Loudspeaker measurement is one of the most difficult aspects of audio quality measurement, and also probably the most relevant, since loudspeakers have long been generally acknowledged to be the 'weak link' in the audio chain.
The standard way to test a loudspeaker requires a fully 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 20Hz.
Measurements made out of doors will usually show ripples in the mid-range caused by ground reflections interfering. Raising both 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 20kHz. If the microphone is left on the tweeter axis the 'suckout' 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 in their measurements.
An 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 absorbant 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 400Hz.
Digging a hole and burying the speaker flush with the ground has even been tried, for even more accurate half-space measurement, creating the loudspeaker equivalent of the boundary effect microphone (all reflections precisely in-phase) but any rear port, must of course be kept open, and any rear mounted amplifier must be allowed cooling air! Refraction effects off the edges of the cabinet are also eliminated, which is not really desirable as these will affect actual performance in use.
Measurements on a single speaker in a listening room, though less satisfactory, can be interesting. 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 (1100ft/s), so a room 20 feet long will have resonances from 25Hz upwards. These resonant modes cause large peaks and dips in response. A speaker in a room does not really radiate low frequencies at all, but couples into the room modes which are resonant standing wave patterns. Because this coupling is impedance dependant, 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 ribbon loudspeakers, couple to the room differently, by velocity rather than pressure, and are generally thought to excite resonant peaks less.
Microphone positioning and mid-range 'suckout'
All multi-driver speakers (unless they are co-axial) are prone to suck out in the crossover region between units (often around 3-5kHz) if the microphone is placed 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, 1kHz has a wavelength of 1 ft in air, and 10kHz a wavelength of only 1 inch. Published results are often only valid for very precise positioning of the microphone to within a centimetre or two, and in fact a speaker that has been designed to give a flat measurement at 1m will not be quite so flat at 2m, a better distance for measurement, but seldom used because of the increased errors caused by residual room reflections.
Measurements made at 2 or 3m, in the actual listening position between two speakers can be much more revealing of what is actually going on in a listening room. Horrendous though the resulting frequency response may be, it provides a basis for real experimentation with absorbent panels or resonators. Driving both speakers is recommended, as this is the only way to stimulate low-frequency room modes in a representative fashion driving at one corner would be very bad. This does mean though that the microphone must be positioned precisely equidistant from the two speakers if a comb-filter effect of alternate peaks and dips is to be avoided. Positioning is best done by moving the mic from side to side for maximum response on a 1kHz tone, then a 3kHz tone, then a 10khz tone. While the very best modern speakers can produce a frequency response flat to +/- 1dB from 40Hz to 20kHz in anechoic conditions, measurements at 2m in a real listening room can be considered good if they are within +/-12dB, and efforts to produce anything like a flat response below 100Hz are likely to provide endless scope for experimentation! This is where the real challenge of audio quality lies
Measurements at less than 1m will not allow proper combining of the radiation from separate drivers around the crossover frequency. Even if the path lengths are the same the tweeter will be off-axis. Very close measurements can be useful though. Placing a microphone carefully on axis a 10 to 50 centimetres in front of a tweeter will allow the high-end response to be measured with minimal room effect. Similarly, placing one a few centimetres in front of the voice coil of a bass driver will give some indication of how it is responding (but not of the output from ports if these are present).
Minimising room modes and equalisation
Using an equaliser to correct for room response is a poor solution, 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 stabilised, 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 equaliser. 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 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 wave-front, with sluggish results. It should be clear from all this that the concept of a fast bass driver is utter nonsense. It is often claimed that small bass drivers are faster, being able to accelerate more quickly. Yes, a light cone is easier to accelerate, but this is only the same as saying that a light cone can reproduce higher frequencies. Given that a driver can generate a given frequency, its ability to generate higher frequencies; its bandwidth, has little to do with the rate at which a low frequency tone builds up or decays. Provided that the driver is operating at reasonably low Q factor (a feature of the driver plus its enclosure) then its contribution to the sluggishness of bass response is likely to be negligible. This is less true of reflex designs, though they remain popular simply because it is rare to find a room in which their defects are not swamped by resonant modes.
Distortion measurements on loudspeakers can only go as low as the distortion of the measuring microphone itself of course which ideally should be 120 to 140dB SPL. A typical top-end speaker, driven by a typical 100watt power amplifier, cannot produce peak levels much above 105dB SPL at 1m (which translates roughly to 105dB 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 130dB 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 100dB SPL, the peaks on percussion will far exceed this. Most speakers give around 3% distortion measured 468-weighted (distortion residue), reducing only slightly at low levels (the Quad electrostatic is exceptional in reducing to around 0.1%. 3% distortion residue corresponds to 1 or 2% Total harmonic distortion. Top professional monitors can maintain this level of distortion up to around 115dB SPL at 1m, but most domestic speakers distort severely above 100dB SPL.
Colouration and waterfall plots
Loudspeakers differ from most other items of audio equipment in suffering from 'colouration'. This refers to the tendency of various part 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 new measuring equipment was introduced to measure the delayed output from speakers and display it in the form of a 'waterfall plot'. Initially this was done using impulse response testing, but this was found to suffer the disadvantage of having very low energy content if testing was within the peak capability of the speaker. Later equipment uses correlation on other waveforms, for example using a 'Maximum length sequence system analyser' or MLSSA. The result of such testing is clearer mid-range and treble, though the effects of bass resonances are usually negligible compared to room modes for most listening rooms.