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on test

Real Sound Lab


Loudspeaker Equalisation System
Real Sound Lab’s innovative speaker correction
system provides high-precision equalisation
tailored to your monitors and your room.
Hugh Robjohns


onitor loudspeakers are, without
doubt, the weakest link of
every audio chain! The perfect
loudspeaker simply doesn’t exist; every
design has a different blend of strengths
and weaknesses, but the distortions inherent
in current electro-mechanical transducers
conspire to degrade the wanted audio
signal in various frustrating ways. Of course,
loudspeaker technology has improved
steadily, if incrementally, over the last 100
years, but it is still very far from flawless —
manufacturers simply choose different sets
of compromises for their designs, to meet
the required performance, size, and cost
goals. However, the digital revolution has
affected pretty much every aspect of the
audio world, and loudspeaker designers are
increasingly now integrating sophisticated
digital equalisation techniques in an attempt
to correct transducer deficiencies and to
optimise loudspeaker system performance.
One interesting variation on this theme
is CONEQ from Real Sound Lab, a company
based in Riga, Latvia. CONEQ is designed
to measure electro-acoustic transducers
such as loudspeakers, and then to calculate
corrective digital equalisation to permit


near-perfect performance. The corrective
filters are applied to the monitoring signal
chain using either bespoke hardware DSP
equalisation or software plug-ins, depending
on practical requirements and budget. The
software is compatible with both Mac (OS
10.5.8 and higher) and Windows (XP SP3
and higher) operating systems.

Many third-party ‘corrective’ equalisation
systems obtain the speaker system
frequency response by measuring the sound
pressure level produced by the speaker
under test on-axis, at a defined listening
position — it is the obvious way to do it,
after all. Armed with a system response, it
is then fairly trivial to calculate the required
corrective filter to address any response
irregularities. However, the total contribution
the loudspeaker makes in a control room
must include the off-axis energy, which is
not taken into account with simple on-axis
measurements. As a result, while many
corrective systems often deliver improved
performance for on-axis listening, they
frequently result in degraded performance
off-axis, and the overall effect can be
disappointing or unsatisfactory because the

February 2013 / w w w . s o u n d o n s o u n d . c o m

frequency response of the total energy in
the room is unbalanced.
Real Sound Lab’s CONEQ system
takes a subtle but significantly different
approach: to determine the acoustic power
frequency-response rather than the simple
sound pressure level response, and adjust
it so that the speaker generates the correct
power spectrum across the listening area.
The idea was developed by Raimonds
Skuruls, who founded Real Sound Lab along
with investor Viesturs Sosars back in 2004.
Their technology is now available in various
software and hardware products, as well as
being licensed to companies like Panasonic
and Hitachi for use in their flagship plasma
TVs, Mitsubishi in their Immersive Sound
Technology surround systems, and in some
of JVC Kenwood’s home audio products,
among others.
I should emphasize that CONEQ
is designed to correct principally for
loudspeaker deficiencies: it is not a ‘room
correction’ system in the conventional
sense. Very sensibly, Real Sound Lab
state that room acoustic issues are best
dealt with using conventional acoustic
techniques. Having said that, the
CONEQ system will inherently attempt
to correct for some of the effects of room
acoustic anomalies captured during the
measurement process, such as direct
reflections from a console or table top.

The Measurement Process
Unlike most simple DSP equalisation
systems, the CONEQ system doesn’t
measure from one fixed, on-axis position.
Instead, the measurement process requires
the measurement microphone to be moved
continuously in a zig-zag pattern up and
down, moving from left to right in an arc
in front of the speaker being measured. In
theory, the measurement mic should be
moved across the surface of a complete
sphere surrounding the speaker, but
that is clearly impractical and generally
unnecessary. Instead, it is sufficient, in

most cases (assuming a reasonably well
treated room), to measure across a surface
representing the manufacturer’s intended
coverage angle. The distance between
mic and speaker affects the accuracy of
low-frequency measurement, and for reliable
results the measurement microphone
should traverse an arc at a distance which
corresponds to between one and two times
the speaker’s largest cabinet dimension.
Each speaker is measured separately, to
generate specific corrective filters for each
channel, but the process only takes about
a minute per speaker. As the microphone
is moved along its zig-zag pattern in front
of the speaker, the measurement software
generates a series of short signals. After
about one minute, the software will have
gathered around 180 sound pressure
measurements from the space in front of the
speaker, and these are used to calculate the
corresponding acoustic power frequency
response, in essence by summing the
squares of all the pressure measurements at
each frequency.
The result of the calculation is the
system’s actual acoustic power frequency
response and, since the desired target
is a flat response, it is straightforward to
calculate the required inverse filter needed
to process the monitoring signal to achieve
an overall flat response. The required
level of precision would be impossible to
achieve in the analogue domain, but it
is relatively simple using a finite impulse
response (FIR) filter running within a digital
signal processor.
By default, CONEQ uses a complex
4096-tap minimum-phase FIR filter to correct
the power response anomalies. The 4096
taps refer to the number of simultaneous
calculations employed to achieve the
desired filter response. More taps basically
means more precision, in this case allowing
5Hz resolution. The minimum-phase
design mimics analogue filters in avoiding
the pre-ringing effects associated with
linear-phase filters — although that is an

As an alternative to
the hardware, the
CONEQ filters can
be implemented using
a range of plug-ins.

option, if preferred —
and it also minimises
the inherent filter
Once calculated,
CONEQ’s proprietary
filter data can be
saved as a small 13kb
file for each measured
channel to any
convenient location on the computer. From
there, it is loaded either into a software
plug-in inserted in the monitor output path
of the DAW, or into a dedicated hardware
DSP equaliser wired into the analogue
(or digital) monitor signal path. While the
CONEQ FIR data format is proprietary,
Real Sound Lab are working with third-party
companies too. For example, the BSS
Soundweb London ‘BLU’ signal processing
and speaker management system,
which is often employed in large-scale
live-sound rigs, can accept CONEQ filter
data to linearise PA speaker responses.
However, the data has to be converted and
exported first using the CONEQ Workshop
software (see below).
So that’s the basic concept and process.
How does it work in practice?

Measurement Software
The measurement process is performed
via a bespoke software program called
CONEQ Workshop, which is licensed via
a USB dongle using eLicenser technology.
The Workshop software (currently v3.2)
is available in three variants: Unlimited,
Annual and Limited, the last being supplied
as standard with any purchased CONEQ
hardware or plug-in system equaliser.
A cut-down CONEQ Starter version also
exists, and provides the basic measurement
and filter synthesis facilities, but omits most
of the more advanced
features of the full
Workshop software.
It is really intended to

provide an easy introduction to the system,
to get you up and running quickly and easily.
Expanding on the basic functionality
of the Starter software, the full Workshop
version includes the ability to create
correction filters (with either minimum
or linear-phase responses) from multiple
measurement sequences to improve
accuracy. It can also calculate filters from
imported pre- recorded files and there
are facilities to adjust the maximum
amount of boost or attenuation that the
corrective filter is allowed to apply (to avoid
overloading amplifiers and exceeding
excursion limits of drivers). By default, the

Real Sound Lab
• Easy-to-use measurement software, with
useful advanced-user options.
• Capable of very impressive results even in
extreme situations.
• Software EQ plug-ins are cost-effective.
• Hardware EQ units provide very flexible
• Compatibility with BSS live-sound

• Initial measurements requires decent test
mic and interface hardware.
• The hardware EQ and unlimited
Workshop software options aren’t cheap!
• No corrective EQ system can deal with
fundamental acoustic room anomalies.
• It might be better to invest in better quality
monitors, rather than use CONEQ to try
to improve inherently poor speakers.


The optional hardware
element of the CONEQ
system consists of
a range of powerful
digital equalisers.

A unique loudspeaker-correction system
based on assessing and correcting the acoustic
power response of a loudspeaker across the
listening area to improve its performance and
accuracy. The calculated correction filters can
be implemented using bespoke plug-ins or
dedicated hardware units.

w w w . s o u n d o n s o u n d . c o m / February 2013


on test
Real Sound Lab CONEQ

The Starter Workshop program features an
animation showing how to move the mic.

correction bandwidth — the frequency
range over which the corrective filter
is allowed to operate — is restricted
to the range between the frequency
extremes where the response falls by 6dB.
However, this can be extended manually.
A measurement microphone calibration
file can also be imported to ensure really
precise measurements, if necessary, and
the FFT time window (used to exclude
room reflections) and filter smoothing can
also be adjusted. Clearly, there’s a lot of
tweakability on offer here for techy nerds
to play with, and it’s all explained well in
the Reference Manual PDF file — but in
reality, most users will not need to tweak
anything, because it works extremely well in
its default mode anyway.
The initial measurement stage uses
whatever audio interface is available to
the computer, and there is a loopback
test to check the linearity of the hardware
and compensate, if required. The main
process begins when you start a new
Workshop project, plug a measurement
mic into the interface input, and route
the interface output to the speaker to be
tested. (A good-quality, flat-response,
small-diaphragm, omnidirectional capacitor
mic will do if a proper measurement mic
isn’t available.) The input and output levels
are then optimised and the measurement
sequence can be started.
The Starter version software
automatically runs a synchronised, animated
on-screen graphic that illustrates how and
where to move the microphone, which is
a useful aid as you get used to the CONEQ
system. The animation isn’t included in the
full Workshop versions, but you’ll know
what to do by the time you’re ready to
explore that program! As the measurement
sequence runs, a repetitive series of
moderately loud ‘chirps’ is generated by


the speaker. When it ends, the
software takes a few moments
to calculate the acoustic power
response and then displays the
measured response, the target and
the corrective filter on screen.
At this stage, you can modify
the corrective filter response, if
necessary, by changing the program
parameters — extending or
reducing the corrective bandwidth
and boost/cut limits, for example.
You can also introduce additional
EQ requirements, perhaps to
conform with the cinema industry’s
X-curve, for example.
Once you are happy with the corrective
filters — separate filters are normally
generated for each channel tested, of
course — they can be named and saved
to a convenient computer location,
from where they can be uploaded into
a suitable CONEQ equalisation system. If
a hardware DSP equaliser is connected to
the computer, the Workshop program can
upload the FIR data directly.

CONEQ Hardware EQ
Real Sound Lab currently offer three
different hardware digital EQ units: the
APEQ-2pro, the APEQ- 2pro DIO, and
the APEQ-8pro DIO. As the names imply,
these are two-channel and eight-channel
variants (the last to support surround sound
applications, primarily, but it could be
used to correct multiple alternative stereo
monitor systems). All three models feature
balanced analogue inputs and outputs,
while the DIO suffix indicates the inclusion of
AES3 digital I/O as well. If a digital input is
detected, it is selected automatically.
Both two-channel units operate with
a fixed processing sample rate of 48kHz,
24-bit, while the eight-channel model will
process at all four standard rates between
44.1 and 96 kHz. All employ 4096-tap FIR
filters which deliver 5Hz filter resolution,
and the A-D and D-A converters in all three
models are claimed to manage a dynamic
range of 115dB (A-weighted) with THD
below 0.01 percent. The nominal analogue
I/O level is +4dBu, with a clipping level of
+21.4dBu (+24dBu for the APEQ-8pro),
and typical latency of 1.6ms. The digital
input can accommodate any source sample
rate between 44 and 96 kHz, 24-bit (up to
192kHz for the APEQ-8pro), and latency
from digital input to digital output is 2.4ms.
The review system included the
APEQ-2pro DIO model, which is equipped
with both XLR and TRS connectivity for the

February 2013 / w w w . s o u n d o n s o u n d . c o m

balanced analogue inputs and outputs,
along with an AES3 input and output on
XLRs, an RS232 serial control port (used
only for service functions), and a USB B-type
socket. The USB link is only required for
uploading FIR filter data, which is done
via a function in the Workshop program,
or via the included C2 Communication
Tool software; once configured, the unit
runs stand-alone. The front panel features
two momentary push buttons, each
with status LEDs, to power the unit and
bypass the DSP filters. (There is a proper
mains isolation rocker switch on the rear
panel.) There is also a simple four-LED
bar-graph level meter, and the analogue
I/O is hard-bypassed with relays should
the APEQ suffer a mains failure. In normal
operation, the bottom -48dBFS LEDs will
be permanently on, with occasional flashes
from the -12dBFS LEDs on high peaks.
The top two LEDs (-6dBFS and Clip) won’t
normally illuminate at all.
The eight-channel APEQ model differs
from the two-channel units in several
ways. It uses the ubiquitous Tascam
D-sub connectors for the analogue I/O
to save space, while retaining separate
XLRs for AES3 interfacing. A word-clock
input is provided on a BNC connector,
along with an Ethernet port, while the
USB socket is moved to the front panel.
Either the Ethernet or USB ports can
be used to control and configure the
APEQ-8pro DIO, and the only switch on
the unit is a mains on-off push-button on
the front panel, while the metering has
been extended to form a block of eight
six-LED bar-graphs. The most significant
difference, though, is the inclusion of
a front-panel microphone input XLR (with
software switchable +48V phantom power)
to enable CONEQ measurements without
a separate audio interface. Save for the
larger analogue headroom and 192kHz
maximum digital input sample rate, the
technical specifications are identical to the
two-channel models.

Filter Software
The alternative to running the calculated
filters in one of Real Sound Lab’s APEQ
hardware units is to run them using
DAW plug-ins instead, and the CONEQ
plug-in software is available in both
VST and RTAS formats (currently 32-bit
only) for compatibility with most DAWs,
audio editing applications and computer
operating systems.
Ideally, the CONEQ plug-in should be
inserted into a dedicated monitoring chain

output bus, but if that isn’t possible, it
can be installed as the final plug-in in the
master mix bus to correct the output for
monitoring. However, in that situation it is
obviously essential to bypass the CONEQ
plug-in before the final mix is bounced or
exported, to prevent the corrective filter
equalisation becoming embedded in the
mix itself!
In a similar way to the APEQ hardware
equalisation units, there are four plug-in
variants. The P2 and P8 versions process
two and eight channels, respectively,
and use 1024-tap FIR filters to minimise
the CPU load. The alternative P2pro and
P8pro versions employ the full-fat 4096-tap
FIR filter algorithms. The FIR length
makes little difference when correcting
the frequency response above about
600Hz, but it has a significant effect on
the filter precision at lower frequencies.
The 4096-tap filter enables frequency
adjustments every 5Hz, while with 1024
taps, the resolution is only 20Hz.
I used the VST versions of the CONEQ
software mostly in Cockos Reaper projects,
but also in Adobe Audition 3.0. The simple

and Pro versions of the CONEQ interface
essentially look and function identically.
A rotary input-level attenuator is provided
in the top left hand corner, with a bypass
button immediately below and an output
meter bar-graph to the right. Below that
are status windows denoting the sample
rate, filter resolution, audio format and
latency (256 samples is the default, but it
can be adjusted between 2 and 2000). The
bottom of the plug-in GUI contains separate
bar-graph meters for each channel and data
boxes that denote the current FIR filter files.
Clicking on these calls up a browser window
for navigating to and selecting alternative
FIR filter files.

In Use
I used the CONEQ system to measure
and correct a number of different
monitoring systems I have to hand, using
both the hardware and plug-in equalisers
on different systems to correct their
responses. In all cases, the results were
definitely beneficial, although the actual
level of improvement varied from virtually
negligible to ‘Wow!’ — depending on

The Trinnov Optimizer ST is a cut-down,
stereo-only version of the original
multi-channel Optimizer hardware system,
which runs proprietary software on a bespoke
computer platform to provide phenomenally
powerful speaker and room correction.

the quality of the underlying system and
room acoustics! Self-evidently, if a speaker
is well-designed and has a pretty flat
acoustic power response across the
listening area, the CONEQ system will
have very little to do. Conversely, if the
speaker is poor, or located in a position
that degrades its performance significantly,
the CONEQ system can make a very
substantial improvement.
The plots shown here illustrate an
extreme example of correction obtained
with a pair of little PMC DB1 speakers
positioned temporarily on high stands at
the sides of a large office desk in a small,
nearly square room, about 18 inches in
front of the back wall and suffering from
hard reflections from the desk surface.
Subjectively, the bass response from this
configuration was extremely weak, and the

w w w . s o u n d o n s o u n d . c o m / February 2013


on test
Real Sound Lab CONEQ

mid-range was noticeably coloured by the
desk surface reflections.
After a couple of minutes spent
measuring each speaker (I used an NTI
calibrated measurement mic connected to
a GML preamp and RME AIO interface),
and a few more minutes fiddling with
the correction bandwidth and EQ limits,
I was able to install the resulting FIR filter
coefficients into both the P2pro plug-in and
the APEQ-2pro DIO hardware.
To correct the bass response, I had to
allow the system to introduce some pretty
extreme bass boost — about 16dB at
40Hz, in fact — and not all speakers would
have the headroom to cope with that.
Fortunately, these little PMCs are remarkably
robust and capable, and I don’t monitor at
silly levels anyway, so there was plenty of
untapped headroom in the Bryston 4B amp
I was using at the time. I should stress that
I wouldn’t recommend this as a practical
solution — dealing with the room’s
horrendous acoustic properties with bass
traps would be a far better (and probably
cheaper) approach! — but it does highlight
the system’s capabilities rather well. As
the plots show, the response corrections
through the mid-range were relatively small,
with a couple of little 1.5dB peaks at 178
and 580 Hz, and a slight 1dB dip around
1kHz, which indicates the inherent quality of
this speaker design.
Switching in the corrective filters using

The three hardware CONEQ equalisers
are the APEQ 2-pro, APEQ 2-pro
DIO and APEQ 8-pro DIO. Suggested
retail prices (including German VAT
at 19 percent) are €1190, €1547 and
€4165 respectively.
The three full versions of the CONEQ
Workshop software differ, as their names
suggest, in the length of time over which
they operate. The eLicenser dongle
supplied with the €1190 Unlimited version
runs straight out of the box forever, while
the other two will only work once the
dongle is authorised on-line, and then
only for a predetermined period. The
€476 Annual edition runs for 53 weeks
(but can be extended by purchasing a new
licence), while the €119 Limited edition
runs for 30 days once activated. That’s
more than enough to measure a few
systems, create the appropriate filter files,
and perhaps fine-tune and re-measure to
optimise the results.
There are four versions of the CONEQ
Filter plug-in: the 1024-tap P2 (€178.50)
and P8 (€535.50), and the 4096-tap P2-pro
(€654.50) and P8-pro (€2082.50).


These plots show the original measurement (green), calculated filter response (yellow) and intended end
result (orange) from the right of my pair of PMC DB1 speakers, mounted next to a large desk. The light
grey areas are the ‘out of bounds’ regions where the software stops trying to correct the response.

either the hardware or plug-options
provided exactly the same quality of
correction and improvement, although
I found the hardware solution slightly more
practical in my application, because it
meant I could use the speakers with sources
other than the computer. In cases where
the monitors are always used to audition
the output of a DAW, the software plug-in
would be an entirely satisfactory and
considerably cheaper option, of course.
The audible improvement in this
lashed-together DB1 office system was
nothing short of miraculous. The bottom
end not only regained all the bass I normally
associate with the DB1s, but actually the
performance was enhanced, with much
of the weight and solidity of a far larger
system set up in a far better room. That
was a phenomenal result that went far
beyond my expectations. While the bass
improvements were surprising and obviously
hugely beneficial, there was no doubt that
the mid-range also sounded noticeably
more transparent and neutral. Bypassing the
correction filters unmistakably confirmed the
improved clarity and sense of openness, and
more natural timbres to voices and acoustic
instruments, in particular.
I’m not going to claim that the CONEQ
system is able to elevate the little DB1s to
match the quality and resolution of high-end
three-way monitors, but the improvement
in mid-range clarity was very audible and
definitely worthwhile, while the corrected
bass response turned a completely
unworkable setup into a reliable reference
monitoring system that enabled me to

February 2013 / w w w . s o u n d o n s o u n d . c o m

deliver decent, transferrable mixes —
a situation that was completely impossible
before CONEQ correction.
I’ve used several room and speaker
corrections systems over the years, but
the only one that really impressed me with
its ability to bring genuine improvements
(without trashing the sound elsewhere
in the room), was the very expensive
French Trinnov system. However, I am very
pleased to be able to add the Real Sound
Lab CONEQ system to my list of useful
loudspeaker correction candidates.
The irony with systems like this,
of course, is that they make the most
impressive difference with poor monitor
speakers, but for the cost of poor monitors
plus this kind of correction tool, you might
just as well have bought better monitors
in the first place, that potentially wouldn’t
need correction! However, placement
issues and local acoustics can degrade the
performance of even the best monitors,
and in those situations the CONEQ system
is a very useful means of regaining the
quality in a remarkably simple, reliable and
user-friendly way. A free demo version of
the software, with simple measurement and
filtering capabilities, can be downloaded
from the Real Sound Lab web site and,
if you have access to an omnidirectional
small-diaphragm capacitor mic, I urge you to
give it a try.  
££ See ‘Pricing’ box.
TT Real Sound Lab +371 6788 9828

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