right(/) Bit Clock(/) WCLK(/) Master Clock(/) GND

Yamaha Cascade Interface This is commonly used with Yamaha digital audio equipment to cascade a number of devices. The interface terminates in an 8-pin DIN-type connector and carries two channels of 24-bit audio data over an RS422 standard differential line.

Figure 142 Yamaha interface connector pins

190 Digital Interface and Synchronization Pin connections are: Pin Function 1 WCLK 2 GND 3 Audio data 4 WCLK 5 Audio data 6 20 H coil to GND 7 20 H coil to GND 8 GND (in), ENABLE (out) The 20 H coils on Pins 6 and 7 are for suppression of r.f. interference.

Timing for Synchronous Signals More than one reference signal can be used to lock the sampling frequency clock of a digital audio device. In a large system it is vital that all devices remain locked to a common sample rate reference clock and there are AES recommendations for this. Frame edges of the input signal should be within ±25% of the frame edge of the reference signal and outputs within ±5%, although tighter accuracy than this is preferred. (See Figure 143.)

Figure 143 Timing determinations for synchronized signals

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

27 Ultrasonic

Piezoelectric transducers Natural materials: quartz, SiO2. Synthetic materials include certain plastics and ceramics containing lead zirconate and titanate (PZT). The latter must be electrically polarized by placing them in a strong electric field. For this reason they are known as 'ferroelectricity' by analogy with the magnetization of a ferromagnet. Four constants characterize the performance and efficiency of piezoelectric transducers: 1. Bandwidth, Q

ffun (via fb)

where ffun is the fundamental resonant frequency, and fa and fb are the frequencies at which the transducer output amplitude drops to 1/⎯ √2 of the amplitude at ffun.

Figure 144 Graph of amplitude versus frequency resulting in the Q factor

192 Ultrasonics 2. Electromechanical coupling coefficient, k k √(fraction of mechanical energy that will produce an electric field in the piezo) or k √(fraction of electrical energy that will produce mechanical energy) 3. g =

electric field in piezo slice voltage applied to piezo planes

(The electric field is in V/m, the voltage is in Pa.) 4. d =

charge on piezo face force applied to wafer face

(Charge is in C, force is in N.) or

change in the length of wafer face tension applied to faces

(The units are m and V, respectively.) Overall: Good Receiver: Good Transmitter:

lage d, hoge d,

hoge g lage g

Electromagnetic Acoustic Transducers (EMAT) Detection using an EMAT is illustrated in Figure 145 and Figure 146.

Magnetostrictive Transducers A change in the dimensions of the material is the result of the application of a magnetic field. The amount of change depends on the material and internal structure. Some materials such as Turfinol (Fe, Tb and Py) can expand up to 1 percent. Ferromagnetic materials can be ultrasonically tested for defects using their own magnetostrictive properties, as shown in Figure 147.

Ultrasonic 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

193

Figure 145 Motion detection in the plane. Bs is the static magnetic field due to the magnet, Be is the varying magnetic field due to the eddy current

Figure 146 Out-of-plane motion detection with an EMAT

194 Ultrasonic

Figure 147 Reflection and transmission of ultrasonic waves in a magnetostrictive sample

Reasons for using ultrasonics in industry 1. Non-destructive – does not affect the sample and does not even require contact between the sample and the ultrasonic generator and detector. 2. Eco-friendly and clean. 3. Easy to use. Ultrasonic equipment operators require little training and many processes can be automated. 4. Accurate. Surface displacements as small as 0.02 nm can be detected using laser interferometry. Thickness measurements up to 10 m are possible.

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

28 Radio studio facilities

General Purpose Studios This category includes most on-air and preparatory studies, news, interviews, presentation and announcement studios. Layout. Figure 148 shows a typical small studio complex. The sound lobbies are essential in providing acoustic separation between studio and control room.

Acoustics A reverberation time of 0.4 to 0.5 is generally acceptable for this type of studio.

The mixing console Number of channels: often about eight for a small studio with only speech. 16–24 channels is more typical. Key desk features include: Prefade listens (PFL) on any channel or group – especially important when live contributions come in. equalization. Auxiliary outputs – at least two, for artificial bounce, foldback or clean feeds to contributors, etc. External sources (OS). A mixing console in a general purpose studio must be able to handle a variety of external sources, including telephone calls over the public network.

Telephone Balance Units (TBU) Such units must be able to: 1. put a call on hold when it is transferred from a normal telephone handset to the mixer;

Figure 148 Typical small studio complex

Radio studio facilities 197 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

2. terminate the incoming line with the correct impedance; 3. electrically isolate the telephone network from the studio equipment; 4. separate transmitting and receiving signals from the two-wire telephone circuit for connection to the desk; 5. provide automatic gain control of the transmitting and receiving signals; 6. provide a voice-over circuit to reduce the level of the telephone conversation when the studio presenter is speaking, reducing unwanted coloration. Due to connections to the public telephone system, some broadcasters insert an audio delay of several seconds ('Profanity Delay') to give a presenter time to respond to unacceptable incoming language.

Communications The following are typical requirements in a general purpose studio: 1. a fast and foolproof talkback system; 2. a special talk-back to tape; 3. an intercom. link with, for example, a newsroom; 4. cue lights - often green. A pre-arranged sequence of signals is used.

Transmission protections Important are: 1. the ability to disable line-up oscillators that can be used for setting signal levels; 2. the routing of talkback only to studio headphones when microphones are live and not to studio speakers or desk output.

Figure 149 Semi-automatic radio station complex

Radio studio facilities 199 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Recording and playback formats: 1. Vinyl disc players - probably still needed. 2. Open-reel (“quarter-inch”) analog tape machines. 3. CD players. 4. Cartridge players (eg for jingles). 5. Cassette recorders. 6. DAT machines. 7. MiniDisc players. 8. Digital mass storage, such as networked hard sick systems. The latter enable semi-automatic systems (see figure 149).

Rock Music Studios These are similar in almost every way to commercial recording studios.

Acoustics Often very short reverberation times, although more flexible working arrangements are often made. These can be 'live' and 'dead' areas. Typical control room facilities. 1. Separate equipment room for multi-track tape machines, desktop power supplies, digital processing racks, etc. 2. Distribution of computer and MIDI data signals in the room. 3. The mixing console is likely to be complex and likely to provide computer-aided mixing and assignable facilities. There will be sufficient aid stations to allow the use of a wide range of outboard equipment.

Listening levels in rock studios SPLs can exceed 100 dBA. Current UK legislation requires precautions to be taken when levels exceed 85 dBA.

200 Radio studio facilities

Classical music studios Acoustics Reverberation times in the region of 1.2 to 2.0 s, as flat as possible between 50 Hz and 10 kHz. Background noise levels should be very low - say 25 dBA.

Live Broadcasting Serious music studios can often be used for live broadcasting and this should be taken into account when specifying facilities.

Radio drama studios Acoustics To allow satisfactory manipulation of acoustics and perspective, a number of conditions are required: 1. dead space. Typical reverberation time is 0.1 s; 2. 'live' area with a reverberation time in the range of 0.5 to 0.8 s; 3. securities room with, among other things, a specially built kitchen, telephone box; 4. narrator's booth. A typical drama studio layout is shown in Figure 150.

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9

29 connectors

Balanced circuits Connector type

Signal Signal Earth (or "hot") (or "cold") (screen)

3-way XLR style jack (e.g. PO style or so-called 'stereo' jack, 6.3mm or miniature, wired for a single balanced circuit) 3-pin DIN (as fitted to some microphones)

Pin 2*

Pin 3*

Pin 1

Tip

Ring

Sleeve

Pin 1

Pin 3

Pin 2

Unbalanced circuits Connector type

Signal (or "hot")

Ground (screen) and signal (or "cold")

Pin 2*

Pins 3* and 1

phono

Sleeve

Jack with 2 terminals (6.3 mm or miniature)

Tip

Sleeve

3-way jack (6.3 mm or miniature)

Tip

Ring and sleeve

3-way jack (6.3mm or miniature) wired for stereo

L: Tip R: Ring

Sleeve

XLR style

* XLR pins 2 and 3 have been swapped in some organizations and some microphones, mainly US. Connecting to equipment using the other standard can cause a phase reversal in balanced circuits and possible loss of signal in unbalanced circuits.

Connectors 203 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

DIN connectors Please note that there may be variations depending on the equipment manufacturer.

DIN 180° (5-pin) microphones

L: Pen 1 R: Pen 4

Pin 2 - Pins 3 and 5 can carry the polarization voltage

Tape recorder inputs

L: Pen 1 R: Pen 4

Pin 2 –

L: Pen 1 R: Pen 4

Pin 2 –

L: Pen 3 R: Pen 5

Pin 2 –

Tape Recorder Outputs: (Low Impedance) (High Impedance)

204 connectors

European voltage standardization For many years the UK mains voltage has been based on a range of 415/240 volts. From January 1, 1995, the permitted range has been changed to a 400/230 volt power supply. In fact, the new range of allowed variations is larger and almost completely covers the older range. User

Supply voltage and allowed variations before January 1, 1995

Supply voltage and permitted variations after January 1, 1995

Most household (single-phase power supplies)

240 volts 225.6 254.4 v

230 volts 216.2 253 v

Most commercial or industrial (three-phase power supplies)

415/240 volt 390.1 439.9 v

400/230 volts 376.0 440 v

Other (split-phase power supplies)

480/240 volts 451.2 508.8 v

460/230 volts 432.4 506 v

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30 Broadcast data

100 volt line loudspeaker systems (data reproduced courtesy of Klark Teknik.) Load impedance vs. power dissipation ohms

watt

10 20 25 33,3 40 50 83 100 111 125 200 250 333 500

1000 500 400 300 250 200 120 100 90 80 50 40 30 20

Load impedance versus power dissipation ohms

watt

1.0k 1.11k 1.25k 1.43 1.67 2.00k 2.50k 3.33k 5.00k 10.0k 20.0k 40.0k 80.0k

10 9 8 7 6 5 4 3 2 1 0,5 0,25 0,125

206 Public Address data

Absorption of sound in air for different relative humidity (dB/30 m) Note that the peak absorption for all frequencies is between 10 and 20 percent RH. The peak at 2 kHz occurs when the RH is about 10 percent; at 10 kHz when RH is about 20 percent. f (kHz) 2 4 6 8 10

RV(%) 20

1,0 2,7 4,8 6,5 8,6

0,5 1,6 2,9 4,3 5,9

0,2 1,0 1,8 3,0 4,0

0,2 0,9 1,5 2,4 3,3

0,1 0,8 1,5 2,3 3,0

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

31 Useful literature

The following list is by no means exhaustive, but nevertheless contains books, papers, and articles that may be helpful if more information is needed. The items in the list are categorized in an effort to help the reader. However, because there is an unavoidable overlap in categories, the same works sometimes appear more than once.

Hearing GREEN, DM, An Introduction to Hearing. Earlbaum, Hillsdale, NY (1976) MOORE, B.C.J., Introduction to the psychology of hearing. Academic Press, London (1986) PICKLES, JO, An Introduction to the Physiology of Hearing. Academic Press, London (1986) ROBINSON, D.W. and DADSON, R.S. A redefinition of the equal loudness relations of pure tones, British Journal of Applied Acoustics, 7, 166-181 (1956)

Transducer BAXENDALL, PJ, Loudspeaker and Headphone Handbook, 2nd edn. Focal Press, Oxford (1994) BORWICK, J., Microphones; Technology and Engineering. Focal Press, Oxford (1990) GAYFORD, ML, Electroacoustics, microphones, earphones and loudspeakers. Newnes-Butterworths, Oxford (1970) JORDAN, EJ, Loudspeakers. Focal Press, London (1963) NESBITT, A., The Use of Microphones, 4th edn. Focal Press, London (1995) OLSON, HF, Acoustical Engineering. Professional and Audio Journals, Philadelphia, PA (1991) ROBERTSON, AE Microphones. Iliffe, London (1963)

Room Acoustics BBC, Guide to Acoustic Practice. British Broadcasting Corporation (1990) BERANEK, LL Music, acoustics and architecture. Wiley, New York (1962) BISHOP, RED and JOHNSON, D.C., Mechanics of Vibration. Cambridge University Press, Cambridge (1979)

208 Useful Literature GILFORD, C.L.S., Acoustics for Radio and Television Studios. Peter Peregrinus, London (1972) HARRIS, CM (ed), Handbook of Noise Control, 2nd edn. McGraw-Hill, New York (1979) KNUDSEN, VO, architectural acoustics. Wiley, New York (1932) SABINE, PE, Acoustics and Architecture. McGraw-Hill, New York (1932) TEMPLETON, DUNCAN (ed.) Acoustics in the Built Environment. Butterworth Architecture, Oxford (1993)

Stereo BLAUERT, J., Spatial hearing: the psychophysics of human sound localization. JS Allen, MIT Press, Cambridge, MA (1983) BLUMLEIN, A.D., British Patent Specification 394325, Journal of the Audio Engineering Society, 6, 91–100 (1958) DAUBNEY, C., Ambisonics - An Operational Insight, Studio Sound ( Aug. 1982) MOORE, B.C.J., Introduction to the psychology of hearing. Academic Press, London (1986) SNOW, W., Basic Principles of Stereophonic Sound, Journal of the Society of Motion Picture and Television Engineers, 61, 567-589 (1953)

Amplifiers and filters AMOS, S.W., Radio, TV and Audio Technical Reference Book. NewnesButterworth, London (1977) BAXANDALL, PJ, Tone control with negative feedback, Wireless World, 58, 402-405 (1952)

Limiters and Compressors BEVILLE, M., Compressors and Limiters: Their Use and Abuse, Studio Sound, 19, 28–32 (1977) DUNCAN, B., VCAs Examined, Studio Sound and Broadcast Engineering, 31(7) 68–62 ( 1989) GLEAVE, MM and MANSON, W.I., The development of soundprogramme limiters in the BBC, BBC Engineering 107, 9–10 (1977)

Analoge opname ARNOLD, AP, Principles of Magnetic Tape Recording (BBC Engineering Training Sheet 45T) BBC, Evesham (1979) HAMMOND, P., Electromagnetism for Engineers. Pergamon Press, Oxford (1978)

Useful literature 209 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Noise reduction BERKOVITZ, R., and GUNDRY, K.J., Dolby B-type noise reduction, Audio Magazine (Sep/Oct 1973) DOLBY, R.M., A 20 dB audio noise reduction system for consumer applications, Journal of the Audio Engineering Society, 31 , 98–113 (1983) DOLBY, R.M., The spectral recording process, Journal of the Audio Engineering Society, 35, 99–118 (1987) EHMER, R.H., Masking of tone v. noise bands, Journal of the Acoustical Society of America , 31, 1253 -1256 (1959)

Compact disc CLIFFORD, M., The Complete Compact Disc Player, Prentice-Hall, Englewood Cliffs, NJ (1987) POHLMANN, KS, Principles of Digital Audio. Macmillan, New York (1985) SONY SERVICE CENTRE, Digital Audio and Compact Disc Technology, 3rd edn. Focuspers, Oxford (1995)

Recording and editing of digital audio tapes BORWICK, J. (ed.), Sound Recording Practice. Oxford University Press, Oxford (1995) HOAGLAND, AS and MONSON, J.E., Digital Magnetic Recording. Wiley, New York IMMINK, KAS, Coding Techniques for Digital Recorders. Prentice Hall, Englewood Cliffs, NJ RUMSEY, FJ, Digital Audio Operations. Focal Press, Oxford (1991) WATKINSON, J., The Art of Digital Audio, 2nd edn. Focus Press, Oxford (1994)

MiniDisc MAES, J., De MiniDisc. Focusers, Oxford (1996)

Digitale apparatuur, enz. POHLMANN, K.S., Principles of Digital Audio. Macmillan, New York (1985) RUMSEY, FJ MIDI Systems and Control, 2e edn. Focal Press, Oxford (1994) SYPHA, The Nonlinear Video Buyer's Guide, 5e edn. Sypha, 216A Gipsy Road, Londen, SE7 9RB (1999) SYPHA, The Tapeless Audio Directory, 7e edn. Sypha, 216A Gipsy Road, Londen SE7 9RB (1998) WATKINSON, J., The Art of Digital Audio, 2e edn. Focuspers, Oxford (1994)

Mobile DEPARTMENT OF TRANSPORT, Road Vehicle Regulations (Construction and Use), current edition. HMSO, London.

210 Useful literature

Airco LUFF, M.G., Airco for Students. Technitrade Journals, London (1980) MILLER, L.M., Students' Textbook of Heating, Ventilation and Air Conditioning. Technitrade Journals, London (1976)

Telephony HILLS, M.T. and EVANS, B.G., Transmission Systems. Allen and Unwin, London (1973) RICHARDS, DL, Telecommunications by speech. Butterworth, Oxford, (1973) SMITH, SF, Telephony and Telegraphy. Oxford University Press, Oxford (1969)

Teletext and RDS Cervix, P.L. and WHITE, N.W., Broadcast Data Systems - Teletext and RDS. Focus press, Oxford (1993)

Digitale interface RUMSEY, F., Digital Audio Operations. Focal Press, Oxford (1991) RUMSEY, F. en WATKINSON, J., The Digital Interface Handbook, 2e edn. Focal Press, Oxford (1995) WATKINSON, J., The Art of Digital Audio, 2e edn. Focuspers, Oxford (1994)

Public Address Systems CAPEL, V., Public Address Systems. Focus Press, Oxford (1992)

Ultrasonics DUBOVY, J., Introduction to Biomedical Electronics. McGraw-Hill, New York (1978) FREDERICK, JR, Ultrasonic Engineering. Wiley, New York (1965) SHUTILOV, VA, Fundamental Physics of Ultrasound. Gordon & Breach, New York (1988) SZILARD, J., (ed.) Ultrasonic Testing, Nonconventional Testing Techniques. Wiley, Chichester (1982)

1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Table of contents

A signals, 87, 88 A weighting network, 23 A-B supply, 71 Absorption coefficients, noise, 42–3, 206 Acoustic elements, electrical equivalents, 34 Acoustic resonances, 4 Acoustics, radio studio, 195, 199, 200 ADAT machines, 142 AES/EBU interface, digital audio transfer, 186, 187 After-fade listen (AFL), 100 Air conditioning, studio, 173–6 Air quality/pollution, 173 Aliasing, 39 All-pass filter, 159 American Standards, 27 Amplitude modulation, 7 Analog noise reduction, 122–30 Analog recording and playback, 112–21 Analog sound mixing, 94, 94–103 Artificial reverb, 159, 162 Audio cassette tape, track layout, 120 Auditory monitoring, sound mixing, 101 Automatic track finding (ATF) , 151 Azimuth Setting, Tape Head, 120 B Signals, 87, 88 Bafﬂes, Speaker, 81 Balanced/Unbalanced Circuits, 94 Connectors, 202

Bases, Mathematical, 3 Fundamentals, 1–11 Bass tip-up see Proximity effect Baxendall tone control, 105 Beats Sensation, 22 Bias Frequency, 116 Bias Setting, 116 Bilinear Companders, 124, 125 Bit Rate, 37 British Standards, acoustic noise , 27 Transmission frequencies, 182 Cardioid microphone, 65–6, 91 Cassette tape, audio, 120 CCIR, 468-4 standard, audio noise, 154, 155 Channel facilities, sound mixing, 97, 100 Circuits, balanced/unbalanced, 94 Classical music studios, 200 Clean feed, 101 Close box bafﬂe, 81 Coincident pair microphones, 88 Comb filter, 160 Compact disc (CD), 131 cutting stages, 131, 133 error compensation, 133 optical system, 139 signal decoding, 140 track and pit dimensions, 132 tracking conditions, 140 Compander: bilinear, 124, 125 constant slope, 124, 128–9 elementary, 123

212 Index Compression Ratio, 104 Limiting, 106 Compressor, 104 Control Methods, 106, 109, 110 Connectors: Balanced Circuit, 202 DIN, 202, 203 Unbalanced Circuit, 203 Constant Slope Companders, 124, 128–9 Control Room Facilities, 199 Crossover Networks , 77, 79–81 Cyclic Redundancy Check (CRC), 133–4 DASH HR Machines, 142 DASH Machines, 142 dB Sufficient, Sound Levels, 154 dbx Compander Systems, 128–9 Decibels (unit), 4 Definitions, 1–3 Degrees of Freedom , system, 35 Delay, audio, 197 Difference tones, 22 Digital audio tape (DAT), 141 automatic track finding (ATF), 151 block diagram, 153 cassette, 146 cassette distinctive holes, 147 drum position, 145 formats, 142, 143, 145, 151 head azimuth system, 144, 145 specifications, 150 bandpad, 148, 149 Digital principles, 37–40 Digital signal processor: basics, 39, 163, 164, 165 mixer, 165 DIN connectors, 203 Directional patterns, microphone, 59-69 Dither, 38

Dolby A system, 125, 126 Dolby B system, 127 Dolby C system, 128 Dolby S type system, 128 Dolby SR system, 128 Doppler effect, 14 DTRS machines, 142 Dual-tone multifrequency signaling (DTMF ), 181 Dynamic range, 94 Ear: frequency discrimination, 19 frequency responses, 19 structure, 19, 20 Effective isotropic radiated power (EIRP), 182 EFM encoding, 137, 138 Eight-to-fourteen modulation see EFM encoding Electret microphone, 180 Electrets, 58 Electrical formulas, 5–6 Electromagnetic acoustic transducers (EMAT), 192, 193 Electromechanical relations, 31–3 Electrostatic microphones, 58–9 Enclosures, loudspeaker, 82 Equal loudness curve, 19, 21 Equivalent sound level, 24 Error concealment, 135 Error detection, 133 –4 digital signal processing, 39 European voltage standardization, 204 Expander, 110 Expansion ratio, 111 Exposure times, 25 Eyring and Norris formula, 41 Ferroelectric, 191 Figure-eight microphones, 62–5, 90

Index 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Filters, 159, 160, 161 digital, 165 Finite impulse response (FIR) filter, 161 Flux density, 112 Fluxivity adjustment, magnetic tape, 118, 121 Frequency, 3, 12, 13 typical sound ranges, 16 Frequency difference limes, 19 Frequency modulation, 7 Group modules, sound mixing, 101 Pistol microphone, 67, 68, 69 Haas effect, 85 Headstock formats, 119 Health and safety regulations, 25 Action levels, 25 Hearing damage, 24 Hearing process, 19–22 Helmholtz resonator, 4–5, 82 Horn load, 82 Humidity requirements, studio , 174 Hypercardioid microphone, 66–7, 90 Hysteresis loop, 112, 113, 115 IEC standards, acoustic noise, 27 Impedance relationships, 31 Input impedances, 95 Integrated digital network for services (ISDN), 181 Cross-channel difference, sound image position, 86 Interconnections, 96 see also Connectors Interference tube microphone, 67, 68, 69 International standards, acoustic noise, 27 Inverse square law, 14 Ionosphere, 182, 184–5 Italian musical terms, 17–18

213

Jack Connector Plugs, 95, 96 JFMG Ltd, 73 Address, 75 Licenses, Transmitters, 75 Lip Ribbon Microphone, 64 Loudness, 20 Design Criteria, 45–6 Loudspeakers, 76–83 100 Volt Line Systems, 205 Impedance and Frequency, 82 Sensitivity, 83 types, 72–7 M signals, 87, 91 M/S microphones, 88, 91 M/S signals, A/B derivation, 88 magnetic tape, 112 recorded wavelengths, 113, 115 standard widths, 114 transfer characteristics, 112 Magnetomotive force, 112 Magnetostrictive transducers, 192, 194 Diaphragm resonance, 4 Microphone transducers, 55–9 Microphones: directional response, 54 electret, 180 electrical characteristics, 55 electrostatic, 58–9 M/S, 88, 91 physical aspects, 54 radio see Radio microphone reliability , 54 sensitivities, 55, 56 sound quality, 53 for stereo, 88–92 types, 55–9 variable aiming sensitivity, 67 vulnerability, 54 MiniDisc system, 151 Mitsubishi interfaces, 188–9

214 Index Mix min see Clean feed Mixing desk, 195 Modal density, 41 Modulation index, 11 Monitoring systems, sound mixing, 101–103 Mono line channel, 98 Most signiﬁcant bit (MSB), 38 Moving-coil drive units, loudspeaker, 76 Moving-coil microphones, 55–6 Moving-coil receiver, 180 Multiple driver units, loudspeaker, 77 Musical Instrument Digital Interface (MIDI), 166 daisy-chaining, 167 hardware interface, 166 message data format, 166, 167–71 sequencer, 171 Synchronization System, 172 Scale, 14, 15 Musical Terms, Italian, 17–18 Nagra D Machines, 142 Sound: Definitions, 23–4 Design Criteria, 45–6 see also Hearing Damage Sound Criteria (NC) Curves, 48 Noise Gates, 111 Sound level meters, 23, 25–6, 157 Sound levels: addition, 26 effect of different materials, 50–52 measurement, 23–4, 26, 154, 157 measurement standards, 24, 158 practical reduction systems, 125–9 typical, 28–30 Noise masking , 21, 122 Noise classification (NR) curves, 47 Normalization (interconnections), 96, 97

Ohm's law, 5 omnidirectional microphones, 59–62 Oversampling, 39 P and Q subcodes, 136, 138 Panpot microphone systems, 92, 93 Parity (error detection), 39–40 Peak programmable (PPM), 103, 157 Free space permeability , 3 Phantom power systems, 69 Phase suppression, 60, 61 Phon (unit), 21 Piezoelectric loudspeaker, 77 Piezoelectric transducers, 191 bandwidth, 191 electromechanical coupling coefficient, 192 Playback formats, 199 Polar diagrams see Directional patterns Power ratios, decibel values , 8 Pre-fade listen (PFL), 100 pressure units, 12 ProDigi (PD) machines, 142 profanity delay, 197 program level measurements, 157, 158 proximity effect, 63, 64 Public Address systems, 205–206 quantization levels, 37 Radio drama studios, 200, 201 Radio microphone frequencies, 72– 5 Radio propagation, 182–5 Radio station, semi-automatic, 198 Radio studio facilities, 195–201 Rayleigh formula, 41 Recording formats, 199 Reference field strength, 182

Index 1111 2 3 4 5 6 7 8111 9 10 1 2 3 4 5 6 7 8 9 20 1 2 3 4 5 6 7 8 9 30 1 2 3 4 5 6 7 8 9 40111

Relations, useful, 1 Relative humidity, 174, 175 Relative permeability, 3 Replay amplifier equalization, 117, 118 Resistance, color code, 6 Resonance formulas, 4, 5, 6 Reverberation, artificial, 159, 162 Reverberation time, 41 effect on sound reduction index , 49 recommendations, 43, 44 studio, 199, 200 Ribbon speaker, 77 Ribbon microphone, 57–8 Rock music studios, 199 listening levels, 199 Rocking armature receiver, 180 S signals, 87, 91 Sabine formula, 41 Safeguards, transmission, 197 Sampling rate, 37 Signal levels, 94 Signal processing, 104–11 Signal-to-noise ratios (S/N), 38 Solo-in-place, sound mixing, 101 Sone (unit), 20 Sony digital interface (SDIF), 186, 187 Sony PCM1630 machines, 144 Sony-Philips digital interface (SPDIF), 186 Sound absorption, for different relative humidity, 206 Sound image position: difference between channels, 86 for stereo microphones, 89 Sound insulation: in the air, 49 building materials, 49–51 Sound level meters see Sound level meters Sound mixing, analog, 94

215

Sound pressure level (SPL): decibel values, 5, 9, 10 speaker sensitivity, 83 reference zero, 19 Sound reduction index (SRI), 49 effect of reverberation time, 49 typical values, 51–2 Speed of sound, 12, 13 Sound waves, physics, 12–18 Standard equalizations, IEC and NAB standards, 118 Standing wave patterns, 16 Star quad cable, 94 Stereo effects, 84–92 M and S signals, 87, 88 terminology, 87 Stereo line channel, 99 Studio reverb times, 44 Synchronous signals timing, 190 Tape head adjustments, 120 Tape head data, 113 Telcom c4 noise reduction system, 129, 130 Phone balance unit (TBU), 195, 197 Phones: electromagnetic, 177–8 electronic, 178–9 Temperature requirements, studio, 173–4 Threshold, 104 Tilt control, 104, 106, 107 Time of arrival (TOA) difference, 84 Tone control, 104, 105 Transmission line housing, 82 Transmission protections, 197 Transmitter licenses, 75 TV channel frequencies, 73, 74 Two's complement, 38 Ultrasonics, 191–3 industrial use, 194 Unbalanced circuit connectors, 202

216 Index Universal asynchronous receiver/transmitter (UART), 166 Microphone with variable directivity, 67 Vented enclosures see Helmholtz resonator Ventilation requirements, studio, 173 VHF frequencies, 72 VHS Hi-Fi system, 130 Vibration frequencies, 14, 17 Vibration isolation analysis, 36 Visual monitoring, sound mixing, 101

Voltage standardization, European, 202, 204 Volume unit (VU) meter, 157 Wavelength, 3, 12, 13 Weighting curves: noise measurement, 154, 155, 156 program level measurement, 157 Yamaha cascade interface, 189–90 Zenith adjustment, tape head, 120

Also available from Focal Press... Audio Engineer's Reference Book Edited by Michael Talbot-Smith An authoritative book covering all aspects of audio engineering and technology, including: • basic math and formulas • acoustics and psychoacoustics • microphones • loudspeakers • studio installations, including air conditioning Compiled by an international team of experts, the second edition has been updated to keep abreast of rapidly evolving areas such as digital audio and transmission technology. For professionals involved in the design, manufacture and installation of all types of audio equipment, this reference will prove to be an invaluable resource. An excellent readable compendium of both theory and practice. It is clear that the writers are experts, making this a valuable reference. It's all in it; from definitions of mass, time and flow, to setting up and running broadcasting and recording studios.” Sound & Communications International

1999 • 640pp • 246 x 189mm • Hardback ISBN 0 240 515285 To order your copy, call +44 (0)1865 888180 (UK) or +1 800 366 2665 (US) Or visit our website at http://www .focalpress. com

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FAQs

Is there a textbook for audio engineering? ›

David Miles Huber's Modern Recording Techniques is the Bible when it comes to audio engineering. You will find this book in almost every AE / music production class as it is the authoritative textbook for audio engineers, old and new.

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How long does it take to learn how do you audio engineer? ›

An associate degree in audio engineering takes two years, while a bachelor's degree takes four years. Some of the courses you can take while pursuing these degrees blend technical, business and music concepts to build you into a holistic professional ready to deploy your chosen fields' skills.

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Can you self teach audio engineering? ›

Typically audio engineers will have some college education or vocational training at a specialized recording studio, however, many audio engineers are also self-taught under the guidance of a mentor.

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Is it hard to learn sound engineering? ›

Newcomers to this task may find it difficult to understand the basics and what is important or what should be focused on when starting this new journey. Though it can be difficult at the start, once you get the hang of what you're doing and how to do it, you can exceed expectations to pass being a beginner.

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Is an audio engineering certificate worth it? ›

While some jobs do not require a certificate or degree, the Bureau of Labor Statistics (BLS) says that most audio engineers have one. And there's a good reason for that; going to an audio engineering school is one of the best things you can do for your career.

Where to find audio college textbooks? ›

The most popular services for audio textbooks include Amazon's premium audiobooks service Audible.com and OverDrive.

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Do you have to be good at math to be an Audio Engineer? ›

Sound engineering course comes with a lot of math calculations. But again, it is not altogether necessary. The sound engineering courses offer required skills for the usage of equipment needed for mixing, to reproduce and record sounds.

Can you be a sound engineer without a degree? ›

Can you be an Audio Engineer without a Degree? You do not need a degree to be an audio engineer. A degree is not necessary for the music industry. Having the knowledge, proficiency, and skills is what is going to help you become a good sound engineer.

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What is the average age of an Audio Engineer? ›

The average age of an employed audio engineer is 42 years old.

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Do audio engineers make money? ›

Working as an Audio Engineer also means you'll earn a decent salary. Entry-level Engineers can expect to make about $40,000 annually as they're building their career while top Engineers can earn up to seven figures! Read on for a deeper look at the salary an experienced Audio Engineer can expect to make.

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Do audio engineers need to know music? ›

A Good Ear For Music: Audio engineers must monitor volume levels, sound quality and so on. Therefore, you should know your stuff when it comes to music and critical listening. Communication: As an audio engineer, you will be working with various creatives.

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How do I start learning sound engineering? ›

Here are four paths you can take to become an audio engineer:

Take a college degree program in audio production.
Attend an audio production trade school.
Earn individual certifications (Pro tools, etc.)
Apprenticeship programs.

Jun 23, 2022

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What are the cons of being a sound engineer? ›

Long working hours: In the beginning a sound engineer or music producer has to work on one or more projects at the same time to get the job. Sometimes one has to work continuously for many hours in order to finish the project on time. Therefore, working hours in this field can be quite long.

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How can I get better at sound engineering? ›

20 Steps to Becoming a Better Audio Engineer

Learn about the gear you need.
Know how to use microphones.
Be aware of your surroundings.
Avoid digital recording mistakes.
Learn to understand the frequency spectrum.
Learn to record vocals.
Know how to get a great guitar sound.
Create a massive acoustic guitar sound.

More items...

Get More Info Here ›

How can I succeed as a sound engineer? ›

How to become a successful audio engineer

Know Your Way Around any Recording Studio. Obviously, there's more to being an audio engineer than just saying that's what you are. ...
Work, Rinse, Repeat. Like everything else in life, you won't get their overnight. ...
Build Your Portfolio. ...
Go Pro in a Commercial Recording Studio.

Sep 28, 2019

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What is the difference between sound engineer and audio engineer? ›

Firstly, audio engineers and sound engineers are terms that are often used interchangeably. However, most use 'audio engineer' to refer to recording or studio work, and 'sound engineer' to refer to live concerts and events. An audio engineer can also be known as a recording engineer.

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What skills do you need to be a sound engineer? ›

10 examples of audio engineering skills

Hardware management. To produce effective sounds, an audio engineer must know how to operate different hardware systems and use their features. ...
Digital sound software. ...
Equalizing methods. ...
Mixing techniques. ...
Acoustics. ...
Music theory knowledge. ...
Problem-solving. ...
Communication.

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Feb 3, 2023

What is the difference between an audio tech and an audio engineer? ›

Most audio technician duties involve the setup, breakdown, and maintenance of equipment. A sound engineer takes on a more supervisory role and has a certain level of creative freedom to influence the sound of a recording or performance.

Keep Reading ›

How to get all college textbooks for free? ›

33 Places to Find Free College Textbooks Online

Internet Archive. Internet Archive is a not-for-profit library that contains millions of e-books, software, music, and so much more — all for free. ...
Bookboon. ...
Project Gutenberg. ...
Library Genesis. ...
Mobilism. ...
OpenStax. ...
OpenEd. ...
BioRxiv.

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Jan 20, 2023

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Are audio books still a thing? ›

While a visual format remained the most-used by readers, the percentage of audiobook listeners rose during the same period — the only format to see an increase in use. More readers are being drawn to audiobooks, often because the format allows the listener to do other things while enjoying a good book.

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What free audible books do college students get? ›

Audible's student discount also includes three free audio books that students can keep forever. That means that students can download English class classics like To Kill a Mockingbird, The Great Gatsby and Fahrenhenheit 451 to read later in the school year.

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What is the highest paid audio engineer? ›

High Paying Audio Engineer Jobs

DSP Engineer. Salary range: $143,000-$167,000 per year. ...
Acoustic Engineer. Salary range: $78,500-$132,500 per year. ...
Voice Engineer. Salary range: $81,000-$127,500 per year. ...
Balance Engineer. ...
Audio Supervisor. ...
Sound Designer. ...
Recording Engineer. ...
Audio Experience Expert.

More items...

Tell Me More ›

What kind of books best for audio? ›

4 Types of Books That are Better as Audiobooks

Humorous Books Read by the Author. This one is a no-brainer to me. ...
Full Cast Audiobooks. Another category of books that are better as audiobooks are those with a full cast recording. ...
Emotionally Raw Memoirs. ...
Fiction With a Narrative Voice Unlike Mine.

May 3, 2019

Know More ›

What is the best way to become an audio engineer? ›

Here are four paths you can take to become an audio engineer:

Take a college degree program in audio production.
Attend an audio production trade school.
Earn individual certifications (Pro tools, etc.)
Apprenticeship programs.

Jun 23, 2022

Discover More ›

How can I get better at audio engineering? ›

20 Steps to Becoming a Better Audio Engineer

Learn about the gear you need.
Know how to use microphones.
Be aware of your surroundings.
Avoid digital recording mistakes.
Learn to understand the frequency spectrum.
Learn to record vocals.
Know how to get a great guitar sound.
Create a massive acoustic guitar sound.

More items...

Discover More ›

What is the average salary in the US today? ›

The average annual average salary in the U.S. is $60,575.

Learn More ›

Do sound engineers get royalties? ›

Do mixing and mastering engineers get royalties? The short answer is: Most mixing and mastering engineers don't receive any share in the song's revenue. However, in some cases, a mixing engineer might get 1 point, which means they will be able to collect 1% of master royalties.

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Do sound engineers make good money? ›

A mid-career Sound Engineer with 4-9 years of experience earns an average salary of ₹4.6 Lakhs per year, while an experienced Sound Engineer with 10-20 years of experience earns an average salary of ₹6.1 Lakhs per year.

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Are audio books good for your brain? ›

According to the Audio Publishers Association, audiobooks help “build and enhance vital literacy skills such as fluency, vocabulary, language acquisition, pronunciation, phonemic awareness, and comprehension—skills that often boost reading scores.” Need some audiobook recommendations for kids?

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Is listening to the audio better than reading? ›

Research has demonstrated that people who listen to audiobooks are able to recall more information than those who read from a traditional book. Additionally, studies have found that people who listen to audiobooks retain information better over time compared to those who read books in the traditional way.

Get More Info Here ›

Is listening to audio book as good as reading? ›

There is little to no difference in comprehension between the two types of consuming literature. Even though the information is processed differently by our brain, recent audiobooks vs reading research from 2021 showed that the overall difference between reading and listening in terms of comprehension was negligible.

What is the average age of an audio engineer? ›

The average age of an employed audio engineer is 42 years old.

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Do you have to be good at math to be an audio engineer? ›

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Is it hard to learn audio engineering? ›

Learning audio engineering isn't something you pick up overnight while messing around with Pro Tools. It often takes a few years to nail down the basics. Many audio engineering programs last from two to four years. Like any other detailed occupation, you are always learning new and useful techniques for your work.

Keep Reading ›

Is audio engineering a skill? ›

To become an Audio Engineer, you need: to be both creative and technically minded. good problem-solving skills. to enjoy working on the recording process from start to finish.