A microphone converts the incoming sound into an electrical voltage. Unlike e.g. vocal microphones, measurement microphones are optimized for audio measurement technology.
Measurement microphones are today available with different connectors. These connectors are completely different, mechanically and electrically. The circuit design concepts are very different. Therefore, adapters are only available in special cases e.g. ICP to P48,
XLR with P48V This method comes from studio technology and uses symmetrical signal transmission with an impressed supply voltage of 48V. The 48V was originally used as an external polarization voltage for condenser microphones and was therefore only designed for very low power consumption. The connection is 3-pin with an XLR connector. Typical XLR measurement microphones only deliver approx. 3V RMS and are therefore not very stable. The 48V were never designed for powerful preamps. This type of connection is more likely to be found in price-conscious measuring devices. However, with the NTI XL2 or the Bedrock SM50, this connection has found wide acceptance for acoustic measurement technology.
Advantages: sufficient signal dynamics in many cases, very cheap and relatively long cable (50m) Disadvantages: input voltage limited to 3V RMS
USB measurement microphones This relatively new connector is designed to connect measurement microphones directly to a computer. Microphone, preamplifier, power supply and direct digitization are integrated into one device. These individual components are optimally matched to each other. High-quality USB measurement microphones generate various auxiliary voltages themselves from the 5V USB supply through DC/DC converters. As a result, the signal dynamics - especially the level stability - is enormous and amounts to approx. 10V RMS. These are 28V peak/peak! High-end USB microphones also generate the external polarization voltage of 200V for non pre-polarized microphone capsules.
Advantages: Very easy connection to computer-aided PC measurement systems such as Akulap. Very high signal dynamics possible. Disadvantages: Cables are limited to around 20m. Due to direct digitization, USB measurement microphones cannot be connected to analog inputs.
How is the sound converted into a voltage?
There are essentially the following types of microphones with different converter principles
A condenser microphone consists of a back-electrode and a membrane that is stretched in front of the back-electrode at a very small distance. The membrane is moved by the sound and this changes the capacitance of the capacitor. This change in capacitance is converted into an electrical signal.
Dynamic microphones play no role in the field of measurement technology. They consist of a membrane that moves a coil in a magnetic field, thereby generating a voltage. Such microphones are mainly found in recording and studio microphones. Dynamic microphones often show a pronounced "crooked" frequency response with pronounced resonances. This type also has low sensitivity and is therefore much more noisy.
A relatively new category are the MEMS microphones. These are special micromechanical components that can be manufactured inexpensively using chip manufacturing technologies. You can find MEMS microphones in almost all smartphones.
In recent years, these microphones have also proven themselves for measurement purposes. A particular advantage of these microphones is their low price and high stability. This makes these microphones particularly suitable for array microphones, where a large number of microphones are used. A current disadvantage of MEMS microphones is their high inherent noise, which currently does not come close to classic 1/2-inch microphone capsules. In addtion, high SPL is still a challange
Nowadays, condenser microphones are mainly used for measurement purposes. These microphones have standardized designs. The size is given in inches (“).
The ½" condenser microphones are the workhorses of acoustic measurement technology.
The most left capsule is a normal working capsule without protective grid. The middle shows the back-elektrode, while the diaphragma has been removed. The most right shows a defective capsule (mechanical shock from dropping to the floor)
There are two basic types of condenser microphones:
The DIN/IEC61672-1 standard defines various requirements for a sound level meter. Only certain deviations are permitted for the frequencies in the range between 10Hz and 20kHz. A distinction is made between class 1 and class 2 microphones. Class 1 devices are often required throughout for professional measurement technology. In practice, the deviation in condenser measurement microphones typically only shows up above 5kHz. The frequency response of the microphone is irrelevant for room acoustic measurements (RT60).
Condenser microphones are divided into categories based on the diaphragm diameter. The most common size is 1/2".
There are also 1-inch microphones with a very large membrane area. As a result, these microphones have less intrinsic noise and higher sensitivity. However, these advantages are bought with a very strong acoustic effect at higher frequencies, even starting from 3kHz.
Therefore, these measurement microphones are rarely used today. However, they are excellently suited for low frequencies down to the infrasound range.
However, a ¼” microphone capsule is much more common. These capsules are suitable for higher frequencies up to 70 to 100 kHz, but also for high sound levels. On the other hand, these microphones have a lower sensitivity and can therefore record much higher sound levels, up to around 170 decibels. On the other hand, these microphones have a higher inherent noise. Typically this is around 35dB(A). Therefore, these microphones are not used for low-noise measurements.
Large diaphragma area: low noise but low maximum frequency and lower maximum sound level
The sensitivity of a measurement microphone is the transmission factor between the incident sound and the electrical voltage generated. This factor is given in mV per pascal.
A common ½" condenser microphone has a sensitivity of 50mV/Pa. At a sound level of 1Pa, a voltage of 50mV is present at the microphone. The sound pressure of 1Pa corresponds exactly to 94dB. This matches the sound pressure level of a typical sound level calibrator.
The significantly smaller 1/4" capsules have a sensitivity of 5mV/Pa.
In acoustic measurement technology, non-directional microphones are usually used (omnidirectional characteristics).
Studio microphones usually have a different directivity, e.g. to attenuate signals behind the speaker.
Due to their design, condenser microphones are initially pure sound pressure receivers. Depending on the signal frequency and the size of the microphone, diffraction effects become noticeable due to the geometry. In a free field, deviations occur at higher frequencies. These deviations are compensated by a built-in correction. This is the free field correction. However, some microphone types are specially designed for diffuse sound incidence and are compensated accordingly.
Free-field equalized measurement microphones are used most frequently. Diffuse field equalized microphones are used in a reverberation room.
Uncompensated pressure field microphones are often used in couplers (e.g. ear simulators).
At lower frequencies below 5 kHz, the behavior of these microphones is almost identical. The compensation only becomes noticeable at higher frequencies. Externally, these microphone types are indistinguishable,
The following picture shows a typical 1/2" capsule (MTG MK222), which is equalized for free field conditions. Here you can clearly see the differences to the pressure field and a diffuse sound incidence.
No, the 200V voltage is only used to polarize the membrane. Practically no current flows. Therefore, the supply for the 200V is extremely high-impedance (Giga-Ohm). When touched, the voltage collapses immediately. The touch is noticeable but harmless. It corresponds to an electrostatic discharge.
Measurement microphones achieve their high level of accuracy under precisely defined sound field conditions. This is usually an open field. In practice, however, measurements are taken in normal rooms. Reflections on walls and other objects create significant resonances that change the sound field considerably. The spatial effect is in the range of up to 20dB and is therefore considerably larger than the differences between class 1 and class 2 microphones. It therefore makes little sense to measure loudspeakers with an ultra-precise microphone with deviations of less than 0.1dB in an acoustically untreated room. In the end, you really only miss the room and neither the microphone nor the loudspeaker have a great influence.
In principle, condenser microphones are able to measure frequencies down to 0Hz. That would then be a barometer. In practice, however, this effect is usually undesirable, since changes in air pressure due to the weather or different altitudes have a strong influence. The membrane would simply swell or dent. Therefore, equalization openings - capillaries - are installed behind the membrane, which ensure pressure equalization. The design defines the lower limit frequency of the microphone. Typically this is below 10Hz.
Caution: There are microphone capsules that vent directly from the capsule. Others vent through the preamp, which then needs an equalization port. Here you have to be careful that you match the capsule and preamplifier to each other. Otherwise there can be significant disruptive effects. We already have deviations at 100Hz! observed from 1-2dB.
For several years, the requirements in the field of acoustic measurement technology have increased significantly. A refrigerator or an air conditioner in the living area should be as quiet as possible. Many of these devices cannot be measured with sufficient accuracy, even with high-quality sound level meters. We have written a dedicated article for low noise measurements.
Typical measurement microphones can easily record levels below 120dB. However, if higher levels need to be measured, many factors must be taken into account. We have summarized this information in a separate article.
Sound can be described using various physical parameters such as sound pressure, sound intensity or sound energy.
However, these values can not sufficiently reflect the subjectively perceived volume. The different weighting curves (A or C) represent a first approximation. They take into account the different sensitivity of the human ear as a function of frequency. However, these sensitivities depend on the level. This results in a variety of different weighting curves. The most common are the A and C curves. A low volume pure sine tone of e.g. 50dB (A) is perceived as the same at different frequencies. These results go back to Barkhausen's research, which he published around 1920. However, pure sine tones are rather rare in nature, so that the sense of hearing is only reproduced inaccurately. The widespread use of these measured values lies solely in the fact that they can be determined comparatively easily from the physical quantities. Therefore, these ratings are now included in almost all handheld sound level meters.
Sound is usually first converted into an electrical signal using a microphone. The RMS value of this signal is a measure of the sound level. This quantity correctly describes the physical level. However, it is not suitable for meaningfully describing the perception of the level. For this purpose, the signal is filtered in time and in the frequency domain in order to achieve a rough approximation of the perception. The most important evaluation curve is the A curve. High and low frequencies are significantly reduced in the calculation. The ear, however, has different evaluation curves depending on the level. The C curve is still common.
Since a display in a sound level meter cannot follow a level that fluctuates over time, the signal is weighted (filtered) in the time domain. This corresponds to a damping of a pointer instrument. The usual time constants are impulse (I 35ms), fast (F 125ms) and slow (S 1s). In addition, the sound level is logarithmized according to the ears' perception (display in dB). You will find these functions in almost every sound level meter, as they can be easily implemented technically. This ensures high comparability of the measured values. However, a sound level in dB (A) describes the sensory perception only inadequately. An identical sound level can be perceived significantly differently. Therefore, psychoacoustic measures such as loudness in sone are increasingly used. There is a much better comparability here.
A broadband noise produces a subjectively perceived volume than a single tone of the same level measured in dB (A). This simple measure is therefore only of limited significance. Zwicker has intensively examined such psycho-acoustic effects and created models for hearing perception. A simple effect is e.g. the masking effect. If a signal consists of a loud tone, quieter tones are not heard. At a symphonic concert, no one would recognize the soft tripping of a mouse, although it would be audible in soft musical phases. These and other effects also serve as the basis for lossy audio compression such as the well-known MP3 method for compressing music. Signal components that cannot be heard according to the models are not stored in the data stream.
Based on its extensive hearing tests, Zwicker has developed a loudness measure, which is a much better measure than dB (A) for stationary signals. The unit of loudness is sone. In contrast to dB (A), this is a linear quantity. This means that e.g. 2 Sone is twice as loud as 1 Sone. The reference point is 1 sone, this corresponds to a pure sine tone with 1000Hz at a level of 40dB.
The applications are e.g. in the assessment of air conditioning systems, ventilation systems, but also typical PC components such as hard drives or CPU coolers etc. The loudness takes into account the subjectively perceived volume. However, it does not describe how pleasant or disturbing a sound is. Many people perceive a sinus tone as more pleasant than the sound of a dentist's drill with the same loudness.
The loudness calculation is based on the results of a third octave analysis. Loudness measurement also requires calibration to absolute sound levels.
Essentially two procedures have been established:
For stationary signals: DIN 45631 or ISO 532B. This method has been available for many decades and is particularly suitable for ventilation systems where the signal is constant, i.e. time-invariant. This simple model does not take into account the temporal masking effects of the human ear.
For time-variant signals: DIN 45631 with Appendix A1 or ISO 532-1. This model is significantly more complex and describes hearing even with signals that change over time and is therefore much more general. This method is often used in NVH (Noise Vibration Harshness).
The specific loudness is the loudness per frequency band. If we show the specific loudness against the frequency, this corresponds to a spectrum. However, if we do not use a linear frequency axis in Hz, but use a Bark scale that is better adapted to hearing.
If we integrate the specific loudness over the Bark scale, we get the loudness as a single value.
Acoustic parameters are subject to strong fluctuations over time. The classic parameter sound level is often averaged in an energy-equivalent manner (LEQ). However, percentiles are predominantly used for psycho-acoustic parameters. The 5, 50 and 95% percentiles are common. A 95% percentile value of e.g. 73 Sone indicates that 95% of all measured values in the time interval are above 73 Sone.
The sharpness is one of the essential hearing sensations according to Zwicker. A tone is perceived as “sharper”, frequency components are higher in the upper frequency range. The sharpness is measured in the unit "Acum". The reference signal with 1 acum consists of a narrow-band noise (920 Hz to 1 080 Hz) at a level of 60dB. The sharpness is calculated from the specific loudness using a standardized weighting method.
If an acoustic event contains perceptible tonal components, this is particularly taken into account by human perception. In general, the "annoyance" increases. A "tonal penalty" in the range of 0 to 6 dB is therefore calculated according to DIN 45681. The analysis is based on a narrowband analysis and takes into account masking effects in the frequency domain.
If an acoustic event contains perceptible impulse-like components, this is particularly taken into account by human perception. In general, the "annoyance" increases. The calculation is based on an evaluation of the loudness curve over time.
Roughness is also one of the core perceptions according to Zwicker. This impression can be clearly explained with two tones. If both tones have the same frequency, this pure single tone has no roughness. If you increase the frequency of one tone, a beat effect arises, which is perceived similarly to an amplitude modulation. The volume fluctuates with the difference frequency. The roughness effect occurs from a modulation frequency of 20 Hz. The roughness is measured in Asper. A signal with a frequency of 1 kHz and an amplitude modulation of 70 Hz (degree of modulation 100%) has a roughness of 1 Asper at a level of 60 dB. The roughness is not defined by international standards. However, the Daniel & Weber algorithm has been established for several years.
In this section we cover important parameters for psycho-acoustics
Sharpness, roughness and other parameters
In this article we cover the measurement of loudness for stationary (not fluctuating noise sources). We offer a turn-key solution for loudness measurement.
In our Webshop you will find many products with focus to audio measurements.
In addition to the simple ISO532B Zwicker method, we offer more complex time-variant loudness measurements and other psycho-acoustic parameters (Sharpness, roughness, tonality etc.).
Sound can be described with various physical parameters e.g. intensity, pressure or energy. These parameters are very limited to describe the perception of the human ear. A first approach is to take into account, that the ear is less sensitive at lower and higher frequencies. The research work around 1920 from Barkhausen result into the well known A,C and other weighting curves. These curves are most widely used and are included in almost any sound level meter.
Today almost any sound level measurement uses the A-curve. The A curve has the strongest attanuation for lower and higher frequencies. If you measure with A instead of C you get lower SPL reading. If you need low results for e.g. product advertisement, choose A. This approach is valid, as long as you mention the weighting curve.
A pure sine at lower levels e.g. 40dB(A) has the same subjective loudness over the complete frequency range.
**But, does the A curve really match with our perception? **
No, it does not. The A curve is defined as an average curve at a sound level of around 40dB. In addition, the curve is simplified, allowing an analog filter with very few components. You can easily implement the A curve with a well defined RC-network. This is one of the reason, why it is widely spread and even included in an 20$ sound level meter.
Unfortunatly, the frequency response of our ear depends on the level. There is not only one curve, there are many
The A curve and others are valid for pure tones, only. However, in practice sine tones are not very common. For complex sounds and noise, measuring with A-weighting does not give reasonable results.
A wideband noise has a different subjective loudness than a pure tone at the same level measured in dB(A). Therefore, dB(A) is very limited in its usage although widely used. Zwicker analyzed various psychoacoustics effects thoroughly. A result was a model for the human hearing.
A famous psychoacoustics effect is acoustical masking. A loud tone hides quiet tones. Nobody would expect to hear the sound of a mouse during a symphonic orchestra, although the mouse would audible in quiet phases. A very famous application of this and other effects are audio compression techniques like MP3. Only parts of the signal, which are audible according to a model, are stored.
Based on its countless hearing tests, Zwicker developed a model for loudness measured in sone. For stationary signals you will achieve much better results than the traditional dB(A).
In contrast to dB(A) sone is a linear parameter. Two sone has double loudness compared to one sone. The reference level is one sone, which is equivalent to a tone with 1kHz at a level of 40dB.
This measurement describes the subjective loudness. However, it does not allow distinguishing between nice or annoying tones. Most people will agree that the sound of a flute is much nicer than a drill used by dentist although they both might have the same loudness.
As mentioned previously, sone according to ISO532B is valid for stationary signals, only. It is ideally in the judgment of air conditioners or typical PC noise like hard disks of fans.
Real world signals are mostly not constant. They change over time. A sound level meter would show strong fluctuations. Therefore we use time-weighting or averaging .e.g. the famous F (fast) and S (slow) setting on a sound level meter. However, F and S are historic. The are leftovers from analog measurement intruments and can be very simply integrated to a sound level meter.
They are NOT related to our perceptions. Our ear is time-variant and handles signals in a very complex way.
Today, we have proven and well standardizid algorithms (the loudness model) to measure noise in a more generic way.
Loudness measurement is far beyound a simple sound level meter. The calculation is very complex. However, modern PC can handle this easily.
By using small notebooks , you can easily build a mobile measurement system. The large color display gives you a quick overview of all relevant parameters.
Akulap is a professional Windows application for powerful real-time signal and system analysis. By using the PC environment, it is not only a cost efficient replacement for classical laboratory equipment. AkuLap offers more powerful features combined with a comfortable user interface. Typical applications are acoustic measurements, room and building acoustics and noise monitoring. If you run Akulap on a notebook or even a tablet PC, you can easily build a mobile measurement system.
The measurement of the loudness is based on the 1/3 octave analyzer. In addition, you have to calibrate the system. All psycho acoustic analysis requires absolute sound levels. This can be achieved with a sound level calibrator.
In this section, we cover important aspects of audio measurements.
Our focus covers:
We will contantly expand this section. Currently, the german section is larger, but this will change during time.
In additon, if you have suggestion for any topics related to audio measurements, let us know.