Live Sound International Tech Topic Why Measured Response

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Tech Topic: What You See Is Not What You Get

Why measured response doesn’t always match what’s heard

By Pat Brown

The measured versus “ideal” response for the direct field of

a loudspeaker.

Many audio field technicians are now in

possession of measurement systems that

can be used to assist the listening process

in equalizing sound reinforcement systems.

But, they’re often surprised to find that the

measured system response correlates poorly

with subjective impression of how the

system sounds.

In other words, the system can sound good

when it looks bad on the analyzer, and it

can sound bad when it looks good on the

analyzer. As a result, some users have

become frustrated and distrustful of analysis

systems in general.

lLet's look at why the eye and ear do not always agree on what is best regarding the response of the sound system.

First, consider the most popular methods of measuring the response of the sound system. By “response,” I am referring to the

magnitude of the frequency response as displayed on a dB (vertical) vs. logarithmic (horizontal) scale. The goal of technical

system equalization is to produce a “flat” horizontal line on this display.

The effect of increasing distance outdoors (top) versus

indoors.

WORKING IN REAL TIME

The real-time analyzer (RTA) is essentially a

bank of meters, each driven with a

1/n-octave constant percentage bandwidth

filter so that only the level of a limited

range of frequencies is displayed by each

meter.

The original RTAs used analog meters, but

current versions use a vertical row of LEDs

for each 1/n-octave band. One-third octave

resolution is the most popular, and

correlates well with the response of the

human auditory system.

The RTA input is fed from an

omnidirectional test microphone located at a

listener position. Omni’s are used because

they typically have a very flat, “benign”

frequency response over most of their band

pass.

RTAs can also be software-based, utilizing

the sound card on a personal computer to

provide the A/D conversion of the

microphone output voltage.

A mathematical algorithm (the FFT) is used to produce the previously described dB vs. frequency display. These “digital”

analyzers emulate their analog counterparts in how the information is displayed, but differ in that the filters and display is the

product of a computer algorithm rather than analog filters. This type of RTA is more versatile, as the octave-fractions, colors,

etc. are under software control.

A target curve can be used with the RTA to compensate for

the low-frequency build-up that occurs in many rooms.

Regardless of which type is used, the

standard method-of-use is to drive the

sound system with pink noise (equal energy

per 1/n octave) and adjust the system

equalizer for a “flat” magnitude response on

the analyzer display.

RTAs are powerful tools when certain

guidelines are followed, but indoors they

can indicate a system response with poor

correlation to what the listener is hearing.

The major consideration is the placement of

the measurement microphone.

If the mic is placed in the near field of the loudspeaker (typically less than 10 feet), the correlation with human hearing is

pretty good. At this position, the direct energy from the loudspeaker dominates what is being observed on the analyzer and

very little of the reflected energy from the room is included in the displayed response. Adjustment of the equalizer for a flat

direct sound field on the analyzer produces a desirable result.

The down side to the near-field placement is that the measured response is very sensitive to small vertical movements of the

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microphone when the loudspeaker has offset vertical components (as most do). This sensitivity can be reduced if the

microphone is moved to a greater distance from the loudspeaker (into the far field) since the path-length difference back to

the individual components becomes more equal. But, as the microphone is moved further away, the reflected energy from the

room begins to dominate the displayed response.

GIVING EQUAL WEIGHT

Microphones have no “perceptual” abilities. They do not localize sound or discriminate early sound energy from late energy like

humans do. A listener at a distance remote from the loudspeaker will pay more attention to the direct field of the loudspeaker

than sound that is building-up in the room. A microphone gives equal weight to all energy without regard to where it is coming

from.

A simple experiment to verify this is to stand at the microphone position and listen to the loudspeaker and then route the mic

through a headphone amplifier and listen to it through headphones - not the same thing at all.

A full-bandwidth transfer function measurement (with SIA

SMAART Live) using variable time windows. This

measurement was made indoors at about 50 feet from the

loudspeaker.

Low frequency sounds tend to linger in

rooms longer than high frequency sounds,

because most rooms have more high

frequency absorption than low frequency

absorption.

As such, the room becomes “bass heavy”

when the total sound field is considered.

This extra low frequency information will

dominate what is observed on the RTA, and

the knee-jerk reaction is to attempt to

“flatten” the response by boosting the high

frequency bands on the equalizer. The

result is a system with excessive high

frequency output and a resultant “harsh”

sound quality.

When RTAs are used in this manner, it is

important to equalize to a “target curve”

rather than for a flat frequency response.

The popular “X” curve for theaters is flat to 2 kHz, where it starts rolling off the high frequency response at about 3 dB per

octave. It is -10 dB at 10 kHz relative to 2 kHz. This represents 1/10th power at 10 kHz relative to flat response. The

one-third-octave analyzer and the target curve have served sound practitioners well for years, and remains a viable approach

to system calibration.

RECENT METHODS

Technology has yielded some new methods for acquiring the system response at a listener position. A complex comparison

(both time and frequency information) of the input and output of a system is called the transfer function. It includes both the

magnitude and phase response of the loudspeaker/room at the microphone position. This has become a popular method of

analysis, as it allows any input stimulus to be used to test the system, since the displayed response is just the difference

between “what you put in” and “what you got out.”

Placing the test microphone on a stand makes it impossible

to observe the loudspeaker’s response without interference.

Transfer function analysis has the added

advantage of the ability to use a “time

window” to exclude late arriving energy

from consideration in the response. This can

prevent the low-frequency build-up problem

that plagues traditional real-time analysis.

With proper implementation of a time

window, the system response can be

adjusted without the need for frequency

weighting via a target curve.

A major difference between transfer

function analysis and 1/n-octave real-time

analysis is that the former requires the

removal of the signal delay between the two

signals being compared.

The stimulus (the reference signal) always

has a much shorter path back to analyzer

input than the output of the measurement

microphone. Sources of delay include the

travel time through the air and the latency

of digital processors.

Failure to properly synchronize the reference signal and the microphone’s signal will result in an erroneous display of the

system’s response. The length of the time window must also be selected - in other words, “how much of the room decay do I

want to include in the response?”

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The test microphone was placed on a stand for this

measurement. Note the comb filtering due to the floor

bounce effect.

Unfortunately, there is not an optimum size

for the entire spectrum. A short time

window excludes much of the room decay at

the expense of low-frequency resolution. A

long time window improves frequency

resolution at the expense of gathering too

much of the room’s decay. A compromise is

required.

The human auditory system perceives pitch

on a proportional (logarithmic) frequency

scale. This is one reason that we use

constant-percentage bandwidth filters for

tuning audio systems - the bandwidth grows

with increasing frequency.

Frequency-dependent bandwidth suggests that the length of the windowing function used in transfer function analysis should

be varied in the same manner - a decreasing length with increasing frequency. This produces a somewhat “anechoic” response

at high frequencies with increasing frequency resolution as frequency decreases. The time window length is a function of

frequency, with even the longest window (highest frequency resolution) excluding much of the late energy from the room.

Another caveat of this type of analysis is that much greater frequency detail is possible than with the typical 1/3-octave

banded display. Phase interference effects from reflections or multiple drivers are clearly visible on the analyzer. Such

anomalies are almost always position-dependent, so careful “corrections” at one seating position will be inappropriate for

another. Both the loudspeaker and the measurement microphone should be carefully positioned to avoid the creation of very

early high-level reflections.

SPECIAL EFFECTS?

The “floor bounce” effect is a common example of a very early reflection (typically within a few milliseconds of the first sound

arrival) that produces a unique acoustic response for each listener seat for all but the lowest octaves of the spectrum. This is

an example of “less is better” when measuring the response, as a 1/3-octave display lacks the resolution to observe the effect

in detail and produces less of a temptation to “fix” it.

The same measurement displayed at one-third octave

resolution. The floor bounce effect is significantly masked by

the lower resolution.

The floor bounce effect can be minimized by

use of an appropriate frequency-dependent

time window or by simply laying the

measurement microphone on the floor, or

on a board placed across the listener seats.

The effect usually disappears with the

presence of an audience, so we do not wish

to consider it when tuning the sound

system.

The use of variable-length time windows

and the synchronous transfer function allow

the system to be tuned in a manner similar

to the near-field RTA method (flat response

on a log frequency display), even at remote

positions in the room. It is superior to the

RTA method in that the effects of air

absorption are readily apparent and can be

compensated for via equalization.

Near-field techniques do not include air absorption for the simple fact that the sound has not traveled very far before it is

picked up by the microphone, so it hasn’t passed though enough air to be significantly attenuated.

By far, the biggest problem with tuning sound systems is failure on the part of the technician to recognize anomalies that

cannot be corrected with equalization. The equalizer is a “global” device, meaning that its response curve will be impressed on

all of the sound radiated from the loudspeaker, regardless of the direction in which it is radiated.

Many, if not most, of the anomalies observed on the analyzer are unique to each listener position. The technician must learn to

recognize and ignore such events. They include:

• Floor-bounce effect;

• Interference between multiple drivers;

• Reflections from objects near the mic or loudspeaker.

Events that produce a more global effect, and can therefore be addressed with equalization include:

• Boundary-loading of loudspeakers;

• Coupling between multiple low-frequency drivers;

• The direct-field loudspeaker response.

With training and experience, the system technician can implement methods that reveal system imperfections that are

correctable, and hide those that are not - regardless of the analysis method used. Better yet, system designers can design

systems with fewer “un-equalizable” problems.

BAD IS ALWAYS BAD

The old adage “an ounce of prevention...” could never be MORE true. System equalization then becomes meaningful and fast,

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providing the “icing on the cake” of the performance of a sound system. It makes a good loudspeaker sound better, and brings

the system to its fullest potential given the acoustic environment into which it is placed. A bad room is a bad room, regardless

of how we process the electrical signal that drives the loudspeakers.

The microphone was placed on the floor for this

measurement. Anomalies inherent to the loudspeaker are

now visible on the analyzer.

When used properly, the traditional

1/n-octave real-time analyzer is a useful

tool outdoors at any distance. Indoors, the

effects of reflected sound and

non-frequency-uniform room absorption

produce some problems for this method at

measurement distances remote from the

loudspeaker.

One solution is to utilize a weighting curve

that reduces the target level of the

high-frequency portion of the spectrum.

Attempts to achieve a flat system response

at remote listener positions without the use

of a weighting curve can result in

harsh-sounding systems and even

component damage.

Transfer function analysis addresses some of the shortcomings of the 1/n-octave RTA, but it requires greater expertise on the

part of the user. Failure to properly compensate for the time differential between the reference and measured signal can

produce wildly erroneous results. The time window length must also be selected by the user, and different lengths will produce

different displayed responses. A frequency-dependent time window produces a display that correlates well with human

perception.

The most important feature of either measurement method is a knowledgeable operator - one who understands the caveats of

each approach along with the basic characteristics of the human auditory system. None of the questions raised here have a

single, correct answer. This means that experience, good judgment, and common sense rooted in Newtonian physics are still a

part of the measurement process.

Sound is a relatively easy quantity to measure, but measurements that correlate with human perception are much more

difficult. Analyzers driven by omni directional microphones do a poor job of emulating the human listener. At this point one

could ask, “So why measure at all? Why not just listen?” Next issue, we’ll have a look at this provocative question.

Pat and Brenda Brown own and operate Syn-Aud-Con, conducting audio training sessions around the world. For more info, go

to

www.synaudcon.com

October 2003 Live Sound International

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