- Loudspeaker measurement
Loudspeaker measurement is one of the most difficult aspects of
audio quality measurementFact|date=March 2008, and also probably the most relevant, since loudspeakers have long been generally acknowledged to be the 'weak link' in the audio chainFact|date=March 2008.
The standard way to test a loudspeaker requires an
anechoic chamber, with an acoustically transparent floor-grid. The measuring microphoneis 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 chamberis needed, with large absorbent wedges on all sides. Most anechoic chambers are not designed for accurate measurement down to 20 Hz.
Measurements 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
tweeterand mid-range unit are equalFact|date=March 2008. 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 ' 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 convert|50|ft|m 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 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 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! Diffraction effects off the edges of the cabinet are also eliminated, which is not desirable as it will affect actual performance in use.
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 convert|20|ft|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, but couples into the
resonant room modes, which are resonant standing wave patterns. Because this coupling is impedance dependent, 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, 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 3–5 kHz) 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, 1 kHz has a wavelength of convert|1|ft|m|abbr=on in air, and 10 kHz a wavelength of only convert|1|in|mm|sing=on. 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 1 m will not be quite so flat at 2 m, a better distance for measurement, but seldom used in increases errors caused by residual room reflections.
Measurements made at 2 or 3 m, 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 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 can be 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! This is where the real challenge of audio quality lies
Minimising room modes and equalisation
equalizerto correct for room response is a poor solutionFact|date=March 2008 (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 equalizerFact|date=March 2008. 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 modesis 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 all this that marketing claims that a bass driver is ‘fast’ or 'quick-responding' are unfounded. Speaker manufacturers often claim that small bass drivers are ‘faster’, or that they have a quicker 'transient response. While a light cone is easier to accelerate, the only result is that 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 bass reflexdesigns, though they remain popular simply because it is rare to find a room in which their defects are not swamped by resonant 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
SPLavailable, especially at low frequencies. Because of the way in which the sensitivity of our ears falls off as shown in Equal-loudness contoursit is desirable 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 bandwidthmeasurement 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 all too common, are in fact meaningless. The situation is even 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 measurements on loudspeakers can only go as low as the distortion of the measuring microphoneitself 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 metermight 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 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 115 dB SPL at 1 m, but most domestic speakers distort severely above 100 dB SPL.
Loudspeakers 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 resonancecause this, by storing energy, and resonances with high Q factorare 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 time vs. frequency spectrogramplot. Initially analysis was done using impulse responsetesting, but that was found to suffer the disadvantage of having very low energy content when testing within the peak capability of the speaker. Later equipment uses correlationon other waveforms such as a ' Maximum length sequencesystem 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 modesfor most listening rooms.
Audio quality measurement
* [http://www.mlssa.com/pdf/MLSSA-Brochure.pdf MLSSA site]
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