|
| accelerometer | A sensor mounted on a structure to measure the acceleration at a particular location in one or multiple directions. |
| aliasing | A phenomenon whereby an analog signal of frequency greater than the Nyquist frequency appears after sampling at a frequency less than the Nyquist frequency. See also anti-aliasing filter. |
| anti-aliasing filter | Analog lowpass filters used before analog to digital conversion to filter out the frequencies greater than half the sampling frequency and prevent aliasing. |
|
| bandedge frequency | The upper and lower cutoff frequencies of an ideal bandpass filter. |
| bandpass filter | A filter with a single transmission band extending from a lower bandedge frequency greater than zero to a finite upper bandedge frequency. |
|
| calibrator | A controlled source generating a known level of excitation used to calibrate a sensor. |
| CCIF | International Telephone Consultative Committee. Also, a standard technique for measuring intermodulation distortion (IMD) of band-limited systems. The CCIF technique uses a test signal composed of two closely spaced tones with an amplitude ratio of 1:1. The CCIF technique is also known as ITU-R. |
| coherence | A measure of the degree of linear dependence between two signals as a function of frequency. |
| coherent output power spectrum | A measure of what part of the (output) power spectrum is fully coherent with the input signal. |
| colormap | A method of displaying 3D data on a 2D plot using color. |
| COP | Coherent output power |
| crest factor | The ratio of the peak value of a signal to its RMS value. For a sine wave, the crest factor is 1.414. For a square wave, the crest factor is 1. |
| critical sampling | Phenomenon that occurs in a Gabor transform when the window length equals the window shift step. In critical sampling, the number of Gabor coefficients cm, n equals the number of original data samples s[k]. |
| cross power spectrum | Measurement of two signals with an amplitude that is the product of the two signal amplitudes and a phase that is the difference of the two phases. |
| crosstalk | The ratio between the level of the unwanted signal on the affected channel to the level of the actual signal on the active channel. Crosstalk usually is expressed in dB. |
|
| DIN | Deutsch Institut fur Nurmung |
| distortion | The production of signal components not in the original signal due to non-linearities in the system or transmission path. |
| DSA | Dynamic signal acquisition |
| dual functions | Pair of window functions for Gabor transform and Gabor expansion. The dual functions are interchangeable. That is, you can use either of the dual functions for a Gabor transform while using the other dual function for a Gabor expansion. |
| dynamic range | The ratio of the highest signal level a circuit can handle to the smallest signal level it can handle, usually equal to the noise level, normally expressed in dB.
|
| DZT | Discrete Zak Transform |
|
| ENOB | Equivalent number of bits. ENOB is equal to the base 2 logarithm of a ratio. |
| equal confidence | Special exponential averaging mode used for fractional-octave analysis. For equal confidence, the time constant for each band is set individually so that the relative confidence in the measurement is equal across all the bands. There is a 68% probability that the results will be within +/- the specified confidence level of the true mean value. |
| equivalent continuous level (Leq) | The energy average level of a signal over a given time interval. |
| exponential averaging | Time-averaging technique that gives recent data more importance than older data. |
|
| fast | Exponential averaging using a time constant of 125 ms. |
| FFT block size | The number of samples used to compute an FFT. |
| FFT lines | The number of FFT lines is related to the FFT block size. Theoretically the number of lines is half of the block size. However, in practice, it is reduced to 80% of the theoretical value due to the anti-aliasing filter. For example, a 400-line FFT is based on a block size of 1,024 points. |
| filter bank | A group of filters. |
| fractional-octave | The interval between two frequencies, one of which is a fractional power of the other. |
| frequency range | The frequency range used to specify the measurement analysis bandwidth. |
| frequency response function | Represents the ratio of output-to-input in the frequency domain and fully characterizes linear, time-invariant systems. |
| FRF | Frequency response function. See also frequency response function. |
| fundamental component | Portion of a signal whose frequency is at the fundamental frequency. |
|
| g | Unit for measuring acceleration. One g = 9.81 m/s2, the acceleration due to gravity at the surface of Earth. |
| Gabor coefficient | The result of Gabor transform. |
| Gabor expansion | The inverse Gabor transform used on Gabor coefficients to recover a time domain input signal. |
| Gabor transform | One of the invertible joint time-frequency transforms. |
| group delay | The relative time delay between different spectral portions of a signal. The group delay is the slope of the plot of phase versus frequency. |
| group delay distortion | Any deviation from a constant time delay. |
|
| H1 | The frequency response function computed as the ratio of the cross spectrum to the input autospectrum: Gxy/Gxx. This technique gives the best performance in the presence of noise for measuring anti-resonances, where the signal to noise ratio tends to be poor. For measurement of resonances, the frequency response function H2 gives a better estimate. In a noise-free environment, both techniques give the same result. Since both measurements are based on the same data set, the choice of technique can be made after the data acquisition is complete. |
| H2 | The frequency response function computed as the ratio of the output autospectrum to the backwards cross spectrum: Gyy/Gyx. This technique gives the best performance in the presence of noise for measuring resonances, where the signal-to-noise ratio tends to be best. For measurement of anti-resonances, the frequency response function H1 gives a better estimate. In a noise-free environment, both techniques give the same result. Since both measurements are based on the same data set, the choice of technique can be made after the data acquisition is complete. |
| H3 | The frequency response function computed as an average of H1 and H2. |
| harmonic | Frequencies that are integer or fractional multiples of a fundamental frequency. |
|
| IEC | International Electrotechnical Commission |
| IEPE | Integrated Electronics Piezoelectric. A type of transducer that operates using a constant current source as the conditioning medium and returns a signal in the form of voltage modulation on the same line as the current source. |
| IMD | Intermodulation distortion. |
| impulse | Exponential averaging using a time constant of 35 ms if the signal is rising and 1,500 ms if the signal is falling. |
| input limits | The upper and lower voltage inputs for a channel. You must use a pair of numbers to express the input limits. |
| intermodulation distortion | The ratio of the RMS sum of all the sum and difference intermodulation distortion products to the RMS level of the high-frequency carrier. Intermodulation distortion is a type of distortion that quantifies the linearity of a device by measuring the intermodulation of two component tones. Intermodulation distortion often is used because you can arrange the test so that many distortion products fall within the passband of a band-limited device. |
| ISO | International Standards Organization |
| ITU-R | See CCIF. |
|
| Leq | See equivalent continuous level (Leq). |
| linear averaging | Time-averaging technique that weights all data in the average equally. |
| LMSE | Least mean square error |
|
| microphone | Sensor used to convert sound pressure variations into an electrical signal, usually when the acoustic medium is air. |
| midband frequency | The center frequency of a bandpass filter, defined as the geometric mean of the bandedge frequencies. |
| MLS | Maximum Length Sequence. A pseudo-random sequence of ones and zeroes you can use as a test signal to perform delay and frequency response measurements. |
| ms | Millisecond |
|
| noise | Any unwanted signal. Noise can be generated by internal sources, such as semiconductors, resistors, and capacitors, or from external sources, such as the AC power line, motors, generators, thunderstorms, and radio transmitters. |
| nominal frequency | Rounded midband frequency for the designation of a particular fractional-octave filter. This term is used by the IEC standards. Nominal frequencies are identical to the preferred frequencies defined in the ANSI standards. |
| nonstationary signal | A signal whose frequency content changes within a captured frame. |
|
| octave | The interval between two frequencies, one of which is twice the other. For example, frequencies of 250 Hz and 500 Hz are one octave apart, as are frequencies of 1 kHz and 2 kHz. |
| order analysis | The analysis of harmonics related to rotational speed. The application of harmonic analysis to rotating machinery. |
| order curves | The high power density curves that indicate order components in a spectral map. |
| overlapping | A method that uses a portion of the previous data block to compute the FFT of the current data block. |
|
| Pa | Pascal. International unit of pressure. |
| peak hold | Peak detection process retaining the maximum value of a signal. |
| phase lag | The delay between two tones of the same frequency measured in angular units of degrees or radians. |
| phase linearity | The maximum deviation from a straight line of a plot of the phase versus frequency in degrees. Constant slope of the plot of phase versus frequency indicates a delay independent of frequency, or constant delay. |
| phon | The unit of loudness on a scale corresponding to the decibel scale of sound pressure level with the number of phons of a given sound being equal to the decibels of a pure 1 kHz sine tone judged by the average listener to be equal in loudness to the given sound. |
| pink noise | A random noise signal that has been filtered so its spectral density is shaped to drop by 3 dB per octave, thus giving it a constant spectral density per octave or third-octave band. |
| pistonphone | Microphone calibrator generating a known sound pressure level, typically at a certain reference frequency. |
| preferred frequency | Rounded midband frequency for the designation of a particular fractional-octave filter. The term preferred frequency is used by the ANSI standards. Preferred frequencies are identical to the nominal frequencies defined in the IEC standards. |
| pregain | Any gain applied to a signal by an external device, such as an amplifier, a preamplifier, signal conditioning, and so on, before the data acquisition board. |
| propagation delay | The delay caused by the measurement circuitry in units of samples. Propagation delay is independent of the sampling frequency. In DSA devices, propagation delay is caused by the anti-imaging and anti-aliasing filters on the output and input, respectively. |
| PSD | Power spectral density |
|
| reference sound pressure | A reference pressure of 20 µPa. This reference pressure was chosen conventionally to correspond to the quietest sound at 1,000 Hz that the human ear can detect. |
| resampling | A method to sample a time sequence at a different time interval. |
| RMS averaging | Averaging technique used to average the energy, or power, of a signal. RMS averaging reduces signal fluctuations but not the noise floor. RMS quantities of single-channel measurements have zero phase. RMS averaging for dual-channel measurements is defined in such a way to preserve important phase information. |
| rpm | Revolutions per minute. |
|
| S/s | Samples per second. Used to express the rate at which a DAQ device samples an analog signal. |
| sampling frequency | The rate at which a continuous waveform is digitized. |
| SFDR | Spurious-free dynamic range. |
| Shannon Sampling Theorem | Theorem stating that to sample a signal properly, the signal must not contain frequencies above the Nyquist frequency. |
| signal in noise and distortion | The ratio of the RMS level of the input signal to the rms level of the noise plus harmonics. |
| SINAD | Signal in noise and distortion. |
| slow | Exponential averaging using a time constant of 1,000 ms. |
| SMPTE | Society of Motion Picture and Television Engineers. Also, a standard technique for measuring intermodulation distortion (IMD). The SMPTE technique uses a test signal composed of a low-frequency tone and a high-frequency tone with an amplitude ratio of 4:1. |
| spectral leakage | A phenomenon whereby the measured spectral energy appears to leak from one frequency into other frequencies. Spectral leakage occurs when a sampled waveform does not contain an integral number of cycles over the time period during which it was sampled. The technique used to reduce spectral leakage is to multiply the time-domain waveform by a window function. See also windowing. |
| spurious-free dynamic range | The ratio of the full-scale amplitude of a device under test to the amplitude of the largest spurious signal. |
| SRS | Shock response spectrum |
| STFT | Short-Time Fourier Transform |
|
| tachometer | Device used to measure the rotational speed of a rotating part. |
| THD | Total harmonic distortion. |
| THD+N | Total harmonic distortion plus noise. |
| third-octave | Ratio between two frequencies, equal to 21/3. |
| time constant | A standardized time constant used in exponential time weighting for acoustical analysis. The standard time constants in the NI Sound and Vibration Measurement Suite are Slow = 1000 ms, Fast = 125 ms, and Impulse = 35 ms while the signal level is increasing or 1,500 ms while the signal level is decreasing. See also fast, impulse, and slow. |
| total harmonic distortion | The RMS sum of all the harmonics relative to the amplitude of the fundamental signal. |
| total harmonic distortion plus noise | The ratio of the RMS level of the noise plus harmonics to the RMS level of the input signal. |
| transient | A very short-duration signal. A transient normally occurs only once, or very infrequently. |
|
| vector averaging | Computes the average of complex quantities directly. In other words, the real and imaginary parts are averaged separately. Vector averaging eliminates noise from synchronous signals and usually requires a trigger. |
|
| waterfall | A 3D plot displaying the amplitude of spectral components as a function of both time and frequency. The frequency spectrum is displayed as a curve for each specified time instant. Several such curves, for different time instants, are displayed simultaneously. |
| weighting filter | A filter used to reproduce the varying sensitivity of the human ear to sound at different frequencies. Originally, A-weighting was intended to represent the varying sensitivity of the ear to sound pressure levels ranging between 40 and 60 dB ref 20E-6 Pa. Subsequently, B-weighting and C-weighting were developed to represent the varying sensitivity of the ear over higher sound pressure level ranges. |
| white noise | Noise that has the same power spectral density at all frequencies. For example, the average power of white noise in a 100 Hz bandwidth between 300 Hz and 400 Hz is the same as the average power of white noise in the 100 Hz bandwidth between 10,000 Hz and 10,100 Hz. |
| window function | A smooth waveform that generally has zero value at the edges. See also windowing. |
| windowing | Technique used to reduce spectral leakage by multiplying the time-domain waveform by a window function. The process of windowing reduces the amplitudes of discontinuities at the edges of a waveform, thereby reducing spectral leakage. See also spectral leakage. |
|
| Z-weighting | Zero-weighting. Weighting with unity gain at all frequencies; equivalent to linear or no weighting. |