First let us quickly understand what a how a speaker functions.
Speakers consist of one or more driver units in a box. The driver is constructed of a metal frame to which is attached a cone, made of paper or plastic and occasionally metal. At the rear end of the cone is attached a coil of wire (the "voice coil") wound around an extension of the cone, called a "former". The two ends of the voice coil are connected to the crossover network, and the crossover network is connected to the speaker binding posts on the rear of the speaker enclosure. The voice coil is suspended inside a permanent magnet so that it lies in a narrow gap between the magnet pole pieces and the front plate. The voice coil is kept centered by a "spider" that is attached to the frame and to the voice coil. A rear vent allows air to get into the back of the driver when the cone is moving, but a dust cap on the cone keeps air from getting in through the front. A rubber surround at the outer edge of the cone allows for flexible movement. In the case of a tweeter, the cone is very light, perhaps made of silk, and is glued directly to the voice coil. It isn't attached to a frame or rubber surround because it needs to be very low mass in order to respond quickly to high frequencies.
When the musical electrical signal from the amplifier passes through the voice coil, the voice coil turns into an electromagnet. Depending on which way the current is travelling in the voice coil, the north and south pole of the magnetic field will be at one end of the voice coil or the other. The permanent magnet has a north and south pole as well. Its magnetic field will push the coil (and the attached speaker cone) outward if the north and south poles of the two magnetic fields are lined up together (north to north, and south to south), or pull the voice coil inward if they are lined up oppositely (north to south, and south to north). So, as the electrical signal from the amplifier, which is a representation of the original musical waveform, passes through the voice coil, and changes direction with the musical waveform, the voice coil and attached speaker cone are driven outward and pulled inward in time with the music. The speaker cone pushes or pulls air in the room, which translates to increases or decreases in air pressure at your eardrums, and there you have it: music.
Speakers have created air pressure. Human ears do just the reverse. Human ears have diaphragm that are like microphone. When you put pressure on the diaphragm, it vibrates and creates electrical signals that are sent to neurons in our brains. These neurons convert the signals to what we perceive as sound.
The human ear hears sound as it is sensitive to pressure in the air. It does not have a flat spectral response. In other words, it's efficiency of detection or reaction to sound pressure), is a function of the frequency or wavelength of the sound signal. Sound pressure is often frequency weighed such that the measured level will match the perceived level. When weighed this way, the measurement is referred to as a sound level.
When you say perceive, you are referring to a comparison or a ratio. To do this you must have a reference level. The commonly used reference sound pressure in air is 20 micropascals (20 µPa) (rms), which is usually considered the threshold of human hearing (roughly the sound of a mosquito flying 3 m away). The issue is this is a very small unit (as it is 2 ten billionths of an atmosphere) and difficult to represent. Thus to measure sound pressure levels, a logarithmic ratio is used.
Thus sound pressure level (SPL) or sound level is a logarithmic measure of the rms sound pressure of a sound relative to a reference value. It is measured in decibel (dB).
The formula for calculating differences in sound pressure is
Lp=10Log10 (P~2 rms / P~2ref) = 20log10 (Prms/Pref) = dB
Where where p(ref) is the reference sound pressure and p(rms) is the rms sound pressure being measured. [Please see below for a representation of the formula is correct mathematical form.]
Most measurements of audio equipment will be made relative to this level, meaning 1 pascal will equal 94 dB of sound pressure.
For instance, suppose we have two loudspeakers, the first playing a sound with power P1, and another playing a louder version of the same sound with power P2, but everything else (how far away, frequency, etc) kept the same.
The difference in power between the two is defined to be
10 log (P2/P1) dB where the log is to base 10.
If the second produces twice as much power than the first, the difference in dB is 10 log (2/1) = 10 log 2 = 3 dB.
If the second had 10 times the power of the first, the difference in dB would be 10 log (10/1)= 10 log 10 = 10 dB.
Remember that decibels measure a ratio. 0 dB occurs when you take the log of a ratio of 2. So 0 dB does not mean no sound, it means a sound level where the sound pressure is equal to that of the reference level. This is a small pressure, but not zero. It is also possible to have negative sound levels: - 20 dB would mean a sound with pressure 10 times smaller than the reference pressure, ie 2 µPa.
Let us see how a speaker specification is shown. If you look at the B&W 683, there are two rows of information.
Frequency response 38Hz - 22kHz ±3dB on reference axis (EFFICIENCY)
Sensitivity 90dB - (2.83V, 1m) (SENSITIVITY)
Both use the same unit of measurement - dB. Well let us not get confused. Decibel or dB is a unit of measurement that is used in both sound signal and electrical signals. Unfortunately since the unit of measurement is the same, a loudspeaker's sensitivity appears to be universally confused with its efficiency.
Sensitivity is strictly defined as how much acoustic power the loudspeaker puts out for how much electrical power it is being driven with. For example, if you feed a loudspeaker with 10 electrical watts, how many acoustic watts of sound does it produce? The answer is "not many," a typical moving-coil loudspeaker being about 1% 'efficient'.
Efficiency, on the other hand, is expressed as sound-pressure level produced by a speaker at a specific distance, 1m, for 1W input; ie, in decibels watt meter (dB/W/m) or simply dB. This is not that simple as it sounds as a loudspeaker's efficiency is dependent upon both impedance and frequency.
To simplify matters, the impedance of a loudspeaker is always assumed to be 8 ohms at 1000 Hz, at 2.83V, As per ohms law, (Power = V x V/R = (2.83 x 2.83)/8 = 1W.), you are now feeding 1 watt to the loudspeaker.
So there we have it. The base frequency for measuring the efficiency of a loudspeaker is 1000Hz or 1KHz.
The efficiency of loudspeakers are measured in what is called a anechoic chamber. This is a room where sound-absorbing materials on every surface soak up every sound emitted by the speaker. The room is therefore removed from the picture and the only sound that reaches the measuring microphone is therefore that from the speaker. A SPL meter is kept 1 meter from the speaker. The loudspeaker is first fed with a continuous sound signal at a frequency of 1000Hz. The sound pressure created by the loudspeaker is measured and this becomes the reference level sound pressure. Then various sound frequencies are fed to the speaker and their sound levels are measured. Sound pressure will increase or decrease at various frequencies as compared to 1000Hz. The differences are specified as ± dB.
A loudspeaker is accepted as efficient if the sound pressure level variation is equal to or below 4 for the stated frequency range. Thus a speaker specified as 20-20,000Hz ±2.5dB is more efficient than a speaker specified as 20-20,000Hz ± 3.5dB.