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Measuring Loudspeaker Low-Frequency ADVERTISEMENT

Response
 

 November 14 2018, 05:10

With the availability of low-cost, PC-based acoustic data acquisition systems, experienced hobbyists
can make highly accurate loudspeaker frequency-response measurements without an anechoic ADVERTISEMENT
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These systems window the time-domain measurement to eliminate room re ections. This, in turn,
limits low-frequency response. Typically, response below 200 to 300 Hz is not possible in reasonably
sized rooms. Industry Events
 

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The near- eld technique — which was proposed by D. B. Keele, circa 1973 — is the commonly with Uni-Core Technology
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Figure 1: Floor re ection geometry.

Typically, if the speaker under test is placed on the middle of the oor, far from re ecting walls, the
rst re ection comes from the oor (see Figure 1). In this gure, the test driver and the test
microphone are both at a 1-m height. The direct distance, d1, from driver to microphone is also 1 m.
The wave re ected by the oor travels the longer path, 2d2 (see Figure 1).

Whenever the distance 2d2 to d1 is equal to an even multiple of a wavelength, the direct and re ected
waves will directly add. Whenever this distance is an odd number of half wavelengths, the re ected
wave will be 180° out-of-phase and subtract from the direct wave. At intermediate distance-to-
wavelength ratios, there will be partial addition or subtraction. The re ected wave will be somewhat
weaker than the direct wave, since it travels a longer distance, so the direct wave will not completely
cancel.
 

Figure 2: Measured impulse response of small two-way monitor.

Figure 2 is a plot of the measured impulse response of a small two-way monitor loudspeaker. The
MLSSA system was used for this measurement. Additional details on the measurement equipment are
outlined at the end of this article. The measurement geometry is similar to that of Figure 1.
Remember, the impulse response is the time-domain equivalent of frequency response. The two are
related by the Fourier transform.

In this test, the direct on-axis wave arrives at the microphone about 2.8 ms after the test signal is
applied to the loudspeaker. The oor re ection arrives at 6.3 ms and then there is 3.5 ms of re ection
free data. If only the data between markers m1 and m2 is analyzed, the relatively smooth frequency
response is shown as curve A of Figure 3.
 

Figure 3: Response data: no bounce (A) with oor bounce (B).

Extending the analysis up to marker m3 would now include the oor re ection and get the response
shown in curve B of Figure 3. (The curves are o set by 10 dB for clarity.) Now you see the e ect of the
alternating addition and subtraction on the direct response caused by the re ected wave. Because the
full anechoic response is not achieved with the windowing process, this response is often termed
“quasi-anechoic.” And there is a catch! By using only 3.5 ms of data in this example, the lowest
frequency resolved is:

fMIN = 1/0.0035 = 285 Hz [1]

This is shown by the black bars in Figure 3 at the top and bottom of the plot. Any part of the curve
plotted below that frequency is simply an artifact of the Fourier transform and does not represent
valid data. Fortunately, there are ways to get the low-frequency data, which can then be spliced to the
high-frequency data to get the full range response. I’ll talk about them next.

One Solution: The Near eld Technique

In the near- eld technique, the microphone is placed very close to the driver diaphragm to swamp out
ba e and room e ects. At low frequencies where the driver diaphragm behaves like a rigid piston, the
measured near- eld response is directly proportional to the far- eld response and independent of the
environment into which the driver radiates. Keele describes this technique in his paper.[1] I will
summarize the approach and its limitations here.

For the near- eld technique to work properly, the microphone should be placed as near to the center
of the diaphragm as possible. Keele shows that a microphone distance less than 0.11 times the
diaphragm e ective radius results in measurement errors of less than 1 dB. As an example, a 6.5"
driver will typically have an e ective cone diameter of 5" or an e ective radius of 2.5". For this driver,
the microphone should be placed within 0.275" of the driver dust cap.

At higher frequencies, where cone break up begins, pressure waves from various areas of the
diaphragm may arrive at the microphone out-of-phase, causing near- eld response cancellations that
are not observed at normal listening distances. For this reason, there is a practical upper limit to the
near eld technique given in terms of driver diaphragm diameter. For a driver mounted in an in nite
ba e, the limit is:

fMAX = 4,311/D [2]

Here fMAX is in hertz and the driver diameter, D, is in inches. For closed-box or ported systems with
nite ba es, this limit may be slightly lower. A 6.5" driver example has:

fMAX = 4,311/5 = 862 Hz [3]

What happens when there are multiple radiating surfaces, as in a vented loudspeaker? Keele has the
answer for that, too. He shows that individual near- eld responses may be added with proper
weighting to get the total near- eld response. If all the radiating surfaces are circular, the addition
looks like this:

PTOT = D1p1 + D2p2 + D3p3 +.....+DNPN [4]

PTOT is the total near- eld pressure. PN is near- eld pressure of the nth circular radiating surface. DN
is the corresponding diameter of the nth radiating surface. For example, if you are testing a vented
loudspeaker with two woofers and two port tubes, you would take a total of four near- eld
measurements and add them together, after multiplying each one by its respective diameter.

If some of the radiating surfaces are rectangular, you can use the diameter of a circle with the same
area. Alternatively, you can weigh all measured near eld pressures by the square root of the area,
each respective radiating surface before adding them together.

The Keele approach seems pretty straightforward, but you must be careful of some things. First, Keele
assumes all radiating surfaces are mounted on an in nite ba e. Under this condition, the radiation is
into a “half-space” or a solid angle of 2π. However, most loudspeakers have relatively narrow ba es
so they become omnidirectional at low frequencies. For this reason, the Keele approach may over
estimate the low-frequency sound pressure level. 

Second, the near- eld pressures are complex quantities — that is, they have both magnitude and
phase. If more than one radiating surface is involved in the testing, a simple pressure magnitude
measurement is not enough. You need a system that measures both magnitude and phase to sum the
responses correctly. Third, if radiating surfaces are close together, measurements may be
contaminated by crosstalk. Finally, the upper-frequency limit on port pressure measurements tends to
be much lower than a diaphragm of the same diameter. Even with these caveats, the technique is
useful when there is no anechoic chamber.

It’s time for an example. Consider the far- eld on-axis response of the two-way monitor rst examined
in Figure 2. Let’s look at the low-frequency response using the Keele approach. This speaker is vented
so, both the woofer and port near- eld responses must be measured. The results are plotted in Figure
4a. For the port measurement, the microphone was placed in the plane of the port exit.

At rst glance, the port output seems to be 3 to 5 dB higher than the woofer output, which is a counter
intuitive result. This is due to the diameter di erence between the woofer and port. The port output
can be scaled to the correct relative level by writing the summing equation in a di erent form:

pTOT = pWOOFER + DPORT/DWOOFER * PPORT [5]

For this example, the woofer e ective diameter is 14 cm and the port diameter is 5.5 cm, which gives:

DPORT / DWOOFER = 5.5/14 = 0.39 = −8.1 dB [6]

The scaled version of the port response is also plotted in Figure 4a. Now you can see that it is more in
line with the woofer level. Before leaving Figure 4a, there are two interesting points not directly related
to near- eld testing. First, the sharp dip in woofer response at 37.6 Hz indicates the tuning frequency
of the vented enclosure. In general, this value is more accurate than one obtained from the
impedance curve, since it is not corrupted by voice coil inductance.
 

Figure 4a: Near- eld woofer, port and scaled port responses.

Figure 4b: Weighted sum of woofer and port near- eld responses.

Second, the up tick in port response around 340 Hz is caused by a standing wave associated with the
internal height of the enclosure. When added to the woofer response, it produces a small dip in the
total near- eld response at the same frequency. However, this is not heard in practice because it is
almost 15 dB lower than the woofer output and because the port exhausts to the rear of our two-way
monitor example.

Adding the woofer and scaled port near- eld responses provides the complete low-frequency
response. This result is plotted in Figure 4b. At this point, the shape of the loudspeaker’s low-end
response is apparent. Now, splice it to the measured far- eld on-axis response of our two-way
monitor example.

The near- eld data are valid below 780 Hz, but we have the small dip at 340 Hz. The far eld data are
valid above 285 Hz. Clearly, the two graphs should be joined somewhere in the 285-to-340-Hz range.
Regardless of the point chosen, the near- eld response should always be brought into coincidence
with the far- eld response since the latter represents the true loudspeaker sensitivity.
 

Figure 5a: Two-way monitor far- eld and near- eld responses.

Figure 5b: Two-way monitor full range response.

In Figure 5a, I have shifted the near- eld response level to meet the far- eld curve just below 300 Hz.
This point selection is somewhat arbitrary, but the result shown in Figure 5b looks reasonable. Notice
that the response in the 70-to-200-Hz range is slightly elevated relative to the average far- eld
response. This may be a consequence of the “half-space” assumption in Keele’s work. Absent an
anechoic chamber, the near- eld approach provides a good estimate of the low-frequency extension
of the two-way monitor.

The Microphone-in-Box Technique

Using the work of Benson, Small showed that, at low frequencies, there is a simple relationship
between the sound-pressure level at a distance from an enclosure and the internal pressure within
the enclosure, regardless of the number of radiating surfaces.[3, 4] This is an amazing result! 

To determine the low-frequency response of the two-way example, only one measurement of
pressure inside the enclosure is needed! Not only that, but because there is only one measurement,
phase information is not required. The governing equation can be written as follows:

pR = kf2pB [7]

pR is the pressure at a distance outside the enclosure. pB is the pressure inside the enclosure. k is a
constant. f is the frequency in hertz.

Equation 7 is not too useful in its present form. The squaring operation will lead to rather large
numbers. This can be avoided by normalizing the equation to the starting frequency, f0. Furthermore,
the result should be in decibels. So, rst rewrite Equation 7 like this:

pR = k (f / f0)2 pB [8]

k is now a di erent constant. Now, take the logarithm of both sides and multiply by 20 to convert
Equation 8 into an equation in decibels:

20 log (pR) = 20 log (k) + 40 log (f/f0) + 20 log (pB)

or:

pR (dB) oc 40 log (f/f0) + pB (dB) [10]

The symbol “oc ” (∝) can be read as “proportional to.” Recall that the logarithm of a squared
quantity is just twice the logarithm of the quantity itself. Now, the large number squaring has been
avoided. Ignore the term “20 log(k)” since it just adds to the level but does not change the shape of the
curve. To make things even easier, I plotted the following term: 

40 log (f/f0)

It is in decibels in Figure 6 for a starting frequency, f0, of 20 Hz.

Now, the process is very simple. All you need is an AC voltmeter with a decibel scale, an audio
oscillator, a good microphone, and an ampli er. Start at 20 Hz. Increase frequency in 10-Hz steps.
Read the internal pressure in decibels o the AC voltmeter. Add the appropriate correction using the
plot in Figure 6. Repeat these steps until reaching 400 Hz, and plot the results. I have dubbed this
process the “Microphone-in-Box” technique, or MIB. PR is the pressure-response shape. Scaling it to
the far- eld response is still necessary.

As with the near- eld approach, there are some caveats to consider when using the MIB technique.
First, there is the same half-space assumption as in the near- eld approach. Second, it is assumed that
the pressure pB within the enclosure is uniform. Once standing waves build up, the equation breaks
down. Small thought the data would be good up to a frequency where the largest dimension of the
enclosure equals 0.125 of a wavelength. 

My experience indicates that this assumption is somewhat pessimistic. In practice, it is fairly obvious
from the data where the technique breaks down. There are also some e ects at higher frequencies
due to enclosure losses that I will not discuss here.
 

Figure 6: The 40*log(f/f0) in dB.

Figure 7: Two-way monitor MIB measurements.

Let’s take a second look at our two-way monitor with the MIB technique. I have programmed my
MLSSA system to take the in-box measurement and automatically apply the correction curve of Figure
6. I passed a microphone through the port tube and placed it close to the geometric center of the
enclosure. Figure 7 shows the in-box pressure measurement taken at that location. The in-box
pressure response peaks close to the box tuning frequency of 37.6 Hz. Above and below that
frequency in-box pressure response fallso by 12 dB/octave. After applying the correction, you get the
low-frequency response also shown in Figure 7. It is clear in this example that the MIB technique
breaks down somewhere above 250 Hz due to the standing wave within the enclosure.
 

Figure 8: Comparing near- eld and MIB responses.

It is interesting to compare the two low-frequency response methods. This comparison is plotted in
Figure 8.

Agreement is quite good between 50 and 200 Hz. However, below 60 Hz, the near- eld result rolls o
more quickly than the MIB result. I tend to believe the MIB result below 50 Hz. There are two reasons
for this. First, below fB, the woofer and port are out-of-phase, so the response in that range results
from a subtraction of two relatively large quantities to get a small di erence. A small error in the
e ective diameters of either the port or woofer, or both, will lead to a response error here. Second,
the near- eld signal-to-noise ratio (SNR) is poor at very low frequencies, requiring several averages of
the data to get a reliable result.

This point is illustrated in Figure 9. Here the woofer near- eld data is plotted after four averages. The
MIB in-box pressure is not averaged at all. The input signal level is the same for both plots. The
di erence is obvious. I used 16 averages to get the near- eld results shown in this article. Of course,
you can increase the test-signal level to get a better SNR, but you must be careful. If the port
velocity gets too high, the port response becomes nonlinear and the results are invalid.
 

Figure 9: An illustration of the signal-to-noise issue.

As a nal example, let’s look at results from the testing of a sealed-box subwoofer with two 12"
drivers. These tests were run to get the unequalized subwoofer response. The data was then used to
design an electronic equalizer for the subwoofer. Near- eld and MIB data are plotted in Figure 10.
Near- eld measurements of both woofers were taken and added together. The woofers are mounted
on opposite sides of the enclosure so there was little chance of cross-contamination. 

The near- eld magnitude responses of the two woofers were identical and could be added together
without regard to phase. Fortunately, in this example, there was no port response to consider. A 0.75"
hole had to be drilled in the test box to insert the microphone for the MIB test. The space around the
microphone cable was sealed with Blue Tac.

Looking rst at the near- eld result, it is relatively smooth from 10 Hz all the way out to 400 Hz. There
is a gentle rise of about 1 dB in response centered around 60 Hz. The MIB response is at in this
region, but breaks down above 200 Hz. Both responses show the expected 12 dB/octave roll o below
40 Hz.

Which one is right? It is di cult to know. But remember, both techniques just give us the response
shape. We can change the relative level. If we set the two responses equal at 100 Hz, the curves then
di er by no more than 1 dB anywhere from 10 Hz to 160 Hz. For design purposes, I averaged the two
responses. The result is shown in Figure 11. For subwoofer equalizer design, data up to 200 Hz was
su cient. 
 

Figure 10: Example of a subwoofer.

Figure 11: Average of near- eld and MIB subwoofer responses.

Reliable Estimates
Both techniques described in this article produce reliable estimates of loudspeaker low-frequency
response in the absence of an anechoic chamber. Both techniques agree reasonably well given their
di erent approaches to the problem. On the plus side, the near- eld approach tends to be valid over a
wider frequency range. But, except for the single-woofer case or the sealed-box subwoofer example,
phase information is required to properly add the individual radiating surface responses. Also,
response error below a vented speaker’s tuning frequency is possible. Increasing the drive level for
better SNR can help in this case.

The big advantage of the MIB technique is that it requires only a single measurement of in-box
pressure to get speaker response, regardless of the number of radiating surfaces. As a result, phase
data is not needed and manual point-by-point measurements may be used.

Generally, the valid frequency range for the MIB measurement is smaller than that of the near- eld
approach. The microphone should be placed near the geometric center of the enclosure away from
walls and interior ba es. With seal-box systems, getting the microphone into the enclosure may
present a problem. aX

Author’s note: The test equipment listed below was used to develop the data presented in this article.

Sources
MLSSA 10.3 acoustic data acquisition card
DRA Laboratories | www.mlssa.com

B&K 4191 0.5" Laboratory-grade condenser microphone, B&K Type 2669 Pre-amp microphone pre-
amp, and B & K Microphone calibrator
Brüel & Kjær | www.bksv.com

Listen Sound Connect microphone power supply and ampli er

Listen, Inc. | www.listeninc.com

CLIO Model 2 Power amp for speaker testing

Audiomatica | www.audiomatica.com

Parts Express
Dayton Audio OmniMic V2 Computer Based Precision Room Measurement System
www.parts-express.com

SoundEasy Software
Bodzio Software Pty. Ltd
www.bodziosoftware.com.au
Also available through Parts Express
https://www.parts-express.com/soundeasy-v23-loudspeaker-design-software-with-usb-key-
-500-912

References
[1] D. B. Keele, “Low-Frequency Loudspeaker Assessment by Near-Field Sound Pressure
Measurement,” Journal of the Audio Engineering Society, 1974.

[2] J. A. D’Appolito, Testing Loudspeakers, Audio Amateur Press, 1998.

[3] J. E. Benson, “Theory and Design of Loudspeaker Enclosures,” IREE (Australia), 1969.

[4] R. H. Small, “Simpli ed Loudspeaker Measurements at low Frequencies,” Journal of the Audio
Engineering Society, 1972.

For more about audio measurements read also Stuart Yaniger's article series on
Practical Test & Measurement Sound Cards for-Data Acquisition in Audio Measurements

This article was originally published in audioXpress, June 2012.

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