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| Equalizing
the room
By Bob McCarthy
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Modern Analysis
Technological progress led to the development and acceptance of
two analysis techniques in the early 80s: Time Delay Spectrometry
(TDS) and dual-channel FFT analysis. Both of these systems brought
to the table whole new capabilities, such as phase response measurement,
the ability to identify echoes and high-resolution frequency response.
No longer could an unintelligible pile of junk look the same as
the real McCoy on an analyzer. The complexity of these analyzers
required a well-trained, highly skilled practitioner in order to
realize the true benefits.
Advocates of both systems stressed the need for engineers to utilize
all tools in their system, not equalizers alone, to remedy the response
anomalies. Delay lines, speaker positioning, crossover optimization
and architectural solutions were to be employed whenever possible.
And now we had tools capable of identifying the different interactions.
But on the issue of “equalizing the room” a division arose.
All parties agreed that speaker/speaker interaction was somewhat
equalizable. The critical disagreement was over the extent the speaker/room
interaction could be compensated by equalization.
The TDS camp advocated that speaker/room interaction was not at
all equalizable and therefore, the measurement system should screen
out the speaker/room interaction, leaving only the equalizable portion
of the speaker system on the analyzer screen. Then the inverse of
the response is applied via the equalizer and that was as far as
one should go.
The TDS system was designed to screen out the frequency response
effects of reflections from its measurements via a sine frequency
sweep and delayed tracking filter mechanism, thereby displaying
a simulated anechoic response. The measurements are able to clearly
show the speaker/speaker interaction of a cluster and provide useful
data for optimization.
Such an approach can be effective in the mid and upper frequency
ranges where the frequency resolution can remain high even with
fast sweeps but it is less effective at low frequencies. Low frequencies
have such long periods that it is impossible to get high-resolution
data without taking long time records, thereby allowing the room
into the measurement.
For example, to achieve 1/12th octave resolution, the equivalent
to the Western Tempered Scale, one must have a time record 12x longer
than the period of the frequency in question. For 30 Hz you will
need a 360ms (12x30ms). If fast sweeps are made to remove echoes
from the measurement, the low frequency data has insufficient resolution
to be of practical use.
Dual-channel FFT analyzers utilize varying time record lengths.
In the HF range, where the period is short, the time record is short.
As the frequency decreases, the time record length increases, creating
an approximately constant frequency resolution. The measurements
reveal a constant proportion of direct sound and early reflections,
the most critical area in terms of perceived tonal quality of a
speaker system.
The most popular FFT systems utilize 1/24th octave resolution, which
means that the measurements are confined to the direct sound and
the reflections inside a 24 wavelengths time period across the board.
This is a good practical level of resolution, allowing us to accurately
equalize at around the 1/8 octave level.
With the FFT approach, more and more of the room enters the response
as frequency decreases. This is appropriate because at low frequencies
the room/speaker interaction is still inside the practical equalizability
window.
For example, the arena scoreboard reflection is 150 ms later than
the direct signal. At 10 kHz, the peaks and dips from this reflection
are spaced 1/1500 of an octave apart. At 30 Hz, they will be only
1/3 octave apart. Thus the scoreboard is in the distant field relative
to the tweeters, and applying equalization to counter its effects
will be totally impractical.
An architectural solution such as a curtain would be effective.
But for the subwoofers, the scoreboard is a near-field boundary
and will yield to filters much more practically than the 50 tons
of absorptive material required to suppress it acoustically.
Many years ago, the FFT camp boldly stated that the echoes in the
room could be suppressed through equalization. Unfortunately, these
statements were made in absolute terms without qualifying parameters,
leaving the impression that the FFT advocates thought it was desirable
or practical to remove all of the effects of reverberation in a
space through equalization.
While it can be proven from a theoretical standpoint that the frequency
response effects of a single echo can be fully compensated for,
that does not mean it is practical or desirable. The suppression
can only be accomplished if the relative level of the echo does
not equal or exceed that of the direct and that no other special
circumstances arise that cause excess delay. (Excess delay causes
a “non-minimum phase” aberration and is outside the scope
of this article.)
If the direct level and echo level are equal the cancellation dip
becomes infinitely deep and the corresponding filter required to
equalize it is an infinite peak. As we know from sci-fi movies,
bad things happen when positive and negative infinity meet up.
Compensating for the response requires adjustable bandwidth filters
capable of creating an inverse to each comb filter peak and dip
in the response. As the echo increases, you will need increasing
numbers of ever narrowing filters.
A 1ms echo corrected to 20 kHz will require some 40 filters because
there are 20 peaks and 20 dips varying in bandwidth from 1 to .025
octave. A 10 ms echo would need 400 with bandwidths down to an 1/400
octave.
Obviously, it would be insane to attempt to remove all of the interaction
at even a single point in the hall. In the practical world, we have
no intention of attacking every minuscule peak and dip, but instead
will go after the biggest repeat offenders. The narrower the filters
are, the less practical value they have because the response changes
over position.
Practical Implications
It is indeed possible and practical to suppress some of the effects
of speaker/room interaction. If this was not possible, it would
be standard practice to equalize your rig in the shop, put a steel
case around the EQ rack and hit the road. The practical side of
this is that we must be realistic about what is attainable and what
are the best means of getting there.
The variations in frequency response due to both speaker/speaker
interaction and speaker/room interaction will always change with
position. Once you have seen high-resolution data at multiple positions,
you can never go back to thinking that your equalization will solve
problems globally.
A system that has the minimal amount of the above interactions will
have the greatest uniformity throughout the listening environment
and, therefore, stand to gain the most practical benefit from equalization.
If it sounds totally different at every seat, let’s just tweak
the mix position and head to catering.
To minimize the speaker/speaker interactions requires directional
components, careful placement and precise arraying. In areas where
the speakers overlap, time delays and level controls will minimize
the damage in the shared area. To minimize speaker/room interaction,
the global solutions lie in architectural modification (it’s
curtain time), the selection of directionally controlled elements
and precise placement.
Finally you are left with equalization. For each subsystem with
an equalizer, map out the response in the area by placing a mic
in as many spots as you can and see what the trends are.
In particular, measure around the central coverage area of the speaker.
Stay away from areas of high interaction, where the response will
vary dramatically every inch. Examples of this include the seam
between two cabinets in an array or very close to a wall. Each position
will be unique, but if you place filters on the top four to six
repeat offenders you will have effectively neutralized the response
in that area.
Conclusion
Modern analyzers are capable of displaying a dizzying array of spectral
data. But little practical benefit will come to us if we continue
with the antiquated approach of the RTA era. To fully take advantage
of the benefits of equalization, we must fully comprehend how to
identify the mechanisms that “unequalize” the system.
With modern tools, it becomes possible to analyze the response such
that the interactive factors of speaker systems can be distilled
and viewed separately. This allows the alignment engineer to prepare
the way for successful equalization by using other techniques that
reduce interaction and maximize uniformity in the system.
“Equalizing the room” will remain in the domain of architectural
acousticians, but with advanced tools and techniques, we can proceed
forward to better equalize the speaker system in the room.
Bob McCarthy specializes in sound system design and alignment
and can be reached at "6o6" bobmcc1@mindspring.com
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