STONES
SOUND STUDIO
Loudspeaker System & Driver Measurements
by
Russell Storey
9999( Updated
Feb-2023 )
|
|
Loudspeaker measurement
is one of the most difficult aspects of audio quality
measurement and also probably the most relevant, since loudspeakers, because
they are transducers, have higher distortion than other audio system components
.
Measurement errors due to temperature compliance and
environment equipment used equipment calibration are a
important
and are described in brief below
Measuring transducer parameters
"Thiele/Small"
parameters is like measuring a " soft balloon " diameter with
a pair of metal verier callipers
" every time you take a measurement
you will get a different reading"
At Stones
Sound Studio
Speakers and Drivers (transducers) are measured
using an array of test equipment Hewlett
Packard Tektronix Pulsed FFT systems in conjunction with
the LMS 4 Loudspeaker Analyser which is calibrated in true SPL db
(Sound
pressure level
).
Test Equipment Calibration:
Our Test
& measurement Equipment
is run for around 2 hours prior to
calibration to warm up the electronics and stabilize
the measurement
circuits .
The Equipment is then individually "calibrated " to
minimise
measuring errors before measurements can be taken.
Taking into account errors in measurements
due to speaker cable and microphone leads etc
Measurement of test lead impedance , microphone db
loss , test speaker cable , cable high and low frequency
insertion losses must be subtracted from the actual
measurement to reduce measurement errors
and provide thus provide accurate measurement data of the
device under test ( speaker or transducer )
Microphone and Pre amp Calibration
First a Brüel
& Kjær
laboratory reference microphone
pressure calibrator @ 94 SPL db is used to calibrate
the microphones
and LMS 4 Speaker Analyser and
pre-amplifier.
SPL db definition re -Wikipedia
Sound pressure level
(SPL) or
sound
level
Lp
is a
logarithmic measure of the
rms sound
pressure of a sound relative to a reference value.
It is measured in
decibels (dB) above a standard reference level
The commonly used reference sound pressure in air is
pref
= 20
µPa (rms)
which is usually considered the
threshold of human hearing (roughly the sound of a
mosquito
flying 3 m away).
Ambient temperature
The Environment ambient temperature is
measured and noted before any measurements are
made.
For frequency response ,measurements
a constant voltage amplitude sine wave sweep set to 2.83V
rms across the speaker terminals or driver
The 2.83V rms across the speaker terminals or driver is
used as a reference point unless otherwise specified .
With Artisan Speakers measurements the microphone is set at a distance of
1 to 2.7 Mtrs "depending on the required measurements "
.
The Microphone
The Microphone is centred between ,at ,and ,or around the tweeter /bass
mid axis on the front baffle depending on the speakers driver /tweeter
configuration be it, MT ,MTM,etc its set different .
Up to 40
different readings are taken over a period 2Hrs as the drivers warm up ,then
the
readings are computer averaged
over a the specific measurement frequency range to
compute the SPL db sensitivity.
Manufacturer Specifications
, Speaker ( transducer ) Sensitivity and graphs errors
SPL db ( pressure level)
measurements of any transducer (speaker)
can vary buy as much as -/+ 3db to -/+ 6db
in the
ambient temperature range from 0 to 45 degs C.
Other
factors (variables) change the transducer
measurement parameters and will affect the SPL db
readings.
Some of the measurement variables to
consider are ambient temperature, humidity ,
barometric pressure
( height above sea level ), transducer motor (magnet) system temperature,
voice coil temperature and
compliance of the cone and
spider materials heating losses in components in the
crossover are all taken into account.
Why, ? All
this if you interested read
more below
Measurement Accuracy and
Measurement errors:
Speaker SPL
db sensitivity ( speaker pressure level) varies depending on the
driver type cone size ( diameter ) materials measurement
location
( room ,car park, shed , hall , anechoic chamber etc ) and
also the measurement type application calibration and
measurement bandwidth
accuracy of equipment and
most importantly the measurement environmental temperature
Speaker
Systems sensitivity in
(SPL )db is usually measured over the 10Hz to 50KHz range (
depending on the type and application of the speaker)
A low distortion constant voltage sine wave of 2.83 V rms ( ref
1KHz) is applied across the speaker terminals with the LMS
loudspeaker Analyzer.
Measurement Bandwidths
Single Drivers( transducers )
Subwoofers 10 to 500Hz,
Woofer 10 to 3KHz ,
Mid /bass 10 to 10 KHz ,
Mid 150 to 12Khz ,
Tweeter
400 to 100 KHz
Measurement
Design and development of High SPL horn
tweeter and diaphragm for
use Communications monitoring
by ,
Fire & Rescue NSW VIC QLD
Ambient temperature
Ambient temperature is the most important factor
when measuring any transducer or loudspeaker system this
is set
normally at 22deg C on average unless other
wise stated.
Speaker temperature under
power
The driver (
transducer ) magnet , voice coil , pole piece temperatures are measured
by Fluke thermal bead direct contact and Laser Infrared
digital temperature meters when doing
IEC
Power or RMS Power Measurements
(See below ) unless other
wise stated.
T/S
Thiele & small parameters New Driver
measurement pre
conditioning mechanical compliance
Movie >
Running in the speakers at
Stones Sound Studio
Laboratory
The Peerless 830883 6.5''
speakers are wired in series /parallel to an
amplifier and driven by a 20 Hz sine wave
for around 8 hours before any measurements
are carried out .
Running in the drivers helps
free up the new surround , spider and lead out wires and also reduces the
resonance frequency and enables changes in
driver mechanical complianceto that of a "real world working
speaker " .Running in the drivers insures more accurate
frequency Impedance and
'' Thiele/Small"
electromechanical parameters that define the specified low frequency
performance of a loudspeaker driver result in more accurate modelling of the crossover and
enclosure design .Running in the drivers in also aids
listening evaluations and improves the speakers bass
mid depth and data
Subwoofer , Woofer , bass /mid
drivers are normally hooked up
to an amplifier with a sine wave source between 20 to 30Hz.
The signal level is adjusted till the cone
excursion is around 70% of the drivers maximum X max (
mm) -/+ specification . The driver is run in
for 6 to 8 hours .
Will loosen the mechanical stiffness of the new
speaker , spider , tinsel and cone surround compliance
Speaker Impedance (T/S
Thiele & small parameters)
Measuring transducer or Speaker
enclosure Impedance or (T/S
Thiele & small parameters) specification is like measuring
a
“sponge with a pair of metal callipers "Every time you take a measurement you will get a
different reading as device , test equipments,
|ambient and internal temperature change
" which measurement is 100% correct ? "the answer is none " the
measurements are only a guide to work from when
designing speaker systems .
At Sound Studio
Studio we have a laboratory with an array of reference
calibrated test equipment to
enable consistent reliable and accurate
measurements for our
customers
and clients " see >>>
Stones Sound Studio , Laboratory
test equipment below
In an ideal world you need to measure several transducer or
speaker samples and then take an average of the readings
however this is not
always practical
Speaker and Driver (transducer ) Power rating
Speaker and Driver (transducer ) Power rating is a very
complex subject and suffers from a lot of debate and arguments world wide
.
Speaker Power rating , rms power handling specifications and
method of testing etc vary from manufacturer to manufacturer and
Engineering Standard
Societies USA and Europe like AES , IEC , ALMA , etc
so unfortunately there are several world standards to chose from
Power Density
Speaker and
Driver (transducer )
Power density
measurements using band limited pseudo random pink/white
noise or multiple modulated sine wave
sweeps can provide a simulated
" average music power" or "Normal speech and music" which
provides a more meaning specification thank a fixed sine wave
or short length pulses as there is a lot more heat and power
energy generated in the driver motor and voice coil , or speaker system
For more
details of IEC
Power Measurements see :
power
test definition-peerless loudspeaker dk
Stones Sound Studio , Laboratory
test equipment
Fig 1
Example of soft dome Tweeter THD -Distortion Test
Fig 1
Example of soft dome Tweeter THD -Distortion Test
Stones Sound Studio , Laboratory
test equipment
Anechoic
measurement
The standard way to test a loudspeaker
requires an
anechoic chamber,
with an acoustically transparent floor-grid. The measuring
microphone
is normally mounted on an unobtrusive boom (to avoid
reflections) and positioned 1 metre in front of the drive
units on axis with the high-frequency driver. While this
will produce repeatable results, such a 'free-space'
measurement is not representative of performance in a room,
especially a small room. For valid results at low
frequencies very large anechoic chamber is needed, with
large absorbent wedges on all sides. Most anechoic chambers
are not designed for accurate measurement down to 20 Hz.Outdoor
measurementMeasurements made outside will usually
show ripples in the mid-range caused by ground reflection
interference. Raising the speaker and microphone helps by
reducing the amplitude of the reflected sound. Positioning
the microphone closer to the speaker helps further, but this
requires it to be moved off the tweeter axis such that the
path lengths from both
tweeter
and mid-range unit are equal[
This usually reduces the high-frequency response, since most
tweeters are very directional at 15 to 20 kHz. If the
microphone is left on the tweeter axis the reduction will
occur in the mid-range (see below). Raising both speaker and
microphone on poles has been used as a way of reducing
ground effect, and some speaker manufacturers specify a
height of 50 feet (15 m) in their measurements. Half-space
measurementAn
alternative is to simply lay the speaker on its back
pointing at the sky on open grass. Ground reflection will
still interfere, but will be greatly reduced in the
mid-range because most speakers are directional, and only
radiate very low frequencies backwards. Putting absorbent
material around the speaker will reduce mid-range ripple by
absorbing rear radiation. At low frequencies, the ground
reflection is always in-phase, so that the measured response
will have increased bass, but this is what generally happens
in a room anyway, where the rear wall and the floor both
provide a similar effect. There is a good case therefore
using such ‘half-space’ measurements, and aiming for a flat
‘half-space’ response. Speakers that are equalised to give a
flat ‘free-space’ response, will always sound very
bass-heavy indoors, which is why monitor speakers tend to
incorporate ‘half-space’, and ‘quarter-space’ (for corner
use) settings which bring in attenuation below about 400 Hz. Digging a hole and burying the speaker flush with the
ground allows far more accurate half-space measurement,
creating the loudspeaker equivalent of the
boundary effect microphone
(all reflections precisely in-phase) but any rear port, must
remain unblocked, and any rear mounted amplifier must be
allowed cooling air. Diffraction from the edges of the
enclosure are reduced, creating a repeatable and accurate,
but not very representative, response curve.
Room
measurements
At low
frequencies, most rooms have resonances at a series of
frequencies where a room dimension corresponds to a multiple
number of half wavelengths. Sound travels at roughly 1 foot
per millisecond (1100 ft/s), so a room 20 feet (6.1 m) long
will have resonances from 25 Hz upwards. These ‘resonant
modes’ cause large peaks and dips in response. A speaker in
a room does not really ‘radiate’ low frequencies at all,
most rooms being smaller than some musically significant
frequency, but in this region instead couples into the
resonant room modes,
which are resonant standing wave patterns. Because this
coupling is in part acoustic
impedance
dependent (and thus reslt from issues in each possible room
or space—though different in every case), it cannot even be
predicted from measurements made of speaker radiation alone.
Put simply, some speakers present a very ‘stiff’ driving
force and will drive a resonant pressure peak at a boundary
more efficiently than a ‘floppy’ one.
Dipole loudspeakers,
such as
electrostatics
or
ribbons,
couple to the room differently, by velocity rather than
pressure (citation?), and are generally thought to less
excite
resonant peaks.Additionally, reflections, dispersion, absorption, etc.
all strongly alter (fortunately or unfortunately) the
perceived sound, not necessarily consciously noticeably for
music or speech, at frequencies above those dominated by
room modes. These depend on speaker location(s) with respect
to reflecting, dispersing, or absorbing surfaces (including
changes in speaker orientation) and on the listening
position. In unfortunate situations, a slight movement of
any of these, or of the listener, can cause considerable
differences. Complex effects, such as stereo (or multiple
channel) aural integration into a unified perceived "sound
stage" can be lost easily.There is limited understanding of how the ear and brain
process sound to produce such perceptions, and so no
measurement, or combination of measurements, can assure
successful perceptions of, for instance, the "sound stage"
effect. Thus, there is no assured procedure which will
maximize speaker performance in any listening space (with
the exception of the sonically unpleasant anechoic chamber).
Some parameters, such as reverberation time (applicable only
to larger volumes in any case), and overall room "frequency
response" can be somewhat adjusted by addition or
subtraction of reflecting, diffusing, or absorbing elements,
but, though this can be remarkably effective (with the right
additions or subtractions and placements), it remains
something of an art and a matter of experience. In some
cases, no such combination of modifications has been found
to be very successful.
Microphone
positioningAll
multi-driver speakers (unless they are
coaxial)
are difficult to measure correctly if the microphone is
placed close to the loudspeaker and slightly above or below
the optimum axis, because the different path length from two
drivers producing the same frequency leads to phase
cancellation. It is useful to remember that, as a rule of
thumb, 1 kHz has a wavelength of 1 ft (0.30 m) in air, and
10 kHz a wavelength of only 1-inch (25 mm). Published
results are often only valid for very precise positioning of
the microphone to within a centimetre or two.
Measurements made at 2 or 3 m,
in the actual listening
position between two speakers can reveal something of what
is actually going on in a listening room. Horrendous though
the resulting curve generally appears to be (in comparison
to other equipment), it provides a basis for real
experimentation with absorbent panels. Driving both speakers
is recommended, as this stimulates low-frequency room
‘modes’ in a representative fashion. This means the
microphone must be positioned precisely equidistant from the
two speakers if ‘comb-filter’ effects (alternate peaks and
dips in the measured room response at that point) are to be
avoided. Positioning is best done by moving the mic from
side to side for maximum response on a 1 kHz tone, then a
3 kHz tone, then a 10 kHz tone. While the very best modern
speakers can produce a frequency response flat to ±1 dB from
40 Hz to 20 kHz in anechoic conditions, measurements at 2 m
in a real listening room are generally considered good if
they are within ±12 dB, and efforts to produce anything like
a flat response below 100 Hz are likely to provide endless
scope for experimentation, and exercise of patience! It is a
major challenge to achieving audio quality. Complex and
expensive
DSP
equipment and state of the art (and so not yet finalized)
algorithms are being used to attempt to address these
issues, but are not yet routinely practicable.
Nearfield
measurements
Room acoustics have much lower effect on
nearfield measurements, so these can be appropriate when
anechoic chamber analysis cannot be done. Measurements
should be done at much lower distances from the speaker than
the speaker (or the sound source, like horn, vent) overall
diameter, where the half-wavelength of the sound is smaller
than the speaker overall diameter. These measurements yield
direct speaker effeiciency, or the average senstivtiy,
without directional information. For a multiple sound source
speaker system the measurement should be carried out for all
sound sources (woofer, bass-reflex vent, midrange speaker,
tweeter...). These measurements are easy to carry out, can
be done at almost any room, more punctual than in-box
measurements, and predicts half-space measurements, but
without directivity information. Minimising
room modes and equalisationUsing
an
equalizer
to correct for room response is a poor solution[citation
needed]
(exception:
digital room correction),
especially at low frequencies, because it relies on reducing
the drive at resonant modes to produce a flat ‘steady state
response’ once the resonant mode has built up and
stabilized, and this can take many tenths of a second. The
result is ‘sluggish’ bass, because the initial wave-front
has been greatly reduced by the equalizer[citation
needed].
Additionally equalization only produces flat response at one
seating position. Bass drums, and bass guitar, produce low
frequencies with sudden onset, and the initial wavefront
accounts for much of the impact that is both heard and felt.
Realistic reproduction requires both the initial radiation
and the steady state level to have a flat response, and
there is no easy way to achieve this — room modes just have
to be eliminated. The commonly recommended approach of
moving speakers around in an attempt to stimulate the
maximum number of
resonant room modes
is also not valid. It amounts to the same thing as using an
equaliser — adjusting the coupling of the speaker to the
mode as a way of controlling the steady state level, but at
the expense of the initial
wavefront,
with sluggish results.It should be clear from the above that marketing claims
regarding a
bass
driver is ‘fast’ or 'quick' are unfounded. Some driver
manufacturers claim that smaller bass drivers are ‘faster’,
or that they have a quicker transient response. While a
light cone is easier to accelerate, the result is that light
cone can reproduce higher frequencies. Given that a driver
can generate a given frequency, its ability to generate
higher frequencies (within its bandwidth) has little to do
with speed. Provided that the driver is operating at
reasonably low ‘Q
factor’ (a feature of the
driver plus its enclosure) then its contribution to any
sluggishness of bass response is negligible. Vented speaker
systems suffer a modest amount of 'group delay' at very low
frequencies, but the human ear is not sensitive to them, and
vented systems remain popular because their minor
deficiencies are typically swamped by room modes. Frequency
response measurement
Frequency response measurements are only
meaningful if shown as a graph, or specified in terms of ±3
dB limits (or other limits). A weakness of most quoted
figures is failure to state the maximum
SPL
available, especially at low frequencies. Because of the way
in which the sensitivity of our ears falls off as shown in
equal-loudness contours
it is desirable[citation
needed]
that a speaker should be able to produce higher levels below
100 Hz, whereas in fact most are limited by cone-excursion
to lower levels. A
power bandwidth
measurement is therefore most useful, in addition to
frequency response, this being a plot of maximum SPL out for
a given distortion figure across the audible frequency
range. Specifications like 'Frequency response 40 Hz to
18 kHz', which are common, are valueless. The situation is
worse for headphones, with manufacturers quoting figures
like '4 Hz to 22 kHz' for headphones that are far from flat
and often as much as 20 to 30 dB down at 4 Hz. Distortion
measurement
Distortion measurements
on loudspeakers can only go as low as the distortion of the
measuring microphone
itself of course, at the level tested. The microphone should
ideally have a clipping level of 120 to 140 dB SPL if
high-level distortion is to be measured. A typical top-end
speaker, driven by a typical 100watt
power amplifier,
cannot produce peak levels much above 105 dB SPL at 1 m
(which translates roughly to 105 dB at listening position
from a pair of speakers in a typical listening room).
Achieving truly realistic reproduction requires speakers
capable of much higher levels than this, ideally around 130
dB SPL. Even though the level of live music measured on a
(slow responding and rms reading)
sound level meter
might be in the region of 100 dB SPL,
programme level
peaks on percussion will far exceed this. Most speakers give
around 3% distortion measured 468-weighted 'distortion
residue' reducing slightly at low levels. Electrostatic
speakers can have lower harmonic distortion, but suffer
higher intermodulation distortion. 3% distortion residue
corresponds to 1 or 2%
Total harmonic distortion.
Professional monitors may maintain modest distortion up to
around 110 dB SPL at 1 m, but almost all domestic speaker
systems distort badly above 100 dB SPL.Colouration
analysisLoudspeakers differ from most other items
of audio equipment in suffering from 'colouration'.
This refers to the tendency of various parts of the speaker:
the cone, its surround, the cabinet, the enclosed space, to
carry on moving when the signal ceases. All forms of
resonance
cause this, by storing energy, and resonances with high
Q factor
are especially audible. Much of the work that has gone into
improving speakers in recent years has been about reducing
colouration, and Fast Fourier Transform, or FFT, measuring
equipment was introduced in order to measure the delayed
output from speakers and display it as a time vs. frequency
waterfall plot or
spectrogram
plot. Initially analysis was performed using
impulse response
testing, but this 'spike' suffers from having very low
energy content if the stimulus is to remain within the peak
ability of the speaker. Later equipment uses
correlation
on other stimulus such as a
Maximum length sequence
analyser or
MLSSA.
Using multiple sine wave tones as a stimulus signal and
analyzing the resultant output, Spectral Contamination
testing provides a measure of a loudspeakers 'self-noise'
distortion component. This 'picket fence' type of signal can
be optimized for any frequency range, and the results
correlate exceptionally well with sound quality listening
tests.
-
STONES SOUND STUDIO
Loudspeaker System & Driver Measurements
by
Russell Storey
Draft
10 _610