A third alternative becomes possible if we can capture the output from the loudspeaker as a digital signal. Once the acoustic signal from the 'speaker has been digitised it can be analysed mathematically. Suppose that we feed an impulse into the loudspeaker. The Fourier transform of the impulse is the frequency response, a sum that can be done easily in a computer. So, if we capture the impulse data we can find the frequency response of the loudspeaker.
The trick is to choose which part of the impulse to transform. By deliberately truncating the 'tail' of the impulse we can effectively cut off the reflections since these arrive at the measurement microphone later than the direct sound.
The reflections are removed by a window in time. This windowing technique is very powerful and is used in many commercial loudspeaker measurement packages. Its chief disadvantage is that it is difficult to calibrate the microphone to show the absolute sound pressure level measured. This is because of the mathematical technique used FFT combined with the normal methods for mic' calibration use of piston-phones and so on.
Even so the method produces good relative measurements. If you need to make a set of measurements for comparison you must use identical settings for each.
Otherwise the relative levels that you record will not be comparable. Measurement Considerations There are a few fundamental rules that apply to this form of measurement. The time to the first reflection determines the window length. This is fixed by the size of the measuring room. The window length determines the low frequency cutoff. The Audio Voice Newsletter. Floyd Toole Cumulative Spectral Decay. Show more Show less. The controversy over subjective versus objective loudspeaker evaluation has raged on for decades.
However, to my mind, there is no controversy. These criteria are simply two faces of the same coin. 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. 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.
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 to dB SPL if high-level distortion is to be measured. A typical top-end speaker, driven by a typical watt power amplifier , cannot produce peak levels much above dB SPL at 1 m which translates roughly to 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 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 dB SPL, programme level peaks on percussion will far exceed this.
Electrostatic speakers can have lower harmonic distortion but suffer higher intermodulation distortion. Professional monitors may maintain modest distortion up to around dB SPL at 1 m, but almost all domestic speaker systems distort badly above 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 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.
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.
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