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Praise for the second edition
“Self provides solid, well-explained technical information throughout the book, all gained from
years of experience and a thorough understanding of the entire topic (. . .) His book exudes skilful
engineering on every page, and I found it a very refreshing, enjoyable, and inspirational read (. . .)
if you have the slightest interest in audio circuit design this book has to be considered an essential
reference. Very highly recommended”.
—Hugh Robjohns, Sound on Sound Magazine
“This book presents a large body of knowledge and countless insider-tips from an award-winning
commercial audio designer (. . .) Douglas Self dumps a lifetime’s worth of thoroughly-tested
audio circuit knowledge into one biblical tome”.
—Joseph Lemmer, Tape Op
Small Signal Audio Design
Small Signal Audio Design is a highly practical handbook providing an extensive repertoire
of circuits that can be assembled to make almost any type of audio system. The publication of
Electronics for Vinyl has freed up space for new material (though this book still contains a lot on
moving-magnet and moving-coil electronics), and this fully revised third edition offers wholly
new chapters on tape machines, guitar electronics, and variable-gain amplifiers, plus much more.
A major theme is the use of inexpensive and readily available parts to obtain state-of-the-art
performance for noise, distortion, crosstalk, frequency response accuracy, and other parameters.
Virtually every page reveals nuggets of specialised knowledge not found anywhere else. For
example, you can improve the offness of a fader simply by adding a resistor in the right place –if
you know the right place.
Essential points of theory that bear on practical audio performance are lucidly and thoroughly
explained, with the mathematics kept to an absolute minimum. Self’s background in design for
manufacture ensures he keeps a wary eye on the cost of things.
This book features the engaging prose style familiar to readers of his other books. You will learn
why mercury-filled cables are not a good idea, the pitfalls of plating gold on copper, and what
quotes from Star Trek have to do with PCB design.
Learn how to:
• make amplifiers with apparently impossibly low noise
• design discrete circuitry that can handle enormous signals with vanishingly low distortion
• use humble low-gain transistors to make an amplifier with an input impedance of more
than 50 megohms
• transform the performance of low-cost opamps
• build active filters with very low noise and distortion
• make incredibly accurate volume controls
• make a huge variety of audio equalisers
• make magnetic cartridge preamplifiers that have noise so low it is limited by basic physics,
by using load synthesis
• sum, switch, clip, compress, and route audio signals
• be confident that phase perception is not an issue
This expanded and updated third edition contains extensive new material on optimising RIAA
equalisation, electronics for ribbon microphones, summation of noise sources, defining system
frequency response, loudness controls, and much more. Including all the crucial theory, but
with minimal mathematics, Small Signal Audio Design is the must-have companion for anyone
studying, researching, or working in audio engineering and audio electronics.
Douglas Self studied engineering at Cambridge University then psychoacoustics at Sussex
University. He has spent many years working at the top level of design in both the professional
audio and hi-fi industries and has taken out a number of patents in the field of audio technology.
He currently acts as a consultant engineer in the field of audio design.
Small Signal Audio Design
Third edition
Douglas Self
Third edition published 2020
by Routledge
52 Vanderbilt Avenue, New York, NY 10017
and by Routledge
2 Park Square, Milton Park, Abingdon, Oxon, OX14 4RN
The right of Douglas Self to be identified as author of this work has been asserted by him in
accordance with sections 77 and 78 of the Copyright, Designs and Patents Act 1988.
All rights reserved. No part of this book may be reprinted or reproduced or utilised in any form
or by any electronic, mechanical, or other means, now known or hereafter invented, including
photocopying and recording, or in any information storage or retrieval system, without
permission in writing from the publishers.
Typeset in Times
by Apex CoVantage, LLC
Dedication
To Julie, with all my love
Contents Contents
Preface � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � �xxiv
Acknowledgments � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � �xxviii
ix
x Contents
Chapter 2 Components � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � 43
Conductors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .43
Copper and Other Conductive Elements. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .43
The Metallurgy of Copper. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .45
Gold and Its Uses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .46
Cable and Wiring Resistance. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .46
PCB Track Resistance. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .47
PCB Track-to-Track Crosstalk. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .49
The Three-Layer PCB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .51
Impedances and Crosstalk: A Case History. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .52
Resistors. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .53
Through-Hole Resistors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .54
Surface-Mount Resistors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .54
Resistor Series. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .56
Resistor Accuracy: Two-Resistor Combinations. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .57
Resistor Accuracy: Three-Resistor Combinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .62
Other Resistor Combinations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .62
Resistor Value Distributions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .64
The Uniform Distribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .66
Resistor Imperfections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .67
Resistor Excess Noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .67
Resistor Non-Linearity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .70
Contents xi
Capacitors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .73
Capacitor Series . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .74
Capacitor Non-Linearity Examined . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .75
Non-Electrolytic Capacitor Non-Linearity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .75
Electrolytic Capacitor Non-Linearity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .80
Inductors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .83
Appendix � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � 745
Index� � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � � 747
Preface Preface
“Another damned thick book! Always scribble, scribble, scribble! Eh, Mr. Gibbon?”
Attributed to Prince William Henry, Duke of Gloucester, in 1781 upon receiving the second
volume of The History of the Decline and Fall of the Roman Empire from its author.
This book deals with small-signal audio design; the amplification and control of audio in the
analogue domain, where the processing is done with opamps or discrete transistors, usually
working at a nominal level of a volt or less. It constitutes a major update of the second edition.
“Small-signal design” is the opposite term to “large-signal design”, which in audio represents
power amplifiers driving loudspeakers rather than the electricity distribution grid or lightning.
As stated on the back cover of this book, the publication of Electronics For Vinyl [1] allowed the
vinyl-oriented material in this book to be much reduced, so the space freed can be used for new
material. All the phono material that was in the second edition of Small Signal Audio Design is
in Electronics For Vinyl, plus a great deal more. Therefore the chapters on moving-magnet inputs
have been reduced to one (Chapter 9). This cannot give comprehensive coverage of a very big
subject, but it does give the most important information with many pointers to where very much
more can be found in Electronics For Vinyl. Chapter 9 also contains new information that has been
acquired since EFV was written.
So much has been added to this edition that it is difficult to summarise, but the new material
includes:
xxiv
Preface xxv
• More on loudness controls. Loudness controls are currently unfashionable, but the
thinking behind them is intriguing. I include a possible solution to the mystery of why
almost everyone disliked them, when consensus of any sort is rare in the hi-fi business.
• Combining tone and balance controls in one stage
• Adjusting the Q of mid tone controls
• Electronics for ribbon microphones
• Bootstrapped balanced line inputs
• More on ground-cancelling outputs
• Improving fader offness
• Crosstalk cancellation in mixers
There is unquestionably a need for high-quality analogue circuitry. For example, a good
microphone preamplifier needs a gain range from 0 to +80 dB if it is to get any signal it is likely
to encounter up to a workable nominal value. There is clearly little prospect of ever being able
to connect an A-to-D converter directly to a microphone. The same applies to other low-output
transducers such as moving-coil and moving-magnet phono cartridges. If you are starting at
line level and all you need is a simple but high-quality tone control, there is little incentive to
convert to digital via a relatively expensive ADC, perform the very straightforward arithmetic
manipulations in the digital domain, then go back to analogue via a DAC; there is also the need
to implement the actual controls as rotary encoders and have those overseen by a microcontroller.
All digital processing involves some delay, because it takes time to do the calculations; this is
called the latency and can cause serious problems if more than one signal path is involved.
The total flexibility of digital signal processing certainly allows greater scope –you might
contemplate how to go about implementing a 1-second delay in the analogue domain, for
example –but there are many times when greater quality or greater economy can be obtained by
keeping the signal analogue. Sometimes analogue circuitry connects to the digital world, and so a
complete chapter of this book deals with the subtleties of analogue/digital interfacing.
Therefore analogue circuitry is often the way to go. This book describes how to achieve high
performance without spending a lot of money. As was remarked in a review of my recent book
Active Crossover Design, duplicating this performance in the digital domain is not at all a trivial
business. You can of course start off in analogue, and when you have identified the filter slopes,
equalisation curves, and whatnot that you want, it is relatively easy to move it over to the DSP
world.
I have devoted the first few chapters to the principles of high-quality small-signal design, moving
on to look closely at first hi-fi preamplifiers and then mixing consoles. These two genres were
chosen partly because they are of wide interest in themselves but mainly because they use a large
number of different functional blocks, with very little overlap between them. They cover a wide
range of circuit functions that will be useful for all kinds of audio systems. You will find out how
to adapt or design these building blocks for audio and how to put them together to form a system
xxvi Preface
without bad things happening due to loading or interaction. You should then be able to design
pretty much anything in this field.
In the pursuit of high quality at low cost, certain principles pervade this book. Low-impedance
design reduces the effects of Johnson noise and current noise without making voltage noise
worse; the only downside is that a low impedance requires an opamp capable of driving it
effectively, and sometimes more than one. The most ambitious application of this approach so far
has been in the ultra-low noise Elektor 2012 Preamplifier.
Another principle is that of using multiple components to reduce the effects of random noise. This
may be electrical noise, in which case the outputs of several amplifiers are averaged (very simply
with a few resistors) and the noise from them is partially cancelled. Multiple amplifiers are also
very useful for driving the low impedances just mentioned. Alternatively, it may be numerical
noise, such as tolerances in a component value; making up the required value with multiple parts
in series or parallel also makes errors partially cancel. This technique has its limits because of
the square-root way it works; four amplifiers or components are required to halve the noise, 16 to
reduce it to a quarter and so on. Multiple parts also allow very precise non-standard values to be
achieved.
There is also the principle of optimisation, in which each circuit block is closely scrutinised to see
if it is possible to improve it by a bit more thinking. One example is the optimisation of Recording
Industry Association of America (RIAA) equalisation networks. There are four ways to connect
resistors and capacitors to make an RIAA network, and I have shown that one of them requires
smaller values of expensive precision capacitors than the others. This finding is presented in
detail in Chapter 9, along with related techniques of optimising resistor values to get convenient
capacitor values.
In many places hybrid amplifiers combining the virtues of discrete active devices and opamps
are used. If you put a bipolar transistor before an opamp, you get lower noise, but the loop gain
of the opamp means the distortion is as good as the opamp alone. This is extremely useful for
making microphone amplifiers, tape replay amplifiers, and virtual-earth summing amplifiers. If
you reverse the order, with an opamp followed by bipolar transistors, you can drive much heavier
loads, with the opamp gain once again providing excellent linearity. This latter technology, among
others, is explained in a chapter on headphone amplifiers.
However, what you most emphatically will not find here is any truck with the religious dogma of
audio subjectivism; the directional cables, the oxygen-free copper, the World War I vintage triodes
still spattered with the mud of the Somme, and all the other depressing paraphernalia of pseudo-
and anti-science. I have spent more time than I care to contemplate in double-blind listening tests
–properly conducted ones, with rigorous statistical analysis –and every time the answer was that if
you couldn’t measure it, you couldn’t hear it. Very often if you could measure it you still couldn’t
hear it. However, faith-based audio is not going away any time soon, because few people (apart of
course from the unfortunate customers) have any interest in it so doing; you can bet your bottom
diode on that. If you want to know more about my experiences and reasoning in this area, there is
a full discussion in my book Audio Power Amplifier Design (sixth edition).
Preface xxvii
A good deal of thought and experiment has gone into this book, and I dare to hope that I have
moved analogue audio design a bit further forward. I hope you find it useful. I hope you enjoy it
too.
I have a website at www.dself.dsl.pipex.com, where I will be adding supplementary material to
this book.
Further information and PCBs, kits and built circuit boards of some of the designs described here,
such as phono input stages and complete preamplifiers, can be found at: www.signaltransfer.
freeuk.com.
Douglas Self
London, October 2019
Acknowledgments Acknowledgments
xxviii
CHAPTER 1
Signals
An audio signal can be transmitted as either a voltage or a current. The construction of the
universe is such that almost always the voltage mode is more convenient; consider for a moment
an output driving more than one input. Connecting a series of high-impedance inputs to a low-
impedance output is simply a matter of connecting them in parallel, and if the ratio of the output
and input impedances is high there will be negligible variations in level. To drive multiple inputs
with a current output it is necessary to have a series of floating current-sensor circuits that can
be connected in series. This can be done [1], as pretty much anything in electronics can be done,
but it requires a lot of hardware and probably introduces performance compromises. The voltage-
mode connection is just a matter of wiring.
Obviously, if there’s a current, there’s a voltage, and vice versa. You can’t have one without the
other. The distinction is in the output impedance of the transmitting end (low for voltage mode,
high for current mode) and in what the receiving end responds to. Typically but not necessarily, a
voltage input has a high impedance; if its input impedance was only 600 Ω, as used to be the case
in very old audio distribution systems, it is still responding to voltage, with the current it draws
doing so a side issue, so it is still a voltage amplifier. In the same way, a current input typically but
not necessarily has a very low input impedance. Current outputs can also present problems when
they are not connected to anything. With no terminating impedance, the voltage at the output
will be very high and probably clipping heavily; the distortion is likely to crosstalk into adjacent
circuitry. An open-circuit voltage output has no analogous problem.
Current-mode connections are not common. One example is the Krell Current Audio Signal
Transmission (CAST) technology, which uses current mode to interconnect units in the Krell
product range. While it is not exactly audio, the 4–20 mA current loop format is widely used in
instrumentation. The current-mode operation means that voltage drops over long cable runs are
ignored, and the zero offset of the current (i.e. 4 mA = zero) makes cable failure easy to detect: if
the current suddenly drops to zero, you have a broken cable.
The old DIN interconnection standard was a form of current mode connection in that it had
voltage output via a high output impedance, of 100 kΩ or more. The idea was presumably that
you could scale the output to a convenient voltage by selecting a suitable input impedance. The
drawback was that the high output impedance made the amount of power transferred very small,
leading to a poor signal-to-noise ratio. The concept is now wholly obsolete.
1
2 Chapter 1
Amplifiers
At the most basic level, there are four kinds of amplifier, because there are two kinds of signal
(voltage and current) and two types of port (input and output). The handy word “port” glosses
over whether the input or output is differential or single ended. Amplifiers with differential input
are very common –such as all opamps and most power amps –but differential outputs are rare and
normally confined to specialised telecoms chips.
Table 1.1 summarises the four kinds of amplifier:
Voltage Amplifiers
These are the vast majority of amplifiers. They take a voltage input at a high impedance and
yield a voltage output at a low impedance. All conventional opamps are voltage amplifiers in
themselves, but they can be made to perform as any of the four kinds of amplifier by suitable
feedback connections. Figure 1.1a shows a high-gain voltage amplifier with series voltage
feedback. The closed-loop gain is (R1+R2)/R2.
Transconductance Amplifiers
The name simply means that a voltage input (usually differential) is converted to a current
output. It has a transfer ratio A = IOUT/VIN, which has dimensions of I/V or conductance, so
it is referred to as a transconductance amplifier. It is possible to make a very simple, though
not very linear, voltage-controlled amplifier with transconductance technology; differential-
input operational transconductance amplifier (OTA) ICs have an extra pin that gives voltage-
control of the transconductance, which when used with no negative feedback gives gain control;
see Chapter 24 for details. Performance falls well short of that required for quality hi-fi or
professional audio. Figure 1.1b shows an OTA used without feedback; note the current-source
symbol at the output.
Current Amplifiers
These accept a current in and give a current out. Since, as we have already noted, current-mode
operation is rare, there is not often a use for a true current amplifier in the audio business. They
should not be confused with current feedback amplifiers (CFAs), which have a voltage output,
the “current” bit referring to the way the feedback is applied in current mode. [2] The bipolar
The Basics 3
transistor is sometimes described as a current amplifier, but it is nothing of the kind. Current may
flow in the base circuit, but this is just an unwanted side effect. It is the voltage on the base that
actually controls the transistor.
Transimpedance Amplifiers
A transimpedance amplifier accepts a current in (usually single-ended) and gives a voltage out.
It is sometimes called an I-V converter. It has a transfer ratio A = VOUT /IIN, which has dimensions
of V/I, or resistance. That is why it is referred to as a transimpedance or transresistance amplifier.
Transimpedance amplifiers are usually made by applying shunt voltage feedback to a high-gain
voltage amplifier. An important use is as virtual-earth summing amplifiers in mixing consoles;
see Chapter 22. The voltage amplifier stage (VAS) in most power amplifiers is a transimpedance
amplifier. They are used for I-V conversion when interfacing to DACs with current outputs;
see Chapter 26. Transimpedance amplifiers are sometimes incorrectly described as “current
amplifiers”.
Figure 1.1c shows a high-gain voltage amplifier transformed into a transimpedance amplifier by
adding the shunt voltage feedback resistor R1. The transimpedance gain is simply the value of R1,
though it is normally expressed in V/mA rather than Ohms.
Negative Feedback
Negative feedback is one of the most useful and omnipresent concepts in electronics. It can
be used to control gain, to reduce distortion and improve frequency response, and to set input
and output impedances, and one feedback connection can do all these things at the same time.
Negative feedback comes in four basic modes, as in the four basic kinds of amplifier. It can be
taken from the output in two different ways (voltage or current feedback) and applied to the
amplifier input in two different ways (series or shunt). Hence there are four combinations.
However, unless you’re making something exotic like an audio constant-current source, the
feedback is always taken as a voltage from the output, leaving us with just two feedback types,
series and shunt, both of which are extensively used in audio. When series feedback is applied to a
high-gain voltage amplifier, as in Figure 1.1a, the following statements are true:
4 Chapter 1
V out A
- (Equation 1.1)
V in
1 + Af
Negative feedback can, however, do much more than stabilising gain. Anything unwanted occurring
in the amplifier, be it distortion or DC drift, or almost any of the other ills that electronics is heir to,
is also reduced by the negative feedback factor (NFB factor for short). This is equal to:
1
NFBfactor = (Equation 1.2)
1 + Af
What negative feedback cannot do is improve the noise performance. When we apply feedback,
the gain drops, and the noise drops by the same factor, leaving the signal-to-noise ratio the same.
Negative feedback and the way it reduces distortion is explained in much more detail in one of my
other books.[3]
Obviously a real signal, as opposed to a test sine wave, continuously varies in amplitude, and the
signal level chosen is purely a nominal level. One must steer a course between two evils:
• If the signal level is too low, it will be contaminated unduly by noise.
• If the signal level is too high, there is a risk it will clip and introduce severe distortion.
The wider the gap between them, the greater the dynamic range. You will note that the first
evil is a certainty, while the second is more of a statistical risk. The consequences of both must
be considered when choosing a level, and if the best possible signal-to-noise is required in a
studio recording, then the internal level must be high, and if there is an unexpected overload
you can always do another take. In live situations it will often be preferable to sacrifice some
noise performance to give less risk of clipping. The internal signal levels of mixing consoles are
examined in detail in Chapter 16.
If you seek to increase the dynamic range, you can either increase the maximum signal level
or lower the noise floor. The maximum signal levels in opamp-based equipment are set by the
voltage capabilities of the opamps used, and this usually means a maximum signal level of
about 10 Vrms or +22 dBu. Discrete transistor technology removes the absolute limit on supply
voltage and allows the voltage swing to be at least doubled before the supply rail voltages get
inconveniently high. For example, +/-40 V rails are quite practical for small-signal transistors and
permit a theoretical voltage swing of 28 Vrms or +31 dBu. However, in view of the complications
of designing your own discrete circuitry and the greater space and power it requires, those 9 extra
dB of headroom are dearly bought. You must also consider the maximum signal capabilities of
stages downstream –they might get damaged.
The dynamic range of human hearing is normally taken as 100 dB, ranging from the threshold of
hearing at 0 dB SPL to the usual “jackhammer at 1 m” at +100 dB SPL; however, hearing damage
is generally reckoned to begin with long exposures to levels above +80 dB SPL. There is, in a
sense, a physical maximum to the loudest possible sound. Since sound is composed of cycles of
compression and rarefaction, this limit is reached when the rarefaction creates a vacuum, because
you can’t have a lower pressure than that. This corresponds to about +194 dB SPL. I thought this
would probably be instantly fatal to a human being, but a little research showed that stun grenades
generate +170 to +180 dB SPL, so maybe not. It is certainly possible to get asymmetrical pressure
spikes higher than +194 dB SPL, but it is not clear that this can be defined as sound.
Compare this with the dynamic range of a simple piece of cable. Let’s say it has a resistance
of 0.5 Ω; the Johnson noise from that will be -155 dBu. If we comply with the European Low
Voltage Directive the maximum voltage will be 50 Vpeak = 35 Vrms = +33 dBu, so the dynamic
range is 155 + 33 = 188 dB, which purely by numerical coincidence is close to the maximum
sound level of 194 dB SPL.
In the field of music, louder is not better. When the level exceeds a threshold, in the range
75–95 dB SPL, the human ear protects itself by with the stapedius reflex (also called the
acoustic reflex). [4] One of the tiny bones that couple the eardrum to the oval window of the
cochlea is the stapedius, and when the reflex is triggered the muscle attached to it contracts and
reduces the sound transmission by about 15 dB. I perceive it as a relatively sudden increase in
6 Chapter 1
intermodulation distortion as the music level increases, though it is not really the same subjective
effect as amplifier distortion. Thus, quite apart from the consideration of hearing damage, which
is important, as it is irreparable and it doesn’t get better, there is no point in being too loud. It is
noticeable that in recent years professional audio has taken notice of this, and rock concerts are
much better for it. Semi-pro practice is less admirable, and there is a venue pub near me where the
SPL’s are much above stapedius and well into the discomfort level, generating a hefty threshold
shift [5] after an hour or so. I probably ought to do something about this.
Frequency Response
The generally accepted requirement for the frequency response of audio equipment is that it
should be flat from 20 Hz to 20 KHz. But how flat? A common spec 50 years ago would have
been ±1 dB. You might wonder why a plus tolerance is required when we are usually talking
about roll-offs; the reason is that valve amplifiers with output transformers may well show a peak
at the HF end due to transformer resonance. There should never be any response peaking with
solid-state equipment. Nowadays if you are aiming for quality kit, the amplitude tolerance is more
likely ±0.1 dB, which is far less than human perception can detect but still easy to attain with a
little thought.
The frequencies are often altered to 22 Hz to 22 kHz because the leading test gear has
measurement bandwidth so defined. It is a sad fact that with increasing chronological age, the
upper reaches of the audio band are lost to us; this is called presbycusis. [6] I was intrigued to
learn that birds, fish, and amphibians do not suffer presbycusis with aging, because they can
regenerate their cochlear sensory cells; mammals including humans have genetically lost this
extremely useful ability, and I call that downright careless.
You will sometimes hear people say that frequencies above 20 kHz are required for true
reproduction, even though every conventional test shows that they cannot be perceived. Usually
someone will mention the notorious 2000 paper by Tsutomu Oohashi in which he claimed that
brain scans of subjects listening to the gamelan music of Bali (with much ultrasonic content)
showed that ultrasonic frequencies caused changes in the brain, although they were reported by
the subjects as inaudible. The experimental methods described in the paper underwent much
criticism, and other researchers could not replicate the results. He never did any more work on the
subject, and the general view is that the paper is a right load of –er, should be disregarded. [7]
Another reason sometimes put forward for an over-extended frequency response is that it reduces
the amount of phase shift undergone by signals towards the top of the audio band. Here’s a
regrettable example at [8]. To illustrate, if the -3 dB cutoff frequency of a single roll-off is 40 kHz,
the phase lag at 10 kHz is 14 degrees with respect to midband. If the cutoff is raised to 100 kHz
the corresponding 10 kHz phase lag is only 5.7 degrees. This might have some weight if phase
shift was perceptible, but since it isn’t (see next section), the whole idea is invalid.
DC-coupled and so have no LF cutoff. It is instructive to see how the roll-offs add up. Bear in
mind that the word “cutoff” does not imply a sharp drop in response such as you would get from
a high-order filter; even with the gentle 6 dB/octave slopes we are talking about here, the -3 dB
point is still referred to as the cutoff frequency.
Take an audio path with n stages, each having the same cutoff frequency of fs. This is hardly
realistic, but stay with me. In this case it is relatively simple to calculate the overall -3 dB cutoff
frequency of the path; use Equation 1.3 for the high frequency roll-off and Equation 1.4 for the
roll-off at the bottom end.
1
foverall fs 21 (Equation 1.3)
n
I2
1
foverall = fs n
-1 (Equation 1.4)
Table 1.2 shows how this works for high-frequency cutoff and Table 1.3 for the low-frequency
cutoff: I have used 10 Hz and 20 kHz for the cutoffs of a single stage because they are nice round
numbers; in practical design they would probably be lower and higher respectively, to obtain the
desired bandwidth, as the tables show.
Table 1.2 makes it clear that two stages have a much lower bandwidth than each individual stage;
three stages halves the HF cutoff frequency. Likewise in Table 1.3, three stages doubles the LF
cutoff frequency. Clearly if you want a final HF cutoff of 20 kHz, then with five identical stages
you would need to select a higher cutoff for each stage at 20×(20/7.71) = 52 kHz or higher.
kHz N kHz
20 1 20
20 2 12.87
20 3 10.20
20 4 8.70
20 5 7.71
20 6 7.00
20 7 6.45
20 8 6.02
20 9 5.66
20 10 5.36
20 15 4.35
20 20 3.76
8 Chapter 1
Hz N Hz
10 1 10.00
10 2 15.54
10 3 19.61
10 4 22.99
10 5 25.93
10 6 28.58
10 7 31.00
10 8 33.24
10 9 35.34
10 10 37.33
10 15 45.98
10 20 53.25
Similarly at LF with five stages you would need to select a lower cutoff for each stage at 10/
(10/25.93) = 3.86 Hz or lower.
When you have the stages designed, the LF cutoff can sometimes be obtained by evaluating a
simple RC time-constant. For anything more involved, I strongly suggest you use a simulator.
I have never enjoyed complex algebra, and it is horribly easy to make mistakes in it.
An alternative way to present this information is to directly calculate the stage cutoffs by working
Equations 1.3 and 1.4 backwards. This can be done very simply using the Goal Seek function in
Excel. Table 1.4 shows that with 20 stages the individual cutoff frequency must be more than five
times the desired overall cutoff frequency. Table 1.5 shows a similar result at the low end –with 20
stages, the individual cutoff must be 1.88 Hz to get 10 Hz overall.
You may be thinking that it is pointless to extend these tables all the way up to 20 stages, but a
signal path can easily be that long in a mixing desk, and such situations require careful design to
stop roll-offs accumulating too much.
While the tables give a good insight into the way roll-offs accumulate, they do not represent a
good design strategy. A lot of identical 6 dB/octave roll-offs give a slow and soggy composite
roll-off, equivalent to a synchronous filter. [9] If we look at the LF cutoff, a better way is to have
one relatively high cutoff frequency and the rest a good deal lower. We can then put the relatively
high cutoff frequency in the first stage, where it will give some protection against subsonic
disturbances. This is not of course an alternative to a proper high-pass filter of second or higher
order in cases where subsonics are a serious problem, such as microphone and vinyl cartridge
amplifiers
The Basics 9
kHz N kHz
20.00 1 20
31.08 2 20
39.23 3 20
45.98 4 20
51.87 5 20
57.15 6 20
61.99 7 20
66.48 8 20
70.68 9 20
74.65 10 20
91.97 15 20
106.50 20 20
Hz N Hz
10.00 1 10
6.44 2 10
5.10 3 10
4.35 4 10
3.86 5 10
3.50 6 10
3.23 7 10
3.01 8 10
2.83 9 10
2.68 10 10
2.17 15 10
1.88 20 10
Figure 1.2 Three ways to obtain an overall -3 dB LF cutoff frequency at 10 Hz, using 5 stages.
Putting the highest cutoff at the start, as in B and C, gives more protection against subsonic
disturbances.
Alternatively we could select a 1 Hz cutoff in stages 2 to 5, perhaps because we are troubled with
capacitor distortion, and the first stage now needs a cutoff only slightly below 10 Hz, at 9.60 Hz;
see Figure 1.2c. The roll-off is slightly slower than for the 2 Hz case.
Exactly the same considerations apply when dealing with the upper end of the frequency
response, though it may be somewhat complicated if the HF cutoffs are partly due to opamp gain/
bandwidth limitations. Figure 1.3a shows that if you have five identical stages, a cutoff of 92.13
kHz is needed to get -1 dB at 20 kHz. Figure 1.3b shows again the principle of putting the most
The Basics 11
restrictive stage first –now we put the lowest cutoff first to help keep out ultrasonic unpleasantness
and have the higher cutoff frequencies later. In this case I have taken the cutoff attenuation as -1
dB at 20 kHz, as -3 dB is certainly too low in most cases.
Spreadsheet calculation shows that if all five stages are identical, the common cutoff frequency
must be 92.13 kHz, as in Figure 1.3a. If instead we set the -3 dB cutoff in stages 2 to 5 to 100
kHz, the cutoff frequency of the first stage can be twiddled with Goal Seek, yielding 72.51 kHz,
as in Figure 1.3b. Alternatively, we can set the -3 dB cutoff in stages 2 to 5 to 120 kHz, and
the correct input stage -3 dB cutoff is now 56.0 kHz, giving rather more input protection, as in
Figure 1.3c. Capacitor distortion should not be a problem at the HF end; the capacitors will be
small and can be polystyrene or NP0 ceramic.
There is of course no requirement, at either the LF or HF ends of the spectrum, that any of the
stages have the same cutoff frequency. The HF stage cutoffs are sometimes determined by the
need to add a stabilisation capacitor; if none is needed, then there may be nothing setting the
cutoff of that stage but the finite opamp gain/bandwidth. This is perfectly acceptable. When a
capacitor is used, the exact HF cutoff frequency will be partly determined by the preferred values
of the capacitors used, and of course their tolerance.
Figure 1.3 Three ways to obtain an overall -1 dB HF cutoff frequency at 20 kHz, using five stages.
Putting the lowest cutoff at the start, as in B and C, gives more protection against ultrasonic
disturbances. Be aware that while the overall response is -1 dB at 20 kHz, the cutoff frequencies
for each are given at -3 dB.
Phase Perception
You don’t have any. Under realistic conditions human hearing cannot detect phase shift, which
is just as well, because most of what you hear will have been phase shifted all over the place
12 Chapter 1
by loudspeakers and subsequent room reflections. Being able to hear phase would probably be
hopelessly distracting and, from an evolutionary point of view, useless. It’s not, as far as I can see,
going to help you detect the tiger stalking you through the undergrowth, whereas stereo location
with binaural hearing might. It’s definitely “might”, because tigers, despite their large size, can
move with terrifying stealth.
We are talking here about having one part of the audio spectrum phase shifted with respect to
another. An example was given in the previous section: a -3 dB cutoff frequency of 40 kHz gives
a phase lag at 10 kHz of 14 degrees relative to low frequencies. Raising the cutoff to 100 kHz
reduces this to 5.7 degrees. Neither is perceptible.
If the phase shift is proportional to frequency, then the group delay is constant with frequency and
this is a linear-phase system, as described above; we just get a pure time-delay with no possible
audible consequences. However, in most cases the phase shift is not remotely proportional to
frequency, and so the group delay varies with frequency. This is sometimes called phase distortion
or group-delay distortion, which is perhaps not the ideal term, as “distortion” implies non-
linearity to most people, while here we are talking about a linear process.
Most of the components in the microphone–recording–loudspeaker chain are minimum-phase; in
other words, they impose only the phase shift that would be expected and which can be predicted
from their amplitude/frequency response. One great exception to this is . . . the multi-way
loudspeaker. Another great exception is the analogue magnetic tape recorder, which showed rapid
phase changes at the bottom of the audio spectrum, usually going several times round the clock.
In the first edition of my book on active crossover design in 2011, I wrote, “Fortunately we don’t
need to worry about that anymore”, but it appears I spoke too soon. There were several vintage
multi-track tape recorders displayed at the Paris AES Convention in June 2016, and tape machines
are now (2019) unquestionably having a revival. Even cassette tapes are having a revival[10],
and there’s a need for machines to play them on. Everybody was certain that the one format that
would not be revived was the eight-track cartridge. Wrong; it’s happening in 2019, see [11]. Wax
cylinders have already been revived once, by The Men Who Will Not Be Blamed For Nothing.
[12] It seems unlikely that experiment will be repeated.
When digital audio began to appear, the possible perception of phase shifts caused by steep
(typically ninth-order elliptical) anti-alias and reconstruction filters was very much a concern,
as no such filtering had been used in audio paths before. That is true no longer, first because
the rapid phase changes at the top of the audio band in fact proved to be inaudible, and even
more since oversampling technology eliminated the need for them altogether. Preis and Bloom
examined this issue in 1983, when it was very much live, and they concluded, “At 15 kHz (cutoff
frequency) the cascade of up to 4 pairs of seventh-order elliptic filters introduced no perceptible
effects”. [13] That is some pretty serious filtering, much more than was actually required in
practice. Mercifully, no-one has suggested reviving ninth-order elliptical anti-aliasing filters –so
far as I am aware.
Whatever happens with tape recorders, we are certainly going to have multi-way loudspeaker
systems around for the foreseeable future, and the vast majority of them have all-pass crossovers.
The Basics 13
Clearly an understanding of what degradation –if any –this all-pass behaviour causes is vital to a
general understanding of phase perception. Much experimentation has been done, as noted above,
and there is only space for a very brief account here.
Some audio commentators have said that the possibility of phase being perceptible has been
neglected by the audio establishment. This is wholly untrue. Searching on “phase perception” in
the archives of the Journal of the Audio Engineering Society brings up no less than 936 technical
papers, the first dated 1956 and the last 2018. It is not a neglected subject.
One of the earliest findings on phase perception was Ohm’s Law. No, not that one, but Ohm’s
Other Law, which is usually called Ohm’s Acoustic Law and was proposed in 1843. [14] In its
original form it simply said that a musical sound is perceived by the ear as a set of sinusoidal
harmonics. The great researcher Hermann von Helmholtz extended it in the 1860s into what today
is known as Ohm’s Acoustic Law by stating that the timbre of musical tone depends solely on the
number and relative level of its harmonics and not on their relative phases. This is a good start, but
it does not ensure the inaudibility of an all-pass response.
An important paper on the audibility of midrange phase distortion was published by Lipshitz,
Pocock and Vanderkooy in 1982 [15], and they summarised their conclusions as follows:
1. Quite small phase nonlinearities can be audible using suitable test signals.
2. Phase audibility is far more pronounced when using headphones instead of loudspeakers.
3. Simple acoustic signals generated in an anechoic environment show clear phase audibility
when headphones are used.
4. On normal music or speech signals phase distortion is not generally audible.
At the end of the abstract of their paper the authors say, “It is stressed that none of these
experiments thus far has indicated a present requirement for phase linearity in loudspeakers for
the reproduction of music and speech”. James Moir also reached the same conclusion. [16]
An interesting paper on the audibility of second-order all-pass filters was published in 2007 [17],
which describes a perception of “ringing” due to the exponentially decaying sine wave in the
impulse response of high-Q all-pass filters (for example Q = 10). It was found that isolated clicks
show this effect best, while it was much more difficult to detect, if audible at all, with test signals
such as speech, music, or random noise. That is the usual finding in this sort of experiment –that
only isolated clicks show any audible difference. While we learn that high-Q all-pass filters
should be avoided in crossover design, I think most people would have thought that was the case
anyway.
Siegfried Linkwitz, sadly no longer with us, did listening tests in which either a first-order all-
pass filter, a second-order all-pass filter (both at 100 Hz), or a direct connection could be switched
into the audio path. [18] These filters have similar phase characteristics to all-pass crossovers and
cause gross visible distortions of a square waveform but are in practice inaudible. He reported,
“I have not found a signal for which I can hear a difference. This seems to confirm Ohm’s
Acoustic Law that we do not hear waveform distortion”.
14 Chapter 1
If we now consider the findings of neurophysiologists, we note that the auditory nerves do not
fire in synchrony with the sound waveform above 2 kHz, so unless some truly subtle encoding
is going on (and there is no reason to suppose that there is), then perception of phase above this
frequency would appear to be inherently impossible.
Having said this, it should not be supposed that the ear operates simply as a spectrum analyser. This
is known not to be the case. A classic demonstration of this is the phenomenon of “beats”. If a 1000
Hz tone and a 1005 Hz tone are applied to the ear together, it is common knowledge that a pulsation
at 5 Hz is heard. There is no actual physical component at 5 Hz, as summing the two tones is a
linear process. (If instead the two tones were multiplied, as in a radio mixer stage, there would be
new components generated.) Likewise, non-linearity in the ear itself can be ruled out if appropriate
levels are used. What the brain is actually responding to is the envelope or peak amplitude of the
combined tones, which does indeed go up and down at 5 Hz as the phase relationship between the
two waveforms continuously changes. Thus the ear is in this case acting more like an oscilloscope
than a spectrum analyser. It does not, however, seem to work as any sort of phase-sensitive detector.
The conclusion we can draw is that for the purposes of designing analogue electronic circuitry,
there is no need to consider phase perception.
Gain Structures
There are some very basic rules for putting together an effective gain structure in a piece of
equipment. Like many rules, they are subject to modification or compromise when you get into
a tight corner. Breaking them reduces the dynamic range of the circuitry, either by worsening the
noise or restricting the headroom; whether this is significant depends on the overall structure of
the system and what level of performance you are aiming at. Three simple rules are:
1. Don’t amplify then attenuate.
2. Don’t attenuate then amplify.
3. The signal should be raised to the nominal internal level as soon as possible to minimise
contamination with circuit noise.
There are exceptions. For an example, see Chapter 10 on moving-coil disc inputs, where
attenuation after amplification does not compromise headroom because of a more severe
headroom limit downstream.
Figure 1.4 a) Amplification then attenuation. Stage 2 will always clip first, reducing headroom.
b) Attenuation then amplification. The noise from Stage 2 degrades the S/N ratio. The lower the
gain setting, the worse the effect.
around its “0 dB” position, where it introduces 10 dB of attenuation, as is typically the case for a
fader on a mixer. To maintain the nominal signal level at 0 dBu we need 10 dB of gain, and a +10
dB amplifier (Stage 2) has been inserted just before the gain control. This is not a good decision.
This amplifier will clip 10 dB before any other stage in the system and introduces what one might
call a headroom bottleneck.
There are exceptions. As noted, the moving-coil phono head amp described in Chapter 10 appears
to flagrantly break this rule, as it always works at maximum gain even when this is not required.
But when considered in conjunction with the following RIAA stage, which also has considerable
gain, it makes perfect sense, for the stage gains are configured so that the second stage always
clips first, and there is actually no loss of headroom.
B. All circuit blocks are assumed to introduce noise at -100 dBu. The noise output for the first
version is -89.2 dBu. Now take a second version of the signal path that has an input amplifier with
5 dB of gain, followed by block A, another amplifier with 5 dB of gain, then block B. The noise
output is now -87.5 dB, 1.7 dB worse, due to the extra amplification of the noise from block A.
There is also more hardware, and the second version is clearly an inferior design.
Noise
Noise here refers only to the random noise generated by resistances and active devices. The
term is sometimes used to include mains hum, spurious signals from demodulated RF, and other
non-random sources, but this threatens confusion, and I prefer to call the other unwanted signals
“interference”. In one case we strive to minimise the random variations arising in the circuit itself,
in the other we are trying to keep extraneous signals out, and the techniques are wholly different.
When noise is referred to in electronics it means white noise unless it is specifically labelled as
something else, because that is the form of noise that most electronic processes generate. There
are two elemental noise mechanisms which make themselves felt in all circuits and active devices.
These are Johnson noise and shot noise, which are both forms of white noise. Both have Gaussian
probability density functions. These two basic mechanisms generate the noise in both Bipolar
Junction Transistors (BJTs) and Field Effect Transistors (FETs), though in rather different ways.
There are other forms of noise that originate from less fundamental mechanisms such as device
processing imperfections which do not have a white spectrum; examples are 1/f (flicker) noise
and popcorn noise. These noise mechanisms are described later in this chapter.
Non-white noise is given a colour which corresponds to the visible spectrum; thus red noise has a
larger low-frequency content than white noise, while pink is midway between the two.
White noise has equal power in equal absolute bandwidth, i.e. with the bandwidth measured in
Hz. Thus there is the same power between 100 and 200 Hz as there is between 1100 and 1200 Hz.
It is the type produced by most electronic noise mechanisms. [19]
The Basics 17
Pink noise has equal power in equal ratios of bandwidth, so there is the same power between 100
and 200 Hz as there is between 200 and 400 Hz. The energy per Hz falls at 3 dB per octave as
frequency increases. Pink noise is widely used for acoustic applications like room equalisation
and loudspeaker measurement, as it gives a flat response when viewed on a third-octave or other
constant-percentage-bandwidth spectrum analyser. [20]
Red noise has energy per Hz falling at 6 dB per octave rather than 3. It is important in the study
of stochastic processes and climate models but has little application in audio. The only place
you are likely to encounter it is in the oscillator section of analogue synthesisers. It is sometimes
called Brownian noise, as it can be produced by Brownian motion; hence its alternative name of
random-walk noise. Brown here is a person and not a colour. [21]
Blue noise has energy per Hz rising at 3 dB per octave. Blue noise is used for dithering in image
anti-aliasing but has, as far as I am aware, no application to audio. The spectral density of blue
noise (i.e. the power per Hz) is proportional to the frequency. It appears that the light-sensitive
cells in the retina of the mammalian eye are arranged in a pattern that resembles blue noise. [22]
Great stuff, this evolution.
Violet noise has energy per Hz rising at 6 dB per octave. (I imagine you saw that one coming.) It
is also known as “differentiated white noise”, as a differentiator circuit has a frequency response
rising at 6 dB per octave. The acoustic thermal noise of water has a violet spectrum, so it usually
dominates hydrophone measurements at high frequencies. It is sometimes called purple noise.
Grey noise is pink noise modified by a psychoacoustic equal loudness curve, such as the inverse
of the A-weighting curve, to give the perception of equal loudness at all frequencies.
Green noise is generally considered to be the background noise of the natural world, i.e. the
ambient noise in natural settings, free from any man-made components. It is like pink noise but
with more energy in the 500 Hz region. In graphics it is used for stochastic half-toning of images
and consists of binary dither patterns composed of homogeneously distributed minority pixel
clusters. I think we had better leave it there.
Black noise usually refers to absolute silence [23] but is sometimes used to mean a noise
spectrum that is zero almost everywhere except for a few spikes. It can also refer to a kind of
noise used to model the frequency of natural disasters; see [24].
Johnson Noise
Johnson noise is produced by all resistances, including those real resistances hiding inside
transistors (such as rbb, the base spreading resistance). It is not generated by the so-called intrinsic
resistances, such as re, which are an expression of the Vbe/Ic slope and not a physical resistance
at all. Given that Johnson noise is present in every circuit and often puts a limit on noise
performance, it is a bit surprising that it was not discovered until 1928 by John B. Johnson at Bell
Labs. [25]
The rms amplitude of Johnson noise is easily calculated with the classic formula:
Where:
vn is the rms noise voltage T is absolute temperature in °K B is the bandwidth in Hz
k is Boltzmann’s constant R is the resistance in Ohms
The only thing to be careful with here (apart from the usual problem of keeping the powers
of 10 straight) is to make sure you use Boltzmann’s constant (1.380662 × 10−23) and NOT the
Stefan-Boltzmann constant (5.67 10−08), which relates to black-body radiation and will give
some spectacularly wrong answers. Often the voltage noise is left in its squared form for ease of
summing with other noise sources. Table 1.6 gives a feel for how resistance affects the magnitude
of Johnson noise. The temperature is 25°C and the bandwidth is 22 kHz.
Johnson noise theoretically goes all the way to daylight, but in the real world, it is ultimately
band-limited by the shunt capacitance of the resistor. Johnson noise is not produced by circuit
reactances – i.e. pure capacitance and inductance. In the real world, however, reactive components
are not pure, and the winding resistances of transformers can produce significant Johnson noise;
this is an important factor in the design of moving-coil cartridge step-up transformers. Capacitors,
with their very high leakage resistances, approach perfection much more closely, and the
capacitance has a filtering effect. They usually have no detectable effect on noise performance,
Ohms μV dBu
and in some circuitry, it is possible to reduce noise by using a capacitive potential divider instead
of a resistive one. [26]
The noise voltage is of course inseparable from the resistance, so the equivalent circuit is of a
voltage source in series with the resistance present. While Johnson noise is usually represented
as a voltage, it can also be treated as a Johnson noise current by means of the Thevenin-Norton
transformation, which gives the alternative equivalent circuit of a current-source in shunt with the
resistance. The equation for the noise current is simply the Johnson voltage divided by the value
of the resistor it comes from:
in = vn/R
When it is first encountered, this ability of resistors to generate electricity from out of nowhere
seems deeply mysterious. You wouldn’t be the first person to think of connecting a small electric
motor across the resistance and getting some useful work out –and you wouldn’t be the first
person to discover it doesn’t work. If it did, then by the First Law of Thermodynamics (the law
of conservation of energy), the resistor would have to get colder, and such a process is flatly
forbidden by . . . the Second Law of Thermodynamics. The Second Law is no more negotiable
than the First Law, and it says that energy cannot be extracted by simply cooling down one body.
If you could build one, it would be what thermodynamicists call a Perpetual Motion Machine
of the Second kind, and they are no more buildable than the common sort of perpetual motion
machine.
It is interesting to speculate what happens as the resistor is made larger. Does the Johnson voltage
keep increasing until there is a hazardous voltage across the resistor terminals? Obviously not,
or picking up any piece of plastic would be a lethal experience. Johnson noise comes from a
source impedance equal to the resistor generating it, and this alone would prevent any problems.
Table 1.6 ends with a couple of silly values to see just how this works; the square root in the
equation means that you need a petaohm resistor (1 ×1015 Ω) to reach even 600 mVrms of Johnson
noise. Resistors are made up to at least 100 GΩ, but petaohm resistors (PΩ?) would really be a
minority interest.
Shot Noise
It is easy to forget that an electric current is not some sort of magic fluid but is actually
composed of a finite (though usually very large) number of electrons, so current is in effect
quantised. Shot noise is so called because it allegedly sounds like a shower of lead shot being
poured onto a drum, and the name emphasises the discrete nature of the charge carriers.
Despite the picturesque description the spectrum is still that of white noise, and the noise
current amplitude for a given steady current is described by a surprisingly simple equation (as
Einstein said, the most incomprehensible thing about the universe is that it is comprehensible)
that runs thus:
Where:
q is the charge on an electron (1.602 ×10−19 Coulomb) Idc is the mean value of the current, and
B is the bandwidth examined
As with Johnson noise, often the shot noise is left in its squared form for ease of summing with
other noise sources. Table 1.7 helps to give a feel for the reality of shot noise. As the current
increases, the Shot noise increases too, but more slowly as it depends on the square root of the DC
current; therefore the percentage fluctuation in the current becomes less. It is the small currents
which are the noisiest.
The actual level of shot noise voltage generated if the current noise is assumed to flow through a 100 Ohm
resistor is rather low, as the last column shows. There are few systems which will be embarrassed by
an extra noise source of even -99 dBu unless it occurs right at the very input. To generate this level of
shot noise requires 1 Amp to flow through 100 Ω, which naturally means a voltage-drop of 100 V and
100 watts of power dissipated. These are not often the sort of circuit conditions that exist in preamplifier
circuitry. This does not mean that shot noise can be ignored completely, but it can usually be ignored
unless it is happening in an active device where the noise is amplified.
Popcorn Noise
This form of noise is named after the sound of popcorn being cooked rather than eaten. It is also
called burst noise or bistable noise and is a type of low frequency noise that is found primarily in
integrated circuits, appearing as low-level step changes in the output voltage, occurring at random
intervals. Viewed on an oscilloscope, this type of noise shows bursts of changes between two or
more discrete levels. The amplitude stays level up to a corner frequency, at which point it falls at a
rate of 1/f2. Different burst-noise mechanisms within the same device can exhibit different corner
frequencies. The exact mechanism is poorly understood but is known to be related to the presence
of heavy-metal ion contamination, such as gold. As for 1/f noise, the only measure that can be
taken against it is to choose an appropriate device. Like 1/f noise, popcorn noise does not have a
Gaussian amplitude distribution.
Any number of noise sources may be summed in the same way, by simply adding more squared
terms inside the square root, as shown by the dotted lines. When dealing with noise in the design
process, it is important to keep in mind the way that noise sources add together when they are not
of equal amplitude. Table 1.8 shows how this works in decibels. Two equal voltage noise sources
give a sum of +3 dB, as expected. What is notable is that when the two sources are of rather
unequal amplitude, the smaller one makes very little contribution to the result.
dB dB dB sum
0 0 +3.01
0 –1 +2.54
0 –2 +2.12
0 –3 +1.76
0 –4 +1.46
0 –5 +1.19
0 –6 +0.97
0 –10 +0.41
0 –15 +0.14
0 –20 +0.04
Another random document with
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TURNER, JOHN HASTINGS. Place in the
world. *$1.75 Scribner
20–3578
“Mr Turner has a clever pen, and the fluttering of the dovecotes
caused by Iris’s unconventionality gives him scope for a number of
incisive character-sketches. Mr Turner is to be congratulated on the
keenness of his observation as well as the liveliness of his style.”
“Fairly amusing.”
Reviewed by H. W. Boynton
+ − Bookm 51:342 My ’20 300w
“Witty comedy.”
“The plot of the tale is extremely slight and at times the novel
drags badly, but the style is often agreeable and the characters of
Henry Cumbers and of the Rev. John Heslop are very well drawn
indeed.”
“Mr Turner’s ‘Simple souls’ was amusing; this novel goes deeper. It
is fine workmanship as to its writing, and in its essence it makes for
soundheartedness and human tolerance.”
“‘The dark wind’ has been most cordially received in London and is
especially interesting to students of poetry because it combines much
of the colorcraft of the imagists with the melodies of the Georgians.
Indeed, none of the young English poets has given us verse in which
sense impressionism plays a more important part than it plays here.”
(N Y Times) “Not the least interesting peculiarity of Mr Turner’s art
is that he has made no startling departures into irregular verse
forms. Nor does Mr Turner seek to startle by the choice of bizarre
subjects. He writes on Haystacks and Sunflowers and Hollyhocks
and Aeroplanes and Recollecting a visit.” (Bookm)
“Turner’s ‘The dark wind’ is first of all a book of color and beautiful
rhythms. He possesses the virtue of flinging lovely pictures before
the reader, not the hard emphasized colors that cry from Miss Amy
Lowell’s efforts, but a soft yet glittering mingling of hues that is
warm with sunlight and harmonious with spring and autumn.”
Reviewed by A. C. Moore
The story of the famous wax works, established in Paris during the
revolution and later brought to London, written by one of the great-
grandsons of the founder, the present proprietor of the exhibition.
Madame Tussaud, altho a young girl at the time of the revolution,
was already famed as a modeler in wax and had been a favorite at
court. She was conscripted and compelled to model the guillotined
heads of Louis XVI and Marie Antoinette, Marat murdered in his
bath, and other horrors, a number of which are reproduced in the
illustrations. The story is brought down to the present day,
describing many of the recent additions, with illustrations. Hilaire
Belloc has written an introduction and the book is indexed.
“The amazing feature of the book is, however, the manner in which
its author has made so intrinsically interesting and romantic a theme
dull and commonplace. It is evident that he possesses absolutely no
qualifications for his task. He is simply adept at the compilation of a
scrap-book. Yet his subject is so fascinating that it is better to have
his account of Madame Tussaud’s life and work than none at all.” E.
F. E.
“Mr Tussaud has appreciated the value of his materials both from
the historic point of view and from the viewpoint of human interest.
His narrative, like his wax figures, simply presents facts of
undeniable interest. But it is the pictures that make the book
unique.”
“There are more adventures to the square inch in this book than
any other that has come to hand since ‘The three musketeers.’ The
manner of telling is swift, humorous, breezy. Reddy is a find.”
Hildegarde Hawthorne
There are two mysteries in this story, that concerning the past of
the beautiful Mrs Davenant and the mystery of Lake House, which
Letty Thorne senses on first coming there to stay with her uncle. In
the solution of the second the secret of the first is also revealed. It is
revealed to the reader and to one other person in the story, but Mrs
Davenant, feeling that there is that in her life which forbids
remarriage says no to the man who loves her and keeps her own
confidence. A minor love story develops between the vicar and Mrs
Davenant’s friend Agnes Howard, and to this affair as well as to the
love story of Letty there is a happy ending.
“So long as there are Christians of even a simple type, Tyrrell will
be read, because of his instinct for the things of Christ. His cruel
ironies and his flaming resentments, his rash speculations and his
tottering syntheses may all be buried in his grave.”
“Miss Petre has done well to publish this selection from his
correspondence. He was a many-sided man, and his letters reflect his
many-sidedness.”
“Any lover of this type of tale must have discovered here [in the
Borzoi books] a number of excellent examples, of which ‘The
pathway of adventure’ is by no means the least successful.”
Goldie is the telephone operator in a large hotel and she tells her
story in slangy letters to her pal Myrtle. Events in which she takes a
share from her switchboard reveal her as a girl “always there with the
helping hand, no matter how much it’s been lacerated in the past.” In
particular the love affairs of her patrons and co-workers interest her,
and she straightens out several tangles, and finally, her own love
story develops happily.
“A love story that is both clever and jolly is so rare nowadays that
one seizes with avidity upon the romance of little Goldie.”
“Mr Ullman deserves full credit for a lot of ‘good lines.’ The wit of
Goldie’s letters is catchy and largely original—not current vaudeville
wheezes warmed over. We wish there was more of it and less of the
‘good-old-ham-and-eggs,’ ‘man-from-home’ brand of philosophy.”
“When the romantic personality falls into the hands of the scholar
there is necessarily something of glamour and delight lost. This is
what has happened in this austerely spiritual biography of Evelyn
Underhill.” L. C. Willcox
“Miss Underhill has converted the old fable of the ant and the
grasshopper into a very modern romance which she calls ‘The white
moth.’ Hilda Plaistead is the earnest plodder, Guy Nearing the gay
and irresponsible hero, and the setting is the town of Cato. The two
have a childhood engagement, become widely separated, and in the
final chapter again discover that they were always meant for each
other, but it is only after Guy has learned the folly of being jack of all
trades and master of none.”—N Y Evening Post
“We can scarcely claim for Miss Underhill’s story either originality
of substance or of treatment. What she does accomplish is an
exceedingly readable and very human story, which possesses certain
scenes of quiet and insistent realism.”
The book is one of the Psychic series and describes the cure of a
case of illness of fifteen years’ standing in the course of a year and
eight months by an invisible spirit doctor. It contains a preface by J.
Arthur Hill, testimonials by several personal friends of the patient
and a report by the physician long in charge of the case in the flesh.
The contents are: A chance paragraph; A chain of coincidences; The
first interview; A further surprise; The invisible hand; Experiences
and experiments; Fellow-lodgers; Royal progress; Learning to walk;
“My little girl”; Six months later; Comments and criticisms;
Appendix and index. The book was published in England as “One
thing I know, or, The power of the unseen.”
“As a ‘psychic’ tale the book is futile and foolish, indeed, rather
fertile in folly.”
“Though it too often misses the authentic current, is too often led
away into stagnant marshes, it is perhaps as good a map as we yet
possess. The editor is a better conversationalist than guide.”
+ − Dial 68:667 My ’20 60w
Dial 69:664 D ’20 60w
Reviewed by R. P. Utter
“If there be any critic in the country who ought not to make a
schoolbook, that critic is Louis Untermeyer. He is much too
brilliantly individual and his likes and dislikes are too pronounced. It
is a book of verse that young people probably will like, if they like
verse at all. Many of the selections included are humorous.... A good
professor would make a better anthology for use in schools.”
Marguerite Wilkinson
− + N Y Times 25:140 Mr 28 ’20 480w
+ School R 28:630 O ’20 160w
“Neither the rhapsodic nor the mocking quality, however, gives the
substance of Untermeyer’s work. The roots of his power lie deeper.
Upright vigor, wide and healthy curiosity describe his own work
excellently.” Babette Deutsch
“The facts are presented with scholarly care, but the style is not too
technical.”