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Basic Signal Procesing for Vibration Data Colleciton

Basic Signal Procesing for Vibration Data Colleciton

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Published by: Mohd Asiren Mohd Sharif on May 30, 2010
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11/14/2012

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DoctorKnow
 ® 
Application Paper
 
Title:
 
Basic Signal Procesing for Vibration Data Colleciton
 
Source/Author:
 
Todd Reeves
 
Product:
 
General
 
Technology:
 
Vibration
 
Classification:
 
Not Classified
 
Basic Signal Processing for Vibration Data Collection
 
Todd Reeves
 
CSI
 
Knoxville, TN
 
Abstract
 Often when setting up measurement points for vibration data collection many choices selected such as themaximum frequency range, the lines of resolution, the window type and the integration mode are made based onrules of thumb instead of an understanding of how each of these are related to the frequency spectrum and thetime waveform. A digital signal analyzer is a powerful tool that can present some problems for the uneducateduser. With an understanding of signal processing basics these problems can be addressed, understood andavoided.
Fast Fourier Transform
 The method used to convert time domain information to frequency domain information is the Fast FourierTransform (FFT).Often a frequency spectrum is referred to as an FFT. However, the FFT is the mathematical conversion from thetime domain to the frequency domain. Since the signal that comes into the analyzer is an analog signal asdiscussed in the previous section, it must be digitally sampled by the analyzer. Therefore, the process used bydigital analyzers is actually a variation of the FFT, called the Discrete Fourier Transform (DFT).The DFT is similar as the analog time waveform is recreated in the analyzer by digitally sampling, and then thetransformation into the frequency domain is done. Part of the reason the FFT process works is the assumption thatthe signal measured and digitally sampled is one period of a periodic signal that extends to minus infinity and toplus infinity. Normally, this is true for most vibrating pieces of equipment.
 
 It is the digital sampling process that makes the signal processing more complicated. The information here unlocksthe mysteries of digital signal processing without getting bogged down in too much theory.In order to understand the FFT digital sampling process, you must understand the relationship between lines ofresolution (LOR), Maximum frequency (Fmax), length of time waveform (Tmax), the digital sample size, aliasing,windowing, filters, and unit conversion.
Resolution
 Once data has been converted to the frequency domain from the time domain, view all of the frequencies ofinterest in as much detail as possible. Resolution is the number of parts that the spectrum is broken into, usuallycalled lines of resolution (LOR). The number of lines of resolution selected are divided into the maximum analysisfrequency (Fmax) to arrive at the bandwidth (BW).
BW = F
max
/LOR
 
 
 The lines are actually the center frequencies of what are often called "bins of energy". Each bin actually containsan infinite number of frequencies, and all of the energy in the bin is summed and represented by a singleamplitude at the center frequency identified at each line of resolution.First, identify your frequencies of interest so that enough resolution is chosen to separate closely spacedfrequencies. Also, be aware that more lines of resolution affects the length of the time waveform. Increasedresolution can also decrease the actual amplitude of the vibration amplitudes due to the separation of energy intomore energy bins.
Time Record Length
 
 
Calculate the time record length of the time waveform, Tmax, from the following basic relationship.
Tmax
 
= 1/BW
 At face value this is a simple and often used equation. However, to understand the limitations of some analyzers, itis important to more fully investigate the relationship between the Fmax
 
the LOR, and the Tmax.
Tmax
 
=
 
sample size/sample rate
 
Sample size = 2.56 * Lines of Resolution
 
Sample rate = 2.56 * Fmax
 These terms have already been defined, but be aware that some analyzers have an upper limit on the samplesize. Usually this number will be 1024 or 2048. Therefore, a 400 line spectrum would require 2.56*400 = 1024samples and an 800 line spectrum would require 2.56* 800 = 2048 samples. However, if the analyzer is limited to1024 samples, then the 800 line spectrum will be created from 1024 samples since it is the upper limit of theanalyzer. This is important when discussing the Tmax in the time waveform, because, in general, raising the Fmaxdecreases Tmax and raising LOR increases Tmax until the point that the multiple of 2.56 * LOR reaches thesample limit in the analyzer. In this case, the sample size for anything greater than 400 lines is forced to be 1024.The table below demonstrates how this limitation affects the Tmax for a maximum 1024 sample size analyzer.
Tmax = sample size/sample rate
 Fmax *2.56 =sample rate LOR*2.56 = sample size Tmax
 
400 1024 100 256 0.25400 1024 200 512 0.50400 1024 400 1024 1.00

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VBLG added this note
The fact is the number of lines or bins are always the half of number of samples. The analyzer often show bins only 80% of calculated bin to avoid the possible occur aliasing of low pass filter's roll of. Example, 2048 samples give 1024 bins and selected to display 80% then display 800 lines
VBLG added this note
Information on page3 is not correct. "However, if the analyzer is limited to 1024 samples, then the 800 line spectrum will be created from 1024 samples since it is the upper limit of the analyzer."
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