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INNOVATIONS IN AUDIO • AUDIO ELECTRONICS • THE BEST IN DIY AUDIO
Questions & Answers
Crispin Herrod-Taylor 
Designs SoundBuckets 
Portable Speakers
By Shannon Becker
Hollow-State Electronics
Power-Supply Stiffness
By Richard Honeycutt
You Can DIY!
The Incredible Egg Loudspeaker
By Thomas Perazella
Building a Tunnel Reflex System
By Ken Bird
Brioso, a Two-Way Speaker
with Transmission-Line Bass
By Paul Kittinger
R&D Stories
AES X223 and X241
Speaker Measurements
By Geoff Hill
Sound Control
Modulation Distortion
in Loudspeakers
By Richard Honeycutt
Standards Review
HDMI-Enhanced Audio
Return Channel (eARC)
By Marshall Goldberg
Audio Praxis—Amplifier Series
Energy-Efficient Multi-Level
Amplifier Solutions
By Ward MaasSEPTEMBER 2018
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September 2018 ISSN 1548-0628
www.audioxpress.com
audioXpress (US ISSN 1548-0628) is published monthly, 
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The Team
President: KC Prescott
Controller: Chuck Fellows
Editor-in-Chief: João Martins
Associate Editor: Shannon Becker
Graphics: Grace Chen
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Technical Editor: Jan Didden
Regular Contributors: Bruce Brown, Bill Christie, Joseph D’Appolito, Vance Dickason, Jan 
Didden, Scott Dorsey, Gary Galo, Gerhard Haas, Chuck Hansen, Richard 
Honeycutt, Charlie Hughes, Mike Klasco, Ward Maas, Oliver Masciarotte, 
Nelson Pass, Christopher Paul, Bill Reeve, Fernando Rodrigues, Steve 
Tatarunis, Ron Tipton, Stuart Yaniger
Speaker Design Starts with Connections
There are good reasons to be excited with speaker design. For the past two 
years, loudspeakers remain the fastest growing product segment in the audio 
industry with consumers embracing the new class of portable wireless, multi-room 
audio, and new categories of smart- and voice-controlled speakers. A trend that is now expanding into 
installation and other applications.
If it’s true that more and more homes are now relying on not necessarily the best quality sound 
coming from portable Bluetooth or cheap smart speakers, it is also true that those fast-growing 
categories are also inspiring people to listen to music more than ever. According to the latest NPR 
and Edison Research Smart Audio Report, Spring 2018, there are now more than 43 million people 
(18+) in the US that own a smart speaker. And around 70% say they are listening to more audio since 
getting a smart speaker. Nearly half of first adopters have meanwhile bought a second smart speaker, 
and the same percentage admit they will buy another one in the near future. Something the speaker 
manufacturers cannot ignore, given the massive scale of this completely new segment. 
The large majority of the households adopting smart speakers are in fact using it two hours a 
day much in the same way they used small tabletop radios in the kitchen, or had audio playing in 
computers with small multimedia speakers. Comparing the experience, in most cases, it’s a major 
quality improvement in sound!
Also, because of the enthusiastic adoption, many homes will acquire a larger and better-quality 
speaker to connect to the original Amazon, Google, or Apple device—opening an opportunity for other 
brands to enter the space, assuming they implement the connectivity.
At the same time, many consumers are also looking at high-quality audio experiences, directly 
inspired by the rediscovery of analog, to enjoy the latest immersive formats, or simply to appreciate 
the potential of extended frequency reproduction propelled by high-resolution audio sources. Among 
those looking for better quality experiences, most are also looking for the convenience of wirelessly 
connecting to streaming sources, and many will look at the existing “higher-quality sound” smart 
speakers (e.g., Apple’s HomePod or a Sonos system). When we look at those households, we now find 
three, four, or more speakers, instead of the traditional two in the living room stereo system.
And “better sound” is generally considered a key focus for the companies researching smart-
speakers. Actually, the R&D departments of those large companies are even exploring innovations 
and early designs that the speaker industry has left behind and mostly forgot about it. Because of the 
demand for compact and highly integrated designs, we see driver manufacturers focusing their efforts 
to improve all key speaker parameters—sometimes exploring options (and patents) that have remained 
ignored for decades. New materials, the use of computer simulation technology, and new production 
methods make it possible to improve on ideas that have been around for some time. Everything that 
was done before can be reinvented and improved. Plus, adding amplification and processing into 
integrated solutions makes it possible to step-up those designs. 
Still, there’s another angle that needs to be urgently addressed. Most current speakers still use 
analog inputs. We now see companies jumping to wireless, adding Wi-Fi and Bluetooth modules and 
keeping the option for “standard” analog audio connectors. Just in case…
As everyone looking at wireless quickly discovers, connectivity is no longer for audio only. By adding 
a network or a USB Type-C connector, speakers can gain bidirectional high-speed data, high-resolution 
digital audio and even power—in one single connector. It is possible to have speakers reconfigured for 
different multichannel or immersive formats adjusting the speakers’ response and optimizing for the 
room acoustics in real time, with an external or built-in processor. If a speaker is designedthat way, 
its “intelligence” can be explored in wireless and wired modes, depending on the most convenient and 
reliable option required by the applications—and this can even significantly reduce BOM costs.
Thinking of the connectivity as a main feature of speaker design is now mandatory and a key 
component of what makes a modern speaker.
J. Martins
Editor-in-Chief
audioxpress.com | September 2018 | 5 
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Vance Dickason has been working as a professional in the 
loudspeaker industry since 1974. He is the author of Loudspeaker 
Design Cookbook—which is now in its seventh edition and published 
in English, French, German, Dutch, Italian, Spanish, and Portuguese—
and The Loudspeaker Recipes. Vance is the editor of Voice Coil: The 
Periodical for the Loudspeaker Industry, a monthly publication. 
Although he has been involved with publishing throughout his 
career, he still works as an engineering consultant for a number of 
loudspeaker manufacturers.
Dr. Richard Honeycutt fell in love with acoustics when his father 
brought home a copy of Leo Beranek’s landmark text on the subject 
while Richard was in the ninth grade. Richard is a member of the North 
Carolina chapter of the Acoustical Society of America. Richard has his 
own business involving musical instruments and sound systems. He 
has been an active acoustics consultant since he received his PhD in 
electroacoustics from the Union Institute in 2004. Richard’s work 
includes architectural acoustics, sound system design, and community 
noise analysis. 
Mike Klasco is the president of Menlo Scientific, a consulting firm for 
the loudspeaker industry, located in Richmond, CA. He is the organizer 
of the Loudspeaker University seminars for speaker engineers. Mike 
specializes in materials and fabrication techniques to enhance speaker 
performance. 
Steve Tatarunis has been active in the loudspeaker industry since 
the late 1970s. His areas of interest include product development 
and test engineering. He is currently a support engineer at Listen, 
in Boston, MA, where he provides front-line technical support to the 
SoundCheck test system’s global user base.
Ron Tipton has degrees in electrical engineering from New Mexico 
State University and is retired from an engineering position at White 
Sands Missile Range. In 1957, he started Testronic Development 
Laboratory, which became TDL Technology, to develop audio 
electronics. All product sales and services were terminated on 
December 31, 2015, but the TDL website is still online with a variety 
of audio information and downloads.
COLUMNISTS
6 | September 2018 | audioxpress.com
Contents
Speaker Special Features
28 Thoughts on AES X223 and 
X241 Speaker Measurement 
Projects
By Geoff Hill
The Audio Engineering Society (AES) is 
working to develop industry standards, 
focusing on measurements of loudspeaker 
drive units. Learn more about the quest. 
34 The Incredible Egg (Part 1)
Sound Reproduction at Its Best
By Thomas Perazella
In the first of this three-part article series, 
Tom Perazella details his latest bold speaker 
project, inspired by the Mach 17 speaker 
built in the 1990s by Waveform Research of 
Canada. This three-way design features the 
latest generation of SB Acoustics’ high-end 
Satori drivers and two Dayton 12” woofers 
in a separate sealed box base.
42 A Modified Version of the 
Bullhorn Speaker System
By Ken Bird
After recalling an unique design that 
produced an amazing sound from small 
drivers, Ken Bird decided to build a modified 
version of this tunnel reflex system. In 
this article, he details how you can make 
your own set of these modified “Bullhorn” 
speakers. 
48 The Brioso Speakers
Lively Two-Way Speakers with 
Transmission-Line Bass
By Paul Kittinger
The Brioso (Italian for lively) uses a 
Scan-Speak Revelator midwoofer and a 
Transducer Labs Alumina ceramic dome 
tweeter in a floor-standing, mass-loaded 
transmission line enclosure (ML-TL). Learn 
how to make these award-winning speakers 
for yourself. 
audioxpress.com | September 2018 | 7 
September 2018Volume 49 – No. 9
4 From the Editor’s Desk
5 Client Index
66 Industry Calendar
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SOUND CONTROL
8 Modulation Distortion 
in Loudspeakers and Its 
Detectability (Part 1)
An Introduction to Doppler Distortion
By Richard Honeycutt
QUESTIONS & ANSWERS
56 Pro Audio Company 
Designs SoundBuckets 
Portable Speakers
An Interview with Crispin Herrod-
Taylor, Co-Founder of Crookwood
By Shannon Becker
HOLLOW-STATE ELECTRONICS
60 The Measured Effect of 
Power-Supply Stiffness 
on Power Amplifier 
Performance
By Richard Honeycutt
Columns
14 Examining Fundamental 
Amplifier Techniques with 
Electron Tubes, 2nd Edition
By Richard Honeycutt
Richard Honeycutt reviews the second and 
more extensive edition of this electronics 
classic, giving high marks for its useful 
information about electron tube technology 
and tube audio amplifiers. 
16 Energy-Efficient Multi-
Level Amplifier Solutions
By Ward Maas
This article takes a closer look at a new 
generation of energy-efficient integrated 
Class-D amplifier solutions developed 
by Merus Audio, recently acquired by 
Infineon Technologies. 
24 HDMI Specification 2.1 
Improves Home Theater 
Audio with eARC
By Marshall Goldberg
Learn more about this new HDMI 
standard, which adds a unique feature 
to preserve AVR compatibility. In this 
article, Marshall Goldberg explains 
how the Enhanced Audio Return 
Channel (eARC) works.
Features
Sound Control
8 | September 2018 | audioxpress.com
Modulation distortion, or intermodulation 
distortion (IMD) as it is more commonly called, is a 
familiar specter to designers and users of electronic 
equipment. In amplifiers, modulation distortion 
consists of the control of the amplitude of one 
signal according to the amplitude of another signal 
of a different frequency. This Amplitude Modulation 
Distortion (AMD) occurs whenever two signals of 
different frequencies are applied to a nonlinear 
system. It is reduced, as harmonic distortion is 
reduced, by improvements in the system’s linearity. 
AMD also occurs in loudspeakers. In addition, 
another type of modulation distortion not present in 
amplifiers—frequency modulation distortion (FMD)—
is present in loudspeakers. 
The mechanism that produces FMD is the Doppler 
Effect: the phenomenon which causes a train whistle 
or a car horn to vary in perceived pitch as the vehicle 
passes an observer. The velocity of a loudspeaker 
cone can become great enough when the speaker 
is reproducing low frequencies (where the cone 
excursion is greatest) to produce a measurable FMD. 
This form of distortion has come to be called Doppler 
distortion (Dd) as shown in Figure 1. 
The existence of Dd has been recognized for 
quite some time—there are articles on it in literature 
dating back at least to 1943. It was, however, largely 
ignored until 1958, and only periodically has there 
been a resurgence of interest in this subtle form of 
distortion. The reasons for this resurgenceare twofold. 
First, improvements in amplifiers have reduced their 
intermodulation distortion to figures that approach our 
present limits of measurability. Second, and much more 
importantly, it is only since the mid-20th Century that 
the serious music listener has been deluged with small-
diameter, long-cone-throw loudspeakers, in which Dd 
is particularly pronounced. 
This deluge began with the introduction of acoustic 
suspension speakers in the mid-1950s, involving 
cabinets using an 8″ woofer rather than the previously 
more common 12″ or 15″ units. The larger speakers 
had cone areas of 2.25 and 3.5 (respectively) times 
that of the 8″ speaker, thus requiring a much lower 
maximum cone velocity and excursion for a given 
sound power output. More recently, the sheer number 
of speakers involved in a surround system has led to 
the common use of speakers much smaller than 8” 
in diameter.
ax
By
Richard Honeycutt
(United States)
Modulation Distortion 
in Loudspeakers and Its 
Detectability (Part 1)
In the first of this three-part article series about Doppler distortion, 
our author details past attempts to assess its importance. 
An Introduction to Doppler Distortion
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Sound Control
10 | September 2018 | audioxpress.com
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Expert Opinions About 
Doppler Distortion
As with the Doppler shift of a train whistle, the 
shift in frequency of a high-frequency tone depends 
upon the cone velocity, with higher velocities 
producing a greater shift in frequency. As with AMD, 
the resulting signal consists of each original frequency 
(the high-frequency and the low-frequency one), plus 
a tone at the sum of the original two frequencies and 
another at the difference frequency. Actually, there 
are also many more frequencies, but only the sum 
and the difference ones are important at the low level 
of modulation produced by a loudspeaker. 
In the usual case for music reproduction, there are 
many low-frequency signals and many high-frequency 
ones all mixed together, and a porridge of the sums 
and differences of all these frequencies, potentially 
resulting in what Paul Klipsch called “a sea of mud” 
in which the desired audio is immersed.
Although Dd is usually analyzed as frequency 
modulation, Rod Elliott has pointed out that the 
process can be viewed as phase modulation. Figure 2
and Figure 3 show unmodulated and Dd-modulated 
waves produced by a speaker. Simple FM analysis 
would predict that the frequency shift is greatest 
at the zero crossing, where instantaneous cone 
velocity is maximum, but the measurements show 
otherwise. Although viewing Dd as an example of 
phase modulation is interesting, it makes no practical 
difference, since angular frequency (2πf, symbolized 
by ω) is the time derivative of phase (φ), and thus 
the frequencies and amplitudes of the distortion 
components are identical. 
As mentioned earlier, there already exists quite 
a bit of literature that discusses Doppler distortion. 
Practically all of it, however, treats the problem from a 
theoretical standpoint. In doing so without the benefit 
of an empirical touchstone against which to test their 
conclusions, the writers have amassed results that 
are simultaneously astounding and baffling. Klipsch 
told us that in typical home listening situations Dd 
may reach 14% of the output of an 8” speaker, and he 
supported his report with spectrograms. Roy Allison 
informed us that total modulation distortion—the RMS 
sum of Dd and AMD—never exceeds 5% in a typical 
situation, and is usually under 1%. Referring to live 
vs. recorded sound demonstrations, he concludes that 
such small amounts of modulation distortion are not 
objectionable. Klipsch points out—quite correctly—that 
Dd and AMD are identical in sound, since they result 
in the production of the same extraneous frequency 
components. 
We have traditionally been led to believe that 
IMD is more audible and more objectionable than 
harmonic distortion (HD), which seems reasonable 
since IMD consists of frequencies that are not 
harmonically related to—or are out-of-tune with—
the desired signal. Although some investigators have 
stated that in a typical speaker, AMD components 
are higher in amplitude than Dd components, James 
Moir’s measurements showed the reverse: The total 
modulation distortion in most of the three units he 
tested (12″, 7″, and 4″) was almost exactly what was 
calculated for the contributions of Dd alone, indicating 
that AMD components were smaller by 20 dB or more 
than Dd components. 
Henri van Hessen has dismissed the problem of 
Figure 1: The Doppler Effect occurs when a sound source is moving. When the speaker 
cone is moving to the right, the sound emitted from the right is shifted to a higher 
frequency, while the sound emitted from the left is shifted to a lower frequency.
Figure 2: Expected Doppler-modulated wave (blue) shows maximum frequency shift at zero 
crossing. (Image courtesy of Elliott Sound Products, http://sound.whsites.net/doppler.htm#ref)
audioxpress.com | September 2018 | 11 
Figure 3: Phase-modulated wave (blue) shows maximum frequency shift at cone excursion 
peaks. (Image courtesy of Elliott Sound Products http://sound.whsites.net/doppler.htm#ref)
Doppler distortion by explaining that a microphone 
introduces Doppler distortion, and the loudspeaker 
removes it. However, he was ignoring, among other 
things, the dependence of Dd upon diaphragm 
velocity, which is several orders of magnitude smaller 
in the microphone than in the loudspeaker. 
Detecting the Doppler Distortion
Most of the papers referenced so far have 
discussed the phenomenon of Dd, but not its 
detectability or unpleasantness. In 1958, Klipsch 
published a report of a test in which an eccentric 
capstan was devised for a tape recorder and used to 
produce flutter in the recording. Thus, he was able to 
create an audio signal in which a 1046-Hz tone was 
frequency modulated by a 10-Hz, 20-Hz, or 40-Hz 
tone. With a 10 Hz modulation, the sound was a fast 
vibrato, described as “not particularly unpleasant.” 
The 20 Hz and 40 Hz modulating frequencies produced 
sounds that were described as “unpleasant” and 
“extremely irritating,” respectively. The modulation 
depth corresponded to 0.35% flutter, although a 
later investigator claimed that there was an error 
in Klipsch’s math, and the actual flutter was 0.75%. 
A review of Klipsch’s paper does not reveal such 
an error. This amount of FMD corresponds to the Dd 
level produced by a 12″ speaker with a peak cone 
displacement of 0.21″ at 50 Hz. In a separate paper, 
Klipsch reports playing a tape of the third movement 
of Mozart’s Concerto for Clarinet and Orchestra along 
with a 28-Hz tone, through a “small speaker” capable 
of 0.25″ excursion at 28 Hz with almost no audible 
output. The “flutter” produced by FM distortion was 
reported as plainly audible in sustained clarinet notes. 
Roy Child performed tests involving two listeners, 
using a single speaker simultaneously reproducing 
a 40-Hz tone and a higher frequency tone. The two 
listeners observed detectible Dd at peak-to-peak 
cone amplitudes of 0.053” and 0.094”, respectively, 
using a 6.5″ speaker. To produce the same volume 
velocity, the cone of a 12″ speaker would only require 
an excursion of 0.0155” and 0.0276”, respectively. 
Note two things:
• Klipsch did not claim to find the minimum 
detectable level of Dd, just that the effect was 
audible at the levels indicated. Indeed, he described 
the effect of hislevel of Dd as “intolerable.”
• Klipsch did not specify the number of listeners 
used in his test, and Child only used two. Since 
significant variations in detectability thresholds 
among listeners are probably to be expected, 
neither Klipsch’s nor Child’s results should be taken 
as having statistical significance.
Peter Fryer built a “Carrier Canceling Doppler 
Machine” based on a variable-length, bucket-
brigade delay line to electronically simulate Dd. 
His one-subject listening test concluded that, even 
when you could hear it, Dd “is not at all unpleasant 
up to quite high levels as long as the modulating 
frequency lies between say 20 Hz and 100 Hz,” and 
that ″Doppler distortion is unlikely to be of importance 
in most speaker systems.″ Detectability, he found, 
corresponded with a total cone movement (peak-to–
peak) of about 10 mm (0.4”). 
Roy Allison and Edgar Villchur later claimed 
that Fryer’s detection criterion was too stringent in 
several ways to reflect real-world loudspeaker use. 
First, Fryer’s work had assumed a full-range drive 
unit, whereas most high-quality speakers divide 
the frequency range between two or more drivers. 
Second, Fryer had taken no account of the fact that 
the amount of Dd generated by a moving diaphragm 
varies with the angle off the forward axis until, at 
right angles to the direction of diaphragm motion, it 
falls to zero. As a result of this, the early reflections 
and reverberant sound in the listening room contain 
lower levels of Dd than the direct sound, which, Allison 
and Villchur claimed, will further raise the threshold 
of detection and thus make the generation of audible 
Dd by most hi-fi loudspeakers even less likely.
Sound Control
12 | September 2018 | audioxpress.com
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Doppler Distortion Research Diminishes 
But Does Not Disappear 
After the publication of Fryer’s paper in 1977 
and the Villchur/Allison paper in the early 1990s, 
Dd dropped off the radar of many audiophiles and 
publishers. However, the careful reader will note 
that simply writing Dd off as insignificant ignores 
results from competent researchers. Thus, in 2004, an 
article was published by Keith Howard in Stereophile, 
comparing the detectability of Dd to the detectability 
of phase jitter in a digital audio signal, a subject that 
had been extensively researched. Howard referenced 
the work of Eric Benjamin and Benjamin Gannon, 
whose work examined the effects of jitter on systems 
using the IEC 60958 Digital Audio Interface. Since 
jitter results in sum and difference sidebands, as does 
Dd, the detectability thresholds should be comparable. 
Howard’s article compares the minimum 
detectable levels of jitter and Dd, based on the the 
work of several authors. He shows the proposed 
tolerance limit for sinusoidal jitter suggested by W.I. 
Manson of the BBC in the early 1970s, compared 
with the most stringent National Association of 
Broadcasters (NAB) standard for wow and flutter 
in tape machines running at 15 IPS. The two curves 
intersect at a modulating frequency of 7 Hz, below 
which the NAB standards predict a slightly higher 
detectability threshold, and above which Manson 
predicts a higher threshold. In both cases, the 
tolerance limit depends on the modulating frequency, 
with much larger amounts of FMD being acceptable 
when the modulating frequency is low—as it will 
predominantly be with loudspeakers. Howard presents 
other experimental data from several authors, and 
this data agrees in a general way with the trends 
and levels shown by the Manson and NAB standards, 
although none of the other data includes more than 
one set of modulating frequency and RMS jitter level, 
so there is not enough data to plot curves.
Despite the long hiatus caused by uncritical 
acceptance of the conclusions of Fryer, Villchur, and 
Allison, there is a long discussion of Dd on Internet 
groups and forums, including some scientifically valid 
posts along with some rambling along the lines of 
unsupported opinions. Future articles in this column 
will address the theoretical background of Dd and 
examine additional tests along with a meta-analysis 
of Dd detectability experiments. ax
Resources
R. Allison, “Doppler Distortion Again,” Audio, Volume 55, No. 3, p. 54, March 1971.
R. Allison and E. Villchur: “On the Magnitude and Audibility of FM Distortion in Loudspeakers,” Journal of the 
Audio Engineering Society, Volume 30, No. 10, p. 664-699, October 1982.
G. L. Beers and H. Belar, “Frequency-Modulation Distortion in Loudspeakers,” Proceedings of the Institute of 
Radio Engineers Australia, Volume 31, pp. 132-138, 1943.
E. Benjamin and B. Gannon, “Theoretical and Audible Effects of Jitter on Digital Audio Quality,” presented at 
the 105th AES Convention, September 1998.
R. Childs, “Doppler Distortion in Loudspeakers,” Audio, Volume 54, No. 8, p. 26, August 1970.
R. Elliott, “Doppler Distortion in Loudspeakers—Real Or Imaginary?” Elliott Sound Products, August 2004,
http://sound.whsites.net/doppler.htm.
P. A. Fryer, “Simulation and Investigation of Doppler Distortion,” Journal of the Audio Engineering Society, 
Preprint # 1197, presented at the 56th AES Convention, March 1977.
K. Howard, “Doppler Distortion in Loudspeakers,” Stereophile, November 2004, 
www.stereophile.com/reference/1104red/index.html.
P. W. Klipsch, “Subjective Effects of Modulation Distortion,” Journal of the Audio Engineering Society, Volume 6, 
No. 2, 1958.
———, “Modulation Distortion in Loudspeakers,” Journal of the Audio Engineering Society, Volume 17, No. 2, pp. 
192-206, 1969.
———, “Loudspeaker Performance,” Wireless World, Volume 77, p. 1412, February 1970.
———, “The Mud Factor,” Audio, Volume 54, No. 10, pp. 38-44, October 1970.
 J. Moir, “Doppler Distortion in Loudspeakers,” presented at the 46th AES Convention, September 1973.
 H. Van Hessen, “Doppler Anyone?” Audio, Volume 54, pp. 42ff, April 1971.
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Book Review
14 | September 2018 | audioxpress.com
ax
My first electronics books were course books 
on radio servicing and radar, given to me by my 
dad. The first electronics book I bought was the 
1963 RCA Receiving Tube Manual, which—at a price 
of $1.00—came to about 2.2 cents/cubic inch. (At 
that time, McDonald’s hamburgers had not gotten 
to my hometown, but when they did, a few yearslater, they cost 19 cents each. Gasoline was 15 cents 
per gallon.) My first physics text when I entered 
college in 1966 cost about 31 cents/cubic inch. When 
Rudolf Moers’ first edition of Fundamental Amplifier 
Techniques with Electron Tubes was published in 
2010, this mammoth volume (822 pages in an 8” 
× 11” format) only cost about 50 cents/cubic inch. 
In my review of Moers’ first edition in the April 
2012 issue of audioXpress, I stated that the author 
“presents a 21st-century perspective on the science 
of hollow-state design as applied to amplifiers and 
power supplies. Beginning with the principles of 
electron emission, the book progresses through 
standard vacuum tube varieties: diodes, triodes, 
tetrodes, and pentodes, after which it covers 
such general principles as frequency-dependent 
behavior, nonlinear distortion, noise, and negative 
feedback. The book concludes with a chapter on the 
construction of electron tube amplifiers.” 
In his second edition (922 pages in an 8.5” × 12” 
format), which—at a price of under 28 cents/cubic 
inch, the same summary is true. Moers describes the 
aim of his second edition as being able “to give you 
useful knowledge about electron tube technology in 
the application of audio amplifiers, including their 
power supplies, for the design and DIY construction 
of these electron tube amplifiers yourself.” In this 
aim he completely succeeds.
The author discusses the physics underlying 
electron tube operation far more comprehensively 
than did any of his predecessors. He does this in a 
way that maximizes presentation of principles while 
minimizing unnecessary mathematics. In many 
cases, the physical explanations can be skipped 
by those whose only interest is design methods. 
For the reader who does take advantage of the 
physical explanations, Moers’ inclusion of an eight-
page listing/definition of mathematical symbols 
Review By
Richard Honeycutt 
(United States)
Richard Honeycutt reviews the second and more 
extensive edition of this electronics classic, 
Fundamental Amplifier Techniques with Electron Tubes.
He gives high marks to this massive tome that provides 
useful information about electron tube technology, 
including their power supplies, for the design and DIY 
construction of these electron tube audio amplifiers. 
Fundamental Amplifier Techniques with Electron Tubes, 2nd Edition
By Rudolf Moers
Pages: 922 
Price: $69.00 (€ 59.00)
ISBN: 978-94-6345-175-8
www.boekenbestellen.nl/boek/
fundamentalamplifiertechniqueswithelectrontubes2ndedition
Examining Fundamental Amplifier 
Techniques with Electron Tubes,
2nd Edition
audioxpress.com | September 2018 | 15 
makes the explanations easy to follow.
The author assumes that the reader knows basic 
AC and DC circuit theory and algebra, although the 
novice can understand the thrust of the explanations 
even in the absence of these basics. As is often true 
when reading the works of authors whose native 
language is not one’s own, the reader must become 
accustomed to some differences from common 
American- and British-English terminology. For 
example, what Americans and Brits call a “common-
cathode” or “grounded-cathode” amplifier stage is 
identified by Moers as a “basic cathode” stage. What 
we call a “phase splitter” is called a “phase shifter” 
in this book. But such minor linguistic issues are 
few and easily handled.
The additional 86 pages in the second edition 
include more extensive coverage of vacuum 
diodes, incorporating formulae more accurate 
than the common ones for output voltage of a 
loaded capacitor-input filter driven by a half- or 
full-wave rectifier. The chapter on pentodes now 
contains extensive analysis of David Hafler and 
Herb Keroes’ ultralinear amplifier circuit (more 
descriptively named—as Moers reminds us—the 
“screen grid negative feedback amplifier”). In this 
analysis, the book clarifies whether the screen-grid 
tap of the output transformer, variously given as a 
“40%” or a “43%” tap, is referring to the percentage 
of primary turns or the percentage of primary 
impedance. The book then examines the effect 
on output-stage linearity for both interpretations 
of the percentage, as well as discussing whether 
maximizing this linearity also minimizes distortion. 
He does this using a single-ended model, which can 
be generalized to predict the behavior of push-pull 
output stages.
The new edition of Fundamental Amplifier 
Techniques with Electron Tubes concludes with eight 
appendices containing some characteristic curves 
for the EL84 and the KT88.
As I stated in concluding my review of the 
first edition, “anyone who wants to go beyond the 
Heathkit level of tube amplifier understanding owes 
it to him/herself to buy and study this excellent 
volume.” To this recommendation, I now add the 
comment that even if you have the first edition, the 
new material in the second edition is more than 
worth the price. ax
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ax Audio Praxis—Amplifier Series
16 | September 2018 | audioxpress.com
In February 2018, 
shortly after a successful 
presentation at the CES show in 
Las Vegas, NV, Infineon Technologies 
announced the acquisition of Merus Audio. The 
Copenhagen-based start-up, founded in 2010 by Hans 
Hasselby-Andersen and Mikkel Høyerby, has developed a 
new generation of energy-efficient integrated Class-D amplifier solutions, which 
stirred the industry’s interest. audioXpress was already working on this article.
There are several development goals in the design 
of Class-D audio amplifiers. A major goal has always 
been to obtain very efficient high-power amplification. 
Nowadays, there are many high-power designs, mainly 
used in pro audio, where that goal clearly has been 
met. Kilowatts of power at an efficiency higher than 
90% are rather common. With the use of switch mode 
power supplies, these amps even became lightweight.
Another aspect of Class-D development is to 
improve the amplifier’s quality to such a level that it 
becomes a match (or an improvement) for traditional 
types of amplifiers. We saw that with the efforts of 
the Dutch company Axign and its AX 5689 (see “Axign 
of Things to Come,“ audioXpress, December 2017). 
The advantages for any manufacturer of a product 
that incorporates an audio amplifier are clear: power, 
efficiency, quality, and cost. 
Looking to maximize audio performance and 
battery playback time for smart home and battery-
powered speakers, while minimizing heat and design 
space, were primary goals for the development of the 
Merus eximo technology by the Danish company now 
acquired by Infineon. Since the acquisition, Infineon 
decided to drop the “eximo” name, now branded 
simply as multi-level Class D amplifier ICs.
There are market segments where audio amplifier 
efficiency is much desired. Unfortunately, one of the 
by-products of that efficiency is electromagnetic 
interference (EMI) levels that are challenging to solve. 
There is where Merus Audio differentiated with its 
multi-level switching technology. An advanced new 
platform that now allows Infineon to offer solutions for 
low-power applications that are suitable for products in 
which efficiency and low EMI—and EMC compliance—
are key.
Multi-Level Amplifiers
The creators of the technology are basedin 
Copenhagen, Denmark, and were able to quickly 
By
Ward Maas
(The Netherlands)
Figure 1: Conventional Class-D output signal
Energy-Efficient Multi-Level 
Amplifier Solutions
Figure 2: Multi-level amplifier output signal
audioxpress.com | September 2018 | 17 
establish sales offices in the US, Singapore, and Hong 
Kong. One year following the company’s creation, 
Merus Audio submitted its first patent for the multi-
level amplifier topology. In 2014, the company publicly 
demonstrated the MA12040 amplifier IC at Stanford 
University and products started sampling to customers 
in 2015. In 2017, four variants of multi-level amplifier 
ICs, from 2× 40 W to 2× 80 W were shipping and 
products from established brands in the US, Europe, 
and Asia were incorporating the technology.
These amplifiers feature a unique power stage 
topology with more—but smaller—power transistors 
to create the amplified output. The power stage 
generates multi-level modulation patterns of up to five 
levels, thereby approximating the audio signal much 
closer to analog signals than conventional Class-D 
amplifiers. 
Figure 1 shows the conventional two-level output 
of a Class-D amplifier. The output signal of the multi-
level amplifier displayed in Figure 2 shows the output 
signal. Figure 3 and Figure 4 show the output stages 
of a conventional Class-D amplifier and a multi-level 
amplifier.
This multi-level output signal has a few clear 
advantages. Frequency multiplication is an inherent 
property of multi-level modulation, which essentially 
makes the effective switching frequency at the speaker 
load much higher than the switching frequency of 
the individual power transistors. This is utilized in 
Merus amplifiers to significantly lower the switching 
activity, reducing the power losses associated with 
each switching event. 
Figure 5 shows how the signal is built up at 
the switching node in a bridge-tied load (BTL) 
configuration for one sinusoid cycle. In this case, the 
speaker load switching frequency is four times the 
MOSFET switching frequency at the output nodes. 
Also note that the switching pattern gives rise to three 
states where complete cancellation of the out-of-band 
switching residue takes place. This directly translates 
to a reduction of the ripple current in the audio system 
output section. With no need to suppress out-of-band 
switching artifacts, many applications do not require 
a common-mode applied L-C filter.
The effect of the reduced ripple current is shown 
in Figure 6. When normalized to the ripple current 
of a traditional Class-D amplifier (red line), 3-level or 
half bridge (blue line), and 5-level or BTL (black line) 
modulated output signals exhibit strongly reduced 
ripple current. Also note the zero states for 3-level 
and 5-level signals—points at which there is no 
ripple current. Overall dissipation and power loss 
are significantly reduced due to the reduced ripple 
current in external components.
Conventional Class-D amplifiers typically require 
Figure 4: Multi-level amplifier output stage
Figure 3: Conventional Class-D output stage
Figure 5: Signal is built up at the switching node in a BTL configuration.
About the Author
Ward Maas is the owner of Pilgham Audio. He studied electronics, marketing, and 
amplifier design. During his career in consumer electronics, Ward worked in areas 
ranging from CD standardization to radio and television to personal GPS navigation. 
Ward has worked on an extreme low-noise magnetic cartridge preamplifier and several 
special amplifier products. As the CTO of “Witchworld,” a theme park near Amsterdam, 
he also works with animatronics. He lives in Almere, Netherlands, with his wife and son.
ax Audio Praxis—Amplifier Series
18 | September 2018 | audioxpress.com
an L-C output filter to reconstruct the audio waveform before the 
amplified audio signal reaches the speaker. The filter accounts for a 
large portion of the size and area for the total amplifier solution as 
well as for the Bill-of-Materials (BOM) cost.
Multi-level switching significantly reduces the output signal’s 
switching content. Therefore, in most cases, Merus amplifiers can 
drive speakers directly, without the use of an L-C filter—even in 
higher-powered applications. This enables ultra-compact solutions 
(see Photo 1) and design freedom in the target application. For 
smaller audio applications (e.g., mobile phones), an L-C filter is not 
required due to the low output power level of the audio amplifier.
The multi-level output stage offers great flexibility and allows 
for the configuring of the amplifier for optimal power performance 
in many applications. Via a digital control interface, various power 
modes can be defined by selecting the modulation method and the 
switching frequency.
During amplifier operation, the integrated power management 
algorithm automatically selects the most optimal power mode for 
a given power level. The seamless transition between power modes 
ensures that power losses are minimized across the entire output 
power range while ensuring high audio performance and low EMI.
Multi-level amplifiers incorporate a fourth-order feedback loop. 
This guarantees excellent audio performance and a robustness against 
real-world system imperfections (e.g., a non-ideal power supply rail 
voltage). This feedback loop is fully integrated on-chip. This minimizes 
the design-in effort by avoiding a time-consuming selection and 
optimization procedure of the external loop filter components for 
achieving loop stability and satisfactory performance.
Several integrated protection features ensure that both the 
amplifier and the speakers are protected from potential damage 
Figure 6: Normalized ripple currents vs. duty cycle Photo 1: Ultra-compact amplifier design made possible
Figure 7: Efficiency vs. output power Figure 8: Input power vs. output power
Multi-level audio amplifiers 
from Infineon
Guaranteeing best-in-class audio performance and 
high power efficiency
These fully integrated IC solutions provide a number of advantages, including: 
› Small and light - enabling high power delivery in a very small form factor.
› Very low power losses and cool, heatsink-free operation.
› EMC compliance without the use of bulky filter components.
› Reduced bill of materials and cost due to elimination of additional components and complex
dynamic rail tracking schemes.
› High dynamic range.
› Down to 0.003% THD and low noise.
› Support for both analog and digital audio input.
› Highly configurable for tailored performance in individual applications.
Explore the product portfolio to maximize the user experience through audio performance 
and design freedom!
Find out more on
www.infineon.com/audiosolutions
ax Audio Praxis—Amplifier Series
20 | September 2018 | audioxpress.com
during various fault events. This includes over-current 
protection circuits, on-chip temperature sensors, and 
under-voltage supply monitors. All warning and error 
signals are stored in registers that can be read via 
the amplifier control bus. A dedicated error pin is also 
available for hardware error signaling. In general, 
multi-level amplifiers are designed to keep playing 
music for as long as possible and only shut down in 
the event of a non-recoverable error.
Maximizing Battery Life
For low-power (battery) operation, battery 
lifetime is undoubtedly very important. Efficiency 
is mostly specified at high power levels. However, 
these maximum output levels are hardly ever met 
in consumer applications. So the efficiency in low 
output conditions is the one that has higher influence 
on battery life. By cutting power losses, multi-level 
amplifiers enable audio product designers to either 
extend battery lifetime or reduce the size of the costly 
Li-Ion battery. As the heat dissipation during typical 
operation is significantly reduced, heatsinking can also 
be much smaller. In fact, in most cases the amplifier 
circuit board will offer sufficientheatsinking capability, 
even in relatively high-powered applications where 
other amplifiers would require a dedicated heatsink.
The low heat dissipation leads to a generally cooler 
operation inside the audio application. This gives a 
much larger headroom for periods of higher output 
power (e.g., during music playback at high volume 
levels). Cooler operation also has an impact on the 
reliability of the audio application, due to a reduction 
of thermally accelerated aging of many system 
components. Figure 7 and Figure 8 show the efficiency 
of the MA12040 IC over power vs. a traditional Class-D 
amplifier. Certainly, the improvement in low-mid power 
use is remarkable.
One of the first companies to make use of 
this technology was Danish company Soundboks 
(https://soundboks.com). The company raised a few 
eyebrows by presenting the Soundboks 1, the loudest 
and the largest “portable” Bluetooth speaker. With 
a battery life of 30 hours, it was presented as the 
ultimate speaker for outdoor parties and sports and 
it even comes with a backpack frame to make it truly 
“portable.” Soundboks 2 (see Photo 2), the current 
product is even louder (122 dB) and has an improved 
battery life of 40 hours and features two MA12070P ICs. 
This made it clear that the technology was very much 
at the top of the Class-D quality efficiency products.
Photo 2: Copenhagen-based Soundboks designed a “portable party speaker,” which was 
one of the first products to benefit from the MA12070P multi-level amplifier ICs, enabling 
higher power and longer battery life.
Figure 9: THD+N vs. frequency Figure 10: THD+N vs. output power
audioxpress.com | September 2018 | 21 
For battery-powered portable speakers, clearly 
power consumption is a key item. With the expected 
growth figures of this market segment a firm rise 
in demand of super-efficient Class-D amplification 
products can be expected as well. As audio quality 
is also becoming an important specification item, 
the new Infineon amplifiers are an extremely good 
fit for the most demanding applications, including 
the latest generation “smart speakers” (see Figure 9
and Figure 10).
The Products
Currently consisting of four products—the MA12040, 
the MA12040P, the MA12070, and the MA12070P—the 
Infineon multi-level amplifier series of products is able 
to improve designs in multiple segments, including 
soundbars, multiroom systems, hi-fi components, and 
commercial integration solutions, including networked 
speakers and modular amplifiers.
The “40” product range are 2× 40 W at 4 Ω using 
a 4 V to 18 V power supply, while the “70” products 
are 2× 80 W at 4 Ω with a 4 V to 26 V power supply. 
The “P” products have a digital I2S audio input 
instead of an analog input. All ICs support sample 
rates from 44.1 kHz to 192 kHz. A complete overview 
can be found on the company’s website and in the 
Supplementary Material section of the audioXpress 
website (www.audioXpress.com).
For product development with multi-level 
amplifiers, there is a comprehensive evaluation kit 
(see Photo 3) which contains a variety of digital/
analog input, output and setup/selection features. It 
Photo 3: The original multi-level amplifier evaluation kit, created to support development 
with the new technology now acquired by Infineon.
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ax Audio Praxis—Amplifier Series
22 | September 2018 | audioxpress.com
also contains two on-board power supply generators 
(5 V and 3.3 V buck-converted), so only one external 
power supply (PVDD) is necessary. With this evaluation 
board, it is possible to verify the claims regarding 
energy efficiency, power losses at normal user 
operating conditions, idle power loss, the adaptive 
power management system, output filter components, 
solution cost, size reduction, and audio performance.
For ease of use, the evaluation board comes 
with an easy-to-use PC GUI to control and program 
the MA120xx device (see Figure 11). The Danish 
engineers also worked with Summit Wireless 
(w w w. s u m m i t w i r e l e s s . c o m) , L i n k p l a y 
(www.linkplay.com), and Eleven Engineering 
(www.elevenengineering.com) on complete reference 
designs (see Photo 4), which can be used for new 
projects supporting wireless audio applications.
The Technology Founders 
Prior to Merus Audio, Hans Hasselby Andersen was with Texas 
Instruments and with switching amplifier pioneer, Toccata Technology, 
holding various positions within R&D, business development, and product 
management. Before that, he was with Nokia Mobile Phones. Hans has an 
M.Sc.EE degree from the Technical University of Denmark and an Executive 
MBA from the Scandinavian International Management Institute.
Mikkel Høyerby was previously employed as a research engineer with 
Motorola in Copenhagen, working on RFPA linearization, general RF/IF 
design as well as portable audio power amplification solutions. Mikkel 
received the M.Sc.EE degree in 2004 and the PhD degree in 2010 from the Technical University of Denmark. He currently holds 
two patents and has another four pending in the areas of switch-mode control and RFPA linearization. Mikkel is a lecturer on the 
“Switch-Mode Audio Power Amplifiers (Class-D)” course at the Technical University of Denmark.
Mikkel HøyerbyHans Hasselby Andersen 
Photo 4: Reference designs for wireless audio products combining the original Merus eximo ICs with solutions from Summit Wireless (a) and Linkplay 
(b) as demonstrated at CES 2018.
a) b)
Figure 11: GUI of the evaluation kit, which is now being rebranded by Infineon.
audioxpress.com | September 2018 | 23 
With these evaluation boards and reference designs, Merus 
Audio promoted very successful demos during CES 2018, in a hotel 
suite of the Mirage Las Vegas where delegations from most major 
consumer electronics manufacturers succeeded one after the other. 
While presentations were promoted in the main room and demo area, 
another room was used to holding meetings for potential customers 
and the room in the suite was used for more practical demos. In the 
bedroom, a modified Canton Musicbox S showed an increased battery 
life of 58% (Photo 5), taking the battery duration from 18 to 27 hours. 
No wonder the company became a target for an acquisition by a major 
semiconductor manufacturer. 
On February 20, 2018, Infineon Technologies, the German 
semiconductor company, announced that it had acquired Merus Audio. 
The multi-level Class-D audio amplifier portfolio now complements 
Infineon’s existing PowIRAudio product line for power levels below 
~70 W and Infineon’s existing sensor and audio processing solutions. 
Employees will remain in Copenhagen, Denmark, and are integrated 
into Infineon’s existing audio business to become part of the Power 
Management & Multimarket Division. ax
Photo 5: Merus Audio’s suite at the Mirage Las Vegas, during CES 2018, 
promoted convincing demonstrations of the company’s multi-level 
amplifier technology efficiency (a), including a retrofit on the existing 
Canton Musicbox S portable Bluetooth Speaker (b), extending the battery’s 
duration 58%.
a)
b)
hypex NCxxxMP series
overview
Highlights
High efficiency
Universal mains operation
Flat, fully load-independent
frequency response
Low output impedance
Very low, frequency
independent THD
Very low noise
Features
One or two channel amplifier
5W standby SMPS
Advanced over current
protection
External controlled operation
Auto-switching (115/230V)
Lowweight
Compact
Applications
Monitor loudspeakers for
recording and mastering
studios
Audiophile power amplifiers
for professional and
consumer use.
Public address systems
Active loudspeakers
NC122MP
NC250MP
NC252MP
NC500MP
NC502MP
Add-on Module NC100HF
The NC100HF is a dedicated
tweeter-amplifier which
fits the NCxxxMP series.
The NCxxxMP amplifier module incorporates a low 
power standby power supply (meets 2013 ERP Lot
6 0.5W requirements), a highly efficient switch 
mode power supply and a high- performance Class 
D amplifier in one compact and easily applicable 
power brick.
ax Standards Review
24 | September 2018 | audioxpress.com
The new HDMI standard adds a unique feature to preserve AVR compatibility. In this article, 
Marshall Goldberg explains how Enhanced Audio Return Channel (eARC) works.
HDMI Specification 2.1 is the standard’s most 
substantial upgrade to date. It quadruples the video 
resolution from 8 megapixels of 4K to 32-megapixel 
resolution of 8K. HDMI Specification 2.1 adds a 
host of additional features, including Dynamic 
HDR, Variable Refresh Rate (VRR), and Quick Media 
Switching. 
One problem with all of these new upgraded 
features is that audio devices such as AV Receivers 
(AVRs) will become incompatible. For example, if 
a gamer purchases a new TV and game console 
with VRR, the AVR that goes in between the game 
console and the TV will need to be upgraded—even 
though the audio features of the AVR don’t change. 
If the AVR isn’t placed in the HDMI path between 
the game console and the TV, it won’t be able to 
play advanced audio formats such as Dolby Atmos 
and DTS:X.
To preserve AVR compatibility long into the 
future, HDMI Specification 2.1 includes the new 
Enhanced Audio Return Channel (eARC) feature. 
eARC technology introduces forward compatibility 
between the AVR and the TV, along with 
uncompromising quality and greater ease of use. 
eARC provides a number of improvements over 
existing home theater audio connectivity methods:
• Forward-compatible, so the AVR will continue 
to work with newer televisions
• Delivers uncompromised audio quality nearly 
30 times the bandwidth of optical
• Far better inter-brand interoperability
• Easier to use
• Simpler connectivity
Challenges of Forward Compatibility
In the past eight years, we’ve seen the upgrade 
from 1080p to 4K video. To use 4K video content, 
new HDCP 2.2 copy protection technology was 
required. This rollout was released in products in 
steps, over several years:
• 4K30 Video with HDCP 1.4 Copy Protection
• 4K30 Video with HDCP 2.2
• 4K60 Video with HDCP 2.2
• 4K60 Video with HDCP 2.2 and High Dynamic 
Range support
As a result, most 4K AVRs could not fully support 
4K video found in Blu-ray and elsewhere. Only the 
most recently released AVRs offer full compatibility 
with today’s 4K Blu-ray discs and other 4K film 
content (see Figure 1). 
By
Marshall Goldberg
(Lattice Semiconductor, Senior Product Marketing Manager)
HDMI Specification 2.1 
Improves Home Theater Audio
with eARC
audioxpress.com | September 2018 | 25 
eARC is designed to safeguard AVR and soundbar 
purchasers against obsolescence, which would 
otherwise occur as the progressive roll-out of 
HDMI Specification 2.1-enabled products continues. 
eARC also substantially improves the overall quality, 
interoperability, usability, and connectivity of home 
theater systems. 
Delivering Forward Compatibility
eARC removes the audio device from the video 
path with a conventional AVR, audio, and video both 
flowing through the AVR and onto the TV. 
With eARC, all HDMI devices are attached directly 
to the TV. The TV then sends the audio-only eARC 
signal to the AVR. This type of connection eliminates 
video format issues, which dramatically extends the 
lifetime of an eARC-enabled AVR. 
To connect an eARC-enabled AVR or soundbar to 
a television, attach an “HDMI Cable with Ethernet” 
from the audio device to the TV’s HDMI input with 
eARC. Then, attach the remaining HDMI devices 
to the television. The connection is illustrated in 
Figure 2.
It is easier to connect a home theater system 
this way. Instead of using the AVR remote to switch 
inputs on the AVR and the TV remote to switch inputs 
on the TV, the user of an eARC-based system simply 
uses the TV remote to handle all input switching. 
This includes not only HDMI inputs, but also apps 
built into the TV and any other inputs (e.g., analog 
video and the TV tuner). If the TV and AVR support 
the optional CEC feature, using the TV remote will 
automatically send Power On/Off, Volume Up/Down, 
and Mute commands to the audio device. 
eARC is designed to ensure compatibility and 
interoperability between mixed-brand TV and AVR 
connections. The improved compatibility comes 
about because video doesn’t flow through the AVR 
and the discovery mechanism of eARC is dedicated 
specifically to audio devices and format discovery, 
as described later in the article. 
eARC is a must-have feature to look for when 
Figure 2: AVR connection 
with eARC
Figure 1: AVR connection 
without eARC
Figure 3: HDMI cable 
with Ethernet to support 
transmission of the eARC 
signal
ax Standards Review
26 | September 2018 | audioxpress.com
shopping for an HDMI-enabled AVR or soundbar 
because eARC is the only way to ensure future 
compatibility with HDMI 2.1 devices. In addition 
to future compatibility, eARC brings substantial 
improvements in simplicity, audio performance, 
and compatibility over any other audio interface, 
and is designed to last decades into the future. 
Technical Details
How Does eARC Work?
eARC transmits a high bitrate audio signal from 
the television to the audio device using an HDMI 
cable with Ethernet, which was designed back in 
2011, with HDMI Specification 1.4. The HDMI cable 
with Ethernet has the same connector and pins 
as an ordinary HDMI cable. However, inside an 
HDMI cable with Ethernet, pins 14, 15, and 19 are 
constructed as a twisted, shielded pair—originally 
intended to serve as an Ethernet channel in addition 
to the HDMI audio/video signal (see Figure 3). In 
HDMI cables without the Ethernet feature, these are 
simply straight-through pins, which cannot support 
transmission of the eARC signal. 
Because it uses this shielded pair to transmit the 
audio signal, eARC can support all of the latest home 
theater surround sound formats, such as DTS:X, 
Dolby Atmos, and Dolby MAT 2.0. Uncompressed, 
audio can be up to eight channels, 192 kHz, 24-bits. 
Discovery is also built into the eARC signal, so the 
TV always knows the audio device’s capabilities, 
and whether it’s powered up and receiving audio.
eARC Compared to Other Audio Connections
There are other methods to transmit audio 
from the television to an audio device, such as 
optical (TOSLINK) and HDMI-ARC. However, these 
methods have limitations, which eARC overcomes. 
The comparisons shown in Table 1 demonstrates 
some of these differences. 
A key benefit of eARC is support for all of 
the home theater audio formats in which can be 
transmitted in an HDMI signal. Previous generation 
ARC transmits over a single pin which only supports 
1 Mb/s audio, making it insufficient for today’s high-
end audio standards. Particularly in Blu-ray formats, 
audio almost always uses higher bandwidth than 
TOSLINK and ARC can support, and some titles use 
the full bandwidth of eARC. 
For eARC to work, both the television and the 
audio device must support eARC. However, eARC 
devices can be designed to automatically fallback 
to ARC mode if the other device does not support 
eARC—so you can buy a new eARC AVR, and it will 
work in ARC mode with your current TV. And when 
you upgrade your TV someday in the future, your 
AVR will be ready. 
eARC can lower the cost of audio devices, 
because there is no need for the AVR or soundbar 
to support four to eight input/output ports of ultra-
high-speed 48 Gb video.
Figure 4: Lattice SiI9437 
and SiI9438 solutions for 
eARC
Function TOSLINKARC eARC
Cable Used Optical SPDIF HDMI HDMI with Ethernet
Stereo Support Yes Yes Yes
Compressed 5.1 Very Limited Yes Yes
Uncompressed 5.1 No No Yes
Uncompressed 7.1 No No Yes
High Bitrate (Dolby, 
True HD, DTS-HD) No No Yes
Dolby Atmos 
Support No No Yes
DTS:X Support No No Yes
Maximum Audio 
Bandwidth (Payload 
size)
~384 Kb/s ~1 Mb/s 37 Mb/s
Link Discovery No CEC eARC data channel
Capability Discovery 
(Audio EDID, etc.) None CEC eARC data channel
Lip Sync No (Optional) (Mandatory)
TV Mutes and 
Controls Volume No Yes (CEC) Yes (CEC)
Powering TV Powers 
Aduio Device No Yes (CEC) Yes (CEC)
ARC Fallback No N/A Yes
Table 1: Comparing eARC with other audio transmission methods 
audioxpress.com | September 2018 | 27 
About the Author
Marshall Goldberg is a Senior Product Marketing Manager at Lattice Semiconductor, 
where he has overseen the company’s HDMI and MHL consumer products for more 
than 10 years. He serves as President of MHL, LLC, and as Chairman of the HDMI Forum 
Marketing Working Group, leading the group responsible for defining future features 
of the HDMI Specification. He is the company’s Technical Representative to the HDMI 
founders and also the HDMI Forum Technical Working Group. 
Lattice Solutions for eARC
SiI9437 and SiI9438
The Lattice SiI9437 eARC Receiver and the 
SiI9437 eARC Transmitter are 32-pin QFN ICs, which 
transmit and receive eARC over an HDMI cable. 
These two ICs were designed so that manufacturers 
of televisions, computer monitors, AVRs and 
soundbars could easily implement eARC onto their 
existing designs, even designs in which HDMI is 
integrated into a system-on-chip (SoC). 
The two ICs don’t transmit or receive HDMI 
signals; they connect only to the eARC pins, 14 
and 19. This enables the ICs to be integrated into 
solutions with existing HDMI transmitters and 
receivers, using any HDMI version (see Figure 4). 
The ICs bridge between the SPDIF/I2S audio 
interfaces and the differential eARC signal, up to the 
full rate of 98 Mb/s. An I2C interface on the ICs is 
used for control, and also for a bridge of the eARC 
Data Channel, which is supported by the SiI9437/
SiI9438 hardware. The SiI9437 and the SiI9438 eARC 
Receiver and Transmitter are available in volume 
now, as well as their corresponding development 
kits, CP9437 and CP9438. 
Conclusion
With a design that wil l ensure forward 
compatibility with future devices, substantial 
improvements in simplicity, audio performance, 
and better compatibility over any other audio 
interface, eARC provides a significant upgrade to 
the home theater sound experience and should be 
a key feature when shopping for an HDMI-enabled 
AVR or soundbar. It is very likely that eARC will find 
its way into most, and perhaps all, HDMI-based 
audio devices in the coming year or two. 
(The Adopted Trademarks HDMI, HDMI High-
Definition Multimedia Interface, and the HDMI Logo 
are trademarks or registered trademarks of HDMI 
Licensing Administrator, Inc,. in the United States 
and other countries.) ax
I know that is putting it bluntly, however I believe 
in speaking of things as they are and not as we would 
like them to be. That is the root of the problem now 
and in the past—it’s crazy when you think about it 
that loudspeakers are between 90 and 100 years 
old now (depending on whether you count Jensen 
or Rice & Kellogg as the first), and we still cannot 
trust our measurements!
Dealing with Anechoic Chambers
I have been involved with loudspeakers and 
transducers for the reproduction of sound for most 
of my career, in the early days, we didn’t have access 
to much equipment and things were initially set up 
by ear. Later at Bowers & Wilkins and Goodmans, I 
was introduced to anechoic chambers and proper test 
equipment. Even then, I thought it seemed odd that 
soundproofed rooms, usually the size and often the 
cost of a house, was considered the way of measuring 
loudspeakers—oddly enough there is not even a lot 
of theory to back it up! Do we listen to loudspeakers 
in an anechoic room?
Anyway, I used anechoic chambers to measure 
loudspeaker systems. These were usually located 
in the middle of the anechoic space equally-spaced 
between the wedges of the chambers. Usually this 
meant mounting big heavy loudspeakers and light 
delicate microphones on floppy mesh floors situated 
around one-third of the way up the vertical height of 
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I have just returned from the recent 
Audio Engineering Society (AES) 
144th International Convention in 
Milan, Italy, where I was among 
many colleagues who are trying 
to solve one of the outstanding 
problems of loudspeaker drive units 
today. Namely, being able to trust 
the measurements that are being 
presented to us. 
By
Geoff Hill
(United Kingdom)
Tetrahedral Test Chambers were demonstrated at the Audio Engineering 
Society (AES) convention in New York, 2015. Contrary to anechoic 
chambers, these systems define the geometry, the environment, and 
a calibration methodology. They minimize internal reflections and 
standing waves and provide internal measurements in a pressurized 
environment, which is readily compensated with modern equipment. 
Finally, they increase the acoustic isolation—making high-quality 
measurements possible in a production environment.
Thoughts on AES X223 
and X241 Speaker 
Measurement Projects
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the chamber. The results of this, of course, was that 
every measurement was different from every other 
one. Because of the inherent imprecision, I remember 
being trained to line up the microphone and the 
loudspeaker using a 1 m rule. And because everything 
uncontrollably bobbed up and down on the mesh 
floor, and because of the variable location within the 
chamber, every result was different from every other 
chamber, and this would show any and every time I 
needed to compare results produced by a contractor 
or sub supplier. Various other unknown factors could 
interfere (e.g., the one chamber I worked that had an 
enormous metal plate supported on a steel beam just 
below the door entrance, in which the loudspeaker 
drivers were mounted!). Crazily enough, this situation 
has remained fundamentally the same throughout 
my career to date.
The AES Documents
Many of us in the Audio Engineering Society 
(AES) have been aware of and working toward a 
unified solution to this problem for a while. The AES 
subcommittee on Acoustics assigned the problem to 
SC-04-03 in 2013, which kicked off project X223 at 
the 134th convention in New York in October 2013. 
And, at the 135th convention in Berlin, Germany, 
in May 2014, it was decided to issue this as an 
Information Document.
An Information Document is probably best 
described as an informal standard that has gone 
through the same full review process that a standard 
has, but it is more advisory in nature than a full 
standard, however, it represents our best collective 
thoughts on a subject.
The problem with Loudspeaker Testing in 
production is one many of you might have come 
across at one time or other. In essence it always 
boils down to the following:
• Production and Sales want to ship as much 
acceptable product as possible.
• Quality and the Customer only want a perfect 
product. Neither want any product with any 
faults as these will cause trouble and usually 
expensive trouble later.
• As a customer or user, you need to know that 
you are getting a loudspeaker as close to 
specification as possible, in an ideal world you 
might want to accept loudspeaker drive units 
purely upon measurement data. But can you?
• How do you make a judgment between perfect 
and acceptable? The standards IEC 60268-5, JIS 
etc., recommend using an anechoic or free-field 
environment. This is fine for a one-off or very 
limited quantities but for production quantities 
that regularly range from tens to hundreds of 
thousandsif not millions, it is not a practical 
option.
• Anyway, do you have a chamber? Unlikely, 
unless you are in the Military, Academia, or in 
a Research and Development department. Even 
many loudspeaker companies don’t, but they 
still need to measure the many characteristics 
of a loudspeaker but in a typically noisy factory 
environment.
As I said earlier, the AES Subcommittee on 
Loudspeakers and Headphones SC-04-03, is aware of 
this problem and has been working toward producing 
advice in this area.
There are two related Information documents, 
X223 and X241. Project X223 concentrates on the 
acoustic test chambers: “To consider factors affecting 
the interchangeability of measurement data from 
simple loudspeaker comparison chambers.”
The problem is essentially simple in theory while 
being complex in practice. Essentially it boils down 
as follows:
(1) The Standards demand anechoic or free-field 
measurements but of loudspeakers drivers on 
baffles (regardless of how they are designed to 
be used in practice—usually in some enclosure).
(2) Practicalities demand measurement in some form 
of test box or cabin if only to reduce the external 
noise produced and to protect workers hearing 
as well as to stop external noises interfering with 
test and measurement equipment.
Statistical Analysis on 50 measurements made using the same driver on a TTC900 test 
chambers, without any disturbances, to test the stability of the system. These were 
within 0.15 dB at 2 dB Standard Deviations over the operating range.
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The problem with using a true anechoic or free-
field environment is that they are horrendously big 
and expensive, while simultaneously difficult to set 
up for precision repeatable measurements especially 
when measuring small loudspeaker drive units.
Test boxes are notoriously individual—often 
having significant differences one to another 
and making it all but impossible to reconcile 
measurements made in facility with another, often 
even within a single facility, unless they use common 
test chambers.
Project X241 intends to: “Specify parameters 
and methods of measurement required for end-of-
line quality control tests performed on loudspeaker 
drivers manufactured for automotive, consumer and 
professional applications. It includes requirements 
for mechanical setup. It does not consider 
measurements of loudspeaker systems or multi-
driver arrangements.”
While X241 originated in Automotive Audio, it 
looks to wider concerns around all the various factors 
that can affect any measurement system.
As of course many things can and do change the 
performance of loudspeaker drive units in practice. 
These include:
• Temperature—think of how a loudspeaker’s 
surround changes its stiffness between usage 
in Finland on a winter morning (-30°C), to 
operating at the height of a summer day in 
Saudi Arabia (+40°C) being quite possible
• Humidity—paper or fiber diaphragms operating 
in dry desert conditions (0% RH to moist 
equatorial rain forests (90% RH)
• Air Pressure 
• External Noise
• Capable of operating correctly with wind, rain, etc.
Between them, X223 and X241 tackle these 
problems and advise current best practice.
The loudspeaker needs to produce its rated 
output without introducing unwanted distortion or 
noises and the quality or otherwise is measured using 
an audio analysis system powering an amplifier, 
usually a digital Fast Fourier Transform (FFT) system 
today, and measured by a microphone back to the 
analysis system. So, it’s quite a complex system with 
an awful lot of ways to go wrong and truthfully just 
a few ways to get it right.
One of the key characteristics of any 
measurement system is accuracy and the other is 
consistency. Both are important and need verification 
and, ultimately, it is determining how accurate and 
the level that this accuracy can be trusted to that is 
the mark of any measurement system.
GageR&R 
Yes, I know the spelling is awkward—but that is 
what it was known as in the Automotive Industry, 
which developed it as a methodology for controlling 
the dimensions of individual parts or complete 
components. One of the core concepts in GageR&R 
is best explained by breaking the name into its 
component parts: 
• Gage is just a shortened version of Gauge
• R for Repeatability 
• R for Reproducibility
The key here is that there is a movement away 
from just a measurement toward a measurement 
system and the realization that consistency is key. 
But how do you effectively measure a loudspeaker’s 
parameters? Let’s remember that GageR&R 
was fundamentally developed to handle single-
dimensional mechanical data.
Apart from specialized Thiele-Small (T-S) 
parameters, most loudspeaker data is inherently 
multi-dimensional simultaneously having frequency 
vs. amplitude, frequency vs. impedance, frequency 
vs. distortion, and so on. The key here is that 
potentially there is a lot of data that needs organizing 
so that we can clearly see what is going on.
Personally, I first started exploring this area 
when working for Ericsson’s Research Triangle Park 
in North Carolina in 2000. There, I had to measure 
and analyze data from batches of thousands at 
a time. These ideas were later developed into a 
more general methodology. I outlined some of the 
principles behind this in a paper I presented to 
the Association of Loudspeaker Manufacturing & 
Acoustics (ALMA) International in 2010 in Shenzhen, 
China. This has been extensively updated in my latest 
A TTC900 being set up for the first time—then and only then do you need to set up or 
calibrate the chamber’s correction curve.
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book, Loudspeaker Design Modeling and Measurement 
a Practical Guide to be published by Focal Press in 
October 2018.
The first requirement is data, and modern FFT-
based systems can produce this, however, even today 
few systems effectively gather and correlate this 
information. So, what is required?
• A list of known frequencies at which the 
parameter of interest is measured.
• A list of the value or level of the parameter.
• A means of correlating these measurements to 
an individual loudspeaker.
We can see the problem clearest if we overlay 
data from a large series of measurements, if we do 
so with typical electronics we get very consistent 
results with relatively small and predictable changes 
in gain, cut-off frequency etc. A loosed tolerance 
being say ±5% or ±0.5 dB. With a tight tolerance 
being ±0.1% or ±0.09 dB.
However, loudspeakers have been notoriously 
variable and unstable with very wide tolerances 
commonly accepted. Partly I believe this has occurred 
for four reasons. A wider tolerance range is easier 
to meet and causes less rejects. Most acoustical 
measurements have a large margin of error often 
±3 dB is accepted as a tight acoustical tolerance. At 
low and high frequencies, ±6 dB would be relatively 
normal. Many loudspeaker test chambers allow large 
variations in setup, and the ruling standards seem 
to accept such variations without question.
GageR&R, ANOVA Analysis of Variation, Statistical 
Analysis-Standard Deviation, Normal distribution, 
Cpk & Ppk, and other methods, all give us tools to 
understand the variations present in large samples 
of data and when we understand thesewe can 
potentially control them.
It was lack of consistency in measurements 
that plagued so many of us through our careers 
whether in hi-fi, loudspeaker drive units, consumer 
electronics, or mobile telephony.
In my mind there are two separate questions 
to answer: Are such variations due to loudspeakers 
themselves or are they due to the methods and 
methodologies that we have or are still applying to 
such measurements?
It is my feeling that it is the methodologies that 
are primarily responsible for this situation—yes, 
undoubtedly some loudspeaker drive units have 
significant variations from ±3 dB to ±6 dB. But let’s 
remember ±6 dB is a 2:1 variation or ±200% tolerance 
range… Such a tolerance range is difficult to justify 
today. I believe it is important to ask why this has 
occurred before we ask what can be done about it.
In loudspeaker drive units, the main causes of 
variation in sensitivity (loudness) are:
• Magnet material—cheaper materials are either 
lower spec or more variable—sometimes the 
grade is subtly changed until the customer 
notices and complains!
• Thinner or thicker grades of wire being used 
or more likely poor control of tension while 
winding. These can change DCR and or wind 
length.
• Thicker or thinner diaphragm materials together 
with under/over doping/damping
• Wrong material—rare but has happened.
• Dimensional differences—excessive tolerances 
especially in the motor unit can lose lots of 
power.
The 68 measurements are of six drivers of the same type but repeatedly measured 
over the course of half a day and demonstrate the stability of both the measurement 
system and the variations of this particular drive unit.
About the Author
Geoff Hill is best described as a passionate engineer and can get frustrated when 
there is a better way to do something, but nobody else can see it. After qualifying 
in electronics (City and Guilds FTC in Telecommunications), he co-founded Impulse, 
a company that made high-efficiency horn loudspeakers at a time when the LS3/5a 
and Linn Kan were all the rage. Later, Geoff has worked with many companies and 
along the way he has designed everything to do with sound, including automotive, 
mobile phones, PA, hi-fi systems, and loudspeaker drivers from 25 mm tweeters 
up to 18” sub-bass drivers. His latest venture, Hill Acoustics, marks a return to 
his roots, combining 40-plus years’ experience. An active member of the Audio 
Engineering Society (AES), he has worked with various committees over the years 
to help and assist where and when he can. He is currently active within AES 
SC-04-03 on the subject of loudspeaker measurements and is a British member of 
IEC TC100. Geoff can be reached via www.geoff-hill.com.
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• Wrong smaller magnet. The inside diameter can 
be increased reducing the magnet volume. This 
is hard to detect and only weighing or stripping 
down the parts reveals exactly what is going on.
• Extra heavy or light dust caps.
Conclusion
As we can see, this is quite a non-exhaustive 
list but provided the measurement system is stable 
and accurate, the final sensitivity can be accurately 
measured. The next point is the accuracy with which 
the frequency response can be measured—in a free-
field or anechoic condition theoretically there are no 
sources of interference, in practice there are many.
Most are fairly benign and change just the levels 
due to distance or change the response shape due 
to the angle and or azimuth of measurement. Or 
changes of response due to diffraction or reflection 
and cancellation which are simple to calculate.
However, when we consider measurements inside 
a test chamber, things get much more complicated 
as there are now multiple sources including those 
previously mentioned. Still, test chambers enable 
us to design and simultaneously control the acoustic 
environment while pinning down the physical 
dimensions and enabling precision equalization, 
thus enabling more accurate measurements to be 
made that are repeatable and reproducible on a 
truly industrial scale for the first time.
All of which sounds great but how should we 
actually go about proving measurement capability? 
As we already mentioned, we should use statistical 
tools. Primarily, these decouple the shape or curve 
of a measurement from the quality or stability of the 
aggregate of a series of measurements of nominally 
similar ones, thus, we can separate out variations in 
loudspeakers themselves from variations due to the 
measurement process itself or the environmental 
aspects. ax
Resources
L. Beranek, Acoustical Measurements, Revised Edition, Acoustical Society of 
America; Revised edition (June 1, 1988)
G. Hill, Loudspeaker Design Modeling & Measurement—A Practical 
Introduction, October 2018.
Project AES-X223 Initiation, “AES-X223, Loudspeaker Driver Comparison 
Chambers,” Audio Engineering Society, October 2014, 
www.aes.org/standards/meetings/init-projects/aes-x223-init.cfm
Project AES-X241 Initiation, “AES Project X241, End of Line Testing for 
Production Loudspeaker Drivers, Audio Engineering Society, March 2017, 
www.aes.org/standards/meetings/init-projects/aes-x241-init.cfm
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Some people might think that I am trying to 
egg them on to better sound. Since I don’t want 
to get egg on my face, enough with the bad puns. 
The reality is that the shape of the egg has several 
amazing characteristics. Some of those have distinct 
advantages in speaker design. I’ll demonstrate how 
that has worked well in previous speakers and is 
the basis for an updated version based on more 
recent technology.
Why an Egg Shape?
If you are not a bird, the oval shape of the egg 
will not help keep your eggs from falling out of the 
nest. Nor will it make it easier to lay one. In fact, 
the complex shape makes it quite difficult for you to 
manufacture one. So why bother? We can take a hint 
from companies that use digesters in waste water 
treatment. Many of them are moving to egg-shaped 
digesters because the flow inside is more even with 
fewer dead spots. Now consider that speakers also 
work with a fluid, that fluid being air, and things that 
can reduce the disturbance of the flow will result in 
fewer acoustic anomalies.
The egg shape has previously been used in 
speaker design and seems to be coming back in 
vogue. Looking at some of the current offerings, I 
am sure the designers are primarily interested in the 
look rather than good sonic results. Others appear to 
be taking advantage of a true egg shape. A search 
on Google for egg-shaped speakers will quickly 
return lots of offerings including products from 
companies such as KEF, Samsung, Fujitsu, Munro, 
HiVi, Eclipse, and others. However my project was 
based on the work of John Otvos of the Canadian 
company Waveform. His flagship speaker, the Mach 
17, garnered rave reviews. One of the most salient 
comments in many reviews is that it was a speaker 
that sounded most like real music, not a high-end 
speaker.
According to what I have read, never having met 
Otvos, the design goal was to produce a speaker 
with very flat frequency and polar response across 
the widest possible frequency band, while retaining 
good transient response and dynamics. In order to 
do that, attention must be paid to diffraction.
Diffraction in Speaker Design
A rigorous discussion of diffraction is beyond 
this article’s scope. However, the references I have 
included do a thorough job of covering this subject. 
Diffraction has the ability to drastically alter the 
frequency response from the driver/housing 
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The Incredible Egg (Part 1)
Speaker buildersrejoice! In the first of this three-part article 
series, Tom Perazella details his latest bold speaker project, 
inspired by the Mach 17 speaker built in the 1990s by 
Waveform Research of Canada. This three-way design features 
the latest generation of SB Acoustics’ high-end Satori drivers 
and two Dayton 12” woofers in a separate sealed box base.
By
Thomas Perazella
(United States)
Sound Reproduction at Its Best
Photo 1: Tom Perazella’s 
completed Egg Speakers are 
shown as they premiered 
at Capital Audio Fest 2017. 
(Photo courtesy of Paul Elliot)
www.ap.com
Check out the
AECM206 Headphone Test Fixture
Testing Headphones?
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system. A well-behaved driver can produce a badly 
behaved acoustic output when in an enclosure where 
diffraction has not been adequately addressed. In 
the simplest form, diffraction is a change in direction 
of a wave as it passes by a solid object. The amount 
of bending is dependent on the object’s size and the 
wavelength of the wave. Without going into the math, 
it is simple to think of diffraction as a response to the 
change in impedance of the solid object at its edge 
compared to the air past that. Imagine a pressure 
wave moving along a flat surface with infinite length 
at one edge that reflects a constant impedance to 
the pressure of the wave. As the wave travels along 
the surface, the waveform is kept intact.
When the pressure wave reaches the edge of 
the flat surface, it sees a lower impedance and the 
pressure drops. That sudden change causes the 
pressure wave to move in another direction. The 
amount of change is a function of the ratio of the 
object’s size and the wavelength of the pressure 
wave. Once we get away from an infinitely long 
edge and to more realistic shapes (e.g., a sphere, 
a cube, or a rectangle), the effect of the diffraction 
changes. For a very comprehensive discussion 
of this effect and how it is affected by different 
shapes see the referenced works by Leo Beranek 
and Harry F. Olson.
Probably the worst shape to employ where 
diffraction is concerned is a flat sharp-edged circular 
disk with an acoustic source mounted in the center. 
The expanding wave will reach the edge where the 
impedance difference will affect the wave at the same 
time for each propagation direction. Two methods to 
mitigate the cumulative effect are to either change 
the distance to the edge so that no path length is 
the same distance from the source and to increase 
the radius of the edge so that the pressure wave 
does not see such a sharp change. A sufficiently 
large radius would minimize diffraction at all audio 
frequencies.
Why an Egg-Shaped Enclosure?
Simply put, an egg-shaped enclosure combines 
two of the techniques that minimize diffraction. One 
is distances from the acoustic center to the edges 
that continuously vary. The second is the absence 
of sharp edges. Probably the second-best shape is 
a sphere but as many researchers have discovered, 
differences in sound pressure along the sphere’s 
surface are not as well mitigated due to the sphere’s 
symmetry. As you get to shapes such as a cube with 
the driver mounted in the center of one side or a 
cylinder with the driver mounted on one flat end, 
the frequency response aberrations magnify.
Design Considerations for the Eggs
Every speaker is a compromise. If anyone tells 
you otherwise, don’t buy a used car from them. To 
determine the best path to take when designing a 
speaker, you should first rank the characteristics of 
sound that you value the most. The most common 
listed in my order of importance are flat frequency 
response, wide dynamic range, low distortion, broad 
frequency response, good transient response, and 
precise imaging. 
My reference system, which has been described 
in past audioXpress articles, superbly fulfills the 
needs of the first five criteria. If there is an area 
where it falls short, it is precise imaging. The mid-
bass and high-frequency line arrays are quite 
large and, unfortunately, in my current house they 
reside in a very small room. As a result, the imaging 
suffers. In my former house where they were in a 
much larger loft with a vaulted ceiling and a rear 
that opened to the first floor, the imaging was 
significantly better. 
One of the positive characteristics attributed to 
the Mach 17 speaker was precise imaging. Many 
mini monitors produce incredibly realistic imaging 
but suffer from narrow frequency range, limited 
dynamic range, and high distortion. My goal was to 
have a speaker that would move the ability to image 
up in the list without undue sacrifices in the other 
parameters. In addition, my reference system is tri-
amped, which I find the best way to go, but it limited 
my ability to test a single amplifier’s full range. The 
new speaker had to work in both full range and tri-
amped modes. Obviously, in the full-range mode 
the speaker needed a passive crossover, which is 
something that is not my favorite path. I basically 
have not built a passive crossover in about 40 years 
since I did my first active. Perhaps more about that 
Resources
L. Beranek, Acoustical Measurements, Revised Edition, American Institute of 
Physics, Cambridge, MA, 1988. 
L. Greenhill, “Waveform Research Mach 17 Loudspeaker,” Stereophile, July 30, 
2006 (first published June 30, 1997), 
www.stereophile.com/floorloudspeakers/697waveform/index.html.
H. Olson, Acoustical Engineering, Professional Audio Journals, 
Philadelphia, PA, 1991.
J. Otvos, “Waveform: Pt2 of 3,” January 2018, 
https://johnotvos.wordpress.com/2018/01/13/waveform-pt2-of-3.
T. Perazella, “True Bass Rides Again (Part 1)” audioXpress, April 2015. 
———, “True Bass Rides Again (Part 2)”, audioxpress, May 2015.
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preference in another article or two as it is such an 
involved and controversial subject.
Compromises
The two areas where I was willing to make 
some degree of compromise were the range of bass 
extension and the ultimate dynamic range. You will 
see that the final design is anything but a wimp, 
but compared to my reference system it gave up 
a little to allow me to use point source midrange 
and high-frequency drivers. On the positive side, 
the midrange and the tweeter are really superb 
drivers. When using the passive crossover, another 
compromise was the inability to immediately change 
speaker parameters as would be possible with an 
active crossover. Because of the difficulties and the 
expense of changing passive components, the ability 
to finely tune the results is limited.
The Three Main Players
In this design, there are three physical enclosures 
for each side of the stereo pair. The first enclosure 
is the egg containing the midrange and the tweeter. 
The second is the bass enclosure, which is a sealed 
box design, each containing two woofers. The third 
is an enclosure for an external passive crossover, 
which also contains alternate connections for bi- 
or tri-amping (see Photo 1). That enclosure also 
has a bottom with an acrylic plastic panel so that 
the components can be viewed when explaining the 
crossover at club meetings or shows.
Driver Selection
One of the factors in my choice of drivers 
was a news release I received from the company 
SB Acoustics about some new drivers in its premium 
Satori line. Two of the drivers were especially 
interesting not only in their description, but also 
the supplied specifications. It was immediately 
clear that these were the result of extensive design 
work and the use of premium materials. I thought 
they would complement the egg shape to produce 
outstanding results.
To match that performance, I needed woofers 
that had high output capability with low distortion 
and relatively high efficiency. In addition, I wanted 
good bass extension in a reasonably sized enclosure. 
I chose to use two 12” woofers from the Dayton 
Audio Reference HODVC series.
The Tweeter
Driver design has progressed immensely since 
I started building speakers. The area of tweeters 
has especially benefited from the introduction of 
new configurations and materials. The tweeter 
I chose, the Satori model TW28BN, is a prime 
example of those improvements (see Photo 2). 
When designing a tweeter, the first choice has to 
be the type such as dynamic, Air Motion Transformer, 
electrostatic, planar magnetic, and plasma. By far 
the most universally used type is dynamic and 
of that category, the dome tweeter has assumed 
prominence. The TW28BN is in that category.
In a dome tweeter, one of the most critical 
components is the dome itself. Because of the 
high frequencies, the dome must exhibit several 
important characteristics. It must be rigid enough 
to stand the acceleration forces at high frequencies 
and amplitudes. The speed of propagation in the 
cone material must be fast to evenly distribute 
the forces over the dome at high frequencies. The 
material must also be light to help with acceleration 
without requiring undue force, which would stress 
the dome. These factors and others are reflected 
in the parameters’ speed of sound in the material; 
density; Young’s modulus, which indicates stiffness, 
Photo 2: The Satori 
beryllium tweeter front 
view (a) and rear view (b)
a)
b)
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tensile strength; and Poisson’s Ratio, the ability to 
keep different modes of propagation separate. Those 
are combined to come up with a figure of merit 
and, of all the current materials, beryllium ranks as 
the best combination of those properties. It is also 
the material used in the Satori tweeter dome. It is 
interesting to note that beryllium can be very toxic 
during the manufacturing or the disposal process so 
the back of the tweeter has an orange label warning 
about improper disposal.
The on-axis response of this tweeter is 
extraordinary. It is virtually flat on axis from 1.5 kHz 
to 20 kHz with a very slight smooth rise to 40 kHz. 
The resonances so often seen that are caused by 
material limitations at high frequencies in aluminum 
and titanium dome tweeters are absent. Nominal 
impedance is 4 Ω and is very flat above 1 kHz. 
Specifications are available on the SB Acoustics 
website (www.sbacoustics.com).
The Midrange
The Satori midrange that I selected was 
the model MR16P-4 (see Photo 3). This 6.5” 
cone driver has a number of interesting design 
features, including a neodymium magnet structure, 
aerodynamic legs on the cast aluminum basket to 
minimize turbulence, a cone made of Egyptian 
papyrus, a copper sleeve on the pole piece, and 
many more. The response was very flat and the off-
axis response to 60° matched the on-axis response 
to about 2.5 kHz. Nominal impedance was 4 Ω and 
very flat through the expected crossover range. It 
is important to note that having the same radiation 
pattern at the crossover point is very important. If 
that criteria is not met, room contributions through 
the crossover region will be different—leading to 
difficulties in perceived timber and image placement.
The Woofer
The Dayton Audio RSS315HO-44 12” driver was 
chosen to do the heavy lifting in the bass range (see 
Photo 4). I have had great success with Dayton 
woofers in the past and my reference subwoofer 
consists of four Dayton DVC 15 drivers, each in a 
sealed box. A description of the performance of those 
drivers can be found in the audioXpress articles 
“True Bass Rides Again” and “True Bass in a Small 
Space.” (see Resources).
Since these speakers occasionally were to be 
used for amplifier testing with no EQ, the bass 
response needed to go low, yet not require a huge 
enclosure while having reasonable sensitivity. If 
you remember Hoffman’s Iron Law, you can have 
your choice of two of three parameters—low bass, 
small enclosure volume, or high sensitivity. That 
is generally true, but you can achieve a better 
combination of the three if you have a very strong 
motor in the driver. Since driver resonance is a 
mass-compliance issue, you can get a low free air 
resonance, Fs, by having a low moving mass and a 
compliant suspension or a high moving mass and 
a less compliant suspension. For a given motor 
strength, the low moving mass solution gives higher 
sensitivity but the driver’s equivalent volume, Vas, is 
higher. That is why for low bass with high sensitivity 
you need a larger enclosure.
Enter today’s monster motors. The above 
scenario is true if you keep the motor strength, 
BL, as expressed in Tesla meters (Tm) constant. In 
the past, for a 12” driver, it was common to have 
a motor strength in the range of 7 Tm to 10 Tm. 
The Dayton driver I used has a BL of a whopping 
26.5 Tm. As a result, you can have a large moving 
mass, a relatively stiff suspension, and a resulting 
Fs of 21.5 Hz. The Vas is relatively low at 61.3 ltr 
and because of the exceptionally strong motor, 
the sensitivity is 84.6 dB at 1 W (2.83 V) with an 
8 Ω impedance. To increase the maximum linear 
Photo 3: Satori midrange 
front view (a) and rear 
view (b)
a)
b)
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volume displacement, two woofers would be used in 
parallel, also increasing the sensitivity to 90.6 dB at 
the same drive level of 2.83 V putting it closer to the 
sensitivities of the tweeter at 95 dB and midrange at 
92.5 dB. This would help minimize the padding that 
would have to be done to get the drivers balanced 
and result in an overall higher sensitivity.
This driver has an extremely flat response out to 
2 kHz thanks in part to the aluminum cone and dust 
cap. However, there is literally a wrinkle to the story 
as reflected in the impedance curve around 2,100 Hz 
that also is reflected in a very substantial peak in the 
response of about 15 dB at that frequency. After some 
research, I found that the peak at 2,100 Hz is the result 
of the aluminum dust cap resonating at that point. The 
combination of the aluminum cone and the dust cap 
result in very high rigidity that greatly reduces the 
tendency to breakup with resulting distortion at high 
power levels. That frequency is way past the intended 
crossover point of 200 Hz, but more about that later 
when discussing the passive crossover.
The Egg Enclosure
How do you make a hollow egg enclosure for the 
midrange and the tweeter? Well, you could probably 
spend a small fortune to tool up for an injection-
molded housing out of some composite material 
that would have excellent properties. For a one-off 
project, that is not very practical. You could also 
spend a lot of time with a CAD program and then 
use the resulting file to produce the egg out of wood 
or other suitable material using a CNC router. You 
could also use the file to build the egg using a 3D 
printer but printers that can handle that size are 
expensive.
I decided to see what could be done just using 
tools that are readily available to most home 
woodworkers. These eggs were produced using a 
table saw, drill press, handheld router, wood lathe, 
drawing tools, a shaping burr, clamps, and three 
clamping fixtures. If you have ever used an egg 
slicer to garnish your salad with hard-boiled egg 
slices, you will understand the method I decided to 
use. It is actually the reverse process. What I did 
was to create horizontal slices of the egg out of 3/4” 
MDF and then glued them together.
The first step was to determine the shape of the 
egg. Enter the Internet. I searched for line drawings 
of eggs and finally found a shape I liked that was 
called the “large Easter egg.” It was obviously not 
drawn for a speaker but the shape looked very 
smooth with no abrupt discontinuities. 
To get it to the correct size I used the scaling 
function of Adobe Reader. When you go to the print 
menu, one of the choices is Custom Scale. I tried Photo 5: Egg enclosure drawing showing the MDF ring construction
Photo 4: DaytonAudio 12” 
Reference HO woofer front view 
(a) and rear view (b)
a)
b)
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various percent values until the image was printed in 
the desired size. My printer will handle a maximum 
paper width of 8” so I had to do the final print in 
two sections and tape them together.
The egg would have a truncated bottom so 
that it could sit on a flat surface. The final external 
dimensions were a height of 13.5” and the width 
at the widest part of 11.5.” With those dimensions, 
it was determined that 18 slices of MDF would be 
needed. Since the mounting surface for the drivers 
would need to be flat, allowances for that shape had 
to be made in the drawing. As a result, a few of the 
slices were circular but the majority of them were 
“D”-shaped so that a flat surface would be created. 
Those shapes were drawn inside the original egg 
outline and served as a guide to determine the sizes 
of the cuts for each piece (see Photo 5). 
Because of the egg’s shape, the distance from 
the flat front surface on each slice to the center 
varied with each piece. To determine all of the 
sizes, I created an Excel spreadsheet that listed all 
the dimensions used in both fractions and decimal 
equivalents. These dimensions were used to set the 
center point of a Jasper Jig router guide back the 
appropriate distance from the straight edge of a 
sheet of MDF. Doing this eliminated the need to cut 
a straight slice off a circular piece.
When the router was used to cut the MDF, it 
would exit past the straight edge of the MDF at 
different points depending on the center point and 
the radius setting on the jig. A second cut was made 
to describe the internal volume of the “D” that used 
the same center point but a smaller radius. That 
resulted in a uniform wall thickness around the slice 
About the Author
Thomas Perazella is a retired Director of IT. He received a BS from the University 
of California, Berkeley campus. He is a Past President of the Rockville Chapter 
of the Izaak Walton League of America, one of the oldest national conservation 
organizations in the US and currently is the Treasurer. Audio has been his passion 
for more than 50 years and he is a member of the Audio Engineering Society, the 
Boston Audio Society, the Philadelphia Area Audio Group, the DC HiFi Group, and 
the DC Audio DIY Group. He has written for audioXpress magazine and prior to 
that for its predecessor, Speaker Builder. In addition to audio, his interests include 
photography, cooking and competition pistol shooting. He has authored several 
articles in professional audio journals and taught commercial lighting at the Winona 
School of Photography. Recently, he received a patent on a cost-effective high-
efficiency LED lighting system for commercial and residential buildings.
Photo 6: MDF ring first stage Photo 7: MDF ring second stage
Photo 8: Aligning rings for registration
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(see Photo 6). Finally, the center piece was removed 
from the slice to produce the final “D” shaped slice 
(see Photo 7). 
I had to devise a method to insure that the 
stacking of all the slices would result in a flat and 
vertically aligned front for driver mounting. To do 
this, I thought back to the days before computers 
where film separations for making CMYK negatives 
used for color lithographic printing were kept aligned 
by using punches and registration pins. I borrowed 
that concept to insure that adjacent rings would be 
aligned during assembly assuring a flat surface. 
First, all the rings were numbered to keep them in 
order. Next, each adjacent pair was placed next to 
each other with the flat part of the “D” resting on 
the surface of my table saw (see Photo 8). They were 
aligned horizontally so the centers were coincident. 
Then they were clamped together. This assured 
that the alignment would remain constant in the 
registration phase.
The easiest was to do the registration was to 
make alignment pins from 1/8” dowel. I cut pieces 
to the appropriate length. I drilled four holes around 
the circumference of each ring pair that would accept 
the dowels. Once the clamps were removed, the 
holes could be used to receive the registration pins 
during gluing assuring the alignment and constant 
positioning during the gluing process. Since the 
dowels were wood, they would remain in the rings 
as part of their structure (see Photo 9).
The second article in this three-part series will 
begin with assembling the eggs. ax
Editor’s Note: All audioXpress articles from 2001 to 
present can be found on the aX Cache, a USB drive 
available from www.cc-webshop.com.
Photo 9: Ring registration pins
In the late 1960s, the American electronics 
industry still enjoyed a predominance in the 
manufacturing and selling of consumer electronics, 
innovation was found in many areas including that 
of home audio. Stereo products had matured since 
their introduction in 1958 and sales of console 
and portable stereos were common place. I was a 
newly minted engineer in 1968 working for Admiral 
Corp., which with five plants in Illinois and a cabinet 
plant in Indiana, was one of the largest consumer 
electronics manufacturers in the US. 
In 1968, Admiral marketing in Chicago was 
looking for a unique product that provided an 
advantage over the competition and generate more 
interest in stereo among consumers. 
Most mass produced stereo consoles of the 
day were not much better than the old open back 
radio consoles of the 1930s and 1940s, and certainly 
were not audiophile quality in sound. The tuners 
and amplifiers produced to drive the dual channel 
consoles were of better quality than the speakers 
they were driving. That was mostly due to the higher 
cost of configuring a single cabinet with two properly 
baffled speakers and except for a few “high end” 
models, the open back cabinet was predominant.
The Tunnel Reflex Concept
One of the engineers at Admiral, John Virva, 
thought about the problem of providing better audio 
in the compact cabinets of the time and came up 
with a simple design that borrowed from the prior 
art of the past. Originally called a “Tunnel Reflex,” 
the design used a single full-range high-compliance 
driver loaded by a transmission line-style tunnel, 
much like the original acoustic labyrinth design of 
Benjamin Olney, of the Stromberg Carlson Co., in 
1936. 
The original, which used an 8” full-range 
driver, was very popular as it eliminated the single 
note bass of the old open back consoles. Admiral 
adapted Virva’s design for use in console radio/
phonographs, TV consoles, and portable stereo 
system (see Photo 1). Driver size varied from 3” 
to 5.25”, depending on the application, and on some 
products a tweeter was added for better highs. 
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A Modified Version of the 
Bullhorn Speaker System
After recalling an unique design that produced an amazing 
sound from small drivers, Ken Bird decided to build a 
modified version of this tunnel reflex system. In this article, 
he details how you can make your own set of these modified 
“Bullhorn” speakers. 
Photo 1: The original Admiral Tunnel Reflex of 1968. This 
version used a 5.4” woofer and a 3” paper tweeter with 
a single 4.7 µF capacitor for the crossover. 
By
Ken Bird
(United States)
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On August 11, 1970, a patent, No. 3,523,589, was 
issued to Virva for the design (see Figure 1). It is 
very similar in description to Olney’s 1936 patent 
on the Acoustic Labyrinth, US Patent No. 2,031,500. 
I encourage readers to download these patents to 
learn more about the concept and its application 
to this project. 
Admiral marketing did not like the name Tunnel 
Reflex, as it was considered “too technical” for the 
average consumer. Instead, theycame up with the 
name “The Bullhorn Speaker.” The Bullhorns did 
produce an amazing sound from the small drivers 
employed so I decided to build a modified version 
of this legacy system. 
The Driver
In his patent, Virva specified a 5.25” driver with 
a resonance of 58 cycles (Hertz) and claimed that 
the vented folded column could achieve a low end 
of 30 cycles (Hertz). This version uses a modified 
MCM coaxial 5.25” ceiling speaker with a simple 
crossover (see Photo 2). Any 5.25” full-range driver 
with a resonance from 50 Hz to 70 Hz and a Qts of 
0.5 and up should work fine in the enclosure. Many 
coaxial drivers that are used in automobile sound 
systems would also be applicable. 
The Enclosure
I constructed the enclosure from 1 × 6 select 
pine for the interior tunnel and 1/4” plywood for 
the sides (see Photo 3 and Photo 4). I lined all of 
the interior surfaces with 1/2” polyester batting to 
dampen the rear highs and mid frequencies (see 
Photo 5 and Photo 6). The original design used 
1/4” felt for the lining but then in later production, 
it was switched to the polyester that worked just 
as well but at less cost. The distance between the 
two upper partitions should be 5.25” with the last 
length of the tunnel at 4.5”.
Figure 1: The original Tunnel Reflex patent drawing from 1970.
Photo 2: The 5.4” MCM driver used in the project has a coaxial neodymium tweeter 
fed by a 4.9 µf capacitor. Any 5.4” full-range driver with a resonance of 50 Hz to 
70 Hz and a Qts of 0.5 or more would work in the enclosure. 
Parts For One Enclosure
Lumber
2× 1 × 6 Pine 18.75” long for top and bottom
1 1 × 6 Pine 17.25” long for the font baffle
2× 1 × 6 Pine 12” long, one for the back panel 
and one for first tunnel partition 
1 1 × 6 Pine 12.5” long for the second tunnel 
partition
2× 1/4” plywood panels 18.75” × 18.5” for the 
side panels
Speaker (see text)
MCM number 50-14005 
5.25” coaxial terminal cup Parts Express #260-303
4.7 µF 100 V capacitor Parts Express #027-332
Miscellaneous
36” length of #18-gauge speaker wire 
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audioxpress.com | September 2018 | 45 
Start by cutting out the 1 × 6 lumber to the 
dimensions shown on the drawing (see Figure 2) 
In the front panel, cut a 4.75” hole for the woofer. 
I used a jasper router jig to make the hole for the 
woofer and a jig saw for the terminal cup hole. 
I assembled the pine frame applying liquid 
nails to the butt surfaces aligned with a corner 
clamp and secured with 1 5/8 drywall screws 
(see Photo 7). 
With the frame assembled, apply a liberal bead 
of liquid nails to the end surfaces and carefully 
place and align the plywood side, making sure the 
tunnel partitions are straight. Nail the plywood 
around edges first using 1” finishing nails. 
Locate the tunnel interior walls by measuring 
from their panel positions and nail them also. You 
should now have one complete side in place and 
be ready to install the polyester padding. I used 
a staple gun to apply the padding to both the side 
walls and the tunnel partitions. Stretch the padding 
over the up side, stapling it to the flat surfaces of 
the partitions. Another alternative is to glue the 
padding down using water based “Tacky Glue.” 
I set all of the nails 1/16” apart using a nail set. 
Before applying the other side wall, pre-wire the 
enclosure by running a 36” length of speaker wire 
between the speaker cut out and the terminal cup. 
Once the padding and wiring is complete the 
other side panel can be installed. Apply a liberal 
bead of liquid nails along edges of the frame and 
nail it in place as you did with the other panel, 
quickly wiping off the excess with a damp rag. 
After allowing the enclosure joints to dry for a few 
hours, fill in the nail and screw holes with wood 
filler. Once that is dry, sand all the surfaces and 
add whatever finish you want. 
Photo 3: This is the 5.5” lumber pre-cut for the enclosure interior. Photo 4: I assembled the frame using picture frame clamps and 
1 1/4” drywall screws. 
Photo 5: With one of the 1/4” side panels attached, the enclosure is 
lined with 1/2” polyester batting and pre-wired with 18-gauge speaker 
wire. 
Photo 6: The polyester is stapled across the opposite side prior to 
attaching the final side panel. Make sure none of the poly material 
overflows on to the frame side.
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Driver Installation and Grille
In the original Bullhorns, those with separate 
tweeters, they only used only a single crossover 
capacitor feeding the tweeter, leaving the woofer 
to run full range. It was primarily a cost savings 
move rather than an engineering decision. 
If you want to add a tweeter, mount it below the 
woofer. The driver used in this version was designed 
for use as a wall-mounted speaker and has a 1/2” 
dome tweeter mounted on a stalk in the center of 
the driver with a coil/capacitor network wired to 
the speaker terminals. Listening tests showed that 
the ferrite coil did little to improve the sound and I 
removed it so that only the single 4.7 µF capacitor 
was used, allowing the woofer to run full range. 
The tweeter crosses at about 4500 Hz. 
Once the enclosures are finished, wire the 
speaker and the terminal cup, noting the proper 
polarities. I used 1/4” foam insulation strips as 
gaskets on both the speaker and the terminal cup. 
I then used 1” drywall screws to mount both items.
I formed the speaker grille from a piece of 1/2” 
plywood to allow space for the protruding tweeter 
and covered it with a loose weave black cotton 
Photo 7: This is the 
assembled enclosure 
frame.
Figure 2: The 
drawing was for the 
2018-version Bullhorn 
Speaker. 
Resources
B. Olney, US Patent No. 2,031,500, www.uspto.gov.
J. Virva, US Patent No. 3, 523,589 www.uspto.gov.
D. B. Weems, “Labyrinth Speakers for Hi Fi,” 
Popular Electronics, January 1972. Photo 8: The completed pair of enclosures are shown on temporary stands from an earlier 
project. The grille of the speaker on the right has been removed to show the driver.
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fabric. The grille is held on with Velcro self stick 
and can easily be removed for maintenance. If you 
use a driver with a whizzer cone, the grilles can be 
made thinner using 1/4” plywood. 
Testing and Sound Quality 
The completed pair are shown in Photo 8, with 
one of the grilles removed to show the driver. The 
room response of the speakers is shown in Figure 3. 
The system does indeed deliver on the low end of 
the spectrum. 
Those who have been striving to get lower bass 
out of smaller woofers will find these speakers will 
meet your requirements. If you want to build a 3” 
or 4” version, just reduce the tunnel height to the 
size of the driver diameter and use 1” × 4” or 1” 
× 6” lumber for the frame adjusting the overall 
dimension accordingly, keeping the tunnel length the 
same. Both 6.5” and 8” versions are also possible, 
but they do get to be rather large compared with 
the 5.25” size.
I think both Olney and Virva would be pleased 
with the revised versions of their Labyrinth and 
Tunnel Reflex/Bullhorn speakers. ax
Figure 3: Here is the in-room response of both channels. 
About the Author
Ken Bird has had a 40-year engineering and product management career focusing 
on consumer, industrial, and telecommunications electronics. Ken retired to 
his native Missouri and now pursues his interest in building speakers and tube 
amplifiers, along with model railroading, amateur radio (WOKLB), and digital 
photography.
For Over 35 Years
ACOustics begins with ACO™
ACO Pacific, Inc
www.acopacific.com sales@acopacific.com
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Custom
My goal with the Brioso(Italian for lively) was to 
create a speaker with really high definition by using 
well-regarded, quality drivers with a transmission-
line (TL) design for a bass response in the low-
30s. I’m fortunate that cost was not particularly a 
limitation for me, but to minimize the complexity 
and keep costs from getting completely out of hand, 
I chose a two-way design (see Photo 1). 
The Speakers
For the midwoofer, I originally planned to use a 
4-Ω, 7” Scan-Speak Illuminator and a Transducer 
Labs Beryllium tweeter. After getting opinions/
recommendations relative to cost/performance 
considerations from several fellow DIYers, especially 
Dan Neubecker, I decided to use the Scan-Speak 
Revelator 18W/8531G00 along with the Transducer 
Labs Alumina ceramic tweeter, N26CR2-A. What 
was “lost” with these choices was about 3 dB in 
sensitivity, but what was gained was a very amplifier-
friendly impedance and a pretty simple crossover, 
plus a reduction in driver costs of approximately 
$200 per cabinet, which ultimately paid for all of 
the crossover components.
The Design
The 18W/8531G00 is mounted in a single-fold, 
mass-loaded transmission line (ML-TL) containing 
about 1.75 ft3 of volume with a physical line length 
of 6’. The effective length of the line, however, is 
almost 6.5’ due to the first half of the line having a 
mild negative taper (decreasing cross-sectional area), 
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The Brioso uses a Scan-Speak Revelator 
midwoofer and a Transducer Labs Alumina 
ceramic dome tweeter in a floor-standing, 
mass-loaded transmission-line (ML-TL) 
enclosure. Paul won first place for this design 
in the over-$200 category in the 2017 Midwest 
Audiofest (MWAF) event held by Parts Express. 
By
Paul Kittinger
(United States)
Lively Two-Way Speakers 
with Transmission-Line Bass
The Brioso Speakers
Photo 1: Cabinet front view (left) and rear view (right)
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a result of the baffle being tilted back at 5°. The 
quarter-wavelength resonance of the line’s effective 
length tunes the enclosure to approximately 44 Hz. 
That is not low enough to tune the system to 28 Hz, 
as is appropriate for this midwoofer’s fs and Qts, 
and achieve an essentially flat system bass response. 
The mass-loading port, mounted on the rear panel, 
completes the system tuning—resulting in a pretty 
smooth overall bass response shape with a predicted 
anechoic f3 of 31 Hz and f10 of 23 Hz (see Figure 1). 
The center of the midwoofer is located 37% of 
the line’s length from the line’s beginning—a little 
further than 33%, one of the two optimum positions 
along a line, but close enough. The port’s center is 
located at one-sixth of the line’s length from the 
end of the line, resulting in the smoothest possible 
overall response shape, although f3 is a couple of 
Hertz higher than it would be if the port was closer 
to the end of the line. 
The baffle was tilted back as Dan recommended 
to aid in reducing diffraction anomalies (along with 
the three edges of the baffle being chamfered). The 
bottom 2” are separated from the rest of the cabinet 
by a solid horizontal divider, above which is the TL 
enclosure, thus creating a pocket at the bottom 
for the crossover assembly mounted on top of the 
base assembly (see Photo 2). With the cabinet on 
its base, the tweeter’s center is located just over 
37” above the floor, a good location relative to the 
typical ear height range of 35” to 38”. I used Martin 
King’s ML-TL Corner worksheet (Revised 8/14/09) to 
model the transmission line.
The Cabinet
Other than the baffle, the cabinet is made 
from 18-mm Baltic Birch Plywood (BBP) that I 
veneered with Jatoba (Brazilian cherry). The baffle 
is constructed from solid, 3/4” hardwood, red oak 
down the middle flanked on both sides and above 
the tweeter by Jatoba. A vertical divider located 
about halfway back in the cabinet’s depth creates 
the single-fold line, with the line starting in front of 
the divider at the bottom of the cabinet, going up 
to the top and making a U-turn, then going down 
the back of the divider to end at the bottom of 
the cabinet. Polyester fiber fills the entire front of 
the cabinet, which is the first half of the line, and 
the stuffing density is 0.75 lb/ft3. I used 1” thick 
Dacron batting sourced from Meniscus Audio, but 
loose polyester “pillow” stuffing performs the same. 
Loose stuffing, however, has to be weighed, then 
teased and fluffed during installation, whereas 
this Dacron batting has an inherent density of 
0.75 lb/ft3. It only needs to be cut to shape and layered
Photo 2: Mounted 
crossover assembly
Figure 1: Modeled 
anechoic bass response 
for an input of 2.83 V/1 m 
(red line)
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in without teasing or compression, and will stay 
where placed.
There are four horizontal “window” braces—
two located between the line divider and baffle, 
and the other two between the line divider and 
the back panel. The window braces are inset into 
1/4”-deep dadoes in each cabinet’s sides, as is the 
line divider, which is also inset into a similar dado 
in the top of the solid, horizontal divider above the 
2” cavity at the bottom of the cabinet. There are 
four additional braces, each 1.5” wide, spanning the 
width of the cabinet, with two located on the back 
and the other two located on the front to create 
additional attachment areas for the rear panel and 
baffle (see Figure 2).
The Jatoba veneer looked much like walnut in its 
unfinished state, but a single coat of polyurethane 
brought out the reddish tint hidden in its primarily 
brown color. The Jatoba hardwood for the baffle, 
however, wasn’t nearly as dark in raw form as 
was the veneer, but I found that a single coat 
of red mahogany oil stain, followed by a coat of 
polyurethane, created a pretty good match. In 
the end, I applied a total of five coats of satin 
polyurethane to the entire cabinet. The first two 
coats were applied with a foam brush for thickness, 
and the last three coats were wiped on, with light 
sanding performed after the brushed-on coats, 
and smoothing out the wiped-on coats with Norton 
Super Fine (0000) plastic “steel wool” pads. All of 
the cabinet, excluding the baffle, was assembled 
and veneered first. 
I created the baffle by edge-gluing together 
four pieces of hardwood, three of Jatoba and 
one of red oak. The baffle was built slightly over-
sized, then clamped to the front edges of the 
assembled, veneered cabinet and trimmed to an 
exact match with a router and flush-trimming bit. 
After removing the trimmed baffle, I chamfered 
the edges of its two sides and top, then cut driver 
through-holes and flange recesses. I made the 
base assembly from 1.5” square red oak on all 
four sides attached to the four edges of a piece 
of 18-mm BBP that form its top plate where the 
crossover assembly is attached.
The Crossover
Dan Neubecker designed the crossover, 
remarking that it was the easiest design he’s ever 
done, adding that he would describe its sound 
with one word—“smooth.” The corner frequency 
is approximately 2.1 kHz with an LR4 slope on the 
tweeter and a relaxed LR4 slope on the woofer. It 
has an amplifier-friendly impedance that drops to 
a “low” of 7.5 Ω between 100 Hz and 250 Hz, plus 
excellent phase matching and tracking. 
Depending on capacitor choices (more on 
this later) only six components are needed: two 
inductors and three capacitors with a single resistor 
for tweeter attenuation. The system’s midrange 
and high-frequency response is very flat up to 
approximately 11 kHz, and rolls off by about 5 dB 
at 20 kHz. I built the crossovers per the original 
design (5-Ω tweeter resistor) and listened to a 
variety of music over a couple of weeks. The sound 
"Window""Window""Window""Window"
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Poly fiber
3 oz.
Poly fiber
3 oz.
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3 oz.
Poly fiber
3 oz.
Poly fiber
3¾ oz.
Poly fiber
3¾ oz.
Poly fiber
3¾ oz.
Poly fiber
3¾ oz.
Wire pass-thruWire pass-thruWire pass-thruWire pass-thru
5˚ Tiltback5˚ Tiltback5˚ Tiltback5˚ Tiltback
2D × 3½L2D × 3½L2D × 3½L2D × 3½L
(Crossover compartment)(Crossover compartment)(Crossover compartment)(Crossover compartment)
"W
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w
"
"W
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"
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1½ × ¾ × 7½
(4)
1½ × ¾ × 7½
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1½ × ¾ × 7½
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Poly fiber
2½ oz.
Poly fiber
2½ oz.
Poly fiber
2½ oz.
Poly fiber
2½ oz.
2¾2¾2¾2¾
Window braces
and line divider
dadoed into
cabinet sides ¼"
Window braces
and line divider
dadoed into
cabinet sides ¼"
Window braces
and line divider
dadoed into
cabinet sides ¼"
Window braces
and line divider
dadoed into
cabinet sides ¼"
14¾14¾14¾14¾14¾14¾14¾14¾
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0000 0 (Ref)0 (Ref)0 (Ref)0 (Ref)
All dimensions in inches
internal width = 7½"
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internal width = 7½"
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internal width = 7½"
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(Solid)(Solid)(Solid)(Solid)
Figure 2: Cabinet side view drawing
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was a bit forward, so I increased the tweeter’s 
resistor to 7 Ω, which starts gradually rolling off 
the response just above 2 kHz, ending up about 
7 dB down at 20 kHz. 
Fine Tuning the Sound
Based on my description of the sound, Dan 
suggested incorporating a 1.5-dB dip in the response 
centered at 3 kHz, covering an octave on either side 
of that frequency, with the tweeter resistor at 5 Ω. 
That was the version attendees heard at InDIYana 
2017 and Dan commented that they didn’t sound 
as engaging to him there as when he listened to 
the single speaker at his home. Frankly, I thought 
the Briosos sounded better at InDIYana than in my 
home. After playing lots more music at home after 
InDIYana, I eventually determined that the left 
channel of my preamp was sometimes distorting 
the music, mostly noticeable on massed violins and 
some percussion instruments (where both are almost 
always located on the left side of a typical symphony 
orchestra). I switched to a backup preamp, which 
resolved this issue. Realizing that all of my listening 
perceptions had likely been negatively affected by 
Figure 3: Individual driver’s responses, plus system’s responses with normal and reversed polarity driver connections via crossover
Figure 4: Impedance magnitude and phase characteristics via crossover
Figure 5: System’s upper response versus tweeter’s attenuation resistor (R1) value of 5 Ω to 7 Ω
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audioxpress.com | September 2018 | 53 
the distortion in the left channel, I decided to revert 
to the crossover’s original design and tweak the 
tweeter’s attenuation resistor to suit my tastes, 
ending up at 7 Ω (see Figures 3–5).
The Capacitors
Regarding capacitors, I decided up front I would 
use premium but not obscenely expensive “boutique” 
capacitors (with questionable advertised or perceived 
attributes) in the crossovers, although I doubted my 
old ears would likely notice any differences. I chose 
Mundorf Supreme capacitors for the three needed, 
4.7 μf and 15 μf in the tweeter circuit and 12 μf for 
the woofer. Unfortunately, the Supreme offerings do 
not include a 12-μf value. So, I paralleled 3.9 μf with 
8.2 μf, giving me 12.1 μf and clearly close enough 
(see Figure 6 and Figure 7). In the end, total costs 
for capacitors, inductors and resistors came to $395 
for the pair ($322 in capacitors). Far less expensive 
polypropylene capacitors are obviously available and 
it would be up to anyone that might replicate this 
design to decide what to use. 
As a note, while my main audio room is the 
living room part of a “great” room and has some 
Figure 6: Crossover schematic
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favorable dimensions, angles, and ceiling heights, 
my speakers sit on opposite ends of a floor-level 
fireplace hearth diagonally in a corner, flanking 
the fireplace, with their backs only about 5” away 
from the wall around the fireplace. As always, I 
had Dan incorporate 3 to 3.5 dB of BSC since there 
is some bass reinforcement from the proximity to 
that wall. The final sensitivity (2.83 V/1 m) ends 
up at 84 dB to 85 dB, a bit on the low side but 
pretty typical, but is mostly offset by being a true 
8-Ω load and easy to drive. An actual input of 
20 W will create an output of 100 dB SPL at 1 m 
from 40 Hz on up with the midwoofer’s excursion 
not exceeding Xmax+15% above approximately 
23 Hz. That is actually quite loud, louder than 
necessary, and translates to about 90 dB SPL in 
my “sweet spot” recliner. 
I typically listen at an average SPL of 75 dB to 
80 dB at my recliner, which means the speakers are 
outputting 85 to 90 dB SPL on average at 1 m. This 
Scan-Speak midwoofer has a maximum mechanical 
excursion specification of 11 mm peak, so there 
is an ample amount of additional SPL available for 
momentary peaks.
Overall Verdict
I am quite pleased with the performance of 
my Brioso speakers. The T-Lab tweeters are very 
revealing and provide lots of detail. Recordings that 
have high sonic qualities will sound their very best, 
but if the recording quality is not good (or there 
are deficiencies in the audio equipment string), 
these tweeters won’t sugar coat the music at all. 
As expected, the bass performance from 
the Scan-Speak Revelators is excellent in depth, 
definition, and dynamics, plus the midrange sound 
is detailed and smooth. And, there’s so much 
detail to hear and enjoy. My capacitor choices 
may have had a positive effect on the outcome 
and the drivers certainly play a large part in the 
performance quality, but Dan’s crossover design 
deserves much of the credit. For me, every single 
penny was well spent. 
As a special note, Dan Neubecker has designed 
at least seven crossovers for speakers I’ve built—
three have taken first place and a couple have taken 
second place in their respective categories at the 
Midwest Audiofest (MWAF) since that event’s debut. 
Dan is not only very talented at crossover design, 
he also is quite inventive and adventurous with 
his own speaker builds, as well being an excellent 
woodworker and craftsman. More important, 
Dan is generous with his help and sage with his 
advice, as is generally typical in the DIY speaker 
community. ax
C3
15 µf
C2
4.7 µf
R1 − 7.0Ω
2.0
mH
C1a
8.2 µf
C1b
3.9 µf
L1
TWT
MW
+
−
L2
0.30
mH
−
+
IN
− +
Figure 7: Crossover assembly drawing showing component layout and wiring
About the Author
Paul Kittinger is a retired BSEE (since 2003) having worked for 40 years with all 
but the first couple in the medical device industry. Paul played trumpet and met 
his future wife in high school orchestra. She instilled an appreciation of symphonic 
music since she was proficient on violin, viola, and piano. Paul built a couple of 
speaker systems “way back when” with little knowledge, but became more serious 
and started gaining knowledge and experience in the late-1990s, with articles 
about four of his designs published in Speaker Builder magazine from 1997 to 
2000. Paul prefers and has focused on transmission line (TL) designs, modeled with 
the accurate software created by Martin King. Not being particularly inclined to 
design crossovers, he solicited help from others in exchange for TL design help. He 
loves to design TL cabinets and has done so many times for other DIYers literally 
around the world, as well as for several commercial entities (Philharmonic Audio, 
Salk Sound, Ars Harmonia and Rhythm Audio Design). Paul also thoroughly enjoysthe woodworking aspects that go along with his hobby that keeps him happily 
occupied in retirement. He took a pair of speakers to its first DIY speaker event in 
Sacramento, CA in 2003, has attended the DIY event in Grinnell, IA, several times, 
and still regularly attends the DIY speaker events of InDIYana and Parts Express’ 
Midwest Audiofest.
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Questions & Answers
56 | September 2018 | audioxpress.com
ax
SHANNON BECKER: Tell us a little about your 
background.
CRISPIN HERROD-TAYLOR: I grew up on a farm in 
Wiltshire, a rural county in the South of England. As 
with most rural places, there wasn’t a lot to do as a 
teen, so we had to create our own entertainment. 
This inevitably meant doing something with music—
everybody I knew was in a band, or DJ’d. I couldn’t 
play an instrument (believe me, I tried!), but I was 
technical, so started to make guitar amps and 
backline gear. I also really got into DIY hi-fi partly 
through lack of budget, and partly to learn, and this 
started an interest in music reproduction, which has 
kept me employed for some 30 years. After school, 
I went onto university in the UK and ended up with 
a BSc in Physics, which was quite an achievement 
considering I spent most of the time in the Students 
Union, working with bands!
My Physics degree wasn’t immediately applicable 
to the audio industry, but it did teach me to think 
out of the box, and look to first principles to work 
out what’s going on, rather than take for granted 
the status quo. This has served me very well, and 
everything we do is slightly different from the norm.
SHANNON: How did you become interested in audio?
CRISPIN: I’m one of six children, and all my older 
siblings and my Dad had the most wonderful record 
collections. I remember listening to a huge range of 
stuff from 50’s Jazz and Blues, through the Beatles 
By
Shannon Becker
(United States)
Pro Audio Company 
Designs 
SoundBuckets 
Portable Speakers
An Interview with Crispin 
Herrod-Taylor, Co-Founder 
of Crookwood Crispin Herrod-Taylor is the designer and co-founder of 
Crookwood and initiated the SoundBuckets concept.
Crookwood (www.crookwood.com) is a high-end pro audio company that has been 
making some of the best specialized pro audio gear on the planet for the last 20 plus 
years. It is also the company behind SoundBuckets. The SoundBuckets design came about 
when Crispin Herrod-Taylor saw a need for a professional small speaker that would let 
professionals check the sound of their work on a small speaker, typical of the new breed of 
consumer products. And, he wanted a small unit for himself. The result is the SoundBucket, 
a Bluetooth rechargeable speaker that’s portable and easy to use.
audioxpress.com | September 2018 | 57 
and Stones to 1970’s experimental stuff. I loved 
the records, the covers, and where the music took 
you. There was no way I couldn’t not be interested 
in audio!
I also grew up in a time when electronics started 
to become affordable, and also started to appear 
in music. I can still remember being mesmerized 
when I first heard Tubeway Army’s “Are Friends 
Electric?” This coincided with a great period in the 
UK, where music was evolving almost daily. About 
this time, I convinced my school to help me purchase 
the bits to make an analog synthesizer, based on 
an Elektor magazine design. Electronics 101 at 16, 
heavy learning curve or what? The synthesizer was 
for the school to use, but a lot of friends’ bands also 
seemed to end up using it…
Perhaps because of my upbringing, I have a very 
eclectic musical interest. I used to say that I liked 
everything except county music, but then I heard 
a Bonnie Raitt album, so I can’t even say that now!
SHANNON: What made you decide to start 
Crookwood? 
CRISPIN: After university, I worked briefly for 
Meridian, the hi-fi company, then for SSL, and finally 
Focusrite, where I worked on the last great British 
console, the Focusrite Studio Console. During this 
period, I saw the writing on the wall for large studio 
production, and decided to start Crookwood, moving 
audio design in a different direction.
It became clear to me that the industry was 
about to be disrupted, and the large studios (and 
manufacturers) would mostly die out, as it became 
ever cheaper to make music. I saw that recording 
quality was going to suffer, as the professionalism 
was eroded, but at the same time there was space 
for a niche manufacturer making quality items for 
the more purist areas of music production.
So we started Crookwood in 1992, and our first 
product the Paintpot preamp was released in 1993. It 
looked strangely like our SoundBucket speakers (more 
below). I must have a thing for leather handles! The 
great thing is, even today, 25 years later the Paintpot 
is reckoned to be one the great preamps of all time, 
and we’ve just released an update of the original 
with the same great neutral sound.
SHANNON: Can you tell us more about Crookwood 
and where the name comes from?
CRISPIN: When you start a company, you initially 
come up with the most awful names! After a few 
rounds of this, I settled on Crookwood, the name of 
the farm I grew up on. It resonates with me, and 
the name is mentioned in the Doomsday book of 
1086, making it one of the oldest names in audio. 
We’ve been in business now for 25 years, and 
we’re best known for our accurate uncolored sound, 
initially with mic preamps, and latterly in our modular 
mastering consoles and routers. We now make the 
widest range of monitor controllers, audio routers, 
and mastering consoles on the planet, and have sold 
to some of the greatest names in the industry.
SHANNON: To what do you attribute Crookwood’s 
continuing success?
Crookwood, a small British niche manufacturer, has quietly been hand making the world’s 
best studio gear, for the world’s best recording engineers. Now, for the first time, the 
company has condensed decades of professional audio knowledge into a compact portable 
speaker, so people can enjoy studio quality sound, wherever they are.
The Paintpot was 
Crookwood’s first product, 
so called because it 
resembles a big can of 
paint. Now in it’s third 
incarnation, this neutral 
two-channel remote 
controllable microphone 
preamp is still widely 
regarded as the most 
neutral mic amp available. 
Questions & Answers
58 | September 2018 | audioxpress.com
ax
CRISPIN: We have established a reputation for very 
accurate sound. Most pro-audio gear has a specific 
sound, and you choose the gear in a recording chain 
to create the overall sound you want. However for 
certain areas, mic preamps and monitoring, you 
need the sound to be as true as possible. These 
are the areas we excel in. The reasoning is simple: 
You are manipulating the sound with your choices 
of optional gear, but the fixed parts of the studio ( 
generally the monitoring) needs to be as accurate 
as possible, or you’ll make false choices and creative 
decisions. Mic amps work similarly, the star should 
be the mic, not the amplifier.
Another unique quality to our products is their 
modular nature. We hate obsolescence and the idea 
that you continuously have to replace gear to be 
current. Using our modular designs, you can grow 
and update your Crookwood products as your needs 
and objectives change. We can look after, repair and 
upgrade nearly everything we have made, every 25 
years on. I like that.
SHANNON: Why design SoundBuckets?
CRISPIN: SoundBuckets are our crossoverprosumer 
portable speaker range. The original SoundBucket 
is a small battery-powered wireless speaker that 
with its leather handle looks like a Bucket! We’ve 
used real speaker drivers and analog circuitry to 
get a proper speaker into a portable package. Most 
other vendors spend the money on impressing you 
with looks and marketing, leaving not much money 
for the actual speaker. To correct this, they try and 
use (inexpensive) DSP to cover up the problems, and 
generally think the consumer doesn’t care anyway.
We’ve taken a different route, focusing on buying 
real speaker drivers with real analog circuitry to make 
a very competent speaker for people who love and 
want to listen to music, rather than show off their 
fashion creds.
SHANNON: Where did the concept originate?
CRISPIN: So I’ve been involved in designing speakers 
for most of my life, although I haven’t done anything 
commercially for a while. Then a couple of years 
ago, my aging hi-fi system eventually broke, and 
I started to look at replacements. A static hi-fi 
system just wasn’t how I listened to music anymore 
(I just don’t have the time, or space to sit in one 
place and focus), so I started to look into multiroom 
setups. However, I soon discovered that they were 
all proprietary—I don’t like being tied to one vendor, 
and also physically, I am only in one place at a time! 
I realized I needed a wireless portable music 
system, and started to look for one. My findings 
were depressing, everything portable was basically a 
fashion toy, pretty, but sonically rubbish. They were 
designed to impress, not play music.
So we designed our own portable speaker, the 
SoundBucket! Designing a speaker is probably the 
audio engineer’s ultimate dream job. It’s impossible 
to get it right, at every turn you have to make a 
compromise, and the job becomes endless. R&D 
heaven!
I was initially worried about Bluetooth as a delivery 
mechanism, but it became apparent (especially with 
aptX encoding) it wasn’t really the weak link—the 
speaker was. A .wav file played back with aptX is 
really very good, certainly for an everyday listen. By 
using an analog path in the SoundBucket, we could 
also bypass any encoder and go analog from input to 
speaker without any DSP messing up the sound, so 
you could truly appreciate the quality of the speaker.
SHANNON: Is SoundBuckets the first product you 
have promoted through Kickstarter?
CRISPIN: Yes, it was an interesting journey, not 
for the faint hearted though. For normal B2B 
products, I’m not sure it’s right, but B2C selling 
is a very different beast. We looked at the various 
crowdfunding sites and Kickstarter shone as the 
more professional. It’s sad that some of the gloss 
Using the latest Bluetooth technology and old school analog design, you can stream 
quality stereo music, and remote control it from your phone, tablet or PC.
audioxpress.com | September 2018 | 59 
has come off it after recent high profile project 
failures, but I still think they do a good job.
SHANNON: Why did you promote it via Kickstarter?
CRISPIN: Although we had an excellent pro audio 
name, we’re not known in the consumer world. 
Trying to get traction there is very expensive. 
Kickstarter enabled us to engage with likeminded 
people, learn about the market, and provide cash flow 
for production. It’s also a very exacting process, by 
the time you’ve finished you have a very good idea 
of what you are trying to bring to market, and why.
SHANNON: How did the Kickstarter process work 
for you?
CRISPIN: Good, we achieved our goals, and 
considering only 1 in 5 projects succeed in the 
Tech section of Kickstarter, we were very pleased. 
In retrospect however, we were naïve in thinking 
that “if you build it, they will come.” We did a 
lot of marketing (like 24/7 for 30 days), but we 
couldn’t break through to a larger audience that 
some projects did. Looking at some other projects, 
they clearly started a long time prior to going to 
Kickstarter to engage with reviewers, and create 
a hype before launching, and the most successful 
ones paid a lot for this engagement. The process 
did, however, make you really focus on what the 
product was about, and what your USP was.
SHANNON: What were some of the challenges?
CRISPIN: There are a lot of things that you cannot 
control. We worked very hard, but we had some 
unforeseen issues with component sourcing and 
keeping up with replies to inquiries. We also ran out 
of energy after the third week, and needed more 
coffee than we could physically digest! 
A four-week project doesn’t sound very long, but 
it becomes your life, and absorbs you completely. 
And that’s before you even start to make the prod-
uct! If I could change anything, I would have mar-
keted the SoundBuckets for longer, building up a 
stronger fan base, before coming to Kickstarter.
SHANNON: How long did it take from concept to 
shipping to customers?
CRISPIN: We probably started prototyping and 
developing 18 months prior to going to Kickstarter. 
After the successful launch, we took another five 
months to start shipping, a bit longer than planned, 
but our backers were brilliant, and still are. 
We purposely didn’t require any special tool-
ing, which I think delays a lot of products, and by 
making everything in the UK, using our Crookwood 
staff, we avoided the low-cost China nightmare that 
plagues the optimistic manufacturer.
SHANNON: Are you working on any new audio 
innovations? If so, can you tell us about them? 
Would you promote them via Kickstarter again?
CRISPIN: It sounds trite, but we’re always playing 
with things. We’ve got a very interesting Crookwood 
product just round the corner, that’s under wraps, 
but we’re close to releasing a big brother to the 
SoundBucket for the person who likes their music 
quite loud. We would promote it through Kickstarter, 
but now with our existing products, we’d run it as 
an early bird-type promotion, to raise awareness, 
rather than look for sales revenue per se.
SHANNON: Do you have any advice for audioXpress
readers who are interested in launching their own 
audio products?
CRISPIN: Don’t do it? Seriously, launching any 
product is very revealing and testing of your resolve 
and character. Be prepared to be stressed! It’s also 
rare to be able to convince other people, customers, 
or vendors that you have something special—the 
audio market is saturated and full of sales noise. 
At the end of the day you have to believe in your 
product yourself, but if it is to be commercial, you 
must ensure it has value to customers. It’s too easy 
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Hollow-State Electronics
60 | September 2018 | audioxpress.com
ax
The power amplifier we will be discussing in this 
article is a Bogen APH-1050, which was designed 
as a power amplifier for public address systems. 
It was advertised as having an output power of 
50 W, and used a pair of 7581 beam power tubes. As 
you can see from the schematic shown in Figure 1, 
the B+ for the output tubes is provided by a full-
wave voltage doubler, and there is a separate full-
wave voltage doubler providing B+ for the preamp 
and phase splitter. Having no use for a 50 W hollow-
state PA power amplifier, I have repurposed this 
unit as a guitar power amplifier, replacing the 7581s 
with 6L6GCs. Otherwise, the amplifier is in original 
condition.
Potential Impact
The concern with power-supply stiffness affecting 
amplifier behavior has to do with possible decreases 
in voltage gain in high-output-power operation, 
and lower clipping-limited maximum output power 
than would be expected if voltage regulation were 
perfect. The 1975 RCA Receiving Tube Manual 
specifies a maximum signal-power output of 55 W 
for two 6L6GCs in a Class AB2 configuration. In my 
experience,a push-pull pair of 6L6GCs will not really 
deliver 55 W, or even the 50 W that Bogen claimed, 
before clipping.
The Experiment
I tested the amplifier by feeding it a 500 Hz sine 
wave, varying the RMS input and output voltages 
and the high B+ voltage, while monitoring the output 
on an oscilloscope. I did not look at the preamp B+ 
voltage, since the preamp and the phase splitter are 
operated in Class A, so their current does not change 
as the output level changes. 
Figure 2 shows the variation of the high B+ 
voltage with output power. The blue line shows the 
performance of the original circuit, and the orange 
line shows the performance after I paralleled the 
voltage doubler’s two 100 µF/250 V capacitors with 
additional 80 µF/250 V units. The voltage regulation 
with a varying load for the original circuit is given by:
load regulation =
V V
V
out no load out load
out rated
, ,max
,
%
− − ×100 ==
− =439 418
440
4 77. %
I have somewhat arbitrarily assumed that the 
design-center B+ voltage was 440 V.
For the circuit with the added capacitance, the 
load regulation is:
440
440
1 7
432 5− =. . %
In the June 2018 issue of audioXpress, we discussed the 
performance of power amplifiers for hollow-state amplifiers. Now 
we’ll look at the effects of supply stiffness on how the amplifier 
performs. Stiffness is an informal term describing the load 
voltage regulation of a power supply. 
By
Richard Honeycutt 
(United States)
The Measured Effect of 
Power-Supply Stiffness on 
Power Amplifier Performance
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Hollow-State Electronics
62 | September 2018 | audioxpress.com
ax
This is pretty good voltage regulation—not 
surprising for a commercial-grade power amplifier. 
Many consumer amplifier power supplies would not 
perform as well. More important for our interests 
is the effect that a varying B+ voltage has on the 
voltage gain, distortion, and maximum output power. 
As shown in Figure 3, the voltage gain decreases 
from 23.3 at 1 W output to 20.7 at the clipping point, 
which was at 37.5W for the original circuit—about a 
1 dB drop. This is a barely noticeable change.
Figures 4-6 show the output waves for the 
original circuit at output power levels of 1 W, 
10 W, and 37.5 W. (The vertical scale of the 
oscilloscope was not the same for these three 
waveforms.) Notice the virtual absence of distortion 
all the way up to the threshold of clipping. This 
is due to the combination of an inherently linear 
preamp, phase splitter and output circuits; and 
the overall negative feedback from the 16 Ω tap 
of the output transformer to the cathode of the 
preamp stage.
The waveforms for this range of output power 
were essentially identical after the insertion of 
additional filter capacitance to stiffen the supply. 
However, with the stiffer supply, it was possible to 
get a full 40 W output before the onset of clipping 
(see Figure 7).
Lessons Learned
What, then, can we learn from this little exercise? 
First, we see that supply stiffness does matter. 
Although dropping B+ voltage only cost about a 
decibel in gain, and 0.78 dB in maximum power 
output for this circuit, we did see that when the 
B+ drops at high output power, gain and maximum 
output power are affected. If our test amplifier had 
been less well designed and built, the effect would 
have been greater. For example, if a hollow-state 
Figure 1: Schematic of the 
Bogen APH-1050 Power 
Amplifier
Figure 2: The amplifier’s voltage regulation
audioxpress.com | September 2018 | 63 
Figure 3: Variation in voltage gain vs. output power.
rectifier had been used, and the filter capacitors for the voltage 
doubler had been half as large, the voltage regulation percentage 
would have been over twice as high.
In an amplifier using a capacitor-input filter, voltage regulation 
and power supply ripple are both affected by the size of the filter 
capacitor. This fact is obvious from the equations for load voltage 
and ripple:
V V
I
fCload peak
load
DC
DC≅ −
2
where f is the charging frequency for the capacitor,
and C is the capacitor value in farads.
γ% %≅ ×1
4 3
100
fR Cload
where γ% is the percent ripple, Rload is the filter’s output voltage 
divided by the load current, and f and C are as defined above.
(The latter equation is for a simple capacitor-input LC filter; a Pi 
or other multi-section filter will produce less ripple, as predicted by 
other equations.) So a stiff power supply will also have less ripple or 
hum. The difference this makes in performance depends upon the 
amplifier design: An amp having a push-pull output stage is not very 
sensitive to ripple in the high B+ supply, because (a) the push-pull 
stage cancels supply ripple, preventing it from being heard in the 
output; and (b) the screen-grid supply of the output stage usually 
has additional filtering that prevents supply ripple from affecting 
the output signal via the screen grids. If the amp has a separate 
low B+ supply, like the Bogen we’ve been examining, that supply’s 
ripple will not be affected by the amount of current drawn from 
the high B+ supply. 
On the other hand, an amplifier using a single-ended output 
stage will produce more hum as the output power is increased 
enough to drive the tube out of Class-A operation, where the 
B+ current is no longer constant. For example, let’s look at the 
Silvertone Model 1471 whose schematic is shown in Figure 8. 
Hollow-State Electronics
64 | September 2018 | audioxpress.com
ax
The high B+ supply is 260 V, and it is filtered by a three-stage 
capacitor-input RC filter. Just looking at the high B+ supply, 
we find that the current drawn from the B+ supply consists of 
0.92 mA from the 143 V supply, plus 4.4 mA from the 235 V supply, 
plus 47 mA from the high B+ (according to the spec sheet for a 
6V6GT operating at 250 V B+ in Class A): a total of 52.32mA. The 
drop in high B+ when loaded is calculated thus:
∆V
I
fC
VloadDC= =
× × ×
=−2
0 05232
2 120 20 10
10 9
6
.
.
This corresponds to a load regulation of:
10 9
260
4 2
.
. %=
as long as we stay in Class A operation. Under these conditions, 
the ripple percentage is:
1
4 3 120 20 10
100 1 2
6× × ×
× =
−
% . %
(The 4969 Ω value for Rload was calculated using Ohm’s Law 
with 260 V at 52.32 mA.)
I f we ove rdr i ve t he 6V6GT so t ha t i t r eache s 
i t s C l a s s-B max imum-s igna l p l a t e cur ren t (abou t 
Figure 6: Output wave of original circuit at 37.5 W (just below clipping) Figure 7: Output wave of circuit with added filter capacitance at 40 W 
(just below clipping)
Figure 5: Output wave of original circuit at 10 W powerFigure 4: Output wave of original circuit at 1 W power
audioxpress.com | September 2018 | 65 
Figure 8: Schematic of 
Silvertone Model 1471 
Guitar Amplifier 
70 mA), the load regulation will increase (worsen) to 
6%, and the ripple will increase to 48%—beyond the 
limits of accuracy for our % ripple equation.
Does this matter? Of course, overdriving any 
amplifier this severely is only done to achieve a 
certain sound, so if the user considers that sound 
desirable, the actual supply stiffness and ripple 
percentage are not important. In actual use, I doubt 
that the amplifier would ever be overdrivento this 
extreme.
Ripple Limit
All this brings up the question of how much ripple 
is objectionable. Traditionally, amp manufacturers 
have specified hum and noise together as a 
percentage of maximum output. From one 
perspective, this makes sense, given that the worst 
hum level would be measured at maximum output if 
the output stage is Class B. But it is also true that hum 
is more objectionable when the amp is idling, with 
no signal to mask the hum and noise. So ideally, the 
hum would be specified as a signal power level during 
idling. That way, a music amplifier driving a speaker 
whose sensitivity is 88 dB at 1 W at 1 m (for example) 
would be known to produce a certain maximum hum 
level that could be calculated in dBSPL. 
For the 6V6GT Class A circuit, maximum output 
power from a 250 V supply is about 4.5 W, so 1.2% 
ripple would amount to a hum output of 0.054 W. 
This is about 13 dB below the maximum sound output 
of the amp when it is operating in Class A. So the 
hum level of this amp when idling would be about 
88 -13 = 75 dBSPL. 
In a lounge, this level of hum would probably be 
masked by ambient noise, but in a quiet living room, 
it could easily be bothersome. Based on personal 
experience, I would say that historically, the freedom 
from hum was at least as big a reason as increased 
power output for many guitarists to go for a more 
expensive (Class-B-output-stage) amp.
As for the effect of supply stiffness on audible 
output, it is probably negligible, although some 
guitarists mention noticing a difference in “feel” 
between an amp with a solid-state rectifier (stiffer 
supply) and one using a hollow-state rectifier. 
This difference could well result from the slight 
compression after an initial signal peak caused by 
drop in high B+. ax
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Industry Calendar
66 | September 2018 | audioxpress.com
ax
Here are a few places where you might find a copy of audioXpress and possibly meet one of our authors and staff members.
August 31-September 5, 2018
IFA Berlin (The International 
Funkausstellung)
The Berlin Exhibition Center, 
Berlin, Germany
www.ifa-berlin.com
IFA Berlin (The International Funkausstellung) is probably the most 
important annual event for the consumer electronics industry, introducing 
new products in time for the most important sales season. If the CES in Las 
Vegas, N.V., is the place where future trends and innovation are highlighted 
and get the most attention, the IFA Berlin show is where many of those 
products and brands make it or break it. Audio at IFA is mainly focused 
on personal/mobile devices and the more accessible and lifestyle-oriented 
products. A wide range of products, solutions, and presentations make a 
visit to IFA worthwhile. IFA 2017 featured 1,805 exhibitors and attracted 
253,000 visitors, with more than half of all trade visitors coming from 
abroad. The 159,000 m2 of fully booked show floor in 2017 was already 
the largest in the show’s history. Expect the 2018 edition to be even larger!
September 4–8, 2018
CEDIA 2018
San Diego Convention Center, San Diego, CA
www.cediaexpo.com
CEDIA, also known as the Custom 
Elec tronic Design & Instal lat ion 
Association, is the primary touch point and international trade association 
for more than 3,500-member companies who design, manufacture, and 
install residential electronic systems and technology. 
CEDIA Expo is now operated by Emerald Expositions, a leading operator 
of business-to-business trade shows in the United States, currently 
operating more than 55 trade shows, as well as numerous other face-
to-face events. The 2018 edition of the CEDIA show promises to take the 
association and the industry to a new expansion era, increasing the focus 
toward the connected home and connected systems in general. 
The CEDIA Expo, held annually in the US, is attended by thousands 
of home technology professionals and more than 500 exhibitors. Major 
exhibiting brands have already committed for the 2018 show, and in 
addition to the already-popular CEDIA Talks, the new Smart Stage will be 
introduced to host a number of new attendee engagement opportunities. 
Tech Bites will also launch in 2018 and will give exhibitors the chance to 
apply for a 3-minute slot to present a new product and/or service in a 
short elevator pitch.
September 12-14, 2018
MWCA18 Mobile World Congress 
Americas
Los Angeles Convention Center (LACC) 
Los Angeles, CA
www.mwcamericas.com
www.gsma.com | www.ctia.org
On September 12-14, the mobile world 
will connect in Los Angeles, CA for the second annual MWC Americas, 
which debuted last year in San Francisco, CA. An alternative event for 
those following the mobile industry and don’t want to or can’t travel to 
Barcelona in February, MWCA is certainly a timely and convenient event 
for the many professionals focusing on the US market and most of the 
Calfornia-based technology companies. This a unique opportunity to learn 
more about the future of mobile—and in particular the 5G rollout in the 
North America—as top companies give an exclusive first look at the latest 
innovations. Like the annual Barcelona gigantic event, MWC Americas allows 
visitors to learn a lot about the latest consumer electronics trends in the 
burgeoning voice recognition and wireless connectivity market perspective, 
meeting with most of the engineering teams from semiconductor and 
development platform vendors. 
September 13–18, 2018
IBC 2018
RAI Amsterdam, The Netherlands
Exhibition: 14-18
www.ibc.org
Happening every year, the IBC 
Exhibition covers 15 halls across the RAI Amesterdam and hosts more 
than 1,700 exhibitors spanning the media, entertainment, and technology 
industry. The most international event for the electronic media and 
entertainment industry—in parallel with the National Association of 
Broadcasters (NAB) convention held in Las Vegas, NV, every April—IBC 
continues to get bigger and better every year as the show continues 
to expand with content production for new platforms on the rise, the 
exciting transition to 4K/8K images, the technology migration toward IP, 
networking and the cloud, and the media landscape becoming increasingly 
Internet-based and on-demand. Most of the Audio exhibits are located in 
Hall 8, and include a unique combination of radio technology suppliers, 
microphone manufacturers, recording and production solutions. 
The 2018 edition is adding specially curated feature areas and events, 
with a special area for startups and leading R&D labs, theaters hosting 
demonstrations, presentations and briefings; and even blockbusters on 
show at the custom-built IBC Big Screen—an opportunity not to miss for 
the latest and greatest in digital cinema!
September 13–15, 2018
Prolight+Sound NAMM Russia
Sokolniki Exhibition and Convention 
Centre,
Moscow, Russia
www.namm-musikmesse.ru 
www.prolight-namm.ru
The Musikmesse and Prolight+Sound NAMM Russia, the international 
trade fair of technologies and services for entertainment, integrated 
systems and creation, will once again return to the ECC Sokolniki in Moscow, 
Russia. As the leading professional event in Russia and CIS region, the 
annual show welcomes exhibitors, producers, and suppliers in the sphere 
of professional lighting and sound equipment and technologies. 
September 16–18, 2018
PLASA Show London
London, Olympia, United Kingdom
www.plasashow.com
The world-renowned PLASA Show 
is the essentialevent for professionals 
working in the entertainment technology 
industry, presenting innovative solutions and fresh product launches by 
the world’s most recognizable brands in lighting, audio, rigging, staging, 
and AV. Still buzzing from 2017’s 40th anniversary, PLASA Show is taking 
it up a notch with even more iconic speakers, immersive workshops, and 
inspiring seminars. Audio heavyweights L-Acoustics, Peavey and Adamson, 
and lighting pioneers DTS, ETC and Philips are among the 200-plus brands 
exhibiting at the show. 
September 23-25, 2018
Prolight+Sound Middle East
Dubai International Convention and
Exhibition Centre, Dubai
www.messefrankfurt.com
www.prolightsoundme.com
Now in its third edition, Prolight + Sound Middle East is shaping up 
to be even bigger and better. With more buzz and more visitors than 
the launch, the second edition of Prolight + Sound Middle East featured 
28 exhibitors from nine countries representing almost 80 brands, while 
2,156 trade visitors from 61 countries attended the show.
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