Sound Power Measurement

Course overview

The course equips you with essential knowledge and practical skills to accurately quantify a noise source’s acoustic output using DewesoftX.

You’ll begin by understanding sound power fundamentals—how it differs from pressure, and why it provides a reliable comparison basis. The course introduces key ISO standards (3741, 3744, 3745), detailing their requirements for testing environments and accuracy levels.

In hands-on modules, you’ll set up test geometry: define measurement surfaces (box, sphere, hemi-sphere), place microphone arrays to conform to standard geometries, and connect calibration devices. Using the Sound Power plugin, you’ll configure these settings—including calibration, frequency-weighting, and trigger logic—to match test requirements.

The course covers acquisition workflow: perform background noise measurement (source off), collect data with the source on, and apply correction factors (K₁ for ambient noise, K₂ for room environment, meteorological corrections, etc.). You’ll manage measurement timing, perform real-time data capture, and execute offline calculations—ensuring results are accurate and standards-compliant.

Finally, you’ll generate results that include frequency-band sound power levels, reports with visual outputs, and data exports suitable for documentation or further analysis. The course concludes with post-measurement reporting, enabling you to produce complete, ISO-compliant deliverables.

Sound is a pressure wave—a vibration that propagates as a mechanical wave of pressure and displacement.

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 sound source (such as a vibrating diaphragm or a stereo speaker), which creates vibrations in the surrounding medium. As the source continues to vibrate, these vibrations propagate outward 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 continuously over time.

Sound pressure

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

Sound pressure level

Sound Pressure Level (SPL), or simply sound level, is a logarithmic measure of the effective sound pressure relative to a reference value. It is expressed in decibels (dB) above this standard reference. The standard reference sound pressure in air or other gases is 20 µPa, which is generally considered the threshold of human hearing (at 1 kHz).

The equation below shows how to calculate the sound pressure level (Lp) in decibels [dB] from the measured sound pressure (p) in pascals [Pa].

where pref is the reference sound pressure, and prms is the RMS sound pressure being measured.

Most sound level measurements are made relative to this reference level. For example, 1 pascal corresponds to an SPL of 94 dB. In other media, such as underwater, a different reference level is used—typically pref = 1 µPa.

The lower limit of audibility is defined as an SPL of 0 dB, but the upper limit is not as clearly defined. In Earth’s atmosphere, the largest pressure variation an undistorted sound wave can have is approximately 1 atm (194 dB Peak or 191 dB SPL). However, larger sound waves can exist in other atmospheres or in different media, such as underwater or within the Earth itself.

Ears detect changes in sound pressure. Human hearing does not have a flat spectral sensitivity (frequency response relative to frequency and amplitude). People do not perceive low- and high-frequency sounds as well as they perceive sounds around 2000 Hz, as shown in the equal-loudness contour.

Because the frequency response of human hearing varies with amplitude, weighting curves have been established to standardize sound pressure measurements.

The sound field is the area in which sound exists. When measuring sound pressure, it is very important to distinguish whether the sound field is free or diffuse:

• Free sound field – This exists in a space without reflections. It can be simulated outdoors or in an isolation room where all sound striking the walls is absorbed. The main characteristic of a free sound field is that sound spreads spherically.

• Diffuse sound field – This exists in a reverberation room, where sound reflects so many times that it travels in all directions with equal probability.

Sound intensity

Sound intensity is defined as the sound power per unit area. It depends on the distance from the sound source and the acoustic environment in which the source is located. Sound intensity is a vector quantity that describes both the amount and the direction of sound energy. Its unit is watts per square meter [W/m²].

It is calculated as the product of sound pressure (p) and the particle velocity (c):

Sound power

Sound power is a characteristic of a sound source and is independent of distance. This makes it a practical way to compare different sound sources. Sound power can be measured in various ways, such as by using sound pressure or sound intensity.

Sound power level is determined as follows:

The SoundPower plugin performs the calculation by measuring the sound pressure.

Sound power is calculated using the following equation:

• \(L_W\) = sound power

• \(L_p\) = sound pressure level

• \(S\) = measurement surface

• \(S_0\) = referenced surface (is by standard 1 m2)

• \(K_1\) = background noise correction

• \(K_2\) = room noise correction

• \(C_1\) = meteorological correction

• \(C_2\) = meteorological correction

If a free-field measurement is performed with microphones arranged on a sphere, then:

• LpL_pLp​ is the sound pressure level measured by 10 microphones.

• The surface SSS is 4πr24\pi r^24πr2.

• The background noise correction K1K_1K1​ accounts for noise emitted from surrounding machinery.

• K2K_2K2​ is zero because the measurement is conducted in free-field conditions.

• C1C_1C1​ and C2C_2C2​ are also zero, since the outside temperature is around 23 °C and the test site altitude is below 500 meters above sea level.

The Sound Power plugin supports several ISO standards for determining sound power by measuring sound pressure.

Both standards define:

• types of noise and noise sources,

• test environment,

• measurement uncertainty,

• criteria for background noise,

• criteria for air temperature and humidity,

• instrumentation equipment, and

• position of microphones.

ISO 3741 – Specifies precision methods for determining the sound power level of a noise source from sound pressure levels measured in a reverberation test room (a room designed to create a diffuse or random-incidence sound field). The methods defined in ISO 3741 are suitable for all types of noise, including steady, non-steady, fluctuating, and isolated bursts of sound energy.

ISO 3744 – Specifies methods for determining the sound power level or sound energy level of a noise source from sound pressure levels measured on a surface enveloping the source (e.g., machinery or equipment) in an environment that approximates a free field near one or more reflecting planes. The sound power level (or, in the case of transient noise emissions, the sound energy level) is then calculated in frequency bands or with A-weighting applied.

ISO 3745 – Specifies various methods for determining the sound power levels and sound energy levels of noise sources, including machinery, equipment, and their subassemblies. The choice of method depends on the purpose of the test (to determine sound power or sound energy levels) and the facilities available.

To facilitate the selection of the shape and dimensions of the measurement surface, the reference box must first be clearly defined. The reference box is a hypothetical surface represented by the smallest right parallelepiped that fully encloses the source under test. When defining its dimensions, elements protruding from the source that are known not to be significant radiators of sound may be disregarded.

The positions of the reference box, the measurement surface, and the microphone locations are defined with respect to a coordinate system with origin O in the ground plane, as illustrated in the figures below. Point O is the midpoint of a box formed by the reference box and its mirrored images in the adjoining reflecting plane(s). The horizontal axes x and y of the coordinate system lie in the ground plane, parallel to the length and width of the reference box.

The characteristic source dimension, d₀, which is used to determine the dimensions of the measurement surface, is also shown in the figures below for reference boxes located on one, two, or three reflecting planes.

The images below illustrate different positions of a reference box near acoustically reflecting planes.

• \(d_0\) - characteristic source dimension

• \(I_1\) - reference box width

• \(I_2\) - reference box length

• \(I_3\) - reference box height

• \(O\) - origin of sound

One acoustically reflecting plane

Characteristic source dimension, in this case:

Two acoustically reflecting planes

Characteristic source dimension, in this case:

Three acoustically reflecting planes

Characteristic source dimension, in this case:

To use the Sound Power plugin, please download it from our webpage and select SoundPower.dll. Copy the file SoundPower.dll into the Addons folder of your DewesoftX installation (e.g., D:\Dewesoft7\Bin\X3\Addons\), and then start DewesoftX.

To enable the plugin, open the Settings menu in DewesoftX. Navigate to the Extensions section and add a new plugin by clicking the plus (+) button. Locate the Sound Power plugin as shown in Image 9, and enable it.

Once enabled, the Sound Power plugin will appear in the Extensions tree list. The next time you open Channel Setup, the Sound Power icon will be available, as shown in Image 10.

ISO 3741 specifies precision methods for determining the sound power level of a noise source from sound pressure levels measured in a reverberation test room (a room designed to create a diffuse or random-incidence sound field).

The methods described in ISO 3741 are suitable for all types of noise, including steady, non-steady, fluctuating, and isolated bursts of sound energy.

The sound power level produced by the noise source, expressed in one-third-octave frequency bands, is calculated from these measurements. Corrections are applied to account for differences between the meteorological conditions during the test and those corresponding to a reference characteristic impedance.

Measurement and calculation procedures are provided for both:

• the direct method (using the equivalent sound absorption area of the reverberation test room), and

• the comparison method (using a reference sound source of known sound power level).

ISO 3741 is applicable to noise sources with a volume not greater than 2% of the volume of the reverberation test room.

Hemisphere

The sound source is placed at the center of the hemisphere. When positioned over one acoustically reflective plane, the measurement surface is expressed as:

S=2πr2S = 2\pi r^2S=2πr2

Key microphone positions are marked with numbers 1–10, and additional positions with 11–20.

When the sound source is placed over two acoustically reflective planes, the measurement surface is expressed as:

S=πr2S = \pi r^2S=πr2

Key microphone positions are marked with numbers 2, 3, 6, 7, and 9, and additional positions with 11, 14, 15, and 18.

When the sound source is placed over three acoustically reflective planes, the measurement surface is expressed as:

S=πr22S = \frac{\pi r^2}{2}S=2πr2​

Key microphone positions are marked with numbers 1, 2, and 3, and additional positions with 4, 5, and 6.

Parallelepiped

The parallelepiped measurement surface must have the same orientation as the reference box. The distance between the reference box and the measurement surface must be at least 0.25 m, denoted as ddd.

The length, width, and height of the reference parallelepiped are denoted as l1l_1l1​, l2l_2l2​, and l3l_3l3​.

When the sound source is placed over one acoustically reflective plane, the measurement surface is expressed as:

S=4(ab+bc+ac)S = 4(ab + bc + ac)S=4(ab+bc+ac)

\(a=0.5l_1 + d\)

\(b=0.5l_2 + d\)

\(c = l_3 + d\)

• When the sound source is placed over two acoustically reflective planes, the measurement surface is expressed as \(S=2(2ab+bc+2ac)\)

\(a=0.5l_2 + 0.5 d\)

\(b = 0.5l_1 + d\)

\(c=l_3 + d\)

• When the sound source is placed over three acoustically reflective planes, the measurement surface is expressed as \(S=2(2ac + cb + ac\)

\(a=0.5l_1 + 0.5 d\)

\(b = 0.5l_2 + d\)

\(c=l_3 + d\)

The parallelepiped in ISO 3744 has, in addition to walls and corners, several defined types/sizes (small, tall, long, medium, large).

All of these types/sizes use the same equation for calculating the measurement surface. The only difference lies in the number of microphones required. This means that you can select the Small type/size and still measure a Large type/size device, as long as you enter the correct value for the number of microphones corresponding to the Large type/size.

Cylinder

The reference box must be positioned at the center of the cylinder. The distances between the cylinder and the reference box are marked as d1d_1d1​, d2d_2d2​, and d3d_3d3​. The radius of the cylinder is expressed as:

The height of the cylinder is expressed as:

The distances d1d_1d1​ and d3d_3d3​ must be set according to the size of the sound source (at least 0.5 m). From d1d_1d1​ and d3d_3d3​, we calculate hhh, RRR, and also d2d_2d2​.

The total measurement surface SSS is the sum of the area of the upper circle STS_TST​ and the area of the cylindrical layer SSS_SSS​.

• When the sound source lies on one acoustically reflective surface:

  ST=πR2andSS=2πRhS_T = \pi R^2 \quad \text{and} \quad S_S = 2\pi R hST​=πR2andSS​=2πRh

• When the sound source lies on two acoustically reflective surfaces:

  ST=πR22andSS=πRhS_T = \frac{\pi R^2}{2} \quad \text{and} \quad S_S = \pi R hST​=2πR2​andSS​=πRh

• When the sound source lies on three acoustically reflective surfaces, the area of the upper circle is

  ST=πR24S_T = \frac{\pi R^2}{4}ST​=4πR2​

  and the area of the cylindrical layer is

  SS=πRh2.S_S = \frac{\pi R h}{2}.SS​=2πRh​.

The standard describes different microphone arrangements around a sound source.

The hemispherical measurement surface shall be centered on a point on the floor of the test room, vertically beneath the assumed acoustic center of the noise source under test—either the actual acoustic center if known, or the geometric center if the acoustic center is unknown. The measurement radius rrr shall satisfy all of the following conditions:

a) r≥2d0r \geq 2d_0r≥2d0​ or r≥3h0r \geq 3h_0r≥3h0​, whichever is larger, where d0d_0d0​ is the characteristic dimension of the noise source under test, and h0h_0h0​ is the distance from the acoustic center of the source to the floor.

b) r≥λ4r \geq \frac{\lambda}{4}r≥4λ​, where λ\lambdaλ is the wavelength of sound at the lowest frequency of interest.

c) r≥1 mr \geq 1 \, \text{m}r≥1m.

The measurement surface must be fully contained within the qualified region of the hemi-anechoic room. For small, low-noise sources measured over a limited frequency range, the measurement radius may be less than 1 m, but not less than 0.5 m. However, conditions a) and b) still apply, and using a radius smaller than 1 m may impose restrictions on the frequency range over which tests can be performed.

• The spherical measurement surface shall be centered on the acoustic center of the noise source under test—either the actual acoustic center if known, or an assumed acoustic center such as the geometric center of the source. The measurement radius rrr shall satisfy all of the following conditions:

  a) r≥2d0r \geq 2d_0r≥2d0​, where d0d_0d0​ is the characteristic dimension of the noise source under test.

  b) r≥λ4r \geq \frac{\lambda}{4}r≥4λ​, where λ\lambdaλ is the wavelength of sound at the lowest frequency of interest.

  c) r≥1 mr \geq 1 \, \text{m}r≥1m.

  The measurement surface must be fully contained within the qualified region of the anechoic room. For small, low-noise sources measured over a limited frequency range, the measurement radius may be less than 1 m, but not less than 0.5 m. However, conditions a) and b) remain applicable, and using a radius smaller than 1 m could impose restrictions on the frequency range over which tests can be performed.

  The area of a spherical measurement surface is given by:

  S=4πr2S = 4\pi r^2S=4πr2

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

Microphones can be calibrated in two ways. First, it is important to understand that the direct output of the microphone represents sound pressure in pascals (Pa). Therefore, it must be scaled to the correct physical quantity.

Scaling with a calibration certificate

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

First, Pa is defined as the physical unit of measurement. Next, open 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 the calibrator

Another method is to calibrate the microphone using a calibrator. In this case, the known parameter is the sound level emitted by the calibrator. For example, in our case, it is 94 dB at 1000 Hz.

First, we need to enter the channel setup of the microphone. The sensitivity is set to 1 by default. On the right side of the microphone scaling section, information from the microphone—placed in the calibrator—can be seen. The calibration frequency is set to 1000 Hz, and the current value detected by the microphone is 127.4 dB. This is, of course, incorrect, since our calibrator has an output value of 94 dB.

After pressing Calibrate, the microphone’s sensitivity will be measured from the highest peak in the frequency spectrum, usually at 1000 Hz (using amplitude correction to obtain the correct amplitude). Microphone sensitivity can also be read from TEDS. In that case, no manual calibration is needed, as the sensitivity is stored in the TEDS.

Once the Calibrate button is pressed, the sensitivity value changes. Under the current value, we can now see 94 dB, which confirms that the microphone is calibrated.

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 difference, frequency weighting curves are used, which approximate the response of the human ear.

The most widely known example is frequency weighting in sound level measurement, where a specific set of weighting curves—A, B, C, and D, as defined in IEC 61672—are applied.

Unweighted measurements of sound pressure do not correspond to perceived loudness because the human ear is less sensitive at both very low and very high frequencies. The weighting curves are applied to the measured sound level using a weighting filter in a sound level meter.

Unlike FFT analysis, which has a specific number of lines per linear frequency (x-axis), CPB (constant percentage bandwidth, also called octave analysis) uses a logarithmic frequency x-axis. This means that lower frequencies have more lines, while higher frequencies have fewer lines. CPB analysis is traditionally used in the sound and vibration field.

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

Filters with the same constant percentage bandwidth (CPB filters), e.g., 1/1 octave, are normally displayed on a logarithmic frequency scale. Sometimes, these are also called relative bandwidth filters. CPB filters with logarithmic scaling are almost always used in acoustic measurements, because they provide a close approximation to how the human ear responds.

The widest octave filter has a bandwidth of 1 octave, but many subdivisions into smaller bandwidths are commonly used. These filters are often labeled Constant Percentage Bandwidth filters. For example, a 1/1 octave filter has a bandwidth of about 70% of its center frequency. The most widely used filters are those with 1/3 octave bandwidths, as this bandwidth corresponds well to the frequency selectivity of the human auditory system above 500 Hz. In specialized cases, bandwidths as narrow as 1/96 octave can be realized.

A detailed signal with many frequency components appears with a dotted curve when analyzed using octave filtering. In contrast, a 1/3 octave analysis produces a solid curve, showing higher resolution and revealing more details.

1/1 octave filter

1/3 octave filter

1/12 octave filter

The Dewesoft X3 Sound Power module can be divided into the following sections:

Basic section

In the Basic section, we must select the standard according to which the measurement will be performed. Depending on the chosen standard, different microphone positions can be selected.

ISO 3741

For both the direct and comparison methods, the minimum distance between the noise source and the nearest microphone position is determined by the reverberation room volume VVV, the reverberation time T60T_{60}T60​, and a constant D1D_1D1​, where:

• D1=0.08D_1 = 0.08D1​=0.08, and

• D1=0.16D_1 = 0.16D1​=0.16 for frequencies below 5000 Hz.

ISO 3744/639x

There are several microphone position options to choose from:

• Parallelepiped

• Cylindrical

• Hemispherical

• Custom

In addition, the device type/size must also be defined.

ISO 3745

According to ISO 3745, the microphone positions can be defined as hemispherical, spherical, or custom.

ISO 3743

ISO 3743 specifies methods for determining the sound power level or sound energy level of a noise source by comparing the measured sound pressure levels emitted by the source (e.g., machinery or equipment) mounted in a hard-walled test room—with characteristics specified in the standard—against those from a calibrated reference sound source.

User entered coordinates for microphone position section 

If we check the Enter mic. coordinates checkbox, we can manually define the microphone positions by specifying the X, Y, and Z coordinates for each position.

Geometry section

To calculate the sound power, we need the surface area of the reference box (the device under test). The required parameters vary depending on the chosen microphone positions.

Parallelepiped

Here, L1, L2, and L3 represent the width, depth, and height of the measuring device, while d represents the distance between the device and the measurement surface.

Sphere and Hemisphere

Insert the radius of the measurement surface.

Cylinder

Enter the values for L1, L2, and L3, which represent the width, depth, and length of the measured device. Also, provide the values for D1, D2, and D3, which represent the width, depth, and length of the distance between the device and the measurement surface.

Surface correction factor

From the geometry distances, the software also calculates the surface correction factor Ls, which is used in the sound power calculation.

Here, S represents the measurement surface, and S₀ is the reference measurement surface, where S₀ = 1 m².

ISO 3741, ISO 3744, and ISO 3745 also specify the required number of microphones, depending on the chosen geometry setup. If the number of microphones differs from the standard, a warning will appear, but you can still continue.

If you have fewer physical microphones than required microphone positions, you can arrange them in groups and move them step by step during the measurement (e.g., rotating them through four positions in a cylindrical arrangement). In the end, all data will be combined. Since some standards require 20 or more microphones, you can reduce equipment costs by, for example, grouping 5 microphones and moving them through all 4 positions step by step. Afterward, the software automatically combines all the data.

Microphone position

In the Microphone Positions section, you can assign the analog channels to specific positions. Simply click on the text fields and select from the drop-down menu. The X, Y, and Z coordinates of the microphones, as well as their group assignments, are displayed.

If the number of microphones exceeds the maximum allowed by ISO 3741, ISO 3744, or ISO 3745, the coordinates will be set to zero.

If you check the Enter Mic. Coordinates checkbox, you can manually enter the coordinates of the microphones.

In the analysis setup, you need to select the CPB resolution along with the bandwidth, frequency weighting, and measurement time. These parameters can also be modified and recalculated offline after the data file has been stored.

Octave settings

For frequency analysis, constant percentage bandwidth (CPB) is used. You can choose between 1/1 octave and 1/3 octave.

Filters with the same constant percentage bandwidth (CPB filters), such as the 1/1 octave, are normally displayed on a logarithmic frequency scale. These filters are sometimes also referred to as relative bandwidth filters. Analysis with CPB filters (and logarithmic scales) is commonly used in acoustic measurements because it provides a close approximation of how the human ear responds.

The widest octave filter has a bandwidth of 1 octave, but it is often subdivided into smaller bandwidths. These are typically labeled as Constant Percentage Bandwidth filters. A 1/1 octave filter has a bandwidth of roughly 70% of its center frequency. Among the most widely used filters are those with 1/3 octave bandwidths, as they correspond well to the frequency selectivity of the human auditory system, particularly at frequencies above 500 Hz.

Bandwidth

In the Bandwidth section, you can select the frequency band for the calculation. Values are chosen from a drop-down menu and depend on the selected acquisition rate and octave setting (1/1 or 1/3).

Weighting

Because human hearing is non-linear, frequency weighting must be applied. The plugin supports A-weighting as well as no weighting (linear). A-weighting is commonly used to account for the relative loudness perceived by the human ear.

Measurement conditions

Measurement time is the operational period or cycle of the noise source under test during which the time-averaged sound pressure level is determined.

The length of the measurement time affects the certainty of the results. To ensure reliability, the measurement interval should be at least 20 seconds or longer.

In offline calculation, cursors (L1 and L2) can be placed to select the calculation range between them (see Image 43).

Repeatability

The uncertainty due to the repeatability of sound pressure level measurements refers to the closeness of agreement between successive results obtained under the same conditions. It may be determined from the standard deviation of repeatability using six measurements of the decibel sound pressure levels, uncorrected for background noise, at a single microphone position.

Measurement repeatability can be strongly influenced by the averaging time. If the averaging time does not cover a sufficient number of machinery cycles, the total uncertainty may become unacceptably large for an engineering-grade standard. For extremely low-noise sources, reducing background noise can lower the sensitivity coefficient and decrease the total uncertainty by up to a factor of two.

The component of uncertainty can also be reduced through better control of machinery operating conditions, longer averaging times, or by averaging multiple measurements under appropriately modified conditions to represent a typical case.

History channels of multiple runs

Select the option Number of runs (e.g., operation modes of a sound source) if you want to measure both sound power level and sound pressure level (see Image 44).

During measurement, select the desired run from the drop-down list.

If you want to rename the runs, click the Runs editor button, located to the right of the Number of runs option, and change the run name to suit your application.

In the exported result matrix, you will see the SPL of each microphone for each run, as well as the average values.

The sound power measurement can also be triggered using a trigger channel (e.g., light barriers) instead of clicking the start and stop buttons.

Select the Use trigger channels checkbox, then choose the first and second trigger channels from the drop-down menu. Trigger channels can be synchronous, asynchronous, or single-value channels.

To define the trigger and retrigger levels, enter the appropriate settings in the trigger configuration.

One trigger channel

The same channel can also be used to both start and stop the measurement.

Measurement starts with the first detected trigger edge on the trigger channel and stops with the next trigger edge on the same channel.

Different trigger channels

Alternatively, one channel can be used to start the measurement and a second channel to stop it.

In this case, the measurement begins with the first detected trigger edge on the first channel and ends with the inverted trigger edge on the second channel.

C1 and C2 meteorological corrections are applied when the air temperature at the test site is below 23 °C or the altitude is higher than 500 meters above sea level. When this option is activated, the correction is immediately calculated and displayed next to the label. You can either enter the barometric pressure or the altitude, and the other value will be calculated automatically.

C1 is the reference quantity correction, expressed in decibels, which accounts for the different reference quantities used to calculate the decibel sound pressure level and the decibel sound power level. It is a function of the characteristic acoustic impedance of the air under the meteorological conditions at the time and place of measurement.

C2 is the acoustic radiation impedance correction, expressed in decibels. It converts the actual sound power relevant to the meteorological conditions at the time and place of measurement into the sound power under reference meteorological conditions. The value should be obtained from the appropriate noise test code. In the absence of such a code, the following equation is valid for a monopole source and provides a mean value for other sources.

K1 background noise correction

K1 is defined in the standard as the background noise correction. It is applied in various situations (refer to the standard for details). K1 can either be measured before switching on the sound source or calculated afterward by correctly placing cursors on the measured data.

The K1 correction is applied to the mean (energy average) of the time-averaged sound pressure levels across all microphone positions on the measurement surface, to account for the influence of background noise. This correction is expressed in decibels [dB].

You can also define the K1 correction factor as a table. Measure K1 once, and reuse the same correction multiple times. In the K1 table editor, enter the correction factors for each frequency band.

K2 room correction

K2 corrects for the influence of room noise. This correction is applied to the mean (energy average) of the time-averaged sound pressure levels across all microphone positions on the measurement surface, to account for reflected or absorbed sound. The environmental correction is expressed in decibels [dB].

Three methods are implemented in the Sound Power plugin:

• Mean absorption grade – Enter the room size and the mean absorption grade according to the standard.

Reverberation time

These parameters can be determined through acoustic measurement.

Enter values

An editor is provided—first click Create K2 table. Ensure that you select the correct bandwidth beforehand; otherwise, the table will be reconfigured when the bandwidth is changed, and the entered values may be lost. The values entered in the table are saved to the Dewesoft setup.

Calculate from RSS

The correction uses measured values from a reference sound source with a known sound power level.

When you switch to Measure mode, an auto-generated display called SoundPower appears. Initially, all graphs are empty.

• Action buttons are provided at the top, allowing the user to proceed through the different steps—acquiring background noise, starting the actual measurement, and switching between microphone groups.

  The Message, Status, and Warning displays show the measurement progress (e.g., 17.2 / 20 sec completed; K1 background acquiring…) and provide warnings if certain conditions are not met (e.g., Acquisition finished early…).

  On the left side, the input signals are displayed: the overall sound pressure levels of the microphones in digital meters and the CPB plots below.

  On the right side, the output signals are displayed: SoundPower and Overall Sound Pressure of all microphones in digital meters, along with the CPB plots of SoundPower, SoundLevel, and corrections.

In this example, four microphones are used and arranged into two groups, which means the setup must be changed once during the process. The measurement time remains at 20 seconds, which is the default value.

The flowchart for this example is shown in Image 65. Using the Previous and Next buttons, you can switch between groups or repeat a measurement, as long as the data for the group has not yet been fully acquired.

Start storing (with the sound source switched off). Note that nothing will change on the display yet. Next, click Acquire background (K1) and wait for 20 seconds. The progress is shown in the status display.

After the K1 acquisition is complete, switch on the sound source. Click Start Acquisition and wait another 20 seconds. The four-microphone CPB plots (Group A) below should then populate with data.

After the acquisition is complete, switch off the sound source and change the microphone setup. The four microphones should then be moved to their second position. Next, switch to Group B.

Click Acquire background (K1) and wait for 20 seconds while keeping the sound source switched off. Since the microphone positions have changed, the K1 measurement must be repeated for this position.

Now switch the sound source back on and select Run 1 instead of the K1 measurement. Click Start Acquisition and wait for another 20 seconds. After this, the four displays on the left side (Group B) should also populate with data.

Finally, the message Sound power measurement finished should appear. If something went wrong, more details will be provided in the warning display. The CPB displays on the right side will show the SoundPower, SoundLevel, and K1 factor results.

The Sound Power plugin also supports offline calculation, which means changes to the calculations can be applied to the measured data (however, changing microphone groups is not possible).

It is even possible to collect only raw data at the test site and perform all calculations later in the office. To ensure accuracy, make sure that enough data is collected—typically 20 seconds of background noise and 20 seconds with the sound source switched on.

Below is an example showing the minimum required data (here using three microphones):

In Analysis mode, with the data file open, go to Offline Math and add the Sound Power module. Configure the settings in the setup.

Then return to Review and click Recalculate. The calculation may not run yet, and a warning can appear if K1 correction is enabled. See the next page for instructions on placing the cursors.

We need to provide the plugin with information about where the background noise is located and where to look for the sound source data. There are four different ways to place the cursors when performing an offline sound power analysis.

If the offline calculation is performed without placing and locking the cursors, the plugin will take 20 seconds of data from the beginning of the file. This option does not take the K1 correction factor into account.

• If only the first cursor is locked at a fixed position, the plugin will take 20 seconds of data starting from that cursor position. This option also does not take the K1 correction factor into account.

• With this option, both the K1 background correction factor and the sound power level of the sound source are calculated. Lock the first cursor at the beginning of the background noise signal and the second cursor at the point where the sound source is turned on.

• Lock the cursors and select Calculate between two cursors (L1 and L2).

To lock the cursors, first place them at the desired positions and then click the cursor symbols on the left. The cursors will then be locked.

Let’s look at an example using offline calculation. Raw data was collected from four microphones. For about 40 seconds, the sound source was off, and then it was switched on.

In the picture below, you can see the data file containing the raw microphone data.

First, place the cursors to indicate where the background noise begins and where the signal from the sound source starts. The first cursor represents the beginning of the noise, and the second cursor marks the point where the sound source is turned on. After placing the cursors, lock them to confirm their positions.

The next step is to set up the SoundPower plugin. Go to Offline Math and select Sound Power. Choose the ISO standard, microphone position, and the number of microphones. The measurement time should be at least 20 seconds—it will take 20 seconds of data starting from the placed cursor.

Then return to Review and click the Recalculate button. Once the recalculation is complete, you will see that the sound power measurement has finished.

We measured the sound power level of a laptop in three different operating states. The minimum measurement time for each state was 20 seconds, as specified by the standard.

For the measurement, we used the hemisphere kit from G.R.A.S with 10 microphones and a radius of 1.36 meters. The distance from the sound source to the microphones was 1 meter. The grid is simple to assemble by following the attached instruction manual, and it took us approximately 30 minutes to prepare everything.

First, we assembled the metal rods to form a stable hemisphere structure. Next, we positioned the microphones in the correct locations according to the standards. The grid supports either 10 or 20 microphones, so we selected the appropriate mounting positions. These positions must match the standard and are already predefined with small, color-coded screws.

We used 10 BNC cables, each 10 meters long, so that the measurement equipment could be placed outside the semi-anechoic room. This setup is very important for minimizing background noise. The microphones were equipped with TEDS chips and had been factory-calibrated. The entire data acquisition process took approximately 10 minutes.

The setup was arranged as shown in the picture below.

The results are presented as a CPB analysis along with an overall dB value.

When the calculations are finished, there are several options for exporting the data from DewesoftX.

You can either print the current display arrangement, export the data to the clipboard, or send it to external software.

For the first option, simply click the Print button and select a printer, which may also be a PDF writer.

A quick way to export data is by using the clipboard. Click on the instrument (e.g., the Octave plot showing the K1 correction data) to activate it. Then, from the Edit menu, select Copy to Clipboard → Widget Data. Open MS Excel and paste the data—the columns and rows will match exactly what you see in the display.

Alternatively, you can use the default export function, which supports many file formats such as Matlab, Excel, Diadem, RPCIII, and CSV, as shown in Image 91.

After completing the sound power measurement, you can easily export the data into a prepared Excel template to automatically generate your report with a single click.

All the necessary files for creating a SoundPower report can be found on our website under the Download section:

Needed prerequisites:

• Microsoft Excel must be installed.

  Import the data header file SoundPowerDataHeader.xml in DewesoftX → Settings → Data Header. Then specify when the user should be prompted to enter details (Ask for header on start or Ask for header on end).

• Copy the template files to your local DewesoftX → Script folder (e.g., D:\Dewesoft\System\X2\Scripts):

  • SoundPowerReport.xps

  • SoundPowerReport1.xlt

Creating the report

Restart DewesoftX, open an existing sound power data file (the data header must have been filled in during the measurement), then go to Export and select the Excel ribbon. The Sound Power template should now appear in the list on the left.

Next, deselect all channels in the list on the right. Synchronous channels take too much time to export and are not needed for the report.

From the Sampling column, select only Single Value Channels. This will include all CPB plots and final results.

From the Sampling column, select only Single Value Channels. This will include all CPB plots and final results.

Then click Export. Excel will automatically open and populate the template with all the data.

The exported Excel file contains multiple sheets:

• DataInfo – data header details

• Events – events such as start/stop storing or notes added during measurement

• Single Value – sound power CPB and overall levels

• Data1 – usually contains full-speed data (e.g., at 50 kHz), but this was deselected

• Results – the final report

The Results sheet is linked to other pages, as shown in Image 99.

Modifying the template

If you want to modify the example template (e.g., to add a company logo), select Edit Template.

Excel will then open the template file (note the .xltx extension).

The template is already filled with the current data. To modify a cell, press the equal sign (=), then navigate to one of the automatically filled sheets.

Select the desired cell and press Enter.

The cell will now be linked.

Finally, press Save in Excel.