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Glossary of Basic Terms Related to FFT Analysis

Isolation

Isolation refers to the process of insulating electrical signals, which is necessary when the common of the signal source differs from the ground potential. If the difference between the signal common and the ground potential (common-mode voltage) is small, measurements can be made using differential inputs; however, when the potential difference exceeds 100 V, it results in excessive input, making measurement impossible. In such cases, isolation amplifiers may be used, and for FFT analyzers, the analog input section up to the amplifier, anti-aliasing filter, and AD converter is isolated from the chassis ground using phot couplers for each channel. This isolation is advantageous for eliminating ground loops or avoiding connections to the common of the signal source.

Impulse response

The response of the system when a unit impulse δ (t ) is added to a linear system is called the impulse response h (t). The impulse response represents the characteristics of the system in the time domain, while the representation in the frequency domain is referred to as the frequency response function.

If the impulse response of the system is known, the output y(t) when the input x(t) is applied to that system can be obtained through the operation of convolution.

Our company's FFT analyzer calculates the impulse response by performing an inverse Fourier transform on the frequency response function.

phase spectrum

Phase representation as a function of frequency is mainly;

(1) Phase spectrum of 1 channel

Phase difference between two channels

There are two types.

(1) Phase spectrum of 1 channel

time functionPhase Spectrum of Channel 1_No.1The Fourier transform of;

  • (1) Phase Spectrum of Channel 1_No.2

(1) formula's(1) Phase Spectrum of 1 Channel_NO.3In this context, it is specifically referred to as the (complex) Fourier spectrum.(1) Phase Spectrum of 1 Channel_NO.4Since it is a complex function, it can be expressed in terms of amplitude and phase from its real and imaginary parts.

  • (1) Phase Spectrum of 1 Channel_NO.5

than

  • (1) Phase Spectrum of 1 Channel_NO.6
  • (1) Phase Spectrum of 1 Channel_NO.7

(2) is the amplitude of the Fourier spectrum (MAG representation), and (3) is specifically referred to as the phase spectrum in this device.

In this device, the phase is set to 0 degrees for the cosine wave, with the starting point of the data frame as the time origin. Even if X (f ) represents the same signal, if the phase spectrum θ (f ) is different, the waveform of the time signal x (t ) will change significantly.

When measuring phase spectra, a trigger function is usually utilized to observe the phase delay relative to a reference position. This method is primarily applied in the field balancing of rotating bodies.

Phase difference between two channels.

The phase difference between the 2 channels can be obtained as the phase representation of the transfer function (or cross spectrum), which is a complex function. Let the transfer function (frequency response function) of the system be H (f );

  • (2) Phase difference between two channels_NO.1
  • (2) Phase difference between two channels_NO.2

| H (f )|represents the gain characteristics of the system, and θ (f ) represents the phase difference between the two channels.

Wigner distribution

The Wigner distribution was proposed by E. Wigner in the field of quantum mechanics. It has properties similar to an extended power spectrum for nonstationary signals. With conventional FFT, time resolution and frequency resolution are complementary (increasing frequency resolution requires a longer sampling time), making it difficult to obtain instantaneous spectra of nonstationary signals with good resolution. In contrast, the Wigner distribution is not directly subject to this complementary constraint between time and frequency resolution, allowing good time-frequency resolution of the power spectrum on the frequency-time plane. However, it was not practical because it requires far more computation points than FFT. Using the OS-0263 Time-Frequency Analysis Software option for the OS-2000 series time-series data analysis tool (software), you can analyze the Wigner distribution.​

Window (time window)

FFT processing is performed on a sampled numerical data series over a certain interval (for example, 1024 points or 2048 points), and this process of cutting out a part of the waveform is referred to as windowing (time windowing). The Fourier transform is defined by processing infinite-length data. This remains true for the Discrete Fourier Transform (DFT), where the FFT analyzer assumes that when the waveform is cut using a window, the waveform in that interval is repeated infinitely for signal analysis. At this time, if the analysis data length (the length of the window) is an integer multiple of the period of each frequency, the waveform assumed by the FFT analyzer will match the actual input waveform, resulting in a single line spectrum. However, if the analysis data length does not match an integer multiple of the period (which does not apply to frequency resolution, where the start and end do not connect), the processing will result in a distorted waveform, and the spectrum will spread out without concentrating power, leading to the occurrence of side lobes. This power leakage is referred to as leakage error. To prevent this leakage error, window processing is employed. By applying a bell-shaped function that makes both ends of the frame zero, the start and end can connect, reducing errors. Such a function is called a window function, and the process of synchronizing the analysis signal using the window function is referred to as window processing. As a result, the shape of the spectrum approaches that of a line spectrum.

The Hanning window is a typical example of a window, but other suitable windows are used depending on the analysis signal.

Aliasing (folding distortion)

According to the sampling theorem, it is necessary to sample at a rate greater than twice the highest frequency component of the signal. The frequency that is half of the sampling frequency is called the Nyquist frequency, and if components above the Nyquist frequency are included in the signal, aliasing occurs.

Energy Spectral Density

For impulse-like finite energy due to methods such as impact methods, this is normalized and displayed using energy. This is determined by further incorporating the time duration (window length, T = 1/Δf) into the power spectral density.

overalls

It is the total power up to the analysis frequency range (overall). The overall value is determined by the following.

1. Based on the peak amplitude value (Our models: CF-350/360*, CF-900 series*, CF-880* etc.)

  • Overalls No. 1

2. Based on effective values (Our models: CF-5000 series*, CF-3000 series*, DS-2000 series*, DS-3000 series*) (*: discontinued)

  • Overalls No. 2

Here
P DC: DC component
P i : the i-th power spectrum obtained (equivalent to squared value)
N : number of frequency lines
H f: window correction value.

Hanning → 2/3
Flat top → 0.316 (CF-350/360)
0.2724 (DS-2000/CF-3000 series)

Others → 1

* H f varies by model, so please refer to the manual for the correct information.

P DC and P i are power, so they will be (peak amplitude value) 2 and (effective value) 2. Overall, it will equal the root mean square of the input time signal.

Partial overall refers to the sum of power within a specified interval.

If the power spectrum values are in dB, please convert dB to squared values before applying the above formula.
P i =10 (dB value/10)

The overall value in dB is 10×log(O.A).

*In frequency calculus, there is an impact due to window correction applied to the overall. If there is no window correction (rectangular), there is no impact.

overlap processing

Real-time analysis can perform FFT analysis by overlapping the window when below the frequency.

For example, we perform FFT processing on data points of 1024, and at this time, we execute FFT analysis by overlapping the newly sampled data with the previous data. A larger overlap means that we can measure changes in signal time more precisely.

Orbit (Lissajous)

The figure formed by synthesizing two signals on orthogonal x and y axes is called an orbit or Lissajous figure, and it exhibits visual characteristics based on the combination of the amplitudes, frequency ratios, and phase differences of the two signals. Additionally, when the frequency ratio is an integer, the trajectory of the drawn figure returns to its original position at a constant period.

Octave Analysis

While the power spectrum divides the analysis frequency into fixed-width bands (fixed-width type) to represent the power for each band, frequency analyzers in the acoustic field often take the frequency axis on a logarithmic scale and perform frequency analysis by passing through band filters of constant relative width that are equally divided on the logarithmic scale. Bandwidths of 1 octave and 1/3 octave are common, and this type of analysis is referred to as octave analysis.

In general, on the frequency axis, when the lower cutoff frequency f 1 is taken, and the upper cutoff frequency f 2 is taken at twice the frequency, that is,Octave Analysis No. 1Then, the spacing between f 1 and f 2 is an octave.

  • Octave Analysis No. 2

is the center frequency of the octave.
1/3 octave is further divided into three parts of the octave, f 1 and f 2 isOctave Analysis_No.3times, that isOctave Analysis_No.4the center frequency is

  • Octave Analysis_No.5

will become.

The IEC 61260 (JIS C 1514) standard specifies the center frequencies of octave bands and filter characteristics, and both analog and digital octave analyzers are unified according to this.

acoustic intensity

SI (Sound Intensity) or AI (Acoustic Intensity)

Acoustic intensity is the sound energy passing through a unit cross-sectional area in a unit time at a point in the sound field, and the sound pressure at that point p(t ) and the particle velocity in the r direction.SI (Sound Intensity) or AI (Acoustic Intensity) No.1It is a vector quantity defined by the time average of a product.

  • SI (Sound Intensity) or AI (Acoustic Intensity) NO.2

On one hand, density. SI (Sound Intensity) or AI (Acoustic Intensity) No.3in a medium without flow, p (t ) andSI (Sound Intensity) or AI (Acoustic Intensity) No. 4The following equation holds between.

  • SI (Sound Intensity) or AI (Acoustic Intensity) _NO.5

However, since it is extremely difficult to directly measure particle velocity accurately, a method has been considered to approximately determine particle velocity from the difference in sound pressure between two closely spaced points. This is known as the SI measurement method using two microphones. That is, the sound pressure at two microphones separated by a distance of ⊿r in the r direction.SI (Sound Intensity) or AI (Acoustic Intensity) No.6usingSI (Sound Intensity) or AI (Acoustic Intensity) No. 7to be approximated as follows.

  • SI (Sound Intensity) or AI (Acoustic Intensity) NO.8
  • SI (Sound Intensity) or AI (Acoustic Intensity) NO.9

Substituting equation (4) into equation (2) gives the particle velocity in the r direction.SI (Sound Intensity) or AI (Acoustic Intensity) No.10is expressed in (5) form.

  • SI (Sound Intensity) or AI (Acoustic Intensity) NO.11

The acoustic intensity Ir will be as follows.

  • SI (Sound Intensity) or AI (Acoustic Intensity) NO.12

This formula directly calculates Ir in the time domain and is called the direct integration method.

Furthermore, any frequency band f 1? f 2 SI value in the r direction (Ir As a way to find ), 7 The formula ) is also frequently used.

  • SI (Sound Intensity) or AI (Acoustic Intensity) NO.13

Here, Im{G 12 (f)} represents the imaginary part of the (one-sided) cross-spectrum of p 1 (t) and p 2 (t). This is obtained by using a 2-channel FFT analyzer to calculate the cross-spectrum between sound pressure signals at two closely spaced points, and by performing the calculations for this imaginary part according to the above formula, Ir for any frequency band can be determined. This method is called the cross-spectrum method. As measurement errors in the SI measurement method, there are finite difference errors due to ⊿r being finite, as well as errors due to sensitivity and phase mismatches between the two microphone systems, but various studies have been conducted on this correction method.

Next, I will provide some examples of the application of the SI measurement method.

Measurement of the power level of the sound source.

Sound intensity represents the amount of sound energy passing through a unit area in a unit time, and the total power P radiated from the sound source is,

  • Measurement of the power level of the sound source.
    (I ri: Acoustic intensity perpendicular to the surface si, si: the i-th area)

Acoustic power is calculated from the measurement of acoustic intensity in the divided areas on a hemispherical surface centered around the sound source, in a direction orthogonal to the sphere.

(2) Sound Insulation Measurement

By measuring the transmission power for each section using the SI method, it is possible to quantitatively assess the sound insulation performance of walls composed of multiple sections and the degree of sound leakage from gaps, making it effective for on-site sound insulation measurements.

(3) Sound Field Analysis

The SI value, being a vector quantity, allows for the visualization and capture of the flow of sound energy by representing the direction and magnitude of sound propagation in two or three dimensions.