lecroy 93XXC-OM-E21 电路图.pdf
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1、 Xn is the resulting complex frequency-domain waveform; We j N = 2/ ; and N is the number of points in xk and Xn. The generalized FFT algorithm, as implemented here, works on N, which need not be a power of 2. 4. The resulting complex vector Xn is divided by the coherent gain of the window function,
2、 in order to compensate for the loss of the signal energy due to windowing. This compensation provides accurate amplitude values for isolated spectrum peaks. 5. The real part of Xn is symmetric around the Nyquist frequency, i.e. Rn = RN-n , while the imaginary part is asymmetric, i.e. In = IN-n . th
3、e energy at frequency 0 is completely contained in the 0 term. The first half of the spectrum (Re, Im), from 0 to the Nyquist frequency is kept for further processing and doubled in amplitude: Rn = 2 ? Rn0 n N/2 In = 2 ? In0 n Mmin angle = 0Mn Mmin . &? $SSHQGL?& Where Mmin is the minimum magnitude,
4、 fixed at about 0.001 of the full scale at any gain setting, below which the angle is not well defined. The dBm Power Spectrum: dBm PS M M M M n ref n ref = = 1020 10 2 2 10 loglog where Mref = 0.316 V (that is, 0 dBm is defined as a sine wave of 0.316 V peak or 0.224 V RMS, giving 1.0 mW into 50).
5、The dBm Power Spectrum is the same as dBm Magnitude, as suggested in the above formula. dBm Power Density: ()dBmPDdBmPSENBWf=10 10 log where ENBW is the equivalent noise bandwidth of the filter corresponding to the selected window, and f is the current frequency resolution (bin width). 7. The FFT Po
6、wer Average takes the complex frequency- domain data Rn and In for each spectrum generated in Step 5, and computes the square of the magnitude: Mn2 = Rn2 + In2, then sums Mn2 and counts the accumulated spectra. The total is normalized by the number of spectra and converted to the selected result typ
7、e using the same formulae as are used for the Fourier Transform. &? ?)7?*ORVVDU *ORVVDU Defines the terms frequently used in FFT spectrum analysis and relates them to the oscilloscope. $OLDVLQJIf the input signal to a sampling acquisition system contains components whose frequency is greater than th
8、e Nyquist frequency (half the sampling frequency), there will be less than two samples per signal period. The result is that the contribution of these components to the sampled waveform is indistinguishable from that of components below the Nyquist frequency. This is aliasing. The timebase and trans
9、form-size should be selected so that the resulting Nyquist frequency is higher than the highest significant component in the time-domain record. &RKHUHQW?*DLQThe normalized coherent gain of a filter corresponding to each window function is 1.0 (0 dB) for a rectangular window and less than 1.0 for ot
10、her windows. It defines the loss of signal energy due to the multiplication by the window function. This loss is compensated in the oscilloscope. This table lists the values for the implemented windows. :LQGRZ?)UHTXHQF?RPDLQ?3DUDPHWHUV Window Type Highest Side Lobe (dB) Scallop Loss (dB) ENBW (bins)
11、 Coherent Gain (dB) Rectangular133.921.0 0.0 von Hann321.421.5 6.02 Hamming431.78 1.37 5.35 Flat Top440.01 2.9611.05 BlackmanHarris671.13 1.71 7.53 &? $SSHQGL?& (1%:?Equivalent Noise BandWidth (ENBW) is the bandwidth of a rectangular filter (same gain at the center frequency), equivalent to a filter
12、 associated with each frequency bin, which would collect the same power from a white noise signal. In the table on the previous page, the ENBW is listed for each window function implemented and is given in bins. )LOWHUVComputing an N-point FFT is equivalent to passing the time- domain input signal t
13、hrough N/2 filters and plotting their outputs against the frequency. The spacing of filters is f = 1/T while the bandwidth depends on the window function used (see Frequency bins). )UHTXHQF?ELQVThe FFT algorithm takes a discrete source waveform, defined over N points, and computes N complex Fourier
14、coefficients, which are interpreted as harmonic components of the input signal. For a real source waveform (imaginary part equals 0), there are only N/2 independent harmonic components. An FFT corresponds to analyzing the input signal with a bank of N/2 filters, all having the same shape and width,
15、and centered at N/2 discrete frequencies. Each filter collects the signal energy that falls into the immediate neighborhood of its center frequency, and thus it can be said that there are N/2 “frequency bins”. The distance in hertz between the center frequencies of two neighboring bins is always: f
16、= 1/T, where T is the duration of the time-domain record in seconds. The width of the main lobe of the filter centered at each bin depends on the window function used. The rectangular window has a nominal width at 1.0 bin. Other windows have wider main lobes (see table). )UHTXHQF?5DQJHThe range of f
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