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Glossary of Basic Terms Related to FFT Analysis

Isolation

Isolation refers to the process of isolating electrical signals, and it is necessary for measurements when the common potential of the signal source differs from the ground potential. When the difference between the signal common potential and the ground potential (common-mode voltage) is small, measurement is possible even with differential input, but when the potential difference is 100 V or more, it results in an overload and measurement is not possible. In this case, isolation amplifiers may be used, but in the case of FFT analyzers, the analog input section from the amplifier to the anti-aliasing filter and AD converter is isolated from the chassis ground for each channel using a photocoupler. As a result, the digital circuit and the analog circuit are isolated, which is advantageous when it is necessary to eliminate ground loops or connections to the common potential of the signal source.

Impulse response

The response h(t) of a linear system when a unit impulse δ(t) is applied is called the impulseless response. The impulse response expresses the characteristics of a system in the time domain, while the frequency response function expresses it in the frequency domain.

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

Our FFT analyzer obtains the impulse response by performing an inverse Fourier transform on the frequency response function.

Phase spectrum

The phase representation as a function of frequency is primarily:

(1) Phase spectrum of one channel

(2) Phase difference between the two channels

There are two types.

(1) Phase spectrum of one channel

Time function(1) Phase spectrum of channel 1_No.1The Fourier transform of is:

  • (1) Phase spectrum of channel 1_No.2

(1)(1) Phase spectrum of channel 1_NO.3In this instrument, this is specifically called the (complex) Fourier spectrum.(1) Phase spectrum of channel 1_NO.4Since it is a complex function, it can be expressed as amplitude and phase, given its real and imaginary parts.

  • (1) Phase spectrum of channel 1_NO.5

twist

  • (1) Phase spectrum of channel 1_NO.6
  • (1) Phase spectrum of channel 1_NO.7

(2) is the amplitude of the Fourier spectrum (in MAG notation), and (3) is specifically called the phase spectrum in this instrument.

In this instrument, the phase is defined with the time origin at the start of the data frame, and the phase of the cosine wave is set to 0 degrees. (2) X (f) Even with the same signal, the phase spectrum If θ (f) is different, the waveform of the time signal x (t) will change significantly.

When measuring phase spectra, a trigger function is typically used to observe the phase lead or lag relative to a specific position. Its primary application is in field balancing of rotating bodies.

(2) Phase difference between the two channels

The phase difference between two channels is obtained as the phase representation of the transfer function (or cross-spectrum), which is a complex function. If the transfer function (frequency response function) of the system is H(f), then:

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

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

Wigner distribution

The Wigner distribution, proposed by E. Wigner in the field of quantum mechanics, possesses properties such as an extended power spectrum for non-stationary signals. In conventional FFT, time resolution and frequency resolution are complementary (increasing frequency resolution increases the sample time), making it difficult to obtain the instantaneous spectrum of a non-stationary signal with good resolution. In contrast, the Wigner distribution is not directly subject to the complementary constraints of time resolution and frequency resolution, thus enabling the acquisition of good time-frequency resolution for the power spectrum on the frequency-time plane. However, it was not practical due to the significantly larger number of calculation points compared to FFT.Oscope Time Series Data Analysis Tool (Software) Option OS-0263 Time-Frequency Analysis SoftwareBy using this, you can analyze the Wigner distribution.

Window (time window)

FFT processing is performed on a certain interval (for example, 1024 or 2048 points) of a sampled numerical data sequence. This act of cutting out a portion of the waveform is called windowing (cutting the waveform with a time window), or applying a window. The Fourier transform is defined as processing infinitely long data. This remains true for the Discrete Fourier Transform (DFT), and in an FFT analyzer, signal analysis is performed under the assumption that when the waveform is cut with a window, the waveform in that interval is repeated infinitely. In this case, if the analysis data length (window length) is an integer multiple of the period of each frequency, the waveform assumed by the FFT analyzer matches the actual input waveform, and a single line spectrum is obtained. However, if the analysis data length does not match an integer multiple of the period (i.e., it does not meet the frequency resolution, and the start and end are not connected), a distorted waveform is processed, and the spectrum does not concentrate power, resulting in spread to the left and right (side ropes). This power leakage is called leakage error. Windowing processing prevents this leakage error. By multiplying the frame by a bell-shaped function such that both ends of the frame are zero, the start and end points connect, reducing errors. Such a function is called a window function, and the process of synchronizing the analysis signal using a window function is called windowing. As a result, the shape of the spectrum approaches that of a line spectrum.

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

Aliasing (folding distortion)

The sampling theorem states that it is necessary to sample a signal at a rate at least twice the rate of the highest frequency component. The frequency that is half the sampling frequency is called the Nyquist frequency, and if the signal contains components above the Nyquist frequency, aliasing (aliasing distortion) occurs.

Energy spectral density

For impulsive, finite energy signals such as those generated by striking techniques, this is normalized and displayed using energy. This is calculated by multiplying the power spectral density by the acquisition time (window length, T = 1/Δf).

overalls

This is the total power (overall) up to the analysis frequency range. The overall value is calculated as follows:

1. When using the single amplitude value (peak value) as the reference (Our models: CF-350/360*, CF-900 series*, CF-880*, etc.)

  • Overalls_No.1

2. When using RMS values as the basis (Our models: CF-5000 series*, CF-3000 series*, DS-2000 series*, DS-3000 series*) (*: Discontinued)

  • Overalls_No.2

Here
P DC: DC component
Pi : The obtained i-th power spectrum (equivalent to the 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)

Other → 1

* HF values vary depending on the model; please refer to the instruction manual for accurate information.

PDC, where Pi is the power, has an amplitude of 2 and an RMS value of 2. The overall value is equal to the mean square of the input time signal.

Partial overall is the sum of power within a specified interval.

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

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

*When performing frequency calculus, the overall result is affected by window correction. There is no effect when there is no window correction (rectangular).

Overlap processing

If the real-time analysis frequency is below the specified frequency, you can overlap the windows and perform the FFT analysis.

For example, when performing an FFT analysis on data at intervals of 1024 points, the newly sampled data is overlaid with the previously sampled data during the FFT analysis. A larger overlap allows for more detailed measurement of changes in signal time.

Orbit (Lissajous)

The figure formed by combining two signals on orthogonal x and y axes is called an orbit or Lissajous curve, and its visual characteristics are determined by the combination of the amplitude, frequency ratio, and phase difference of the two signals. Furthermore, when the frequency ratio is an integer, the trajectory of the drawn figure returns to its original position with a constant period.

Octave Analysis

While power spectra divide the analysis frequency into constant-width segments (constant-width type) to represent the power of each band, frequency analyzers in the field of acoustics often use a logarithmic scale for the frequency axis and perform frequency analysis by passing the signal through a bandpass filter with a constant-ratio width that divides the logarithmic scale equally. Bandwidths of one octave and one-third of an octave are common, and this type of analysis is called octave analysis.

Generally, the lower limit of the cutoff frequency is on the frequency axis. For f1, the upper limit cutoff frequency f2 When we take twice the frequency, that isOctave Analysis_No.1Therefore, the interval between f1 and f2 is an octave.

  • Octave Analysis_No.2

This is the center frequency of the octave.
A 1/3 octave is an octave that has been further divided into three parts. f 1 In contrast, f 2 isOctave Analysis_No.3double, that isOctave Analysis_No.4The center frequency is

  • Octave Analysis_No.5

It will be.

The IEC 61260 (JIS C 1514) standard defines the center frequency and filter characteristics of octave bands, and analog and digital octave analyzers are standardized to this standard.

Acoustic intensity

SI (Sound intensity) or AI (Acoustic intensity)

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

  • SI (Sound intensity) or AI (Acoustic intensity) - No. 2

On the other hand, density SI (Sound intensity) or AI (Acoustic intensity) _ No.3In a fluid medium, p(t) andSI (Sound intensity) or AI (Acoustic intensity) - No. 4The following equation holds between [the two points].

  • SI (Sound intensity) or AI (Acoustic intensity) - No. 5

However, since it is extremely difficult to accurately measure particle velocity directly, a method has been considered to approximate particle velocity from the difference in sound pressure between two adjacent points, and this is the SI measurement method using two microphones. That is, the sound pressure of two microphones separated by a distance Δr in the r directionSI (Sound intensity) or AI (Acoustic intensity) _ No. 6UsingSI (Sound intensity) or AI (Acoustic intensity) _ No.7We can find an approximate value of it 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), we get the particle velocity in the r direction.SI (Sound intensity) or AI (Acoustic intensity) - No. 10This is expressed by equation (5).

  • SI (Sound intensity) or AI (Acoustic intensity) - No. 11

From here on, acoustic intensity Ir It 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, equation (7) is often used as a method to determine the SI value (Ir) in the r direction in an arbitrary frequency band f1 to f2.

  • 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 can be done by using a 2-channel FFT analyzer to obtain the cross-spectrum between sound pressure signals at two adjacent points, and then performing the above calculation on the imaginary part of this cross-spectrum to determine Ir for any frequency band. This method is called the cross-spectrum method. Measurement errors in the SI measurement method include finite difference errors due to the finite Δr and errors due to sensitivity and phase mismatch between two microphone systems, and various correction methods have been studied.

Next, I will give some examples of applications of the SI measurement method.

(1) Measurement of the power level of the sound source

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

  • (1) Measurement of the power level of the sound source
    (Iri: Acoustic intensity perpendicular to surface si, si: Ith area)

It is given by [formula]. From this, the acoustic power is calculated from the measurement of acoustic intensity in areas divided in a hemispherical surface centered on the sound source, in directions perpendicular to the sphere.

(2) Sound insulation measurement

By measuring the transmission power for each part using the SI method, the sound insulation performance of a wall consisting of multiple parts and the degree of sound leakage through gaps can be quantitatively measured, making it effective for on-site sound insulation measurements.

(3) Sound field analysis

Since SI values are vector quantities, the energy flow of sound can be visualized and understood by representing the direction and magnitude of sound propagation in two or three dimensions.