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Analogue Sound Restoration

A wide ranging document on the restoration of "older" sound recordings. Chapter six is especially worth reading on the different types of equalization used on LP records.

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Charles F Port
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100% found this document useful (2 votes)
2K views340 pages

Analogue Sound Restoration

A wide ranging document on the restoration of "older" sound recordings. Chapter six is especially worth reading on the different types of equalization used on LP records.

Uploaded by

Charles F Port
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

MANUAL OF ANALOGUE

SOUND RESTORATION
TECHNIQUES
by Peter Copeland

The British Library


Analogue Sound Restoration Techniques

MANUAL OF ANALOGUE SOUND


RESTORATION TECHNIQUES

by Peter Copeland

This manual is dedicated to the memory of

Patrick Saul

who founded the British Institute of Recorded Sound, *


and was its director from 1953 to 1978,
thereby setting the scene which made this manual possible.

Published September 2008 by The British Library


96 Euston Road, London NW1 2DB
Copyright 2008, The British Library Board
www.bl.uk

*
renamed the British Library Sound Archive in 1983.

ii
Analogue Sound Restoration Techniques

CONTENTS

Preface ................................................................................................................................................................1
Acknowledgements .............................................................................................................................................2
1 Introduction ..............................................................................................................................................3
1.1 The organisation of this manual ...........................................................................................................3
1.2 The target audience for this manual .....................................................................................................4
1.3 The original sound................................................................................................................................6
1.4 Operational principles ..........................................................................................................................8
1.5 A quadruple conservation strategy .....................................................................................................10
1.6 How to achieve objectivity .................................................................................................................11
1.7 The necessity for documentation........................................................................................................12
2 The overall copying strategy....................................................................................................................13
2.1 The problem to be solved...................................................................................................................13
2.2 General issues.....................................................................................................................................13
2.3 The principle of the “Power-Bandwidth Product” ..............................................................................14
2.4 Restricting the bandwidth ..................................................................................................................16
2.5 Deciding priorities...............................................................................................................................17
2.6 Getting the best original power-bandwidth product...........................................................................17
2.7 Archive, objective, and service copies .................................................................................................19
2.8 “Partially objective” copies.................................................................................................................20
2.9 Documentation strategy.....................................................................................................................20
2.10 Absolute phase..............................................................................................................................22
2.11 Relative phase ...............................................................................................................................23
2.12 Scale distortion..............................................................................................................................23
2.13 Conclusion ....................................................................................................................................24
3 Digital conversion of analogue sound......................................................................................................25
3.1 The advantages of digital audio..........................................................................................................25
3.2 Technical restrictions of digital audio - the “power” element .............................................................26
3.3 Technical limitations of digital audio: the “bandwidth” element ........................................................28
3.4 Operational techniques for digital encoding .......................................................................................30
3.5 Difficulties of “cloning” digital recordings ..........................................................................................30
3.6 Digital data compression ....................................................................................................................33
3.7 A severe warning................................................................................................................................35
3.8 Digital watermarking and copy protection..........................................................................................37
3.9 The use of general-purpose computers...............................................................................................38
3.10 Processes better handled in the analogue domain .........................................................................39
3.11 Digital recording media .................................................................................................................40
4 Grooves and styli.....................................................................................................................................42
4.1 Introduction .......................................................................................................................................42
4.2 Basic turntable principles ....................................................................................................................43
4.3 Pickups and other devices ..................................................................................................................44
4.4 Conventional electrical pickup considerations.....................................................................................45
4.5 Operational procedure for selecting a stylus.......................................................................................47
4.6 U-shaped and V-shaped grooves .......................................................................................................48
4.7 The principle of minimising groove hiss ..............................................................................................52
4.8 “Soft” replay styli...............................................................................................................................53
4.9 “Hard” replay styli .............................................................................................................................55
4.10 Stereo techniques..........................................................................................................................57
4.11 “Elliptical” and other styli..............................................................................................................59
4.12 Other considerations .....................................................................................................................61
4.13 Playing records backwards ............................................................................................................63
4.14 Half-speed copying .......................................................................................................................65
4.15 Distortion correction......................................................................................................................65

iii
Analogue Sound Restoration Techniques

4.16 Radius compensation ....................................................................................................................67


4.17 Electronic click reduction ...............................................................................................................70
4.18 Electronic hiss reduction ................................................................................................................74
4.19 Eliminating rumble ........................................................................................................................76
4.20 De-thumping ................................................................................................................................76
4.21 Future developments.....................................................................................................................77
4.22 Recommendations and conclusion ................................................................................................78
5 Speed setting...........................................................................................................................................81
5.1 Introduction .......................................................................................................................................81
5.2 History of speed control .....................................................................................................................82
5.3 History of speed-control in visual media .............................................................................................84
5.4 Setting the speed of old commercial sound records ............................................................................86
5.5 Musical considerations .......................................................................................................................90
5.6 Strengths and weaknesses of “standard pitch” ..................................................................................91
5.7 Non-“standard” pitches .....................................................................................................................91
5.8 The use of vocal quality......................................................................................................................92
5.9 Variable-speed recordings ..................................................................................................................93
5.10 Engineering evidence ....................................................................................................................95
5.11 Timings .........................................................................................................................................97
6 Frequency responses of grooved media...................................................................................................99
6.1 The problem stated ............................................................................................................................99
6.2 A broad history of equalisation.........................................................................................................100
6.3 Why previous writers have gone wrong ...........................................................................................100
6.4 Two ways to define “a flat frequency response”..............................................................................101
6.5 Equalisation ethics and philosophy ...................................................................................................102
6.6 Old frequency records as evidence of characteristics ........................................................................103
6.7 Two common characteristics ............................................................................................................104
6.8 Practical limits to equalisation...........................................................................................................106
6.9 Practical test discs.............................................................................................................................107
6.10 International standard microgroove test discs..............................................................................107
6.11 Coarsegroove (78rpm) test discs .................................................................................................109
6.12 Generalised study of electromagnetic cutters ..............................................................................111
6.13 Characteristics of “simple” cutterheads.......................................................................................111
6.14 High-resistance cutterheads ........................................................................................................115
6.15 Western Electric, and similar line-damped recording systems ......................................................116
6.16 Western Electric revolutionises sound recording ..........................................................................117
6.17 The Western Electric microphone ................................................................................................117
6.18 The Western Electric amplifier .....................................................................................................118
6.19 Documentation of HMV amplifier settings, 1925-1931...............................................................118
6.20 The Western Electric cutterhead..................................................................................................119
6.21 How to recognise recordings made with Western Electric 1A and 1B systems .............................121
6.22 Summary of equalisation techniques for the above .....................................................................122
6.23 Western Electric developments after 1931 ..................................................................................122
6.24 Recognising recordings made on RCA-modified and Western Electric 1C and 1D systems..........124
6.25 Summary of equalisation techniques for the above .....................................................................124
6.26 Other systems using line-damped cutterheads: British Brunswick, Decca, DGG 1925-1939 ........124
6.27 Other systems using line-damped cutterheads: The Lindström system ........................................125
6.28 Conclusion to line-damped systems ............................................................................................126
6.29 The Blumlein system....................................................................................................................127
6.30 The Blumlein microphone............................................................................................................127
6.31 The Blumlein moving-coil cutterhead ..........................................................................................128
6.32 Test discs made by Blumlein’s system..........................................................................................129
6.33 How to recognise Blumlein cutters on commercial records, 1931-1944.......................................130
6.34 Summary of how to equalise the above ......................................................................................130
6.35 The Gramophone Company system ............................................................................................130
6.36 How to recognise the Gramophone Company system on commercial records.............................131
6.37 Summary of how to equalise “Gramophone system” recordings.................................................131
6.38 Extended-Range Blumlein recordings (1943-5) and later systems................................................131
6.39 Summary of how to equalise “Extended-Range” and subsequent systems .................................132

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Analogue Sound Restoration Techniques

6.40 Early EMI long-playing and 45 r.p.m. records..............................................................................132


6.41 Other systems giving a “Blumlein-shaped” curve - amateur and semi-pro machines ..................133
6.42 British Decca-group recordings 1935-1944 .................................................................................134
6.43 Summary of how to equalise Decca-group recordings 1935-1944 ..............................................135
6.44 Synchrophone .............................................................................................................................135
6.45 Summary of how to equalise Synchrophone recordings ..............................................................135
6.46 Conclusion to “Blumlein-shaped” equalisation............................................................................135
6.47 The Marconi system ....................................................................................................................136
6.48 BBC Disc Record equalisation - Introduction................................................................................137
6.49 The subject matter of these sections............................................................................................138
6.50 Pre-history of BBC disc recording ................................................................................................139
6.51 BBC matrix numbers....................................................................................................................139
6.52 BBC “current library” numbers....................................................................................................140
6.53 BBC M.S.S. recordings.................................................................................................................141
6.54 BBC American equipment............................................................................................................142
6.55 BBC transportable equipment......................................................................................................142
6.56 BBC coarsegroove 33rpm discs....................................................................................................142
6.57 Later BBC coarsegroove systems .................................................................................................143
6.58 Early BBC microgroove discs........................................................................................................145
6.59 BBC “CCIR characteristics” .........................................................................................................145
6.60 RIAA and subsequently ...............................................................................................................146
6.61 Brief summary of BBC characteristics...........................................................................................146
6.62 “Standard” equalisation curves ...................................................................................................147
6.63 General history of “standards”....................................................................................................147
6.64 Defining standard curves.............................................................................................................149
6.65 Discographical problems..............................................................................................................150
6.66 General history of changeover procedures ..................................................................................151
6.67 NAB (later “NARTB”) characteristics ...........................................................................................152
6.68 Decca Group (UK) “ffrr” characteristics ......................................................................................152
6.69 American Columbia LPs...............................................................................................................155
6.70 “AES” characteristics...................................................................................................................156
6.71 Various RCA characteristics .........................................................................................................157
6.72 “CCIR” characteristics.................................................................................................................158
6.73 “DIN” characteristics...................................................................................................................158
6.74 Concluding remarks ....................................................................................................................159
7 Analogue tape reproduction ..................................................................................................................163
7.1 Preliminary remarks ..........................................................................................................................163
7.2 Historical development of magnetic sound recording .......................................................................164
7.3 Bias ..................................................................................................................................................166
7.4 Magnetised tape heads ....................................................................................................................168
7.5 Print-through ...................................................................................................................................169
7.6 Azimuth ...........................................................................................................................................170
7.7 Frequency responses of tape recordings ...........................................................................................172
7.8 “Standard” characteristics on open-reel tapes..................................................................................173
7.9 Standards on tape cassettes..............................................................................................................176
7.10 Operational principles .................................................................................................................177
7.11 “Mono” and “half-track” tapes..................................................................................................180
7.12 “Twin-track” tapes .....................................................................................................................181
7.13 “Quarter-track” tapes.................................................................................................................182
7.14 Practical reproduction issues........................................................................................................183
7.15 “Hi-fi” Tracks on domestic video. ...............................................................................................184
8 Optical film reproduction.......................................................................................................................187
8.1 Introduction .....................................................................................................................................187
8.2 Optical sound with moving pictures .................................................................................................187
8.3 Considerations of strategy................................................................................................................188
8.4 Basic types of optical soundtracks.....................................................................................................189
8.5 Soundtracks combined with optical picture media ............................................................................190
8.6 Recovering the power-bandwidth product .......................................................................................191
8.7 Frequency responses ........................................................................................................................193

v
Analogue Sound Restoration Techniques

8.8 Reducing background noise .............................................................................................................194


9 Reciprocal noise reduction .....................................................................................................................196
9.1 Principles of noise reduction .............................................................................................................196
9.2 Non-reciprocal and reciprocal noise-reduction..................................................................................196
9.3 Recognising reciprocal noise reduction systems ................................................................................197
9.4 Principles of reciprocal noise reduction systems ................................................................................198
9.5 Dolby “A”........................................................................................................................................199
9.6 Dolby “B” ........................................................................................................................................201
9.7 DBX systems.....................................................................................................................................202
9.8 JVC ANRS (Audio Noise Reduction System) .....................................................................................205
9.9 Telcom C4........................................................................................................................................205
9.10 Telefunken “High-Com”.............................................................................................................206
9.11 The “CX” systems.......................................................................................................................206
9.12 Dolby “C”...................................................................................................................................207
9.13 Dolby SR and Dolby S .................................................................................................................208
9.14 Other noise reduction systems ....................................................................................................209
9.15 Noise reduction systems not needing treatment..........................................................................211
9.16 Conclusion ..................................................................................................................................211
10 Spatial recordings ..................................................................................................................................213
10.1 Introduction ................................................................................................................................213
10.2 “Two-channel” recordings ..........................................................................................................213
10.3 Archaeological stereo ..................................................................................................................216
10.4 Matrixing into two channels........................................................................................................218
10.5 Three-channel recordings. ...........................................................................................................218
10.6 Four-channel recordings - in the cinema .....................................................................................219
10.7 Four-channel audio-only principles..............................................................................................220
10.8 Matrix “quadraphonic” systems..................................................................................................221
10.9 The QS system ............................................................................................................................221
10.10 The SQ system ............................................................................................................................222
10.11 Matrix H .....................................................................................................................................223
10.12 “Dolby Stereo” ...........................................................................................................................224
10.13 Developments of “Dolby Stereo” - (a) Dolby AC-3 ....................................................................226
10.14 Developments of “Dolby Stereo” - (b) Dolby headphone ...........................................................227
10.15 Developments of “Dolby Stereo” - (c) Pro Logic 2......................................................................227
10.16 Discrete 4-channel systems - The JVC CD-4 system....................................................................227
10.17 The “UD-4” system ....................................................................................................................228
10.18 “Ambisonics”..............................................................................................................................229
10.19 Other discrete four-channel media..............................................................................................230
10.20 More than five-channel systems..................................................................................................230
10.21 Multitrack master-tapes ..............................................................................................................231
11 Dynamics...............................................................................................................................................233
11.1 Introduction ................................................................................................................................233
11.2 The reasons for dynamic compression .........................................................................................234
11.3 Acoustic recording.......................................................................................................................235
11.4 Manually-controlled electrical recording......................................................................................235
11.5 Procedures for reversing manual control .....................................................................................237
11.6 Automatic volume controlling .....................................................................................................238
11.7 Principles of limiters and compressors..........................................................................................241
11.8 Identifying limited recordings ......................................................................................................243
11.9 Attack times ................................................................................................................................244
11.10 Decay-times ................................................................................................................................245
11.11 The compression-ratio and how to kludge It ...............................................................................246
12 Acoustic recordings ...............................................................................................................................249
12.1 Introduction ................................................................................................................................249
12.2 Ethical matters.............................................................................................................................249
12.3 Overall view of acoustic recording hardware...............................................................................251
12.4 Performance modifications ..........................................................................................................252
12.5 Procedures for reverse-engineering the effects............................................................................254
12.6 Documentation of HMV acoustic recording equipment...............................................................255

vi
Analogue Sound Restoration Techniques

12.7 The Recording horn - why horns were used ................................................................................256


12.8 The lack of bass with horn recording...........................................................................................257
12.9 Resonances of air within the horn ...............................................................................................257
12.10 Experimental methodology..........................................................................................................257
12.11 Design of an Acoustic Horn Equaliser ..........................................................................................259
12.12 Resonances of the horn itself.......................................................................................................259
12.13 Positions of artists in relation to a conical horn ............................................................................261
12.14 Acoustic impedances ...................................................................................................................262
12.15 Joining two horns........................................................................................................................263
12.16 Joining three or more horns ........................................................................................................264
12.17 Another way of connecting several horns....................................................................................264
12.18 Electrical equalisation of recordings made with parallel-sided tubes.............................................265
12.19 Electrical equalisation of recordings made with multiple recording horns.....................................266
12.20 The recording soundbox..............................................................................................................267
12.21 “Lumped” and “distributed” components ..................................................................................267
12.22 How pre-1925 soundboxes worked ............................................................................................268
12.23 Practicalities of acoustic recording diaphragms ............................................................................270
12.24 The rest of the soundbox ............................................................................................................271
12.25 Notes on variations .....................................................................................................................271
12.26 When we should apply these lessons ..........................................................................................274
12.27 Summary of present-day equalisation possibilities .......................................................................274
12.28 Conclusion ..................................................................................................................................275
13 The engineer and the artist....................................................................................................................279
13.1 Introduction ................................................................................................................................279
13.2 The effect of playing-time upon recorded performances .............................................................279
13.3 The introduction of the microphone............................................................................................283
13.4 The performing environment ......................................................................................................284
13.5 “Multitrack” issues......................................................................................................................287
13.6 Signal strengths...........................................................................................................................289
13.7 Frequency ranges ........................................................................................................................290
13.8 Monitoring sound recordings ......................................................................................................291
13.9 The effects of playback ...............................................................................................................292
13.10 The Costs of recording, making copies, and playback .................................................................295
13.10.1 The dawn of sound recording ...........................................................................................295
13.10.2 Mass produced cylinders...................................................................................................297
13.10.3 Coarsegroove disc mastering costs....................................................................................297
13.10.4 Retail prices of coarsegroove pressings..............................................................................298
13.10.5 One-off disc recording ......................................................................................................300
13.10.6 Magnetic tape costs ..........................................................................................................301
13.10.7 Pre-recorded tapes and cassettes ......................................................................................302
13.10.8 Popular music ...................................................................................................................303
13.10.9 Digital formats ..................................................................................................................303
13.10.10 Conclusion for pure sound recordings ...............................................................................304
13.11 The cinema and the performer ....................................................................................................304
13.12 Film sound on disc.......................................................................................................................305
13.13 The art of film sound...................................................................................................................307
13.14 Film sound editing and dubbing ..................................................................................................309
13.15 The automatic volume limiter......................................................................................................311
13.16 Films, video, and the acoustic environment .................................................................................312
13.17 Making continuous performances ...............................................................................................313
13.18 Audio monitoring for visual media ..............................................................................................313
13.19 Listening conditions and the target audience and its equipment..................................................314
13.20 The influence of naturalism .........................................................................................................316
Appendix 1. Preparing media for playback ......................................................................................................318
Appendix 2: Aligning analogue tape reproducers.............................................................................................328

vii
Preface
With the rapid pace of change in audio technology, analogue formats have all but disappeared as
a means for current production and distribution of sound recordings. Nevertheless, many audio
archivists are responsible for large numbers of valuable audio recordings in analogue formats.
These require dedicated playback machines that have become obsolete, so the only way to ensure
lasting access to this legacy is digitisation. To do this properly requires firstly that the optimum
signal is extracted during playback from an analogue carrier, and this in turn necessitates extensive
knowledge of the engineering processes and standards used at the time of its creation. The
passing on of expertise and detailed knowledge gained during a time when analogue technology
represented the cutting edge, is therefore of vital importance to subsequent generations, and it is
with this in mind that this work was written.
The manual was written by Peter Copeland when he was Conservation Manager at the
British Library Sound Archive over a ten-year period up until 2002, as an aid to audio engineers
and audio archivists. Peter started his professional career at the BBC in 1961, initially working from
Bush House in London for the World Service. From 1966 he worked as a dubbing engineer at BBC
Bristol, taking on responsibility for sound in many productions by the famed Natural History Unit,
among them the groundbreaking David Attenborough series, Life on Earth. In 1986 he joined
the British Library as Conservation Manager for the Sound Archive where he started work on this
manual. After Peter retired from the Library in 2002, we began work with him to polish the text
ready for publication. The work was near to completion when Peter died in July 2006.
The original intention was to thoroughly edit the manual to bring it up to date, with
appropriate diagrams and other illustrations added. However in the extremely fast-moving world
of audiovisual archiving this would have entailed a great deal of rewriting, such that it would no
longer have been the manuscript that Peter left us. After much discussion therefore, the decision
has been taken to publish it now, largely unchanged from its original form, and make it freely
available on the internet as a service to professional audiovisual engineers and archivists.
Readers should bear in mind that the manual was written over a period of time some years
ago, and so should be considered as a snapshot of Peter’s perspective during that time. Some
information, particularly on digital conversion in chapter 3, is outdated: the R-DAT format is now
obsolete, professional audio engineers routinely use computer-based hardware and software for
audio processing with 24-bit converters at sample rates exceeding 96kHz, and sigma delta
processors are now available. The core of the manual however, being concerned with making
sense of the incredibly rich and complex history of analogue technology, remains a singular feat of
rigorous, sustained research, and is unlikely to date. Some statements in the text represent
personal opinion, although this is always clearly stated. The somewhat quirky, personal style in
which the manual is written will be very familiar to anyone who knew Peter, and was as much a
part of him as was his passion for the subject of this work. Minor amendments to the original
texts have been made, mostly ironing out inconsistencies in style and for this I would like to thank
Christine Adams, Nigel Bewley, Bill Lowry, Will Prentice, Andrew Pearson and Tom Ruane who
helped check through the manual in its entirety.
This work assembles in one place a tremendous amount of factual information amassed
during Peter’s long career as an audio engineer - information that is difficult to find anywhere
else. The breadth of research into the history of sound playback is unequalled. Peter himself
sometimes referred to his role as that of an “audio archaeologist”. This manual is a fitting and
lasting testament to Peter’s combination of depth of knowledge with clarity of expression.

Richard Ranft
The British Library Sound Archive
September 2008

1
Acknowledgements

Few types of torture can be more exquisite than writing a book about a subject which is
instinctive to the writer. This has been the most difficult piece of writing I have ever
undertaken, because everything lives in my brain in a very disorganised form. My first
thanks must be to the inventors of the word processor, who allowed me to get my very
thoughts into order.
Next I must acknowledge the many talented people who gave up considerable
amounts of time to reading printouts at various stages. First, I would like to thank Lloyd
Stickells, a former engineer of the British Library Sound Archive, who looked through a
very early draft of this work while he was still with the Archive, and made sufficiently
encouraging noises for me to want to continue. Steve Smolian, of Smolian Sound Studios,
Potomac, Maryland, also made encouraging noises; moreover, he helped with many
suggestions when I sought advice from the American point of view. Alistair Bamford also
made encouraging noises as the work proceeded. Ted Kendall and Roger Wilmut read it
more recently when it was nearly at full length, and Crispin Jewitt (Head of the Sound
Archive) was rash enough to put his money where his mouth was, financing a Pilot
Digitisation Project to test some of the methodology. As a result, Serena Lovelace was in
the front line, and gave me very clear (and polite) feedback on every section of the book.
But the biggest bouquet must go to my editor, David Way, who was obliged to read the
whole thing many times over.
I must thank the following individuals for supplying me with specific pieces of
information or actual artefacts, and I just hope I have reported the information correctly:
The British Library Science Research and Information Service, Sean Davies (S. W. Davies
Ltd.), Eliot Levin (Symposium Records), Alistair Murray, Noel Sidebottom (British Library
Sound Archive), Pete Thomas (BBC), and Roger Wilmut. George Overstall, the guru of
acoustic soundboxes and horns, gave me an afternoon of his time to the mechanics of
treating and playing 78rpm discs; chapter 3 could not have been written without his help.
Mrs. Ruth Edge, of the EMI Archive in Hayes, kindly allowed me to disassemble the
world’s last surviving professional acoustic recording machine, and measure its horns,
without which Chapter 11 would have been impossible.
I was able to try several chapters of this manual upon an unsuspecting public,
when Jack Wrigley asked me to write for his magazine The Historic Record. On the whole
the readership did not argue with me, which boosted my morale despite my torture; the
few comments I received have certainly resulted in greater clarity. So thanks to all Jack’s
readers as well.
But nothing at all could have been done without the continuous support of the
Information Service of the British Library Sound Archive. It seems invidious to name
anyone in particular, but I must thank Mavis Hammond-Brake, who happened to draw
the short straw every time I required some information from a different department of the
British Library. Sorry, Mavis, it wasn’t planned that way! Since then, the same straw has
been drawn by Lee Taylor - sorry, Lee.

Peter Copeland
London, 28th February 2001

2
1 Introduction

1.1 The organisation of this manual

This manual gives details of some of the techniques to be followed when old sound
recordings are transferred to more modern carriers. It is aimed primarily at the professional
archivist transferring them to conserve them, and it is generally assumed that the purpose
is TO PRESERVE THE ORIGINAL SOUND. (That is what I understand to be the function of
a “National Sound Archive”, rather than preservation of the artefacts; that would be the
function of a “Museum.” Please feel free to disagree with me, though!)
Here is one disagreement I myself accept. In many cases the work of people
behind the scenes is just as important as that of the performer, for example in editing
defective sections of a performance; so this must often be modified to read “to preserve
the original intended sound.” I would enlarge this in two ways. When it comes to the
subject matter, it must surely mean “intended by the producer of the recording” (or the
film or the broadcast), although this will become rather a subjective judgement. And
when it comes to technical matters, it must mean “Intended by the Sound Engineer.”
Hence this manual!
I also need to define the word “sound”. Do we mean a psychoacoustic sensation,
or objective variations in pressure at the ear(s)? In other words, when a tree fell in a
prehistoric forest before animals evolved ears, did it make a “sound” or not? In this
manual I use the second definition. It even seems possible that some form of genetic
engineering may enable us to develop better ears (and brains to perceive the results, plus
ways of storing and reproducing nerve pulses from the ears) in future. But the objective
nature of sound pressures is what a sound archivist can (and must) preserve at present.
The arrangement of the book is as follows. After two chapters on overall copying
strategy and the conversion of analogue sound to digital, we have five chapters on the
techniques for getting the sound accurately from various analogue media. Each has some
history and some scientific facts, and shows how we may use this knowledge to help us
get back to the original sound today. The section on setting playing speeds, for example,
covers both objective and subjective techniques, and contains a summary of our objective
knowledge for reference purposes.
Next come three chapters for special techniques to ensure an old recording is heard
the way the engineers originally intended. One deals with noise reduction systems, the
second where spatial effects occur (e.g. stereo), and the third where the original dynamic
range of a compressed recording might be recovered. These are all problems of
reproduction, rather than of recording.
There are vast areas where we do not have objective knowledge, and we must rely
upon future developments or discoveries. So I have left the discussion of acoustic
recording techniques until this point. (I define “acoustic recordings” as “sound recordings
made without the assistance of electronic amplification”). Besides our shortage of
objective knowledge, the whole subject is much more complicated, and we must use
previous vocabulary to express what little we know. Although large amounts of research
are under way as I write, I can only indicate what I consider to be an appropriate strategy
for a sound archive, which will give better fidelity for listeners until ideal technology
becomes available.

3
In many other cases, we are comparatively close to our goal of restoring the
original sound (or the original intended sound). We now have techniques former
engineers could never have dreamt of, and even acoustic recordings will probably
succumb to such progress. This brings three new difficulties.
In a recent broadcast (“Sunday Feature: Settling the Score” BBC Radio 3, Sunday
4th July 1999), the presenter Samuel West described how he had taken a (modern)
performance of a solo piano piece, and deliberately distorted it until it sounded like an
acoustic recording. He then played the two versions to a number of musically literate
listeners. Not only were all the listeners fooled, but they put quite different artistic
interpretations on their two responses, even though the actual performance was the
same. This seems to show modern day listeners may have quite different artistic responses
to historic records, if progress in sound restoration continues!
However, the programme then went on to outline the second of the three
difficulties - the compromises forced upon performers by obsolete recording techniques.
So my last chapter is an account of some techniques of former sound recording engineers,
so you may judge how they balanced scientific and aesthetic thought processes, and
understand some of the differences between the original sound and the intended original
sound.
The third of the three difficulties is that sound recording is now becoming
subservient to other media, because it is comparatively cheap and easy, and less of a self-
contained discipline. Thus audio is becoming centred on applications, rather than
technologies. All this means my final chapter will go out of date much faster than the rest
of the book.

1.2 The target audience for this manual

Considerable research is needed to understand, and therefore to compensate for, the


accidental distortions of old sound recording systems. This manual consists largely of the
fruits of my research. Although the research has been extensive (and quite a lot of it is
published for the first time), I do not intend this manual to fill the role of a formal
historical paper. It is aimed at the operator with his hands on the controls, not the
qualified engineer at his drawing board, nor the artist in his garret, nor the academic
student of history.
So I have tried to describe technical matters in words, rather than using circuit
diagrams or mathematical formulae. Some professional engineers will (correctly) criticise
me for oversimplifying or using “subjective language.” To these people, I plead guilty; but
I hope the actual results of what I say will not be in error.
I have standardised upon some vocabulary which might otherwise cause confusion.
In particular, what do we call the people who operated the original sound recording
equipment, and what do we call the people restoring the sound today? I have adopted an
arbitrary policy of calling the former “engineers” and the latter “operators.” I apologise if
this upsets or misrepresents anyone, but there is no universally accepted phraseology. The
English language also lacks suitable pronouns for a single person who may be of either
sex, so again I apologise if male pronouns suggest I am ignoring the ladies.
This manual includes a certain amount of sound recording history. I shall have to
look at some aspects in great detail, because they have not been considered elsewhere;
but to reduce unnecessary explanation, I have had to assume a certain level of historical
knowledge on the part of the reader. For analogue media, that level is given by Roland

4
Gelatt’s book The Fabulous Phonograph in its 1977 edition. If you want to get involved in
reproducing the original sound and your knowledge of sound recording history isn’t up to
it, I strongly recommend you to digest that book first.
Psychoacoustics plays a large part, because recording engineers have intuitively
used psychoacoustic tricks in their work. They have always been much easier to do than
to describe, so the word “psychoacoustic” appears in most of my chapters! But it is very
difficult to describe the tricks in scientific language. Furthermore, our present day
knowledge is accumulated from long lines of scientific workers following in each other’s
footsteps - there are very few seminal papers in psychoacoustics, and new discoveries
continue to be made. So I have not given references to such research, but I recommend
another book if you’re interested in this aspect: An Introduction to the Psychology of
Hearing by Brian C. Moore. However, you do not have to read that book before this one.
I must also explain that human beings are not born with the ability to hear. They
have to learn it in the first eighteen months of their lives. For example, as they lie
wriggling in their prams, Grandma might shake a rattle to get their attention. At first the
child would not only be ignorant of the sound, but would lack the coordination of his
other senses. Eventually he would turn his head and see the rattle, coordinating sight and
sound to gain an understanding of what rattles are. There are six or seven senses being
coordinated here, the sense of sight (which in this case is three senses combining to
provide stereoscopic vision - the sense of left eye versus right eye, the sense of parallax,
and the sense of the irises “pulling focus”), the sense of hearing (which is stereophonic,
combining the difference in times and in amplitudes at the two ears), and the sense of
balance and how this changes as the muscles of the neck operate. All this has to be learnt.
Individual people learn in slightly different ways, and if an individual is defective in some
physiological sense, psychological compensation may occur.
All this combines to make the sense of hearing remarkably complex. It is therefore
even more amazing that, in the first 100 years of sound recording history, it was possible
to fool the brain into thinking a sound recording was the real thing - and to a higher
standard than any of the other senses.
A further difficulty I face is that of the reader’s historical expertise. An expert can
take one look at a disc record and immediately pronounce upon its age, rarity, what it will
sound like, the surname of the recording engineer’s mother-in-law, etc. An expert will be
able to recognise the characteristics of a record just by looking at it. Much of my material
will seem redundant to experts. The restoration operators employed by a single record
company also do not need such detail, since they will be specialising in recordings whose
characteristics are largely constant. But there are innumerable examples of operators
getting it wrong when stepping beyond the areas they know. So I consider it important
for every operator to read the book at least once, just to see how things may differ
elsewhere.
The most difficult part is to know which technology was used for making a
particular recording. This is practically impossible to teach. A recipe book approach with
dates and numbers is easy to misunderstand, while the true expert relies on the “look and
feel” of a particular artefact which is impossible to describe in words. I just hope that
experts will not be upset by apparent trivia; but I have made a deliberate attempt to
include such details if there is no convenient alternative. I must also confess that the
archivist in me wants to get unwritten facts into print while it is still possible.
Yet another problem is caused by the frequent changes in hardware preferred by
sound operators. So I shall not give recipe book instructions like “Use a Shure M44
cartridge for playing 78s,” except when there are no alternatives. Instead I shall describe

5
the principles, and leave operators to implement them with the resources they have
available. It is my opinion that transferring recordings made during the last century will
continue for at least the next century. No-one can predict how the hardware will evolve in
that time; but I am reasonably certain the principles will not change.
Past history also shows that you can sometimes have a “low-tech” and a “high-
tech” solution to the same problem. Do not assume the “high-tech” solution will always
give the best results. Practical skills - in handling and listening to old media - often
outweigh the best that modern technology can offer. I can even think of some disgraceful
cases where professional sound restoration operators have been thrown out of trade
associations or engineering societies, because they favour “low-tech” solutions. Do not
allow yourself to be corrupted by such ideas. There is always one optimum solution to a
technical problem, and it is up to you to choose that solution. I cannot always teach you
the craftsmanship aspects by means of the printed word; but I can, and do, explain the
principles of recovering sound with optimum engineering quality. This will allow you to
assess your own skills, and balance them against those of others, for yourself.
I cannot consider every sound medium which has ever been invented. I have
therefore concentrated upon “mainstream” media, or media which illustrate a particular
technical point I wish to make. More advanced or primitive types of sound recordings
have been ignored, since when you know the basic principles, you will be able to figure
out how to transfer them yourself. So you will not find the first stereo experiments
mentioned, or early magnetic recordings, or freakish cylinders, or media never intended to
have a long shelf life (such as those for dictation machines).
Finally, I shall only be describing the techniques I consider essential for someone
trying to restore the original intended sound. There are many others. People have said to
me “Why don’t you mention such and such?” It is usually because I disapprove of “such
and such” on principle. I shall be placing great emphasis upon ways giving the most
accurate results, and I shall simply ignore the rest - otherwise the length of the book
would be doubled.

1.3 The original sound

I had been a professional sound engineer for a quarter of a century before I joined the
British Library Sound Archive. I was the first such person to be appointed as a
“Conservation Manager,” in overall charge of sound conservation strategy. And I must
make it clear that I was regarded as an outsider by the library community. (“A sound
expert? Working in a Library!?”)
As soon as I had cleared away the short ends of quarter-inch tape which stopped
me putting my feet under my desk, I soon became aware that the “culture” of the sound
engineer was not appreciated within the building. So I ask your forgiveness if this manual
appears to say “You must do this” or “You should do that” with no rational explanation
being given. The culture of a successful analogue sound operator appears to be learnt at
his mother’s knee. It isn’t always clear whether he’s generated the rules himself, learnt
them from his peers, or had it hammered into him on a formal training course. To
illustrate my meaning, the very first task I witnessed my staff doing was a straight copy of
a quarter-inch analogue tape. The operator never even looked at the stroboscope of the
playback deck to check if it was running at the right speed! However, he did go through a
prescribed procedure for checking the performance of the destination machine with tone
signals; but he obviously did not understand why this was being done, he only knew who

6
to ask if he got the wrong figures on the meter. All this would be second nature to a
professional sound operator from his mother’s knee. So I apologise again if I keep stating
what I consider to be obvious.
A broader problem is this. I am the first to recognise that “restoring the original
intended sound” may not be the motivation for all transfer operators. The success of
Robert Parker in making old recordings accessible to the modern public is proof of that,
and he has been followed by numerous other workers bringing various corners of the
recorded repertoire out of obscurity. Parker has been the subject of criticism for imposing
artificial reverberation and fake stereo effects upon the original transfers. He replies that
without these techniques the music would not have had such a wide appeal, and anyway
he has lodged the untreated tapes in his vault. I think this is the right attitude. Even if
commercial release is anticipated, I consider that recovering the “original sound” should
always be the first step, whatever happens to it afterwards. Describing subjective
improvements would again double the length of this manual and cause it to go out of
date very rapidly (partly because of changes in fashion, and partly because of new
technical processes). But I hope my remarks will also be helpful when exploitation is the
driving force, rather than preservation.
Since older media often distorted the sound, it is first necessary to decide whether
we should attempt to restore the sound in an undistorted form. It is often argued that the
existing media should be transferred as they are, warts and all, on the grounds that better
restoration technology may be available in the future. Another argument says that such
warts are part of the ambiance in which such media were appreciated in the past, and
should be preserved as a significant part of the artefact.
Having been a professional recording engineer myself, I challenge these views. I
should not wish that the sound recordings I made before I joined the British Library Sound
Archive should be reproduced “warts and all”. I should certainly demand that the ravages
of time, and the undocumented but deliberate distortions (the “recording
characteristics”), should always be compensated, because listeners will then get my
original intended sound. So I consider it’s my responsibility to perform similar services for
my predecessors. As for attempts to tidy up my work in ways which weren’t possible
when I made the recordings, I hold the view that where the warts are accidental (as
opposed to deliberate distortions, such as might be applied to a guitar within a pop music
balance), I have no objection to their being corrected, so long as the corrections result in
more faithful intended sound.
I shall now respond to the assumption of the “warts and all brigade” that future
technology will be better than ours. Frankly, I am not fully convinced by this argument,
because with a scientific approach we can usually quantify the effects of technology, and
decide whether or not future technology can offer any improvement. I only fear
technology when it doesn’t exist at all, or when it exists in the form of “trade secrets”
which I cannot judge. (I shall be indicating these cases as we come to them). Rather, I fear
that sound recording will become more and more “idiot-proof,” and eventually we shall
forget the relationships between past artists and engineers. If we misunderstand this
relationship, we are likely to misunderstand the way the recording equipment was used,
and we will be unable to reproduce the sounds correctly, even with perfect technology. I
shall illustrate the point with the same example I mentioned above. Enjoying popular
music when I was young, I generally know which distortions were deliberate - the guitar
in the pop mix - and I know which were accidental; but I must not assume these points
will always be appreciated in the future. Indeed, I strongly suspect that the passage of
time will make it more difficult for future operators to appreciate what is now subliminal

7
for us. But few people appreciate these “cultural” factors. They have never been written
down; but they’re there, so I shall be making some references to them in the final chapter.
I shall, however, mention one now. Recording sound to accompany pictures is a
completely different business from recording sound on its own. I have spent much of my
life as a film and video dubbing mixer, and I cannot think of a single case where it would
be justifiable to take any of my final mixes and “restore the original sound,” even if it
were possible. I would only want people to go as far as indicated above - to undo the
ravages of time and equalise the known recording characteristics. All the rest of the
“distortions” are deliberate - to distract from compromises made during the picture
shooting process, to steer the emotional direction of the film by the addition of music
and/or the pace of the mixing, to deliberately drive the dynamics of the sound to fit
imperfect pictures, etc. In these circumstances pictures are dominant while sound is
subservient - the sound only helps to convey a film’s message. (Films of musical
performances seem to be the principal exception).
Most people find their visual sense is stronger than their aural sense, even though
sound recording has achieved a higher degree of “fidelity” than moving pictures. Thus
films and videos become art-forms with rules of their own, built into them at the time
they went through “post-production.” When we do want to restore the “original sound,”
rather than the original intended sound, we should clearly divorce the sound from the
pictures, and use “rushes” or other raw material in an unmixed state rather than the final
mix.
Finally, I should like to mention that some workers have argued that old recordings
should be played on old equipment, so we would hear them the way contemporary
engineers intended. I have a certain amount of sympathy with this view, although it does
not agree with my own opinion. I would prefer my recordings to be played on state-of-
the-art equipment, not what I had thirty years ago! But if we wish to pursue this avenue,
we meet other difficulties. The principal one is that we have very few accounts of the
hardware actually used by contemporary engineers, so we don’t actually know what is
“right” for the way they worked.
Even if we did have this knowledge, we would have to maintain the preserved
equipment to contemporary standards. There was a great deal of craftsmanship and taste
involved in this, which cannot be maintained by recipe book methods. Next we would
need an enormous collection of such equipment, possibly one piece for every half decade
and every format, to satisfy any legitimate historical demand for sound the way the
original workers heard it. And we would inevitably cause a lot of wear and tear to our
collection of original recordings, as we do not have satisfactory ways of making modern
replicas of original records.
But it so happens that we can have our cake and eat it. If we transfer the sound
electrically using precise objective techniques, we can recreate the sound of that record
being played on any reproducing machine at a subsequent date. For example, we could
drive its amplifier from our replayed copy, its soundbox from a motional feedback
transducer, or its aerial from an RF modulator.

1.4 Operational principles

I shall first state what I believe to be an extremely important principle. I believe the
goal of the present-day restoration expert should be to compensate for the known
deficiencies objectively. He should not start by playing the recording and twiddling the

8
knobs subjectively. He should have the courtesy first to reproduce the sound with all
known objective parameters compensated. For archival purposes, this could be the end of
the matter; but it may happen that some minor deficiencies remain which were not
apparent (or curable) to contemporary engineers, and these can next be undone. In any
event, I personally think that only when the known objective parameters have been
compensated does anyone have the moral right to fiddle subjectively - whether in an
archive, or for exploitation.
The aim of objectivity implies that we should measure what we are doing. In fact,
considerable engineering work may be needed to ensure all the apparatus is performing
to specification. I know this goes against the grain for some people, who take the view
that “the ear should be the final arbiter.” My view is that of course the ear should be the
final arbiter. But, even as a professional recording engineer deeply concerned with artistic
effects, I maintain that measurements should come first. “Understanding comes from
measurement” as physical scientists say; if we can measure something’s wrong, then
clearly it is wrong.
On numerous occasions, history has shown that listeners have perceived
something wrong before the techniques for measuring it were developed; this is bound to
continue. Unfortunately, “golden-eared” listeners are frequently people who are
technically illiterate, unable to describe the problem in terms an engineer would
understand. My personal view (which you are always free to reject if you wish), is that
measurements come first; then proper statistically-based double-blind trials with “golden-
eared” listeners to establish there is a valid basis for complaining about problems; then
only when this has been done can we reasonably research ways to cure the problem. I
certainly do not wish to discourage you from careful listening; but accurate sound
reproduction must at the very least begin with equipment whose performance measures
correctly.
On the other hand, the ear is also important in a rather coarse sense - to get us
back on the right track if we are catastrophically wrong. For example, if the tape box label
says the tape runs at 15 inches per second and the tape sounds as if it’s at double speed,
then it will probably be a fault in the documentation, not a fault in our ears!
For intermediate cases, we should be able to justify subjective decisions in objective
terms. For example, if we switch the tape reproducer to 7.5 inches per second and we
perceive music at slightly the wrong pitch, then we should proceed as follows. First we
check our own sense of pitch with a known frequency source properly calibrated. Then
we quantify the error and we seek explanations. (Was it an unreliable tape recorder? or
an historic musical instrument?) If we cannot find an explanation, we then seek
confirmatory evidence. (Is the background hum similarly pitch-shifted? Does the tape play
for the correct duration?) But, at the end of the day, if there is no objective explanation, a
sound archive must transfer the tape so that at least one copy is exactly like the original,
regardless of the evidence of our senses.
The question then arises, which subjective compensations should be done in the
environment of a sound archive? A strictly scientific approach might suggest that no such
compensations should ever be considered. But most professional audio operators are
recruited from a background which includes both the arts and the sciences. It is my
personal belief that this is only to the good, because if these elements are correctly
balanced, one doesn’t dominate over the other. But it is impossible for anyone’s artistic
expertise to stretch across the whole range of recorded sound. It may be necessary to
restrict the artistic involvement of an operator, depending upon the breadth of his

9
knowledge. To be brutal about it, a pop expert may know about the deliberately distorted
guitar, whereas an expert in languages may not.
This assertion has not answered the potential criticism that the artistic input should
ideally be zero. I shall counter that argument by way of an example. For the past thirty
years Britain has had an expert in restoring jazz and certain types of dance music in the
person of John R. T. Davies. I do not think he will mind if I say that his scientific
knowledge is not particularly great; but as he played in the Temperance Seven, and has
collected old records of his own particular genre ever since he was a boy, his knowledge
of what such music should sound like has been his greatest asset. Long before the present
scientific knowledge came to be formulated, he had deduced it by ear. He therefore
systematically acquired the hardware he needed to eliminate the various distortions he
perceived, and although the methods he evolved appear a little odd to someone like me
with a scientific background, he ends up with very accurate results. John Davies does not
claim to be perfect, but he holds the position that his particular musical knowledge
prevents him making silly mistakes of the type which might befall a zombie operator.
I therefore accept that a certain level of artistic input is advantageous, if only to
insure against some of the howlers of a zombie operator. But my position on the matter is
that each individual must recognise his own limited knowledge, and never to go beyond
it. We shall be encountering cases in the following pages where specialist artistic
knowledge is vital. When such knowledge comes from an artistic expert, I consider it is no
less reliable than pure scientific knowledge; but I would feel entitled to query it if I wasn’t
certain the knowledge was “right”. Such knowledge may not just be about the original
performance. It may also be knowledge of the relationship between the artist and the
contemporary recording engineer. It is not always realised that the sounds may have been
modified as they were being created, for very good reasons at the time.

1.5 A quadruple conservation strategy

The contradictions which can arise between “technical” and cultural” factors have caused
passionate debate within the British Library Sound Archive. Indeed, the writer once
addressed an external public meeting on the subject, and the audience nearly came to
blows. There is often no satisfactory compromise which can be taken between the
contradictions. I propose to leave some of the considerations until we get to the final
chapter; but in the meantime the best answer seems to be a “quadruple conservation”
strategy. This would mean the archive might end up with as many as four versions of a
recording for long term storage, although two or more might often be combined into one.

(1) The original, kept for as long as it lasts.


(2) A copy with warts-and-all, sounding as much like the original artefact as possible,
which I shall call “The Archive Copy.”
(3) A copy with all known objective parameters compensated, which I shall call “The
Objective Copy.”
(4) A copy with all known subjective and cultural parameters compensated, which I shall
call “The Service Copy.”

I recognise that such an ambitious policy may not always be possible. As we reach
critical decision points during the course of this manual, I shall be giving my personal
recommendations; but I am writing from the point of view of a professional sound

10
recordist in a publicly funded national archive. Each reader must make his own decision
for himself (or his employer) once he understands the issues.
Fortunately, there are also many occasions where, even from an ivory tower
viewpoint, we don’t actually need an original plus three copies. For example, when basic
engineering principles tell us we have recovered the sound as well as we can, the
objective and service copies might well be identical. Sometimes we do not have the
knowledge to do an “objective copy”, and sometimes cultural pressures are so intense
that we might never do an “archive copy.” (I shall describe an example of the latter in
section 13.2). But I shall assume the quadruple conservation strategy lies behind our
various attempts, and I advise my readers to remember these names as the work
proceeds.

1.6 How to achieve objectivity

Given that the purpose of conservation copying is to “restore the original intended
sound”, how do we go about this? How can we know we are doing the job with as much
objectivity as possible, especially with older media made with temperamental recording
machinery, or before the days of international standards? The present-day archivist has
the following sources of knowledge to help him, which I list in approximate order of
importance.

1. Contemporary “objective recordings.” This generally means contemporary frequency


discs or tapes, together with a small handful of other engineering test media for
intermodulation distortion and speed. Many large manufacturers made test recordings, if
only for testing the reproducers they made, and provided they have unambiguous written
documentation, we can use them to calibrate modern reproducers to give the correct
result. Unfortunately, not all test records are unambiguously documented, and I shall
allude to such cases as they arise. Even worse, many manufacturers did not make test
recordings. Yet objective measurements are sometimes available accidentally. For
instance, several workers have analysed the “white noise” of the swarf vacuum pipe near
an acoustic recording diaphragm to establish the diaphragm’s resonant frequency. And
we will be coming across another rather curious example in section 6.47.
2. Contemporary, or near-contemporary, objective measurements of the recording
equipment. Written accounts of contemporary measurements are preferable to present-
day measurements, because it is more likely the machinery was set up using
contemporary methods of alignment, undecayed rubber, new valves, magnetic materials
at the appropriate strength, etc. As for the measuring equipment, there are practically no
cases where present-day test gear would give significantly different results from
contemporary measuring equipment. As the machinery improved, so did measuring
equipment; so contemporary measurements will always be of the right order of
magnitude.
On the other hand, alleged objective measurements made by the equipment
makers themselves should always be subject to deep suspicion (this applies particularly to
microphones). This last point reminds us that there may be “specifications” for old
recording equipment, besides “objective measurements.” These must be regarded with
even more suspicion; but when there were international standards for sound recordings,
we must at least examine how well the equipment actually conformed to the standards.

11
3. Present-day analysis of surviving equipment. This suffers from the disadvantages hinted
at above, where the equipment isn’t necessarily in its best state. There are also the
“cultural” factors; the way in which the machinery was used was often very important,
and may invalidate any scientific results achieved.
4. Analysis of drawings, patents, etc. It is often possible to work out the performance of a
piece of recording equipment from documentary evidence; no actual artefact or replica is
required.
5. Interviews with the engineers concerned. Reminiscences of the people who operated
old recording equipment often reveal objective information (and sometimes trade secrets).
6. Reverse engineering surviving recordings. In general, this is not possible for one
recording alone; but if a number of different recordings made by the same organisation
exhibit similar characteristics, it is possible to assume they are characteristics of the
machine which made them, rather than of the performances. It is therefore possible to
reverse engineer the machine and neutralise the characteristics. A similar technique occurs
when we have a recording which is demonstrably a copy of another. We can then use the
original to deduce the characteristics of the copy, and thence other recordings made on
that equipment.
7. Automatic analysis. It is early days yet, but I mention this because mathematical
analysis of digital transfers is being invoked to identify resonances in the original. One aim
is to eliminate the “tinnyness” of acoustic horn recording. Identifying resonances by such
an objective technique is clearly superior to the subjective approach of the next
suggestion.
8. Intelligent artistic input. If it’s known that a type of recording had resonances, it may be
possible to listen out for them and neutralise them by ear. This implies there is a “general
structure” to the recorded characteristics, which can be neutralised by appropriate
“tuning.” So in the following pages I have included a few such “general structures.” But
I’ve placed this evidence last, because it’s very easy to cross the borderline between
objective and subjective compensation. Although the result may be more faithful than no
compensation at all, there will be no proof that the particular form of tuning is exactly
correct, and this may give difficulties to subsequent generations who inherit our copies. As
digital signal processing evolves, it will be feasible to do statistical analysis to determine a
“level of confidence” in the results. Then, at some point (which would need a consensus
among archivists), the process might even be adopted for the “objective copy”.

1.7 The necessity for documentation

If we continue to assume that “recovering the intended original sound” is a vital stage of
the process, this implies that we should always keep some documentation of what we
have done. It might take the form of a written “recording report” or a spoken
announcement. It has three functions. It advises our successors of any subjective elements
in our work, enabling them to reverse engineer it if the need arises after the original has
decayed away. It also shows our successors the steps we have taken ourselves and our
thought processes, so later generations will not be tempted to undo or repeat our work
without good reason. I must also say that I personally find the documentation useful for a
third reason, although not everyone will agree with me. I find it forces me to think each
case through logically.

12
2 The overall copying strategy

2.1 The problem to be solved

In this manual I do not propose to discuss the major strategies of running a sound archive;
instead, I shall refer you to a book by my mentor Alan Ward (A Manual of Sound Archive
Administration, pub. Gower, 1990). But this chapter includes wider issues than just
analogue sound reproduction and copying.
Some philosophers have considered the possibility of a replicating machine which
might build an exact replica of an original recording, atom by atom. This is science fiction
at present, so the only other way is to play such a recording back and re-record it.
But even if we could build such a replicating machine, I suspect that the universe
may contain something more fundamental even than sub-atomic particles. Here is a
rhetorical question for you to ponder: What is “Information”?
It may even be what holds the Universe together! When certain sub-atomic
particles separate under the laws of Quantum Physics, they may be connected by
“information” which travels even faster than light, but which does not actually travel until
you make the observation. This is still a novel concept amongst the scientific community
as I write (Ref. 1); but within a few decades I suspect it will be as familiar to
schoolchildren as “Relativity” is now. And, since sound recording is by definition a way of
storing “information,” such philosophical issues aren’t completely irrelevant to us.

2.2 General issues

Most of this manual is designed to facilitate the playback process so as to recover the
information - the sound - without any intentional or unintentional distortions. It is aimed
at the operator whose hands are on the controls, rather than the manager planning the
overall strategy. For the latter, politics, cost, space and time are paramount; he is less
concerned with “mechanics.” But it would be wrong for me to ignore the operational
aspects of overall strategy, if only because in smaller archives the manager and the
operator is the same person; so I shall now say a few words on the subject.
First, the law of copyright. This differs from one country to the next, and may also
have exemptions for archival applications. For many years the British Library Sound
Archive had special permission from The British Phonographic Industry Ltd. to make
copies of records for internal purposes, since published records had no “fair dealing”
exemptions. Under procedures laid down under the 1988 Copyright Act, archival copying
work might always then be possible provided the Secretary for State was persuaded that
the archive was “not conducted principally for profit”; but I must stress that, whatever I
recommend, it does not absolve you from observing the law of copyright in your country.
The manager will certainly be concerned with cost, perhaps thinking of getting the
maximum amount of work done for a particular budget. Frankly, I believe this is
inappropriate for an archive dedicated to conserving sounds for centuries, but I recognise
this will be a consideration in the commercial world. A manager must therefore
understand the principles, so he may see clearly how the work will suffer if the ideal
scenario is not followed. It may not be a catastrophe if it isn’t, but there will be trade-offs.
The procedure actually used should certainly be documented, and then originals should be
kept so that future generations can have another bite at the cherry. So the manager must

13
assess the costs of storing the originals and then financing another bite of the cherry,
comparing them with the costs of the ideal scenario.
I am afraid that experience also shows that “unexpected hitches” are frequent. It is
usually impossible to copy sounds using production-line techniques. Whatever overall
strategy you adopt, your schedule is certain to be disrupted sooner or later by a recording
which requires many times the man-hours of apparently-similar items.

2.3 The principle of the “Power-Bandwidth Product”

As I said, the only way of conserving sounds which are at risk is to copy them. “At risk”
can mean theft, wilful destruction, accidental erasure, biological attack, miscataloguing, or
wear-and-tear, as well as plain chemical breakdown. But if it’s considered there is little
chance of these, then there is much to be said for simply keeping the original recording
uncopied, for the following reason.
Analogue recordings cannot be copied without some loss of quality, or
“information” as engineers call it. Despite the idea of “information” being a fundamental
property of matter, to an analogue engineer “information” is an objective measurement
of the quality of a recording. It is obtained by multiplying the frequency range, by the
number of decibels between the power of the loudest undistorted signal and the power of
the background noise. The result is the “power-bandwidth product.” This term is used by
analogue engineers to measure the information-carrying capacity of such things as
transformers, landlines, and satellites, besides sound recordings.
It is always possible to trade one parameter against the other. To return to sound
recording, a hissy disc may have a full frequency range to the limits of human hearing (say
16kHz), but if we apply a high-frequency filter when we play it, the hiss is reduced. In
fact, if the filter is set to 8kHz, so that the frequency range is halved, the hiss will also be
halved in power. We can therefore trade frequency range against background noise. Of
course, there may be other parameters which are not covered by the power-bandwidth
formula - such as speed constancy - but because it’s a fundamental limitation, we must
always consider it first. It is true that in Chapter 3 we may learn about a potential process
for breaching the background-noise barrier without touching the wanted sound; but that
process is not yet available, and in any case we must always consider the matter
“downstream” of us. In objective terms, there’s no way round it. (For further details, see
Box 2.3).
The first strategic point about copying analogue sound recordings is therefore to
minimise the loss of power-bandwidth product caused by the copying process. If the hissy
disc mentioned above were copied to another hissy disc with the same performance, the
hiss would be doubled, and we would irrevocably lose half the power-bandwidth product
of the original. An archive should therefore copy analogue recordings to another medium
which has a much greater power-bandwidth product, to minimise the inherent losses.

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BOX 2.3
THE POWER-BANDWIDTH PRODUCT IN AUDIO RECORDINGS

This box is aimed at engineers. It is relatively easy to assess the information-


carrying capacity of analogue devices such as transformers, landlines, and
satellites. They tend to have flat responses within the passband, Gaussian noise
characteristics, and clear overload points. But sound recordings generally do
not have these features, so I must explain how we might quantify the power-
bandwidth product of sound recordings.
Analogue media overload “gently” - the distortion gradually gets worse as
the signal volume increases. So we must make an arbitrary definition of
“overload.” In professional analogue audio circles, two percent total harmonic
distortion was generally assumed. As this is realistic for most of the analogue
media we shall be considering, I propose to stick to this.
For electronic devices, the bandwidth is conventionally assumed to be the
points where the frequency response has fallen to half-power. This is distinctly
misleading for sound recordings, which often have very uneven responses; the
unevenness frequently exceeds a factor of two. There is another complication
as well (see section 2.4). For my purposes, I propose to alter my definition of
“bandwidth” to mean the point at which the signal is equal to random
(Gaussian) noise - a much wider definition. Yet this is not unrealistic, because
random noise is in principle unpredictable, so we can never neutralise it. We
can only circumvent it by relying upon psychoacoustics or, for particular
combinations of circumstances (as we shall see in Chapter 3). Thus random
noise tends to form the baseline beyond which we cannot go without
introducing subjectivism, so this definition has the advantage that it also forms
the limit to what is objectively possible.
But most recording media do not have Gaussian noise characteristics. After
we have eliminated the predictable components of noise, even their random
noise varies with frequency in a non-Gaussian way. We must perform a
spectral analysis of the medium to quantify how the noise varies with
frequency. And because we can (in principle) equalise frequency-response
errors (causing an analogous alteration to the noise spectrum), the difference
between the recorded frequency-response and the noise-spectrum is what we
should measure.
The human ear’s perception of both frequencies and sound-power is a
“logarithmic” one. Thus, every time a frequency is doubled, the interval sounds
the same (an “octave”), and every time the sound power increases by three
decibels the subjective effect of the increase is also very similar to other three-
decibel increases. Following the way analogue sound engineers work, my
assessment of the power-bandwidth product of an analogue sound recording is
therefore to plot the frequency response at the 2% harmonic-distortion level,
and the noise spectrum, on a log-log graph; and measure the AREA between
the two curves. The bigger the area, the more information the recording holds.

15
2.4 Restricting the bandwidth

With an older record, we may be tempted to say “There’s nothing above 8 kiloHertz, so
we can copy it with the hiss filtered off without losing any of the wanted sound, and
make it more pleasant to listen to at the same time.” This is a very common attitude, and
I want to take some space to demolish the idea, because it is definitely wrong for archival
copying, although it might be justifiable for exploitation work.
The first point is that if it ever becomes possible to restore the frequencies above
8kHz somehow, three considerations will make it more difficult for our successors. First,
the copy will add its own hiss above 8kHz, perhaps much fainter than that of the original;
but when the original high frequencies are further attenuated by the filter, the wanted
signal will be drowned more efficiently. Secondly, by making the high frequencies weaker,
we shall make it much more difficult for our successors to assess and pursue what little
there is. Thirdly, we actually have means for restoring some of the missing frequencies
now - imperfectly and subjectively, it is true; but to eliminate such sounds with filters is an
act of destruction exactly analogous to theft, erasure, or wear-and-tear.
The second point is, how do we “know” the “fact” that there is “nothing” above
8 kiloHertz? Actually, there is no known method for cutting all sounds off above (or
below) a fixed frequency. Whether the effect is done acoustically, mechanically, or
electronically, all such systems have a slope in their frequency responses. A disc-recording
cutterhead, for example, may work up to 8kHz, and above that its response will slope
away at twelve decibels per octave, so that at 16kHz the cutter will be recording twelve
decibels less efficiently. So it is never true to say there’s nothing above 8kHz. In a well-
matched system, the performance of the microphone, the amplifier, and the cutterhead
will be very similar, so the overall result might be a slope of as much as 36 decibels per
octave; but this hardly ever seems to happen. Certainly, experiments have shown that
there is audible information above the “official” limit. Often it is highly distorted and
difficult to amplify without blowing up the loudspeaker with hiss, but it’s there all right.
The question then becomes “how do we go about making the high frequencies more
audible”, rather than “where do we cut them off.”
I regret having to labour this point, but a very respected digital sound expert once
fell into this trap. He did a computer analysis of the energy of an acoustic recording at
different frequencies, and observed that noise dominated above 4kHz, so ruthlessly cut
those frequencies off. In his published paper he devoted some puzzled paragraphs to why
experienced listeners found the resulting recordings muffled. The reason, of course, is that
(using psychoacoustics) human beings can hear many sounds when they are twenty or
thirty decibels fainter than noise.
The only possible justification for filtering is if the subsequent recording medium is
about to overload. Fortunately digital media are relatively immune from such problems,
but it is a perpetual problem with analogue copy media.
In practice, this recovery of sound above an official cut-off frequency is a
technique in its infancy. We can do it, but as Peter Eckersley is reported to have said, “the
wider you open a window, the more muck blows in.” Practical “muck” comprises both
background-noise and distortion-products. The techniques for removing these are in their
infancy. Unless such sounds can be restored perfectly, it is probably better that we should
not try. But it is equally wrong for us to throw them away. The logical compromise is to
transfer the high frequencies “flat” on the “archive copy,” so future researchers will have

16
the raw material to work on. The copy medium must therefore have suitable power-
bandwidth characteristics so that it will not alter the noise or distortion of the original
medium. From the present state-of-the-art, we suspect that harmonic-distortion removal
will depend critically upon phase linearity; therefore the copy medium must not introduce
phase distortion either, or if it does it must be documented somehow (e. g. by recording
its impulse-response - see section 3.4).

2.5 Deciding priorities

Our strategy is therefore to copy the vulnerable recording to another medium which has
ample power-bandwidth product so that we don’t lose very much, and not to filter the
recording. Unfortunately, all media have a finite power-bandwidth product, so in fact we
shall always lose something. The strategy must therefore balance the inevitable losses
against the financial costs of making the copy and the likelihood of the original surviving
to another day. This adds another dimension to the equation, because when analogue
media degrade, their power-bandwidth product suffers (it’s usually because their
background noise goes up). So, one must decide when to do one’s copying programme,
depending on the power-bandwidth product of the original, its likely power-bandwidth
product after future years in storage, the ability to recover power-bandwidth product at
any particular point in time, and the power-bandwidth capacity of the copy medium.
Clearly we must always give top priority to media whose power-bandwidth
product seems to be degrading faster than we can reproduce it. At the British Library this
means wax cylinders and cellulose nitrate discs. (Acetate tapes are also vulnerable because
the base-material is getting more brittle, but this does not directly affect the power-
bandwidth product). There is a race against time to save these media, and the matter is
not helped by two further circumstances. These media tend to be less-well documented,
and it is impossible to play them without risk of damage; so the sounds must be copied
before anyone can make an informed judgement on whether it is worth copying them !
Other media are less vulnerable, so we can afford to make a considered judgement about
when to start copying them, and the balance will tilt as our knowledge improves. Also it is
quite possible (although, in this digital age, less likely) that the technology for obtaining
the maximum power-bandwidth product will improve. I am not going to talk about the
present state-of-the-art here, because any such discussion will quickly go out of date; but
I believe the basic principles will not change.

2.6 Getting the best original power-bandwidth product

The power-bandwidth product of an analogue recording always suffers if it is copied, so


we must ensure we are working with “an original”, not a copy. Thus we need to know
the provenance of the analogue record. Does an earlier generation exist elsewhere? Does
a manufacturer have master-tapes or metal negatives in his vault? Do masters exist of
copies donated to your archive? We find ourselves picking up political hot potatoes when
we examine this aspect, but the issue must be faced.
A knowledge of sound recording history is vital here, or at least that part of sound
recording history which has a bearing upon your particular archive. The ideal strategy
would be to collect full lists of the holdings of originals in various collections; but an
adequate substitute might take the form of a generalised statement. At the British Library

17
Sound Archive, we have an interest in commercial records made by Britain’s leading
manufacturer EMI Records Ltd. It is useful for us to know that: “The British EMI factory
has disposed of the metalwork for all recordings of black-label status or below, which
were deleted by the outbreak of the Second World War.” This sentence shows us the
recordings we must seek and process in order to complete the collection for the nation. I
advise you to collect similar statements to describe the genres you are interested in.
The strategy will also be determined by which media have the greatest power-
bandwidth product, not just their mere existence. Although the metalwork mentioned in
the previous paragraph is amongst the most rugged of all sound recording media, that
situation isn’t always the case. From about 1940 onwards, for example, Columbia Records
in the United States did their mastering on large nitrate discs in anticipation of long-
playing records (“L.P.”s), which they introduced in 1948. These nitrates, if they still
survive today, will be nearing the end of their useful life. Similar considerations apply to
early tape. Thus a properly-planned conservation strategy will also take account of the
lifetime of the “masters.”
The archive must have some idea about the existence or non-existence of such
holdings, because I frankly don’t see the point of wasting time recovering the original
sound from a second or third-generation copy when a version with a better power-
bandwidth product exists somewhere else. The only reason might be to make service
copies for use as long as the earlier generation remains inaccessible, or “partially
objective” copies for reference purposes in the future (I shall explain this idea in section
2.8).
The overall strategy should be planned in such a way that, if a better version turns
up, you can substitute it. There should be minimum disruption despite the obvious
difficulties. Re-cataloguing must be possible to ensure the old version doesn’t get used by
mistake, but documentation of the old one should not be destroyed.
With nearly all analogue media it is easy to establish the order of the generations
by ear - exact provenances aren’t always essential. For example, if an analogue tape is
copied onto a similar machine similarly aligned, the hiss will double, the wow-and-flutter
will double, and the distortion will double. These effects are quite easy to hear so long as
the two tapes are running simultaneously into a changeover switch under the control of
the operator. Difficulties only occur when the two tapes are on opposite sides of the
world and neither owner will allow them to be moved, or one is suspected of being a
copy of the other on a medium with a better power-bandwidth product (but it isn’t
certain which is the original and which is the copy), or the original has disappeared and
you must choose between two different copies of the same generation.
This latter case, two or more copies of an “original,” is not uncommon. If the
original has disappeared, it behoves us to choose the copy with the maximum power-
bandwidth product. To put it more simply, if there are two copies available, we must
choose the better one. It seems almost self-evident; but it’s a principle which is often
ignored.
A further dimension is that it may be possible to combine two copies to get an
even better power-bandwidth product than either of them alone, and we shall be looking
at this in Chapter 3. There may be political and practical difficulties; but every effort
should be put into securing several good copies before the copying session starts.
Copies manufactured in other countries may often be better quality than locally-
made ones. Meanwhile, recordings made for foreign broadcasters may only survive in
foreign vaults. Thus you may be kept very busy chasing versions in other countries,
usually with different catalogue numbers.

18
All this means that someone qualified to do discographical work may be kept just
as busy as the actual sound operator. The two should work in close collaboration for
another reason as well. Often technical factors depend on the date of the recording, or its
publication-date; so the operator (or manager) may need this information before work
starts.

2.7 Archive, objective, and service copies

With these considerations in mind, it now seems appropriate to address the issue of the
versions we wish to make. (We considered the “three possible copies” in section 1.5)
Until now, most copying has been “demand-led” - the demand from listeners and
customers dictates what gets copied. While this is all right so far as it goes, the result is
usually that only “service copies” are achieved, because copies are tailored to listeners’
needs with subjective and cultural factors incorporated.
In my view, a proper programme of archival copying cannot be demand-led for
that reason, and the following as well. The technical standards for service copies can be
less critical, so general standards are lowered; I confess I have been guilty of this myself.
Service copies are often done “against the clock”, when loving care-and-attention is in
short supply. And since the demand always comes from someone familiar with the subject
matter, documentation tends to be less rigorously done.
Thus a programme incorporating several separate copies will take longer as well. It
may be necessary to do three versions and document them. And it is advisable to have a
procedure to prevent the same job being done twice.
On the other hand, there are ways to save time if a proper programme is planned.
Demand-led hopping between different media with different characteristics wastes time
connecting and aligning equipment, and may mean research and experiment if the plan
does not confine itself to known areas. It requires “technical rehearsal time,” which I shall
consider shortly. Thus it is best to allocate at least a full working day specifically to archival
copying without risk of interruption, and during that time a slab of technically-similar
technically-understood work should be tackled.
There are many cases in which the various copy versions may be combined. If a
disc record is so good that modern technology can do nothing to improve the sound, then
the objective and service copies might as well be identical. Many professionally-made
tapes can be copied to fill all three roles.
The overall strategy must always be capable of giving predictable results. If two
different operators do the same job with different equipment, there should be no audible
difference between their two “archive copies” and their two “objective copies”. This
implies that the operators should be supported by technical staff ensuring that all the
equipment operates to international standards. A programme of routine measurement of
equipment is essential, and if a machine is discovered to have been operated in a
misaligned state, all the work done by that machine in the meantime should be checked
and, if necessary, re-done. I shall not impose my ideas of the tolerances needed in such
measurements, as standards are bound to rise with time; but managers must ensure such
checks take place at frequent intervals.
Top-of-the-range copying facilities have high capital costs. These might be diluted
by arranging a shift-system, so the equipment is in constant use. Alternatively, one shift
might be doing exploitation work while another is doing strict archival work and a third is
doing routine maintenance.

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2.8 “Partially objective” copies

Sometimes the maximum power-bandwidth product exists on a source without proper


documentation, so we are not sure if it qualifies as an “objective copy” or not. This quite
often happens when a record manufacturer has had privileged access to metal-parts or
vinyl pressings or master-tapes. He may have used them specifically for a modern reissue,
but given the reissue subjective treatment using undocumented trade secrets, so we
cannot reverse-engineer it to get an objective copy. However, even if we have a poor
“original,” we can use it to see whether the reissue qualifies as an “objective copy” or
not. I call this the “partially objective” copy. Straightforward comparison with a
changeover switch is usually sufficient to determine whether the new version is
“objective.” If the manufacturer hasn’t added irreversible effects, we may even be able to
re-equalise or alter the speed of his version to match the original, and achieve a better
end-result. To assess the re-equalisation objectively, we may need to compare the two
versions with a spectrum analyser or use a Thorn-EMI “Aquaid” System (Ref. 2). All this
underlines the need for rigorous discographical research before the session.
The Power-Bandwidth Principle shows quite unambiguously the advantages of not
copying a recording if we don’t have to. Furthermore, an analogue original will always
contain a certain amount of extra information hidden in it which may become available to
future technologists. It will often be lost if we copy the original, even if we use the best
technology we have. The late talented engineer Michael Gerzon (Ref. 3) claimed that over
99% of the information may be thrown away. Personally I consider this an overestimate;
but I could agree to its being in the order of 25%. The difference may partly be because
we have different definitions of the word “information.” But, either way, Gerzon’s
message agrees with mine - KEEP THE ORIGINALS.
A final point is that it goes without saying that the facilities for cleaning originals,
and otherwise restoring them to a ready-to-play state, must be provided (see Appendix
1).

2.9 Documentation strategy

I will not dwell upon the well-known empirical rule, confirmed upon numerous occasions,
that it takes at least twice as long to document a recording as it does to play it. Note that
I’m only talking about documenting the recorded contents now, not the technical
features!
Personally, I am rather attracted by the idea that there should be a version of the
documentation on the copy itself. This means that as long as the recording survives, so
does the documentation, and the two can never be separated. In olden times this was
achieved by a spoken announcement, and there is much to be said for this technique; as
long as the copy is playable, it can be identified. For really long term purposes I consider
such an announcement should be made by an expert speaker, since questions of
pronunciation will arise in future years.
On the other hand, a spoken announcement isn’t “machine-readable.” With a
digital copy the documentation might be stored as ASCII text, or as a facsimile of a
written document; but as yet I have no practical experience of these techniques. There are

20
(unfortunately) several “standard” proposals for storing such “metadata” in digital form.
And the end of Section 3.7 will warn you of potential problems with this idea.
As we proceed through this manual, we shall see that technical documentation
could also become very complex, and the strategy for your archive will largely depend on
what has gone before. My former employer, the BBC, always had a paper “recording
report” accompanying every radio recording. It was useless without one, because it had
to have the producer’s signature to say it was ready for transmission before the network
engineer would transmit it. But my current employer, the British Library Sound Archive,
does not have such a system. It’s probably too late to introduce it, because the idea of a
recording-report can only work if alarm-bells ring in its absence.
By adding technical documentation, I don’t wish it to take even longer to
document a recording. This is especially important, because a technical report can only be
completed by technical staff, and if both the operators and the equipment are
unproductive while this goes on, it is very expensive. My suggested alternative is to
establish a standard set of procedures, called something simple like “X1”, and simply
write down “Copied to procedure X1” followed by the operator’s signature. (I consider
the signature is important, since only the operator can certify that the copy is a faithful
representation of the original).
This implies that “Procedure X1” is documented somewhere else, and here we
must face the possibility that it may become lost. This actually happened to both my
employers. When I was in the BBC the change from CCIR to IEC tape-recording
equalisation at 19cm/sec took place (section 7.8), and was implemented immediately on
quarter-inch tape; but not immediately on 16mm sepmag film, which happens to run at
the same speed. For the next year I got complaints that my films sounded “woolly” on
transmission, despite rigorous calibration of the equipment and my mixing the sound with
more and more treble. When the truth dawned I was very angry; the problem (and
damage to my reputation) could have been avoided by proper documentation. I was
determined this should not happen when I joined the Sound Archive. Unhappily, a new
director was once appointed who decided there was too much paperwork about, and
scrapped most of it. The result is that, to this day, we do not know how-and-when the
change to IEC equalisation took place at the Archive, so we often cannot do objective
copies.
The two problems of “technical rehearsal” and “time to do the documentation”
might both be solved by a system of rehearsals before the transfers actually take place.
Thus, a working day might consist of the operator and the discographer working together
in a low-tech area to decide on such things as playing-speeds, the best copy or copies to
transfer, and whether alternatives are the same or not. The catalogue-entry can be started
at the same time, and perhaps spoken announcements can be pre-recorded. There is also
time to research anything which proves to need investigation. The actual “high-tech”
transfers could then take place at a later date with much greater efficiency.
The advantages and disadvantages of converting analogue sounds to digital are
the subject of the next chapter. We shall learn that there are some processes which should
always be carried out in the analogue domain - speed-setting, for example - and some
best carried out in the digital domain - various types of noise reduction, for example. Thus
the overall strategy must take the two technologies into account, so that the appropriate
processes happen in the right order. Also digital recordings are often inconvenient for
“service copies.” It may be necessary to put service-copies onto analogue media to make
it easier to find excerpts, or because analogue machinery is more familiar to users.

21
This writer happens to believe that the analogue and digital processes should be
carried out by the same people as far as possible. Up till now, digital signal processing has
been rather expensive, and has tended to be farmed out to bureau services as resources
permit. Not only are there communications problems and unpredictable delays which
inhibit quality-checking, but a vital feedback loop - of trying something and seeing how it
sounds - is broken. The overall strategy should keep the analogue and digital processes as
close together as possible, although the special skills of individuals on one side of the
fence or the other should not be diluted.

2.10 Absolute phase

The next few chapters of this manual will outline the different techniques for copying
sounds, so I shall not deal with them here. But there are three considerations which affect
all the techniques, and this is the only logical place to discuss them.
At various times in history, there have been debates whether a phenomenon called
“absolute phase” is significant. Natural sounds consist of alternating sound pressures and
rarefactions. It is argued that positive pressures should be provided at the listener’s ear
when positive pressures occurred at the original location, and not replaced with
rarefactions. Many experienced listeners claim that when this is done correctly, the
recording sounds much more satisfactory than when the phases are reversed; others claim
they cannot hear any difference. I freely admit I am in the latter category; but I can see
that the advantages of “absolute phase” could well exist for some people, so I should
advise the sound archivist to bear this in mind and ensure all his equipment is fitted with
irreversible connectors and tested to ensure absolute phase is preserved.
Since the earliest days of electrical recording, the effect has been so subtle that
most equipment has been connected in essentially random ways. Furthermore, bi-
directional microphones cannot have “absolute phase,” because the absolute phases of
artists on the two opposite sides of the microphone are inherently dissimilar. But this
doesn’t apply to acoustic recordings. As sound-pressures travelled down the horn, they
resulted in the groove deviating from its path towards the edge of a lateral-cut disc record
and going less deep on a hill-and-dale master-recording (due to the lever mechanisms
between diaphragms and cutters). Thus, we actually know the absolute phase of most
acoustic recordings - those which have never been copied, anyway. I therefore suggest
that the copying strategy for all discs and cylinders should follow the convention that
movements towards the disc edge for a lateral stylus and upwards movements for a hill-
and-dale stylus should result in positive increases in the value of the digital bits. This
won’t mean that absolute phase is preserved in all electrical recordings, because electrical
recording components were connected in random ways; but it will ensure that this aspect
of the originals is preserved on the copies, whether it ever proves to be critical or not.
The English Decca Record Company has recently adopted the standard that
positive-going pressures at the microphone should be represented by positive-going digits
in a digital recording, and this idea is currently under discussion for an AES Standard. It
seems so sensible that I advise everyone else to adopt the same procedure when planning
a new installation. There is also a convention for analogue tape (Ref. 4). But virtually no-
one has used it, and there is no engineering reason why the absolute phase of a tape-
recording should be preserved on a copy when it is only a subjective judgement. Yet
because there is a standard, archives should follow it.

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2.11 Relative phase

It is even possible to enlarge upon the above ideal, and insist on correct “relative phase”
as well. Please allow me to explain this, even though we shan’t encounter the problem
very often. Practical recording equipment (both analogue and digital) introduces relative
phase shifts between different components of the same sound, which may occur due to
acoustic effects, mechanical effects, or electronic effects. Any piece of equipment which
“rolls off” the extreme high frequencies, for example, also delays them with respect to
the low frequencies - admittedly by not very much, half a cycle at most. Since this
happens every time we shout through a wall (for example), our ears have evolved to
ignore this type of delay.
Many years ago at my Engineering Training School, our class was given a
demonstration which was supposed to prove we couldn’t hear relative phase distortion.
The test-generator comprised eight gearwheels on a common axle. The first had 100
teeth, the second 200, the third 300, etc. As the axle rotated, eight pickup coils detected
each tooth as it passed. The eight outputs were mixed together, displayed on an
oscilloscope, and reproduced on a loudspeaker. The pickup coils could be moved slightly
in relation to the gearwheels. As this was done, the relative phases of the components
changed, and the waveform displayed on the oscilloscope changed radically. The sound
heard from the loudspeaker wasn’t supposed to change; but of course there was one
sceptic in our class who insisted it did, and when the class had officially finished, we spent
some time in the lab blind-testing him - the result was that he could indeed hear a
difference.
But I don’t mention this for the small proportion of listeners who can hear a
difference. I mention it because the elimination of overload distortion may depend
critically upon the correct reproduction of “relative phase.” So I shall be insisting on
reproduction techniques which have this feature, and on using originals (since we usually
don’t know the relative-phase characteristics of any equipment making copies).

2.12 Scale distortion

The third consideration has had several names over the years. The controversy flared up
most brightly in the early 1960s, when it was called “scale distortion”. It arises from the
fact that we almost never hear a sound recording at the same volume as the original
sounds. Various psychoacoustic factors come into this, which I won’t expound now, but
which may be imagined by knowledgeable readers when I mention the “Fletcher-Munson
Curves.” Where does the controversy arise? Because it is not clear what we should do
when the sounds are reproduced at the wrong volume. I think everyone agrees that in the
ideal world we should reproduce the original volume. The trouble for archivists is that we
do not usually have objective knowledge of what this original volume was. A standard
sound-calibration would be needed at every location, and this would have to be included
on every recording. Such calibrations do occasionally appear on recordings of industrial
noises or historic musical instruments, but they are the exception rather than the rule. Yet
every time we have even a tiny scrap of such information we should preserve it.
The acoustic-recording system is again a case where this applies. It was not
possible to alter the sensitivity of an acoustic recording machine during a “take”, so it
would be silly to transfer such a recording without including a calibration signal to link the
original waveform with the transferred version. And I would point out that many early

23
commercial electrically-recorded discs were subject to “wear tests” before being approved
for publication. At least one studio kept documentary evidence of the settings of their
equipment, in case they were forced to retake an item because a test-pressing wore out
(Section 6.19). Thus we do have an indication of how loud the performance was,
although we have not yet learnt how to interpret this information.

2.13 Conclusion

Unfortunately, in the real world, procedures cannot be perfect, and ad-hoc decisions
frequently have to be made. In the remainder of this manual, we shall see there are many
areas for which technical information is incomplete. We must avoid making any objective
copies unless we have the technology and the knowledge to do them. There must also be
a deliberate and carefully-constructed policy of what to do in less-than-ideal
circumstances. For example, in today’s world with very restricted facilities, which should
have maximum priority: vulnerable media, media in maximum demand, or media which
could result in further knowledge? What should be the policy if the archive cannot get
hold of a good copy of a record? What should be the policy if appropriate technology is
not available? And is this decision affected when fundamental engineering theory tells us
such technology is always impossible?
Fortunately, there’s more than enough work for my recommendations to be
implemented immediately. We do not have to wait for answers to those questions!

REFERENCES

1: Mark Buchanan, “Beyond reality” (article), London: New Scientist Vol. 157 No. 2125
(14th March 1998), pp. 27-30. A more general article is: Robert Matthews, “I Is
The Law” (article), London: New Scientist Vol. 161 No. 2171 (30th January
1999), pp. 24-28.
2: Richard Clemow, “Computerised tape testing” (article), London: One to One
(magazine) No. 53 (September 1994), pp. 67-75.
3: Michael Gerzon, “Don’t Destroy The Archives!”. A technical report, hitherto
unpublished, dated 14th December 1992.
4: Lipshitz and Vanderkooy, “Polarity Calibration Tape (Issue 2)” (article), Journal of the
Audio Engineering Society Vol. 29 Nos. 7/8 (July/August 1981), pp. 528-530.

24
3 Digital conversion of analogue sound

3.1 The advantages of digital audio

There is a primary reason why digital recordings appeal to sound archivists. Once digital
encoding has been achieved, they can in principle last for ever without degradation,
because digital recordings can in principle be copied indefinitely without suffering any
further loss of quality. This assumes: (1) the media are always copied in the digital domain
before the errors accumulate too far, (2) the errors (which can be measured) always lie
within the limits of the error-correction system (which can also be measured), and (3)
after error-correction has been achieved, the basic digits representing the sound are not
altered in any way. When a digital recording goes through such a process successfully, it is
said to be “cloned.”
Both in theory and in practice, no power-bandwidth product (Section 2.3) is lost
when cloning takes place - there can be no further loss in quality. However, it also means
the initial analogue-to-digital conversion must be done well, otherwise faults will be
propagated forever. In fact, the Compact Digital Disc (CD) has two “layers” of error-
correction, and (according to audio folk-culture) the format was designed to be rugged
enough to allow a hole one-sixteenth of an inch diameter (1.5mm) to be drilled through
the disc without audible side-effects.
For all these reasons, the word “digital” began to achieve mystical qualities to the
general public, many of whom evidently believe that anything “digital” must be superior!
I am afraid much of this chapter will refute that idea. It will also be necessary to
understand what happens when a recording is copied “digitally” without actually being
“cloned”.
The power-bandwidth products of most of today’s linear pulse-code modulation
media exceed most of today’s analogue media, so it seems logical to copy all analogue
recordings onto digital carriers anyway, even if the digital coding is slightly imperfect. But
we must understand the weaknesses of today's systems if we are to avoid them (thus
craftsmanship is still involved!), and we should ideally provide test-signals to document
the conversion for future generations.
If you are a digital engineer, you will say that digital pulse-code modulation is a
form of “lossless compression”, because we don’t have to double the ratio between the
powers of the background-noise and of the overload-point in order to double the power-
bandwidth product. In principle, we could just add one extra digital bit to each sample, as
we shall see in the next section. Now I am getting ahead of myself; but I mention this
because there is sometimes total lack of understanding between digital and analogue
engineers about such fundamental issues. I shall therefore start by describing these
fundamental issues very thoroughly. So I must apologise to readers on one side of the
fence or the other, for apparently stating the obvious (or the incomprehensible).
A digital recording format seems pretty idiot-proof; the data normally consists of
ones and zeros, with no room for ambiguity. But this simply isn’t the case. All digital
carriers store the digits as analogue information. The data may be represented by the size
of a pit, or the strength of a magnetic domain, or a blob of dye. All these are quantified
using analogue measurements, and error-correction is specifically intended to get around
this difficulty (so you don’t have to be measuring the size of a pit or the strength of a tiny
magnet).

25
Unfortunately, such misunderstandings even bedevil the process of choosing an
adequate medium for storing the digitised sound. There are at least two areas where
these misunderstandings happen. First we must ask, are tests based on analogue or digital
measurements (such as “carrier-to-noise ratio” or “bit error-rates”)? And secondly, has
the digital reproducer been optimised for reproducing the analogue features, or is it a
self-adjusting “universal” machine (and if so, how do we judge that)?
Finally, the word “format” even has two meanings, both of which should be
specified. The digits have their own “format” (number of bits per sample, sampling-
frequency, and other parameters we shall meet later in this chapter); and the actual digits
may be recorded on either analogue or digital carrier formats (such as Umatic
videocassettes, versions of Compact Digital discs, etc.). The result tends to be total
confusion when “digital” and “analogue” operators try communicating!

3.2 Technical restrictions of digital audio - the “power” element

I shall now look at the principles of digital sound recording with the eyes of an analogue
engineer, to check how digital recording can conserve the power-bandwidth product. I
will just remind you that the “power” dimension defines the interval between the faintest
sound which can be recorded, and the onset of overloading.
All digital systems overload abruptly, unless they are protected by preceding
analogue circuitry. Fortunately analogue sound recordings generally have a smaller
“power” component to their power-bandwidth products, so there is no need to overload
a digital medium when you play an analogue recording. Today the principal exception
concerns desktop computers, whose analogue-to-digital converters are put into an
“electrically noisy” environment. Fortunately, very low-tech analogue noise-tests define
this situation if it occurs; and to make the system “idiot-proof” for non-audio experts,
consumer-quality cards often contains an automatic volume control as well (chapter 10).
Unfortunately, we now have two philosophies for archivists, and you may need to
work out a policy on this matter. One is to transfer the analogue recording at “constant-
gain,” so future users get a representation of the signal-strength of the original, which
might conceivably help future methods of curing analogue distortion (section 4.15). The
other is that all encoders working on the system called Pulse Code Modulation (or PCM)
have defects at the low end of the dynamic range. Since we are copying (as opposed to
doing “live” recording), we can predict very precisely what the signal volume will be
before we digitise it, set it to reach the maximum undistorted volume, and drown the
quiet side-effects.
These low-level effects are occupying the attention of many talented engineers,
with inevitable hocus-pocus from pundits. I shall spend the next few paragraphs outlining
the problem. If an ideal solution ever emerges, you will be able to sort the wheat from the
chaff and adopt it yourself. Meanwhile, it seems to me that any future methods of
reducing overload distortion will have to “learn” - in other words, adapt themselves to
the actual distortions present - rather than using predetermined “recipes”.
All PCM recordings will give a “granular” sound quality if they are not “dithered.”
This is because wanted signals whose volume is similar to the least significant bit will be
“chopped up” by the lack of resolution at this level. The result is called “quantisation
distortion.” One solution is always to have some background noise.
Hiss may be provided from a special analogue hiss-generator preceding the
analogue-to-digital converter. (Usually this is there whether you want it or not!)

26
Alternatively it may be generated by a random-number algorithm in the digital domain.
Such “dither” will completely eliminate this distortion, at the cost of a very faint steady
hiss being added. The current debate is about methods of reducing, or perhaps making
deliberate use of, this additional hiss.
Today the normal practice is to add “triangular probability distribution” noise,
which is preferable to the “rectangular probability distribution” of earlier days, because
you don’t hear the hiss vary with signal volume (an effect called “modulation noise”).
Regrettably, many “sound cards” for computers still use rectangular probability
distribution noise, illustrating the gulf which can exist between digital engineers and
analogue ones! It also illustrates why you must be aware of basic principles on both sides
of the fence.
Even with triangular probability distribution noise, in the past few years this
strategy has been re-examined. There are now several processes which claim to make the
hiss less audible to human listeners while simultaneously avoiding quantisation distortion
and modulation noise. These processes have different starting-points. For example, some
are for studio recordings with very low background noise (at the “twenty-bit level”) so
they will sound better when configured for sixteen-bit Compact Discs. Such hiss might
subjectively be fifteen decibels quieter, yet have an un-natural quality; so other processes
aim for a more “benign” hiss. A process suggested by Philips adds something which
sounds like hiss, but which actually comprises pseudo-random digits which carry useful
information. Another process (called “auto-dither”) adds pseudo-random digits which can
be subtracted upon replay, thereby making such dither totally inaudible, although it will
still be reproduced on an ordinary machine. Personally I advocate good old triangular
probability distribution noise for the somewhat esoteric reason that it’s always possible to
say where the “original sound” stopped and the “destination medium” starts.
All this is largely irrelevant to archivists transferring analogue recordings to digital,
except that you should not forget “non-human” applications such as wildlife recording.
There is also a risk of unsuspected cumulative build-ups over several generations of digital
processing, and unexpected side-effects (or actual loss of information) if some types of
processing are carried out upon the results.
If you put a recording through a digital process which drops the volume of some or
all of the recording so that the remaining hiss (whether “dither” or “natural”) is less than
the least-significant bit, it will have to be “re-dithered.” We should also perform re-
dithering if the resulting sounds involve fractions of a bit, rather than integers; I am told
that a gain change of a fraction of a decibel causes endless side-effects! One version of a
widely-used process called The Fast Fourier Transform splits the frequency-range into
2048 slices, so the noise energy may be reduced by a factor of 2048 (or more) in some
slices. If this falls below the least-significant bit, quantisation distortion will begin to affect
the wanted signal.
In my personal view, the best way of avoiding these troubles is to use 24-bit
samples during the generation of any archive and objective copies which need digital
signal processing. The results can then be reduced to 16-bits for storage, and the side-
effects tailored at that stage. Practical 24-bit encoders do not yet exist, because they have
insufficiently low background noise; but this is exactly why they solve the difficulty.
Provided digital processing takes place in the 24-bit domain, the side-effects will be at
least 48 decibels lower than with 16-bit encoding, and quantisation distortion will hardly
ever come into the picture at all.
On the other hand, if 16-bit processing is used, the operator must move his sound
from one process to the next without re-dithering it unnecessarily (to avoid building up

27
the noise), but to add the dither whenever it is needed to kill the distortion. This means
intelligent judgement throughout. The operator must ask himself “Did the last stage result
in part or all of the wanted signal falling below the least-significant bit?” And at the end
of the processes he must ask himself “Has the final stage resulted in part or all of the
signal falling below the least-significant bit?” If the answer to either of these questions is
Yes, then the operator must carry out a new re-dithering stage. This is an argument for
ensuring the same operator sees the job through from beginning to end.

3.3 Technical limitations of digital audio: the “bandwidth” element

In this section I shall discuss how the frequency range may be corrupted by digital
encoding. The first point is that the coding system known as “PCM” (almost universally
used today) requires “anti-aliasing filters.” These are deliberately introduced to reduce the
frequency range, contrary to the ideals mentioned in section 2.4.
This is because of a mathematical theorem known as “Shannon’s Sampling
Theorem.” Shannon showed that if you have to take samples of a time-varying signal of
any type, the data you have to store will correctly represent the signal if the signal is
frequency-restricted before you take the samples. After this, you only need to sample the
amplitude of the data at twice the cut-off frequency. This applies whether you are taking
readings of the water-level in a river once an hour, or encoding high-definition television
pictures which contain components up to thirty million per second.
To describe this concept in words, if the “float” in the river has “parasitic
oscillations” due to the waves, and bobs up and down at 1Hz, you will need to make
measurements of its level at least twice a second to reproduce all its movements faithfully
without errors. If you try to sample at (say) only once a minute, the bobbing-actions will
cause “noise” added to the wanted signal (the longer-term level of the river), reducing
the precision of the measurements (by reducing the power-bandwidth product).
Any method of sampling an analogue signal will misbehave if it contains any
frequencies higher than half the sampling frequency. If, for instance, the sampling-
frequency is 44.1kHz (the standard for audio Compact Discs), and an analogue signal
with a frequency of 22.06kHz gets presented to the analogue-to-digital converter, the
resulting digits will contain this frequency “folded back” - in this case, a spurious
frequency of 22.04kHz. Such spurious sounds can never be distinguished from the
wanted signal afterwards. This is called “aliasing.”
Unfortunately, aliasing may also occur when you conduct “non-linear” digital
signal-processing upon the results. This has implications for the processes you should use
before you put the signal through the analogue-to-digital conversion stage. On the other
hand, quite a few processes are easier to carry out in the digital domain. These processes
must be designed so as not to introduce significant aliasing, otherwise supposedly-
superior methods may come under an unquantifiable cloud - although most can be shown
up by simple low-tech analogue tests!
The problem is Shannon’s sampling theorem again. For example, a digital process
may recognise an analogue “click” because it has a leading edge and a trailing edge, both
of which are faster than any naturally-occurring transient sound. But Shannon’s sampling
theorem says the frequency range must not exceed half the sampling-frequency; this
bears no simple relationship to the slew-rate, which is what the computer will recognise in
this example. Therefore the elimination of the click will result in aliased artefacts mixed up

28
with the wanted sound, unless some very computation-intensive processes are used to
bandwidth-limit these artefacts.
Because high-fidelity digital audio was first tried in the mid-1970s when it was very
difficult to store the samples fast enough, the anti-aliasing filters were designed to work
just above the upper limit of human hearing. There simply wasn’t any spare capacity to
provide any “elbow-room,” unlike measuring the water-level in a river. The perfect filters
required by Shannon’s theorem do not exist, and in practice you can often hear the result
on semi-pro or amateur digital machines if you try recording test-tones around 20-25
kHz. Sometimes the analogue filters will behave differently on the two channels, so stereo
images will be affected. And even if “perfect filters” are approached by careful
engineering, another mathematical theorem called “the Gibbs effect” may distort the
resulting waveshapes.
An analogue “square-wave” signal will acquire “ripples” along its top and bottom
edges, looking exactly like a high-frequency resonance. If you are an analogue engineer
you will criticise this effect, because analogue engineers are trained to eliminate
resonances in their microphones, their loudspeakers, and their electrical circuitry; but this
phenomenon is actually an artefact of the mathematics of steeply bandwidth-limiting a
frequency before you digitise it. Such factors cause distress to golden-eared analogue
engineers, and have generated much argument against digital recording. “Professional-
standard” de-clicking devices employ “oversampling” to overcome the Gibbs effect on
clicks; but this cannot work with declicking software on personal computers, for example.
The Gibbs effect can be reduced by reasonably gentle filters coming into effect at
about 18kHz, when only material above the limit of hearing for an adult human listener
would be thrown away. But we might be throwing away information of relevance in other
applications, for instance analysis by electronic measuring instruments, or playback to
wildlife, or to young children (whose hearing can sometimes reach 25kHz). So we must
first be sure only to use linear PCM at 44.1kHz when the subject matter is only for adult
human listeners. This isn’t a condemnation of digital recording as such, of course. It is only
a reminder to use the correct tool for any job.
You can see why the cult hi-fi fraternity sometimes avoids digital recordings like
the plague! Fortunately, this need not apply to you. Ordinary listeners cannot compare
quality “before” and “after”; but you can (and should), so you needn’t be involved in the
debate at all. If there is a likelihood of mechanical or non-human applications, then a
different medium might be preferable; otherwise you should ensure that archive copies
are made by state-of-the-art converters checked in the laboratory and double-checked by
ear.
I could spend some time discussing the promise of other proposed digital encoding
systems, such as non-linear encoding or delta-sigma modulation, which have advantages
and disadvantages; but I shall not do so until such technology becomes readily available
to the archivist. One version of delta-sigma modulation has in fact just become available;
Sony/Philips are using it for their “Super Audio CD” (SACD). The idea is to have “one-
bit” samples taken at very many times the highest wanted frequency. Such “one-bit
samples” record whether the signal is going “up” or “down” at the time the sample is
taken. There is no need for an anti-aliasing filter, because Shannon’s theorem does not
apply. However, the process results in large quantities of quantisation noise above the
upper limit of human hearing. At present, delta-modulation combines the advantage of
no anti-aliasing filter with the disadvantage that there is practically no signal-processing
technology which can make use of the bitstream. If delta modulation “takes off”, signal
processes will eventually become available, together with the technology for storing the

29
required extra bits. (SACD needs slightly more bits than a PCM 24-bit recording sampled
at 96kHz). But for the moment, we are stuck with PCM, and I shall assume PCM for the
remainder of this manual.

3.4 Operational techniques for digital encoding

I hesitate to make the next point again, but I do so knowing many operators don’t work
this way. Whatever the medium, whether it be digital or analogue, the transfer operator
must be able to compare source and replay upon a changeover switch. Although this is
normal for analogue tape recording, where machines with three heads are used to play
back a sound as soon as it is recorded, they seem to be very rare in digital environments.
Some machines do not even give “E-E monitoring”, where the encoder and decoder
electronics are connected back-to-back for monitoring purposes. So, I think it is absolutely
vital for the operator to listen when his object is to copy the wanted sound without
alteration. Only he is in a position to judge when the new version is faithful to the
original, and he must be prepared to sign his name to witness this. Please remember my
philosophy, which is that all equipment must give satisfactory measurements; but the ear
must be the final arbiter.
Even if E-E monitoring passes this test, it does not prove the copy is perfect. There
could be dropouts on tape, or track-jumps on CD-R discs. Fortunately, once digital
conversion has been done, automatic devices may be used to check the medium for
errors; humans are not required.
I now mention a novel difficulty, documenting the performance of the anti-aliasing
filter and the subsequent analogue-to-digital converter. Theoretically, everything can be
documented by a simple “impulse response” test. The impulse-response of a digital-to-
analogue converter is easy to measure, because all you need is a single sample with all its
bits at “1”. This is easy to generate, and many test CDs carry such signals. But there isn’t
an “international standard” for documenting the performance of analogue-to-digital
converters. One is desperately needed, because such converters may have frequency-
responses down to a fraction of 1Hz, which means that one impulse may have to be
separated by many seconds; while the impulse must also be chosen with a combination of
duration and amplitude to suit the “sample-and-hold” circuit as well as not to overload
the anti-aliasing filter.
At the British Library Sound Archive, we use a Thurlby Thandar Instruments type
TGP110 analogue Pulse Generator in its basic form (not “calibrated” to give highly
precise waveforms). We have standardised on impulses exactly 1 microsecond long. The
resulting digitised shape can be displayed on a digital audio editor.

3.5 Difficulties of “cloning” digital recordings

For the next three sections, you will think I am biased against “digital sound”; but the fact
that many digital formats have higher power-bandwidth products should not mean we
should be unaware of their problems. I shall now point out the defects of the assumption
that digital recordings can be cloned. My earlier assertion that linear PCM digital
recordings can be copied without degradation had three hidden assumptions, which seem
increasingly unlikely with the passage of time. They are that the sampling frequency, the
pre-emphasis, and the bit-resolution remain constant.

30
If (say) it is desired to copy a digital PCM audio recording to a new PCM format
with a higher sampling frequency, then either the sound must be converted from digital
to analogue and back again, or it must be recalculated digitally. When the new sampling-
frequency and the old have an arithmetical greatest common factor n, then every nth
sample remains the same; but all the others must be a weighted average of the samples
either side, and this implies that the new bitstream must include “decimals.” For truthful
representation, it cannot be a stream of integers; rounding-errors (and therefore
quantisation-distortion) are bound to occur.
The subjective effect is difficult to describe, and with practical present-day systems
it is usually inaudible when done just once (even with integers). But now we have a quite
different danger, because once people realise digital conversion is possible (however
imperfect), they will ask for it again and again, and errors will accumulate through the
generations unless there is strict control by documentation. Therefore it is necessary to
outguess posterity, and choose a sampling-frequency which will not become obsolete.
For the older audio media, the writer’s present view is that 44.1kHz will last,
because of the large number of compact disc players, and new audio media (like the DCC
and MiniDisc) use 44.1kHz as well. I also consider that for the media which need urgent
conservation copying (wax cylinders, acetate-based recording tape, and “acetate” discs,
none of which have very intense high frequencies), this system results in imperceptible
losses, and the gains outweigh these. For picture media I advocate 48kHz, because that is
the rate used by digital video formats.
But there will inevitably be losses when digital sound is moved from one domain to
the other, and this will get worse if sampling-frequencies proliferate. Equipment is
becoming available which works at 96kHz, precisely double the frequency used by
television. Recordings made at 48kHz can then be copied to make the “even-numbered”
samples, while the “odd-numbered” samples become the averages of the samples either
side. The options for better anti-aliasing filters, applications such as wildlife recording,
preservation of transients (such as analogue disc clicks), transparent digital signal-
processing, etc. remain open. Yet even this option requires that we document what has
happened - future workers cannot be expected to guess it.
Converting to a lower sampling frequency means that the recording must be
subjected to a new anti-aliasing filter. Although this filtering can be done in the digital
domain to reduce the effects of practical analogue filters and two converters, it means
throwing away some of the information of course.
The next problem is pre-emphasis. This means amplifying some of the wanted high
frequencies before they are encoded, and doing the inverse on playback. This renders the
sound less liable to quantisation distortion, because any “natural” hiss is about twelve
decibels stronger. At present there is only one standard pre-emphasis characteristic for
digital audio recording (section 7.3), so it can only be either ON or OFF. A flag is set in
the digital data-stream of standardised interconnections, so the presence of pre-emphasis
may be recognised on playback. And there is a much more powerful pre-emphasis system
(the “C.C.I.T” curve) used in telecommunications. It is optimised for 8-bit audio work,
and 8 bits were once often used by personal computers for economical sound recording.
Hopefully my readers won’t be called upon to work with CCIT pre-emphasis, because
digital sound-card designers apparently haven’t learnt that 8-bit encoders could then give
the dynamic range of professional analogue tape; but you ought to know the possibility
exists! But if the recording gets copied to change its pre-emphasis status, whether
through a digital process or an analogue link, some of the power-bandwidth product will
be lost each time. Worse still, some digital signal devices (particularly hard-disk editors)

31
strip off the pre-emphasis flag, and it is possible that digital recordings may be reproduced
incorrectly after this. (Or worse still, parts of digital recordings will be reproduced
incorrectly).
I advise the reader to make a definite policy on the use of pre-emphasis and stick
to it. The “pros” are that with the vast majority of sounds meant for human listeners, the
power-bandwidth product of the medium is used more efficiently; the “cons” are that this
doesn’t apply to most animal sounds, and digital metering and processing (Chapter 3) are
sometimes more difficult. Yet even this option requires that we document what has
happened - future workers cannot be expected to guess it.
Converting a linear PCM recording to a greater number of bits (such as going from
14-bit to 16-bit) does not theoretically mean any losses. In fact, if it happens at the same
time as a sample-rate conversion, the new medium can be made to carry two bits of the
decimal part of the interpolation mentioned earlier, thereby reducing the side-effects.
Meanwhile the most significant bits retain their status, and the peak volume will be the
same as for the original recording. So if it ever becomes normal to copy from (say) 16-bits
to 20-bits in the digital domain, it will be easier to change the sampling frequency as well,
because the roundoff errors will have less effect, by a factor of 16 in this example. Thus,
to summarise, satisfactory sampling-frequency conversion will always be difficult; but it
will become easier with higher bit-resolutions. Yet even this option requires that we
document what has happened - future workers cannot be expected to guess it.
All these difficulties are inherent - they cannot be solved with “better technology”
- and this underlines the fact that analogue-to-digital conversions and digital signal
processing must be done to the highest possible standards. Since the AES Interface for
digital audio allows for 24-bit samples, it seems sensible to plan for this number of bits,
even though the best current converters can just about reach the 22-bit level.
It is nearly always better to do straight digital standards conversions in the digital
domain when you must, and a device such as the Digital Audio Research “DASS” Unit
may be very helpful. This offers several useful processes. Besides changing the pre-
emphasis status and the bit-resolution, it can alter the copy-protect bits, reverse both
relative and absolute phases, change the volume, and remove DC offsets. The unit
automatically looks after the process of “re-dithering” when necessary, and offers two
different ways of doing sampling-rate conversion. The first is advocated when the two
rates are not very different, but accurate synchronisation is essential. This makes use of a
buffer memory of 1024 samples. When this is either empty or full, it does a benign digital
crossfade to “catch up,” but between these moments the data-stream remains
uncorrupted. The other method is used for widely-differing sampling-rates which would
overwhelm the buffer. This does the operation described at the start of this section,
causing slight degradation throughout the whole of the recording. Yet even this option
requires that we document what has happened - future workers cannot be expected to
guess it.
Finally I must remind you that digital recordings are not necessarily above criticism.
I can think of many compact discs with quite conspicuous analogue errors on them. Some
even have the code “DDD” (suggesting only digital processes have been used during
their manufacture). It seems some companies use analogue noise reduction systems
(Chapter 8) to “stretch” the performance of 16-bit recording media, and they do not
understand the “old technology.” Later chapters will teach you how to get accurate
sound from analogue media; but please keep your ears open, and be prepared for the
same faults on digital media!

32
3.6 Digital data compression

The undoubted advantages of linear PCM as a way of storing audio waveforms are being
endangered by various types of digital data compression. The idea is to store digital sound
at lower cost, or to transmit it using less of one of our limited natural resources (the
electromagnetic spectrum).
Algorithms for digital compression are of two kinds, “lossless” and “lossy.” The
lossless ones give you back the same digits after decompression, so they do not affect the
sound. There are several processes; one of the first (Compusonics) reduced the data to
only about four-fifths the original amount, but the compression rate was fixed in the
sense that the same number of bits was recorded in a particular time. If we allow the
recording medium to vary its data-rate depending on the subject matter, data-reduction
may be two-thirds for the worst cases to one-quarter for the best. My personal view is
that these aren’t worth bothering with, unless you’re consistently in the situation where
the durations of your recordings are fractionally longer than the capacity of your storage
media.
For audio, some lossless methods actually make matters worse. Applause is
notorious for being difficult to compress; if you must use such compression, test it on a
recording of continuous applause. You may even find the size of the file increases.
But the real trouble comes from lossy systems, which can achieve compression
factors from twofold to at least twentyfold. They all rely upon psychoacoustics to permit
the digital data stream to be reduced. Two such digital sound recording formats were the
Digital Compact Cassette (DCC) and the MiniDisc, each achieving about one-fifth the
original number of bits; but in practice, quoted costs were certainly not one-fifth! While
they make acceptable noises on studio-quality recordings, it is very suspicious that no
“back-catalogue” is offered. The unpredictable nature of background noise always gives
problems, and that is precisely what we find ourselves trying to encode with analogue
sources. Applause can also degenerate into a noisy “mush”. The real reason for their
introduction was not an engineering one. Because newer digital systems were designed so
they could not clone manufactured CDs, the professional recording industry was less likely
to object to their potential for copyright abuse (a consideration we shall meet in section
3.8 below).
Other examples of digital audio compression methods are being used for other
applications. To get digital audio between the perforation-holes of 35mm optical film,
cinema surround-sound was originally coded digitally into a soundtrack with lossy
compression. Initial reports suggested it sometimes strained the technology beyond its
breaking-point. While ordinary stereo didn’t sound too bad, the extra information for the
rear-channel loudspeakers caused strange results to appear. An ethical point arises here,
which is that the sound-mixers adapted their mixing technique to suit the compression-
system. Therefore the sound was changed to suit the medium. (In this case, no original
sound existed in the first place, so there wasn’t any need to conserve it.)
A number of compression techniques are used for landline and satellite
communication, and here the tradeoffs are financial - it costs money to buy the power-
bandwidth product of such media. Broadcasters use digital compression a lot – NICAM
stereo and DAB have it - but this is more understandable, because there is a limited
amount of electromagnetic spectrum which we must all share, especially for consistent
reception in cars. At least we can assume that wildlife creatures or analytical machinery
won’t be listening to the radio, visiting cinemas, or driving cars.

33
The “advantages” of lossy digital compression have six counterarguments. (1) The
GDR Archive found that the cost of storage is less than five percent of the total costs of
running an archive, so the savings are not great; (2) digital storage (and transmission) are
set to get cheaper, not more expensive; (3) even though a system may sound transparent
now, there’s no evidence that we may not hear side-effects when new applications are
developed; (4) once people think digital recordings can be “cloned”, they will put lossy
compression systems one after the other and think they are preserving the original sound,
whereas cascading several lossy compression-systems magnifies all the disadvantages of
each; (5) data compression systems will themselves evolve, so capital costs will be
involved; (6) even digital storage media with brief shelf-lives seem set to outlast current
compression-systems.
There can be no “perfect” lossy compression system for audio. In section 1.2 I
described how individual human babies learned how to hear, and how a physiological
defect might be compensated by a psychological change. Compression-systems are
always tested by people with “normal” hearing (or sight in the case of video
compression). This research may be inherently wrong for people with defective hearing
(or sight). Under British law at least, the result might be regarded as discriminating against
certain members of the public. Although there has not yet been any legal action on this
front, I must point out the possibility to my readers.
With all lossy systems, unless cloning with error-correction is provided, the sound
will degrade further each time it is copied. I consider a sound archive should have nothing
to do with such media unless the ability to clone the stuff with error-correction is made
available, and the destination-media and the hardware also continue to be available. The
degradations will then stay the same and won’t accumulate. (The DCC will do this, but
won’t allow the accompanying text, documentation, and start-idents to be transferred;
and you have to buy a special cloning machine for MiniDisc, which erases the existence of
edits).
Because there is no “watermark” to document what has happened en route,
digital television is already giving endless problems to archivists. Since compression is vital
for getting news-stories home quickly - and several versions may be cascaded depending
on the bitrates available en route - there is no way of knowing which version is “nearest
the original”. So even this option requires that we document what has happened if we
can - future workers cannot be expected to guess it.
Ideally, hardware should be made available to decode the compressed bit-stream
with no loss of power-bandwidth product under audible or laboratory test-conditions.
This isn’t always possible with the present state-of-the-art; but unless it is, we can at least
preserve the sound in the manner the misguided producers intended, and the advantages
of uncompressed PCM recording won’t come under a shadow. When neither of these
strategies is possible, the archive will be forced to convert the original back to analogue
whenever a listener requires, and this will mean considerable investment in equipment
and perpetual maintenance costs. All this underlines my feeling that a sound archive
should have nothing to do with lossy data compression.
My mention of the Compusonics system reminds me that it isn’t just a matter of
hardware. There is a thin dividing line between “hardware” and “software.” I do not
mean to libel Messrs. Compusonics by the following remark, but it is a point I must make.
Software can be copyrighted, which reduces the chance of a process being usable in
future.

34
3.7 A severe warning

I shall make this point more clearly by an actual example in another field. I started writing
the text of this manual about ten years ago, and as I have continued to add to it and
amend it, I have been forced to keep the same word-processing software in my computer.
Unfortunately, I have had three computers during that time, and the word-processing
software was licensed for use on only one machine. I bought a second copy with my
second machine, but by the time I got my third machine the product was no longer
available. Rather than risk prosecution by copying the software onto my new machine, I
am now forced to use one of the original floppies in the floppy disk drive, talking to the
text on the hard drive. This adds unnecessary operational hassles, and only works at all
because the first computer and the last happen to have had the same (copyright)
operating system (which was sheer luck).
For a computer-user not used to my way of thinking, the immediate question is
“What’s wrong with buying a more up-to-date word-processor?” My answer is threefold.
(1) There is nothing wrong with my present system; (2) I can export my writings in a way
which avoids having to re-type the stuff for another word-processor, whereas the other
way round simply doesn’t work; (3) I cannot see why I should pay someone to re-invent
the wheel. (I have better things to spend my time and money on)! And once I enter these
treacherous waters, I shall have to continue shelling out money for the next fifty years or
more.
I must now explain that last remark, drawing from my own experience in Britain,
and asking readers to apply the lessons of what I say to the legal situation wherever they
work. In Britain, copyright in computer software lasts fifty years after its first publication
(with new versions constantly pushing this date further into the future). Furthermore,
under British law, you do not “buy” software, you “license” it. So the normal provisions
of the Consumer Protection Act (that it must be “fit for the intended purpose” - i.e. it
must work) simply do not apply. Meanwhile, different manufacturers are free to impose
their ideas about what constitutes “copying” to make the software practicable. (At
present, this isn’t defined in British law, except to say that software may always legally be
copied into RAM - memory in which the information vanishes when you switch off the
power). Finally, the 1988 Copyright Act allows “moral” rights, which prevent anyone
modifying anything “in a derogatory manner”. This right cannot be assigned to another
person or organisation, it stays with the author; presumably in extreme cases it could
mean the licensee may not even modify it.
It is easy to see the difficulties that sound archivists might face in forty-nine years
time, when the hardware has radically changed. (Think of the difficulties I’ve had in only
ten years with text!) I therefore think it is essential to plan sound archival strategy so that
no software is involved. Alternatively, the software might be “public-domain” or “home-
grown,” and ideally one should have access to the “source code” (written in an
internationally-standardised language such as FORTRAN or “C”), which may
subsequently be re-compiled for different microprocessors. I consider that even temporary
processes used in sound restoration, such as audio editors or digital noise reduction
systems (Chapter 3) should follow the same principles, otherwise the archivist is bound to
be painted into a corner sooner or later. If hardware evolves at its present rate, copyright
law may halt legal playback of many digital recording formats or implementation of many
digital signal processes. Under British law, once the “software” is permanently stored in a
device known as an “EPROM” chip, it becomes “hardware”, and the problems of

35
copyright software evaporate. But this just makes matters more difficult if the EPROM
should fail.
I apologise to readers for being forced to point out further “facts of life,” but I
have never seen the next few ideas written down anywhere. It is your duty to understand
all the “Dangers of Digital”, so I will warn you about more dangers of copyright software.
The obvious one, which is that hardware manufacturers will blame the software writers if
something goes wrong (and vice versa), seems almost self-evident; but it still needs to be
mentioned.
A second-order danger is that publishers of software often make deliberate
attempts to trap users into “brand loyalty”. Thus, I can think of many word-processing
programs reissued with “upgrades” (real or imagined), sometimes for use with a new
“operating system”. But such programs usually function with at least one step of
“downward compatibility”, so users are not tempted to cut their losses and switch to
different software. This has been the situation since at least 1977 (with the languages
FORTRAN66 and FORTRAN77); but for some reason no-one seems to have recognised
the problem. I regret having to make a political point here; but as both the computer-
magazine and book industries are utterly dependent upon not mentioning it, the point has
never been raised among people who matter (archivists!).
This disease has spread to digital recording media as well, with many backup media
(controlled by software) having only one level of downwards compatibility, if that. As a
simple example, I shall cite the “three-and-a-half inch floppy disk”, which exists in two
forms, the “normal” one (under MS-DOS this can hold 720 kilobytes), and the “high-
density” version (which can hold 1.44 megabytes). In the “high density” version the
digits are packed closer together, requiring a high-density magnetic layer. The hardware
should be able to tell which is which by means of a “feeler hole”, exactly like analogue
audiocassettes (Chapter 6). But, to cut costs, modern floppy disk drives lack any way of
detecting the hole, so cannot read 720k disks. The problem is an analogue one, and we
shall see precisely the same problem when we talk about analogue magnetic tape. Both
greater amplification and matched playback-heads must coexist to read older formats
properly.
Even worse, the downwards-compatibility situation has spread to the operating
system (the software which makes a computer work at all). For example, “Windows NT”
(which was much-touted as a 32-bit operating system, although no engineer would see
any advantage in that) can handle 16-bit applications, but not 8-bit ones. A large
organisation has this pistol held to its head with greater pressure, since if the operating
system must be changed, every computer must also be changed - overnight - or data
cannot be exchanged on digital media (or if they can, with meaningless error-messages or
warnings of viruses). All this arises because pieces of digital jigsaw do not fit together.
To a sound archivist, the second-order strategy is only acceptable so long as the
sound recordings do not change their format. If you think that an updated operating-
system is certain to be better for digital storage, then I must remind you that you will be
storing successive layers of problems for future generations. Use only a format which is
internationally-standardised and widely-used (such as “Red Book” compact disc), and do
not allow yourself to be seduced by potential “upgrades.”
A “third-order” problem is the well-known problem of “vapourware.” This is
where the software company deliberately courts brand-loyalty by telling its users an
upgrade is imminent. This has four unfavourable features which don’t apply to
“hardware.” First, no particular delivery-time is promised - it may be two or three years
away; second, the new version will need to be re-tested by users; third, operators will

36
have to re-learn how to use it; and almost inevitably people will then use the new version
more intensely, pushing at the barriers until it too “falls over.” (These won’t be the
responsibility of the software company, of course; and usually extra cash is involved as
well). Even if the software is buried in an EPROM chip (as opposed to a removable
medium which can be changed easily), this means that sound archivists must document
the “version number” for any archive copies, while the original analogue recording must
be preserved indefinitely in case a better version becomes available.
And there are even “fourth-order” problems. The handbook often gets separated
from the software, so you often cannot do anything practical even when the software
survives. Also, many software packages are (deliberately or accidentally) badly-written, so
you find yourself “trapped in a loop” or something similar, and must ring a so-called
“help-line” at a massive cost to your telephone bill. Even this wouldn’t matter if only the
software could be guaranteed for fifty years into the future . . . .
Without wishing to decry the efforts of legitimate copyright owners, I must remind
you that many forms of hardware come with associated software. For example, every
digital “sound card” I know has copyright software to make it work. So I must advise
readers to have nothing to do with sound cards, unless they are used solely as an
intermediate stage in the generation of internationally-standardised digital copies played
by means of hardware alone.

3.8 Digital watermarking and copy protection.

As I write this, digital audio is also becoming corrupted by “watermarking”. The idea is to
alter a sound recording so that its source may be identified, whether broadcast, sent over
the Internet, or whatever. Such treatment must be rugged enough to survive various
analogue or digital distortions. Many manufacturers have developed inaudible techniques
for adding a watermark, although they all corrupt “the original sound” in the process. As I
write this, it looks as though a system called “MusiCode” (from Aris Technologies) will
become dominant (Ref. 1). This one changes successive peaks in music to carry an extra
encoded message. As the decoding software will have the same problems as I described in
the previous section, it won’t be a complete answer to the archivist’s prayer for a
recording containing its own identification; but for once I can see some sort of advantage
accompanying this “distortion.” On the other hand, it means the archivist will have to
purchase (sorry, “license”) a copy of the appropriate software to make any practical use
of this information. And of course, professional sound archivists will be forced to license
both it and all the other watermarking systems, in order to identify unmodified versions
of the same sound.
Wealthy commercial record-companies will use the code to identify their products.
But such identification will not identify the copyright owner, for example when a product
is licensed for overseas sales, or the artists own the copyright in their own music. This
point is developed on another page of the same Reference (Ref. 2), where a rival
watermarking system is accompanied by an infrastructure of monitoring-stations listening
to the airwaves for recordings with their watermarks, and automatically sending reports to
a centralised agency which will notify the copyright owners.
I am writing this paragraph in 1999: I confidently predict that numerous
“watermarking” systems will be invented, they will allegedly be tested by professional
audio listeners using the rigorous A-B comparison methods I described in section 3.4, and
then discarded.

37
All this is quite apart from a machine-readable identification to replace a spoken
announcement (on the lines I mentioned in section 2.9). Yet even here, there are currently
five “standards” fighting it out in the marketplace, and as far as I can see these all depend
upon Roman characters for the “metadata”. I will say no more.
Another difficulty is “copy protection.” In a very belated attempt to restrict the
digital cloning of commercial products, digital media are beginning to carry extra “copy
protection bits.” The 1981 “Red Book” standard for compact discs allowed these from
Day One; but sales people considered it a technicality not worthy of their attention! But
people in the real world - namely, music composers and performers - soon realised there
was an enormous potential for piracy; and now we have a great deal of shutting of stable
doors.
The Compact Disc itself carries any “copy protect” flag, and many countries now
pay royalties on retail sales of recordable CDs specifically to compensate music and record
publishers. (Such discs may be marketed with a distinctly misleading phrase like “For
music”, which ought to read “For in-copyright published music, and copyright records,
only.” Then, when it doesn’t record anything else, there could be action under the Trades
Descriptions Act.) So a professional archive may be obliged to pay the “music” royalty
(or purchase “professional” CD-Rs) to ensure the copy-protect flags are not raised.
Meanwhile, blank CD-ROM discs for computers (which normally permit a third layer of
error-correction) can also be used for audio, when the third layer is ignored by CD players
(so the result becomes “Red Book” standard with two layers). The copy-protect bit is not
(at the time of writing) raised; so now another company has entered the field to corrupt
this third layer of error-correction and prevent copying from on a CD-ROM drive. (Ref. 3)
Most other media (including digital audio tape and MiniDisc) add a “copy-protect”
bit when recorded on an “amateur” machine, without the idea of copyright being
explained at any point - so the snag only becomes apparent when you are asked to clone
the result. The “amateur” digital connection (SP-DIF) carries start-flags as well as copy-
protect flags, while the “professional” digital connection (AES3) carries neither. So if you
need to clone digital recordings including their track-flags, it may be virtually impossible
with many combinations of media.
A cure is easy to see - it just means purchasing the correct equipment. Add a digital
delay-line with a capacity of ten seconds or so using AES connectors. Use a source-
machine which does a digital count-down of the time before each track ends. Then the
cloning can run as a background task to some other job, and meanwhile the operator can
press the track-increment button manually with two types of advance-warning - visual
and aural.

3.9 The use of general-purpose computers

Desktop computers are getting cheaper and more powerful all the time, so digital sound
restoration techniques will become more accessible to the archivist. So far these processes
are utterly dependent upon classical linear PCM coding; this is another argument against
compressed digital recordings.
Unfortunately, desktop computers are not always welcomed by the audio
fraternity, because most of them have whirring disk-drives and continuous cooling fans.
Analogue sound-restoration frequently means listening to the faintest parts of recordings,
and just one desktop computer in the same room can completely scupper this process.
Finally, analogue operators curse and swear at the inconvenient controls and non-intuitive

38
software. The senses of touch and instant responses are very important to an analogue
operator.
It is vital to plan a way around the noise difficulty. Kits are available which allow
the “system box” to be out of the room, while the keyboard screen and mouse remain on
the desktop. Alternatively, we might do trial sections on noise-excluding headphones, and
leave the computer to crunch through long recordings during the night-time. In section
1.4 I expressed the view that the restoration operator should not be twiddling knobs
subjectively. A computer running “out of real-time” forces the operator to plan his
processing logically, and actually prevents subjective intervention.
This brings us to the fact that desktop computers are only just beginning to cope
with real-time digital signal processing (DSP), although this sometimes implies dedicated
accelerator-boards or special “bus architectures” (both of which imply special software).
On the other hand, the desktop PC is an ideal tool for solving a rare technical problem.
Sound archivists do not have much cash, and there aren’t enough to provide a user-base
for the designers of special hardware or the writers of special software. But once we can
get a digitised recording into a PC and out again, it is relatively cheap to develop a tailor-
made solution to a rare problem, which may be needed only once or twice in a decade.
Even so, archivists often have specialist requirements which are needed more often
than that. This writer considers an acceptable compromise is to purchase a special board
which will write digital audio into a DOS file on the hard disk of a PC (I am told Turtle
Beach Electronics makes such a board, although it is only 16-bit capable, and requires its
own software). Then special software can be loaded from floppy disk to perform special
signal processing.

3.10 Processes better handled in the analogue domain

The present state-of-the-art means that all digital recordings will be subject to difficulties
if we want to alter their speeds. To be pedantic, the difficulties occur when we want to
change the speed of a digital recording, rather than its playback into the analogue
domain. In principle a digital recording can be varispeeded while converting it back into
analogue simply by running it at a different sampling-frequency, and there are a few
compact disc players, R-DAT machines, and multitrack digital audio workstations which
permit a small amount of such adjustment. But vari-speeding a digital recording can only
be done on expensive specialist equipment, often by a constant fixed percentage, not
adjustable while you are actually listening to it. Furthermore, the process results in
fractional rounding-errors, as we saw earlier.
So it is vital to make every effort to get the playing-speed of an analogue medium
right before converting it to digital. The subject is dealt with in Chapter 4; but I mention it
now because it clearly forms an important part of the overall strategy. Discographical or
musical experts may be needed during the copying session to select the appropriate
playing-speed; it should not be done after digitisation.
With other processes (notably noise-reduction, Chapter 3, and equalisation,
Chapters 5, 6 and 11) it may be necessary to do a close study of the relationship between
the transfer and processing stages. The analogue transfer stage cannot always be
considered independently of digital processing stage(s), because correct processing may
be impossible in one of the domains. For readers who need to know the gory details, most
digital processes are impotent to handle the relative phases introduced by analogue

39
equipment (section 2.11), and become impotent as zeroes or infinities are approached
(especially very low or very high frequencies).
To take this thought further, how do we judge that a digital process is accurate?
The usual analogue tests for frequency-response, noise, and distortion, should always be
done to show up unsuspected problems if they exist. But measuring a digital noise
reduction process is difficult, because no-one has yet published details of how well a
process restores the original sound. It may be necessary to set up an elementary “before-
and-after” experiment - taking a high-fidelity recording, deliberately adding some noise,
then seeing if the process removes it while restoring the original high-fidelity sound. The
results can be judged by ear, but it is even better to compare the waveforms (e.g. on a
digital audio editor). But what tolerances should we aim for? This writer’s current belief is
that errors must always be drowned by the natural background noise of the original - I do
not insist on bit-perfect reproduction - but this is sometimes difficult to establish. We can,
however, take the cleaned-up version, add more background-noise, and iterate the
process. Eventually we shall have a clear idea of the limitations of the digital algorithm,
and work within them.
Draft standards for the measurement of digital equipment are now being
published, so one hopes the digital audio engineering fraternity will be able to agree
common methodology for formal engineering tests. But the above try-it-and-see process
has always been the way in which operators assess things. These less-formal methods
must not be ignored - especially at the leading edge of technology!
One other can-of-worms is becoming talked about as I write - the use of “neural
methods.” This is a buzzword based on the assumption that someone’s expertise can be
transferred to a digital process, or that an algorithm can adapt itself until the best results
are achieved. You are of course free to disagree, but I consider such techniques are only
applicable to the production of service-copies which rely on redundancies in the human
hearing process. For the objective restoration of the power-bandwidth product, there can
be no room for “learning.” The software can only be acceptable if it always optimises the
power-bandwidth product, and that is something which can (and should) be the subject
of formal measurements. Ideally, any such assumptions or experiences must be
documented with the final recording anyway; so when neural techniques do not even
allow formal documentation, the objective character of the result cannot be sustained.

3.11 Digital recording media

Although I don’t regard it of direct relevance to the theme of my book, I must warn
innocent readers that conservation problems are not necessarily solved by converting
analogue sounds to digital media. Some digital formats are more transitory than analogue
formats, because they have shorter shelf-lives and less resistance to repeated use. One
paper assessed the shelf life of unplayed R-DAT metal-particle tapes as 23 years, and you
certainly don’t want to be cloning your entire collection at 22-year intervals! Your
digitisation process must therefore include careful choice of a destination medium for the
encoded sounds. I could give you my current favourite, but the technology is moving so
rapidly that my idea is certain to be proved wrong. (It could even be better to stick
everything on R-DAT now, and delay a firm decision for 22 years).
It is also vital to store digitised sounds on media which allow individual tracks and
items to be found quickly. Here we must outguess posterity, preferably without using
copyright software.

40
I shall ask you to remember the hassles of copy-protection (section 3.8 above), and
next I will state a few principles you might consider when making your decision. They are
based on practical experience rather than (alleged) scientific research.
(1) It is always much easier to reproduce widely-used media than specialist media.
It is still quite cheap to install machinery for reproducing Edison cylinders, because there
were over a million machines sold by Edison, and both enough hardware and software
survives for experience also to survive.
(2) The blank destination-media should not just have some “proof” of longevity
(there are literally thousands of ways of destroying a sound recording, and nobody can
test them all)! Instead, the physical principles should be understood, and then there
should be no self-evident failure-mechanism which remains unexplained. (For example, an
optical disc which might fade).
(3) The media should be purchased from the people who actually made them. This
is (a) so you know for certain what you’ve got, and (b) there is no division of
responsibility when it fails.
(4) Ideally the media (not their packaging) should have indelible batch-numbers,
which should be incorporated in the cataloguing information. Then when an example
fails, other records from the same batch can be isolated and an emergency recovery-
programme begun.
(5) On the principle “never put all your eggs into one basket,” the digital copy
should be cloned onto another medium meeting principles (2) to (4), but made by a
different supplier (using different chemicals if possible), and stored in a quite different
place.

So, after a long time examining operational strategy, we are now free to examine
the technicalities behind retrieving analogue signals from old media.

REFERENCES

1: anon, “SDMI chooses MusiCode from Aris to control Internet copying” (news item),
London: One To One (magazine), Issue 110 (September 1999), page 10.
2: ibid, pages 73-74 and 77.
3: Barry Fox, “technology” (article), London: Hi-Fi News & Record Review (magazine),
Vol. 44 No. 10 (October 1999), page 27.

41
4 Grooves and styli

4.1 Introduction

This chapter will be about “mechanical” sound recordings, in which the sound waves
were translated into varying waveshapes along the length of a spiral groove. These waves
are nearly always “baseband” - that is, the sound frequencies were recorded directly,
instead of being modulated onto a “carrier frequency” in some way. The principal
exception will be quadraphonic recordings using the CD-4 system, which I shall leave until
we explore spatial sounds in section 10.16.
A century of development lies behind sound recording with mechanical techniques.
The technology is unlike any other, since a frequency range of some ten octaves might be
involved, with signal-to-noise ratios in the upper-sixties of decibels. Thus the power-
bandwidth products of the best mechanical recordings can rival anything else analogue
technology has achieved.
In this chapter I shall be considering the best ways to extract the original sound
from disc or cylinder grooves. It will deal with the groove/stylus interface and also
electronic techniques for minimising surface noise and distortion. Thus we will have taken
the story to the point where electrical waveforms are coming out of a socket in
anticipation of adjustments to playing-speed or frequency-equalisation. At this socket we
will have extracted all the frequencies from the groove while minimising distortion, and
will have reduced the surface noise as far as we can; thus we will have recovered the
maximum power-bandwidth product. This is the fundamental limitation. After that we
may refine the sound if we want to.
It is an old adage that if the groove/stylus interface is wrong, it is pointless to rely
on electronics to get you out of your difficulties. I think this is something of an
oversimplification, although the basic idea is correct. My point is that we should really
work upon the groove/stylus interface at the same time as electronic noise reduction.
Some electronic treatments are very good at removing showers of loud clicks, for
instance. When this is done, it is then possible to choose a stylus to reduce the remaining
steady hiss. Had the stylus been chosen without the de-clicker in place, it might not have
been the one which gave the least hiss. This would be wrong, because the current state-
of-the-art is that it is much more difficult to approach the intended sound in the presence
of steady hiss.
Although I shall leave electronic treatments until the sections following 4.15, please
remember the two topics should actually go hand-in-hand. And once again, I remind you
of the importance of having the best possible originals, as we saw in section 2.6.
I shall be making great use of the term “thou” in this chapter. This means
“thousandths of an inch”, and has always been the traditional unit of measurement for
styli and grooves. In North America, the term “mil” has the same meaning. Although it
would be perfectly acceptable to use the metric term “microns”, where one micron is
one-millionth of a metre, specialist stylus-makers still speak in “thou”, so I shall do the
same. One thou equals 25.4 microns.
Two other terms I must define are “tracing” and “tracking.” I do so because many
textbooks confuse them. “Tracing” concerns the way a stylus fits into, and is modulated

42
by, the groove. “Tracking” concerns the alignment of the reproducer compared with the
geometry of the original cutting lathe.

4.2 Basic turntable principles

I assume my readers know what a turntable is, but they may not be aware that the
equivalent for cylinder records is called a “mandrel.” My first words must be to encourage
you to use “a good turntable (or mandrel) and pickup”. It isn’t easy to define this; but to
recover the best power-bandwidth product, the unit must have a lower background-noise
and a wider frequency-range than any of the media being played on it. This is something
we can measure, and does not need any “black magic.”
The unit should also have reasonably stable speed, although we shall come to a
technique in section 4.16 where this need not necessarily be to the highest standards. In
Chapter 4 we shall be studying the problem of establishing the correct playing-speed for
rogue records. These are in the minority, but at least one machine must have variable
playing-speed so this matter may be researched. (It may be necessary to have two or
more machines with different features to get all the facilities).
The unit should be insulated so it is not significantly vibrated by sounds from the
loudspeakers, loose floorboards, etc; this may mean fixing it to a massive mounting-plate,
and suspending it in compliant supports. All this is good engineering practice, and needs
no further comment from me.
Rather more difficult to define is that the unit should have “low colouration.” This
is much more a black magic subject. Basically it means that the wanted sounds are
reproduced without “hangover” - that is, the machinery should not contribute resonances
or sounds of its own which continue after the wanted sounds have stopped.
The main point about hangover is that the stylus and record themselves generate
it. As the stylus is vibrated by the groove, reaction-forces develop in the disc or cylinder
itself. It is important that such vibrations are quelled instantly, especially in the presence of
clicks and pops. When we remove these clicks and pops using electronic techniques, we
may not be able to get a clean-sounding result if hangover exists in the record itself or the
pickup arm.
To deal with the record first. We cannot prevent the vibrations being set up; we
can only ensure they are eliminated as quickly as possible. One method is to use a heavy
rubber turntable-mat in close contact with the back of the disc. The only suitable kind of
mat is the kind with a recessed area about four inches across in the middle, so the disc
makes good contact irrespective of whether the label-area is raised or not. As I say, it
helps if the mat is thick and heavy; the kind with lightweight ribs is pretty ineffective.
Some vibrations can also be attenuated by a clamp - a heavy weight placed over the
record label.
The Pink Triangle Turntable is designed without a mat at all. It is made from a
plastics material of the same specific gravity as vinyl. Any vibrations set up in the vinyl are
efficiently conducted away without being reflected back towards the pickup. Basically this
is a good idea, but it cannot give perfect reproduction unless the disc is in intimate contact
all over. You should be prepared to flatten warped records to cut down wow and tracing
errors, anyway; please see Appendix 1 for details. In my experience, Pink Triangle’s
method is as good as the best turntable mats, but not better.
I can see no virtue in “anti-static” turntable mats, especially ones that aren’t thick
and heavy. A more suitable method of neutralising electrostatic charges is to keep an

43
open saucer of lukewarm water near the turntable, perhaps with a sponge in it for
increased surface-area. Turntable lids are usually provided to keep dust off, and in my
opinion they are potential sources of hangover; but in cold frosty weather (when the
natural humidity is zero), they can define a microclimate where a high degree of humidity
may be established. Probably the optimum solution is a removable turntable lid, perhaps
with a motor-operated pickup lowering mechanism for use when the air is particularly dry
and the lid must be shut.
The Pink Triangle company also make a mechanism which brings eccentric records
on-centre. While it is important to play records on-centre, you may not need this if the
records are your own property. Frankly, attacking the centre-hole with a round file is just
as good; but after the disc has been positioned, you may need the clamp to hold it in
place through the subsequent operations.
To extend the above ideas to cylinders: Edison invented the “tapered mandrel,”
and the vast majority of cylinders have tapered orifices so the cylinder cannot be loaded
backwards. But the system has the disadvantage that the cylinder is subjected to tensile
stresses as it is pushed on, which may split it. After you have done this once, you know
how hard not to push! Seriously though, practicing with some sacrificial cylinders is a
good way of learning.
Another problem is that playing the cylinder may generate reaction stresses
between cylinder and mandrel. After a minute or two, these have the effect that the
cylinder will try to “walk off the mandrel,” by moving towards the narrow end in a series
of very small steps. My answer is very low-tech: a rubber-band pressed up against the
end of the cylinder! Eccentric or warped cylinders can often be helped with pieces of
paper packed between the ends of the cylinder and the mandrel. All these disadvantages
can be avoided with a machine which presses the cylinder between two contact points,
holding the cylinder by its ends (Ref. 1, for example).
To reduce the overhang difficulty, I recommend a plain solid tapered mandrel
made from anodised aluminium. I do not recommend anything compliant (analogous to a
turntable-mat), although Reference 1 involves just that! Once again, the ideal might be
two machines which provide all the features between them.

4.3 Pickups and other devices

“Pickup” is the term for a device which touches the groove as the record rotates, and
translates the sound recorded in the groove into analogue electrical signals. But before we
go any further, it is my duty to mention three other ways of doing the job, and why they
aren’t satisfactory for a sound archive. It is quite possible one of these techniques may
blossom into something suitable one day. In my judgement this hasn’t happened yet; but
you should know the principles, so you may judge them if they improve.
The first is the original principle invented by Edison, which is to couple the stylus to
something (usually a “diaphragm”) which vibrates, and radiates sound directly into the
air. Having stated this principle, it would normally be followed by some hundreds of pages
dealing with the design of diaphragms, horns, and other acoustico-mechanical devices for
improved fidelity and “amplification” ! I have put that last word in quotation-marks
deliberately. The laws of conservation of energy make it clear that you cannot get
“amplification” without taking energy from the rotating record, and the more you
attempt this, the more wear you cause to the original record. Obviously we don’t wish to
wear out our records; so I propose to abandon this principle, although there will be other

44
lessons for us further on. The remaining principles imply electronic amplification
somewhere, which does not stress the record itself.
The next principle is to play the record by looking at it with beams of light. There is
a fundamental difficulty here, because the sound waves recorded in the groove may be
both larger, and smaller, than typical wavelengths of light. Thus it is necessary to invent
ways of looking at the groove so both large and small waves are reproduced accurately.
At the time of writing the most successful of these is an analogue machine using infra-red
light modulated at an ultrasonic frequency, and demodulating it using radio-reception
techniques. This has three practical disadvantages over mechanical pickups. First, dirt is
reproduced as faithfully as the wanted sound. Second, the frequency response is limited at
the high end in a way which makes it difficult to cure the first problem. Third, the
hardware can only play grooves with straight-sided walls (which we shall come to later),
and only those made of something which will reflect infra-red light.
The third principle is to use an optical sensor. Here the idea is to measure the entire
recorded surface in three dimensions. This might be done by scanning it with a laser-light
detector at intervals of less than one micron, measuring the third dimension (“depth”) by
sensing when the laser light spot is in focus. This results in a vast file of some gigabytes of
numerical data, which might be processed into a digital recording of the original sound.

4.4 Conventional electrical pickup considerations

We now come to the way a pickup is carried across the record to minimise geometrical
sources of distortion. As this is not a textbook on conventional audio techniques, I shall
not describe tracking distortion in detail, but only the specific problems encountered by
present-day operators playing old discs.
It is generally assumed that discs were mastered by a cutter which moved across
the disc in a straight line, whereas most pickups are mounted on an arm which carries the
stylus in a curved path. In 1924 Percy Wilson did the basic geometrical study for
minimising distortion from this cause (Ref. 2). This study remains valid today, but
nowadays we use Wilson’s formulae slightly differently. Wilson originally sought to
minimise tracking error as averaged over the whole disc. Nowadays we minimise the
tracking error at the inner recorded radius, which is usually taken as two and a quarter
inches (56mm). There are two reasons for this: (1) the effects of tracking error are much
worse at the inner radius; (2) most music ends with a loud passage, and loud passages are
more vulnerable to tracking distortion.
The result of all this is that a pivoted pickup-arm should, in effect, have a bend in it
so that the cartridge is rotated clockwise with respect to an imaginary straight line joining
the stylus to the pivot. The exact angle varies with the arm’s length, but is usually in the
order of twenty degrees. In addition, the stylus should overhang the centre-pin of the
turntable by an amount which also varies with arm length, but is of the order of about
15mm. When a pickup arm is installed in this manner, minimum tracking distortion is
assured from conventional records. But operators should be aware that the “alignment
protractor” supplied with many pickup arms will not give the correct alignment for
unconventional records.
A pivoted arm is much more amenable to experimental work than a non-pivoted
one such as a parallel tracking arm. This doesn’t mean that either type is inherently
superior, only that one must use the right tool for the job. Many pivoted arms have an
oval-shaped base for the actual pivot, and the whole arm can be moved bodily towards or

45
away from the turntable centre. This enables you to neutralise tracking distortion when
the disc ends at an unusual radius. Coarsegroove 33rpm discs, for example, may finish
100mm from the centre; but tracking-distortion can be quite noticeable in these
circumstances because the sound waves are packed close together in a coarse groove. The
arm can be moved towards the turntable slightly to make the cartridge perfectly
tangential to the inner radius. On the other hand, many small 78rpm discs of the late
1920s and early 1930s were recorded much further in; the worst example I know ends
only 20mm from the centre hole. The tracking distortion is terrible under these conditions,
and may be reduced by moving the whole arm a centimetre or so away from the
turntable centre.
However, tracking distortion can be totally eliminated from conventional records
by means of a “parallel tracking arm” - a mechanism which carries the pickup across the
disc in a straight line. In practice this is difficult to achieve without causing other problems,
so parallel tracking arms are more expensive; but in this situation, the centre-line of the
cartridge should be aligned perpendicular to the direction of motion, and the path of the
stylus should pass through the centre-pin of the turntable.
However, I must report that, although a parallel tracking arm eliminates tracking
distortion on conventional records, it is practically impossible to do anything about
unconventional ones. In practice, these fall into two types. (1) Discs cut on a lathe where
the cutterhead was correctly aligned, but the cutting stylus was inserted askew. In this
case the tracking error is the same at all radii. (2) Discs cut on a lathe which carried the
cutter in a straight line not passing through the centre of the disc, but along a line parallel
to the radius; or, what comes to the same thing, discs cut on a lathe whose cutterhead
was not mounted squarely. In these cases the tracking error varies with radius. These
features result in various types of distortion which we shall examine in detail later.
Whether it is a pivoted arm or a parallel-tracker, the arm itself should not
contribute any significant resonances after being shock-excited by cracks and bumps. You
may need a long arm (such as the SME 3012) for playing outsized discs; but any long arm
will have noticeable resonances. Since the original Model 3012 was made, SME have an
upgrade-kit comprising a trough of silicone damping fluid. This greatly reduces the
resonances, but a later design (such as their Series V) is preferable for conventional-sized
discs. All other things being equal, a parallel-tracker has less metal which can vibrate;
experience with massive de-clicking operations tends to show that this type is better for
badly cracked and scratched records.
All cylinders were mastered with a straight-line movement parallel to the axis, so
this geometry should be followed in principle. All other things being equal, it is of course
irrelevant whether it is actually the pickup or the cylinder which moves; but other
considerations (such as groove-jumping) may force one method or the other. A machine
called “Ole Tobias” at the National Library of Norway combines the principle of having
no mandrel as mentioned at the end of 4.2, with a pivoted tonearm whose pivot is driven
in a straight line parallel to the axis. This would seem to combine all the advantages of
both; but I have no knowledge of the time it takes to get each cylinder running
concentrically.

46
4.5 Operational procedure for selecting a stylus

Because there are techniques for dealing with crackle and clicks, the maximum power-
bandwidth product comes from choosing a record with the best balance between the
basic hiss and the distortion due to wear. Although psychoacoustic tricks exist for
reducing hiss, there are currently no cures for wear, so priority should be given to an
unworn copy.
An experienced transfer operator will be able to choose the correct stylus simply by
looking at the grooves. I am afraid this is practically impossible to teach; the operator has
to look at both groove walls, the groove bottom, and the “horns” (if any; see section
4.6). A “point source” of light is needed (not fluorescent tubes), preferably in an
“Anglepoise” so you may choose different ways of looking at the surface. For selecting a
stylus, put the anglepoise behind your shoulder so you are looking down the beam of
light; then turn a disc about a horizontal axis between the two hands, while watching
how the overall amount of reflected light varies with the angle of the disc. I know that’s a
totally inadequate explanation, but I simply can’t do any better. Until you have learnt the
trick, you will be obliged to go through a procedure of consecutive feedback-loops to
identify the best stylus.
Thus you may have to compare two or more styli to see which gives the greater
power-bandwidth product. Unfortunately practical pickup cartridges cannot withstand
frequent stylus-changes (which in many cases can only be done by the manufacturer
anyway). So we must use exchangeable headshells, which will annoy the hi-fi buffs. Allow
me to deal with this objection first. Exchangeable headshells are inherently heavier than
fixed headshells. But the reduction of mass is only significant when dealing with warped
or eccentric records at very low playing forces; various types of distortion can occur if
there is any tendency for the pickup to move up and down or to and fro. Frankly, it is
much better to have flat concentric discs to start with! For an archive copy this is essential
anyway, as it is the best way to minimise speed inconsistencies.
So the professional transfer operator will have several cartridges mounted in
headshells ready for comparison. Sometimes, however, we find ourselves at the point of
diminishing returns. When we have got reasonably sensible noises out of the groove, it
may require a lot of work to make a not-very-significant improvement to the power-
bandwidth product. By the time I have unscrewed one head-shell and tried another, I find
I have forgotten what the first one sounded like. There are two cures: (1) Transfer one
version before changing the shell; (2) to have two pickup arms playing the same record
and switching between them (this is a better way, as it reduces wear-and-tear on the
headshell contacts).
To close the feedback loop, and expedite the choice of one stylus from dozens of
possibilities, we must learn the mechanisms involved and their effects upon the
reproduction. This chapter therefore continues with a look at the history of grooves and
styli. Whenever we come across a technique which is still applicable today, I shall interrupt
the history lesson and examine the technique in more detail. I am afraid this will mean a
rather zig-zag course for my argument, but I hope that sub-headings will allow you to
concentrate upon one strand or the other if you wish.

47
4.6 U-shaped and V-shaped grooves

I shall talk mainly about two kinds of groove – “U-shaped” and “V-shaped” - but I shall
not formally define these terms. I use them to differentiate between two philosophies for
playback purposes; you should not assume that all “V-shaped” grooves have straight
walls and sharp bottoms, for example. And recordings made at the dawn of sound
recording history do not fit either category.
Edison’s tinfoil phonograph did not cut grooves. It indented them in a sheet of
metal commonly known as “tinfoil.” The noise-level of the groove was determined
principally by the physical properties of the foil. It was virtually impossible to remove it
and replace it correctly without either corrupting the indentations or crinkling the sheet;
and there was inevitably a once-per-revolution clunk as the stylus crossed the seam where
the foil was wrapped round the mandrel. These features were largely responsible for the
eclipse of the phonograph as a practical recording machine during the years 1878 to
1887. They also explain why so few tinfoils survive today, and those in unplayable
condition.
Bell and Tainter’s “Graphophone” circumvented these difficulties by using pre-
shaped cardboard cylinders coated in a wax-like substance called “ozokerite.” Thus the
problems of coiling up the tinfoil, aligning it on the mandrel, and arranging for an
inoffensive seam, were avoided. But Bell & Tainter’s fundamental improvement was that
the groove was cut instead of indented. The recording machine was fitted with a stylus
which actually removed a continuous thread of ozokerite, leaving behind a fine clean
groove with much lower noise. (In parentheses, I add that the Graphophone used an
untapered mandrel. So there may be ambiguity about which is the start and which is the
end of the recording).
Edison’s “Improved Phonograph” of 1888 adopted the cutting idea, but he
favoured cylinders made of solid wax much thicker than the Graphophone’s layer of
ozokerite. It was therefore possible to erase a recording by shaving it off. This was much
better suited for dictation purposes, which is how both the Graphophone and the
Improved Phonograph were first marketed. I do not know the details of Graphophone
cutters, but I do know that Edison’s Improved Phonograph used a sapphire cutting-tool.
Sapphire is a jewel with a hardness greater than any metal. Anything less hard was found
to wear out quickly. This was the main reason behind the commercial failure of the
Graphophone, because a blunt cutter would not make a quiet groove.
Artificial sapphires were made for the jewelled bearings of watches. They were
cylindrical in form and smaller than a grain of rice, about one-hundredth of an inch in
diameter. To make a phonograph cutter, one end was ground flat and mounted so it
would dig end-on into the rotating wax. The sharp edge where the flat end met the
curved rim would be where the swarf was separated from the cylinder, leaving behind a
groove so smooth it would reflect light. In practice, the cutter would be tilted at a slight
angle, and the front face ground to a complimentary angle. This left a groove bottom
which wasn’t shaped like an arc of a circle, but an arc of an ellipse with a low degree of
eccentricity. This is what I mean when I talk about “U-shaped” grooves.
In the case of Edison machines, recordings were reproduced by another jewel, this
one deliberately “blunt” so it would not cut the wax again, but small enough to run along
the bottom of the groove. The vertically-modulated sound waves would cause the
reproducing stylus to be vibrated up and down as it pressed against the groove bottom,
and thus the sound would be extracted. Later developments resulted in playback styli
made to a specific diameter to fit the grooves, minimising noise and wear. Edison

48
established standards for his two-minute cylinders, his four-minute cylinders, his
“Voicewriter” dictation-machines, and his “Diamond” discs. Edison also showed that
minimum distortion occurred with a button-shaped playback stylus (the correct
geometrical term is an oblate spheroid). This was designed to sit across a plain groove
whilst remaining in contact all round, while its minor radius was sufficiently small to follow
the most intricate details of the recorded waveform.
Meanwhile, back in 1888, Emile Berliner was developing a quite different way of
recording sound. There were three fundamental differences. (1) He preferred discs to
cylinders, which gave him two advantages. His reproducing machines needed no
mechanism to propel the reproducing stylus, the disc itself would do it; and he could
mass-produce copies of his records like printing. (2) His styli vibrated side-to-side rather
than up-and-down. The groove walls therefore pushed the playback styli to and fro rather
than the unidirectional propulsion of the hill-and-dale (vertical cut) format. (3) He did not
cut grooves, but used an acid-etching process.
“Acid-etched” disc recordings, made between 1888 and 1901, therefore have
grooves of rather indeterminate cross-section. Partly because of this, and partly because
Berliner was competing with cylinder manufacturers on cost grounds, Berliner used
relatively soft steel reproducing needles and made his discs in relatively abrasive material.
The first few seconds of groove would grind down the tip of the reproducing needle until
it had its maximum area of contact, thereby ensuring the needle would be propelled by
the groove walls, while his machines avoided the cost of jewelled playback styli. On the
other hand, steel needles could only be used once; and this philosophy remained the
norm until the 1950s.
In 1901 Eldridge Johnson (founder of the Victor Company) adapted the wax-
cutting process to the mastering of disc pressings, so the groove now had consistent
cross-section throughout the side of the disc. For several decades they were usually U-
bottomed like hill-and-dale recordings. Although the abrasive nature of the pressings did
much to hide the advantages, the wax masters and the stampers had smoother surfaces
than acid-etched recordings, and today much of our restoration work consists of trying to
get back to the low noise-level of the wax masters.
The vast majority of such pressed records were played with steel needles. The only
exceptions were collections belonging to wealthier or more careful collectors, who used
“fibres” (see section 4.8).
In 1911 a British inventor, P. J. Packman, patented a new type of cutting stylus in
which a cylindrical sapphire had its axis perpendicular to the wax, rather than substantially
parallel to it. (Ref. 2). His aim was to cut deeper grooves. He wanted to pack more sound
into a given space, and reasoned that if one used hill-and-dale recording, one would not
have to leave space between the grooves for lateral modulation. By combining hill-and-
dale recording with a finer groove-pitch and the technique of an abrasive record to grind
a steel needle, he hoped to make inexpensive long-playing disc records; a couple of
hundred were actually published under the tradename “Marathon.” They were not a
success; however, the principle of Packman’s cutter was gradually adopted by the rest of
the sound recording industry.
There were several advantages to a relatively deep groove. The deeper it was, the
less likely the sound would be corrupted by scratches and dirt. Also a reproducing stylus
was less likely to “skid,” or to be thrown out of the groove by heavy modulation. These
advantages meant it was easier to accommodate louder sounds. There could be a greater
area of contact between stylus and groove, so there could also be less hiss as we shall see
in section 4.8.

49
If one tries to cut a groove of U-shaped cross-section which is impracticably deep,
the walls will become nearly vertical at the surface of the disc. A number of difficulties
come to light if this happens. During the cutting process, the swarf does not separate
cleanly, because material being forced up from the bottom of the groove causes shearing
action (rather than cutting action) at the top. Even if this problem were overcome, it
would be much more difficult to press or mould records from a negative with ridges of
near-semicircular shape. The material would adhere to the stamper rather than separate
cleanly, because of different coefficients of thermal contraction as stamper and material
cooled. When the groove walls are less than 45 degrees, the thermal contraction results in
the record being pushed away from the stamper; when it is greater than forty-five
degrees, the record tends to be gripped by the stamper.
Therefore the deepest possible groove can only be a V-shape rather than a U-
shape, with no part of the groove walls greater than forty-five degrees from the
horizontal. This therefore represents the ultimate practicable groove-shape to make use of
the advantages I have just described.
Nowadays, when most people have done applied mathematics at school, the idea
of a force being resolved into two components is commonplace. But evidently this wasn’t
the case in the first quarter of this century; it was thought grooves must have flat bottoms
if the record was to take the playing-weight of acoustic soundboxes (over a hundred
grams). Today we know that a V-shaped groove is equally capable of bearing such a
weight, since the force exerted upon each inclined groove wall can be resolved into
horizontal and vertical components, and the two horizontal components from each of the
two walls cancel. Packman’s groove worked this way, although he did not claim it as part
of his invention. During the first World War the English Columbia company adopted V-
shaped grooves for its records, I suspect largely because they had much worse surface
noise than their competitors at that time. But the mistaken idea of U-bottomed grooves
being inherently better remained the dominant philosophy until the early 1930s.
What forced the change was the advent of the auto-changer for playing a stack of
records without human intervention. Reproducing needles suddenly had to be capable of
playing eight consecutive sides. Less wealthy customers still used relatively soft steel
needles, so the records had to retain abrasive qualities to grind them until there was a
perfect fit - an early case of “downwards compatibility.” Only the very hardest stylus
materials would stand up to eight abrasive sides, and various forms of tungsten were
tried, followed eventually by the renaissance of jewels. To ensure the grooves would
always propel such styli backwards and forwards in the lateral plane, the walls had to be
in control. This was impossible so long as different records had U-bottomed grooves of
different sizes and the playback styli couldn’t adapt themselves. So the industry gradually
changed to V-shaped grooves cut by Packman-type cutters, a process which was
complete by 1945.
Although Packman’s patent shows a V-shaped cutter coming to a definite point,
sapphires of this design are very fragile. Sapphire is very hard and it resists compression
tolerably well, but it has little shear strength. Cutting a V-shaped groove with a sharp
bottom is practically impossible. Instead, the tip of the cutter is deliberately rounded, and
the resulting groove actually has a finite radius in its bottom. In 78rpm days this might be
anything between 1 thou and 2.5 thou, even in nominally V-shaped grooves. With the
introduction of microgroove, the bottom radius might be 0.3 to 0.7 thou. If we consider
mass-produced pressings, the radius tended to increase as the stamper wore, and greater
radii may be encountered in practice.

50
Before it became possible to copy a disc record electrically (in about 1930), a
factory might “restore” a negative by polishing it, so the pressing would have a groove
with a flat (or flattish) bottom, no matter what the original groove shape was. This was
done to clear up background-noise due to irregularities in the bottom of the groove,
which were reproduced loud and clear when played with a steel needle ground to fit.
Background-noise was certainly ameliorated, but the process was not without side-effects.
A steel needle would take longer to grind down, resulting in an extended period of wear;
and before modern stylus-shapes became available, such blunter styli could not trace the
high frequency detail.
To continue my history lesson: Cecil Watts invented the cellulose nitrate lacquer
recording blank in 1934. This comprised a layer of lacquer upon a sheet aluminium base,
and although there were many alterations in the detailed composition of the lacquer and
in the material used for the base, as far as I know cellulose nitrate was always the principal
constituent. His development accompanied rivals such as “gelatine”, “Simplat,”
“Permarec,” and others, but these were all aimed at amateur markets. Only “nitrate” was
adopted by professionals, because (when new) it had lower background-noise than any
other analogue medium, either before or since. (For some reason it was called “acetate”
for short, although as far as I know there was never a formulation relying on cellulose
acetate as its principal component. I shall call it “nitrate.”)
Nitrate was gradually adopted by the whole disc-recording industry to replace wax,
which was more expensive and too soft to be played back; the changeover was complete
by 1950. Wax cutters had had a plain sharp cutting edge (known, for some reason, as a
“feather edge.”) Packman-type sapphires with feather-edges could not withstand the
extra shear stresses of nitrate, so steel cutters were widely used for such discs. However it
was noticed they sometimes gave lower surface noise as they wore. There was a great
deal of hocus-pocus pronounced about this subject, until the New York cutting stylus
manufacturer Isabel Capps provided the correct explanation. The swarf was being
separated by the front face of the cutter, as intended; but when it was slightly blunt, the
following metal pushed the groove walls further apart and imparted a polishing action.
When this was replicated by adding a “polishing bevel” to a sapphire cutter, there was a
major improvement in background-noise and the sapphire was better able to withstand
the shear stresses at the same time. From the early 1940s sapphire cutters with polishing-
bevels became normal for cellulose nitrate mastering.
These polishing-bevels had the effect of increasing the minimum possible recorded
wavelength. Although insignificant compared with the losses of playing contemporary
grooves with contemporary styli, they caused a definite limit to the high-frequency
reproduction possible today.
Because the polishing-bevel pushed some of the nitrate aside, the result was that
miniature ridges were formed along the top edge of the groove walls, called “horns.” If
not polished off the “positive,” they are reproduced upon the pressed records, and when
you have learnt the trick of looking at them, the horns provide conclusive evidence that
nitrate was used rather than wax. We may need to know this when we get to section
4.11.
With microgroove recording it was necessary to adopt another technique to allow
small recorded wavelengths. The cutting stylus was heated by a red-hot coil of wire. The
actual temperature at the cutting-edge proved impossible to measure in the presence of
the swarf-removal suction, but it was literally like playing with fire. Cellulose nitrate is
highly inflammable, and engineers have an endless supply of anecdotes about the
resulting conflagrations. By almost melting the lacquer at the cutting-edge, it was found

51
possible to make the polishing-bevels much smaller and improve the high frequencies, to
reduce the background-noise of the master-lacquer, and to extend the cutter’s life at the
same time. (Ref. 3).
The final development occurred in 1981-2, when “direct metal mastering” was
invented by Telefunken. To oversimplify somewhat, this involved cutting a groove directly
into a sheet of copper. A diamond cutter was needed for this, and for a number of
reasons it was necessary to emulate the polishing action by an ultrasonic vibration of the
cutter. A decade later, more than half the disc-cutting industry was using the process.
This has been a grossly oversimplified account of what became a very high-tech
process; but I mention it because operators must often recognise techniques used for the
master-disc to ensure correct geometry for playback.

4.7 The principle of minimising groove hiss

We now come to the problems of reproducing sound from grooves with fidelity. This is by
no means a static science, and I anticipate there will be numerous developments in the
next few years. The power-bandwidth principle, however, shows us the route, and
quantitatively how far along the road we are. Most of what I shall say is applicable to all
grooved media; but to save terminological circumlocutions, I shall assume we are trying to
play a mono, lateral-cut, edge-start shellac disc, unless I say otherwise.
The irregularities in the walls of the groove cause hiss. These irregularities may be
individual molecules of PVC in the case of the very best vinyl LPs, ranging up to much
larger elements such as grains of slate-dust which formed a major constituent of early 78s.
The hiss is always the ultimate limit beyond which we cannot go on a single copy, so we
must make every effort to eliminate it at source. In fact, steady hiss is not usually the most
noticeable problem, but rather crackles and pops; but we shall see later that there are
ways of tackling those. It is the basic hiss that forms the boundary to what is possible.
The only way to reduce the basic hiss from a single disc is to collect the sound from
as much of the modulated groove walls as we can. It is rather like two boats on a choppy
sea; a dinghy will be tossed about by the waves, while a liner will barely respond to them.
Playing as much of the groove as possible will imitate the action of a liner. We can
quantify the effect. If our stylus touches the groove with a certain contact area, and we
re-design the stylus or otherwise alter things so there is now twice the area of contact, the
individual molecules or elements of slate dust will have half their original effect. In fact,
the hiss will reduce by three decibels. So, if the basic hiss is the problem, we can reduce it
by playing as much of the groove as we possibly can.
Please note that last caveat – “If the basic hiss is the problem.” That is an
important point. If the noise we hear is not due to the basic structural grain of the record,
this rule will not apply. Suppose instead that the record has been scratched at some time
in the past, and this scratch has left relatively large protruding lumps of material in the
walls of the groove. If we now attempt to double the contact area, the effects of the
scratch will not be diluted; to the first approximation, the stylus will continue to be driven
by the protruding lumps, and will not be in contact with the basic groove structure. Thus
the effects of the scratch will be reproduced exactly as before. To use our boat analogy
again, both the liner and the dinghy will be equally affected by a tidal wave.
So whatever we subsequently do about the scratches and clicks, our system must
be capable of playing as much of the groove as possible, in order to reduce the basic hiss
of an undamaged disc. I shall now consider some ways of achieving this ideal - which, I

52
must repeat, is not necessarily an ideal we should always aim at, because of other sources
of trouble.

4.8 “Soft” replay styli

I must start by making it quite clear that there are several quite different approaches we
might take. Nowadays we tend instinctively to think in terms of playing a mono lateral-
cut edge-start shellac disc with an electrical pickup fitted with a jewel stylus, but it is only
right that I should first describe other ways of doing it. The Nimbus record company’s
“Prima Voce” reissues of 78s on compact discs were transferred from an acoustic
gramophone using bamboo needles, and whatever your opinion might have been about
the technique, you could only admire the lack of surface noise.
For modern operators used to diamonds and sapphires, it is necessary for me to
explain the thinking behind this idea. I use the unofficial term “soft” needle for any stylus
which is softer than the record material. It forms a collective term for the needles better
known as “bamboo”, “thorn”, or “fibre”, but please do not confuse my term with so-
called “soft-toned” needles; I am referring to physical softness.
Most shellac lateral-cut discs were deliberately designed to be abrasive, because
they had to be capable of grinding down a steel needle. Microscopic examination would
then show that the grooves were populated with iron filings embedded in the walls; the
result was additional surface noise. From about 1909, quality-conscious collectors used
soft needles instead. (Ref. 4). They were obtained from various plants and treated in
various ways, but all worked on the same principle. A slice from the outer skin of a
bamboo, for example, was cut triangular in cross-section. Soundboxes with a triangular
needle-socket were obtainable. The needle could be re-sharpened by a straight diagonal
cut; you could do this with a razor, although a special hand-tool was easier to use. This
action left a point sharp enough to fit any record groove. “Bamboos” were favoured for
acoustic soundboxes. The smaller “thorn” needles for lightweight electrical pickups had to
be sandpapered or otherwise ground to a conical point. “Fibre” needles seems to be a
collective term for the two types. All such materials had much more hardness along the
grain of the fibre, and it was possible to achieve a really sharp point - as little as 0.5 thou
was often achieved. (Ref. 5). Sometimes specialist dealers provided needles impregnated
with various chemicals, and much effort was expended in getting the optimum balance
between lubricant (when the needle was relatively soft) and dryness (when the needle
was much harder, transmitted more high frequencies, and lasted longer in the groove).
At the outside edge of a shellac disc, a “soft” needle wore very rapidly until it was
a perfect fit for the groove - this happened typically within one revolution, so the parts of
the groove containing the music were not affected by wear. After that, the close fit with
the groove-shape minimised the wear and the hiss, as we saw in our boat analogy. By
having a very fine point with an acute included angle (semi-included angles of only ten
degrees were aimed for), the shape would fit the groove perfectly in the direction across
the groove, but would be relatively short along the groove. This was found empirically to
give less distortion. I am not aware that a printed explanation was ever given, although
clearly users were emulating Edison’s oblate spheroid of four decades earlier, and the
“elliptical” jewels of four decades later.
The hiss was also attenuated by the compliance of the needle, which however
affected the reproduction of wanted high frequencies (Ref. 6). However, greater
compliance meant less strain between needle and groove, so less wear at high frequencies

53
- exactly where steel and jewel styli had problems. Modulation of high amplitude would
sometimes cause the point to shear off a “soft” needle, but collectors considered this was
a small price to pay for having a record which would not wear out. Only one playing with
steel would damage the record, adding crackle, and would render further fibres useless by
chipping at the ends of the bundle of fibrous material. Collectors therefore jealously
guarded their collections, and did not allow records out of their hands in case the fibred
records became corrupted.
“Soft” needles were in common use through the 1920s and 1930s, and the pages
of The Gramophone magazine were filled with debates about the relative merits.
However these debates always concentrated upon the subjective effect; there is nothing
objective, and nowadays we find it difficult to tell which gives the optimum power-
bandwidth product. Soft needles were even tried on microgroove records in the early
1950s, evidently with some success (Ref. 7).
The motivation was, of course, to make contact with all the groove so as to reduce
the hiss, as we saw earlier. Reference 6 quite correctly described the disadvantages of
thorn needles (the attenuation of high frequencies and the risk that the tip might shear
off), but it perversely did not say what the advantages were. To archivists nowadays,
those two disadvantages are less important. We can equalise frequency response
aberrations (so long as they are consistent), and we can resharpen the stylus whenever
needed (as Nimbus did in the middle of playing their records, editing the resulting sections
together).
A number of experiments by myself and others may be summarised as follows. A
“soft” needle undoubtedly gives lower surface noise than any other, although the
differences are less conspicuous when the high-frequency losses due to the needle’s
compliance are equalised. (The response starts to fall at 3 or 4 kiloHertz when a bamboo
needle is used in an acoustic soundbox). This is apparently because, with “hard” styli, in
most cases we are not playing the basic hiss of the record, but the damage; this does not
mean reduced noise. But damage shears bits off a “soft” point, and is then reproduced
more quietly - a sort of mechanical equivalent of the electronic “peak clipper.” The
reproduction is particularly congested at the end of the disc, because the needle has a
long area of contact along the groove. Scientific measurements of frequency response and
distortion give inconsistent results, because the tip is constantly changing its shape; thus I
must regretfully conclude that a soft needle is not worthy of a sound archive. Also the
process requires a great deal of labour and attention. However, it shows that better results
are possible, and we must try to emulate the success of the soft needle with modern
technology.
There seems to be only one case where the soft needle is worth trying - when the
groove is of indeterminate shape for some reason - perhaps an etched “Berliner” or a
damaged cutter. Grooves like this sometimes turn up on nitrate discs, and are not
unknown in the world of pressings; operators derisively describe them with the graphic
expression “W-shaped grooves.” Obviously, if the groove is indeed W-shaped, or
otherwise has a shape where a conventional stylus is useless, then a soft needle is worth
trying.
I should like to conclude this topic by advising potential users to experiment. It is
comparatively easy to construct soft needles to one’s own specification, and there seems
to be little harm to be caused to shellac 78s. I should also like to see a stereo pickup which
takes such needles; in section 0 we shall be examining another method of reducing noise
which depends upon the record being played with a stereo pickup. But I should like to
remind you that if the needle develops a “flat” (which means it has a long area of contact

54
in the dimension along the groove), various forms of distortion become apparent on
sounds which cause sharp curves in the groove. So if you are doing any serious
experimenting, I recommend you make use of an Intermodulation-Distortion Test Disc. In
America the obvious choice is RCA Victor 12-5-39, and in Britain EMI JH138. References
8 and 9 give some hints on how to use them and the results which may be expected.

4.9 “Hard” replay styli

I count steel needles as being neither “soft” nor “hard.” They were soft enough to fit the
groove after a few turns when played with pickups whose downwards force was
measured in ounces, but the knife-edges worn onto the tip during this process cut into
the waveform and caused distortion. They are not used today for this reason, and I can
say with confidence that sacrificial trials of steel needles upon shellac discs do not give a
better power-bandwidth product. If anyone wants to confirm my experiment, I should
mention that “soft-toned” steel needles (soft in volume, that is) have extra compliance.
This gives high-frequency losses like fibres, which must be equalised for a fair trial.
On the contrary, from the end of the second World War onwards, it has been
considered best practice to use hard styli, by which I mean styli significantly harder than
the record material. Styli could be made of sapphire, ruby, or diamond; but I shall assume
diamond from now on, because sapphires and rubies suffer appreciable amounts of wear
when playing abrasive 78s, and do not last very long on vinyl. The cost of a stylus
(especially a specialist shape) is now such that better value is obtained by going for
diamond right from the start. In the author’s experience there are two other advantages.
Diamonds are less likely to be shattered by the shocks imparted by a cracked disc. Also
diamonds can play embossed aluminium discs; sapphire is a crystalline type of aluminium
oxide, which can form an affinity with the sheet aluminium with mutual destruction.
At this point I will insert a paragraph to point out the difficulties of worn “hard”
styli. The wear changes a rounded, and therefore inherently “blunt”, shape, into
something with a cutting edge. In practice, U-bottomed grooves cause a worn patch
whose shape approaches that of the curved surface of a cylinder; V-bottomed grooves
cause two flats with plane surfaces. Where these geometrical features intersect the
original curved surface of the stylus, we get an “edge” - a place where the stylus has a
line separating two surfaces, rather than a curved (blunt) surface. This can (and does) cut
into record grooves, particularly where the groove contains sharp deviations from the line
of an unmodulated spiral. Thus we cause distortion and noise on loud notes. The damage
is irreversible. Therefore it is better to use diamond tips, which develop such cutting edges
less easily. The increased cost is greatly outweighed by the increased life. Inspection with
a 100-diameter microscope and suitable illumination is sufficient to show the
development of flats before they become destructive.
The “bluntest” shape, and therefore the one least likely to cause wear, is the
sphere. “Spherical” tips were the norm from about 1945 to 1963. The spherical shape
was ground onto the apex of a substantially-conical jewel mounted onto the armature or
cantilever of an electrical pickup. Being spherical, there were relatively few problems with
alignment to minimise tracing distortion (section 4.10), so long as the cantilever was
pointing in about the right direction and there was no restriction in its movement. The
spherical shape gave the maximum area of contact for a given playing-weight. So
acceptable signal-to-noise ratio and reasonable wear was achieved, even upon the earliest
vinyl LPs with pickups requiring a downward force of six grams or more.

55
From 1953 to 1959, the British Standards Institution even went so far as to
recommend standardised sizes of 2.5 thou radius for coarsegroove records and 1 thou for
microgroove records. This was supposed to ensure disc-cutting engineers kept their
modulation within suitable limits for consumers with such styli; it did not directly influence
the dimensions of grooves themselves. However, the idea had some difficulties, and
neither recommendation lasted long. You may often find older records needing larger styli
(particularly microgroove from the Soviet Union). Something larger than 1 thou will be
absolutely vital for undistorted sound here.
But first I shall mention the difficulties for coarsegroove records. We saw in section
4.6 that manufacturers were forced to change to V-shaped grooves to enable harder
needles to last in autochangers. 2.5-thou spherical styli could be relied upon to play such
V-shaped grooves successfully, but they were often troublesome with older U-shaped
grooves. When you think about it, a hard spherical tip is very likely to misbehave in a U-
shaped groove. If it is fractionally too small, it will run along the bottom of the groove and
not be propelled by the laterally-modulated groove walls at all; the result is noisy
reproduction and distortion at low levels. If it is fractionally too large, it will not go
properly into the groove at all, but sit across the top edges. The result, again, is noisy; but
this time the distortion occurs on loud notes and high-frequency passages. As long as
customers could only buy spherical tips, the only way forward was to buy tips of different
diameters. Enthusiasts had a range of spherical-tipped styli such as 2.5-thou, 3-thou, 3.5-
thou, and 4-thou, specifically for playing old U-shaped grooves. We saw in section 4.6
that U-shaped grooves had, in fact, the cross-section of a ellipse with a low degree of
eccentricity. It was found that a range of styli with differences of 0.5 thou, could trace
nearly all such grooves. It was also found that for modern coarsegroove discs with V-
shaped grooves and high levels of modulation (or high frequencies), smaller tips were
helpful, such as 2-thou and 1.5-thou.
For microgroove discs, the recommended 1-thou tip was found unsatisfactory for a
different reason. Pickup development was rapid, aided by Professor Hunt’s papers about
the effects of pickup design upon record-wear. These showed the advantages of high
compliance, high areas of contact, and low effective tip-mass. Below certain limits in these
parameters, Professor Hunt showed that pickups would work within the elastic limits of
vinyl and cause no permanent wear (although instantaneous distortions could still occur).
(Refs. 11 and 12). Great efforts were made to approach the ideals laid down by Professor
Hunt, which the introduction of stereo and the high volumes of pop discs did little to
hinder. By the end of the 1950s it was apparent that greater fidelity could be achieved
with smaller spherical tips, 0.7 thou or 0.5 thou. Although such styli often “bottomed” on
older records, they could trace the finer, high-frequency, details of newer discs with less
distortion; thus more power-bandwidth product was recovered. There was increased disc
wear due to the smaller area of contact, but this was soon reduced by improvements in
compliance and effective tip-mass.
There was another consideration in the days of mono, which is of less importance
now. Consider a spherical stylus playing a lateral-cut mono groove with loud modulation.
Where the groove is slewed, its cross-section also narrows. To overcome this, the stylus
must also rise and fall twice a cycle - in other words, it must possess vertical compliance.
Even if we are only interested in the horizontal movement, the centre of the spherical
stylus tip does not run exactly along the centre-line of the groove. The resulting distortion
is called “pinch effect distortion,” and permanent deformation of a groove was caused if
a stylus had low vertical compliance.

56
For years two sources of distortion tended to cancel each other. The harmonic
distortion which resulted from the large tip being unable to trace the fine detail was
almost exactly opposite to the harmonic distortion caused when a massive stylus modified
its course by deforming the groove wall. (Ref. 13). The fact that two types of distortion
neutralised each other was often used deliberately, but at the cost of permanent damage
to the groove. This explains why so many “pop singles” could be recorded at such high
volumes. If you don’t have a copy in unplayed condition, your only chances of getting
undistorted sound are to use a large tip with low vertical compliance, or find another
version of the same song.

4.10 Stereo techniques

To carry a stereo recording, a disc has to be capable of holding two channels of sound in
accurate synchronism. Several different methods were tried at different dates. In the
1930s both Blumlein in Britain and the Bell Labs engineers in America tried cutting the
left-hand sound as lateral modulation and the right-hand channel as vertical modulation.
This worked, but the two systems have different distortion characteristics. So the
reproduced sound had asymmetrical distortion, which was very noticeable because this
cannot occur in nature. Arnold Sugden of Yorkshire England attempted the same thing
using microgroove in the years 1955-6 (Ref. 14). Meanwhile Cook Laboratories in
America recorded stereo by using two lateral cutters at different radii, an idea taken up by
Audio Collector and Atlantic. It was found that the inevitable tracking errors of pivoted
pickup arms caused small time-shifts between the two channels, which were very
noticeable on stereo images. Back in the UK, Decca tried an ultrasonic carrier system. One
channel was recorded using conventional lateral recording, while the other was
modulated onto an ultrasonic 28kHz carrier-wave, also cut laterally.
But the solution ultimately adopted was one originally patented by Blumlein,
although not actually used by him as far as I know: to record the sum of the two channels
laterally, and their difference vertically. Not only do the two channels have symmetrical
distortion characteristics, but the record has the advantage of “downwards compatibility,”
so a mono record will reproduce in double-mono when played with a stereo cartridge.
This is geometrically equivalent to having one channel modulated at an angle of 45
degrees on one wall of a V-shaped groove, and the other at right-angles to it upon the
other groove-wall. The convention adopted was that the wall facing the centre of the disc
should carry the right hand channel, and the one facing away from the centre should
have the left-hand channel. This standard was agreed internationally and very rapidly in
April 1958, and I shall be assuming it from now on. The other (very rare) systems amount
to “incunabula.”
V-shaped grooves were standard by now, as we have seen. There were no
immediate consequences to the design of hard styli, except to accelerate the trend
towards 0.5 thou sphericals and ellipticals (which form the topic of the next section) as
part of the general upgrading process.
But a new source of distortion rapidly became noticeable – “vertical tracking
distortion.” Cutters in the original Westrex 3A stereo cutterhead, and its successor the
model 3B, were mounted on a cantilever whose pivot was above the surface of the
master-disc. So when “vertical” modulation was intimated, it wasn’t actually vertical at
all; it was at a distinct angle. In those Westrex cutterheads the angle of the cantilever was
twenty-three degrees, while in another contemporary cutterhead (the Teldec) there was

57
no cantilever at all, so when the cutter was meant to be moving vertically, it was moving
vertically.
So besides the various tracing and tracking distortions known from lateral records,
there was now a new source of trouble. When a perfect vertical sine wave is traced by a
cantilever, the sine wave is traced askew, resulting in measurable and noticeable
distortion. This is exactly analogous to the “tracking distortion” on lateral modulation
when you play the groove with a pivoted arm (section 4.4), so the phenomenon is called
“vertical tracking distortion.” The solution is to ensure that cutter cantilevers and pickup
cantilevers operate at the same angle. It proved impossible to design a rugged stereo
pickup without a cantilever, so the angle could not be vertical. Some years of research
followed, and in the meantime non-standard stereo LPs continued to be issued; but the
end of the story was that a vertical tracking angle of fifteen degrees was recommended.
(Ref. 15).
The first difficulty was that the actual physical angle of the cantilever is not
relevant. What is important is the angle between the tip of the stylus and the point (often
an ill-defined point) where the other end of the cantilever was pivoted. Furthermore,
variations in playing-weight and flexing of the cantilever at audio frequencies had an
effect. All this took some time to work out, and it was only from about 1964 onwards
that all the factors were understood, and a pickup could be guaranteed to have a fifteen-
degree vertical tracking angle at high frequencies (which is where the worst of the trouble
was). Unfortunately, by 1980 a gradual drift had become apparent among pickup makers,
if not disc cutting engineers; the average angle was over twenty degrees.
Similar problems applied to cutting the master-disc. Some designs of cutter proved
impossible to tilt at the required angle. And more research was needed because of a
phenomenon known as “lacquer-springback.” We saw in section 4.6 that cellulose nitrate
lacquer discs gradually took over for disc mastering, the changeover years being roughly
1936 to 1948. It was found that elasticity of the lacquer also caused some vertical
tracking error, because after the cutter removed a deep section of groove, the lacquer
tended to creep back under its own elasticity. This effect had not been noticed before,
because the springback was consistent for lateral cutting (with constant groove depth),
and for wax (which was not elastic). But the vertical tracking error from lacquer alone
might be twenty degrees or so. It varied with the make of lacquer, the size of the
polishing bevels, and the temperature of the cutter. When this effect was added to the
twenty-three degrees of the Westrex cutterheads, or the fifteen degrees of the proposed
standard, major redesigns were needed in three dimensions. The Westrex 3D, the
Neumann SX68, and the Ortofon cutterheads were the result. It proved possible to
modify the Teldec by chamfering off a bottom edge and tilting it (to oversimplify greatly);
thus all discs mastered after late 1964 should be to the fifteen-degree standard. But we
should be prepared to mess about when playing stereo discs mastered before 1964. The
best technique is to mount the pickup rigidly in its headshell, and arrange for the turntable
to swing in gimbal mountings beneath it while listening to the vertical component of the
sound, choosing an angle for minimum distortion.
A new type of cutting stylus was introduced in 1964 called the “Cappscoop.” (Ref.
16). This was specifically intended to make the lacquer-springback more consistent, giving
straighter groove-walls; but I have no experience of it or its results.

58
4.11 “Elliptical” and other styli

I have said several times that difficulties were caused when styli failed to trace the smaller
wiggles in grooves, but I have not yet formally mentioned the solution. The smaller the
zone of contact, the more accurately fine wiggles can be traced; but with small spherical
tips the result is generally an increase in hiss and an increase in disc wear, because the
contact takes place over a smaller area. Both problems can be ameliorated if we use a “bi-
radial” stylus - that is, with a small size in one dimension and a large size in another.
Edison’s “button-shaped” stylus was one solution, and a sharp-tipped fibre was another.
The so-called “elliptical” stylus was a third.
This is really only a trivial modification of Edison’s idea. Edison made an oblate
spheroid sit across the groove; the “elliptical stylus” comprises a conical tip rounded, not
to a spherical shape, but to an ellipsoidal shape. When either Edison’s or an ellipsoidal
stylus sits in a V-shaped groove, the effect is the same. If you draw a horizontal cross-
section of the stylus at the level of the points of contact, both styli are shaped like an
ellipse; hence the shorter but inaccurate term, “elliptical stylus.” Historically, the Ferranti
ribbon pickup of 1948 was the first to be marketed with an elliptical sapphire stylus (Ref.
17), followed by a version of the Decca “ffrr” moving-iron pickup. Decca had been
cutting full frequency-range coarsegroove shellac discs for four years, but towards the
centre of the record the high frequencies were so crammed together that spherical styli
could not separate them. An elliptical stylus not only extracted them better, but did so
with less distortion.
In the case of Decca’s stylus, the “dimensions” were 2.5 thou by 1.0 thou. By
convention, this means that if you look at the horizontal cross-section at the point where
the jewel sits across a V-shaped groove with walls of slope 45 degrees, the ellipse has a
major axis of 5 thou (twice 2.5 thou) across the groove, and 2 thou (twice 1.0 thou) along
the groove. It is also understood that the third dimension, up and down in the groove,
also has a radius of 2.5 thou, but this axis may be slightly off vertical. When we speak of
the “dimensions of an elliptical stylus,” this is what we mean.
This turns out to be a useful compromise between a small spherical tip and a large
spherical tip. In the example given, the stylus will follow the groove wiggles as
satisfactorily as a 1 thou tip, while riding further up the groove walls (so there is less risk
of the stylus running along the curved bottom of the groove, or hitting any noisy debris in
the bottom). The disadvantage is that, although the area of contact is much the same, the
ability to trace smaller wiggles can mean greater accelerations imparted to the stylus, and
greater risk of disc wear on loud passages. Your organisation may like to consider the
policy of using spherical tips for everyday playback purposes, and more sophisticated
shapes for critical copying.
Reduced distortion can only be achieved if the major axis of the ellipse links two
corresponding points of the opposite groove walls. It has been shown that the major axis
has only to be in error by a few degrees for the reduction in distortion to be lost. Thus the
pickup must be aligned for minimising both tracking and tracing distortions, particularly
on inner grooves (section 4.4). The conventional alignment procedure assumes that the
edges of the cutter which cut the two groove walls were oriented along the disc’s radius.
This was nearly always the case on discs mastered on cellulose nitrate, or the swarf
wouldn’t “throw” properly; but it is not unknown for wax-mastered discs to be
substantially in error. A feather-edged cutter would cut a clean groove at almost any

59
angle. It may be necessary to “misalign” the arm, or the cartridge in the headshell, to
neutralise the recorded tracing distortion.
At the time elliptical tips became popular, hi-fi enthusiasts were encouraged to
spend time aligning their record-playing decks for optimum performance. This generally
meant balancing record-wear against quality, and if you didn’t want to damage any
records in your collection, you needed to investigate the issues very thoroughly. But if you
do not play vinyl or nitrate very much, most of the risk of wear can be circumvented by
playing and transferring at half-speed.
Elliptical styli did not become commonplace until the mid-1960s. In the meantime,
an attempt was made to reduce tracing distortion by pre-distorting the recorded groove.
The RCA “Dynagroove” system was designed to neutralise the tracing distortion which
occurred when a 0.7 thou spherical tip was used. (Ref. 18). So presumably that’s what we
should use for playing “Dynagroove” records today. But the “Dynagroove” system was
also combined with a dynamic equalizer supposed to compensate for the Fletcher-
Munson curves (a psychoacoustic phenomenon). The rationale behind this was essentially
faulty, but the characteristics were publicly defined, and can be reversed. (Ref. 19)
If an elliptical tip does not have its vertical axis at the same angle as the vertical
tracking angle, a phenomenon known as “vertical tracing distortion” occurs. This doesn’t
occur with spherical tips. I suspect the simultaneous existence of “vertical tracking” and
“vertical tracing” distortion was responsible for the confusion between the words, but the
terminology I have used is pedantically correct. Vertical tracing distortion can occur with
mono lateral-cut discs, under extreme conditions of high frequencies and inner diameters.
To put it in words, if the minor axis of the elliptical tip is tilted so that it cannot quite fit
into the shorter modulations of the groove, results similar to conventional tracing
distortion will occur. John R. T. Davies had some special styli made to play cellulose nitrate
discs. These suffered from lacquer-springback even when the recording was made with a
mono cutterhead, and for some reason the surface noise is improved by this technique as
well. But a turntable in gimbals seems equally effective so long as the clearance beneath
the cartridge is sufficient.
I said earlier that Edison’s oblate spheroid was equivalent to an elliptical in a V-
shaped groove; but it’s not quite the same in a U-shaped groove. An elliptical will have
the same difficulties fitting a U-shaped groove as a spherical, because looking in the
direction along the groove, it appears spherical. The problem was solved by the
“Truncated Elliptical” tip, a modern development only made by specialist manufacturers.
It’s an “elliptical” shape with the tip rounded off, or truncated, so it will always be driven
by the groove walls and never the bottom. This shape is preferred for the majority of
lateral coarsegroove records. (It even gives acceptable, although not perfect, results on
most hill-and-dale records).
Although a range of sizes is offered, it is usually only necessary to change to avoid
damage on a particular part of a groove wall, or to play lateral U-shaped grooves which
have such a large radius that even a truncated tip is too small. Truncation can reduce the
contact area and increase disc wear. Fortunately it is hardly ever needed for vinyl or
cellulose nitrate records, which nearly always have V-shaped grooves.
We now reintroduce the lessons of “soft” styli, which had a large area of contact
giving less hiss from particles in the pressed disc. Electronic synchronisation techniques
permit us to play grooves with several styli of different sizes and combine the results.
Thus, given a family of truncated ellipticals of different sizes, we emulate fibre needles
without their disadvantages. I shall say more about this in section 0.

60
In the microgroove domain, the success of the elliptical stylus stimulated more
developments, which are known collectively as “line-contact” styli. There were several
shapes with different names. The first was the “Shibata” stylus, introduced in 1972 for
playing the ultrasonic carriers of CD-4 quadraphonic discs (Section 10.16). The idea was
to pursue lower noise, better frequency response, or lower wear, (or all three), by making
contact with more of the groove walls.
But all line-contact styli suffer the same disadvantage. If the line of contact is not
exactly correct - parallel to the face of the cutting stylus in the horizontal plane and
fifteen degrees in the vertical - tracing distortion becomes very obvious. When everything
is right they work well; but when anything is slightly misaligned, the result is
disappointing. In 1980 an article in Hi-Fi News listed some of the types of line-contact
stylus, mentioning that fundamentally faulty manufacturing principles and bad finish were
adding to the difficulties. The author advocated the new “Van den Hul” stylus as being
the solution; but a review of the very first such cartridge in the very same issue revealed
that it had more distortion than half-a-dozen others. That review seems to have killed the
idea for widespread use.
The trouble is that variations in the lacquer-springback effect and the tracking
distortions of pivoted pickup arms made the ideal impossible to achieve without much
fiddling. Cartridges with line-contact styli were expensive and delicate, and hi-fi buffs
preferred fixed headshells, so fiddling was not made easier. So perfect reproduction was
hardly ever achieved. It is significant that professionals have never used them. From the
archival point of view, there is little need; most master tapes of the period still exist, and
the subject matter is often available on compact digital disc. But clearly there is an avenue
for exploration here. The reproduction of some older full-range records might well be
improved, so for a general article on line-contact styli I refer you to Reference 20.

4.12 Other considerations

The above history should enable you to choose suitable styli and playing-conditions for
yourself, so I do not propose to ram the points home by saying it all again. Instead, I
conclude with a few random observations on things which have been found to improve
the power-bandwidth product.
Many coarsegroove discs with V-shaped grooves have bottom radii which are
smaller than the stylus sizes laid down by the British Standards Institution. Try a drastically
smaller tip-radius if you can, but learn the sound of a bottoming stylus and avoid this. Not
only does a small radius minimise pinch-effect and tracing distortions, but the bottom of
the groove often survives free from wear-and-tear. This particularly applies to cellulose
nitrates and late 78s with high recorded volumes. Indeed, these is some evidence that
record companies did not change the sapphire cutter between a microgroove master and
a 78 master. Eventually you will train your eye to tell the bottom radius of a groove,
which will cut down the trial-and-error.
On the other hand, it sometimes happens that an outsized stylus is better. This is
less common, because (all other things being equal) you will get increased tracing
distortion, and there will be greater vulnerability to noise from surface scratches. But just
occasionally the wear further down in the groove sounds worse. For various reasons, it
seems unlikely we shall ever be able to counteract the effects of wear, so evasive action is
advised. You can then concentrate upon reducing the distortion with elliptical or line-
contact styli, and the noise with an electronic process.

61
Although my next point is not capable of universal application, there is much to be
said for playing records with downward pressures greater than the pickup manufacturer
recommends. To reduce record-wear, an audio buff would set his playing-weight with a
test disc such as the Shure “Audio Obstacle Course”, carrying loud sounds which might
cause loss of groove contact. He would set his pickup to the minimum playing-weight to
keep his stylus in contact with the groove walls at the sort of volumes he expected
(different for classical music and disco singles!), thereby getting optimum balance
between distortion and wear. But nowadays, little wear is caused by higher playing
weights; most is caused when the grooves vibrate the stylus, not by the downward
pressure. There can be several advantages in increasing the downward pressure for an
archival transfer. The fundamental resonant frequency of the cantilever is increased
(according to a one-sixth power law - Ref. 21), thereby improving the high frequency
response. Clicks and pops are dulled, partly because the stylus can push more dirt aside,
and partly because the cantilever is less free to resonate. But most important of all, the
stylus is forced into the surface of the disc, thereby increasing the contact area and
reducing the basic hiss. Obviously the operator must not risk causing irreparable damage
to a disc; but if he is sufficiently familiar with his equipment, he will soon learn how far to
go whilst staying within the elastic limits of the medium.
Shellac discs seem practically indestructible at any playing-weight with modern
stereo pickup cartridges. Modern pickup arms are not designed for high pressures, but a
suitably-sliced section of pencil-eraser placed on top of the head-shell increases the down-
force with no risk of hangover. Pressures of six to ten grams often give improved results
with such discs; special low-compliance styli should be used if they are available. With
ultra-large styli, like those for Pathé hill-and-dale discs, it may even be necessary to jam
bits of pencil-eraser between cantilever and cartridge to decrease the compliance further;
twenty to thirty grams may be needed to minimise the basic hiss here, because the area of
contact is so large.
Records should, of course, be cleaned before playback whenever practicable (see
Appendix 1). But there are sometimes advantages in playing a record while it is wet,
particularly with vinyl discs. Water neutralises any electrostatic charges, of course; but the
main advantages come with discs which have acquired “urban grime” in the form of
essence-of-cigarette-smoke, condensed smog, and sweaty fingerprints. Also, if previous
owners have tried certain types of anti-static spray or other cleaning agents relying upon
unconventional chemicals, there may be a considerable deposit on the groove walls which
causes characteristic low-volume sounds. Conventional cleaning does not always remove
these, because the sludge gets deposited back in the grooves before the record can be
dried. Unfortunately it is impossible to give a rule here, because sometimes cleaning
makes matters worse (particularly with nitrates - it may be essential to transfer each disc
twice, once dry and once wet, and compare the results of the two transfers).
Centrifugal force often makes it difficult to play 78rpm discs wet. But for slower-
speed discs, distilled water may be spread over the surface while it plays, perhaps with a
minute amount of photographic wetting agent. The liquid can be applied through the
outer casing of a ballpoint pen with the works extracted; this can be used as a pipette to
apply the liquid, and as a ruler to spread it. Some types of disc have a “vinyl roar” which
is caused when the stylus runs across the surface and excites mechanical resonances
within the plastic. Although a proper turntable-mat and centre-clamp should eliminate the
effect on most records, the liquid also helps. However, some transfer engineers have
reported that dry playing of discs previously played wet can reveal a subsequent increase
in surface noise. The author accepts no responsibility for damage to record or pickup!

62
I deliberately concentrated upon laterally-modulated records from section 4.7
onwards, but I shall now deal with a specific problem for hill-and-dale records. It is
essential to take vertical tracking and vertical tracing into account of course, and strike a
compromise between tracing distortion (caused by a large area of contact) and hiss
(caused by a small area of contact). Even so, much even-harmonic distortion may remain,
and in many cases this will be found to be recorded in the groove. The reason for this will
be dealt with in section 4.15, where we look at electronic techniques for improving the
power-bandwidth product.
Finally, the archivist should be aware that the metal intermediate stages in the disc-
record manufacturing process – “master”, “mother” and “stamper” - sometimes survive.
Since these do not contain abrasives, the power-bandwidth product is usually better. I
have no experience in playing metalwork myself, but a consensus emerged when I was
researching this manual, which was that most people preferred to play fresh vinyl
pressings rather than metal. There are a number of difficulties with metal - it is usually
warped and lacks a proper-sized central hole, the nickel upsets the magnetic circuit of the
pickup, you can have only “one bite of the cherry” whereas you may have several vinyl
pressings, etc. However, as vinyl pressing plants are decommissioned, it will become
increasingly difficult to get fresh vinyl pressings made, and the risk when a unique
negative is clamped in a press by an inexperienced worker will increase. Until sound
archives set up small pressing-plants, I think we are more likely to be playing metalwork in
the future.
Pressing factories usually had the wherewithal to play a metal negative (with ridges
instead of grooves), if only to be able to locate clicks or noise. The turntable must rotate
backwards (see section 4.13), and the stylus must obviously have a notch so it can sit
astride the ridge. Top quality isn’t essential for factory-work; it is only necessary to locate
problems without having to examine every inch of ridge under a microscope. The Stanton
company makes a suitable stylus for their cartridges. In effect, it comprises two ordinary
diamond tips side-by-side on the same cantilever. I am not aware that there are any
options over the dimensions, so this could conceivably give disappointing results; but I
must say the few I’ve heard sounded no worse than vinyl, and often better.

4.13 Playing records backwards

I shall now continue with a couple of “hybrid” topics. They combine mechanical
techniques with electronic techniques. After that, the remaining sections will deal with
purely electronic signal-processing. It has often been suggested that playing a record
backwards and then reversing the transfer has some advantages. Among those cited are:

1. The opposite side of any steep wavefront is played, so wear has less effect.
2. Resonances and other effects which smear the signal in time are neutralised.
3. It is easier to extract the first milliseconds of modulation if the cutter has been
lowered with sound on it.
4. It is easier to distinguish between clicks and music for electronic treatment.
5. If you are using fibre needles, the problems which would be caused by the
needle being most-worn at the middle of the disc are ameliorated.
6. Needle-digs and other sources of repeating or jumping grooves are more easily
dealt with.

63
Unfortunately the author simply does not agree with the first two reasons,
although he has tried the idea several times. Worn records still sound worn (if the needle
is tracing the groove correctly, of course). The theory of neutralising resonances is wrong.
Even if electronic anti-resonance circuitry is proposed, the original waveform can only be
recreated if the sound passes through the anti-resonant circuit forwards.
However, the other four arguments for playing a record backwards do have
slightly more validity, but not much. In the case of argument (3), the writer finds that (on
coarsegroove records, anyway) it is quicker to lower the pickup onto the correct place,
repeating the exercise until it’s done correctly! For argument (4), analogue click detectors
work more efficiently because the circuitry is less confused by naturally-occurring
transients, such as the starts of piano notes. But since all current analogue click detectors
remove the click without replacing the original sound, they are not suited to archival uses.
Computer-based declicking systems do not care whether the record is playing backwards
or not; in effect, they shuttle the sound to and fro in RAM anyway. The writer has no
experience of argument (5), because there is not yet a satisfactory electrical pickup using
fibre needles, so you cannot reverse an electronic transfer anyway.
This leaves only the groove-jumping argument. For some records the reverse
process can be very helpful. It will, of course, be necessary to use a reverse-running
turntable, with a pivoted arm with a negative offset angle or a parallel-tracking system.
Seth Winner, of the Rogers and Hammerstein Archives of Recorded Sound, has a
conventional headshell with the cartridge facing backwards. He made this for playing
disc-stamper negatives rather than records liable to groove-jumping. If his cartridge were
to be used for groove-jumping, one would have to risk the cantilever being bent, because
it will be compressed when it was designed to work under tension.
Also there are distinct disadvantages to the reverse-playing process. To start with,
we need another turntable, or one which can be modified. A practical difficulty is that if
the operator cannot understand the music, he may well miss other faults, such as wow, or
lack of radius compensation (section 4.19). When some defects of equipment (such as
tone-arm resonances) are reproduced backwards, the result is particularly distracting,
because backward resonances cannot occur in nature.
To get the recording the right way round again, an analogue tape copy has to be
reversed. For stereo, the left and right have to be swapped when the tape is recorded, so
they will come out correctly on replay. Although I’d count it a luxury, if you were thinking
of buying a digital audio editor, I’d advise getting one with the additional feature of being
able to play a digital recording backwards while you were at it.
Since I haven’t said much about groove-jumping, I shall now devote a paragraph
to the subject, although I hesitate because any operator worth his salt should be able to
invent ways round the difficulties much more quickly than I can advise him. The obvious
way, adjusting the bias on the pickup-arm, causes the whole disc to be affected; so ideally
you need a short-term aid. My method (which can also be applied to a parallel-tracking
arm) is to apply some side-pressure through a small camel-hair paintbrush. With grossly-
damaged records this isn’t enough, so you may simply have to grab the cartridge lifting-
handle between finger and thumb and push. This latter idea works best when you are
copying at half-speed, which is the topic of the next section. You can’t always get a
transfer of archival quality under these conditions; so you may have to use your digital
editor for its intended purpose, editing the results! For some notes on playing broken
records, please see Appendix 1.
I shall now share an idea which I have not tried personally. We have seen that
tracing distortions occur because a cutting-stylus does not have the same shape as a

64
replay stylus. Obviously, if we play a groove with a cutting stylus, we shall cut into it. But
this wouldn’t happen with a cutting stylus running backwards, and this could eliminate
many kinds of tracing distortion. Extremely accurate matching between the shape and
dimensions of the two styli would be needed, plus considerable reduction in the effective
mass of the replay one to avoid groove deformation.

4.14 Half-speed copying

This is a technique which is useful for badly-warped or broken records which would
otherwise throw the pickup out of the groove. It is particularly valuable for cylinders. It is
almost impossible to get warped cylinders back to the original shape, and most of them
rotate faster than discs anyway. The solution is to transfer the item at half the correct
speed to a system running at half the desired sampling frequency.
The principal disadvantage is that the low-end responses of all the equipment have
to be accurate beyond their normal designed limits. Another is that the natural
momentum of all moving parts is lower, so speed variations in the copying equipment are
always higher. It is true that, given good modern equipment, the errors are likely to be
swamped by those of the original media; but you should remember the danger exists.

4.15 Distortion correction

You will note that this is the first significant area in which the word “maybe” occurs. I
shall be talking about processes which have yet to be invented. I don’t intend to infuriate
you, but rather to show where techniques are possible rather than impossible. In the
archival world time should not be of the essence, so you could leave “possible but not yet
practical” work until a later date.
At present, harmonic and intermodulation distortion are faults which never seem to
be reverse-engineered electronically. In principle, some types of such distortion could
easily be undone; it seems the necessary motivation, and therefore the research, hasn’t
happened. I can only recall one piece of equipment which attempted the feat during
playback - the Yamaha TC800 cassette-recorder of 1976. It certainly made the Dolby
tone (Section 8.4) sound better; but personally I could hear no difference to the music!
In the circumstances, I can only advise readers to make sure as little distortion as
possible comes off the medium at source, because (as we shall see later) there are
electronic ways of dealing with noise. Until someone breaks the mould, we must assume
that retrospective distortion-removal will never be possible, and therefore we must
concentrate upon it at source.
Harmonic distortion is, in practice, always accompanied by intermodulation
distortion. For a reasonably complete survey of this idea I refer you to Reference 22; but
in the meantime I will explain it briefly in words. If two frequencies are present at the
same time, say m and n, we not only get harmonics (2m, 3m, 4m . . . and 2n, 3n, 4n . . . ,
the conventional “harmonic distortion”), but we also get “sum-and-difference
frequencies” (m+n, m-n, 2m-n, 2n-m, etc). The latter case is called “intermodulation
distortion.” Subjectively, the worst case is usually (m-n), because this means extra
frequencies appear which are lower in pitch than the original sounds, and very
conspicuous. They are often called “blasting.” If they have come from this source (they
could also come from transient effects in the power-supply of the recording amplifier), the

65
only hope for removing them without filtering is to generate equal-and-opposite sum-
and-difference frequencies by reverse-engineering the original situation.
Gazing into my crystal ball, I can see no reason why distortion-removal should
always remain impossible. One can visualise a computer-program which could look at a
musical signal in one octave, pick up the harmonic and intermodulation products in other
octaves, and by trial-and-error synthesise a transfer-characteristic to minimise these. By
working through all the frequency-bands and other subsequent sections of sound, it
should be possible to refine the transfer characteristic to minimise the overall distortions at
different volumes and frequencies. It would be an objective process, because there would
be only one transfer characteristic which would reduce all the distortion products in the
recording to a minimum, and this would not affect naturally-occurring harmonics. If
future research then finds a transfer characteristic which is consistent for several separate
recordings done with similar equipment, we might then apply it to an “objective copy.” I
admit that, unless there is a paradigm shift because of a completely new principle, it
would mean billions of computation-intensive trials. But computer-power is doubling
roughly each year, so ultimate success seems inevitable.
The conventional approach - reverse-engineering the original situation - would
depend upon having access to the sound with the correct amplitudes and relative phases.
I have already mentioned the importance of phase in section 2.13. When we come to
frequency equalisation in later chapters, I shall be insisting on pedantically correct ways of
doing equalisation for this reason.
The first progress is likely to be made in the area of even-harmonic distortion,
which occurs on recorded media which do not have a “push-pull” action. These include
hill-and-dale grooves, the individual channels of a stereo groove, and unilateral optical
media. Sometimes these show horrendous distortion which cries out for attention.
Sometimes they are essentially reproduction problems, but at other times the recording
medium will cause varying load on a cutter, meaning distortion is actually recorded into
the groove.
In the late 1950s even harmonic tracing distortion was heard (for the first time in
many years) from stereo LP grooves. The two individual groove walls did not work
together to give a “push-pull” action to a stylus; they acted independently, giving only a
“push” action. It was suggested that record manufacturers should copy a master-nitrate
with the phases reversed so as to exactly neutralise the tracing distortion when the second
reproduction took place. Fortunately, this was not necessary; as we saw in section 4.11,
new types of playback styli were developed to circumvent the difficulty. And there was
very little recorded distortion, because by that time the cutterheads were being controlled
by motional negative feedback, which virtually eliminated distortion due to the load of the
nitrate.
Many decades before, some manufacturers of hill-and-dale records did actually
copy their masters, incidentally cancelling much of this sort of distortion. Pathé, for
instance, recorded on master-cylinders and dubbed them to hill-and-dale discs (Ref. 23),
and at least some of Edison’s products worked the opposite way, with discs being dubbed
to cylinders. And, of course, “pantographed” cylinders were in effect dubbed with phase-
reversal. So there are comparatively few cases where hill-and-dale records have gross
even-harmonic distortion. It is only likely to occur with original wax cylinders, or moulded
cylinders made directly from such a wax master.
The fact that it was possible to “correct” the even harmonic distortions shows that
it should be easy with electronics today; but archivists must be certain such processes do
not corrupt the odd harmonics, and this means we need more experience first.

66
The CEDAR Noise reduction System includes an option which reduces distortion.
This uses a computerised “music model” to distinguish between music and other noises.
Details have not yet been made public, so it is impossible to assess how objective the
process is, so I cannot yet recommend it for archive copies.

4.16 Radius compensation

Edison doggedly kept to the cylinder format long after everyone else, for a very good
engineering reason. With a disc rotating at a constant speed, the inner grooves run under
the stylus more slowly than the outer grooves, and there is less room for the higher
frequencies. Thus, all things being equal, the quality will be worse at the inner grooves.
Cylinders do not have this inconsistency. Earlier we saw some of the difficulties, and some
of the solutions, for playing disc records. But I shall now be dealing with the recording
side of the problem, and how we might compensate it.
A “feather-edged” cutter was not affected by the groove speed. Such cutters were
used for wax recording until the mid-1940s. With spherical or “soft” styli, there would be
problems in reproduction; but today we merely use a bi-radial or line-contact stylus to
restore the undistorted waveform. We do not need to compensate for the lack of high
frequencies electrically.
The problem only occurred when the cutter did not have a sharp edge, e.g.
because it had a polishing bevel. Here the medium resisted the motion of the cutter in a
manner directly proportional to its hardness. For geometrical reasons it was also inversely
proportional to the groove speed, and inversely proportional to the mechanical impedance
of the moving parts. (A stiff armature/cutter will be less affected than a floppy one. A
cutter with motional feedback has a high mechanical impedance). Finally, the effect was
also dependent upon the size of the polishing bevel and the temperature of the wax or
lacquer at the point of contact. All these factors affected the high-frequency response
which was cut into the disc.
Thus, even with perfect groove contact, we may notice a high-frequency loss
today. The effect will be worst on a recording “cut cold” in lacquer, using a duralumin-
and-sapphire coarsegroove cutting-tool in a wide-range cutterhead with low mechanical
impedance. In practice, the effect seems worst on semi-pro nitrate 78s recorded live in the
1950s.
Because of the complexity of the problem, and because no systematic analysis was
done at the time, the effect cannot be reversed objectively. On any one record, it’s usually
proportional to the groove speed; but human ears work “logarithmically” (in octaves
rather than wavelength). The subjective effect is usually imperceptible at the outside edge
of the disc. It is often inaudible half-way through; but the nearer the middle, the worse it
starts to sound.
We do not know precisely when recording engineers started compensating for the
effect as they cut the master-disc. It is thought Western Electric’s Type 628 radius-
compensator circuit was in use by 1939. Before this date, the official upper-frequency
limits of electrical recording systems prevented the effect from demanding much
attention. After 1939, it can be assumed that commercial master-disc cutting incorporated
radius compensation in some form. We may have to play the pressings with line-contact
or elliptical styli to minimise the pinch-effect distortion, but this should not affect the
intended frequency-range; compensation for the recorded losses will have been
performed by the mastering engineer.

67
For other discs, the present-day transfer operator should compare the inner and
outer radii. The usual procedure is to assume that the outside edge suffers no radius loss,
and compensate for the high-frequencies at other radii by ear on the service-copy only.
The operator will certainly have to do this if the subject matter requires the sides to be
seamlessly joined!
Because the effect is wavelength-dependent, the compensation circuit should
ideally vary the frequency of the slope continuously, not the slope itself. There is a limit to
the compensation possible without making drastic increases in hiss and harmonic
distortion. When we consider this, we observe that objective compensation is impossible
for another reason. The transfer operator must use subjective judgement to balance the
effects and minimise them for the listener.
The author knows of only one organisation which treated radius-compensation
scientifically, and unfortunately its research was based on a different foundation. During
the second world war, the BBC was attempting to stretch the performance of its nitrate
lacquer disc-cutting operation to 10kHz, and the engineers considered the whole system
(recording and reproduction) together. So far as reproduction was concerned, they settled
on a standard stylus (2.5 thou spherical sapphire) and a standard pickup (the EMI Type 12
modified so its fundamental resonance was 10kHz), and they devised radius-
compensation which gave minimum distortion when nitrates were played with this
equipment. And higher frequencies were ignored, because the landline distribution system
and the characteristics of double-sideband amplitude-modulation transmission usually
eliminated frequencies above 10kHz anyway. The compensation was designed for cold
cutting of V-shaped grooves into cellulose nitrate blanks. The result was a family of
resonant circuits in the recording electronics, each with a different resonant frequency and
peak level. An electrical stud system (like a stud fader) switched between these circuits
about five times during every inch of recorded radius. (Ref. 24). This continued until the
BBC abandoned coarsegroove nitrate discs in about 1965.
From today’s viewpoint, this puts us in a dilemma. It would seem that we should
play such discs with a 2.5 thou spherical sapphire in an EMI Type 12 cartridge; but this is a
destructive instrument by today’s standards, and it will damage the disc. Furthermore the
BBC assumed the nitrate had consistent hardness and elasticity. Several decades later the
material has altered considerably, so accurate reconstruction of the intended situation is
impossible anyway. Finally it may be impossible for academics running a sound-archive to
recover the original intended sound, because of the tradeoffs made to minimise side-
changes after the sound was broadcast with limited bandwidth.
The current policy at the British Library Sound Archive is to compensate only for
the steady-state recording characteristics (which we shall be considering in chapter 5). We
generally play the discs with the usual truncated elliptical styli to recover the maximum
power-bandwidth product, but we do not attempt to neutralise the resonant artefacts at
high frequencies, which are audible (but not severe) under these conditions. It is possible
that some form of adaptive filtering may analyse the high-frequency spectrum and
compensate it in future; in the meantime we have preserved the power-bandwidth
product, which is the fundamental limitation.
The remainder of this chapter is concerned with the state-of-the-art in audio
restoration technology, but can only be considered to be so at the time of writing. While
much of the information will inevitably become outdated, it may still remain instructive
and of some future use.

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BOX 4.17
METHODS OF REPLACING CLICKS

1. The first is to cut both the click and the sound which was drowned by it, and
to pull the wanted sounds on either side together in time. This is how tape editors
have worked for the past forty years. Not only does it destroy part of the original
waveform, but in extreme cases it can destroy tempo as well.
2. Another answer is to replace the click with nothing. Certainly, it is true that
leaving a hole in the music is less perceptible than leaving the click; but we can
hardly call it “restoring the original sound” - at least, if we mean the objective
sound-wave rather than the sensation.
3. Another answer is to synthesise something to fill the gap. A very popular
method is the “two-band method” (where there are two processors, one dealing
with high frequencies, which leaves little holes as before, and one dealing with
low frequencies, which holds the instantaneous voltage throughout the gap). This
is subjectively less noticeable, but again you cannot call it “restoring the original
sound.”
4. John R. T. Davis was the inventor of the “Decerealization” technique, which
emulates this process. It involves a quarter-inch analogue magnetic tape of the
clicky disc. A special jig which incorporates a tape-head and an extremely accurate
marking-device holds the tape. Its dimensions are such as to permit a “stick-slip
action” as the tape is pulled by hand. The operator listens on a selected
loudspeaker, and as the individual short segments of sound are reproduced, the
click stands out conspicuously. After the position is marked, the surface layer of
the tape is scraped off where the click is. Although very labour-intensive, this
remains the best way to deal with some types of material, because the operator
can scrape off different degrees of oxide, thus creating the effect of the previous
method with variable crossover frequencies for each click. In addition, when you
can’t hear the click, the waveform isn’t attacked.

5. Another technique is to take a piece of sound from somewhere else in the


recording and patch it into the gap. This technique was first described by D. T. N.
Williamson, and although automatic devices using the idea have been proposed,
they have never appeared. (It was envisaged that sound delayed by a few
milliseconds could be patched into place, but it was found difficult to change to
the delay-line without a glitch). Manual equivalents of the principle have been
used successfully by tape editors. It has the “moral” advantage that you can say
nothing has been synthesised. All the sounds were made by the artist!
6. More elaborate synthesis is used by the digital computer-based noise reduction
methods “No-Noise” and “CEDAR.” They analyse the sound either side of the
click, and synthesise a sound of the same spectral content to bridge the gap.
7. The final solution to the question “what do we replace the click with” only
works if you have two copies of a sound recording and each of them suffers from
clicks in different places. Then we can take the “best of both” without
interpolating the waveform.

69
4.17 Electronic click reduction

The elimination of clicks has been a tantalising goal for more than half a century, because
it is a relatively simple matter to detect a click with simple electronic techniques. The
problem has always been: What do we replace the click with? (see Box 4.17)
All but the last method disqualify themselves because they do not pretend to
restore the original sound waves; but if you need more details, please see Ref. 25.
Unfortunately, there are relatively few pieces of equipment which can reduce noise
without affecting the wanted waveform. In fact there are so few that I must mention
actual trade-names in order to make my points; but I should remind readers these pieces
of apparatus will be displaced in time. Some of them may be needed only occasionally,
and may be hired instead of purchased; or recordings might be taken to a bureau service
to cut down the capital expenses. You will need to know the various options when
formulating your strategy.
The most important objective technique is Idea (7) in the Box, which is employed
as the “first stage” of the Packburn Noise Reduction System (Packburn Electronics Inc,
USA; Ref. 26). This is an analogue processor widely used in sound archives, and it has
three stages. The first is used when a mono disc is being played with a stereo pickup, and
the machine chooses the quieter of the two groove walls. It cannot therefore be used on
stereo records.
Analysis of the actual circuit shows that it only attenuates the noisier groove wall
by 16dB, so the description I have just given is something of an oversimplification; but it is
certainly effective. The result is a little difficult to quantify, because it varies with the
nature of the disc-noise and how one measures the result; but on an unweighted BBC
Peak Programme Meter an average EMI shellac pressing of the inter-war years will be
improved by about ten decibels. And, as I say, the waveform of the wanted sound is not,
in principle, altered.
I should, however, like to make a few points about the practical use of the circuit.
The first is that if we play one groove-wall instead of both groove walls, we find ourselves
with a “unilateral” medium. Thus we risk even-harmonic distortion, as we saw in section
4.15. Actually, there is a mid-way position on the Packburn such that the two groove
walls are paralleled and the whole thing functions as a lateral “push-pull” reproduction
process. Theoretical analysis also shows that optimum noise reduction occurs when the
groove walls are paralleled whenever they are within 3dB of each other. The problem is to
quantify this situation.
The manufacturers recommend you to set the “RATE” control so the indicator-
lights illuminate to show left groove-wall, right groove-wall, and lateral, about equally. I
agree; but my experience with truncated-elliptical styli is that there is very little even-
harmonic distortion reproduced from each groove wall anyway. You shouldn’t worry
about this argument; there are other, more-significant, factors.
The next point is that, in the original unmodified Packburn, control-signals broke
through into the audio under conditions of high gain, giving a muffled but definite
increase in background noise which has been described subjectively using the words “less
clarity” and “fluffing”. Therefore the RATE control must be set to the maximum which
actually improves the power-bandwidth product and no more. My personal methodology
is based on playing HMV Frequency Test Disc DB4037, which we shall be considering in
chapter 5. Using a high frequency test-tone, we can easily hear the best noise reduction
happens when the three light-emitting diodes are lit for about the same overall time. Thus
the manufacturer’s recommendation is confirmed. Do this on real music, and the optimum

70
power-bandwidth is assured, even though it is less easy to hear the side-effects. Now that
“The Mousetrap” manufactured in the UK by Ted Kendall has replaced “The Packburn,”
this problem has been eliminated by the use of high-speed insulated-gate field effect
transistors (IGFETs).
Another point is that, if the machine is to switch between the two groove walls
successfully, the wanted sound on those two groove walls must be identical in volume
and phase. (Otherwise the switching action will distort the waveform). The Packburn
therefore has a control marked “SWITCHER - CHANNEL BALANCE.” When you are
playing a lateral mono disc, you switch the main function switch to VERTICAL, and adjust
this control to cancel the wanted signal. Then, when you switch back to LATERAL, the
two groove walls will be going through the processor at equal volumes.
All this is made clear in the instruction-book. But what if you cannot get a null? In
my view, if the wanted sound is always audible above the scratch, there’s something
wrong which needs investigating. Assuming it isn’t a stereo or fake-stereo disc, and you
can get a proper cancellation on known mono records (which eliminates your pickup),
then either the tracking angle is wrong (most of Blumlein’s discs, section 6.31 below), or
you’ve found a record made with a faulty cutterhead (e. g. Edison-Bell’s - section 6.16
below).
The former fault can be neutralised by slewing the pickup cartridge in its headshell.
The latter faults have no cures with our present state of knowledge, but cures may be
discovered soon, which would be important because sometimes there is useful power-
bandwidth product in the vertical plane. In the meantime, all you can do is slew the
cartridge as before in an attempt to cancel as much sound as possible, and then try the
Packburn in its usual configuration to assess whether its side-effects outweigh the
advantage of lower surface-noise.
To decide between the two groove walls, the machine needs access to undistorted
peak signals at frequencies between 12kHz and 20kHz. It has been said that even the best
stereo analogue tape copy of a disc will mar the efficiency of the unit, because it “clips”
the peaks or corrupts their phase-linearity, and it is rather difficult to keep azimuths
(section 9.6) dead right. This makes it difficult to treat a record unless you have it
immediately beside the Packburn. Actually, I do not agree; I have even got useful noise
reduction from a stereo cassette of a disc. But certainly the Packburn isn’t at its best under
these conditions.
But digital transfers seem “transparent” enough. So it is practicable to use a two-
channel digital transfer for the “archive” (warts-and-all) copy, provided no disc de-
emphasis is employed (sections 3.5 or 6.23). Meanwhile, for objective and service copies
it is best to place the Packburn following a flat pickup preamplifier with the usual
precautions against high-frequency losses. Any frequency de-emphasis must be placed
after the Packburn. (This is indeed how the manufacturers recommend the unit should be
used).
The “second stage” of the Packburn is “the blanker”, a device for removing clicks
which remained after the first stage, either because both groove walls were damaged at
the same place, or because it was a stereo disc. The Packburn’s blanker rapidly switches to
a low-pass filter, whose characteristics are designed to minimise the subjective side-effects
(as paragraph (3) of Box 4.17. It does not restore the original sound wave, so it should
only be used for service copies. Likewise, the “third stage” comprises a quite good non-
reciprocal hiss reduction system (chapter 10), but this too alters the recorded waveform,
so it too should be confined to service copies. To remove the remaining hiss and crackle

71
whilst keeping the waveform, we must use alternative techniques; but the Packburn “first
stage” is a very good start.
There are two such alternative techniques. One is to emulate the action of the
Packburn first stage, but using two different copies of the same record. I shall be talking
about this idea here and in section 4.20. The other is to use computer-based digital
processing techniques to synthesise the missing sound.
The first idea is still in the development stage as I write, but the principle of its
operation is very simple. Two copies of a disc pressed from the same matrix are played in
synchronism. If the discs are mono, each goes through a “Packburn first-stage” (or
equivalent). The difficult part is achieving and keeping the synchronism, for which the
geometrical errors must be kept very low; but once this is achieved, a third Packburn first-
stage (or equivalent) cleans up the result. Using the same example as I had in section
4.16, the result is a further 8dB improvement in signal-to-noise ratio. The noise coming
from each disc is not actually steady hiss (although it usually sounds like it), but a very
“spiky” hiss which responds to the selection process. If it had been pure white-noise,
equal on the two copies but uncorrelated, the improvement would only be 3dB. (Which
would still be worth having). Isolated clicks are generally completely eliminated, and no
synthesis of the waveform is involved.
For this principle to work, the two discs have to be synchronised with great
accuracy - better than 0.2 milliseconds at the very least - and this accuracy must be
maintained throughout a complete disc side. Although digital speed-adjustment
techniques exist, we saw in section 3.4 these have disadvantages which we should avoid
if we can. So use a deck with minimal geometrical errors. For example, use a parallel-
tracking arm whose pickup is pivoted in the plane of the disc, or provided with an
effective means of keeping it a constant distance above the disc surface, so warps do not
have an influence. The sampling-frequency of the analogue-to-digital converter is then
“locked” to the turntable speed; there are other reasons in favour of doing this, which I
shall mention towards the end of section 5.5. In the British Library Sound Archive’s case, a
photoelectric device has been used to look at the stroboscope markings at the edge of the
turntable, giving a 100Hz output. The result is frequency-multiplied by 441, giving a
44.1kHz clock-signal for the analogue-to-digital converters. The transfers of the two discs
are done using the same stylus and equalisation, and at the same level, through a
Packburn first-stage. The results are transferred to a digital audio editor and adjusted until
they play in synchronism. The result is fed back through another Packburn at present,
although a digital equivalent is being written to avoid unnecessary D-A and A-D
conversions.
It has been found advantageous to combine the two discs using methods which
operate on different frequency-bands independently. The Packburn only switches in
response to noises in the range 12 – 20kHz. But if we have uncorrelated low frequency
noises (e.g. rumble introduced during pressing), the switching action will generate
sidebands, heard as additional clicks. In a prototype digital equivalent of the Packburn
First Stage, we divide the frequency range into discrete octaves and treat each octave
separately. The switching action takes place in each octave at the optimum speed for
minimising sideband generation, and of course we get the quieter of the two grooves at
all frequencies (not just 12-20kHz). We also get the version with the least distortion-
effects in each band. The wanted waveform is never touched; all that happens is that
background-noises and distortions due to the reproduction process are reduced. But at
least two “originals” must be available.

72
We return now to when we have only one “original.” It is always possible to
combine two plays of the same disc with different-sized styli, using the technology I have
just described. This imitates the action of a “soft” stylus!
Several systems synthesise the missing section of waveform (previously drowned
by the click) and insert it into place. Most digital processes use a technique known as the
Fast Fourier Transform, or FFT, to analyse the wanted sound either side of the click. This is
a speedy algorithm for a binary computer; in audio work, it is some hundreds of times
faster than the next best way of doing the job, so it can run usefully even on a desktop
microcomputer. (Ref. 27). When the click is eliminated, random digits are re-shaped
according to the FFT analysis, and when the inverse FFT is performed, the missing
waveform is synthesised. Both “No-Noise” and “CEDAR” offer realtime implementation,
so the operator can switch between the processed and unprocessed versions and check
that nothing nasty is happening to the music. Both replace the clicks with synthesised
sound, so in principle we are not actually restoring the original waveform; but it’s a matter
of degree. Experiments can be set up taking known waveforms, adding an artificial click,
and seeing what sort of job the computer does of synthesising the original. The result may
be judged aurally, or visually (on a waveform display of some sort).
The CEDAR people have various pieces of hardware and software for click
removal. This includes a computer-based platform offering several powerful restoration
algorithms (not just clicks), free-standing boxes which cannot be used for any other
purpose (the cheaper one has no analogue-to-digital or digital-to-analogue converters),
and cards for slotting into a SADiE PC-based hard-disk editor. The last-mentioned is
usually the most recent version, since it is easier to make both “beta versions” and proven
hardware.
Although “real-time,” the free-standing DC.1 unit may require the signal to be
played through the machine three times, since three different algorithms are offered; they
should be performed starting with the loudest clicks and ending with the quietest. CEDAR
very bravely admit that the DC.1 process does not always synthesise the waveshape
correctly for a long scratch, but in 1994 they claimed it was correct for clicks up to fifty
samples long. CEDAR have been extending the number of samples; the latest version is in
the range 250-300 samples. (This clearly shows archivists must log the version-number of
the software). If the results were put to a scientific test on both aural and visual grounds
with 100% successful results, there would presumably be no objection to using the
algorithm for objective copies as well as service copies.
Unfortunately, since one must start with the biggest clicks, and the DC.1
sometimes blurs these (making it more difficult for future processes to detect them), there
are relatively few records for which the DC.1 gives archivally-acceptable results. Malcolm
Hobson’s solution is to run his process several times from hard disc in batch mode (this
avoids having to accumulate several R-DATs which must work in real time). He starts with
an FFT looking for high frequency transients less than six samples long (these are almost
bound to be components of crackle), then interpolates these (which is certain to give
faithful results for such small clicks). The process then works upwards towards larger
clicks. Each time the surrounding music has less crackle, so interpolation is easier.
However, much loving care-and-attention is needed for the benign replacement of the
largest clicks, which may be done manually. So the trade-off is between leaving blurred
clicks and possibly-inaccurate interpolation.
The No-Noise process is obliged to work in conjunction with a Sonic Solutions
editing-system, which could be a restriction for some customers; but it is possible to
concatenate several processes (for example, de-clicking, de-hissing, and equalisation), and

73
run them all in real-time. This helps the operator to listen out for unwanted side-effects to
any one of the processes. No-Noise has an option to mark the start and end of long clicks
manually, and then do a trial synthesis of the missing signal, which you can adopt if you
are satisfied. I have heard it do a convincing job on many thousands of missing samples,
but I do not know how “accurate” this was. Although it has nothing to do with click
removal, this process seems to be the best way to synthesise sound for a sector of a
broken record which has vanished. This will probably never be an objective technique; but
over the years many similar jobs have been done in the analogue domain. No-Noise can
help in two ways, firstly by synthesising the missing sound, and secondly by performing
edits non-destructively.
CEDAR’s computer-platform system and their free-standing de-crackle unit Type
CR.1 offer another process. To oversimplify somewhat, the recording is split digitally into
two files, one through a “music model” and the other comprising everything else. It is
then possible to listen to the “music model” on its own, and adjust a control so that even
the biggest clicks are eliminated. (This file lacks high frequencies and has various digital
artefacts along with the music, but it is easy to listen for loud clicks if they are there).
When a satisfactory setting has been achieved which eliminates the loudest clicks but
goes no further, the two files are recombined. This process has been found empirically to
reduce many of the effects of harmonic distortion, as I mentioned in section 4.15.
As we go to press, audio engineers are exploring other mathematical strategies for
synthesising missing data. So far, these studies seem to comprise “thought experiments”,
with no “before-and-after” comparisons being reported. The only one to have appeared
is the newly developed SASS System (invented by Dr. Rudolf Bisping). Prony’s Method is
used to analyse the music and express it as a sum of exponentially-decaying frequencies,
which enables complete remodelling of the amplitude spectrum, including signals which
change in pitch and notes which start or stop during the click. To do all this requires a
computer some hundreds of times more powerful than hitherto. The SASS System has a
dedicated architecture involving many transputers; but once again I have not had an
opportunity to test it for “accuracy.”
Interpolation for replacing the biggest clicks is still not reliable. But it is well-known
that interpolation is easier on sounds which change slowly, and that clicks appear
subjectively louder on these same sounds. I consider we need a click-replacement process
which automatically adapts itself to the subject matter. To end with a crudely-expressed
dream, we need an interpolation strategy which automatically knows the difference
between clicks during slow organ music, and clicks during a recording of castanets.

4.18 Electronic hiss reduction

No-Noise, CEDAR, and SASS also offer “hiss-reduction” algorithms. I wish to spend some
time talking about these, because they offer conspicuously powerful methods of reducing
any remaining noise; but frankly I am sure they are wrong for archival storage purposes.
The idea is to carve up the frequency-range into a number of bands, then reduce
the energy in each band when it reaches a level corresponding that of the basic hiss. The
process can reduce hiss very spectacularly; but it can cut audible low-level signals fainter
than the hiss, so listeners sometimes complain there is “no air around” the performance.
At its best it can reduce so much noise that it is possible to extract wanted high-frequency
sounds which are otherwise inaudible, thereby apparently making a dent in the power-
bandwidth principle (section 2.2).

74
I must also report that an analogue equivalent was being marketed by Nagra at
one point (the Elison Model YSMA 18 with eighteen frequency bands); but it did not
become available for some reason, which was a pity as it could be operated more
intuitively than any digital equivalents.
Unfortunately these processes must make use of psychoacoustics to conceal side-
effects. The width of the sample being analysed (in both the time and frequency
domains), the amplitude below which attenuation can take place, the degree of
attenuation, the times of response and recovery, and the volume at which reproduction is
assumed to take place may all need to be taken into account. To make matters worse,
recent psychoacoustic experiments suggest that our ears work differently when listening
to speech as opposed to music. Most hiss-reduction units have user-adjustable controls
for some of these factors. Although offered on a try-it-and-see basis, this subjective
approach rules it out for archival applications. The controversies about records which have
been processed by both No-Noise and CEDAR are usually attributable to the fact that the
operator and the consumer have different psychoacoustic responses and/or listening
conditions. Nevertheless, psychoacoustic measurements have made strides in the last
decade, and many of the factors can now be quantified with precision, and, equally
importantly, with a clear understanding of the variances.
The Fast Fourier Transform requires the number of frequency bands to be an exact
power of two, with linear spacing. At present, No-Noise uses 2048 bands and CEDAR
uses 1024, giving bands about 11 and 22 Hertz wide respectively when the sampling-
frequency is 44.1kHz. This is not the optimum from the psychoacoustic point of view; it is
well known that the human ear deals with frequency bands in quite a different way.
To reduce hiss whilst (apparently) leaving the wanted sounds intact, wanted
sounds must “mask” unwanted ones. The unit of masking is the “bark”. Listening tests
suggest that the human ear has a linear distribution of barks at frequencies below about
800Hz, and logarithmic above that. Thus any computer emulating the masking properties
of the human ear needs a rather complex digital filter. Furthermore, the slopes of the
filtered sections of the frequency range are asymmetrical, and vary with absolute volume.
The last-mentioned parameter has been circumvented by Werner Deutsch et. al. (Ref.
28), whose team chose values erring on the side of never affecting the wanted sound
(“overmasking”). In my view, this algorithm is the best available method of reducing hiss
while leaving the (perceived) wanted sound untouched. It is surprisingly powerful. Even
when the hiss and the music are equal in volume, the hiss can be reduced by some 30dB;
but its quality then becomes very strange. There is also the difficulty that, owing to the
vital contribution of psychoacoustics, any frequency or volume changes must be made
before the process, not after it.
Bisping divides the spectrum into 24 bands that correspond to critical bands in
Corti’s organ of the inner ear, and hiss-removal is inhibited when it is isn’t needed. The
trouble is that many experts in psychoacoustics consider 24 bands an oversimplification!
Such processes might be applied to the service copies of recordings meant to be
heard by human adults. (But not to other sounds, or to sounds which might go through a
second process involving audio masking). Even so, the correct archival practice must
surely be to store the recording complete with hiss, and remove the hiss whenever it is
played back.
We may yet find that the human ear has evolved to make the best use of the
information presented to it, with little room for manoeuvre. We already know that the
threshold of hearing is almost exactly at the level where the Brownian-movement of
individual air molecules lies. Thus we might find that our pitch-detection and our

75
tolerance to background-noise have evolved together to give a performance which
cannot be improved. If so, we could never reduce wideband hiss to reveal completely
inaudible sounds. But I very much look forward to further developments, because they
might permit a real breakthrough in sound restoration. The limitations of the power-
bandwidth principle could be breached for the very first time.

4.19 Eliminating rumble

Apart from the use of linear filters (which affect the wanted sound, of course), there have
been very few attempts to neutralise the low-pitched noises caused by mechanical
problems in the cutting machine. Not all such noises are capable of being removed
“objectively,” but there are a few exceptions. Usually these occur when the machine was
driven by gears (as opposed to idler-wheels, belts, or electronic servo-systems). Here the
pattern of rumble may form a precise relationship with the rotational speed of the cylinder
or disc. Using the principle of digital sampling being locked to rotational speed, as
mentioned in section 0 above, it is possible simply to add together the sounds from each
turn of the record. When this is done, you may often find a consistent pattern of low-
frequency rumble builds up, which may be low-pass filtered and then subtracted from
each of the turns in the digital domain to reduce the noises without affecting the wanted
sound. This is particularly valuable when you’re trying to get some bass back into acoustic
recordings (Chapter 11).

4.20 De-thumping

This section deals with the side-effects of large clicks when played with many practical
pickup arms. The click may shock-excite the arm or disc into resonances of its own, so
that even when the click is eliminated, a low-frequency “thud” remains. From one or two
cases I have known, I suspect the actual cartridge may also be excited into ringing-noises
at much higher frequencies (a few kiloHertz). Sometimes these exhibit extremely high Q-
factor resonances, which cause long “pinging” noises. In the ordinary course of events
these artefacts are not audible, because they are masked by the click itself. Only when the
click is removed does the artefact become audible.
Quite frankly, the best solution is not to generate the thumps in the first place.
Very careful microscopic alignment of cracked records may be needed to ensure that the
pickup is not deviated from the centre-line of its travel. The cartridge may need to have
silicone grease packed in or around it to reduce its tendency to make mechanical or
electrical signals of its own. The pickup arm must have well-damped mechanical
resonances; alternatively, a “parallel-tracking” arm may be used (section 4.2). This
suspends the cartridge above the record in a relatively small and light mechanism, and, all
things being equal, has less tendency to resonate. (Some parallel-trackers are capable of
dealing with misaligned cracks which would throw a pivoted tone-arm, because they are
guided by a relatively inert motor mechanism. This is probably the best way to play a
warped and/or broken disc which cannot be handled any other way).
The Tuddenham Processor not only removes clicks; it also has a de-thump option,
which applies a transient bass-cut with an adjustable exponential recovery. However, this
is a subjective process, which should only be applied to the service-copy.

76
CEDAR have an algorithm for de-thumping which relies on having a large number
of similar thumps from a cracked record. When twenty or thirty are averaged, the
components of the wanted sound are greatly reduced, leaving a “template” which can be
subtracted from each one. It does not significantly corrupt the original waveform, so it has
its place. Sounds intended for such treatment should have the basic clicks left in place, as
CEDAR uses them to locate the thumps. The makers of the SADiE digital editor can supply
a de-thump card for the machine, but I simply do not have experience of its principles or
its ease of operation.

4.21 Future developments

As I write, the industry is excited by the possibilities of adaptive noise-cancellation. This is


similar to the hiss-reduction processes I mentioned in section 4.18, except that instead of
using a fixed sample of hiss to define the difference between hiss and music, it can be a
dynamic process. (Ref. 29). Given a sample of “pure” noise varying with time, the
computer can (in theory) do a Fast Fourier Transform of the noise, and use it to subtract
the appropriate amount of energy from the signal.
The makers envisage it could be used on the following lines. If it were desired to
eliminate the background noise of speech picked up in an aircraft cockpit, for example,
the usual noisy recording would be made, plus another track of pure aircraft noise (from
another microphone). In theory, it would then be possible to sample the pure aircraft
noise and use it to reduce the noise behind the speech, without having to rely upon
phase-accurate information. This is actually an old idea (Ref. 30), which hitherto has been
used by the US military for improving the intelligibility of radio-telephone communication.
Only now is enough computing power becoming available for high-fidelity applications.
For once, sound archivists don’t have to plead their special case. It is an application with
many uses in radio, films, and television, and I anticipate developments will be rapid.
With mono disc records there are new possibilities, because “pure noise” is largely
available. It is possible to extract a noise signal from a lateral disc by taking the vertical
output of the pickup, although the rumble of the recording-turntable usually differs
between the two planes. It offers the only hope for ameliorating cyclic swishing noises
when there is only one copy of the disc, or when all surviving copies are equally affected.
Some of the effects of wear might also be neutralised, although I wouldn’t expect to get
all the sound back; indeed, we might be left with conspicuous low-frequency
intermodulation distortion. Certain types of modulation noise on tape recordings could
also be reduced, by splitting a tape track in two and antiphasing them to provide a “clean
noise” track. Since it is possible to insert timeshifts into either signal, tape “print-through”
might even be reduced. In chapter 11 I shall be considering the feasibility of dynamic
expansion; this would almost certainly have to be done in conjunction with adaptive
noise-cancellation to conceal the effect of background noise going up and down. But it
seems these applications must always be subjective processes, only to be considered when
drastic treatment is essential for service copies.
I should stress that adaptive noise-cancellation still has not achieved success in
audio. One attempt to reduce long disc clicks failed, because there was insufficient
processing power to analyse rapidly-varying noise. Disc clicks are distinguished from
wanted sound because they change so rapidly.
At one time CEDAR were developing an algorithm which emulated the dual-
processing method described in section 0 above, although it did not actually take the

77
wanted sound from two copies. No “hard-lock” synchronisation was involved, so it could
be used on wild-running transfers from two widely different sources. The reason it is not
yet available is that it was very computation-intensive, and did not always offer
satisfactory results because of difficulties synchronising recordings in the presence of
noise. In the case of disc records, the two copies each underwent click-reduction first, so
the whole point of dual processing (to avoid interpolation) was missed. (Refs. 31 and 32).
Nevertheless, this process might be used for reducing the basic hiss of already-quiet
media, such as two magnetic tapes of the same signal. But it will always be computation-
intensive. One can only hope that increased processing power and further research might
make this process successful.

4.22 Recommendations and conclusion

This ends my description of how to recover the power-bandwidth product from grooved
media, but I have not yet formally stated my views about what the three versions should
comprise.
The archive copy should be a representation of the groove reproduced to constant-
velocity characteristics (see chapter 6) using a stereo pickup, so that the two groove walls
are kept separate. Ideally, there should be several separate transfers done with styli of
different sizes, to provide samples at different heights up the groove wall. It is natural to
ask how many transfers this should be. Experiments with the earliest version of the
program mentioned in section 0 have been done, in which additional counting procedures
were inserted to quantify the number of samples taken from each transfer. This was
checked by both digital and analogue methods of measuring the resulting signal-to-noise
ratio. All three methods suggest that, for coarsegroove shellac discs played with truncated
elliptical styli, four such transfers should be done with styli whose larger radii differ by 0.5
thou. Softer media, such as vinyl or nitrate, will have a greater commonality between the
transfers, because the styli will penetrate deeper into the groove walls; so four is the most
which will normally be needed. Of course, this means a lot of data must be stored; but if
you accept that Program J1 * does the job of combining the transfers optimally, you can
use this, and still call it an “archive copy.” To help any future anti-distortion processes,
the stylus dimensions, and the outer and inner radii of the disc grooves, should be logged.
And I remind you that absolute phase must be preserved (section 2.11).
For the objective copy, the same procedure should be followed, except that known
playing-speeds (chapter 5 and recording characteristics (chapter 6) should be
incorporated. Clicks may be eliminated, so long as accurate interpolation of the
previously-drowned waveform occurs. It is natural to ask what tolerance is acceptable. My
answer would be to do some before-and-after tests on the declicker; if subtracting “after”
from “before” results in no audible side-effects, then the waveforms were synthesised
accurately enough. But I recognise readers might have other ideas.
For the service copy, radius compensation may be applied, speed adjustments for
artistic reasons can be incorporated, hiss-reduction may be considered, and sides may be
joined up where necessary (section 13.2).
I hope my peek into the future won’t leave you cross and frustrated. Digital
techniques are admittedly costly and operationally cumbersome, but there has been
enormous progress in the last few years. By the time you read these words, the above
paragraphs are certain to be out-of-date; but I include them so you may see the various
*
Editors’ note: program J1 was probably written by Peter Copeland but has not been found.

78
possibilities. Then you can make some sensible plans, start the work which can be done
now, and put aside the jobs which may have to wait a decade or two.

REFERENCES

1: Franz Lechleitner, “A Newly Constructed Cylinder Replay Machine for 2-inch Diameter
Cylinders” (paper), Third Joint Technical Symposium “Archiving The Audio-Visual
Heritage,” Ottawa, Canada, 5th May 1990.
2: Percy Wilson, “Modern Gramophones and Electrical Reproducers,” (book), (London:
Cassell & Co., 1929), pp. 126-128.
3: P. J. Packman, British patent 23644 of 1909.
4: The earliest reference I have found is an anonymous article in Talking Machine News,
Vol. VII No. 89 (May 1909), page 77.
5: Carlos E. R. de A. Moura, “Practical Aspects of Hot Stylus,” Journal of the Audio
Engineering Society, April 1957 Vol. 5 No. 2, pp. 90-93.
6: A. M. Pollock, Letter to the Editor. London: Wireless World, April 1951, page 145.
7: S. Kelly, “Further Notes on Thorn Needles.” Wireless World, June 1952, pages 243-
244.
8: A. M. Pollock, “Thorn Gramophone Needles.” Wireless World, December 1950, page
452.
9: H. E. Roys, “Determining the Tracking Capabilities of a Pickup” (article), New York:
Audio Engineering Vol. 34 No. 5 (May 1950), pp. 11-12 and 38-40.
10: S. Kelly, “Intermodulation Distortion in Gramophone Pickups.” Wireless World, July
1951, pages 256-259.
11: F. V. Hunt, “On Stylus Wear and Surface Noise in Phonograph Playback Systems.”
Journal of the Audio Engineering Society, Vol. 3 No. 1, January 1955.
12: J. A. Pierce and F. V. Hunt, J.S.M.P.E, Vol. 31, August 1938.
13: J. Walton, “Stylus Mass and Distortion.” Paper presented to the Audio Engineering
Society Convention in America in October 1962, but only published in printed
form in Britain. Wireless World Vol. 69 No. 4 (April 1963), pp. 171-178.
14: Roger Maude: “Arnold Sugden, stereo pioneer.” London: Hi-Fi News, October 1981
pages 59 and 61. (Includes complete discography)
15: John Crabbe: “Pickup Problems, Part Two - Tracking Error,” Hi-Fi News, January
1963, pp. 541-545.
16: Richard Marcucci, “Design and Use of Recording Styli,” J.A.E.S., April 1965 pp. 297-
301.
17: Wireless World, April 1948 page 135.
18: John Crabbe: “Dynagroove Hullabaloo,” Hi-Fi News, November 1963 pages 417 and
419, and December 1963 pages 521 and 523.
19: Harry F. Olsen: “The RCA Victor Dynagroove System” (paper), Journal of the Audio
Engineering Society, April 1964, pp. 203-219.
20: Basil Lane, “Improving groove contact,” Hi-Fi News, August 1980 pages 75-77.
21: C. R. Bastiaans, “Factors affecting the Stylus/Groove Relationship in Phonograph
Playback Systems,” Journal of the Audio Engineering Society, (1967?), pages 107-
117.
22: “Cathode Ray”: “More Distortion . . . What Causes Musical Unpleasantness?”
(article), Wireless World Vol. 61 No. 5 (May 1955), pp. 239-243.

79
23: Girard and Barnes, “Vertically Cut Cylinders and Discs” (book), pub. The British
Library Sound Archive.
24: For a general article on the BBC philosophy, see J. W. Godfrey and S. W. Amos,
“Sound Recording and Reproduction” (book), London: Iliffe & Sons Ltd (1952),
page 50 and pages 80-82. Details of the effect of the circuit for coarsegroove 33s
and 78s on the BBC Type D disc-cutter may be found in BBC Technical Instruction
R1 (October 1949), page 8; and there is a simplified circuit in Fig. 1.
25: Adrian Tuddenham and Peter Copeland, “Record Processing for Improved Sound”
(series of articles), “Part Three: Noise Reduction Methods,” London, Hillandale
News (the journal of the City of London Phonograph and Gramophone Society),
August 1988, pages 89 to 97.
26: Richard C. Burns, “The Packburn Audio Noise Suppressor” (article), Sheffield, The
Historic Record No. 7 pages 27-29. (March 1988).
27: The Fast Fourier Transform was invented in several forms by several workers at several
times, and there does not seem to be a definitive and seminal article on the
subject. For readers with a maths A-level and some programming experience with
microcomputers, I recommend Chapter 12 of the following:
William H. Press, Brian P. Flannery, Saul A. Teukolsky, and William T. Vetterling:
“Numerical Recipes - The Art of Scientific Computing,” (book), Cambridge
University Press (1989). This is available in three editions, the recipes being given in
three different computer languages.
28: Werner A. Deutsch, Gerhard Eckel, and Anton Noll: “The Perception of Audio Signals
Reduced by Overmasking to the Most Prominent Spectral Amplitudes (Peaks)”
(preprint), AES Convention, Vienna, 1992 March 24-27.
29: Francis Rumsey, “Adaptive Digital Filtering” (article), London: Studio Sound, Vol. 33
No. 5, pp. 34-5. (May 1991).
30: (Pioneer adaptive noise cancellation paper)
31: Saeed V. Vaseghi and Peter J. W. Rayner, “A New Application of Adaptive Filters for
restoration of Archived Gramophone Recordings” (paper), I.E.E.E Transcriptions on
Acoustics Speech and Signal Processing, 1988 pages 2548-2551.
32: UK Patent Application GB 2218307.

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5 Speed setting

5.1 Introduction

Few things give more trouble than setting the speed of an anomalous recording correctly.
There are many factors in the equation, and often they are contradictory. This writer
therefore feels it is important, not only to take corrective action, but to document the
reasons why a decision has been made. Without such documentation, users of the
transferred recording will be tempted to take further corrective action themselves, which
may or may not be justified - no-one knows everything!
I must (with respect) point out that “psychoacoustics” can often play a dominant
role in speed-setting. Personally, I can’t do the following trick myself, but many musicians
consistently and repeatedly get a sensation that something is “right” when they hear
music at the correct pitch. They usually can’t articulate how they know, and since I don’t
know the sensation myself, I can’t comment; but it’s my duty to point out the potential
traps of this situation. It’s a craft that musicians will have learnt. I am not saying that such
musicians are necessarily “wrong”. I am, however, saying that musical pitch has changed
over the years, that actual performances will have become modified for perfectly good
scientific reasons, and yet hardly anybody has researched these matters. Ideally therefore,
analogue sound restoration operators should make themselves aware of all the issues, and
be prepared to make judgements when trying to reach “the original sound” or “the
intended original sound.”
When we come to make an objective copy, there are two types of analogue media
which need somewhat different philosophies. One occurs when the medium gives no
indication of where a particular sound is located, the main examples being full-track
magnetic tape and magnetic wire. In these cases it is impossible even to add such
information retrospectively without sacrificing some of the power-bandwidth product,
because there are no sprockets, no pulses, no timecode, nor any “spare space” to add
them. But other cases have a “location-mechanism by default.” For example, we could
refer to a particular feature being at “the 234th turn of the disc record”. It is very possible
that future digital processes may use information like this; and ideally we should not
sacrifice such information as we convert the sound to digital. During this chapter we shall
see that it is often impossible to set a playing-speed with greater accuracy than one
percent. In which cases, it may be advantageous to invoke a “digital gearbox” to lock the
rotational speed of the disc with the sampling-frequency of the digital transfer, so the
rotations of the disc do not evaporate from the digital copy.
Pure sound operators are sometimes unaware that a very close lock is vital in some
circumstances, so I shall define that word “lock.” It means that the speed of the original
medium and the speed of the transferred sound must match to a very tight tolerance
(typically one part in a million). This is much tighter than most ordinary sound media can
do; so we may need to create our own “digital gearbox,” especially for digital signal-
processes downstream of us. And this means we may have to do some creative thinking
to establish a suitable gear-ratio.
On the other hand, it is impossible to “lock” analogue media which gradually
change in speed with a fixed “gearbox.” But obviously some form of gearbox is essential
for a sound medium intended to accompany moving pictures, since it is always implied

81
that “Pictures are King,” and sound must follow in synchronism, even if it’s actually the
wrong speed! As an illustration of the point I am trying to make, to provide consistently
reproducible sound for a film running at 24 frames per second, we could multiply the
frame-rate by 2000 (making 48000), and clock our analogue-to-digital converter from
that.

5.2 History of speed control

I shall start with a brief history of speed-control in the history of analogue sound
recording, and ask you to bear it in mind as different situations come up.
The very earliest cylinder and disc machines were hand-cranked, but this was soon
found unsatisfactory, except for demonstrating how pitch varied with speed !
Motor-drive was essential for anything better. Early recorders of the 1880s and
1890s were powered by unregulated DC electric motors from primitive electrochemical
cells. Several percent of slow speed drift is the normal result.
Clockwork motors, both spring and weight powered, quickly replaced electric
motors because of the attention which such early batteries demanded. But mainsprings
were less reliable than falling weights, so they tended to be used only where portability
was essential (location recording), and for amateur use. The centrifugal governor was
adopted at the same time to regulate such motors; the one in the surviving acoustic lathe
at the EMI Archive, which is weight-powered, is made to exactly the same pattern as in
spring gramophones. Oddly enough, a spring would have given better results for edge-
start disc-records. According to Hooke’s law, the tension in a spring is proportional to its
displacement, so there was more torque at the start of the recording, precisely where it
was most needed. Yet professional disc recordists actually preferred the weight system,
justifying the choice with the words “Nothing is more consistent than gravity.”
The governor could be adjusted within quite wide limits (of the order of plus or
minus twenty percent). Most commercial disc records were between 70 and 90rpm, with
this range narrowing as time progressed. Likewise, although location or amateur cylinders
might well differ widely from the contemporary standards (section 5.4), they were often
quite constant within the recording itself.
In the late 1920s alternating-current electric mains became common in British
urban areas, and from the early 1930s AC electric motors began to be used for both disc
and film recording. These motors were affected by the frequency of the supply. During
cold winters and most of the second World War, frequencies could vary.
BBC Radio got into such trouble with its programmes not running to time that it
adopted a procedure for combating it, which I shall mention in detail because it provides
one of the few objective ways of setting speeds that I know. It applies to “Recorded
Programmes” (i.e. tape and disc recordings with a “R. P. Number”, as opposed to BBC
Archive or Transcription recordings) made within a mile or two of Broadcasting House in
London. (I shall be mentioning these further in section 6.52). The various studios took
line-up tone from a centrally-placed, very stable, 1kHz tone-generator (which was also
used for the “six pips” of the Greenwich Time Signal). When a recording was started, a
passage of this line-up tone was recorded, not only to establish the programme volume
(its main purpose), but as a reference in case the frequency of the supply was wrong.
When the disc or tape was played back, it was compared with the tone at the time, and
the speed could be adjusted by ear with great accuracy. We can use this technique today.
If you are playing a “BBC Recorded Programme” and you have an accurate 1kHz tone-

82
source or a frequency-counter, you can make the recording run at precisely the correct
speed. This applies to recordings made at Broadcasting House, 200 Oxford Street, and
Bush House; you can recognise these because the last two letters of the R. P. Reference
Number prefixes are LO, OX and BU respectively. But do not use the system on other
BBC recordings, made for example in the regions. The master-oscillators at these places
were deliberately made different, so that when engineers were establishing the landlines
for an inter-regional session, they could tell who was who from the pitch of the line-up
tone. But there was an internal procedure which stated that either accurate 1kHz tone
was used, or tone had to be at least five percent different. So if you find a line-up tone
outside the range 950Hz - 1050Hz, ignore it for speed-correction purposes.
To continue our history of speed-stability. Transportable electrical recording
machinery became available from the late 1930s which could be used away from a mains
supply. It falls into three types. First we have the old DC electric motor system, whose
speed was usually adjusted by a rheostat. (For example, the BBC Type C disc equipment,
which a specialist can recognise from the appearance of the discs it cut. In this case an
electronic circuit provided a stroboscopic indicator, although the actual speed-control was
done manually by the engineer). Next we have mains equipment running from a
“transverter” or “chopper”, a device which converted DC from accumulators into mains-
voltage A.C. (For example, the machine used by wildlife recordist Ludwig Koch. These
devices offered greater stability, but only as long as the voltage held up). Finally we have
low-voltage DC motors controlled by rapidly-acting contacts from a governor. (For
example, the EMI “L2” portable tape recorder). All these systems had one thing in
common. When they worked, they worked well; but when they failed, the result was
catastrophic. The usual cause was a drop in the battery voltage, making the machine run
at a crawl. Often no-one would notice this at the time. So you should be prepared to do
drastic, and unfortunately empirical, speed correction in these cases.
It wasn’t until the “transistor age” that electronic ways of controlling the speed of
a motor without consuming too much power became available, and in 1960 the first
“Nagra” portable recorder used the technology. From the late 1960s electronic speed
control became reliable on domestic portable equipment. Similar technology was then
applied to mains equipment, and from about 1972 onwards the majority of studio motors
in Britain began to be independent of the mains frequency.
But do not assume your archive’s equipment is correct without an independent
check. I insist: an independent check. Do not rely on the equipment’s own tachometers
or internal crystals or any other such gizmology. I regret I have had too much experience
of top-of-the-range hardware running at the wrong speed, even though the hardware
itself actually insists it is correct! You should always check it with something else, even if
it’s only a stroboscopic indicator illuminated by the local mains supply, or a measured
length of tape and a stopwatch. As an example of this problem, I shall mention the
otherwise excellent Technics SL.1200 Turntable, used by broadcasters and professional
disc-jockeys. This is driven from an internal crystal; but the same crystal generates the
light by which the stroboscope is viewed. The arithmetic of making this work forces the
stroboscope around the turntable have 183 bars, rather than the 180 normally needed for
50Hz lighting in Europe. So the actual speed may be in error, depending how you
interpret the lighting conditions!

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5.3 History of speed-control in visual media

I must also give you some information about the methods of speed-control for film and
video. Pure sound archivists may run into difficulties here, because if you don’t
understand the source of the material, you may not realise you have something at the
wrong speed. And if your archive collects picture media as well, you need a general idea
of the history of synchronisation techniques, as these may also affect the speed of the
soundtrack. But if your archive doesn’t use any sound which has accompanied moving
pictures in the past, I suggest you jump to Section 5.4.
As this isn’t a history of film and video, I am obliged to dispense with the
incunabula of the subject, and start with the media which the European archivist is most
likely to encounter.
In silent-film days, cameras were generally hand-cranked, and the intended speed
of projection was between twelve and twenty frames per second. For the projector, the
shutter had two blades, or sometimes three; this chopped up the beam and raised the
apparent frequency of flicker so that it was above the persistence of vision. Moving
scenes did, of course, jerk past slower than that, but this was acceptable because the
brightness of cinema screens was lower in those days, and the persistence of vision of
human eyes increases in dim light.
But there was a sudden and quite definite switch to a higher frame-rate when
sound came along. This happened even when the sound was being recorded on disc
rather than film, so it seems that the traditional story of its being necessary to allow high
audio frequencies onto the film is wrong. I suspect that increasing viewing-standards in
cinemas meant the deficiencies of slower speeds were very obvious to all. So, with
motorised cameras now being essential if the accompanying soundtrack were to be
reproduced with steady pitch, the opportunity was taken for a radical change. Anyway,
nearly all sound films were intended for projection at 24 frames per second, and all studio
and location film crews achieved this speed by appropriate gearing in conjunction with
A.C. motors fed from a suitable A.C. supply.
There were two basic methods which were used for both recording and projection;
they ran in parallel for a couple of years, and films made by one process were sometimes
copied to the other. They were optical sound (which always ran at the same speed as the
picture, whether on a separate piece of celluloid or not), and coarsegroove disc (always
exactly 33 1/3rpm when the picture was exactly 24 frames per second). Most film crews
had separate cameras and sound recorders running off the same supply, and the clapper-
board system was used so the editor could match the two recordings at the editing-
bench.
Because location-filming often required generous supplies of artificial light, location
crews took a “power truck” with them to generate the power; but this does not mean the
A.C supply was more vulnerable to change, because of a little-known oddity. The speed
of 24 frames per second had the property of giving steady exposure whether the camera
looked at 50Hz or 60Hz lighting. If however the lights and the camera were running off
separate supplies, there was likely to be a cyclic change in the film exposure, varying from
perhaps once every few seconds to several times per second. Location electricians
therefore spent a significant amount of time checking that all the lights plus the picture
and sound cameras were all working off the same frequency of supply, wherever the
power actually came from.

84
Since the invention of the video camera, this fact has been “rediscovered”,
because 50Hz cameras give “strobing” under 60Hz lights and vice-versa. So, since the
earliest days of talkies, location power supplies have never been allowed to vary, or
strobing would occur. Thus we can assume fairly confidently that feature-films for
projection in the cinema are all intended to run at 24 frames per second. And whether the
sound medium is sepmag, sepopt, commag, comopt, or disc, it can be assumed that 24
fps working was the norm - until television came along, anyway.
But when television did come along in the late 1940s, this perfection was ruined.
Television was 25 fps in countries with 50Hz mains, and 30 fps in countries with 60Hz
mains, to prevent “rolling hum-bars” appearing on the picture. This remains generally
true to this day. (For the pedantic, modern colour NTSC pictures - as used in America and
Japan - are now 29.97 frames per second). In Europe we are so used to 50Hz lighting and
comparatively dim television screens that we do not notice the flicker; but visiting
Americans often complain at our television pictures and fluorescent lighting, as they are
not used to such low frequencies of flicker at home.
Before the successful invention of videotape (in America in 1956), the only way of
recording television pictures was “telerecording” (US: “kinetoscoping”) - essentially
filming a television screen by means of a film camera. Telerecording is still carried out by
some specialists, the technique isn’t quite dead. All current television systems use
“interlacing,” in which the scene is scanned in two passes called “fields” during one
frame, to cut down the effect of flicker. To record both halves of the frame equally, it is
necessary for the film camera to be exactly “locked” to the television screen, so that there
are precisely 25 exposures per second in 50Hz countries, and 30 (or later 29.97)
exposures per second in 60Hz countries.
So whether the sound medium is comopt, commag or sepmag, the speed of a
telerecording soundtrack is always either 25, 30 or 29.97 frames per second. Thus, before
you can handle an actual telerecording, you must know that it is a telerecording and not a
conventional film, and run it on the appropriate projector. A cinema-type projector will
always give the wrong speed.
The real trouble occurs when film and video techniques are mixed, for example
when a cinema film is shown on television. We must not only know whether we are
talking about a film or a telerecording, but we must also know the country of
transmission.
In Europe, feature films have always been broadcast at 25 frames per second.
Audio and video transfers from broadcast equipment are therefore a little over four
percent fast, just under a semitone. Thus music is always at the wrong pitch, and all voices
are appreciably squeaky. There is quite a lot of this material around, and unless you know
the provenance, you may mistakenly run it at the wrong speed. Keen collectors of film
music sometimes had their tape-recorders modified to run four percent faster on record,
or four percent slower on playback; so once again you have to know the provenance to
be certain.
Meanwhile, cinema films broadcast in 60Hz countries are replayed at the right
speed, using a technique known as “three-two pulldown.” The first 24fps frame is
scanned three times at 60Hz, taking one-twentieth of a second; the next frame is scanned
twice, taking one-thirtieth of a second. Thus two frames take one-twelfth of a second,
which is correct. But the pictures have a strange jerky motion which is very conspicuous to
a European; but Americans apparently don’t notice it because they’ve always had it.
Optical films shot specifically for television purposes usually differed from the
telerecording norm in America. They were generally 24 frames per second like feature

85
films. This was so such films could be exported without the complications of picture
standards-conversion. But in Europe, cameramen working for TV have generally had their
cameras altered so they shoot at 25 frames per second, like telerecordings. Thus stuff shot
on film for television is at the right speed in its country of origin; but when television films
cross the Atlantic in either direction they end up being screened with a four percent error.
Only within the last decade or so have some American television films been shot at 30
frames per second for internal purposes.
Up to this point I have been describing the conventional scenario. To fill in the
picture, I’m afraid I must also mention a few ways in which this scenario is wrong, so you
will be able to recognise the problems when they occur.
Although feature-films are made to world-wide standards, there was a problem
when commercial videos became big business from about 1982 onwards. Some videos
involving music have been made from American films (for example Elvis Presley movies),
and these were sometimes transferred at 24 fps to get the pitch right. This was done by
what is laughingly called “picture interpolation.” To show twenty-four frames in the time
taken for twenty-five, portions of the optical frame were duplicated at various intervals;
this can be seen by slow-motion analysis of the picture. The sound therefore came out
right, although the pictures were scrambled. In cases of doubt, still-frame analysis of a
PAL or SECAM video can be used as evidence to prove the audio is running correctly!
More often, it is considered preferable not to distort the picture. Here I cannot give
you a foolproof recipe. My present experience (1999) suggests that most of the time the
sound is four percent fast; but I understand some production-houses have taken to using
a “Lexicon” or “harmonizer” or other device which changes pitch independently of speed
(Ref. 1). Thus if the musical or vocal pitch is right and there are no video artefacts, it may
mean that the tempo of the performance is wrong.
But there have now been two more twists in the story. Sometimes American
television material is shot on film at 24 fps, transferred to 30 fps videotape for editing and
dubbing into foreign languages, and then subjected to electronic standards-conversion
before being sent to Europe. This gives the right speed of sound, but movement-
anomalies on the pictures; but again, you can regard the presence of movement
anomalies as evidence that the sound is right. The second twist came with Snell and
Wilcox’s “DEFT” electronic standards-converter, which has sufficient solid-state memory
to recognise when “three-two pulldown” has taken place. It is then possible to reverse-
engineer the effect to “two-two pulldown,” and copy steady images to a video recorder
running at 48 fields per second, ready for transmission on a conventional video machine
at 50Hz. Again, the steady pictures warn you something is wrong with the sound.

5.4 Setting the speed of old commercial sound records

In setting the speed of professional audio recordings, my opinion is that the first
consideration (which must predominate in the absence of any other evidence) is the
manufacturer’s recommended speed. For the majority of moulded commercial cylinder
records, this was usually 160 revolutions per minute; for most coarsegroove records, it
was about 80rpm until the late 1920s and then 78rpm until microgroove came along; for
magnetic tapes, it was usually submultiples of 60 inches per second. (Early
“Magnetophon” tapes ran a little faster than 30 inches per second, and this is thought to
apply to EMI’s earliest master-tapes made before about 1951. Ref. 2).

86
Unfortunately it isn’t always easy for the modern archivist to discover what the
recommended speed actually was. It does not always appear on the record itself, and if it
is mentioned at all it will be in sales literature, or instructions for playback equipment
made by the same company.
The recommended speeds of Edison commercial pre-recorded cylinders have been
researched by John C. Fesler (Ref. 3). The results may be summarised as follows:
1888-1892: 100rpm
Mid-1892 to at least 1st November 1899: 125rpm
June 1900 to the beginning of moulded cylinders: 144rpm
All moulded cylinders (late 1902 onwards): 160rpm.
It is also known that moulded cylinders by Columbia were intended to revolve at 160rpm,
and this forms the “baseline” for all moulded cylinders; so do not depart from 160rpm
unless there is good reason to do so.
The following is a list of so-called 78rpm discs which weren’t anywhere near 78, all
taken from “official” sources, contemporary catalogues and the like.

Berliner Gramophone Company. Instructions for the first hand-cranked


gramophones recommended a playing-speed of about 100rpm for the five-inch
records dating from 1890-1894, and 70rpm for the seven-inch ones dating from
about 1894-1900. But these are only “ballpark” figures.

Brunswick-Cliftophone (UK) records prior to 1927 were all marked 80rpm.


Since they were all re-pressings from American stampers, this would appear to fix
the American Brunswicks of this time as well.

Columbia (including Phoenix, Regal, and Rena): according to official


company memoranda, 80rpm for all recordings made prior to 1st September 1927,
from both sides of the Atlantic; 78rpm thereafter. But I should like to expand on
this. The company stated in subsequent catalogues that Columbia records should
be played “at the speed recommended on the label.” This is not quite true,
because sometimes old recordings were reissued from the original matrixes, and
the new versions were commonly labelled “Speed 78” by the printing department
in blissful ignorance that they were old recordings. The best approach for British
recordings is to use the matrix numbers. The first 78rpm ones were WA6100 (ten-
inch) and WAX3036 (twelve-inch). At this point I should like to remind you that I
am still talking about official speeds, which may be overridden by other evidence,
as we shall see in sections 5.6 onwards. Note too that Parlophone records, many of
which were pressed by Columbia, were officially 78.

Edison “Diamond discs” (hill-and-dale): 80rpm.

Grammavox: 77rpm. (The Grammavox catalogue was the pre-war


foundation for the better-known UK “Imperial” label; Imperial records numbered
below about 900 are in fact Grammavox recordings).

Vocalion: All products of the (British) Vocalion company, including


“Broadcast”, “Aco”, and “Coliseum”, and discs made by the company for
Linguaphone and the National Gramophonic Society, were officially 80rpm.

87
Finally, there are a few anomalous discs with a specific speed printed on the label. This
evidence should be adopted in the absence of any other considerations.
There also has to be a collection of “unofficial” speeds; that is to say, the results of
experience which have shown when not to play 78s at 78.

It is known that for some years the US Victor company recorded its master-
discs at 76rpm, so they would sound “more brilliant” when reproduced at the
intended speed of 78rpm. (This seems to be a manifestation of the syndrome
whereby musicians tune their instruments sharp for extra brilliance of tone). At the
1986 Conference of the Association of Recorded Sound Collections, George Brock-
Nannestad presented a paper which confirmed this. He revealed the plan was
mentioned in a letter from Victor to the European Gramophone Company dated
13th July 1910, when there had been an attempt to get agreement between the
two companies; but the Gramophone Company evidently considered this search
for artificial brilliance was wrong, and preferred to use the same speeds for
recording and playback. George Brock-Nannestad said he had confirmed Victor’s
practice upon several occasions prior to the mid-1920s.

Edison-Bell (UK) discs (including “Velvet Face” and “Winner”) tended to be


recorded on the high side, particularly before 1927 or so; the average is about
84rpm.

Pathé recordings before 1925 were made on master-cylinders and


transferred to disc or cylinder formats depending upon the demand. The speed
depends on the date of the matrix or mould, not the original recording. The earliest
commercial cylinders ranged from about 180rpm to as much as 200rpm, and then
they slowed to 160 just as the company switched to discs in 1906. The first discs to
be made from master-cylinders were about 90rpm, but this is not quite consistent;
two disc copies of Caruso’s famous master-cylinders acquired by Pathé, one
pressed in France and one in Belgium, have slightly different speeds. And some
Pathé disc sleeves state “from 90 to 100 revolutions per minute.” But a general
rule is that Pathé discs without a paper “label” (introduced about 1916) will have
to run at about 90rpm, and those with a paper label at about 80. The latter include
“Actuelle,” British “Grafton,” and some “Homochord.”

In 1951 His Master’s Voice issued their “Archive Series” of historic records
(VA and VB prefixes). The company received vituperation from collectors and
reviewers for printing “SPEED 78” in clear gold letters upon every label, despite
the same records having been originally catalogued with the words “above 78”
and “below 78.”

Quite often official recommended speeds varied from one record to the next. I will
therefore give you some information for such inconsistent commercial records.

Odeon, pre-1914. The English branch of the Odeon Record company, whose
popular label was “Jumbo”, were first to publicise the playing speeds for their disc
records. They attempted to correct the previous misdemeanours of their recording-
engineers in the trade magazine Talking Machine News (Vol.VI No.80, September
1908), in which the speeds of the then-current issues were tabulated.

88
Subsequently, Jumbo records often carried the speed on the label, in a slightly
cryptic manner (e.g. “79R” meant 79 revolutions per minute), and this system
spread to the parent-company’s Odeon records before the first World War. We
don’t know nowadays precisely how these speeds were estimated. And, although I
haven’t conducted a formal survey, my impression is that when an Odeon record
didn’t carry a speed, it was often because it was horribly wrong, and the company
didn’t want to admit it.

Gramophone, pre-1912. The leading record company in Europe was the


Gramophone Company, makers of HMV and Zonophone records. In about 1912
they decided to set a standard of 78rpm, this being the average of their
contemporary catalogue, and they also conducted listening experiments on their
back-catalogue. The resulting speed-estimates were published in catalogues and
brochures for some years afterwards; for modern readers, many can be found in
the David and Charles 1975 facsimile reprint “Gramophone Records of the First
World War.”
Experienced record-collectors soon became very suspicious of some of these
recommendations. But if we ignore one or two obvious mistakes, and the slight
errors which result from making voices recognisable rather than doing precise
adjustments of pitch, the present writer has a theory which accounts for the most
of the remaining results. Gramophones of 1912 were equipped with speed-
regulators with a central “78” setting and unlabelled marks on either side. It seems
there was a false assumption that one mark meant one revolution-per-minute. But
the marks provided by the factory were arbitrary, and the assumption gave an
error of about 60 percent; that is to say, one mark was about two-and-a-half rpm.
So when the judges said “Speed 76” (differing from 78 by two), they should have
said “Speed 73” (differing from 78 by five). If you’re confused, imagine how the
catalogue editors felt when the truth began to dawn. It’s not surprising they
decided to make the best of a bad job, and from 1928 onwards the rogue records
were listed simply as “above 78” or “below 78”. Nevertheless, it is a clear
indication to us today that we must do something!

Speeds were usually consistent within a recording-session. So you should not make
random speed-changes between records with consecutive matrix numbers unless there is
good reason to do so. But there are some exceptions. Sometimes one may find alternative
takes of the same song done on the same day with piano accompaniment and orchestral
accompaniment; these may appear to be at different speeds. This could be because the
piano was at a different pitch from the orchestra, or because a different recording-
machine was used. When a long side was being attempted, engineers would sometimes
tweak the governor of the recording-machine to make the wax run slower. I would
recommend you to be suspicious of any disc records made before 1925 which are
recorded right up to the label edge. These might have been cut slower to help fit the song
onto the disc.
I must report that so-called 78rpm disc records were hardly ever recorded at
exactly 78rpm anyway. The reason lies in the different mains frequencies on either side of
the Atlantic, which means that speed-testing stroboscopes gave slightly different results
when illuminated from the local mains supply, because the arithmetic resulted in decimals.
In America a 92-bar stroboscope suggests a speed of 78.26087rpm; in Europe a 77-bar
stroboscope suggests a speed of 77.922078rpm. The vast majority of disc recording lathes

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then ran at these speeds, which were eventually fixed in national (but not international)
standards. From now on, you should assume 78.26 for American recordings and 77.922
for European recordings whenever I use the phrase “78rpm disc.” A similar problem
occurs with 45rpm discs, but not 33 1/3s; stroboscopes for this speed can be made to
give exact results on either side of the Atlantic.

5.5 Musical considerations

My policy as a sound engineer is to start with the “official” speed, taking into account the
known exceptions given earlier. I change this only when there’s good reason to do so.
The first reason is usually that the pitch of the music is wrong.
It’s my routine always to check the pitch if I can, even on a modern recording.
(Originals are often replayed on the same machine, so a speed error will cancel out; thus
an independent check can reveal an engineering problem). I only omit it when I am
dealing with music for films or other situations when synchronisation is more important
than pitch. Copies of the two “Dictionaries of Musical Themes” may be essential
(unfortunately, there isn’t an equivalent for popular music). Even so, there are a number
of traps to setting the speed from the pitch of the music, which can only be skirted with
specialist knowledge.
The first is that music may be transposed into other keys. Here we must think our
way into the minds of the people making the recording. It isn’t easy to transpose; in fact it
can only be done by planning beforehand with orchestral or band accompaniments, and
it’s usually impossible with more than two or three singers. So transposition can usually be
ruled out, except for established or VIP soloists accompanied by an orchestra; for
example, Vera Lynn, whose deep voice was always forcing Decca’s arranger Mantovani to
re-score the music a fourth lower. Piano transposition was more frequent, though in my
experience only for vocal records. Even so, it happened less often than may be supposed.
Accompanist Gerald Moore related how he would rehearse a difficult song transposed
down a semitone, but play in the right key on the actual take, forcing his singer to do it
correctly. So transposition isn’t common, and it’s usually possible to detect when the
voice-quality is compared with other records of the same artist. For the modern engineer
the problem is to get sufficient examples to be able to sort the wheat from the chaff. A
more insidious trap is the specialist producer who’s been listening to the recordings of a
long-dead artist all his life, and who’s got it wrong from Day 1 !
A useful document is L. Heavingham Root’s article “Speeds and Keys” published in
the Record Collector Volume 14 (1961; pp. 30-47 and 78-93). This gives recommended
playing-speeds for vocal Gramophone Company records during the “golden years” of
1901 to 1908, but unfortunately a promised second article covering other makes never
appeared. Mr. Root gave full musical reasons for his choices. Although a scientific mind
would challenge them because he said he tuned his piano to C = 440Hz (when
presumably he meant A = 440Hz), this author has found his recommendations reliable.
Other articles in discographies may give estimated playing-speeds accurate to four
significant figures. This is caused by the use of stroboscopes for measuring the musically-
correct speed, which can be converted to revolutions-per-minute to a high degree of
accuracy; but musical judgement can never be more accurate than two significant figures
(about one percent), for reasons which will become apparent in the next few paragraphs.
Tighter accuracy is only necessary for matching the pitches of two different recordings
which will be edited together or played in quick succession.

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5.6 Strengths and weaknesses of “standard pitch”

The next difficulty lies in ascertaining the pitch at which musical instruments were tuned.
There has always been a tendency for musicians to tune instruments sharp for “extra
brilliance”, and there is some evidence that standard pitches have risen slowly but
consistently over the centuries in consequence. There were many attempts to standardise
pitch so different instruments could play together; but the definitive international
agreement did not come until 1939, after over four decades of recording using other
“standards.” You will find it fascinating to read a survey done just prior to the
International Agreement. Live broadcasts of classical music from four European countries
were monitored and measured with great accuracy. (Ref. 4). There were relatively small
differences between the countries, the averages varying only from A = 438.5 (England) to
A = 441.2 (Germany). Of the individual concerts, the three worst examples were all solo
organ recitals in winter, when the temperature made them go flat. When we discount
those, they were overshadowed by pitch variations which were essentially part of the
language of music. They were almost exactly one percent peak-to-peak. Then, as now,
musicians hardly ever play exactly on key; objective measurement is meaningless on
expressive instruments. Thus, a musically trained person may be needed to estimate what
the nominal pitch actually is.
Other instruments, such as piano-accordions and vibraphones, have tuning which
cannot be altered. When a band includes such instruments, everyone else has to tune to
them, so pitch variations tend to be reduced. So ensembles featuring these instruments
may be more accurate.
Instead of judging by ear, some operators may prefer the tuning devices which
allow pop musicians to tune their instruments silently on-stage. Korg make a very wide
range, some of which can also deal with obsolete musical pitches. They can indicate the
actual pitch of a guitar very accurately, so it could be possible to use one to measure a
recording. But they give anomalous results when there is strong vibrato or harmonics, so
this facility must be used with care, and only on recordings of instruments with “fixed
tuning” (by which I mean instruments such as guitars with frets, which restrict “bending”
the pitch as a means of musical expression). In other cases (particularly ensembles), I
consider a musical ear is more trustworthy.

5.7 Non-“standard” pitches

Before the 1939 Agreement, British concert pitch (called “New Philharmonic Pitch” or
“Flat Pitch”) was A = 435 at 60 degrees Fahrenheit (in practice, about 439 at concert-hall
temperatures). The International Agreement rounded this up to A = 440 at 68 degrees
(20 Celsius), and that was supposed to be the end of the matter. Nevertheless, it is known
that the Berlin Philharmonic and the Philadelphia orchestras use A = 444Hz today (a
Decca record-producer recalled that the Vienna Philharmonic was at this pitch in 1957,
with the pitch rising even further in the heat of the moment). The pianos used for
concertos with such orchestras are tuned even higher. This may partly be because the
pitch goes up in many wind instruments as the temperature rises, while piano strings tend
to go flatter. Concert-halls in Europe and America are usually warmer than 68 Fahrenheit;
it seems that only us poor Brits try using 440Hz nowadays!

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The next complication is that acoustic recording-studios were deliberately kept very
warm, so the wax would be soft and easy to cut. Temperatures of 90F were not
uncommon, and this would make some A=440 instruments go up to at least 450. On the
other hand, different instruments are affected by different amounts, those warmed by
human breath much less than others. (This is why an orchestra tunes to the oboe). The
human voice is hardly affected at all; so when adjusting the speed of wind-accompanied
vocal records, accurate voice-quality results from not adjusting to correct tuning pitch.
Thus we must make a cultural judgement: what are we aiming for, correct orchestral pitch
or correct vocal quality? For some types of music, the choice isn’t easy.
There are sufficient exceptions to A = 440 to fill many pages. The most important
is that British military bands and some other groups officially tuned to “Old Philharmonic
Pitch” or “Sharp Pitch” before 1929, with A at a frequency which has been given
variously as 452.5 and 454. Since many acoustic recordings have military band
accompaniments, this shows that A = 440 can be quite irrelevant. Much the same
situation occurred in the United States, but I haven’t been able to find out if it applied
elsewhere. Nor do I know the difference between a “military” band and a “non-military”
one; while it seems British ones switched to A = 440 in about 1964. Before that, it was
apparently normal for most British wind players to possess two instruments, a “low pitch”
one and a “high pitch” one, and it was clearly understood which would be needed well in
advance of a session or concert.
Of course, most ethnic music has always been performed at pitches which are
essentially random to a European, and the recent revival of “original instrumental
technique” means that much music will differ from standard pitch anyway.
Because of their location in draughty places, pipe organs tended to be much more
stable than orchestras. Many were tuned to Old Philharmonic Pitch when there was any
likelihood of a military band playing with them (the Albert Hall organ was a notorious
example). Because an organ is quite impossible to tune “on the night”, the speed of such
location recordings can actually be set more precisely than studio ones; but for perfect
results you must obviously know the location, the history of the organ, the date of the
record, and the temperature!
With ethnic music, it is sometimes possible to get hold of an example of a fixed-
pitch instrument and use it to set the speed of the reproduction. Many collectors of ethnic
music at the beginning of the century took a pitch-pipe with them to calibrate the cylinder
recordings they made. (We saw all sorts of difficulties with professionally-made cylinders
at the start of section 5.4, and location-recordings varied even more widely).
Unfortunately, this writer has found further difficulties here - the pitch of the pitch-pipe
was never documented, the cylinder was often still getting up to speed when it was
sounded, and I even worked on one collection where the pitch-pipe was evidently lost,
some cylinders were made without it, and then another (different) one was found! In
these circumstances, you can sometimes make cylinders play at the correct relative pitches
(in my case, “nailed down” more closely by judging human voices), but you cannot go
further. Sometimes, too, if you know what you’re doing, you can make use of collections
of musical instruments, such as those at the Horniman or Victoria & Albert museums.

5.8 The use of vocal quality

Ultimately, however, the test must be “does it sound right?” And with some material,
such as solo speech, there may be no other method. Professional tape editors like myself

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were very used to the noises which come from tape running at strange speeds, and we
also became used to the effect of “everything clicking into place”. One of the little games
I used to play during a boring editing-session was to pull the scrap tape off the machine
past the head, and try my luck at aiming for fifteen inches per second by ear, then going
straight into “play” to see if I’d got it right. Much of the time, I had. All the theories go
for nothing if the end-result fails to “gel” in this manner. Only when one is forced to do it
(because of lack of any other evidence) should one use the technique; but, as I say, this is
precisely when we must document why a solution has been adopted.
The operator must, however, be aware of two potential difficulties. The first is that
the average human being has become significantly bigger during the past century. It is
medically known that linear dimensions have increased by about five percent. (Weights,
of course, have increased by the cube of this). Thus the pitch of formants will be affected,
and there would be an even greater effect with the voices of (say) African pygmies.
John R. T. Davies also uses the “vocal quality” technique. He once tried to
demonstrate it to me using an American female singer. He asserted that it was quite easy
to judge, because at 78rpm the voice was “pinched”, as if delivered through lips clenched
against the teeth. I could hear the difference all right, but could not decide which was
“right”. It wasn’t until I was driving home that I worked out why the demonstration
didn’t work for me. I am not acquainted with any native American speakers, and I am not
a regular cinemagoer. So my knowledge of American speech has come from television,
and we saw earlier that American films are transmitted four percent faster in Europe. So I
had assumed that American vocal delivery was always like that. The point I’m making is
that personal judgements can be useful and decisive; but it’s vital for every individual to
work out for himself precisely where the limits of his experience lie, and never to go
beyond them.
Nevertheless I consider we should always take advantage of specialist experience
when we can. When John R. T. Davies isn’t certain what key the jazz band is playing in,
he gets out his trumpet and plays along with the improvisation to see what key gives the
easiest fingering. I sometimes ask a colleague who is an expert violinist to help me. She
can recognise the sound of an “open string”, and from this set the speed accurately by
ear. And, although the best pianists can get their fingers round anything, I am told it is
often possible to tell when a piano accompaniment has been transposed, because of the
fingering. This type of “evidence”, although indisputable, clearly depends upon specialist
knowledge.

5.9 Variable-speed recordings

So far, we’ve assumed that a record’s speed is constant. This is not always the case. On
78rpm disc records, the commonest problem occurs because the drag of the cutter at the
outside edge of the wax was greater than at the inside edge, so the master-record tended
to speed up as the groove-diameter decreased. I have found this particularly troublesome
with pre-EMI Columbias, though it can crop up anywhere. It is, of course, annoying when
you try to join up the sides of a multi-side work. But even if it’s only one side, you should
get into the habit of skipping the pickup to the middle and seeing if the music is at the
same pitch. On the other hand, some types of performances (particularly unaccompanied
singers and amateur string players) tend to go flatter as time goes by; so be careful. A
technique for solving this difficulty was mentioned in paragraph 6 of section 1.6; that is,

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to collect evidence of the performance of the disc-cutter from a number of sessions
around the same date.
Until about 1940 most commercial recording lathes were weight-powered,
regulated by a centrifugal governor like that on a clockwork gramophone. A well-
designed machine would not have any excess power capability, because there was a limit
to how much power could be delivered by a falling weight. The gearing was arranged to
give just enough power to cut the grooves at the outside edge of the disc, while the
governor absorbed little power itself. The friction-pad of a practical governor came on
gently, because a sudden “on/off” action would cause speed oscillations; so, as it was
called upon to absorb more power, the speed would rise slightly. By the time the cutter
had moved in an inch or two, the governor would be absorbing several times as much
power as it did at the start, and the proportion would remain in the governor’s favour. So
you will find that when this type of speed-variation occurs, things are usually back to
normal quite soon after the start.
The correction of fluctuating speeds is a subject which has been largely untouched
so far. Most speed variation is caused by defects in mechanical equipment, resulting in the
well-known “wow” and “flutter.” The former is slow speed variation, commonly less than
twenty times per second, and the latter comprises faster variations.
Undoubtedly, the best time to work on these problems is at the time of
transferring the sound off the original medium. Much “wow” is essentially due to the
reproduction process, e.g. eccentricity or warpage of a disc. One’s first move must be to
cure the source of the problem by re-centering or flattening the disc.
Unfortunately it is practically impossible to cure eccentric or warped cylinders. The
only light at the end of the tunnel is to drive the cylinder by a belt wrapped round the
cylinder rather than the mandrel, arranged so it is always leaving the cylinder tangentially
close to the stylus. (A piece of quarter-inch tape makes by far the best belt!) If the
mandrel has very little momentum of its own, and the pickup is pivoted in the same plane,
the linear speed of the groove under the pickup will be almost constant. But this will not
cure wow if differential shrinkage has taken place.
Another problem concerns cylinders with an eccentric bore. With moulded
cylinders the only “cure” is to pack pieces of paper between mandrel and cylinder to
bring it on-centre. But for direct-cut wax cylinders, the original condition should be
recreated, driving the mandrel rather than the surface (Ref. 5).
However, it is possible to use one source of wow to cancel another. For example, if
a disc has audible once-per-revolution recorded wow, you may be able to create an
equal-but-opposite wow by deliberately orienting the disc off-centre. This way, the
phases of the original wow and the artificial wow are locked together. This relationship
will be lost from any copy unless you invoke special synchronisation techniques.
It is often asked, “What are the prospects for correcting wow and flutter on a
digitised copy?” I am afraid I must reply “Not very good.” A great deal has been said
about using computers for this purpose. Allow me to deal with the difficulties, not
because I wish to be destructive, but because you have a right to know what will always
be impossible.
The first difficulty is that we must make a conscious choice between the
advantages and disadvantages. We saw in chapter 1 that the overall strategy should
include getting the speed right before analogue-to-digital conversion, to avoid the
generation of digital artefacts. Nevertheless it is possible to reduce the latter to any
desired degree, either by having a high sampling-frequency or a high bit-resolution. So
we can at least minimise the side-effects.

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Correction of “wow” in the digital domain means we need some way of telling the
processor what is happening. One way to do this objectively is to gain access to a
constant frequency signal recorded at the same time, a concept we shall explore for
steady-state purposes in the next section. But when the speed variations are essentially
random the possibilities are limited, mainly because any constant frequency signal is
comparatively weak when it occurs. To extract it requires sharp filtering, and we also need
to ignore it when it is drowned by wanted sound. Unfortunately, information theory tells
us we cannot detect rapid frequency changes with sharp filters. To make things worse, if
the constant frequency is weak, it will be corrupted by background noise or by variations
in sensitivity. Although slow wow may sometimes be correctable, I am quite sure we shall
never be able to deal with flutter this way. I am sorry to be pessimistic, but this is a
principle of nature; I cannot see how we shall ever be able correct random flutter from
any constant frequency which happens to be recorded under the sound.
But if the wow or flutter has a consistent element, for example due to an eccentric
capstan rotating twenty-five times per second in a tape recorder, then there is more hope.
In principle we could tell the computer to re-compute the sampling-frequency twenty-five
times per second and leave it to get on with it. The difficulty is “slippage.” Once the
recording is a fiftieth of a second out-of-sync, the wow or flutter will be doubled instead
of cancelled. This would either require human intervention (implying subjectivism), or
software which could distinguish the effect from natural pitch variations in the wanted
sound. The latter is not inconceivable, but it has not yet been done.
The computer may be assisted if the digital copy bears a fixed relationship to the
rotational speed of the original. Reverting to discs, we might record a once-per-revolution
pulse on a second channel. A better method is some form of rigid lock - so that one
revolution of a 77.92rpm disc always takes precisely 33957 samples, for example. (The US
equivalent would be a 78.26rpm disc taking 33776 samples). This would make it easier
for future programmers to detect cyclic speed variations in the presence of natural pitch-
changes, by accumulating and averaging statistics over many cycles. So here is another
area of development for the future.
Another is to match one medium to another. Some early LPs had wow because of
lower flywheel effect at the slower speed. But 78rpm versions are often better for wow,
while being worse for noise. Perhaps a matching process might combine the advantages
of both.
In 1990 CEDAR demonstrated a new algorithm for reducing wow on digitised
recordings, which took its information from the pitch of the music being played. Only
slow wow could be corrected, otherwise the process would “correct” musical vibrato!
Presumably this algorithm is impotent on speech, and personally I found I could hear the
side-effects of digital re-sampling. But here is hope when it’s impossible to correct the
fault at source. Unfortunately, CEDAR did not market the algorithm.
I hope this discussion will help you decide what to do when the problem occurs.
For the rest of this chapter, we revert to steady-state situations and situations where
human beings can react fast enough.

5.10 Engineering evidence

Sometimes we can make use of technical faults to guide us about speed-setting.


Alternating-current mains can sometimes get recorded - the familiar “background hum.”
In Europe the mains alternates at a nominal frequency of 50Hz, and in America the

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frequency is 60Hz. If it is recorded, we can use it to compensate for the mechanical
deficiencies of the machinery.
Before we can use the evidence intelligently, we must study the likelihood of errors
in the mains frequency. Nowadays British electricity boards are supposed to give advance
notice of errors likely to exceed 0.1 percent. Britain was fortunate enough to have a
“National Grid” before the second World War, giving nationwide frequency stability
(except in those areas not on 50Hz mains). Heavy demand would slow the generators, so
they had to be speeded up under light demand if errors were not to build up in
synchronous electric clocks. So the frequency might be “high” as well as “low.”
Occasional bombing raids during the second World War meant that isolated pockets of
Britain would be running independently of the Grid, but the overall stability is illustrated
by Reference 6, which shows that over one week in 1943 the peak error-rate was only 15
seconds in three hours, or less than 0.15 percent. (There might be worse errors for very
short periods of time, but these would be distinctly uncommon). After the war, the
Central Electricity Generating Board was statutorily obliged to keep its 50Hz supplies
within 2Hz in 1951 and 0.5Hz in 1971. My impression is that these tolerances were
extremely rarely approached, let alone exceeded. However there is ample evidence of
incompetent engineers blaming “the mains” for speed errors on their equipment.
An anecdote to illustrate that things were never quite as bad as that. In the years
1967-1971 I worked on a weekly BBC Radio programme lasting half-an-hour, which was
recorded using A.C. mains motors on a Saturday afternoon (normally a “light current
load” time), and reproduced during Monday morning (normally a “heavy load time,”
because it was the traditional English wash-day). The programmes consistently overran
when transmitted, but only by ten or fifteen seconds, an error of less than one percent
even when the cumulative errors of recording and reproduction were taken into account.
In over twenty-five years of broadcasting, I never came across another example like that.
However, I have no experience of mains-supplies in other countries; so I must urge you to
find the tolerances in other areas for yourself.
We do not find many cases of hum on professional recordings, but it is endemic on
amateur ones, the very recordings most liable to speed errors. So the presence of hum is a
useful tool to help us set the speed of an anomalous disc or tape; it can be used to get us
“into the right ballpark”, if nothing else. This writer has also found a lot of recorded hum
on magnetic wire recordings. This is doubly useful; apart from setting the “ballpark”
speed, its frequency can be used to distinguish between wires recorded with capstan-drive
and wires recorded with constant-speed takeup reels. But here is another can-of-worms;
the magnetic wire itself forms a “low-reluctance” path for picking up any mains hum and
carrying it to the playback head. It can be extremely difficult to hear one kind of hum in
the presence of the other.
Portable analogue quarter-inch tape-recorders were used for recording film sound
on location from the early 1960s. Synchronisation relied upon a reference-tone being
recorded alongside the audio, usually at 50Hz for 50Hz countries and 60Hz for 60Hz
countries. Back at base, this pilot-tone could be compared with the local mains frequency
used for powering the film recording machines, so speed variations in the portable unit
were neutralised. In principle it might be possible to extract accidental hum from any
recording and use it to control a playback tape-recorder in the same way. This is another
argument in favour of making an “archive copy” with warts and all; the hum could be
useful in the future.
We can sometimes make use of a similar fault for setting the speed of a television
soundtrack recording. The “line-scan” frequency of the picture sometimes gets recorded

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amongst the audio. This was 10125Hz for British BBC-1 and ITV 405-line pictures until
1987; 15625Hz for 625-line pictures throughout the world (commencing in 1963 in
Britain); 15750Hz for monochrome 525-line 30Hz pictures, and 15734.25Hz for colour
NTSC pictures. These varied only a slight amount. For example, before frame-stores
became common in 1985, a studio centre might slew its master picture-generator to
synchronise with an outside broadcast unit. This could take up to 45 seconds in the worst
possible case (to achieve almost a full frame of slewing). Even so, this amounts to less
than one part in a thousand; so speed-setting from the linescan frequency can be very
accurate. In my experience such high frequencies are recorded rather inefficiently, and
only the human ear can extract them reliably enough to be useful; so setting the speed
has to be done by ear at present.
Although my next remark refers to the digitisation procedures in Chapter 2, it is
worth noting that the embedding of audio in a digital video bitstream means that there
must be exactly 1920 digital samples per frame in 625-line television, and exactly 8008
samples per five frames in NTSC/525-line systems.
The 19kHz of the stereo pilot-tone of an FM radio broadcast (1st June 1961
onwards in the USA, 1962 onwards in Britain) can also get recorded. This does not vary,
and can be assumed to be perfectly accurate - provided you can reproduce it.
John Allen has even suggested that the ultrasonic bias of magnetic tape recording
(see section 9.3) is sometimes retained on tape well enough to be useful. (Ref. 7). We
usually have no idea what its absolute frequency might be; but it has been suggested that
variations caused by tape speed-errors might be extracted and used to cancel wow and
flutter. I have already expressed my reasons why I doubt this, but it has correctly been
pointed out that, provided it’s above the level of the hiss (it usually isn’t), this information
should not be thrown away, e. g. by the anti-aliasing circuit of a digital encoder. Although
it may be necessary to change the frequency down and store it on a parallel track of a
multi-channel digital machine, we should do so. Again, it’s a topic for the future; but it
seems just possible that a few short-term tape speed variations might be compensated
objectively one day.
There is one caveat I must conclude with. The recording of ultrasonic signals is
beset with problems, because the various signals may interfere with each other and result
in different frequencies from what you might expect. For example, the fifth harmonic of a
television linescan at 78125Hz might beat with a bias frequency of 58935Hz, resulting in
a spurious signal at 19190Hz. If you did not know it was a television soundtrack, this
might be mistaken for a 19kHz radio pilot-tone, and you’d end up with a one percent
speed error when you thought you’d got it exactly right. So please note the difficulties,
which can only be circumvented with experience and a clear understanding of the
mechanics.

5.11 Timings

There’s a final way of confirming an overall speed, by timing the recording. This is useful
when the accompanying documentation includes the supposed duration. Actually, the
process is unreliable for short recordings, because if the producer was working with a
stopwatch, you would have to allow for reaction-time, the varying perception of decaying
reverberation, and any “rounding errors” which might be practised. So short-term timings
would not be reliable enough. But for longer recordings, exceeding three or four minutes,

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the documentation can be a very useful guide to setting a playing-speed. The only trouble
is that it may take a lot of trial-and-error to achieve the correct timing.
I hope this chapter will help you to assess the likelihood, quantity, and sign of a
speed error on a particular recording. But I conclude with my plea once again. It seems to
me that the basis of the estimate should also be documented. The very act of logging the
details forces one to think the matter through and helps against omitting a vital step. And
it’s only right and proper that others should be able to challenge the estimate, and to do
so without going through all the work a second time.

REFERENCES

1: anon., “Lexiconning” (article), Sight and Sound Vol. 57 No. 1 (Winter 1987/8), pp. 47-
8. It should be noted this article’s main complaint was a film speeded by ten percent.
A Lexicon would be essential to stop a sound like “Chipmunks” at this rate,
although the same technology could of course be used for a four-percent change.
2: Friedrich Engel (BASF), Letter to the Editor, Studio Sound, Vol. 28 No. 7 (July 1986), p.
147.
3: John C. Fesler, London: Hillandale News, No. 125 (April 1982), p. 21.
4: Balth van der Pol and C. C. J. Addink, “Orchestral Pitch: A Cathode-Ray Method of
Measurement during a Concert” (article), Wireless World, 11th May 1939, pp.
441-2.
5: Hans Meulengracht-Madsen, “On the Transcription of Old Phonograph Wax Records”
(paper), J.A.E.S., Jan/Feb 1976.
6: H. Morgan, “Time Signals” (Letter to the Editor), Wireless World, Vol. L No. 1 (January
1944), p. 26.
7: John S. Allen, “Some new possibilities in audio restoration,” (article), ARSC Journal,
Volume 21 No. 1 (Spring 1990), page 44.

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6 Frequency responses of grooved media

6.1 The problem stated

The subject of this chapter raises emotions varying from deep pedantry to complete lack
of understanding. Unfortunately, there has never been a clear explanation of all the issues
involved, and the few scraps of published material are often misquoted or just plain
wrong, whilst almost-perfect discographical knowledge is required to solve problems from
one recording-company to the next. Yet we need a clear understanding of these issues to
make acceptable “service copies”, and we are compelled to apply the lessons rigorously
for “objective copies”. (My research shows we now have the necessary level of
understanding for perhaps seventy-five percent of all grooved media before International
Standards were developed). But for the “warts-and-all” copy, it isn’t an issue of course.
Grooved disc records have never been recorded with a flat frequency response.
The bass notes have always been attenuated in comparison with the treble, and when
electrical methods are used to play the records back, it is always implied that the bass is
lifted by a corresponding amount to restore the balance. You may like to demonstrate the
effect using your own amplifier. Try plugging a good quality microphone into its “phono”
input, instead of a pickup-cartridge. If you do, you will notice a boomy and muffled sound
quality, because the phono circuitry is performing the “equalisation” function, which will
not happen when you use a “Mic” input.
The trouble is that the idea of deliberately distorting the frequency response only
took root gradually. In the days of acoustic recording (before there was any electronic
amplification), it was a triumph to get anything audible at all; we shall be dealing with this
problem in Chapter 11. Then came the first “electrical recording” systems. (I shall define
this phrase as meaning those using an electronic amplifier somewhere - see Ref. 1 for a
discussion of other meanings of the phrase, plus the earliest examples actually to be
published). At first, these early systems were not so much “designed”, as subject to the
law of “the survival of the fittest.” It was some years before objective measurements
helped the development of new systems.
This chapter concentrates on electrical recordings made during the years 1925 to
1955, after which International Standards were supposed to be used. I shall be showing
equalisation curves the way the discs were recorded. If you are interested in restoring the
sound correctly, you will have to apply equalisation curves which are the inverse of these;
that is, the bass needs to be boosted rather than cut.
The major reason for the importance of this issue is different from the ones of
restoring the full “power-bandwidth product” that I mentioned in Chapter 1. Incorrect
disc equalisation affects sounds right in the middle of the frequency range, where even
the smallest and lowest-quality loudspeaker will display them loud and clear – usually at a
junction between a modern recording and an old one. The resulting “wooliness” or
“harshness” will almost always seem detrimental to the archived sound.

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6.2 A broad history of equalisation

Electrical recording owed its initial success to the Western Electric recording system.
Although this was designed using scientific principles to give a “flat frequency response,”
it had at least one undefined bass-cut which needs correction today, and other features if
we are ever to achieve “high fidelity” from its recordings. So its success was partly
accidental. The recording equipment dictated the equalisation, rather than the other way
round.
During the next twenty years the whole process of making an acceptable record
was a series of empirical compromises with comparatively little scientific measurement.
During the Second World War accurate methods of measurement were developed, and
after the war the knowledge of how to apply these to sound reproduction became more
widely known. Thus it became possible to set up “standards”, and modify equipment
until it met those standards. Thus professionals (and, later, hi-fi fanatics) could exchange
recordings and know they would be reproduced correctly.
This last phase is particularly troublesome. There were nine “standards” which
users of disc recording equipment were invited to support between 1941 and 1953, and
the ghastly details will be listed in sections 6.62 onwards. If you put your political
thinking-cap on, and conclude that such chaos is typical of a Free Market Economy, I
reply that State Monopolies could be just as bad. For example, between 1949 and 1961
the British Broadcasting Corporation had three “standards” used at once, none of which
were International ones!
Most record manufacturers had different recipes which we can describe in scientific
language. The number of recipes isn’t just because of the “Not Invented Here” syndrome,
but there was at least one manufacturer who kept his methods a trade secret because he
feared his competitive advantage would be harmed! Two international standards were
established in 1955, one for coarsegroove records and one for microgroove records. The
latter has several names, but most people call it by the name of the organisation which
promoted it, the Recording Industries Association of America. It is on any present-day
“Phono Input,” and I shall call it “RIAA” from now on.
So if you are interested in the faithful reproduction of pre-1955 records, you
should at least know that an “equalisation problem” may exist.

6.3 Why previous writers have gone wrong

This section is for readers who may know something already. It summarises three areas in
which I believe previous writers have got things wrong, so you can decide whether to
read any more.
(1) Equalisation is largely independent of the make of the disc. It depends only upon who
cut the master-disc and when. (I shall be using the word “logo” to mean the trademark
printed on the label, which is something different again!) I’m afraid this implies you
should be able to detect “who cut the master-disc and when” by looking at the disc, not
the logo. In other words, you need discographical knowledge. I’m afraid it’s practically
impossible to teach this, which may explain why so many previous writers have made
such a mess of things.
(2) It is best to define an equalisation curve in unambiguous scientific language. American
writers in particular have used misleading language, admittedly without committing gross

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errors along the way. I shall be using “microseconds”, and shall explain that at the end of
section 6.7 below.
(3) The names of various “standards” are themselves ambiguous. For instance, when
International Standards became operational in 1955, most old ones were re-named “the
new Bloggs characteristic” or words to that effect. I recently found a microgroove nitrate
dated 24-1-57 whose label bore the typed message: “Playback: New C.C.I.R., A.E.S.,
Orthoacoustic.” (This was clearly RIAA, of course!) Similar considerations apply to curves
designed for one particular format (for example, American Columbia’s pioneering long-
playing disc curve of 1948), which may be found on vintage pre-amplifiers simply called
“LP” only - or worse still “Columbia” only - when neither name is appropriate, of course.

6.4 Two ways to define “a flat frequency response”

Equalisation techniques are usually a combination of two different systems, known for
short as “constant velocity” and “constant amplitude.” The former, as its name implies,
occurs when the cutting stylus vibrates to and fro at a constant velocity whatever the
frequency, provided the volume remains the same. This technique suited an “ideal”
mechanical reproducing machine (not using electronics), such as the Orthophonic Victrola
and its HMV equivalent gramophone of 1926. These scientifically-designed machines
approached the ideal very closely. On such records, as the frequency rises the amplitude
of the waves in the grooves falls, so the high frequencies are vulnerable to surface noise.
On the other hand low frequencies cause high amplitudes, which have the potential for
throwing the needle out of the groove (Fig. 1a). Thus all disc records are given some
degree of bass cut compared with the idealised constant-velocity technique.

Fig 1a. Constant Velocity waveshape Fig 1b Constant Amplitude waveshape


These two diagrams depict how two musical notes, the second an octave above the first,
would be cut onto lateral-cut discs using these two different systems.

Constant-amplitude recording overcomes both these difficulties. Given constant


input, if varying frequencies are cut, the amplitude of the waves in the groove stays the
same (Fig. 1b). Thus the fine particulate matter of the record is always overshadowed and
the hiss largely drowned, while the low notes are never greater than the high notes and
there is less risk of intercutting grooves. Unfortunately, the result sounded very shrill upon
a clockwork gramophone.
Most record-companies therefore combined the two systems. In the years 1925-
1945, most European record companies provided constant-velocity over most of the

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frequency range to give acceptable results on acoustic gramophones, but changed to
constant-amplitude for the lower frequencies (which were generally below the lower limit
of such machines anyway) to prevent the inter-cutting difficulty. The scene was different
in America, where the higher standard of living encouraged electrical pickups and
amplifiers, and it was possible to use a greater proportion of constant-amplitude thanks to
electronic compensation. From the mid-1930s, not only did many record manufacturers
use a higher proportion of constant-amplitude, but another constant-amplitude section
may have been added at high frequencies, which is usually called “pre-emphasis.” More
high-frequency energy is recorded than with the “constant-velocity” system. Thus the
wanted music dominates hiss and clicks, which are played back greatly muffled without
the music being touched. An equivalent process occurs today on FM broadcasts, TV
sound, and some digital media.
In principle, magnetic pickups (not crystal or ceramic ones) give a constant voltage
output when playing constant-velocity records. But this can be converted to the
equivalent of playing a constant-amplitude record by applying a treble cut (and/or a bass
boost) amounting to 6 decibels per octave. This can be achieved in mono with just two
simple components - a resistor and a capacitor - so it is a trivial matter to convert an
electronic signal from one domain to the other.
What exists on most electrically-recorded discs can be defined by one or more
frequencies at which the techniques change from constant-amplitude to constant-
velocity, or vice versa. “Phono” equalisers are designed to apply 6dB/octave slopes to the
appropriate parts of the frequency spectrum, so as to get an overall flat frequency
response. And graphs are always drawn from the “velocity” viewpoint, so constant-
velocity sections form horizontal lines and constant-amplitude sections have gradients.
If you wish to reproduce old records accurately, I’m afraid I shan’t be giving you
any circuit diagrams, because it depends very much on how you propose to do the
equalisation. Quite different methods will be needed for valves, transistors, integrated
circuits, or processing in the digital domain; and the chronology of the subject means you
will only find circuits for valve technology anyway. Personally, I don’t do any of those
things! I equalise discs “passively,” using no amplification at the equalisation stage at all;
but this implies neighbouring circuitry must have specific electrical impedances. This
technique automatically corrects the relative phases (section 2.11), whether the phase
changes were caused by acoustic, mechanical, or electronic processes in the analogue
domain.
Next we have the problem of testing such circuitry. Over the years, many record
companies have issued Frequency Test Discs, documenting how they intended their
records to be reproduced. (Sometimes they published frequency response graphs with the
same aim, although we don’t actually know if they were capable of meeting their own
specifications!). Such published information is known as “a Recording Characteristic,” and
I follow Terry’s definition of what this means (Ref. 2): “The relation between the R.M.S
electrical input to the recording chain and the R.M.S velocity of the groove cut in the
disc.” This gives rise to the following thoughts.

6.5 Equalisation ethics and philosophy

In the late ’twenties, a flat frequency response seems to have been the dominant
consideration in assessing fidelity. Regardless of any other vices, a piece of equipment
with a wide flat frequency range was apparently described as “distortionless.” (Ref. 3).

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Unless there is definite evidence to the contrary, we should therefore assume that 1920s
engineers wanted their recordings to be reproduced to a flat frequency response, with the
deficiencies of their equipment reversed as far as possible. This would certainly be true
until the mid-thirties, and is sometimes true now; but there is a counter-argument for later
recordings.
I mentioned how empirical methods ruled the roost. At the forefront of this process
was the session recording engineer, who would direct the positions of the performers,
choose a particular microphone, or set the knobs on his control-desk, using his judgement
to get the optimum sound. As archivists, we must first decide whether we have the right
to reverse the effects of his early microphones or of his controls.
The engineer may have had two different reasons behind his judgements, which
need to be understood. One is an aesthetic one - for example, using a cheap
omnidirectional moving-coil microphone instead of a more expensive one on a piano,
because it minimises thumping noises coming from the sounding-board and clarifies the
“right hand” when it is mixed with other microphones. Here I would not advocate using
equalisation to neutralise the effects of the microphone, because it was an “artistic”
judgement to gain the best overall effect.
The other is to fit the sound better onto the destination medium. In 78rpm days,
the clarity would perhaps be further enhanced to get it above the hiss and scratch of the
shellac. For reproduction of the actual 78 this isn’t an issue; but if you are aiming to move
the sound onto another medium, e.g. cassette tape or compact disc, it might be
acceptable to equalise the sound to make it more faithful, so it suits today’s medium
better. (After all, this was precisely how the original engineer was thinking). This implies
we know what the original engineer actually did to fit the sound onto his 78. We either
need industrial archaeology, or the original engineer may be invited to comment (if he’s
available). I am very much against the principle of one lot of empirical adjustments being
superimposed on someone else’s empirical adjustments.
Personally I take the above argument to its logical conclusion, and believe we
should only compensate for the microphone and the controls if there were no satisfactory
alternatives for the recording engineer. Thus, I would compensate for the known
properties of the 1925 Western Electric microphone, because Western Electric licencees
had only one alternative. It was so conspicuously awful that contemporary engineers
almost never used it. But listeners soon discovered that the silky sound of Kreisler’s violin
had been captured more faithfully by the acoustic process