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Loudspeaker measurement is the practice of determining the behaviour of loudspeakers by measuring various aspects of performance. This measurement is especially important because loudspeakers, being transducers, have a higher level of distortion than other audio system components used in playback or sound reinforcement.
One 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 the axis with the high-frequency driver. While this can 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, a 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 and most are not capable of measuring below 80 Hz.
A tetrahedral chamber is capable of measuring the low frequency limit of the driver without the large footprint required by an anechoic chamber. This compact measurement system for loudspeaker drivers is defined in IEC 60268-21:2018, [1] IEC 60268-22:2020 [2] and AES73id-2019. [3]
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 backward. 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 is reduced, creating a repeatable and accurate, but not very representative, response curve.
At low frequencies, most rooms have resonances at a series of frequencies where a room dimension corresponds to a multiple of half wavelengths. Sound travels at about 1,100 feet per second (340 m/s), so a room 20 feet (6.1 m) long will have resonances from 27.5 Hz upwards. These resonant modes cause large peaks and dips in the sound level of a constant signal as the frequency of that signal varies from low to high.
Additionally, reflections, dispersion, absorption, etc. all strongly alter the perceived sound, though this is not necessarily consciously noticeable for either music or speech, at frequencies above those dominated by room modes. These alterations depend on speaker locations 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 that will maximize speaker performance in any listening space (with the exception of the sonically unpleasant anechoic chamber). Some parameters, such as reverberation time (in any case, really applicable only to larger volumes), 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.
All multi-driver speakers (unless they are coaxial) are difficult to measure correctly if the measuring 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 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.
Room acoustics have a much smaller effect on nearfield measurements, so these can be appropriate when anechoic chamber analysis cannot be done. Measurements should be done at much shorter 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 efficiency, or the average sensitivity, 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. [4]
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 a failure to state the maximum SPL available, especially at low frequencies. 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.
Distortion measurements on loudspeakers can only go as low as the distortion of the measurement 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 the 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.
Loudspeakers differ from most other items of audio equipment in suffering from colouration, 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, an 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 system analyser (MLSSA). [5] 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.
A weighting filter is used to emphasize or suppress some aspects of a phenomenon compared to others, for measurement or other purposes.
A subwoofer is a loudspeaker designed to reproduce low-pitched audio frequencies, known as bass and sub-bass, that are lower in frequency than those which can be (optimally) generated by a woofer. The typical frequency range that is covered by a subwoofer is about 20–200 Hz for consumer products, below 100 Hz for professional live sound, and below 80 Hz in THX-certified systems. Thus, one or more subwoofers are important for high-quality sound reproduction as they are responsible for the lowest two to three octaves of the ten octaves that are audible. This very low-frequency (VLF) range reproduces the natural fundamental tones of the bass drum, electric bass, double bass, grand piano, contrabassoon, tuba, in addition to thunder, gunshots, explosions, etc.
A loudspeaker is an electroacoustic transducer that converts an electrical audio signal into a corresponding sound. A speaker system, also often simply referred to as a speaker or loudspeaker, comprises one or more such speaker drivers, an enclosure, and electrical connections possibly including a crossover network. The speaker driver can be viewed as a linear motor attached to a diaphragm which couples that motor's movement to motion of air, that is, sound. An audio signal, typically from a microphone, recording, or radio broadcast, is amplified electronically to a power level capable of driving that motor in order to reproduce the sound corresponding to the original unamplified electronic signal. This is thus the opposite function to the microphone; indeed the dynamic speaker driver, by far the most common type, is a linear motor in the same basic configuration as the dynamic microphone which uses such a motor in reverse, as a generator.
A microphone, colloquially called a mic, or mike, is a transducer that converts sound into an electrical signal. Microphones are used in many applications such as telephones, hearing aids, public address systems for concert halls and public events, motion picture production, live and recorded audio engineering, sound recording, two-way radios, megaphones, and radio and television broadcasting. They are also used in computers and other electronic devices, such as mobile phones, for recording sounds, speech recognition, VoIP, and other purposes, such as ultrasonic sensors or knock sensors.
A woofer or bass speaker is a technical term for a loudspeaker driver designed to produce low frequency sounds, typically from 20 Hz up to a few hundred Hz. The name is from the onomatopoeic English word for a dog's deep bark, "woof". The most common design for a woofer is the electrodynamic driver, which typically uses a stiff paper cone, driven by a voice coil surrounded by a magnetic field.
Audio power is the electrical power transferred from an audio amplifier to a loudspeaker, measured in watts. The electrical power delivered to the loudspeaker, together with its efficiency, determines the sound power generated.
Audio system measurements are a means of quantifying system performance. These measurements are made for several purposes. Designers take measurements so that they can specify the performance of a piece of equipment. Maintenance engineers make them to ensure equipment is still working to specification, or to ensure that the cumulative defects of an audio path are within limits considered acceptable. Audio system measurements often accommodate psychoacoustic principles to measure the system in a way that relates to human hearing.
A horn loudspeaker is a loudspeaker or loudspeaker element which uses an acoustic horn to increase the overall efficiency of the driving element(s). A common form (right) consists of a compression driver which produces sound waves with a small metal diaphragm vibrated by an electromagnet, attached to a horn, a flaring duct to conduct the sound waves to the open air. Another type is a woofer driver mounted in a loudspeaker enclosure which is divided by internal partitions to form a zigzag flaring duct which functions as a horn; this type is called a folded horn speaker. The horn serves to improve the coupling efficiency between the speaker driver and the air. The horn can be thought of as an "acoustic transformer" that provides impedance matching between the relatively dense diaphragm material and the less-dense air. The result is greater acoustic output power from a given driver.
An equal-loudness contour is a measure of sound pressure level, over the frequency spectrum, for which a listener perceives a constant loudness when presented with pure steady tones. The unit of measurement for loudness levels is the phon and is arrived at by reference to equal-loudness contours. By definition, two sine waves of differing frequencies are said to have equal-loudness level measured in phons if they are perceived as equally loud by the average young person without significant hearing impairment.
Loudspeaker acoustics is a subfield of acoustical engineering concerned with the reproduction of sound and the parameters involved in doing so in actual equipment.
Digital room correction is a process in the field of acoustics where digital filters designed to ameliorate unfavorable effects of a room's acoustics are applied to the input of a sound reproduction system. Modern room correction systems produce substantial improvements in the time domain and frequency domain response of the sound reproduction system.
Speech Transmission Index (STI) is a measure of speech transmission quality. The absolute measurement of speech intelligibility is a complex science. The STI measures some physical characteristics of a transmission channel (a room, electro-acoustic equipment, telephone line, etc.), and expresses the ability of the channel to carry across the characteristics of a speech signal. STI is a well-established objective measurement predictor of how the characteristics of the transmission channel affect speech intelligibility.
A loudspeaker enclosure or loudspeaker cabinet is an enclosure in which speaker drivers and associated electronic hardware, such as crossover circuits and, in some cases, power amplifiers, are mounted. Enclosures may range in design from simple, homemade DIY rectangular particleboard boxes to very complex, expensive computer-designed hi-fi cabinets that incorporate composite materials, internal baffles, horns, bass reflex ports and acoustic insulation. Loudspeaker enclosures range in size from small "bookshelf" speaker cabinets with 4-inch (10 cm) woofers and small tweeters designed for listening to music with a hi-fi system in a private home to huge, heavy subwoofer enclosures with multiple 18-inch (46 cm) or even 21-inch (53 cm) speakers in huge enclosures which are designed for use in stadium concert sound reinforcement systems for rock music concerts.
An acoustic transmission line is the use of a long duct, which acts as an acoustic waveguide and is used to produce or transmit sound in an undistorted manner. Technically it is the acoustic analog of the electrical transmission line, typically conceived as a rigid-walled duct or tube, that is long and thin relative to the wavelength of sound present in it.
Acoustic suspension is a loudspeaker cabinet design that uses one or more loudspeaker drivers mounted in a sealed box. Acoustic suspension systems reduce bass distortion which can be caused by stiff suspensions required on drivers used for open cabinet designs.
A-weighting is the most commonly used of a family of curves defined in the International standard IEC 61672:2003 and various national standards relating to the measurement of sound pressure level. A-weighting is applied to instrument-measured sound levels in an effort to account for the relative loudness perceived by the human ear, as the ear is less sensitive to low audio frequencies. It is employed by arithmetically adding a table of values, listed by octave or third-octave bands, to the measured sound pressure levels in dB. The resulting octave band measurements are usually added to provide a single A-weighted value describing the sound; the units are written as dB(A). Other weighting sets of values – B, C, D and now Z – are discussed below.
A parabolic loudspeaker is a loudspeaker which seeks to focus its sound in coherent plane waves either by reflecting sound output from a speaker driver to a parabolic reflector aimed at the target audience, or by arraying drivers on a parabolic surface. The resulting beam of sound travels farther, with less dissipation in air, than horn loudspeakers, and can be more focused than line array loudspeakers allowing sound to be sent to isolated audience targets. The parabolic loudspeaker has been used for such diverse purposes as directing sound at faraway targets in performing arts centers and stadia, for industrial testing, for intimate listening at museum exhibits, and as a sonic weapon.
An audio analyzer is a test and measurement instrument used to objectively quantify the audio performance of electronic and electro-acoustical devices. Audio quality metrics cover a wide variety of parameters, including level, gain, noise, harmonic and intermodulation distortion, frequency response, relative phase of signals, interchannel crosstalk, and more. In addition, many manufacturers have requirements for behavior and connectivity of audio devices that require specific tests and confirmations.
Veritone Minimum Phase Speakers, or VMPS, was a loudspeaker manufacturer founded in 1977 by speaker designer Brian Cheney. Many VMPS speakers received favorable reviews from audio critics, such as the RM40, which was awarded Best of CES in the High-End Audio category in 2002. VMPS was in operation for over 35 years, from January 1977 to December 2012, when it closed soon after the death of company owner Brian Cheney on December 7, 2012.
Diffuse field acoustic testing is the testing of the mechanical resistance of a spacecraft to the acoustic pressures during launch.