Sound Pressure Measurement

Course overview

The course introduces the fundamentals of acoustic pressure detection using DewesoftX and standard microphones. You’ll begin with theory: understanding how sound waves cause local pressure fluctuations and how to convert these to SPL in dB using the reference level of 20 µPa.

Next, you’ll examine the frequency response of human hearing and how A, B, C, and Z-weighting filters help match the analysis to perceived loudness. The course also covers CPB (octave) analysis, allowing you to inspect sound pressure across defined log-frequency bands.

Hands-on segments walk through selecting the appropriate microphone type—pre-polarized condenser or hydrophone—understanding directionality patterns, and setting up IEPE excitation and sensitivity scaling in DewesoftX. You’ll perform calibration using reference tone generators, ensuring accurate decibel levels during measurement.

In Measure mode, you’ll configure real-time tracking of SPL metrics like Leq, peak, and true‑peak. The course also demonstrates handling different environmental conditions—whether you’re measuring in a free field outdoors or in enclosed spaces with reflections.

By the end, you’ll be capable of delivering reliable sound pressure measurements, fully calibrated and in compliance with acoustic standards—perfect for noise monitoring, acoustic testing, NVH, or environmental sound surveys.

Sound is a wave motion that travels through air or other elastic media. It is produced by objects vibrating at specific frequencies, such as loudspeakers, reeds, or machinery.

When an air particle is displaced from its original position, the elastic forces of the air act to restore it. Due to inertia, the particle overshoots the resting position, which in turn activates elastic forces in the opposite direction, and the process repeats. Sound is readily conducted in gases, liquids, and solids—such as air, water, steel, and concrete—all of which are elastic media.

Sound cannot propagate without a medium. It travels through compressible media such as air, water, and solids as longitudinal waves, and through solids also as transverse waves. Sound waves are generated by a source (for example, a vibrating diaphragm or stereo speaker), which creates vibrations in the surrounding medium. As the source continues to vibrate, these disturbances propagate away from it at the speed of sound, forming a sound wave. At a fixed distance from the source, the pressure, velocity, and displacement of the medium vary over time.

Wavelength and frequency

In the picture below, a sine wave is illustrated. The wavelength (λ) is the distance a wave travels during the time it takes to complete one cycle. A wavelength can be measured between successive peaks or between any two corresponding points in the cycle. This applies not only to sine waves but also to other periodic waveforms. The frequency (f) specifies the number of cycles per second and is measured in hertz (Hz).

Sound pressure

Sound pressure, or acoustic pressure, is the local deviation from the ambient (average or equilibrium) atmospheric pressure caused by a sound wave. In air, sound pressure can be measured with a microphone, while in water it is measured with a hydrophone. The SI unit for sound pressure (p) is the pascal (Pa).

Sound pressure level

Sound pressure level (SPL), or sound level, is a logarithmic measure of the effective sound pressure relative to a reference value. It is expressed in decibels (dB) above a standard reference level. The standard reference for sound pressure in air or other gases is 20 µPa, which is generally considered the threshold of human hearing (at 1 kHz). The following equation shows how to calculate the sound pressure level (Lp) in decibels [dB] from sound pressure (p) in pascals [Pa].

where p_ref is the reference sound pressure and p_RMS is the RMS sound pressure being measured.

Most sound level measurements are made relative to this reference level, meaning that 1 pascal corresponds to an SPL of 94 dB. In other media, such as underwater, a reference level (p_ref) of 1 µPa is used.

The minimum level detectable by a healthy human ear is an SPL of 0 dB, while the upper limit is not as clearly defined. In Earth’s atmosphere, 1 bar (equivalent to 194 dB Peak or 191 dB SPL) represents the maximum pressure variation an undistorted sound wave can have. However, larger sound waves may occur in other atmospheres or in different media, such as underwater or through the Earth.

The ears detect changes in sound pressure, but human hearing does not have a flat frequency response with respect to amplitude. Humans do not perceive low- and high-frequency sounds as effectively as they perceive sounds around 2000 Hz. Since the frequency response of human hearing also varies with amplitude, weighting curves have been established for measuring sound pressure.

In this course, we will learn how to perform such measurements and how to “distort” the results so they correspond more closely to what the human ear actually perceives.

The human ear does not have equal “gain” at different frequencies. For example, we perceive the same level of sound pressure at 1 kHz as louder than at 100 Hz. To compensate for this effect, frequency weighting curves are used, which approximate the response of the human ear.

The most commonly known application is frequency weighting in sound level measurement, where specific sets of weighting curves—A, B, C, and D weighting—are defined in IEC 61672. Unweighted measurements of sound pressure do not correspond to perceived loudness, because the human ear is less sensitive to very low and very high frequencies. The curves are applied to the measured sound level using a weighting filter in a sound level meter.

• A-weighting: A-weighting is applied to measured sound levels to account for the relative loudness perceived by the human ear, which is less sensitive to both low and high audio frequencies.

  B-weighting: B-weighting is similar to A-weighting, but the low-frequency attenuation is less pronounced (–10 dB at 60 Hz). It was historically considered the best weighting for musical listening purposes.

  C-weighting: C-weighting is similar to A and B at high frequencies but provides very little attenuation at low frequencies. It is typically used for measuring high-level noise.

  D-weighting: D-weighting was specifically designed for measuring high-level aircraft noise in accordance with the IEC 537 standard. The large peak in the D-weighting curve reflects the fact that humans perceive random noise differently from pure tones, particularly around 6 kHz.

  Z-weighting (linear): Z-weighting is linear across all frequencies, applying the same effect to all measured values.

Unlike FFT analysis, which uses a specific number of lines per linear frequency on the x-axis, CPB (constant percentage bandwidth, also called octave analysis) uses a specific number of lines when a logarithmic frequency axis is applied. As a result, lower frequencies contain more lines, while higher frequencies contain fewer. CPB analysis is traditionally used in the fields of sound and vibration.

A CPB filter is defined by having a bandwidth that is a fixed percentage of its center frequency. The width of each filter is relative to its position within the frequency range of interest. The higher the center frequency, the wider the bandwidth. Bandwidth is typically expressed in octaves or as a fixed percentage of the center frequency.

Filters with the same constant percentage bandwidth (CPB filters), such as 1/1 octave, are usually displayed on a logarithmic frequency scale. These are sometimes also referred to as relative bandwidth filters. CPB filter analysis, together with logarithmic scaling, is commonly used in acoustic measurements because it provides a close approximation to the way the human ear perceives sound.

The widest octave filter used has a bandwidth of 1 octave, but many subdivisions into smaller bandwidths are also common. CPB filters are often labeled according to their fractional-octave resolution. For example, a 1/1 octave filter has a bandwidth of about 70% of its center frequency, while 1/3 octave filters are among the most widely used. One advantage of 1/3 octave filters is that, at frequencies above 500 Hz, their bandwidth closely matches the frequency selectivity of the human auditory system. Filter bandwidths as fine as 1/96 octave have been realized.

When a signal with many frequency components is analyzed, an octave filter produces a response shown as the dotted curve. Using a 1/3 octave analysis provides higher resolution, represented by the solid curve, which reveals more detailed information.

Example of a 1/1 octave filter

Example of a 1/3 octave filter:

Example of a 1/12 octave filter:

A microphone is an acoustic-to-electric transducer, or sensor, that converts sound in the air into an electrical signal.

Microphones function essentially as pressure sensors, but they are designed to measure very small variations in pressure around the atmospheric baseline.

All microphones convert sound energy into electrical energy, but there are several different methods of achieving this—using electrostatics, electromagnetism, piezoelectric effects, or even changes in the resistance of carbon granules. The majority of microphones used today are either capacitor (electrostatic) or dynamic (electromagnetic) types. Both employ a moving diaphragm to capture sound, but they rely on different electrical principles to convert mechanical energy into an electrical signal.

The efficiency of this conversion is critical, as the amount of acoustic energy produced by voices and musical instruments is relatively small.

How does a microphone work?

Different types of microphones use different methods to convert sound energy, but they all share one essential component: the diaphragm. This is a thin piece of material (such as paper, plastic, or aluminum) that vibrates when struck by sound waves. In a typical handheld microphone, the diaphragm is located in the head of the device.

When the diaphragm vibrates, it sets other components of the microphone into motion. These vibrations are then converted into an electrical current, which becomes the audio signal.

The way a microphone converts an acoustic input signal into an electrical signal is known as its transducer principle.

The condenser microphone is the most widely used type. It operates on the principle that the capacitance between two electrically charged plates changes with their separation distance. In microphones, this capacitor is formed by a backplate and a lightweight diaphragm that moves in response to acoustic pressure variations.

The diaphragm serves as one element of the capacitor. When sound waves cause it to move, the distance between the diaphragm and the backplate changes, which alters the capacitance. These variations in capacitance are then converted into an electrical output signal. In essence, a condenser microphone functions as a capacitor with one movable plate that responds to sound waves. The resulting changes in capacitance are amplified to produce a measurable signal.

Because the capacitor requires a voltage across its plates, condenser microphones usually need an external power supply (such as phantom power) or a small battery.

A capacitor consists of two plates with a voltage applied between them. In a condenser microphone, the diaphragm vibrates when struck by sound waves, changing the distance between the plates and thus altering the capacitance. When the plates move closer together, the capacitance increases and a charging current flows. When they move farther apart, the capacitance decreases and a discharging current occurs. Because of this design, condenser microphones typically require a small battery or an external power source to maintain the voltage across the capacitor.

A dynamic microphone operates on electromagnetic principles. When a magnet moves past a wire (or a coil of wire), it induces an electrical current in the wire. In a dynamic microphone, sound waves cause the diaphragm to move either a coil or a magnet, and this motion generates a small current. Using this electromagnetic principle, the dynamic microphone converts mechanical vibrations into an audio signal via the interaction of a wire coil and a magnet.

Piezoelectric Microphone – Piezoelectric microphones use sensitive crystals that respond to the physical vibrations of incoming acoustic signals and generate an electrical output. While this technology was once widely used in tape recorders, it is most commonly applied today in pickup devices for acoustic instruments.

Loudspeakers perform the opposite function of microphones by converting electrical energy into sound waves. This principle is demonstrated in the dynamic microphone, which essentially operates as a loudspeaker in reverse.

A microphone’s directionality, or polar pattern, indicates how sensitive it is to sound arriving from different angles relative to its central axis. The polar pattern describes the microphone’s three-dimensional orientation in space in relation to sound sources in the surrounding environment. Microphones are generally classified as either omnidirectional or unidirectional.

An omnidirectional microphone (also called a non-directional microphone) captures acoustic signals equally well from all directions.

A unidirectional microphone, by contrast, uses specific polar patterns or shapes that determine the directions from which it most effectively receives sound.

Unidirectional patterns are ideal for focusing on a specific sound source while reducing ambient noise. These microphones are highly directional and must be pointed directly at the subject to achieve the best sound quality.

A common type of unidirectional microphone is the cardioid microphone, which captures sound in a heart-shaped pattern. This makes cardioid microphones well suited for vocals, as they can capture the dynamic range of a performance while minimizing unwanted sounds from other directions.

An omnidirectional microphone captures a wide range of sounds from all directions. This type of microphone provides the most accurate representation of the overall environment, as it collects sound equally well from every angle.

Bidirectional microphones pick up sound evenly from both the front and back of the microphone element. They are especially useful when capturing the interaction between two sound sources.

Selecting a microphone involves several key considerations:

• Externally polarized or pre-polarized

• Free-field, pressure, or random incidence

• Dynamic range

• Frequency range

Externally polarized or pre-polarized

Condenser microphones require a polarization voltage, which can either be supplied from an external power source or achieved by injecting a permanent electrical charge into a thin PTFE layer on the microphone backplate.

Externally polarized microphones

These microphones are used with standard preamplifiers. The preamplifier must be connected to a power module or an analyzer input capable of supplying both operating power for the preamplifier and 200 V for polarization. Externally polarized microphones are the most accurate and stable, making them the preferred choice for highly critical measurements.

Pre-polarized microphones

Pre-polarized microphones are typically used with constant current power (CCP) preamplifiers. They must be connected either to an input stage designed for CCP transducers or powered by a constant current source.

CCP preamplifiers use standard coaxial cables. However, the long-term and high-temperature stability of pre-polarized microphones is generally not as good as that of externally polarized microphones.

Free-field, pressure or random incidence

Measurement microphones can be divided into three main categories: free-field, pressure, and random incidence. The differences between these types become significant at higher frequencies, where the physical size of the microphone is comparable to the wavelength of the sound being measured.

Free-field microphones

A free-field microphone is designed to measure sound pressure as it would exist if the microphone were not present in the sound field. At higher frequencies, the presence of the microphone can locally disturb the sound pressure. To address this, the frequency response of a free-field microphone is carefully adjusted to compensate for these disturbances.

Free-field microphones are recommended for most sound pressure level measurements, such as those made with sound level meters or during sound power measurements.

Pressure microphones

A pressure microphone is designed to measure the actual sound pressure on the surface of its diaphragm. A typical application is measuring sound pressure in a closed coupler or, as shown below, measuring sound pressure at a boundary or wall. In this case, the microphone forms part of the wall and measures the pressure directly at its surface.

Pressure microphones are recommended for studies of sound pressure inside closed cavities.

Random incidence microphones

A random incidence microphone is designed for measurements in sound fields where sound arrives from many directions, for example in a reverberation chamber or other highly reflective environments. The combined influence of sound waves arriving from all directions depends on their distribution. For measurement microphones, a standardized distribution has been defined based on statistical considerations, resulting in the standardized random incidence microphone.

Random incidence microphones are typically used for sound pressure level measurements in accordance with ANSI standards.

The dynamic range of a microphone

The dynamic range of a microphone is defined as the range between the lowest and highest levels the microphone can accurately handle. This is not only determined by the microphone itself but also by the preamplifier used with it. To a large extent, the dynamic range of a microphone is directly linked to its sensitivity.

In general, a microphone with high sensitivity can measure very low levels but not very high ones, while a microphone with low sensitivity can measure very high levels but not very low ones. The sensitivity of a microphone is influenced by its size and the tension of its diaphragm: a large microphone with a loose diaphragm will have high sensitivity, whereas a small microphone with a stiff diaphragm will have low sensitivity.

The upper limit of the dynamic range

The highest measurable levels are limited by the amount of diaphragm movement before it comes into contact with the microphone’s backplate.

As sound pressure increases, the diaphragm deflects further until it eventually strikes the backplate inside the microphone body. This point represents the maximum level the microphone can measure.

The lower limit of the dynamic range

Even in perfectly quiet conditions, the thermal agitation of air molecules generates a small output signal. This thermal noise is typically around 5 µV and is superimposed on any acoustically excited signal detected by the microphone. Because of this, no signal below the level of the thermal noise can be measured.

The frequency range of a microphone

The frequency range of a microphone is defined as the interval between its upper and lower limiting frequencies. Modern microphones can cover a range from about 1 Hz up to 140 kHz.

• Low-frequency measurements require a microphone with well-controlled static pressure equalization and very slow venting. Special versions are available for infrasound measurements.

• High-frequency measurements are highly sensitive to diaphragm stiffness, damping, mass, and diffraction effects.

Upper limiting frequency

The upper limiting frequency depends on the size of the microphone, or more precisely, its diameter relative to the wavelength of the sound. Since wavelength decreases as frequency increases, smaller microphones can measure higher frequencies. However, microphone sensitivity is also linked to size, which in turn affects dynamic range.

Lower limiting frequency

The lower limiting frequency is determined by the microphone’s static pressure equalization system. A microphone essentially measures the difference between its internal pressure and the ambient pressure.

If the microphone were completely airtight, changes in barometric pressure or altitude would cause static deflection of the diaphragm, leading to changes in frequency response and sensitivity. To prevent this, microphones are built with a static pressure equalization channel that balances internal and ambient pressure. Equalization, however, must be slow enough to avoid interfering with the measurement of dynamic signals.

We can connect a condenser microphone directly to the ACC or ACC+ module. In this example, we used a pre-polarized free-field microphone with an integrated preamplifier.

A condenser microphone requires a power supply. When the IEPE mode is selected, the necessary excitation for the microphone is provided. The excitation current can be set to either 4 mA or 8 mA.

To perform accurate scientific measurements with a microphone, its precise sensitivity must be known (in volts per pascal, V/Pa). Since sensitivity may change over the lifetime of the device, it is necessary to calibrate measurement microphones regularly.

Microphones can be calibrated in two ways. First, it is important to understand that the direct output value from the microphone represents sound pressure in pascals (Pa). Therefore, scaling is required to match the physical quantity.

Scaling with a calibration certificate

If a calibrator is not used but the microphone’s sensitivity is available, the value can be entered directly in the Channel Setup.

First, Pa is defined as the physical unit of measurement. Next, in Scaling by Function, check Sensitivity and enter the value in mV/Pa, which can be found on the microphone’s calibration certificate.

Calibrating the microphone with calibrator

Another method of calibrating the microphone is by using a calibrator. In this case, the known parameter is the sound level emitted by the calibrator—in our example, 94 dB at 1000 Hz.

First, open the Channel Setup of the microphone. By default, the sensitivity is set to 1. On the right side of the microphone scaling section, information from the microphone connected to the calibrator is displayed. The calibration frequency is set to 1000 Hz, and the current value detected by the microphone is 127.4 dB. This reading is incorrect, since the calibrator output is 94 dB.

After pressing Calibrate, the sensitivity of the microphone is measured based on the highest peak in the frequency spectrum, usually at 1000 Hz (with amplitude correction applied to obtain the correct amplitude).

Microphone sensitivity can also be read directly from TEDS. In that case, manual calibration is not required because the sensitivity value is already stored in the TEDS chip.

After pressing the Calibrate button, the sensitivity value updates. The current value now shows 94 dB, which confirms that the microphone has been successfully calibrated.

The Sound Level Math section enables the calculation of typical parameters for sound level measurements from a single microphone. This allows Dewesoft to function as a standard sound level meter. With appropriate hardware (such as Sirius ACC), it can meet all the requirements for a Class I sound level meter. It also supports multiple standards, including IEC 60651, IEC 60804, and IEC 61672.

Sound level measurements can be enabled by selecting the Sound Level Meter checkbox under the Math section.

Once this option is selected, a new tab labeled Sound Levels appears in the Dewesoft Setup screen.

Basic Procedures for Sound Level Measurement:

1. Channel setup

2. Microphone calibration

3. Measurement

This value is calculated directly in the Medium & Calibration field of the sound level module’s channel setup. To perform the calibration, connect the calibrator to the microphone and switch it on. The signal is displayed in the small overview window and, in this case, should appear as a sine wave at 1000 Hz. Since all frequency-weighted curves are referenced to 1000 Hz, this is the standard frequency used for calibrating microphones.

It is also possible to select the medium in which the measurement is performed. The options are Air or Water, with the difference being the reference sound pressure.

Once the sound is correctly recognized as a sine wave at 1000 Hz, click the Calibrate button to perform the calibration. The sound module then calculates the microphone’s sensitivity based on the highest FFT amplitude and the reference value.

The sensitivity is automatically corrected in the source channel, so no additional analog scaling is required. The calibrated sensitivity can be verified against the information provided on the calibration certificate.

Next, check whether the calibration was successful:

1. Set a sampling rate of at least 5 kS/s (a range of 20–50 kS/s is recommended).

2. Open the FFT Analysis.

3. Set the FFT options to Flat Top Filter and the Y-axis scale type to dB Noise.

4. Press the RMS icon to display the RMS values in the FFT graph.

5. Switch on the microphone calibrator again. The RMS values should display 94 dB.

If the values do not match, repeat the calibration process.

Because of differences in their construction, microphones have unique characteristic responses to sound. These differences result in non-uniform phase and frequency responses.

The dynamic range of a microphone is the difference in SPL (sound pressure level) between the noise floor and the maximum measurable sound pressure level. The sensitivity of a microphone indicates how effectively it converts acoustic pressure into an output voltage (unit: mV/Pa). A high-sensitivity microphone produces more voltage for a given pressure, while a low-sensitivity microphone produces less.

Microphones are not uniformly sensitive to sound pressure and can handle varying levels without distortion. For scientific applications, microphones with a more uniform response are preferred. However, in music recording, a non-uniform response can be desirable, as it adds coloration to the sound. This variability makes it difficult to compare published data from different manufacturers, since different measurement techniques are often used.

The frequency response diagram plots microphone sensitivity in decibels across a range of frequencies (typically 20 Hz to 20 kHz). For example, a stated response of 30 Hz – 16 kHz ±3 dB indicates a nearly flat, linear response between the specified frequencies, with amplitude variations no greater than ±3 dB. By contrast, broad claims such as 20 Hz – 20 kHz are essentially meaningless without a tolerance value. For directional microphones, the frequency response also depends on the distance from the sound source and the geometry of the source itself.

The noise level is the sound pressure level that produces the same output voltage as the microphone generates in the absence of sound. This represents the lowest point of the microphone’s dynamic range and is important when recording quiet sounds. Noise level is often specified in dB(A), which expresses the equivalent loudness of the noise on a decibel scale, frequency-weighted to approximate human hearing (A-weighting).

Example 1

In the example below, we measured the sound produced by an accordion. A condenser microphone was placed near the instrument, and we captured the beating frequency of the signal. The microphone output clearly shows that the accordion does not produce a single pure tone but rather two closely spaced frequencies. These are perceived as a beating effect.

Example 2

The next example with the condenser microphone was performed using a cantilever beam. We measured the beam’s natural frequency. The microphone was connected to the ACC module and set to IEPE mode. Calibration was carried out as described on the previous pages. The beam was excited with a hammer, after which it vibrated at its natural frequency.

The microphone signal was analyzed using the FFT analyzer.

The first peak in the FFT spectrum appeared at 91.55 Hz, which closely matched the beam’s natural frequency determined using the frequency response function (FRF) method (91 Hz).