INNOVATIONS IN AUDIO • AUDIO ELECTRONICS • THE BEST IN DIY AUDIO

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ADVANCING THE EVOLUTION OF AUDIO TECHNOLOGY

AUDIOXPRE SS | FEBRUARY 2014

Standards Review
Audinate Dante
Making Digital Audio Networking Easy
You Can DIY!
Vacuum Tube Oscillator
By Atto Rinaldo

Audio Electronics
A High-Voltage Delay
for Tube Amplifiers
By Jan Didden

You Can DIY! Acoustics

The Ultimate Living Room Home Theater
By Ethan Winer

Sound Control
Acoustical Diffusion and Scattering

Audio Praxis
The Brave New World
of Loudness Control
FEBRUARY 2014 By Jon Schorah
The Authority on Hi-Fi DIY

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 Contents
Features
12 Solid State Logic’s Sigma
Innovative Mix Engine
By Miguel Marques
This review details the features
of Solid State Logic’s stand-
alone summing mixer, a digitally
controlled device that can provide
automation to any console.
14 High-Definition Digital
By Gary Galo
“Move Over PCM and Make Room
for HiRes DSD and DXD Audio,”
a presentation at the 135th
Audio Engineering Society (AES)
Convention.
24 The Brave New World of
Loudness Control (Part 1)
New Post-Production Workflows
Transcend Compliance for Enhanced
Audio Creativity
By Jon Schorah
Reintroduce creative freedoms
4 | February 2014 | audioxpress.com
lost in the old paradigm of peak
normalization.
44 The Ultimate Living Room
Home Theater
By Ethan Winer
Good acoustics affect the sound
quality more than the choice of
receiver or loudspeaker.
48 Vacuum Tube
Low-Frequency Oscillator
By Atto Rinaldo
Design and build a vacuum
tube oscillator utilizing current
components.
54 Tube Amplifier
High-Voltage Delay
By Jan Didden
Conserve tube life by adding this
high-voltage delay to your tube
amplifier.
 Volume 45 – No. 2 February 2014

Columns
Departments
STANDARDS REVIEW
18 Audinate Dante (Part 1) QUESTIONS & ANSWERS
Making Digital Audio Networking 40 The Original 6 From the Editor’s Desk
Easy Sound Designer
7 Client Index
By João Martins An Interview with Dan Dugan—
Audio Engineer, Inventor, and
8 What’s News
SPEAKERS Nature Sounds Recordist
32 Super Subwoofers By Shannon Becker
64 Member Profile
A Battle of the Titans
By Mike Klasco and Steve Tatarunis HOLLOW-STATE ELECTRONICS 66 Industry Calendar
58 The Tone Character of
SOUND CONTROL Tube Guitar Amplifiers
36 Acoustical Diffusion and By Richard Honeycutt
Scattering Websites
By Richard Honeycutt

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audioxpress.com | February 2014 | 5

ax

February 2014 ISSN 1548-0628 One More Take

www.audioxpress.com Remember that joke? The producer in the recording studio
says to the band: “Not bad, fellas. Let’s do one more take, this
audioxpress (US ISSN 1548-0628) is published monthly,
time with more emphasis on tone, harmony, melody, rhythm,
at $50 per year for the US, at $65 per year for Canada,
composition, lyrics, musicianship, tempo, and originality.”
and at $75 per year Foreign/ROW, by Segment LLC, Hugo
Maybe it’s time for the audio industry to try “one more take.”
Van haecke, publisher, at 111 Founders Plaza, Suite 300,
During last year’s 135th Audio Engineering Society (AES)
East Hartford, CT 06108, US Periodical Postage paid at
Convention in New York, it was apparent that the audio engineer-
East Hartford, CT, and additional offices.
ing community unites several generations. Also, the younger but
much more technically perceptive generation is fascinated by the
achievements of those who had the “privilege” of working in the big
studios and doing audio production for live concerts, or during great
Head Office: broadcast moments from the 1960s, the 1970s, and the 1980s.
Segment LLC The younger generations have learned to value the tools and what they can do with them. They
111 Founders Plaza, Suite 300 even value the “good old analog” electronics, essentially by using plug-in emulations of the real
East Hartford, CT 06108, US things inside Pro Tools or Logic. Yet, this generation also encodes studio recordings to MP3s.
Phone: 860-289-0800 From one content format to another, the music industry continually re-released its content in
Fax: 860-461-0450 physical media until the Super Audio CD (SACD) and the Blu-ray disc (on video) formats appeared.
And that was it. Suddenly, the Internet, mobile devices, and digital files changed everything. With
that change came the MP3, the iPod, iTunes, and mobile networks. This accelerated the demise of
Subscription Management: physical media, on which the entire music industry had become over-dependent.
audioxpress Meanwhile, technology continues to evolve. Even though SACD is dead and gone, the key
P.O. Box 462256
developments remain valid and high-resolution audio is still a logical proposition. But is it well
Escondido, CA 92046, US
Phone: 800-269-6301 understood by the “plug-in” generation? A very faint sign of hope emits from the enthusiasm
E-mail: audioxpress@pcspublink.com detected at events such as the AES conventions and the NAMM shows.
Internet: www.audioxpress.com With new 64-bit processors and OSes becoming the norm, large bandwith networks available
everywhere, and memory and storage increasing faster than consumers’ actual needs, it seems
the industry is ripe for another go at quality.
Postmaster: send address changes to: As our contributing author Gary Galo noted in his impressions of the 135th AES Conven-
audioxpress tion, it seems consumers are rediscovering the virtues of high-resolution sound and finding
P.O. Box 462256 compressed formats such as MP3 unacceptable. But at the same time, mobile platforms and
Escondido, CA 92046, US
wireless networks have created new consumer behaviors. People are increasingly listening to
music via headphones, soundbars, and portable wireless loudspeakers. Therefore, we need a
US Advertising: new approach to address that changing landscape, and it’s not going to be with $20,000 home
Strategic Media Marketing, Inc. stereo (or multichannel) systems.
2 Main Street If downloading high-resolution audio files is practical and inspires a new group of record
Gloucester, MA 01930, US companies to reinvest in high-quality content production, it is clear that 1-bit DSD record-
Phone: 978-281-7708
Fax: 978-281-7706
ings could also breathe new life into studios, the pro audio industry in general, and even
E-mail: audioxpress@smmarketing.us many high-end audio brands.
And it is at forums such as the Winter NAMM Show in Anaheim, CA—where those same
Advertising rates and terms available on request.
generations again meet with producers and musicians—that the conscience needs to be raised.
Not at the Venetian Hotel demo rooms in Las Vegas, NV. The signs are still fragile, the economic
environment remains unstable, and the market trends are uncertain, but it all seems to be
Editorial Inquiries: aligning for a “new take” in the audio industry.
Send editorial correspondence and
manuscripts to:
audioxpress, Editorial Department
João Martins
111 Founders Plaza, Suite 300 Editor-in-Chief
East Hartford, CT 06108, US
The Team
Publisher: Carlo van Nistelrooy President: Hugo Van haecke
Legal Notice:
Each design published in audioxpress is the intellectual Associate Publisher: Shannon Barraclough Art Director: KC Prescott
property of its author and is offered to readers for their
personal use only.
Editor-in-Chief: João Martins Customer Service: Debbie Lavoie
Any commercial use of such ideas or designs without prior Associate Editor: Shannon Becker Advertising Coordinator: Kiki Donaldson
written permission is an infringement of the copyright
protection of the work of each author.
Technical Editors: Jan Didden, David J. Weinberg

© Segment LLC 2014 Regular Contributors: Bill Christie, Dennis Colin, Joseph D’Appolito, Vance Dickason, Jan
Printed in the US Didden, Gary Galo, Chuck Hansen, Richard Honeycutt, Charlie
Hughes, Mike Klasco, G. R. Koonce, Ward Maas, Miguel Marques,
Nelson Pass, Bill Reeve, Steve Tatarunis, David J. Weinberg

6 | February 2014 | audioxpress.com

 OUR NETWORK

United States
Carlo van Nistelrooy
860-289-0800
c.vannistelrooy@elektor.com

Elektor Labs
Wisse Hettinga SUPPORTING COMPANIES
+31 46 4389428
w.hettinga@elektor.com
ACO Pacific, Inc. 38 Home Theater Shack 64

Germany All Electronics Corp. 64 KAB Electro-Acoustics 17

Ferdinand te Walvaart
+49 241 88 909-17 Amtrans Corp. 35 Linear Integrated Systems 39
f.tewalvaart@elektor.de
Analog Emporium 17 Materion Electrofusion 30-31
France
Denis Meyer
Avel Lindberg, Inc. 62 Menlo Scientific, Ltd. 61
+31 46 4389435
d.meyer@elektor.fr Axpona 47 OPPO Digital, Inc. 9

BTB Elektronik-Vertriebs, GmbH 53 Parts Connexion 2

Netherlands
Harry Baggen Celestion 27 Parts Express 68
+31 46 4389429
h.baggen@elektor.nl Earthworks, Inc. 21 Silicon Ray 63

Front Panel Express 55 Solen Electronique, Inc. 51

Spain
Eduardo Corral
+34 91 101 93 95
Hammond Manufacturing Co. 3 Tymphany 67
e.corral@elektor.es

Italy NOT A SUPPORTING COMPANY YET?

Maurizio del Corso
+39 2.66504755 Contact Peter Wostrel (audioxpress@smmarketing.us, Phone 978-281-7708, Fax 978-281-7706)
m.delcorso@inware.it to reserve your own space for the next edition of our members’ magazine.

COLUMNISTS
Brazil
João Martins
+31 46 4389444
j.martins@elektor.com
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
Portugal English, French, German, Dutch, Italian, Spanish, and Portuguese—and The Loudspeaker Recipes.
João Martins
+31 46 438944 Vance is the editor of Voice Coil: The Periodical for the Loudspeaker Industry, a monthly publication.
j.martins@elektor.com Although he has been involved with publishing throughout his career, he still works as an engineering
consultant for a number of loudspeaker manufacturers.
India
Sunil D. Malekar Dr. Richard Honeycutt fell in love with acoustics when his father brought home a copy of Leo Beranek’s
+91 9833168815
ts@elektor.in 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
Russia
Nataliya Melnikova electroacoustics from the Union Institute in 2004. Richard’s work includes architectural acoustics, sound
+7 965 395 33 36
system design, and community noise analysis.
Elektor.Russia@gmail.com

Mike Klasco is the president of Menlo Scientific, a consulting firm for the loudspeaker industry, located
Turkey
in Richmond, CA. He is the organizer of the Loudspeaker University seminars for speaker engineers. Mike
Zeynep Köksal
+90 532 277 48 26 specializes in materials and fabrication techniques to enhance speaker performance.
zkoksal@beti.com.tr

Steve Tatarunis has been active in the loudspeaker industry since the late 1970s. His areas of
China interest include product development and test engineering. He is currently a support engineer at
Cees Baay
+86 21 6445 2811
Listen, in Boston, MA, where he provides front-line technical support to the SoundCheck test system’s
CeesBaay@gmail.com global user base.

audioxpress.com | February 2014 | 7

ax What’s News — Innovation Takes Center Stage

The New Schoeps V4 U Microphone

The “retro” look has been popular in pro audio design for some time now, and
that preference is very apparent in the new styles of recording microphones. In 1951,
Schoeps introduced the CM 51/3, a condenser microphone in an unusually trim pack-
age. Its compact size was due to a low-noise Telefunken EF-94 (6AU6) vacuum tube
and a compact output transformer. Both omnidirectional and figure-8 capsules were
included, and a cardioid pattern was obtained by electrically combining their outputs.
Recently, Schoeps introduced its new V4 U microphone, which looks similar
to the earlier CM 51/3, but inside it has an entirely new capsule and electronics.
The new microphone is cardioid only and contains a small-diaphragm capsule
with a 33-mm beveled collar that causes the directivity to increase steadily and
smoothly at high frequencies. However, it maintains the smooth frequency and
polar response typical of small-diaphragm capsules.
The electronics contain a newly developed bridge-type field-effect transistor (FET)
balanced output circuit, which is transformer-less and free of coupling capacitors.
Output impedance is specified at 97 Ω and is constant with frequency. Schoeps
microphones have traditionally been known for their wide dynamic range, and
the V4 U is no exception. It offers a 144-dB maximum sound pressure level, cor-
responding to a 4.8-V output level. The microphone operates with standard 48-V
phantom powering. Billed as a vocal microphone, Schoeps notes a “warm, clear
sonic character with smoothly rolled-off diffuse-field response.”
The Schoeps V4 U condenser microphone is modeled after
the CM 51/3, but contains an entirely new “tiltable” capsule Schoeps
and electronics. (Photo courtesy of Schoeps GMBH) www.schoeps.de

QSC Wins 2013 PLASA Innovation Award

The QSC CXD/PLD amplifier platform (featured in audioXpress, products released over the past year that, in the view of the industry
November 2013) was first introduced at Infocomm 2013 in Florida. judges, have a design that demonstrates a new style of thinking,
The QSC CXD/PLD amplifier platform was one of eight new products improved technical practice, new technology, innovative materials
recognized with a PLASA Award for Innovation at the Pro Lighting or techniques, improved safety, or unique intellectual property.
And Sound Association (PLASA) show in London (October 6–9, 2013).
Featuring user-configurable, built-in DSP, the amplifier platform QSC Audio Products
includes Flexible Amplifier Summing Technology (FAST), which enables http://qsc.com
the amp’s total power allocation to be distributed
across one, two, three, or all four of its outputs.
The distribution enables amplifier channels to be
combined for maximum current or voltage output,
depending on the requirements.
The judging panel, comprised of industry
experts in the field of lighting, sound, and installa-
tion technology, commented: “This is an incredibly
flexible amplifier that can be used in almost any
situation. It can be subdivided and reconfigured,
and will work well in installations or as hire stock.”
“CXD and PLD amplifiers are a reflection of our
use of cutting-edge technologies and our deep
understanding of power amplification, DSP and
loudspeaker engineering and we’re delighted to
see that the PLASA judges feel the same way,”
said Ray van Straten, QSC’s Senior Director, Mar-
keting Communications and Training & Education.
The PLASA Awards for Innovation focus on

8 | February 2014 | audioxpress.com

ax What’s News — Innovation Takes Center Stage
Audio Precision Brings AP Performance to Loudspeaker Tests
Audio Precision (AP) announced a new software release that
expands the features of the electroacoustic test suite for APx audio
analyzers. APx v3.4 adds Thiele-Small (T-S) parameters, complex
impedance, and loudspeaker production test to the APx platform.
This expanded electroacoustic capability makes APx audio
analyzers the preferred choice for designing and testing inte-
grated audio products that incorporate electronics, DSP, and
loudspeakers.
The APx electroacoustic suite includes an energy time curve
(for quasi-anechoic measurements), impulse response, frequency
response, relative level, phase, distortion product ratio, distor-
tion product level, Rub & Buzz, and modulated noise (for air leak
detection). Output options include waterfall charts and polar plots
via APx utilities.
Converged Audio Test covers all aspects of today’s integrated
audio products. AP is the recognized standard in analog audio
tests with ultra-low distortion, wide input and output ranges,
and high accuracy measurements, while APx I/O options provide
native connectivity for a wide range of digital formats including
Bluetooth, HDMI, PDM, and Digital Serial. With the expansion of the
APx electroacoustic test suite, integrated audio products can be
tested in every domain, and at every step—from R&D to production.
R&D engineers who work on integrated audio products must
be able to obtain reliable results from each part of the signal
chain, from analog inputs to digitally processed compensation to
The APx525 analyzer can be combined with the new Loudspeaker Production Test suite
for work on integrated devices such as powered speakers, TVs, and Bluetooth headsets.
For work involving driver evaluation or final production test, a simpler APx515 can be
used.
10 | February 2014 | audioxpress.com
power amplifiers and loudspeakers. The complete APx electro-
acoustic suite, which includes incredible flexibility and reporting
capacity, is ideal for these tasks. The full range of T-S parameters
may be obtained using added mass, known volume, or known
mass methods.
For high-speed production tests, impedance magnitude and
phase plus a limited set of T-S parameters are calculated from a
single log sweep.
Because APx is a unified platform, R&D can define tests and
acceptable ranges of results and send this information directly
to the factory for complete quality control of the manufacturing
process. With this extended electroacoustic functionality, R&D
groups working on different sections of a device (e.g., the power
amplifier, Bluetooth, HDMI, DSP, and speakers) can seamlessly
share APx projects and results, and then send a single, lockable
test sequence with limits and reports to manufacturing.
The new electroacoustic measurements are enabled through
two software options. Both options are available concurrent with
the APx500 v3.4 release. An APx analyzer is required to run the
measurements. All models support the new options.
The APX-SW-SPK-RD (for R&D) tests impedance magnitude,
impedance phase, impedance real, impedance imaginary, com-
plete T-S parameters, energy time curve, impulse response, fre-
quency response, relative level, phase, distortion product ratio,
distortion product level, Rub & Buzz, and modulated
noise. It includes all measurements in the loudspeaker
production test measurement detailed below and polar
plots and waterfall graph utilities. The APX-SW-SPK-RD
costs $1,500.
The APX-SW-SPK-PT (for production tests) mea-
sures frequency response, relative level, phase, dis-
tortion product ratio, distortion product level, Rub
& Buzz, impedance magnitude, impedance phase,
limited T-S parameters, and modulated noise. The
APX-SW-SPK-PT costs $750.
The complete new Loudspeaker Production Test
suite uses AP’s highly reliable hardware and intui-
tive software instead of the sound card-based solu-
tions currently on the market. A full production test
package (including analyzer hardware and software)
costs $7,200, with no recurring license fees.
The hardware and software can measure
speakers, power amplifiers, and other electronics.
APx’s easy automation enables faster line setup
or adjustment as test needs change. Because the
hardware is reliable and ships with an ISO:17025
accredited calibration, test stations have fewer
problems and no time is spent questioning the
accuracy of results.
Audio Precision
www.ap.com
 Multichannel Audio Testing
Audio Precision’s APx582 Multi-Channel Audio Analyzer is a more
cost-effective version of its APx585, which is a true multichannel audio
analyzer with eight analog input channels and eight analog output channels.
The APx582 retains the eight input channels, but the number of analog
outputs has been reduced to two. The test system has been designed for multichannel power
amplifiers, automotive head units, multichannel receivers, multichannel converters, as well as Dolby/ Audio Precision’s APx582
dts confidence testing. It also has 192-kHz digital inputs and outputs with AES/EBU, S/PDIF and Toslink is a more cost-effective
connectivity. I/O is expandable with optional digital serial, HDMI with ARC, PDM and Bluetooth mod- version of the APx585.
It retains the eight input
ules. The analyzer is flat from 5 Hz to 20 kHz ±0.008 dB, and residual THD+N is specified as –105 dB
channels but it only has
within a 20-kHz bandwidth. The AG52 generator option is included, which the manufacturer claims
two analog outputs.
generates perfect square waves. The APx582 is compatible with Windows-platform PCs, including
Windows 8, 7, XP, and Vista.

Audio Precision
www.ap.com

Rohde & Schwarz

The Rohde & Schwarz UPP line of audio analyzers includes
three models—the UPP 200, the UPP 400, and the UPP 800, fea-
turing two, four, and eight channels, respectively. Designed to be
compact and cost-efficient, the units can be used as stand-alone
devices when connected to a keyboard, a mouse, and a monitor.
Alternately, they can be interfaced with a computer for remote
operation. Testing capabilities include level and signal/noise (S/N)
measurements, THD+N, THD alone, modulation distortion in accor-
dance with IEC 60268-3, intermodulation distortion, DC voltage,
frequency, phase and group delay, polarity, FFT analysis, 1/n octave
analysis, and HDMI device testing.
User-programmable filters designed for fast operation supple-
ment the standard weighting filters included with the analyzers.
XLR connectors, selectable as balanced or unbalanced, are used for
analog inputs and outputs. Digital connectivity includes nine-pin
D-Sub male, transformer-coupled 110-Ω interfacing, 75 Ω S/PDIF
via BNC connectors, Toslink optical, and HDMI. I2S is also avail-
able via a 25-pin D-Sub male connector. Sine wave performance is
specified for a frequency range of 20 Hz to 20 kHz, ±0.1 dB, with
THD+N of –100 dB from 20 Hz to 22 kHz.
Everything is included with the Rohde & Schwarz’s UPV audio
analyzer, which is a stand-alone device with a built-in display,
including an integral computer—no peripherals are required. The
manufacturer notes that when an audio analyzer is used with an
external computer, “the monitor or the interface connectors can
emit a level of disturbance that impedes the measurement of the
audio DUT. In contrast, the R&S UPV’s EMUI properties, including
the integrated PC, have been tested and therefore meet all of the
requirements placed on a measuring instrument. No conventional
Rohde & Schwarz’s UPV audio analyzer (top) is a stand-alone device
complete with a built-in display. It includes an on-board computer and
does not require any peripherals. The UPP 800 (bottom) is a compact,
eight-channel analyzer that can be used as a stand-alone device when
connected to a keyboard, mouse and monitor.
PC offers such extensive shielding measures, which include mag-
netically shielded AC power transformers or filter plates in front
of the display.”
Recently, Rohde & Schwarz updated its R&S UPP and R&S UPV
audio analyzer platforms with new firmware that enables Dolby
licensees to perform compliance self-testing and simplifies con-
figuration management, at no additional cost.
Dolby protocol dictates that before a licensee can launch new
products to the market they are required to pass a comprehen-
sive compliance test. This collaboration between Rohde & Schwarz
and Dolby simplifies the procedure and reduces the compliance
testing time for Dolby licensees significantly. Depending on the
type of device being tested, the programs will select and run the
tests required by Dolby for that specific device.
All measurements follow Dolby specifications,
test routines are carried out automatically, and
results and device information are summarized
in a test report.

Rohde & Schwarz

www.rohde-schwarz.com

audioxpress.com | February 2014 | 11

ax New to the Market

Solid State Logic’s Sigma

Innovative Mix Engine
In early 2013, British-based Solid State Logic (SSL) produced Sigma, its first
stand-alone summing mixer. Sigma is a remote-controlled analog mix engine
based on an updated version of the SSL SuperAnalogue mix engine, which powers
its top-of-the-line studio mixers. Unfortunately, we are not yet able to share any
more “inside” details. Nevertheless, one of the 32-input analog summing box’s
immediately notable features is that any Digital Audio Workstation can use HUI,
Media Control Interface (MCI), or even general MIDI to control all the levels.
By
Miguel Marques
(Portugal)
The digitally controlled summing mixer was an The SSL Summing
immediate success when it was introduced at the SSL is known for its large-format mixers, which
2013 Prolight+Sound in Frankfurt, Germany. But as have been a worldwide reference in recording and
sometimes happens with new product announce- broadcast studios for more than three decades. SSL
ments, most people didn’t realize there’s a lot more is also known for its product quality and its innova-
to the Sigma than what meets the eye. The Sigma tions in the audio industry. After practically inventing
can solve problems in any studio configuration. Not on-board automation systems for large consoles, SSL
only can it be a digitally controlled summing box also was the first to produce analog mixers with both
with talkback and monitoring functions, it can also digital control and connectivity. This is a notable fea-
provide an automation system to any console. It can ture in its Duality and AWS series of consoles, which
also be an easy way to integrate analog hardware can control and be controlled by any major DAW while
into a digital setup without worrying about latency still operating as full-format analog mixers. The SSL
or phasing issues between channels. Sigma is, in its essence, the centerpiece of an AWS

I/O meters mostly occupy the Sigma front panel. A knob level can control monitoring and talkback functions, and two user-defined buttons can
be assigned to functions in the control panel. The Sigma also includes an external audio input using mini TRS input and has a standard 0.25” TRS
headphone output.

12 | February 2014 | audioxpress.com

series mixer in compact stand-alone 2RU gear. The
Sigma’s front panel features a large section of input
metering and a single knob that controls most of the
monitoring section’s features.
With 32 inputs, the Sigma can work as a sim-
ple summing box, providing analog summing of
those signals using SSL’s SuperAnalogue circuitry.
The SuperAnalogue concept is the basis of most of
SSL’s recent products. SSL’s Sigma provides clean
sounding results and its signals can be easily and
pleasantly saturated, if needed. The 32 inputs can
work as 16 stereo channels, with just one level con-
trol for each pair of inputs. Each pair is assigned to
left and right buses, or as 16 mono channels with
pan and level controls for each one. All 32 inputs are
available on the back of the unit, using DB25 connec-
tors, which is a typical design in products that need
many channels. Inside, the Sigma has two mixing
busses, conveniently called “Mix A” and “Mix B” with
balanced insert points for using, for example, a bus
compressor or a program EQ over a whole mix. Both
mixing busses have their own level controls, and as
with each individual channel, they can be controlled
from a DAW or a computer.
Connecting the Sigma
Digital control is provided using high-resolution
multiplying digital-to-analog converter (MDAC) chips,
which is the key component on all the SSL digitally
controlled products. In this case, the MDAC chips con-
vert the digital control signals received through HUI/
MCI or MIDI to an analog signal in a level relative to
their original value. This is the control signal that will
later set the levels or gain of each channel and pan
(when available) in the analog domain. All knobs and
functions directly control these chips, which not only
ensures the program’s integrity is maintained, but
also makes it possible to store and later recall all the
settings and levels from the Sigma. The same concept
is used in the larger SSL products (e.g., the AWS and
the Matrix mixers) where even the faders are assigned
to a MDAC chip instead of directly affecting the under-
lying audio signal’s level. SSL mentioned it uses very
high-resolution MDACs in these products, though it
does not specify how much resolution or which chips.
The Sigma connects via Ethernet to any com-
puter or uses Wi-Fi through a wireless router. It is
compatible with almost any software and OS. All the
Sigma’s settings and configurations can be accessed
through any modern web browser to a HTML5 back
end, which means iPad or other mobile devices can
be used to set up the Sigma. Through this setup it’s
possible to control Sigma’s main functions, select one
of the two monitoring outputs, control the volume,
and store and recall presets. For each DAW and each
desired configuration, SSL provides installation tips
to help configure the Sigma, whether you use the HUI
or MCI control protocols. Automating the Sigma is
also possible using dedicated MIDI tracks assigned to
each channel of the unit. Automation can be written Most of the I/O in the
on the DAW or controlled in real time using a MIDI SSL Sigma is done using
controller, and the Sigma will respond instantly to DB25 connectors, with the
any volume change. exception of the main and
monitor outputs, available
The really clever part of the Sigma’s design is the
on XLR connectors. There’s
32 direct outputs, also in DB25 connectors, which
a USB connection and an
enable you to use the unit side by side with any con- Ethernet RJ45 connector for
sole to provide automation for its channels. By using pairing the unit with any
each channel insert of the mixer connected to each computer or wireless router.
I/O channel of the Sigma, or by placing the Sigma A foot switch can enable
between the DAW and the mixer, you get instant and disable user-defined
DAW-controlled automation. functions from the main
It is also possible to use the Sigma even if some- configuration panel.
one’s using dedicated digital recorders (e.g., a Radar)
or tape machines. You can use a dedicated computer
to run a lightweight DAW for automation to those
transports. Obviously the time code must be shared
between the transport and the automation DAW. The
Sigma can be extremely helpful even in studios that
don’t have an analog mixer but want to incorporate
analog process in the mixes.
Because all channels are summed in analog,
there’s pretty much no latency and the analog pro-
cessing can be inserted before the Sigma. Because it
is added before the Sigma, the volume automations
will not affect the settings on your compressor or
other dynamics processing tools.
Even though we couldn’t yet test and measure a
Sigma, we can attest to the good design and engi-
neering that’s gone into this product. This is definitely
the most complete solution in its category, consid-
ering Sigma is also packed with impressive features.
It costs $4,499.
Solid State Logic
www.solidstatelogic.com
Manufacturer Specifications
Maximum I/O Level: 18, 22, or 24 dBu
Frequency response: 20 Hz–40 kHz, ±0.3 dB
THD+N: < 0.025% (20 Hz–20 kHz)
Noise CHIP to CHOP: < –83 dBu at 24d Bu (20 Hz–20 kHz)
Noise CHIP to MIX A: < –75 dBu at 24 dBu (20 Hz–20 kHz) (stereo, all channels routed)
audioxpress.com | February 2014 | 13
ax It’s About the Sound

High-Definition Digital
Direct-Stream Digital (DSD) recording/playback technology takes a
different approach to existing pulse-code modulation (PCM)-based
systems and delivers the high-resolution stereo or multichannel
By
Gary Galo audio found on the Super Audio CD (SACD). DSD encodes audio
(United States) data using 1-bit samples taken at 2,822,400 samples per second.
This is 64 times faster than the rate used on standard audio CDs,
which enables the digital representation on SACD audio to more
closely follow the analog source signal that is being encoded.
DSD technology appears to be gaining in popularity among
audio industry professionals.

In October 2013, the Audio Engineering Society
(AES) 135th Convention kicked off AES week in New
York with a 2-h monthly meeting focused on “Move
Over PCM and Make Room for Hi Res DSD and DXD
Audio.” Three of the world’s leading proponents of
the new ultra-high-resolution recording formats—
Dominique Brulhart of Merging Technologies, Morten
Lindberg of 2L, and John Newton of Soundmirror—
lead the discussion.
Sony developed the DSD recording format for the
SACD. A 1-bit system, DSD originally operated at a
2.8-MHz sampling frequency, providing a 100-kHz
bandwidth. Conventional 2.8-MHz DSD recording is
often referred to as DSD64 because the sampling rate
is 64 times that of the 44.1-kHz CD standard. (The
“2.8 MHz” designation has been rounded off.) But in
recent years, the base-sampling frequency has been
doubled to 5.6 MHz and even quadrupled to 11.2 MHz.
These new DSD formats are often referred to as
DSD128 and DSD256 (i.e., Double DSD and Quad DSD).
Although the SACD has never become much more than
a fringe audiophile format, the DSD recording system
has a small but loyal following in the audio industry.
One of the main problems with DSD is the difficulty and
expense of editing. PCM editing offers much greater
The configuration screen
of Merging Technologies’
Horus network interface
supports recordings from
44.1 to 192 kHz, DXD and
up to DSD256 modes.
flexibility and ease of use and far greater signal pro-
cessing capability, than raw DSD editing.
Digit a l e X t r e m e D e f initio n ( DX D) is a n
ultra-high-resolution recording format developed
by Merging Technologies, which offers the Pyramix
multitrack recording system. DXD operates at sam-
pling frequencies of either 352.8 kHz (8 × 44.1) or
384 kHz (8 × 48), up to 32-bit floating. It also sup-
ports all the standard sampling frequencies from
44.1 to 192 kHz. DXD is a PCM format (the title of
the presentation was a bit misleading in that regard)
and is not new.
Merging Technologies exhibited an earlier version
of the Pyramix system at the 123rd AES Convention
in 2007. One of DXD’s virtues is its conversion tech-
nology to and from DSD is incredibly sophisticated
(352.8 kHz is the preferred sampling frequency for
optimum conversion to DSD). Because DXD is a PCM
format, the Pyramix recording system offers the ease
and flexibility recording engineers expect from PCM.
However, because the conversion to DSD is so trans-
parent, DXD makes an excellent recording and mas-
tering system for a finished product in DSD format,
whether an SACD or a raw DSD file.
Brulhart gave an overview of the formats and
described the new ultra-high sampling frequencies
used for DSD. Indeed, the “merging” in Merging
Technologies focuses on bridging the gap between
the two digital formats, and the panelists noted that
the conversion algorithms will further improve in
the future.
Demos and Testimonies
Lindberg is the proprietor of 2L, a small Norwe-
gian company specializing in high-resolution digital

14 | February 2014 | audioxpress.com

 The Trondheim Solistene ensemble is
featured on DXD recordings produced
by 2L. The recordings are available on
Pure Audio Blu-ray and SACD discs, or
as high-resolution downloads. DXD and
high-resolution DSD were among the
topics discussed at an AES, New York
Section, monthly meeting that featured
representatives from 2L, Merging
Technologies, and Soundmirror.
(Photo courtesy of 2L)

audio. 2L has also issued recordings in high-resolu- only those spots—often only 10 ms in length—to DXD
tion PCM formats on Pure Audio Blu-ray discs and on to perform the edit, and then convert back to DSD. The Horus Networked
SACDs. About 50% of its sales are high-resolution Newton noted that Merging Technologies’ recording Audio Interface from
downloads. equipment follows the Ravenna protocol described Merging Technologies
Lindberg noted that attendees who were hoping in my 2008 article about the AES 2007 Convention quickly became one of the
for a boxing match between DSD and DXD would (see Resources). The technology uses CAT-5 cabling most popular AD/DA and
microphone pre-amplifiers
be disappointed. Both formats are excellent, in his from the performance space to the control room.
solutions used for high-
view, providing a level of transparency that enables This minimizes the amount of microphone cabling
resolution audio recordings.
the recording engineer to forget about the recording he transports for a location recording, which results In the picture, Merging
system and concentrate on microphone placement. in a substantial cost savings. He normally buys the Technologies’ President,
Lindberg prefers Brüel & Kjær 130-V omnidirectional CAT-5 cable locally and leaves it on site—it’s less Claude Cellier is supervising
condenser microphones and makes nearly all of 2L’s expensive than shipping it back and forth and it can a recording session where
recordings in a five-channel surround format, gen- always be reused for future sessions at the same the Horus is used as an I/O
erally using only one microphone per channel. Spot location. Newton said the differences between DSD connection node to RAVENNA
microphones are not used in most of their record- IP Audio Networks.
ings, and equalization is only needed on about 10%
of them. Lindberg records and masters in the DXD
format and does not go back and forth between DXD
and DSD. When the master is finished, he makes a
one-time conversion to DSD.
Newton has been the proprietor of Soundmirror
since 1971. Soundmirror is a Boston, MA-based com-
pany that specializes in high-resolution digital record-
ing. It was responsible for the remastering of the
RCA Victor Living Stereo material issued by BMG on
SACD. Newton prefers recording directly in DSD for-
mat. But, he admitted that however spectacular the
results, using DSD for recording, mixing, and edit-
ing is very expensive and time consuming. In cases
where cross-fade edits are needed, he will convert

audioxpress.com | February 2014 | 15

ax It’s About the Sound

“I asked Merging
[Technologies] to pay
attention to the headphone
amplifier when Horus was
being developed. These
often sound very poor on
other devices. I must say
they definitely listened
to me because it is really
outstanding. It also saves
me from taking a separate
amp and all the cables
and power supply,” says
Jean-Daniel Noir from
the specialist recording
company, JDN—based in
Gland-Vich, Switzerland.
at 2.8, 5.6, and 11.2 MHz is audible even in simple
recording/playback demonstrations, and not just in
editing and processing.
An assortment of high-resolution digital record-
ings was played to demonstrate the working methods
and recording philosophies of the participants. Lind-
berg played a military band recording that was very
impressive in terms of detail and depth perspective.
The stereo layer from this SACD was created from
his five-channel pickup. Newton played a Haydn Sym-
phony performed by the Oregon Symphony Orchestra,
recorded in DSD at 11.2 MHz using a pair of Brüel &
Kjær 2006 microphones for the stereo pickup. The
playback loudspeakers were a pair of NHT Pro Model
A-20 Active Monitors, a powered two-way loudspeaker
system with a 6.5” treated paper cone woofer and a
butyl rubber surround, and a 1” metal dome tweeter
with a textile surround. Amplifier power was rated at
250-W continuous RMS, and the system also included
its own outboard control amplifier.
The meeting venue was a fairly intimate space.
The loudspeakers had no difficulty supplying clean,
detailed sound, though the bass, as expected from
such a small woofer, was deficient. Unfortunately, the

ARSC New York Chapter

Following four days of technical presentations and exhibits at
the 135th Audio Engineering Society (AES) Convention (October
2013), it was easy to lose sight of what the audio field is really
about—namely music. One of the most important branches of
the audio field is the transfer, preservation, and restoration of
historical recordings by the great performers of the past, and
the Association for Recording Sound Collections (ARSC) is the
leading professional organization dedicated to that aim.
The ARSC’s membership is diverse, and includes audio archi-
vists, transfer and restoration engineers, and librarians (some
employed by university libraries and sound archives and others
who work in the private sector). Private collectors are the tradi-
tional backbone of ARSC, and many of its members pursue their
interests as an avocation and a profession. Several ARSC mem-
b e r s a r e a l s o belong to AES, myself included. In addition to
ARSC’s annual conferences, the New York
Chapter offers presentations related to
sound recording history at its monthly meetings, held at the City
University of New York’s Sonic Arts Center at West 140th Street
and Convent Avenue.
On October 14, 2013, the New York ARSC meeting featured
Jon M. Samuels and Joseph Patrych, who discussed Sony Classi-
cal’s new 41-CD collection Vladimir Horowitz: Live at Carnegie
Hall. Horowitz was one of the most significant pianists of the
20 th century. His art devotees often prefer the spontaneity of
his live performances to the more sanitized music found on his
familiar studio recordings, which makes this collection a trea-
sure trove for Horowitz admirers. The collection spans the years
1943 through 1976. Many of the performances have not been
previously released in any form. A generous sampling of the
collection’s recordings was played during the presentation to an
enthusiastic and attentive audience.
Samuels, the proprietor of Recorded Legacy and an individ-
ual well known by collectors of historical reissues, served as the
collection’s producer, engineer, and booklet annotator. Judging
from the playbacks presented at the meeting, he has performed
admirable restoration work on the diverse sources, which include
78- and 33 1/3-RPM lacquer discs, plus mono and stereo tapes.
Patrych is a New York-based recording engineer, owner of
Patrych Sound Studios, and a life-long collector of historical piano
recordings. He served as consultant to the project and was a
gracious and informative host for the ARSC meeting.

Association for Recording Sound Collections

www.arsc-audio.org

At the October 2013 New York Chapter of the Association for Recorded Sound
Collections (ARSC) meeting, producer/engineer Jon M. Samuels and project
consultant Joseph Patrych gave a presentation on Sony Classical’s 41-CD collection
Vladimir Horowitz: Live at Carnegie Hall. (Photo courtesy of Sony Classical)

16 | February 2014 | audioxpress.com

playbacks were hampered by ventilation noise and
musicians practicing in adjoining rooms.
Recently, I heard another Newton DSD recording
of the Oregon Symphony conducted by Carlos Kalmar.
The recording was a Pentatone Music SACD featur-

Resources
2L, www.2l.no.

G. Galo, “AES 2011: Details of a Full Technical Program,”

audioXpress, April 2012.

———, “Digital Audio: Progress, Stagnation and

Regression,” Linear Audio, Vol. 2, September 2011.
ing the “Four Sea Interludes and Passacaglia” from A view of the control
———, “New York AES 2009: The Audio Industry room shows the Chicago
Benjamin Britten’s opera Peter Grimes. Symphony Orchestra
feels the Economic Pinch,” audioXpress, June 2010,
(online edition only). On my own two-channel system, Newton’s record- recording an upcoming
ing sounded wonderful. It was spacious and natural CSO Resound release in
———, “Is SACD Doomed? The New York AES 2007: which David Frost was the
with an excellent sense of the recording venue. Digital
A Convention Notable for What Wasn’t There,” producer (reading scores
audioXpress, June 2008. audio has evolved to the point where the technology
at table); with Tim Martyn,
need not draw attention to itself, leaving only the engineer (at console); and
Merging Technologies, www.merging.com.
music. I should also mention the Oregon Symphony, Charlie Post, engineer (near
Pentatone Music, www.pentatonemusic.com. under Kalmar’s direction, is a fine ensemble, and the computer monitors). (Photo
conductor demonstrates an understanding of Britten’s courtesy of Todd Rosenberg)
Soundmirror, Inc., www.soundmirror.com.
musical language. Very well done, indeed! ax

Merrill Replica
ES-R1
Turntable manufactured in the USA by George Merrill

audioxpress.com | February 2014 | 17

 ax Standards Review

Audinate Dante
(Part 1)

Making Digital Audio

Networking Easy

The story of Audinate and its Dante audio

networking implementation is one of
the most interesting examples of
perseverance and focus in the
industry. It demonstrates
the evolution to an
By industry standard
João Martins results from a clear
(Editor-in-Chief) vision of market needs and
the ability to fulfill those necessities
with the practical implementation of
converging technologies.

Sydney, Australia-based
Audinate became a global
company in just 10 years
thanks to the success of its
Dante technology.
In a two-part article series, we will explore the
reasons why Dante technology became the “de facto”
standard in audio networking and reveal the com-
pany’s perspective of “surfing the wave” of audio
over IP (AoIP). We will also discuss the company’s
ability to cleverly manage existing industry efforts
and commercial requirements to build a unique
example of marketing success. As Audinate’s CEO
Lee Ellison explains, because “the great thing about
standards is that there are so many of them.”
Built on existing networking protocols and stan-
dards, Audinate’s Dante technology is a self-de-
scribed “plug-and-play networking solution, which
delivers synchronized media with ultra-low latency,
simplifying the installation and configuration of digi-
tal media networks.” Dante is currently the market’s
leading digital audio networking solution, adopted
by approximately 150 OEMs to date (some still to
be publicly disclosed). Those OEMs have already
developed a unique ecosystem of hundreds of com-
patible products. Currently, applications that use
Dante can be found throughout the pro audio live
sound market. But most importantly, Dante is also
reaching the commercial installation, broadcast,
and recording studio segments.
The Dante networking solution became widely
accepted among many pro audio manufacturers and
is currently deployed in thousands of installations
worldwide because of its self-configuring network
architecture. Dante uses standard IP over 100-Mbps
and 1-Gb Ethernet, enabling easy setup and auto-
matic discovery of devices on the network with
one-click signal routing and user-editable names.
In contrast to previously existing audio network
technologies, Dante distributes digital audio plus inte-
grated control data with imperceptible latency, sam-
ple-accurate playback synchronization, and high-chan-
nel counts. The technology reliably uses standard
network infrastructures, even with high-quality digital
signals in high-sampling frequencies.
As licensed technology, Dante also offers a
combined hardware and software toolset, enabling

18 | February 2014 | audioxpress.com

manufacturers to quickly implement their own solu-
tions while leveraging continuously improved soft-
ware tools (e.g., the Dante Controller application
for Windows 7 and Windows 8 and Mac OS X). This
application enables manufacturers to set up, manage
audio routes, and configure Dante devices in any
sized network, and also enables real-time network
monitoring functionalities with channel metering,
multicast bandwidth usage, and clock health mon-
itoring and event logging, which enables users to
quickly identify and fix potential network issues.
But possibly the most interesting software com-
ponent in a Dante solution—resulting from Audi-
nate’s IT standards approach—is the availability of
the Dante Virtual Soundcard technology. The tech-
According to Lee Ellison,
nology enables any PC/Mac connected to a Dante directly to many of the best digital consoles on the
Audinate’s CEO, “The
audio network to use a computer’s Ethernet port market. By incorporating Dante, we have both sim- adoption of Dante among
to communicate with other Dante-enabled devices. plified the setup of systems…and added versatility A/V manufacturers doubled
Popular DAW applications (e.g., Nuendo, Cubase, for the user.” just in the past year. Our
Logic, or Pro Tools) can transmit and receive up to customers recognize that
64 bidirectional channels to networked audio equip- The IP Formula networking matters and
ment, via the Dante Virtual Soundcard, without any To understand how Audinate surpassed compet- having an interoperable,
additional hardware soundcard. ing existing network technologies propositions and proven networking
technology to get to market
Dante also appeals to the recording and studio thrived when discussions about the need for indus-
quickly has enormous
markets, attracting several important new OEMs try standards in media networks are prevalent, we
benefits.”
to join the Audinate licensing program. In the last interviewed Audinate CEO Lee Ellison.
two years, interesting developments in that market Ellison brought to the company more than 25 years
revealed Audinate’s ability to handle new develop- of executive leadership experience that encompasses
ments and implementations, working in coopera-
tion with Yamaha/Steinberg on its ambitious Nuage
post-production platform and with Focusrite on its
Rednet studio platform.
Demonstrating an ability to quickly respond to
existing market demands, Audinate attracted broad-
cast equipment manufacturers as diverse as DHD
from Germany, AEQ from Spain, and NTP from Den-
mark, among many others. In turn, Dante became
the most recognized and interoperable media net-
working solution in today’s market. And, broadcast
manufacturers are always willing to invest in their
own network developments, while others prefer to
adhere to standards or at least to steer existing devel-
opment efforts toward those potential standards.
But even among the companies with enough
resources to pursue such an endeavor, we are start-
ing to see “signs of fatigue” while decisions are
being made to quickly introduce Dante-compatible
solutions, as required by its own clients.
Aviom is another Audinate licensee. Its pres-
ident and CEO Carl Bader explains, “Dante has
become a standard in digital audio networking for
the pro audio industry, and incorporating Dante Live sound was the first industry segment targeted by Audinate. Working in cooperation
connectivity into our personal mixing system makes with companies such as Yamaha, the Dante network audio technology became essential.
it possible for our personal mixers to be connected (The Yamaha CL Series is pictured.)

audioxpress.com | February 2014 | 19

 ax Standards Review
Dante Controller is a free
software application for
Windows (7 and 8) and Mac
OS X. It can be used to set
up and manage audio routes
in a Dante audio network
and configure Dante devices,
providing real-time network
monitoring functionalities.
Dante uses automatic
device discovery and Zero
Configuration Networking—
Internet Engineering Task
Force (IETF) Zeroconf
protocols.
20 | February 2014 | audioxpress.com
a broad range of technology sectors including soft-
ware, computer, and telecommunications. His expe-
rience enabled him to quickly become an industry
spokesman on all media network-related subjects.
As Ellison is quick to instantiate when we refer
to Dante’s success as a “de facto” standard, “we
are becoming an enabling technology, rather than a
competing technology. We are enabling things that
could have never been done before…
“Maybe the reason is that, from the beginning,
we architected Dante looking at using the
IP technology as a foundation, unlike other
companies who approached this industry in
the past. Primarily, those were audio compa-
nies developing another network technology,
while we are not an audio company, we are
a networking company. We have also the
largest engineering team out there with a
background on all various forms of network-
ing, so that’s why we where able to do that.
“We knew there would be some additional
challenges and more stringent requirements
in transmitting uncompressed audio. From
the beginning our initial focus was really
the Live Sound. Commercial installation is
a larger market than Live, but we felt we
could address the more stringent require-
ments of live sound and we knew we had an
architecture from the perspective of being a scalable
platform, because we were using IP.”
But this is not the entire story. Essentially, Audi-
nate responded efficiently to a diversity of require-
ments and different goals from the manufacturers
that approached the company. Audinate showed
it could answer the key challenge of conflicting
requirements (e.g., latency, number of channels,
and higher sampling frequency), without compro-
mising the IP and networking standards approach.
It’s important to understand what was in
question here and how the industry handled those
dilemmas.
A Little History
In the analog domain, the main concern dealt
with noise interferences and audio signal degrada-
tion from long cables. With the adoption of com-
puter networks, the focus evolved from the simple
transmission of digital audio signals in point-to-
point, to finding the best way to efficiently distribute
and route audio without compromises in quality—if
only using, in practice, the Ethernet physical layers.
From standards such as AES/EBU (currently
AES3), a multichannel version was created and
Multichannel Audio Digital Interface (MADI) AES10
became the first solution to enable large routing sys-
tems and the distribution of up to 56 channels (later
expanded to 64 in the 2003 revision). The hardware
was expensive because it was based on coaxial
cable, but the possibilities still generated interest,
especially for use with broadcast applications.
Other digital point-to-point audio transmission
interfaces, such as TDIF and ADAT, survived for
quite some time together with AES3 and MADI, but
they required dedicated cables. Digital signals were
“transmitted” or distributed point-to-point in most
of the applications. But it become apparent that cre-
ating a true network topology, in which the signals
can be routed freely to any number of points with
bidirectional flows of independent data sharing the
same infrastructure, made sense. The flexibility this
would enable encouraged many R&D departments to
start examining solutions that would use standard
Ethernet network switchers, routers, and cabling.
Audio Engineering Society (AES) convention
exhibitors have demonstrated such systems since
2000. Among them was Sony’s demonstration of the
Pro-Audio Lab Oxford (2003) of SuperMac (10/100 Mb)
and HyperMac (1 Gb), which were later recognized
as the AES50 standard (2005). The technology was
later sold by Sony Oxford to Klark-Teknik. Today it
serves as the foundation for the successful range
of Midas and Behringer (The Music Group) digital
consoles and live sound solutions.
Interestingly enough, the technology was intro-
duced as a “multichannel audio interconnection”
solution running on CAT-5/6 concurrently with con-
trol data and (potentially) enabling independent
 ax Standards Review
With the Nuage project—
developed in close
cooperation between
Yamaha, Steinberg, and
Audinate—the Dante
technology was brought to
another level. An example
of the implementation
is the Dante Accelerator
audio interface card for
the Nuendo DAW, which
provides extra low-latency
transfers of as many as
128 simultaneous channels
of 24-bit/96 kHz audio
(64 channels at 192 kHz)
plus redundant Gigabit
connections.
audio channel switching and routing using low-
latency TDM technology. The technology was a pro-
prietary implementation in terms of networking
protocol, but was converted to an open standard
and temporarily supported by companies (e.g.,
Merging Technologies and SADiE) for studio and
post-production applications.
The impact of this implementation was that it
demonstrated the benefits of moving away from the
typical packet-based Ethernet systems because of
the latency in favor of higher sampling frequency
support (up to 96 kHz) and frame synchronization.
The AES50 approach only used the Ethernet tech-
nology’s physical layer—the cables and transceiv-
ers at each end—creating point-to-point transmis-
sions within the generic network infrastructure. The
change enabled a more efficient use of the available
bandwidth, which made the technology incompati-
ble with off-the-shelf IT equipment.
Simultaneously, competing technologies were
adopted in the market. The main one was EtherSound
from the French company Digigram. EtherSound used
a daisy-chain topology, in which all the devices, includ-
ing standard IT switchers, could be connected on a
continuous string of signals running over Ethernet
with “deterministic” latency. That is, the full net-
work, once running, achieved the same low latency
in all connected points. The problem was EtherSound
was never fully developed as promised, including
a star architecture, and it remained dependent of
single points of failure in a daisy-chain structure.
EtherSound for Gigabit Ethernet (ES-Giga) was
promised in 2006 but it was not compatible with
previous implementations.
CobraNet, a technology developed originally by
Peak Audio in the 1990s and later by Cirrus Logic,
was a different approach being developed around the
same time as EtherSound. In this implementation,
the signal routing was determined by network soft-
ware and not the network devices. The technology
used real audio distribution over Ethernet, unfor-
tunately at the cost of latency. It was designed for
large, fixed commercial installations where it was
not an issue to have more than 5 ms. CobraNet uses
standard Ethernet technology (IEEE 802.3) for audio
transport, transmitting to Layer 2 to distribute up
to 32 channels to and from each CobraNet device
over a single Cat-5. Unfortunately the solution was
fairly complicated in terms of network manage-
ment and was designed for people with extensive
IT knowledge and network administrators. It was
also limited to 100 Mbps Ethernet.
Recently, we have started to see more advanced
implementations in audio networking over IP/Ethernet
for studio, live broadcast, and live sound, which man-
age different requirements in terms of bandwidth,

Resources
Audio Engineering Society (AES), “AES standard for Audio Appli-
cations of Networks—High-Performance Streaming Audio-over-IP
(AoIP) Interoperability,” AES67-2013.
———, “Analog to Digital Audio in the 21st Century,” White Paper,
www.audinate.com.
———, “Audio Networks Past, Present and Future (CobraNet and
Dante), White Paper, www.audinate.com.
———, “Dante: Digital Audio Networking Just Got Easy 2.0US-09A09.
pdf, White Paper, www.audinate.com.
S. Schmitt and J. Cronemeyer, “Audio Over Ethernet: There are Many
Solutions—But Which One Is Best for You?” www.dspecialists.com/
sites/default/files/publication/110126a_networking_paperembedded
world2011_paperen_final_st_jc.pdf.
T. Shuttleworth, “Emerging Technology Trends Report,” Audio Engi-
neering Society (AES) Technical Committee on Network Audio Systems,
November 2011.
Sony-Oxford Technologies, “Oxford SuperMac/HyperMac AES50 Digital
Audio Interconnection,” technology brief, 2005, www.philippe-lahaye.fr/
IMG/pdf/xmac_brochure.pdf.

J. Berryman, “OCA Alliance Open Control Architecture: The OCA Alli- M. Johas Teener, A. Huotari, Y. Kim, R. Kreifeldt, and K. Stanton,
ance Overview,” www.oca-alliance.com.Audinate, “Evolving Networks “No Excuses Audio/Video Networking: The Technology Behind AVnu,”
to Audio Video Bridging (AVB)” White Paper, 2011, www.audinate.com. AVnu Alliance White Paper, 2009.

A. Hildebrandt, “Networked Audio: Aktuelle Entwicklungen & Tech- M. Vest, “High-Quality, Low-Latency IP-Based Audio Routing via
nologische Perspektiven für den Broadcast-Markt (Networked Audio: Ethernet,” NTP Technology Presentation, 2013, www.ntp.dk/media/
Current Developments & Perspectives for the Broadcast Market), VDT NTP_presentation_IP_Audio_2013.pdf.
International Convention, November 2010.

22 | February 2014 | audioxpress.com

latency, and synchronism. Examples include QSC’s
Q-LAN in the commercial installation segment, Axia
Livewire in the broadcast radio market, Riedel Rock-
net in the Live events market, and industry efforts
such as Audio Video Bridging (AVB—IEEE 802.1) and
Ravenna. Even with the first practical implementa-
tion demonstrations, which were supposed to be
based on existing standards, experience has shown
that it is not easy to make AoIP routing that enables
the use of low-cost off-the-shelf IT components while
using advanced Layer 3 Ethernet implementations.
At the same time, Ethernet is quickly evolving
from 1 to 10 Gbps. With 40- and 100-Gb equipment
promising to become a market reality very soon,
there is a clear demonstration of the importance
of building audio network technologies on top of
existing networking standards.
Figuring It Out
Ellison explains how all those requirements were
put in perspective with Dante. “One of the early
technologies was CobraNet, but it had a really high
latency and that was the reason why it never took
off in live sound. That’s why EtherSound did OK for
a while, but they had this daisy-chain ring topol-
ogy and it didn’t scale well. You couldn’t use it in a
convention center or an airport…
“The way we did it, was figure all that out by
using the right kind of clocking mechanism. By using
clocks that had precision time protocol and that
where accurately timed to one another, within plus
or minus 1 µs, we could solve a lot of those prob-
lems. Other technologies where also always limited
to 100 Mbps. They could never go beyond that, and
network speed/bandwidth is your friend when it
comes to latency. Depending on the manufacturer,
some might setup a small point-to-point system
at 250 µs, were as, you might typically setup the
rest of your network for 1 ms and we were able to
achieve that. For monitoring purposes, you would
try to set it up to be very short, where as for the
rest of the system you would try to put in a milli-
second and have everything else transmit plus or
minus one microsecond.
“We can also support multiple sample rates in
the same network and doing those kind of things.
We are also developing improved health monitor-
ing information and helping the person understand
the network. We know there are other competitive
technologies. Usually, we know that if companies
put enough engineering people on a project—and
there are good enough people and they are directed
in the right direction—you can potentially do that,
if their background is on networking. We are the
largest company doing this for this industry by far. Dante has already started to
By far! What a lot of the senior management in make inroads in the studio
these companies are realizing is ‘why should they and recording market with
Focusrite and the RedNet
develop a technology themselves, which at the end
range of modular Ethernet-
of the day isn’t as good as Dante is, cause they are
networked audio interfaces,
stealing those resources away from their core com- even bridging between Pro
petencies that differentiate their product to make Tools|HD or MADI systems
really cool professional audio equipment. I think and any Dante audio
people understand the argument of ‘build versus network.
buy.’ Once we build it, it continues to change and
evolve and we are always doing new things to it
and that is one of the benefits our customers get
from Audinate and the reason why our customers
stand behind us.”
Another key factor for Audinate growth is that
the company was intelligent enough to learn from
previous experiences and work with the implement-
ers of other technologies, as Lee Ellison acknowl-
edges, although clearly separating the “implement-
ers of an existing technology,” from the “creators.”
“We work with great implementation partners
and we have authorized implementers, like ZP The Dante-MY16-AUD card
Engineering and Attero Tech. We like to work with is a fully compatible Yamaha
really experienced implementers. Auvitran did some Mini-YGDAI standard card
that instantly Dante-enables
work—and they were the guys behind EtherSound;
any Yamaha console mixer,
and Attero Tech where among the largest CobraNet
processor, or power amp
installers—so we work with great authorized imple- with 16 bidirectional audio
menters who help our customers layout the products channels and full Dante
quickly and get to market quickly.” ax network audio redundancy
over Gigabit Ethernet.
audioxpress.com | February 2014 | 23
ax Audio Praxis

The Brave New World of

Loudness Control (Part 1)
New Post-Production Workflows
Transcend Compliance for Enhanced
By
Audio Creativity
Jon Schorah Coalescing around the ITU-R BS.1770 standard—Algorithms to Measure Audio
(NUGEN Audio)
Program Loudness and True-Peak Audio Level—loudness regulations or recommen-
dations are in place across the US as well as South America, Europe, Japan, and
much of the world. No matter where you produce audio and for whom, it’s almost
certain the broadcast will need to be loudness-compliant. It’s just a question of
how to make it happen.

As loudness management becomes the required
norm, the responsibility for compliance increasingly
falls on audio post-production engineers. But there’s
some good news. Since several world regions have
been delivering loudness-compliant broadcasts for
two years or more, a methodology for best practice
is beginning to emerge. The solutions for loudness
control are not only maturing, but also becoming
versatile tools in the engineer’s toolbox.
Initially, loudness compliance may appear
to be just another task added to the seemingly
ever-growing list of tickboxes engineers must check
before delivery. But if loudness normalization is
correctly integrated, it can become more than a
means of avoiding consumer complaints and poten-
tial fines. It can reintroduce creative freedoms lost
in the old peak normalization paradigm and become
a tool to improve broadcast audio quality.
Loudness and Post-Production:
A Creative Marriage
At first glance, playout processing is an obvi-
ous solution to loudness compliance. By definition,
playout processing achieves loudness normalization
as an afterthought by adjusting station output to
ensure compliance after post-production. However,

Figure 1: This section of

a feature film audio has
been repurposed for TV by
reducing the dynamic range.
The top image shows the
original signal analysis.

24 | February 2014 | audioxpress.com

 a) b)

experience has shown that playout processors are
not particularly good at solving loudness issues in a
consumer-satisfying manner. They can even intro-
duce their own new loudness problems. In regions
where loudness recommendations have been in
place for more than a year, there is a clear move-
ment away from correction after the fact.
A better solution would be to consider loudness
compliance as part of the creative process during
pre-production and even at acquisition. In this sce-
nario, simply pushing the mix against the limiter is
no longer a viable option. Instead, incorporate loud-
ness compliance into an experienced post-production
professional’s creative decision-making process. In
general, noncompliant material that falls outside a
tolerance margin is rejected. Near-compliant audio
can be satisfactorily corrected with loudness batch
processing that brings it into compliance with a gain
offset and possibly some true-peak limiting. Here, the
playout processor’s role shifts to that of an error-han-
dling stop-gap and is bypassed with the delivery of
compliant material.
Instead of adding another complicating factor to
the mixing process, the right loudness tools provide Figure 2: A section of
an engineer more creative freedom. Compressing heavily compressed audio
a mix to achieve a consistently loud level under has been normalized to –23
loudness normalization will cause the audio to be LUFS (a). The same section
turned down. So the best way to achieve a mix of audio is shown with light
compression normalized to
that stands out in the crowd is to creatively engage
–23 LUFS (b). A comparison
with the content and make the most of the avail-
clearly shows that instead
able dynamics. of producing a “louder”
Loudness normalization coupled with the use result, much of the heavily
of true-peak maximum levels enables you to cre- compressed signal is below
atively use dynamic range and increase headroom. that of the corresponding
When experienced operators ensure compliance in lightly compressed
post-production, the playout processor becomes alternative after loudness
largely inactive. These are great developments, normalization.
especially for professionals involved in short-form
production.
Loudness can be considered in pre-production
and acquisition. Normalizing archive and library
material and ensuring field recordings and outside
broadcasting (OB) sources are already compliant
speeds the production process, enabling a faster
turnaround while ensuring the audio remains faith-
ful to its original context. This can be especially

S Loudness
I Loudness
M Loudness minimum
M Loudness maximum
Variance maximum
Loudness range
True peak maximum
True peak clip
Alert
Figure 3: The typical log
file output from a real-time
loudness meter (VisLM-H)
shows the variation of
loudness parameters over
time.

audioxpress.com | February 2014 | 25

ax Audio Praxis
Figure 4: The same section
of audio is normalized to
–23 LUFS and to 0 dBFS
to show loudness vs. peak
normalization.
important during live sports events, breaking news
broadcasts, event coverage, and studio interviews.
Building a Post-Production Loudness
Workflow
New workflows call for new tools. Fortunately,
companies are developing products designed spe-
cifically for post-production engineers. Intuitive
audio-editing tools (e.g., real-time metering, offline
correction, and loudness-compliant limiting) enable
post-production editors to put their creative exper-
tise to work while ensuring compliance.
Ears are the best tools when it comes to making
creative decisions in audio post-production. That
rule also generally applies to loudness normalization.
The new loudness standards hold the potential for
increased dynamic range and contrast. But com-
puters also play a key role with their ability to take
measurements and make smaller adjustments to
get things exactly on target. Computers can work
quickly, saving considerable time near the end of
the process. With these tools in mind, a post-pro-
duction workflow begins to emerge.
Clear, intuitive loudness metering is the key to
delivering high-quality, loudness-compliant audio.
Because the new loudness measurements are
designed to correspond to the human ear, a good
engineer can almost mix sound without a meter. It’s
possible for skilled engineers in a calibrated room
to simply occasionally glance at the meter during
the creative process to maintain their bearings or
to check something in particular. But in the end,
Figure 5: Production to
loudness standards are
shown within a typical
nonlinear editor (NLE).
26 | February 2014 | audioxpress.com
even experts need to confirm that they’ve met target
values. Likewise, anyone looking to push creative
boundaries needs checks to ensure their work is
compliant. With visual meters, editors can keep an
eye on the meter and loudness profile while relying
on their trained ears to make most of their decisions.
Another important factor in this workflow is a
high-quality true-peak limiter that can handle the
new standard’s intersample true-peak requirement,
which is something traditional sample-peak limiters
cannot do. It’s tempting for engineers to use their
existing sample peak limiters with a setting that
would yield results “safe enough” to be compliant
with the loudness standards’ true-peak measure.
But, those who follow this practice do so at their
peril. Simply put, it’s impossible to arrive at an
accurate true-peak reading with a sample peak
limiter because the measurements are different.
What may seem like safe settings on a sample peak
limiter would not guarantee compliance.
Therefore, the best true-peak limiters offer a
true brick-wall solution, measuring inter-sample
peaks and enabling the user to define the audio
output’s true-peak limit (rather than the more tra-
ditional threshold control at which limiting begins to
take effect). Based on ITU-R BS.1770’s standardized
true-peak algorithms, these tools are suitable for
controlling audio for post-production and broadcast
applications. True-peak limiting can also be used to
ensure that downstream codecs (e.g., MP3, AAC, and
others) do not introduce distortion into the signal.
Once a mix is more or less loudness compliant,
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But you can bend the rules
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That’s exactly what you get with Celestion’s full-range, compact array drivers. The space-efficient
2”, 2.75” and 3.5” designs are perfect for close-coupling applications.
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Find out more www.celestion.com

ax Audio Praxis
Figure 6: There are
complexities involved in
reducing the dynamic
range of a feature film
while continuing to respect
dialogue clarity.
editors can use offline tools to fine tune the mix
and speed up the last part of the normalization
process. These timesaving tools can be plugged
into the editing environment to bring a mix into
line quickly, correcting any true-peak overshoots
along the way.
Batch analysis is another highly useful tool for
busy post-production operations, enabling them to
automate part of their loudness processing. Acting
as a rapid fail-safe system and internal QA compo-
nent, a batch processor can automatically assess
files for compliance and correct or reject as needed.
An Expanded Role for Loudness
Loudness measurement doesn’t have to end
when broadcast criteria are met. In regions where
loudness compliance is an accepted part of the
audio workflow, the same normalization tools can
also be employed in several areas that go beyond
technical compliance to support new and improved
Figure 7: A sample-by-
production techniques.
sample analysis shows a
True-peak over. As I mentioned, loudness consideration during
28 | February 2014 | audioxpress.com
acquisition is a significant timesaving pre-production
technique that brings audio into the editing suite
at the right time. Another area in which loudness
can play an important role is dialog clarity. Even
today, mixes are occasionally broadcast with the
background music too loud, which makes dialog
indistinct and results in viewer complaints. Using
a meter to preserve loudness separation for dia-
log above other mix components can help guard
against these mistakes. Measuring the loudness of
background music beds and FX spots can also help
maintain consistency from section to section. In
sessions that require significant complex editing,
loudness normalization can quickly match dialog
levels to a far more useful control than 0 dBFS (i.e.,
the maximum possible digital level).
Audio libraries can also benefit from loudness
normalization, ensuring that audio is always inter-
nally consistent and available at an expected level.
Consideration of the loudness range (LRA) param-
eter can also be useful when mixing material for a
specific target device.
Work Still To Be Done
Fortunately, the various international loudness
recommendations are based on the same Inter-
national Telecommunication Union (ITU) standard,
so there is general agreement within the industry
about how to approach loudness control. Even so,
some issues still need resolving.
One problem that can arise in a maturing market
relates to internal loudness jumps, which can cause
viewer irritation or discomfort. In the pressurized
advertising environment, commercial imperatives
demand every method of capturing viewer attention
is employed. Therefore, a clever mix engineer may
carefully mix a spot with a long quiet section, which
would enable a very loud burst of audio without
affecting the spot’s overall loudness compliance.
How to handle this new consumer annoyance is far
from standardized. Some regions have introduced
additional loudness constraints for commercials that
go beyond the integrated program loudness (e.g.,
maximum momentary or short-term loudness) to
avoid this exploit, but they’re walking a fine line
between loudness control and over-specification
that can trample on desirable creative freedoms.
A more technical issue is the difference between
a 5.1 mix and its corresponding downmix. It is com-
mon for the downmix to differ slightly in loudness
from the 5.1 mix. However, the difference can be in
either direction. This can further confuse the issue
and preclude a simple offset as a viable solution.
Similar situations arise with dual-language, multi-
mono stereos, in which a consumer’s television can
produce an unexpected 3-dB loudness jump depend-
ing on the configuration. Relying on the metadata
is one solution, but this only works if the metadata see an increased use of flexible alerts to identify Figure 8: The NUGEN Audio
is accurate and the appropriate device is capable potential clarity concerns based on a local set of VisLM-H provides a simple
of properly reading and responding. preferences. With awareness of the overall program standard-compliant way
to measure, compare, and
Work also needs to be done in the application loudness target, the producer could retain clarity
contrast loudness during
of loudness compliance to other audio fields. The and meet loudness targets in real-time situations
production, broadcast, and
loudness standards for broadcast were not designed that require fast and accurate decision making. post production.
with radio, game audio, music production, or film Looking further ahead, new loudness algorithms
in mind, and there is room for further research that measure the loudness of dialog in audio relative
into how these areas may benefit and what specific to the overall program loudness could be developed to
requirements may be needed. enhance the repurposing of film audio for television.
For example, loudness measures are only defined Currently, traditional limiting takes no account of the
for up to 5.1 audio, but much film and game sound level of the dialog and tends to push the dialog too
is now produced in 7.1, which is not covered by the low in the mix as the dynamic range is reduced for
ITU-R BS.1770 definition. Now that iTunes Radio has reasonable television broadcast. An algorithm that
incorporated a sound-check algorithm for level har- is aware of the dialog level would be able to com-
monization, loudness considerations during music press around this anchor and preserve the dialog
production are also coming to the forefront. level within the resulting repurposed work.

What’s Next? Workflow Changes Are Necessary

As a product designer, this is one of the most Establishing a post-production workflow that
common questions I get in a mature loudness mar- integrates loudness normalization leads to improved
ket. The answer lies in two areas since new devel- audio quality, with compliance becoming an inte-
opments are not just about the tools. First, we will grated and natural part of the creative process. The About the Author
see new applications arising from the continuing practice of loudness compliance leads to better audio Jon Schorah is the
education process and a deepening understanding consistency and enables a greater dynamic range creative director and
of existing tools. This will lead to a demand for with the introduction of more headroom resulting co-founder of NUGEN
enhancements to existing tools (e.g., warnings when in the opportunity for more creative expression. Audio, one of the world’s
leading manufacturers
loudness differentials between differing signals or Thanks to loudness normalization, there are of loudness products.
signal types become too large or too close). Multi- fewer complaints from consumers, which means Jon has a background
ple-stream loudness metering could be employed that the changes are headed in the right direction in mastering and en-
in a broadcast studio to monitor the microphone so far. Now it’s time to start refining solutions for gineering and has con-
levels during a live-panel interview. If the meters loudness control and build on the solid foundation siderable experience
in wider aspects of the
are aware of a preferred target range, they could already established. industry. A 1992 Leeds
alert the operator to potential dialog clarity issues Ultimately and ideally, loudness will become a University (UK) gradu-
if signals were to become too divergent. primary consideration during production. As tools ate, in recent years Jon
Similarly, the opposite could be applied in an OB improve, loudness parameters and transferable has focused on product
production if the crowd noise at a sporting event objective measures can be used to check whether design with a particular
interest in the usability
rises too close to the commentary level. At the audio is compliant and target-appropriate. At the and workflow aspects of
moment, most automated alerts are based around same time, audio engineers can use these tools to audio software.
compliance issues, but as products develop we will produce better-sounding, more creative mixes. ax

audioxpress.com | February 2014 | 29

www.truextent.com/audioxpress www.focal.com
Truextent is a registered trademark of Materion Brush Inc. Focal® is a trademark of Focal-JMlab®
Accurate . Fast . Revealing.
Focal relies on Truextent ® brand acoustic
beryllium for their legendary dome
tweeter products.
 ax Speakers
Super
Subwoofers
A Battle of the Titans
To achieve intense deep bass requires considerable air
movement. In audio, there is no sensory experience more intense than
profuse deep bass. If nothing else, you get your clothes dry cleaned for free! This month, we focus
on “the titans.” Throughout audio history, there have been several large to huge subwoofers at the top
are the extraordinary speakers exceeding 18” diameter.
By Although there are many transducer technologies
Mike Klasco and for sound reproduction, generating a lot of deep bass
Steve Tatarunis requires several woofers, a huge horn, or one really big
woofer. Of course, high excursion and a lot of power
(United States)
handling and sensitivity does not hurt. And while we
don’t want to disparage some of these efforts, several
of these designs are flawed and never saw much com-
mercial success. The failures were due to size, and in
some cases, questionable engineering. Yet, these giant
subwoofers are entertaining to revisit and a couple of
them do contain inspired engineering.
While good engineering does not always scale,
loudspeaker physics may. No matter what the dia-
phragm’s diameter in these giant woofers, they all
require cabinet volume to maintain efficiency at
very low frequencies. Speaker manufacturers that
make 15” and 18” woofers say the sales ratio is
approximately 10:1 in favor of the smaller woofer
because larger piston areas need larger internal
box volumes. Anything larger than an 18” woofer
becomes abnormally sized.
How Do the Big Woofers Differ?
There are no standards for speaker frames or
cones beyond 18”; however, 21” is the logical next
size increase. Frame fabrication and construction
is limited by projected sales vs. tooling costs, so
aluminum gravity-cast or sand-cast frames are
more common than aluminum, zinc, or magne-
sium alloy injection castings. Stamped steel tooling
is expensive and the likelihood of a heavy magnetic
structure warping the thinner gauge steel frames
requires some sort of metal casting. However, some
32 | February 2014 | audioxpress.com
trade shows have also displayed some “erector-set”
frames for one-off concept speakers.
For these large woofers, a 4” diameter voice
coil is barely enough, and the tooling is limited for
voice coils beyond 4”. Large-diameter voice coils
are not just needed for thermal power handling. If
the voice coil diameter remains constant and the
cone’s diameter increases, then the cone’s unsup-
ported area increases. This puts increased strain
on the cone’s strength.
For example, take a large powerful speaker
motor and a tight box then add a lot of excursion.
As the autosound subwoofer aftermarket discovered,
it results in a cone-crunching machine.
Paper cones rule the woofer world, at least for
the majority of units. For these subwoofer giants,
many of the cone processing machines have size and
material limitations. Fostex circumvented this chal-
lenge by using pie slices pieced together on its 31”
paper cone. Paper can be strong, especially with the
right additives such as aramids (Kevlar), carbon fiber,
or hemp. Some woofers use woven and non-woven
high-performance composites. Others have metal
cones. Electro-Voice even had some success using
Styrofoam on its 30W, a 30” woofer. And, recently
developed high-performance ultra-high density poly-
ethylene materials (e.g., Teijin Endumax) promise
significantly enhanced strength at lighter weights.
Large-size speakers do not always have high
excursion, but when they do there are some seri-
ous design issues with the coil centering spiders.
The large cones with big-diameter voice coils usually
result in high-moving mass, which means the spider
must keep the voice coil centered and provide at
least a portion of the restoring force. Large voice
coils, for a spider’s given outer diameter, reduce
the number of corrugations thereby reducing the
maximum excursion.
The RTR loudspeaker company put the spider
at the inside of the voice coil. Electro-Voice used
this design on its EVX1800. Superior materials for
spiders include aramids (e.g., Conex, Nomex, and
Technora). Other approaches include glass epoxy
composites and high flex-life metals such as Be-Cu
(which is about 95% copper with 5% beryllium).

History
Historically, large speakers appeared relatively
early because amplifiers were limited in power,
which made loudspeaker efficiency crucial. One
way to provide higher bass efficiency was to use a
large radiating area (i.e., a large cone).
As a young man, Rudy Bozak worked at a com-
pany called Cinaudagraph. While there, he helped
prepare a tower topped with a cluster of eight 27”
Cinaudagraph model PE-27 loudspeakers in 30”
frames with large 450-lb field coil magnets for the
1939 New York World’s Fair.
This package provided the low-frequency signals
for a two-way sound system at Flushing Meadows in
Queens, NY. An ad for the PE-27 loudspeaker claimed
a 20-to-10,000-Hz response and a 600-lb shipping
weight! The loudspeakers were mounted into bass
horns with 14’ wide mouths and were each driven
by a 500-W amplifier derived from a high-power Photo 1: Hartley Products
radio broadcast tube. Bozak went on to found Bozak continues to produce its 24”
Loudspeakers in Stamford, CT, which was famous woofer. (Photo courtesy of
for a strong bass signature sound. Hartley Products)
Hartley Products—founded in England in 1927 and
still operating today in North Carolina—was another
early large woofer contributor. The company began
developing a unique polymer cone for its own 21”
in 1956. The suspension uses a silicon rubber and
the spider shape was the old style “45-RPM” type
made from trilaminate fiberglass.
Hartley Products’s magnetic damping suspension
transformed the driver’s voice coil into a part-time
electromagnet. In the presence of an AC signal, the
coil and cone could move freely. In the absence of
a signal, a strong restorative force pulled the coil
back to its starting position. In doing so, it created
an electrodynamic damping effect. The Hartley 220
MSG is built on a sand-cast aluminum frame with
a 6-lb ferrite magnet (see Photo 1).
Electro-Voice first introduced its 30W with a 30” Photo 2: This is an early
Styrofoam cone about 50 years ago (see Photo 2). ad for Electro-Voice’s 30W
The loudspeakers matching enclosures were the loudspeaker. (Photo courtesy
corner-loaded Patrician 700 and 800. The enclosures of Electro-Voice)

audioxpress.com | February 2014 | 33

 ax Speakers
Photo 3: This promotional shot for the 64” Diatone demonstrates the speaker’s
enormity. (Photo courtesy of Mitsubishi Corp.)
were approximately the size of a large refrigerator!
The 30W contained a cast frame containing a 2.5”
diameter voice coil with an alnico-magnetic struc-
ture. It offered high sensitivity with medium power
handling, which was typical of the power amplifiers
of the time. The surround was multi-piece pleated
fabric. The 30W was also used in an early 1970s
Rickenbacker Transonic bass guitar
amplifier.
Photo 4: Fostex FW800 HS is the large-diameter super woofer for ultra-low frequency
reproduction. (Photo courtesy of Fostex)
34 | February 2014 | audioxpress.com
In the 1970s, author Mike Klasco built a pair of
RTR 25” subwoofers into a custom-made bed. The
speakers were driven by a Phase Linear 700, which
was the highest output power amplifier at the time.
While he was living in Manhattan, NY, he increased
the volume during the Pyotr Ilyich Tchaikovsky’s
1812 Overture late one evening and all the lights in
the neighboring building illuminated, soon followed
by a couple New York police officers responding to
reported cannon shots.
The RTR speakers ended up in huge scaled-up
horn enclosures in New York’s famed Infinity Club.
This subwoofer had a 6” voice coil with a spider
inside the coil, a double-roll S rubber surround, a
heavy cone, and a huge alnico slug pot structure.
Fane introduced its Colossus 24 Bass in the
1980s, and while Fane and the Colossus woofer
series continues, its 24” is no longer listed as a
standard product. The Colossus 24 Bass had a 4”
voice coil diameter and its 101-dB sensitivity was
quite good.
Mitsubishi Electric’s Diatone brand offered even
larger woofers (32” and 64”) in the 1980s. The
woofers were intended for broadcast and studio
monitoring applications (see Photo 3). One of the
largest woofers of the time was Fostex’s FW800HS,
which was a 31.5″ bass driver. The FW800HS fea-
tured distinctive seamed pleated pie slice paper
cone. Power handling was only 150 W continuous
and 450-W program, which was not very high, with
a 4” voice coil diameter (see Photo 4).
Clarion’s SRW8000 subwoofer was comparable in
size to Fostex’s FW800HS. The SRW8000 contained a
32” polypropylene cone. It was intended for the auto-
sound aftermarket “crank-it-up” contests and boasted
1,000 W (claimed) power handling (see Photo 5).
BEYMA of Spain’s contribution to the super-sized
subwoofers is the 21L50, which is a 21” subwoofer
with a power handling of 1,000-W AES and 2,000-W
program. The 21L50’s voice coil is 4.5” diameter with
a triple-roll surround, and a double spider system.
P.Audio System has offered 21” or larger speak-
ers for quite some time. Its first offering seemed
to have an inappropriate light cone and a spider
that did not have much excursion. However, future
P.Audio products appeared to have competent
engineering.
After inquiring about the speakers, P.Audio said
its first offering’s design was the “winner of a Bud-
dhist prayer call speaker contest.” P.Audio recom-
mended its large subwoofers to those interested in
pro-sound applications.
The SD21-2000N has a 21” diameter piston that
uses neodymium magnetics to achieve a high acoustic
 Photo 5: Clarion’s SRW8000
is shown mounted in the
rear of a station wagon.
(Photo courtesy of Clarion
Corp.)

output to weight ratio. The 6” diameter voice coil

has an inside/outside geometry with square wire
to improve efficiency. The SD21-2000N has an AES-
rated 2,000 W of continuous power handling and a
full 8,000 W of peak power handling. This model is
intended for vented enclosures.
Its system linearity is enhanced with flux
demodulation devices in the magnetic structure
to increase sonic accuracy. The system suspension’s
double-spider design ensures high displacement
and linearity. The cone was treated with a con-
formal coating providing damping and moisture
resistance.
The SD21-2000N’s chassis is die-cast aluminum
Photo 6: This is the Peavey Versarray 124
for good structural integrity. There are also some
Subwoofer. (Photo courtesy of Peavey Corp.)
variants available in the P.Audio line of 21” woofers
(e.g., the SD21 1800EL with 5” voice coil diameter).
Another super-sized subwoofer is MTX Audio’s
22” Super Woofer, which contains a 900-oz stron-
tium ferrite magnet. The woofer is 23” long and
weighs 320 lb. The 22” Super Woofer claims a
5,000-WRMS power handling with 10,000-W peaks.
It includes an aluminum heatsink on dual 6.5”
voice coils and enables 2.5” of linear cone move-
ment “one way.”
Finally there is Peavey Electronics’s Versarray
124 Sub, which uses a 24” driver with a 5” diameter
voice coil. The matching enclosure is a direct radia-
tor vented subwoofer 2,200-W program rating and
extra-long cone excursion with dual spiders and a
long-throw construction designed for professional
touring and high-performance permanent install.
The Versarray 124 Sub boasts continuous 1,100-W
power handling, a 2,200-W program, and 4,400-W
peak (see Photo 6).
Of course, you can always find a few more super-
sized subwoofers (e.g., designs from Precision Devices
and 18 Sound), so take some time to explore the
large woofers.
In the next few months, we will explore the
“weird science” subwoofer turf, which includes
the Danley servo-drive, Bruce Thigpin’s rotary
subwoofers, and Tymphany’ LAT. Later, we will
examine the “shakers,” which are built for haptic
structure-borne bass vibration rather than audible
airborne sound energy. ax

audioxpress.com | February 2014 | 35

 ax Sound Control
Acoustical Diffusion
and Scattering
By
Richard
Honeycutt An incredible amount of
(United States)
processing occurs in our brains when we
hear a sound. First, we compare the sound heard in
our two ears. From that comparison, we determine the direction of origin. This involves
small differences in timing, frequency response, and phase response. Then the “early
response”—the first 50 to 80 ms—is analyzed, and reflections are identified.
There will always be reflections, even in a
so-called “anechoic chamber” (although the reflec-
tions in a chamber will be very low in amplitude,
especially at high frequencies). The time gap between
the onset of the initial sound and the first reflection
is called the “initial time-delay gap.” This gap cor-
responds to the distance from the sound source to
the nearest reflecting surface, and from there to the
listener’s ears, compared to the sound’s direct path.
Figure 1 shows the early part of a decaying sound
wave with the gap marked. The brain concludes that
the acoustical space is large if the initial time-delay
gap is large. As an example, Leo Beranek studied
listener preferences and concluded that listeners
prefer halls with a gap of about 21 ms. This would
direct sound
initial time-delay gap
first reflection
Figure 1: There is an important initial time-delay gap between the direct sound and the
first reflected sound a listener hears.
36 | February 2014 | audioxpress.com
imply the reflected sound’s time-of-flight is about
24’ longer than the direct path.
The Haas Effect
In 1949, Helmut Haas published research indi-
cating that reflections occurring within 50 to 80 ms
of each other are merged by the ear-brain system
into a single acoustical event. This phenomenon is
called the “Haas Effect.”
Actually, the critical time separating sound into
multiple acoustical events depends on the type of
sound. Well-defined percussive sounds, such as
snare drum rim shots or single bongo strikes, can be
distinguished at intervals much closer than 50 ms.
The commonly mentioned 50-ms criterion applies
to speech and some types of music.
Many types of music have an integration time
closer to 80 ms. Some very legato types have much
longer times. The integration of multiple sound
arrivals into a single aural event does not mean
that later arrivals within that 50-to-80-ms window
have no aural effect.
If early reflections after the initial time-delay
gap (including early “artificial echoes” produced by
loudspeakers close to the listener) are sufficiently
strong, they can confuse the sense of aural localiza-
tion, making the sound source seem to be located
somewhere besides its actual location. Such strong
early off-axis reflections can also cause music and
speech to sound unnatural. These detrimental effects
only occur with discrete early echoes. If the off-axis
echoes are diffuse (i.e., spread out in time), they add
to the sense of spaciousness and apparent source
width. Classical-music audiences consider spacious-
ness desirable, although it must be balanced so the
a) b)
Incident ray Incident ray
55º 55º
Reflected ray
ratio of direct to reverberant sound is high enough
to yield good clarity, definition, and intimacy.
The bottom line is that, while Professor Wallace
Clement Sabine of the Harvard University physics
faculty was correct in identifying the importance of
reverberation time (RT) in determining the sound of
an auditorium, it is also true that not only the late
reverberation (which determines the RT) but also
the early reverberant sound play important roles
in concert or lecture hall acoustics. The direction
from which the early reflections arrive, as well as
their diffusiveness, significantly affect the listening
experience.
Scattering and Diffusion
Two technical terms have come into use in
acoustics in recent years: “scattering” and “diffu-
sion.” These terms describe similar but not iden-
tical phenomena.
Scattering is the redirection of sound by specular
(light-like) reflection from the surface of the scat-
terer. It is often considered as though it involved
random or nonspecular reflections, since the many
tiny scattering surfaces of a material make for a
situation too complex to analyze except as an aggre-
gate effect. But in fact, each reflection is specular.
Diffusion involves wave interactions (diffraction) to
spread the reflected sound.
Figure 2 shows the difference between scattering
and diffusion. Figure 2a shows specular reflection
from a plane surface. Figure 2b shows diffusion from
a rough surface. Notice that the rays representing
reflected waves are spread out. Figure 2c shows
redirection or scattering of incident rays striking
the angled surfaces of a pyramidal diffuser and
diffusion of rays striking the peak of the pyramid.
The redirection provided by a diffuser changes
the average direction of the reflected rays, while
diffusion only adds rays in more or less the same
direction as the main specular reflection. An excel-
lent illustrated discussion of scattering and diffu-
sion is available from RPG Diffusor Systems (see
Resources).
Historically, excellent concert halls had good
scattering and diffusion provided by architectural
features such as deeply coffered and ornamented
ceilings, crystal chandeliers, niches, cornices, side-
wall balconies, columns, and statues. The Boston
Symphony Hall, designed by Sabine, benefits from
diffusion provided by 18 statues placed in niches
above the second balcony.
c) Ray scattered
and diffused
Incident ray by pyramidal Reflected ray
55º 55º diffuser
Reflected ray
Gold lines are
diffuse rays
Less architecturally complex (and also less Figure 2: Scattering and
diffusion are not the same.
expensive) means to provide diffusion include geo-
Variations include specular
metrical diffusers of various kinds. Among them are
reflection from a plane
barrel diffusers and pyramidal diffusers, which have surface (a), diffusion from
been available since the early 20 th century. Barrel a rough surface (b), and
diffusers look like sections sliced from a barrel and redirection or scattering of
provide 1-D diffusion in the plane perpendicular to incident rays (c).
the axis (see Photo 1). Pyramidal diffusers provide
2-D diffusion (see Photo 2). Some diffusers have
a gentle curvature in either a cloud pattern or in
repeating suspended-ceiling tiles.
Quadratic Residue Diffusers
There is nothing magical about cylindrical or
pyramidal shapes for diffusers. Many different
shapes have been utilized, including the one shown
in Photo 3. This product is one of many that pro-
vide absorption as well as diffusion.
In 1983, RPG introduced a new type of diffuser
based on slots (for 1-D diffusers) or square wells
(for 2-D diffusers) with the depth of each slot or
well chosen according to the branch of number
theory called “quadratic residues.” These diffusers
are also called refraction phase gratings (RPGs).
These quadratic residue diffusers (QRDs) operate
on the basis of constructive and destructive inter-
ference between sound reflecting from adjacent
wells. The relative phase of adjacent reflections
depends on the well depth.
QRD variations are available from several
Photo 1: This barrel diffuser
redirects sound that strikes
it, reducing flutter echo.
(Photo courtesy of Acoustic
Surfaces, Inc.)
audioxpress.com | February 2014 | 37
 a x7052PH
Sound Control
Phantom Powered
Measurement Mic System
NOW 4Hz to 25+kHz Photo 2: These pyramidal diffusers
redirect sound in two dimensions.
<16dBA >135dBSPL (Photo courtesy of RPG, Inc.)
IEC61094-4
Compliant 1dB/div
30 kHz

manufacturers. Photo 4 shows a

wide-bandwidth wooden diffuser
used in a music rehearsal room.
For a diffuser to be effective at
lower frequencies, the dimensions
of the diffuser and its wells or
slots are crucial. Therefor, it is
important for anyone specifying
Anechoic diffusers to carefully examine the
Free Field frequency range through which
it is designed to operate. rays spread in all directions. Some man-
Response ufacturers show the polar pattern formed
Included Scattering and Diffusion by the reflected rays.
Standards Although there are a few ISO standards
This brings up the issue of how scatter- (ISO 17497-1 and-2) for specifying scat-
ing and diffusion are now specified. Both tering and diffusion, there is not uniform
are stated in terms of specific coefficients. agreement that the resulting measurements
The scattering coefficient is the ratio of the are accurate, or that they measure what we
Rugged acoustic energy scattered from a surface think we are measuring.
STAINLESS to the total reflected energy (specular plus In fact, some manufacturers and mea-
Body scattered. The diffusion coefficient expresses surement specialists question whether cer-
Operational the evenness with which the reflected sound tain classes of diffusers marketed by some
<-20 to >70C
Storage
<-30 to >85C
Phantom
<18Vdc
to
>56Vdc

Removable
Capsule
Titanium
Diaphragm

www.acopacific.com Photo 3: This product

combines diffusion and
absorption. (Photo courtesy
of Jocavi, Inc.)

38 | February 2014 | audioxpress.com

well-known companies even provide effective dif-
fusion. Needless to say, this leads to confusion for
specifiers.
As with all international standards, ISO 17497-1
and -2 undergo periodic review, and issues with pres-
ent measuring methods and specification format are
being examined. Since adherence to standards is
completely voluntary on the part of each manufac-
turer, we can hope that future revisions of these
standards will encourage more widespread use of the
new standards, provide a more uniform specification
format, and make it easier to compare products.

Acoustical Diffusion
In addition to providing a desirable quality of
early reflections, acoustical diffusion is often used
to eliminate flutter echo in rectangular rooms, and
slap echo from rear walls or balcony faces. When Photo 4: This quadratic residue diffuser is constructed of wood. (Photo courtesy of RPG, Inc.)
using diffusion for these purposes, the acoustician
should remember that diffusers do not get rid of
sound; they only redirect it.
For example, in an auditorium or worship center Resource
in which echoes from the balcony face inhibit speech RPG Diffusor Systems, Inc., www.rpginc.com/pdfs/news/ScatteringvsDiffusion.pdf.
intelligibility for those on the stage or platform, dif-
fusion can be used to redirect the reflected sound
to another place. The question becomes, “which
other place?”
If the reflected sound is redirected upward, it
can then reflect from the wall behind the stage or
platform, producing objectionable echoes for people
seated in the front balcony rows. If the reflections
are redirected downward, the result may be that
those in the front rows of the main floor will be
affected by objectionable echoes or comb-filtering
effects. If the sound is redirected to the side walls,
the result can be “round-robin” reflections causing
late echoes at numerous places in the room. This
is not to say that diffusion should not be used to
deal with balcony-face reflections, just that such a
treatment must be carefully engineered.
Another use for diffusion is to help projection
of unamplified sound from the stage to the seat-
ing areas. In many new auditoriums, the ceiling is
carefully shaped to provide helpful early reflections.
While this approach works well in new construction,
it is difficult to retrofit to existing rooms. Appropriate
use of diffusion can redirect sound from the stage
out to the audience. The diffusive materials can be
directly attached to the ceiling, or applied as clouds.
Although the engineering use of scattering and
diffusion for acoustical design is a fairly young sci-
ence, it provides an important tool for improving
the listening experience and speech intelligibility in
a variety of acoustical spaces. ax

audioxpress.com | February 2014 | 39

 ax Questions & Answers
The Original
By
Sound Designer
Shannon Becker
(United States) An Interview with Dan
Dugan—Audio Engineer,
Inventor, and Nature
Sounds Recordist

SHANNON BECKER: When and how did you first become

interested in audio electronics?

DAN DUGAN: As a child! I was most interested in theater

lighting. I was raised in San Diego, CA, and when my parents
took me to the Old Globe Theatre or the summer musicals
in the Ford Bowl, I always wanted to go backstage to see
the light board. In grade school, I operated the projectors,
the tape recorders (Wollensak and Revere), and the sound
systems (Bogen).

SHANNON: When did you attempt your first audio project?

DAN: In grade school, I remember making up a program on tape. Something historical, but I can’t
remember what it was about.

SHANNON: Describe some of the jobs you had prior to inventing the automatic microphone mixer.

DAN: After doing all the lighting for four years at the University of San Francisco (USF) College Players
and for concerts in the USF Gym, I did sound for the Globe Theatre in 1964 and lighting and sound in
1965, and lighting for the first production of the San Diego Opera in 1965. In 1967, I switched to doing
theater sound, working for the San Diego National Shakespeare Festival and the American Conservatory
Theatre in San Francisco.
[The title “Sound Designer” was created in 1968 to describe what Dan was doing. He provided sound
services for many seasons of the Mondavi Jazz Festival, and engineered several independent record
albums, including Kate Wolf’s first two albums which are still in print, now as CDs].

SHANNON: Describe what the term “sound designer” means to you.

Photo 1: Dan Dugan was DAN: In theaters a “sound designer” supervises the sound from the microphones to the audience’s ears.
the first person in regional In motion picture production there are two meanings. The first is the same as in theater, also called
theater to be called a “sound
supervising sound editor, and the second usage is for a person who creates novel sounds like monsters.
designer.” He also developed
the first effective automatic
microphone mixer—the
SHANNON: How did you come up with the idea for the automatic microphone mixer?
automixer. He is shown here
with his museum rack of DAN: In 1968, I did sound design for the resident companies of Hair in Chicago, Las Vegas, and Toronto.
Dugan automatic mixers. There were 36 microphones and one operator working rotary-knob mixers in a rack. I thought there

40 | February 2014 | www.audioxpress.com

had to be a way to help. I experimented for about
six years and hit on a solution.

SHANNON: Tell us about some of your other inven-

tions. Which is the most popular? Are any currently
in production?

DAN: The Dugan Speech System is my most popular

invention. It is 40 years old and still finding new
applications. There’s also the Dugan Music System,
a distant second, and Dugan Gain Limiting. In lim-
ited use but with more coming soon is the Dugan
Automatic Level Control. Unrealized as of yet but in
the wings are Dugan Foldback Limiting and a Dugan
Speech Equalizer.

SHANNON: Tell us about “A New Music and Sound

Effects System for Theatrical Products,” which
is the sound design paper you presented to the Photo 2: Dugan was raised in San Diego, CA, where his parents took him to the Old Globe
Audio Engineering Society (AES) at its 37th Con- Theatre. Here he is pictured editing sound in a dressing room at the Old Globe Theatre in
1978.
vention. Did you realize its future implications
when you wrote it?

DAN: In the paper, I described a system in which

the signals from three stereo tape players were
routed to 10 loudspeaker zones in the theater. Audio
mixing boards generally combine a large number
of inputs to a small number of outputs—that’s mix-
ing. For playback of theater cues, the opposite was
desired, routing a small number of channels over a
large number of speakers. As there was nothing like
that available, I designed and built a system from
scratch. It was the first multi-scene preset board
for theatrical cues playback, sending three stereo
tape decks to ten speaker channels. And I described
my work in that paper. Subsequently, Charles Rich-
mond, of Richmond Sound Design, designed prod-
ucts developing the concept further.

SHANNON: Your patented equipment has been used

in thousands of places, including the courtroom
where Saddam Hussein’s trial took place and on
the David Letterman Show. Can you share other
locations where your equipment may be found?

DAN: My equipment is used in corporate meetings

everywhere, from ESPN sports to PAC-12 sports to
US Presidential debates and on several television
set locations including Washington Week and PBS
News Hour.
SHANNON: Your San Francisco, CA-based company
Dan Dugan Sound Design (www.dandugan.com)
produces automixing solutions. Are you currently
developing any new products?
Photo 3: One of Dugan’s most popular products is the Dugan-MY16, a 16-channel
automatic mixing controller that plugs into a slot on Yamaha consoles. The controller
enables sound engineers to manage multiple live microphones without continually riding
individual faders. The Dugan-MY16 automatically detects the active microphones and
makes fast, transparent cross-fades without the distracting sonic artifacts common to
noise gates. It tracks unscripted dialogue and maintains consistent system gain for up to
16 open microphones.

www.audioxpress.com | February 2014 | 41

 ax Questions & Answers
Photo 4: The Dugan Model E-2 automatic mixing controller is used with multiple live
microphones. This updated unit replaces the Dugan Model D-2 as the company’s top-
of-the-line automatic mixing controller with analog I/O and is useful for users who are
working in tight spaces or who need portability in their analog Dugan system.
Photo 5: The Dugan-VN16 is a soon-to-be-released option card for Avid live mixers.
Photo 6: Dugan records ice falls at Upper Yosemite Falls in Yosemite Falls National Park,
Mariposa, CA.
42 | February 2014 | www.audioxpress.com
DAN: We recently added the Model E-2 to complete
the E-series (E-1A, E-2, E-3). We are also just about
to ship the Dugan-VN16, an option board for Avid
live sound mixers.
Next out for our company will be a new physi-
cal control panel for Dugan automixers. It can be
used when you are working under pressure and real
knobs and buttons are better than mouse clicks.
SHANNON: To what do your attribute your com-
pany’s continuing success?
DAN: Persistence, good luck.
SHANNON: You are known for your use of natural
sound recordings. When and why did you first
begin capturing the sounds of nature?
DAN: I was the Northern California service shop for
Nagra Audio. Around 1987 or 1988 one of the found-
ers of the Nature Sounds Society worked at the Oak-
land Museum and he brought a Nagra recorder in
for service. He mentioned that every summer they
had a camp in the Sierras and invited me to come. I
started mentoring with the Nature Sounds Society,
teaching people how to get the best sound from their
equipment. I started recording for myself at the end
of 2001 when I took a borrowed MiniDisc recorder
for a trip to New Zealand and I recorded an album’s
worth of good stuff.
SHANNON: Where do you conduct your outdoor
recordings?
DAN: One of my favorite locations is Muir Woods
[National Monument in Mill Valley, CA] because it’s
so accessible. I also enjoy recording in Mariposa
Grove in Yosemite National Park because it is sub-
lime and at Joshua Tree National Park [in south-
eastern California].
SHANNON: What do you see as some of the great-
est audio innovations of your time?
DAN: “Of my time” meaning in my career? I think
there are several including solid-state electronics,
integrated circuits, electret condenser microphones,
and digital audio.
SHANNON: Would you recommend any promising
technologies to audioXpress readers?
DAN: Audio over Ethernet.
 Seeking acoustical design
projects for listening rooms,
music venues,
and beyond!

Completing the successful acoustical design of a space—

whether it’s an 8' × 10' home listening room, an intimate 200-seat music hall,
or a 10,000' club—is one part art and one part engineering.

Acoustical Design Projects

Have you completed such a project? Do you have a listening room decked out with absorptive
and reflective wall panels, baffles, banners, and geometric diffusers? Have you planned and equipped
a small venue for optimum listening pleasure?

The staff at audioXpress wants to know! Share pictures and descriptions

of your designs.

Submit your photos and descriptions

via e-mail to editor@audioxpress.com
for a chance to have your
design featured in audioXpress.
Please include “AX SPACE ACOUSTICS”
in the subject line.
 You Can DIY!

The Ultimate Living

Room Home Theater
You can combine high-tech audio devices and acoustic knowledge to
By turn your living room into a home theater. This article describes how to
Ethan Winer acoustically treat a room and explains the importance of good acoustics.
(United States)

Photo 1: The home theater
system boasts a 159” screen
and plenty of acoustic
treatment to let music and
movie soundtracks appear
crystal clear.
In some ways, my living room home theater
looks more high-tech than some of the million-dollar
home theaters you see in high-end designer pho-
tographs (see Photo 1). And, it definitely sounds
like a million bucks. To replicate the sound, you
need to understand how I have acoustically treated
this room and why having good acoustics affects
the sound quality more than the type of receiver
or even which loudspeakers you use. Understand
that all acoustic problems are caused by reflections
from the walls, floor, and ceiling. However, there
are several types of acoustic problems, and each
one requires a different solution.
Acoustic Reverberation
In a large space (e.g., a gymnasium) the main
acoustic problem is excess reverberation. If you clap
your hands or yell, the sound
may continue for 5 s
or longer. But,
home-
sized
rooms suffer more from bass peaks and nulls that
cause music to sound boomy or thin and individual
“early” reflections that mask musical details and
make movie dialogue difficult to understand. Both
of these problems can be solved using absorbers
made from rigid fiberglass or similar materials.
I’ll address bass problems first because they’re
more difficult. Excess ambience and reflections
at mid and high frequencies are easily tamed
using relatively thin absorbers, but low
frequencies require bass traps that
are much larger and thicker.
F i g u r e 1 s h ow s t h e
low-frequency response
measured in a small
listening room
before and
after
 Figure 1: I measured the
low-frequency response in a
16’ × 11.5’ × 8’ room, with
adding bass traps. This highly skewed response is and without bass traps. It
not only common, but typical. If you tried to sell should be obvious which
trace is with the traps and
an amplifier or loudspeaker with a response this
which is without!
skewed, you would be laughed out of business.
Yet, many people have no idea their rooms impart
similar responses—regardless of the sound. In an
attempt to document the responses, I used Room
EQ Wizard (REW) acoustics analysis software. This
Figure 2: Derived from
excellent freeware program is available for Win-
the same “before” data as
dows, Mac OS, and Linux.
Figure 1, this plot shows
Generally speaking, the more bass traps you each peak’s decay time as
have, the closer you’ll get to a flat response. It’s well as its amplitude. The
impossible to make any domestic-size room per- numbers along the upper
fectly flat, but reducing the span between peaks right show the decay times
and nulls to 10 dB or less greatly improves bass in milliseconds.
fullness and clarity.
The best locations for bass traps are a room’s
corners because that’s where bass waves tend to
collect. Note that a rectangle room has 12 corners, Figure 3: After adding
not just four where each wall meets another wall. bass traps, the response is
Bass traps can also be placed in the corners at the much flatter and the peak
wall’s top where it meets the ceiling or in the floor decay times are also greatly
corners at the bottom of each wall. Photo 1 shows reduced making the bass
instruments sound tighter
traps in the wall-ceiling corners. There is also a
and clearer.
smaller trap in the wall-floor corner to the right of
the right speaker. Another floor trap is just visible
behind the receiver and satellite TV box.
All rooms have peaks and nulls, so a room can
sound boomy and thin at the same time, depend-
ing on what musical notes happen to be playing. a distinct echo. Most loudspeakers are designed
Low-frequency peaks in a room make some bass to disperse sound horizontally over a wide range
notes sound too loud, and nulls make other notes to fill the entire room with sound. Therefore, a
sound too soft. fair amount of sound reaches the side walls and
Another problem, “modal ringing” is just as dam- reflects toward the listening spot. Since the reflec-
aging. If you’ve ever clapped your hands in an empty tions travel farther, they arrive at your ears a few
room, you have probably heard the resonant “boing” milliseconds after the direct sound. These reflec-
sound as the waves bounce repeatedly between tions reduce clarity and harm imaging because the
opposing surfaces until they diminish after a few same sound arrives from two different locations
seconds. The same thing happens at low frequen- at different times. The ceiling is another source of
cies, although you won’t hear bass resonance with early reflections. The three panels shown at the top
hand claps. Low frequencies, whose wavelengths of Photo 1 avoid reflections from the left, center,
correspond to the room’s dimensions, resonate and and right loudspeakers.
linger after the source sound has stopped. Figure 2 The rear of my living room has yet more bass
and Figure 3 are “waterfall plots” that demonstrate traps, as well as diffusers to scatter sound rather
where the peak decays come forward over time. than reflect it straight back to the seating area About the Author
(see Photo 2). Most people make all the acoustic Ethan Winer is co-owner
Bass Traps treatment in a room the same color. However, my of RealTraps, an acoustic
Bass traps work well in any room corner, though living room also occasionally serves as a product treatment company based
damaging reflections at mid and high frequen- showroom, so I have a mix of colors in this room, in New Milford, CT. His
late s t b o o k is titled
cies occur at specific locations. Referring again to as well as off-white and gray (not shown) in my The Audio Expert. You
Photo 1, the panels in the foreground at left and home recording studio, which is located in another can communicate with
right are placed to absorb early reflections from part of the house. Ethan Winer by visiting
the side walls. Reflections that arrive within about There are a total of 55 acoustic panels in the www.realtraps.com and
20 ms of the direct sound are considered “early” living room. However, many people are not willing www.ethanwiner.com.
because the delay is too short to be perceived as to install that many acoustic panels in one room.

audioxpress.com | February 2014 | 45

ax You Can DIY!
Photo 2: The rear of the
living room is treated
equally well with bass traps
in the wall-wall, wall-ceiling,
and wall-floor corners (just
visible under the outer
two diffusers). Diffusers
on the wall and in front
of the fireplace scatter
sound horizontally to avoid
sending reflections directly
toward listeners seated on
the couch. The fireplace is
blocked and not used, so
another bass trap, which is
just visible, sits on the floor
behind those two diffusers.
Photo 3: This planter is a
bass trap in disguise!
Photo 4: Additional
absorbers under the ceiling
peak avoid the focusing
effect that otherwise occurs.
46 | February 2014 | audioxpress.com
In hi-fi terms, this is sometimes referred to as the
“Spouse Acceptance Factor.”
Acoustic panels come in many shapes and sizes.
They can be painted with watercolors, hidden behind
wall tapestries, or even disguised as something else
entirely. The planter in Photo 3 is actually a bass
trap. If you look closely at Photo 1 you’ll see this
trap on the floor in the front-right corner. The good
news for people who consider sound quality more
important than appearance is the growing popular-
ity of dedicated listening rooms and theaters. Hav-
ing a spare room or finished area in the basement
enables you to do what’s best sound-wise, without
as much concern for the room’s appearance.
My living room is 25’ long (front to back), 16’
wide, and the ceiling rises from 8’ high at the
front and rear to a peak 11’ in the center. A ceiling
peak is also important to treat because the angled
shape focuses sound to the area directly under-
neath the peak. Focusing is the opposite of diffu-
sion, and it directs instead of scatters reflections.
Photo 4 shows the panels hanging under the ceil-
ing peak to avoid focusing sound toward listeners
seated on the couch.
You may think that with this much acoustic treat-
ment my living room sounds dead and unnatural.
I assure you it does not! Ideally, bass traps should
absorb a range of bass frequencies, but absorb
less at mid and high frequencies. This lets you put
enough bass traps in a room to really clean up the
low end, without making the room dead sounding,
which can happen with plain rigid fiberglass or
acoustic foam. However, absorbers at the side wall
and ceiling reflection points should be broadband
to absorb all frequencies.
My room has a mix of bass traps, broadband
absorbers, and diffusers from my company Real-
Traps. The bass traps have a semi-reflective mem-
brane behind the front fabric, which increases
absorption at low frequencies by about 50% while
reducing absorption at mid and high frequencies.
Electronic Gear
After all this talk about acoustics, I’ll mention
that the electronic gear’s frequency response, noise,
and distortion affects fidelity far less than the acous-
tics of even a good room. In other words, I consider
my electronic gear to be sufficiently “transparent”
even though it’s not particularly expensive. I have
a modest Pioneer receiver and basic Sony Blu-ray
player, with Mackie HR624-powered studio monitor
speakers and an SVS PB12-Ultra/2 subwoofer with
two 12” drivers in a ported box. You can see the
subwoofer in Photo 1’s front left corner. I also have
a Dell laptop with a Presonus FireBox sound card
that I use to listen to music files and mix original
music in 5.1 surround. ax
Resource
Real Traps, LLC, http://realtraps.com.
Source
Room EQ Wizard (REW) acoustics analysis
software
Home Theater Shack | www.hometheatershack.com
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ax You Can DIY!

Vacuum Tube
Low-Frequency
Oscillator
Why build a vacuum tube oscillator when, with a few
hundred bucks, you can buy a semi-professional piece of
gear that performs better and contains a larger frequency
span? The answer is simple. It is fun and challenging to
replicate state-of-the-art technology from the 1950s. The
project is also a great way to really appreciate those who
made historic instrumentation technology.

By
Atto Rinaldo
(Italy)
The oscillator described here resembles a 1950
HP-200CD, which is a Hewlett-Packard (H-P) design.
I have implemented several modifications to make
it easier to build with modern components. I did so
while maintaining the vacuum tube technology that
characterizes most of my projects.
I had the opportunity to use the HP-200AB at
school. At the time, it was an outstanding piece of
hardware and superior to any other test instrument
available on the market. While the HP-200AB was
limited to 20 Hz to 40 kHz, the HP-200CD had a fre-
quency range from 5 Hz to 600 kHz in five ranges.
Design Changes
When building my own vacuum tube oscillator,
I made a few changes and added some features
to the original design. I replaced the frequen-
cy-controlled oscillator components with vari-
able resistors and a fixed capacitor. The original
used a variable capacitor and fixed resistors.
I eliminated the output transformers, mostly
because they are impossible to find. I replaced
the tubes with ones that are currently available.
I changed the output to make it unbalanced as
opposed to balanced. I added an output voltage
measurement device, a 10-step attenuator, and
a frequency counter. I also used solid-state reg-
ulated power supplies.

Figure 1: Here is the block diagram of my modified Hewlett-Packard 200CD oscillator.
Theory of Operation
The circuit includes a frequency-controlling
bridge and a balanced push-pull amplifier, which
form the oscillator circuit (see Figure 1).
The frequency and amplitude controlling cir-
cuits are arranged as a floating bridge. They are
symmetrical with respect to ground. This feature
ensures frequency stability, constant amplitude,
and high reliability. The bridge is fed by the bal-
anced voltage developed at the cathode of V2 and
V4 (see Figure 2).
The output of the bridge’s frequency-controlling
branch is applied to the grid of V3 and the output
of the amplitude branch is applied to the grid of

48 | February 2014 | audioxpress.com

 Figure 2: The oscillator’s
design includes the power
supply schematic and the
component values.

V1. Table 1 shows the oscillator’s frequency range The first condition is relatively easy to achieve.
with the components value’s formula. However, the second condition is slightly more dif-
Potentiometer P3 adjusts the bridge’s ampli- ficult to meet. It requires an automated control
tude-stabilizing branch, while lamp L1 stabilizes the mechanism on the feedback network by the lamp
oscillation amplitude across the entire frequency L1, which acts as a gain stabilizer.
About the Author
span. Potentiometers P1 and P2 vary the frequency To simplify the concept, if the signal voltage Atto Rinaldo retired
from IBM after 32 years.
within the ranges set by the value of capacitors across lamp L1 increases, (the oscillator tends to
While there, he held var-
C5 to C12. go toward saturation mode) the resistance of its ious managerial respon-
The balanced push-pull circuit includes V1 and filament will increase. In turn, this will make the sibilities in Italy, the
V3 as the true amplifier and V2 and V4 as the output V1 grid more negative decreasing its gain and vice- US, and other parts of
cathode follower. A criss-cross positive feedback is versa if the oscillator signal tends to decrease. For the world. He received
his Radio and Television
applied to keep output impedance as low as pos- more information about the concept, visit the Clifton
Te c h n i q u e s d i p l o m a
sible as seen by the cathode-to-cathode load. The Laboratories website (see Resources). in 1957 when vacuum
feedback paths travel from the plate of V2 to the tubes were the primary
grids (i.e., the control and the screen) of V4 and technology. Atto wrote
Frequency Range three books in Italian
from the plate of V4 to the grids of V2.
Valvole e dintorni (which
The oscillation is maintained through a positive 10 to 100 Hz According to: translates to “The Rise
feedback from its output to the input stage. Two a n d Fa l l o f Va c u u m
1
conditions are required to maintain this oscilla- 100 Hz to 1 kHz f = Tube Technology”) and
2π RC
tion. First, the feedback signal must be in phase Valvole e dintorni HI-
with the input signal to sustain oscillations. Second, 1 to 10 kHz With: FI (which translates to
R in Ohms “High Fidelity at Home”)
the system’s net gain at the oscillation frequency C in Farads
10 to 100 kHz and Il Fascino del Violi-
must be equal to one. If it drops below one, the no (which translates to
oscillation stops. If it is higher, the oscillator goes Table 1: The oscillator’s frequency range with the “The Fascinating Violin.”
into saturation. component value’s formula are shown.

audioxpress.com | February 2014 | 49

ax You Can DIY!

Figure 3: To construct the

oscillator, you will need the
attenuator and metering
schematic.
From output 1 (out 1), a level control P4 sets
the input voltage to the attenuator at a value of
10 V RMSf, while, with P5, the meter M indi-
cator should be set to a reference level—mid
scale or full scale (see Figure 3). The attenuator
steps down the voltage from 10 V to 100 µV in
11 steps under a constant 600-Ω load, from the
1-V position down.
The oscillator’s second output (out 2) is applied
to LM 393, a comparator IC, to generate a square
wave signal to properly drive a solid-state frequency
counter and to a BNC connector to provide a sync
signal for a scope (on a high impedance). The fre-
quency counter is a ready-made solid-state commer-
cial device. Two power supplies provide the required
voltage.
A simple high voltage power supply, which is
regulated via multiple Zener diodes and a power
MOSFET, provides a 325-V output which, by means
of R3 drop resistor, turns into 190 and –135 V to
feed the vacuum tubes (see Figure 4). A three-
pin solid-state LM7805 regulator generates the
5 V required by the frequency counter and the
LM393 IC.

Figure 4: The high-voltage power supply details and components are shown.
Construction Tips
The oscillator’s construction is not recommended
for people who lack electronic assembly experience.
During a debug operation it may be necessary to
perform some troubleshooting, which is best done
by someone with electronics experience.
Photo 1 shows the oscillator’s front panel.
Photo 2 shows the internal layout and the point-
to-point wiring.
I built my oscillator on a Ballantine vacuum-tube
voltmeter enclosure. Its construction is rather com-
plex and requires the use of some critical compo-
nents to achieve maximum performance. Specif-
ically, potentiometers P1 and P2 must have a 1%
tolerance or better. This is necessary to get an
undistorted and stable waveform. I chose a ganged
10 + 10 kΩ linear type from Burns. Ideally, they
should be of antilogarithmic curve for better scale

50 | February 2014 | audioxpress.com

linearity. However I was not able to
find them. Maybe they do not exist. I
do not recommend carbon or plastic
potentiometers for this project due
to their poor tolerance.
The original HP-200CD oscil-
lator uses a variable capacitor
and fixed resistors. However, this
design requires the use of high-
value (50 MΩ), strict tolerance
high-stability resistors, which
are difficult to find. Using a vari-
able resistor and fixed capacitors
simplifies the parts procurement
task and enables you to achieve
the same quality results.
All capacitors associated with
the bridge oscillator (C5-C12) must
have 1% tolerance. Matched cou-
ples may be a good solution for
those who have a suitable capaci-
meter to measure them. The
value I chose for this the oscilla-
Photo 1: The front panel contains all the readouts
tor ranges continuously from 7 to required to operate the oscillator.
100,000 Hz in four steps. DIYers
may experiment with different val-
ues for P1 and P2 and C5 to C12, to get different frequencies. However, potentiometer
values above 25 kΩ, may introduce excessive spurious capacitance and/or inductance.

a) b)

Photo 2: The components layout and point-to-point wiring are visible in this top view (a) and bottom
view (b).

audioxpress.com | February 2014 | 51

ax You Can DIY!

Frequency Range 10 Hz to 100 kHz in four ranges range, particularly when switching from one range
to the next, select C8 and C12 with 2,100 pf and
Accuracy ±1 digit (frequency counter tolerance) parallel them with a 0-to-50-pf trim cap. You must
use caution during the components layout pro-
Signal level Monitored by meter
cess to avoid performance degradation over the
Output Maximum 10 V 70-to-100-kHz range due to stray capacitance.
Lamp L1 is critical. I used a 220-V, 3-W filament
3 to 10 V (on >10-kΩ load) lamp, but you could experiment with other lamps.
1 V–300–100–30–10-3–1 mV (Internal 600-Ω load)
Attenuator (±5%) Just adjust P3 for minimum distortion every time
300–100 µV (600-Ω load)
From 1 V down, adjustable by P6, uncalibrated lamp L1 is changed.

Sync BNC output to trigger a scope (50 kΩ) Performance

Less than 0.5% from 10 Hz to 50 kHz (–60 dB)
Distortion (see HP spectrum analyzer Photo 3)
Less than 1% from 50 kHz upward
Table 2: Specifications are listed for my updated Hewlett-Packard 200CD oscillator.
If this occurs, the oscillator’s performance would
significantly deteriorate.
To achieve better accuracy on the 10-to-100-kHz
My oscillator design exhibits performances that
are similar to H-P’s original oscillator. However, it
is limited to a 10-Hz-to-100-kHz frequency (see
Photo 3). Table 2 details the specifications for
my modified HP-200CD oscillator.
This oscillator design combines vacuum-tube
technology with current components. Many advanced
DIYers may fnd it a challenging and enjoyable proj-
ect. Once it is built, the oscillator is will also be quite
useful. ax

c)
a)

b)
Photo 3: The Hewlett-Packard 3585A spectrum analyzer waveform is
shown at 100 Hz (a), 1,000 Hz (b), and 10 kHz (c).

Resources
Clifton Laboratories, ”Bill Hewlett and His Magic Lamp,“
www.cliftonlaboratories.com/Bill%20Hewlett%20and%20
his%20Magic%20Lamp.htm.

Hewlett-Packard, “200CD Wide-Range Oscillator Operating

and Service Manual,” 1955, www.hparchive.com/Manuals/
HP-200CD-Manual-SNP_605.PDF.

52 | February 2014 | audioxpress.com

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audioxpress.com | February 2014 | 53
ax Audio Electronics

Tube Amplifier
High-Voltage
Delay
Tube amplifiers need some time for
the heaters to warm before a tube
starts conducting. Depending on
the tube type, this can range from
a dozen seconds to a minute. This
delay is not typically a problem—you
just wait a little bit before your music
starts playing.

But normally the tube’s high-voltage supply comes on at the same time as the heater
voltage, often supplied by the same transformer. That means the tube sits there for up to
a minute with high-voltage applied and no conduction. This can appreciably decrease tube
life. Again, it depends on the tube type and the actual high voltage, but the effect is there.
By It is compounded by the fact that the high voltage in the initial, unloaded situation can have a
Jan Didden much higher value than when the amplifier is operating and the high-voltage supply is loaded down. To
preserve the tube life, a device should delay the high voltage until after the tube heaters have warmed.
(The Netherlands)
There are many ways this can be done. The simplest method is probably to use a delayed relay that
switches on the high voltage after a preset time delay. This can be achieved with a discrete or integrated
timer circuit to activate the relay. You can also buy a timer unit that includes a relay and an integrated
delay you can set with a dial. But I don’t like relatively bulky and mechanical systems with their power
dissipation and mechanical failure modes. So, I thought there must be a more elegant and more reli-
able way to do this.
To solve my dilemma, I built a power supply unit (PSU) delay. However, I had a few additional require-
ments for the device including a solid-state switch used for noiseless and long-life operation, low power
dissipation, a programmable delay time, easy integration even in existing tube amplifiers, and no impact
on amplifier audio performance.
Figure 1 shows the topology I selected for the last two requirements. By inserting the switch in the
return line to the power transformer, I left everything after the first reservoir cap unchanged so there is
no impact on the power supply quality and no changes in any carefully laid out ground circuitry. It can
be used either with a bridge-type rectifier or a double-phase rectifier as shown.
The actual switch is a MOSFET connected between D and S. The unit also contains a time-delay cir-
cuit. The switch and the delay circuits are powered by a spare heater winding on the power transformer.
Figure 2 shows the complete circuit.
The spare heater voltage on J3 and J4 is followed by a voltage doubler to sufficiently raise the volt-
age for the control circuit and MOSFET gate drive. The control circuit is simple but you need to examine
it a bit closer to see how it works.
Let’s start with the switch Q1, which is an N-channel high-voltage MOSFET. The drain is connected to
the transformer return line at J1 (see Figure 1). The source at J2 is connected to ground. If the switch
is closed, the transformer’s secondary return is connected to ground and the high voltage is applied to
the amplifier.

54 | February 2014 | audioxpress.com

 The
Convenient
All-in-One
Solution
for Custom-
Designed Front Panels
Figure 1: The high-voltage switching topology shows the switch inserted in the return line to the & Enclosures
power transformer.

FREE
Software
On the right you see microcontroller U1, which controls the switching. U1 is a small
eight-pin dual in-line (DIL) chip that is programmed to turn the MOSFET on after a set
delay. I will discuss that later. R3 provides a sample of the AC mains voltage to the
controller. This enables the controller to detect the mains zero crossing and switch
the supply on at the right moment. The result makes for a smoother increase in the
high voltage and avoids any high in-rush currents. You design it
Initially, the controller holds the MOSFET in the Off position by shorting the gate to to your specifications using
the drain via the activated transistor Q2. R2 and C4 provide that function for the first our FREE CAD software,
Front Panel Designer
few milliseconds until the controller is operational. Following the delay, the controller
releases Q2 by pulling down its GP10 Pin 7. R2 increases the gate voltage. The MOSFET
switches on at the next mains zero crossing and the high-voltage supply circuit is
enabled. The controller also drives a bi-color LED to show a delay or an active state.
The controller has an internal shunt regulator, which is supplied via R7. VSS is
the controller ground pin, and VDD is its supply pin (not shown on the chip drawing).
Since I had one pin left at the controller chip, I connected that to J5. Depending
on whether J5 is jumpered to ground, the controller selects one of two delay times:
low or high. We machine it
and ship to you a
Controller professionally finished product,
You may wonder what business a digital (there, I said it!) controller has in audio. no minimum quantity required
The answer is none, of course. That’s why it isn’t in the audio. And if you are afraid
of “digital hash” or whatever, rest assured: After the initial delay and the switching
●
Cost effective prototypes
on of the high voltage, the controller goes to sleep and switches itself off until the
next time you switch on the amplifier’s mains voltage. and production runs with
If you are new to controllers or are unfamiliar with them, it may seem like a bit of no setup charges
magic, but it is not rocket science. The controller program is simply several instruc- ● Powder-coated and anodized
tions executed by the chip after it is powered on. For example (in order of execution):
finishes in various colors
• Set Pin 7 to VDD to keep the MOSFET switched off ●
Select from aluminum,
• Read the voltage at Pin 5 to determine whether the jumper is placed and choose
acrylic or provide your
either the Low or the High delay value from memory
own material
●
Standard lead time in
5 days or express
manufacturing in 3
or 1 days

FrontPanelExpress.com
1(800)FPE-9060
Figure 2: The complete circuit for the high-voltage delay unit contains a time delay.

audioxpress.com | February 2014 | 55

ax Audio Electronics
Photo 1: The top (a) and
bottom (b) of the Psudelay
PCB are shown.
About the Author
Jan Didden has written
for audioXpress since
the 1970s. He is retired
following a career with
the Netherlands Air Force
and NATO. He worked in
logistics, air defense, and
information technology.
Retirement has provid-
ed him with the time to
finish all the audio proj-
ects that have piled up
for decades. He writes
about them on his web-
site linearaudio.nl. Jan is
also the publisher and
managing editor of the
twice-yearly bookzine
Linear Audio.
Category Quantity
Resistors
Capacitors
ICs
Transistors
Diodes
Miscellaneous
Table 1: Bill of Materials
56 | February 2014 | audioxpress.com
a) b)
• Switch on the LED for the Delay state to make life easier, you can download the compiled
• Wait the required time until the delay has passed program from my website (www.linearaudio.nl) and
• Wait until the voltage at Pin 6 goes through zero program the chip yourself or you can order the chip
(mains zero crossing) and the small PCB from Circuit Cellar’s CC-Webshop.
• Set Pin 7 to 0 V to switch on the MOSFET
• Switch on the LED for the Active state Construction
I have laid out a small PCB for this device (see
The way I set up the program, the delay is 40 s Photo 1). If you want to roll your own, the PCB
without the jumper at J5; with the jumper, the delay Gerbers are available through my website under My
is 60 s. Projects. The website also contains a stuffing guide.
For small microcontroller s such as the The designated MOSFET has an insulated tab so
PIC12HV615, there are many open-source develop- you can simply attach it to the amplifier chassis.
ment programs you can use to translate the previous Personally, I prefer a more secure approach, so I
program into instruction codes and read it into the use a small TO-220 isolator. Actually, the dissipation
PIC though an inexpensive USB programmer. But in the MOSFET is so small you can also just solder
it in an upright position to the top of the board.
Reference Value Comment Connect the GND and CT to J2 and J1, and connect
a heater winding for the supply at J3 and J4.
2 R1,R4 22 kΩ 0.25 W MF One word of caution: The heater used for this
1 R2 4.99 kΩ 0.25 W MF supply floats with the full transformer high-volt-
age output voltage. So, it should have adequate
2 R3,R7 10 kΩ 0.25 W MF
isolation from the other parts of the transformer
1 R5 390 0.25 W MF and the chassis. If in doubt, use a separate small
2 C1,C2 100 uF/35 V heater transformer (or anything that delivers a
secondary 6 to 10 VAC at a few milliamps). Note
2 C3,C4 100 nF
that you cannot use this heater winding to power
1 U1 PIC12HV609 eight-pin DIL other tube heaters in the amplifier. You would need
1 Q1 STF21NM60ND a separate winding (or transformer).
or equal I have used this little circuit for more than a
1 Q2 BC546BP year. I found it to be very reliable and it does what
Small-signal
it’s told to do. Have fun! ax
1 D1 LED-BICOLOR
2 D2,D3 1N4002
1 D4 Z15 Zener 400 mW Source
1 J5 CONN-H2 sil-2 header PIC12HV615 microcontroller
Microchip Technology, Inc. | www.microchip.com
 engineering
embedded
electronics

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 ax Hollow-State Electronics
The Tone Character of
Tube Guitar Amplifiers
By
Richard A tube guitar amplifier’s character can be described in three parts: the tone, the
Honeycutt nature of the distortion, and the “feel.” The tone is a function of the frequency
(United States)
response and the design of the tone controls. The nature of the distortion depends
on the type of tube circuit that is distorting (e.g., triode preamplifier, pentode
preamplifier, phase splitter, single-ended output stage, or push-pull output stage)
and how hard that stage is driven. The “feel” is controlled by the power supply’s
voltage regulation. Good regulation provides a responsive feel, while poor regu-
lation provides a more laid-back, relaxed feel. This article will examine different
amplifiers’ tones.

The fundamental frequency range of a guitar
using standard tuning extends from 80 to 640 Hz,
assuming 24 frets. Figure 1 shows the waveform
produced by an electric guitar with two humbucking
pickups. The low E string was open and the upper
three strings were fretted at fret 24.
Figure 2 shows significant harmonic content is
present up to about 2 kHz (roughly the third har-
monic of the highest note). If the waveform is heav-
ily clipped, the frequency range is much wider (see
Figure 3). To accurately hear the guitar, including the
distortion components, we would need the amplifier
and speaker to have a response from almost DC to
over 10,000 Hz. Actually, the upper harmonics are
unpleasant, so most guitar speakers roll off well
below 10 kHz. Also, few speakers, even bass speak-
ers, have responses below 40 Hz.
The Speaker Helps Set the Tone
The speaker is an important ingredient in deter-
mining a guitar amplifier’s tone. Many of the earliest
amplifiers used 10” speakers with seamed paper
cones and surrounds (see Photo 1).
Figure 4 shows the response of such a speaker.
Later amplifiers often used 12” speakers. Begin-
ning in the 1960s, the speakers were classified as
having either an “American” or a “British” tone (see
Figure 5 and Figure 6).
Some guitarists prefer premium speakers such as
the instrument speakers produced by Electro-Voice,
JBL, and (in the past) Altec-Lansing. These premium
speakers have much flatter frequency responses and,
therefore, do not affect the tone as much as the
more common speakers. Since the speaker’s sound
is inextricably a part of the guitar amplifier’s overall
sound, you must keep in mind the characteristics of
the speaker when considering the tone of an amplifier.
The tone of the guitar amplifier itself depends
on the low- and high-frequency cutoffs and the
design of the tone controls. Most amplifiers (not
including the speaker) have responses that are
reasonably flat from about 60 Hz to 15 kHz, if you
ignore the tone control circuit. The speaker takes
care of the rolloff of undesirable high-frequency
distortion products.
There are two notable exceptions. One is the

Figure 1: This waveform

includes the full-frequency
range produced by an
electric guitar.

58 | February 2014 | audioxpress.com

Figure 2: This frequency analysis shows the frequencies contained in the Figure 3: The frequency range widens if the signal is distorted.
waveform shown in Figure 1.

110

100

90

80
SPL
(dB) 70

60

50
Photo 1: Early guitar amplifiers used 10” speakers such as this one.
40

110 500 30
20 50 100 200 500 1,000 2,000 5,000 10,000 20,000
105
200 Frequency (Hz)
100
100
95

90 50 Figure 4: The frequency response of an early 10” guitar speaker includes

85
20
Ω a boost from about 1.5 to 5 kHz and a sharp rolloff above 5 kHz.
80

75 10

70
6
65
3
60
20 50 100 200 500 1,000 2,000 5,000 10,000 20,000
90
Frequency (Hz)

Figure 5: The frequency response of this 12” “American” speaker is 70

similar to that of the 10” speaker in Figure 1, but the midrange dip
centered about 400 Hz imparts a different characteristic tone. The lower
curve shows the speaker’s impedance.
20 100 1,000 20,000
Frequency (Hz)

Figure 6: This “British” speaker does not have the midrange dip so it has
a different tone.

Figure 7: The Gibson EH-150 included a tone switch located between the Figure 8: This “losser” 1-potentiometer tone control has been popular,
preamplifier and the driver. especially in lower-cost amplifiers.

audioxpress.com | February 2014 | 59

 ax Hollow-State Electronics
Mike Matthews Freedom Amp manufactured by
0 Electro-Harmonix in the 1970s. This amplifier could
be powered by the AC mains or by 40 D-cell bat-
–5 teries. Partially to conserve battery life by eliminat-
ing power-hogging low frequencies, this amplifier
–10 sharply cut off lows below about 160 Hz (if mem-
ory serves). A side result was that the power was
–15 delivered only in the frequency range to which the
ear is most sensitive, making for an amplifier that
–20 sounded incredibly loud! The other exception is cer-
tain models of VOX amplifiers that also cut off the
–30
20 30 100 300 1,000 3,000 10,000 lows well above 80 Hz.
Frequency (Hz) The earliest guitar amplifiers had no tone controls
or even volume controls. Before too long, though,
Figure 9: The 1-potentiometer tone control provides only treble cut. amplifiers began including a tone switch (see Figure 7).
This switch increased the low-frequency cutoff of the
interstage circuit from about 24 Hz (with the switch
closed) to 450 Hz (with the switch open).

Potentiometer-Type Tone Control

The addition of a potentiometer-type tone control
Figure 10: This Fender improved the tone switch (see Figure 8). The fre-
circuit has separate bass quency-response curve shown in Figure 9 reveals
and treble controls. this is simply a treble-cut control.
Many of the better amplifiers from the 1950s and
later incorporated separate bass and treble controls.
0 Although some of them used circuitry that resemble
the Baxandall tone control circuits used in hi-fi equip-
–10
ment, the performance was very different. For exam-
–20 ple, the “turnover point” above which the bass control
had little effect and below which the treble control had
–30
little effect was not usually 1 kHz, as it was in hi-fi
–40
tone controls. Also, some of these control sets had
positions enabling you to obtain a flat response and
–50 some did not. For some of these controls, the closest
–60
thing to a flat position was the full-counterclockwise
position of both the bass and treble potentiometers.
–70
10 30 100 300 1,000 3,000 10,000 30,000
For example, Figure 10 and Figure 11 show the circuit
Frequency (Hz)
and curves for the “normal” channel of the Fender
Bassman 6G6B bass amplifier.

Figure 11: The bass control in the circuit shown in Figure 10 is boost-only. The green curve Midrange Control
shows full boost of both controls. Red is the full cut of both controls. To provide even more control over the amplifi-
er’s tone, some manufacturers added a midrange
control. The so-called “bass-mid-treble tone stacks”
reception has been mixed. Some guitarists like them
very much, while others cannot find any setting of
these controls that suits them.
The circuitry and curves are shown in Figure 12
and Figure 13. This amplifier also had “deep” and
Figure 12: This bass-mid- “bright” switches. The curves shown in Figure 13
treble tone stack was used were made with deep switch on.
by Fender in the AB-864 A few amplifier manufacturers (e.g., Orange)
guitar amplifier. experimented with design to provide a unique tone.

60 | February 2014 | audioxpress.com

 ax Hollow-State Electronics
10 which this switch is located. Basically, the circuit is a
0
bass-cut switch that works in tandem with the bass
and treble controls and the DEEP switch. Figure 15
–10
shows the behavior of this circuit with the switch in
–20 position 3 (two 4,700-pF capacitors in series with a
0.068-µF coupling capacitor). In modeling for this
–30
graph, the bass, treble, and volume controls were
–40 set at 50% rotation (approximately 10% rotation,
since they are logarithmic potentiometers).
–50

–60 Graphic Equalizers

10 30 100 300 1,000 3,000 10,000
Frequency (Hz)
Figure 13: These curves show the effect of the bass-mid-treble stack. The green curve
represents the controls at their midpoint, red is minimum treble, blue is maximum treble,
magenta is minimum bass, and yellow is maximum bass.
Orange incorporated a “contour” switch. The first
model with this feature had only icons on the front
panel and no text. On later models, the contour switch
was identified as “FAC,” (i.e., frequency analyzer con-
trol). Figure 14 shows the portion of the circuitry in
To provide even more flexibility in amplifier tone
control, a few models in the late 1970s and the 1980s
included graphic equalizers such as those used for
sound reinforcement systems. Some manufacturers
offered them in footswitch (i.e., “stompbox”) versions
(see Photo 2). These were best used in the effects
loop of an amplifier, which consisted of a line-level
output and input that enabled the user to connect an
external device between guitar amplifier’s pream-
plifier and phase splitter. (In Figure 14, the effects
“send “output is connected to the bottom of R12, and
the “return” input is connected to the bottom of R8.)
Most players do not seem to need the level of
tone control provided by this 10-band graphic equal-
izer, although there are still some on the market.

offering an extensive range of ready-to-go

toroidal transformers
to please the ear, but won’t take you for a ride.

Avel Lindberg Inc.

47 South End Plaza, New Milford, CT 06776
p: 860.355.4711 / f: 860.354.8597
sales@avellindberg.com • www.avellindberg.com Photo 2: This 1970s-vintage outboard graphic equalizer provided even
more tone control.

62 | February 2014 | audioxpress.com

The six-band equalizers now available better fit an
electric guitar’s frequency range. Needless to say,
these outboard devices are solid-state, since hol-
low-state graphic equalizers are very bulky. (The
Blonder-Tongue B-9 Audio Baton was a nine-band
graphic equalizer using six tubes, and it weighed
approximately 15 lb to 20 lb!) A few amplifiers offered
built-in graphic equalizers, and some—though by no
means most—still do. Figure 14: This section of an Orange graphic amplifier schematic shows the tone controls
and contour switch.
Distortion
The stage in which most of the distortion is gen- –20

erated affects the preferred tone control setting.

If the distortion occurs before the tone controls, a –30

guitarist may choose to reduce the treble control to

avoid a razzy, metallic sound. However, this decreased –40

treble may not be ideal when the guitarist wants a

clean sound. Therefore, some amplifier manufac- –50
turers incorporate a “lead” and a “rhythm” channel Figure 15: The
with separate volume and tone controls, often with midrange hump
–60
is created by the
a footswitch to switch between channels. In this
tone controls
way, the player can change between a distorted
–70 and the contour
lead sound with reduced treble and a brighter clean 10 30 100 300 1,000 3,000 10,000
switch working
rhythm sound. Frequency (Hz)
together.
An interesting aspect of this interaction between
distortion considerations and tone preferences is
the change in many guitarists’ opinions of some Simplify your electronics projects
older, low-cost amplifiers such as the Silvertone
“case amplifier.” (The amplifier, which was built
into a guitar case, was invented and manufactured
by Danelectro). SR-84 Hifi EL-84 Push-Pull Tube Amplifier
12AX7 Preamp · EL84 Push Pull Power Stage · Hi-End Building
During its commercial life, the case amplifier was
110/220V Compatible · Loundness Control · Cateye Indicator
considered a “starter amp,” and it was marketed as
such. It was not capable of much volume without a
Custom photo engraving available!
lot of distortion, especially at low frequencies. In the
early 1960s when this amplifier was on the market,
distortion was unwanted. After distortion came into
vogue, the case amplifier came to be prized as a
practice amp, precisely because of its distortion.
This little hot-chassis amplifier had no tone control,
but if you like the sound of a 12AU6 pentode pream-
plifier driving a single-ended Class-A 50C5 pentode
power amp into distortion, coupled with the limited
frequency response of the small speaker, you could
practice with high distortion, without getting evicted
from your apartment!
In fact, there is a new trend among some guitarists
toward a more classic amplifier design, as exempli-
fied by the Shaw Audio Retro Mod 15, a push-pull-
output 6V6-based amp with a single tone control, IC · Kits · Parts · Enclosure · Amplifier · Chassis · Tube · DACs
a volume control, and a “fat” switch. Fans of these For
amplifiers have almost come full circle, although I Companies · Engineers · Professionals · Students · Amateurs
doubt if we’ll ever see amplifiers again without any Website: http://www.siliconray.com Email: sales@siliconray.com
tone or volume controls! ax

audioxpress.com | February 2014 | 63

 ax Member Profile
Member Name:
Bob Cordell
Location:
Holmdel, NJ
Education:
Bob has a MSEE from Stanford University, Stanford, CA,
and a BSEE from Rensselaer Polytechnic Institute
(RPI) in Troy, NY.
Occupation:
He is a retired electrical engineer.
Member Status:
Bob said he has been subscribing to audioXpress “since
the beginning of time.” Most Recent Purchase:
Affiliations: Bob recently upgraded his equipment with an OPPO BDP-95 CD/SACD player.
Bob retains memberships to the Institute of Electrical and Current Audio Projects:
Electronics Engineers (IEEE) and the Audio Engineering Bob is writing the second edition of his book, “Designing Audio Power Amplifiers,”
Society (AES). and working on his website, www.cordellaudio.com.
Audio Interests: Dream System:
He enjoys working on amplifiers, loudspeakers, and test Bob said, “I am not a dreamer, I’m a builder.” His current system includes a
equipment. He also spends his time authoring books and 3.5-way active loudspeakers, each with four 125-W MOSFET power amplifiers
articles about his various audio projects. built into the cabinets. ax

www.hometheatershack.com
Audio/Video Discussion Forum
FO R SE R IO U S AUDIO PHILES !

64 | February 2014 | audioxpress.com

 ax AX Shop
FundamentalAmplifierTechniques3_Opmaak 1 22-9-10 15:55 Pagina 1
LOUDSPEAKER INDUSTRY SOURCEBOOK 2013
1 2
1 LINEAR AUDIO SERIES
(VOLUMES 0-6)
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Fundamental Amplifier Techniques
with Electron Tubes
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For years, author Rudolf Moers searched for a textbook about electron tube amplifiers in
order to learn how to design amplifiers using electron tubes, given that one has a reasonable Amplifier Techniques
with Electron Tubes
knowledge of analog electronics. Since he never found such a book, he has written one
himself in a style and content such as he would like to buy. To achieve this, he studied many
radio books and did many measurements. This was followed by the organization of all the
information, making many drawings and calculations and a lot of writing. By all this effort,
a lot of knowledge about electron tube electronics, which was under the threat of being lost,
will be retained.
There are two aspects of a technical book that readers like to see, but which are often poorly
presented. These two aspects are accuracy and a pleasing readability. Readers less skilled
in mathematics and interested only in formulae that provide a valid ‘recipe’ like to skip the
Theory and practice with design
derivation of that ‘recipe’. In this book this is possible without prejudice to the readability of
the story. These ‘recipes’ are framed at the end of formulae derivations. methods for self construction
There are readers who are not satisfied with formulae that ‘fall from the sky’ without
derivation.
For these readers, a ‘recipe’ without substantiation is unsatisfactory and the application of
■ COMPONENTS the formulae will be accepted only when the derivation is shown. The formulae are proved in
such a way that the reader can easily follow them through, thus a deep knowledge of mathe-
■ DESIGN SOFTWARE matics is not necessary.
The introduction in chapter 1 will be followed in chapter 2 by the principles of electron
■ DRIVERS
emission. Chapters 3, 4, 5 and 6 treat the diode, triode, tetrode and pentode respectively,
each with a different framework, but only at middle frequencies and without discussing
distortion and negative feedback. Chapter 7 treats the limits of audio frequencies and in
■ ENCLOSURES
chapter 8 are discussions about distortion and noise. Chapter 9 concerns negative feedback
& CABINETS
and in chapter 10 you can read how to build your own electron tube amplifier. Much of the
theory is checked against practice, and design methods give the reader help to start with
the design and construction of electron tubes amplifiers.
■ FINISHED SYSTEMS
The aim of the book is to give the reader useful knowledge about electron tube technology
■ HEADPHONES
in the application of audio amplifiers, including their power supplies, for the design and DIY
construction of these electron tube amplifiers. This is much more than just building an
electron tube amplifier from a schematic made from the design from someone else: not
■ INTEGRATED AMPLIFIED
only academic theory for scientific evidence, but also a theoretical explanation of how the
SPEAKER
tions, then SOLUTIONS
practice works. No modern simulations, but because you first understand the circuit calcula-
you can work with your hands to build the circuit and last, but not least, if you
have a multimeter, a signal generator and an oscilloscope, you can measure the circuit
■ MEASUREMENT
parameters yourself to see that theory and practice are very close. That is the aim, and
MICROPHONES
makes this book a unique reference source.
■ MICROSPEAKERS
Rudolf Moers
■ SPEAKER PARTS
ISBN 978-0-905705-93-4
■ TEST EQUIPMENT
Elektor International Media
www.elektor.com
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3 FUNDAMENTAL AMPLIFIER
Masterclass Masterclass
TECHNIQUES
Feedback in Audio Amplifier Feedback in Audio Amplifier
This book provides the reader with use-
ful knowledge about electron tube technol-
In this Masterclass we address several as-
Masterclass
pects of feedback in audio amplifiers. The Jan Didden
ogy in the application of audio amplifiers.
focus of this Masterclass, although not enti-
rely math-free, is on providing insight and un-
derstanding of the issues involved. Amongst
Recognized audio designer and feedback expert
Serving as a unique reference source, it
others, we discuss:
• Basics of negative feedback and its infl -
provides insight as you learn critical tech-
Feedback in Audio Amplifiers
ence on amplifier pa ameters;
• ‘Gotchas’ like Slew Rate Limiting;
niques. Then you can build the circuitis your-
• The impact of feedback on internal over-
This Masterclassload;
• Stability of feedback amplifiers
presented by Jan Didden
self. This is a must-have resource!
•
•
Feedback ‘speed’ and the difference
between delay and phase shift;
Positive feedback and its effects;
Jan Didden has spent most of his
working life with the Netherlands Air
• Simultaneous application of positive and Force and NATO, and in that period
negative feedback in a 1950s tube am- wrote many audio design articles for
the audio press. After his retirement,
SKU #BK-ELNL-978-0-905705-93-4
plifier
• Error Correction according to professor he finally found time to finish al
Malcolm Hawksford; those accumulated projects.
• Error Correction as used in the paX Am- At present his main interest is on er-
plifier published by Elektor in April and ror correction in power amplifiers,
May 2008; which leads to unusual designs. His
• Feedback or Error Correction - a case latest design, the paX power amplifi-
study: QUAD Current Dumping; er, was published by Elektor in 2008.
Since a few years Jan also publishes
This Masterclass provides a clear overview of Linear Audio, a unique, six-monthly
the benefits that can be obtained by feedback ‘bookzine’ dedicated to technical On this DVD:
and its sibling, error correction; but also of audio, with articles written by inter-
• 140 min. video presentation
4 MASTERCLASS FEEDBACK
its limitations and disadvantages. Recom- nationally recognized audio design
mended to audio designers and serious audio
hobbyists.
experts. • All material as PDF documents
Bonus:
IN AUDIO AMPLIFIERS DVD
J. Didden
ISBN:978-907920-16-5 • complete Amplifier Desig
Recorded at Elektor Castle, Limbricht, The Netherlands.
This video-DVD can be played stand-alone on any DVD- or • More than 100 pages additional
The Masterclass: Feedback in Audio
BluRay player; no separate computer is required. A PC is
required to read or print the included additional material
in PDF documents.
PDF documentation
r DVD
Picture format PAL 16:9 | Scene-selection | Stereo sound |
Dual-laye entation
Amplifiers DVD provides a clear over-
playing time 140 min. | © 2012 Elektor International Media B.V.
140 min.
video pres
view of the benefits that can be obtained
from feedback and error correction. It 4
also details their limitations and disad-
vantages. It is recommended to all audio
designers and serious audio hobbyists!
Further information and ordering
SKU #DVD-ELNL-978907920-16-5
www.cc-webshop.com
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audioxpress.com | February 2014 | 65
 ax Industry Calendar
Here are a few places where you might find a copy of audioXpress or possibly meet one of our authors and staff members:

February 4–6, 2014 Integrated Systems Europe (ISE) 2014

Amsterdam RAI, The Netherlands
www.iseurope.org
This is a very successful European show, promoted jointly by InfoComm and Custom Electronic Design and
Installation Association (CEDIA). The show has already surpassed all growth expectations and is becoming
larger than any of the original North American events. Integrated Systems Europe (ISE) is also becoming an
interesting convergence point for new Internet Protocol (IP)-based technology solutions, from home networks
and automation to building management. Audio over IP is becoming the hot topic and audioXpress will be
there, checking for the latest innovations.

February 24–27, 2014 Prolight+Sound Guangzhou

China (Guangzhou) International Professional Light and Sound Exhibition
Area A, China Import and Export Fair Complex, Guangzhou, China
www.messefrankfurt.com.hk
www.prolightsound-guangzhou.com
Benefiting from the strategic alliance between Messe Frankfurt and Guangdong International Science
and Technology Exhibition Company (STE), Prolight+Sound Guangzhou is now China’s largest professional
audio, lighting, and stage products show. Exhibition space at the 2014 show will expand from nine to 11
halls (110,000 m2) and the event is expected to attract an estimated 50,000 professional visitors. Messe
Frankfurt Shanghai will debut three new halls, that feature several pro audio brands including AKG, Beyer-
dynamic, Clair Bros, Crown, dbx, JBL, Lexicon, L-Acoustics, Martin Audio, Mediamatrix, Peavey, Sennheiser,
Shure, Soundcraft, and Studer.
Located in the Guangdong province, Enping City has long been China’s major import and export base for microphones and entertainment
equipment. For many years, the Enping Audio Industry Association has organized a pavilion at Prolight+Sound Guangzhou to strengthen trade
between the city and international markets. For the 2014 show, the Enping pavilion will expand from one to one-and-a-half halls to showcase more
than 1,000 different types of audio visual systems, conference systems, consoles, and microphones.

March 12–15, 2014 Musikmesse e Prolight+Sound Frankfurt

Exhibition Centre, Frankfurt, Germany
www.musikmesse.com
www.prolight-sound.com
www.messefrankfurt.com
This is by far the world’s largest professional show dedicated to music instruments, music technol-
ogy, and professional audio. audioXpress will attend in full force covering all the audio novelties in the
Prolight+Sound halls and the technology-oriented halls of the Musikmesse, where “the return of the
analog synths” is apparent. This is also the place to look for acoustical solutions, recording studio equip-
ment, and lots of sound reinforcement solutions. In fact, we believe no other show in the world unites as many audio companies from different
segments as the annual Frankfurt event. Look for audioXpress in the International Press distribution areas.

May 13–15, 2014 18th AES Brazil Expo

Expo Center Norte Pavilhão Amarelo, Vila Guilherme, São Paulo, Brazil
www.aesbrasilexpo.com.br
The AES convention in Brazil is not like any other Audio Engineering Society (AES) Convention held in the US or elsewhere. In fact, even
though the conference sessions follow the traditional association model—almost 100% in Portuguese—the exhibition floor is predominantly
dedicated to sound reinforcement and PA systems due to the importance of that professional audio segment in the Brazilian market, which is
one of the largest music markets in the world. The event, held in the megalopolis of São Paulo—the business capital of Brazil—is an important
annual gathering for the domestic pro audio industry, including speaker component and amplification manufacturers. In fact, the AES Brazil
is the main pro audio event in South America and provides an opportunity to find local partners and learn more about Brazil.

June 14–20, 2014 InfoComm 2014

Las Vegas Convention Center, Las Vegas, NV, US
www.infocommshow.org
The audiovisual communications marketplace, as the InfoComm association defines this major event, increasingly represents the inter-
section of AV and IT, technology and architecture, business and entertainment, government and hospitality, healthcare and education, live
events and installed systems. InfoComm is undoubtedly the most important professional AV show on the American continent. For 2014, with
more than 950 exhibitors—of which more than 300 are audio companies—and 35,000 plus attendees expected from more than 110 countries,
this is a unique opportunity to see the latest technology, learn new skills, and grow our professional network.

66 | February 2014 | audioxpress.com

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