STONES SOUND STUDIO    
Loudspeaker System & Driver Measurements

 
                                                       
by  Russell Storey
                                   ( Updated   Aug-2012 ) 

   
 

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  "d
epending 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 th
e 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
positioning
All 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 equalisation
Using 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