Fourier Analysis. Chapter Introduction Distortion Harmonic Distortion

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Chapter 5 Fourier Analysis 5.1 Introduction The theory, practice, and application of Fourier analysis are presented in the three major sections of this chapter. The theory includes a discussion of Fourier transforms, including the Fourier integral and the discrete Fourier transform, along with suggestions on computing the spectrum of quasiperiodic waveforms. The section on practice presents details on error mechanisms as well as on how Fourier analysis is implemented in Spice, Spectre, and external Fourier analyzers. Suggestions are given on how to achieve accurate results. The chapter concludes by presenting various types of circuits that present challenges to Fourier analysis and how those challenges might be addressed. 5.1.1 Distortion Fourier analysis is almost exclusively used to measure distortion. When measuring distortion, there are two important cases to consider: wide-band and narrow-band circuits. 5.1.1.1 Harmonic Distortion To measure the distortion of wide-band circuits, one simply applies a pure single-tone sine wave to the input, and determines how much 251

252 Chapter 5. Fourier Analysis the output deviates from being a pure sinusoid after the circuit has reached steady state. The distortion products in the output signal fall at frequencies that are integer multiples (harmonics) of the input frequency. In this case, the distortion measure of choice is total harmonic distortion or THD. Total Harmonic Distortion THD is a measure of the amount of energy in the harmonics relative to the energy in the fundamental. k=2 X k THD = 2 X 1 2 100% (5.1) where X k is the magnitude of the k th harmonic. 5.1.1.2 Intermodulation Distortion If this same technique were applied to measure the distortion of a narrow-band amplifier, the harmonic distortion products would normally fall outside the bandwidth of the amplifier and so would be attenuated. The amount of distortion calculated using (5.1) would be much too low. Distortion is measured in narrow-band amplifiers by applying two pure sinusoids with frequencies well within the bandwidth of the amplifier (call these frequencies f 1 and f 2 ). The harmonics of these two frequencies are outside the bandwidth of the amplifier, however there are distortion products that fall at the frequencies 2f 1 f 2, 2f 2 f 1,3f 1 2f 2,3f 2 2f 1, etc. As shown in Figure 5.1 on the next page these frequencies should also be well within the bandwidth of the amplifier and so can be used to measure accurately what is referred to as the intermodulation distortion, or IMD, produced by the amplifier. 5.1.1.3 Compression and Intercept Points Harmonic and intermodulation distortion both characterize signals rather than the underlying circuit. At low frequencies, it is common

5.1. Introduction 253 3f1 2f2 2f1 f2 f1 f2 2f2 f1 3f2 2f1 Figure 5.1: Two-tone intermodulation products superimposed on a narrow-band filter s passband. to describe the distortion of a circuit by indicating the distortion in the output signal when driven by a sinusoid to achieve a certain output level. For example, an audio power amplifier might be described as having 0.1% THD at 100 Watts. At high frequencies it is more common to characterize the distortion produced by a circuit in terms of a compression point or an intercept point. These metrics characterize the circuit rather than the signal, and as such it is not necessary to specify the signal level at which the circuit was characterized. In order to more easily understand the definition of a compression point and an intercept point, consider the output power of the first three harmonics for a GaAs traveling-wave amplifier, as shown in Figure 5.2 on the following page. The 1 db compression point is the value of the output power when the gain of the amplifier has dropped 1 db from it small-signal asymptotic value, as shown in Figure 5.3 on page 255. The compression point is a difficult measurement to make with a time-domain circuit simulator (Spice) because one must measure the

254 Chapter 5. Fourier Analysis output 30 dbm 20 dbm 10 dbm 0 dbm -10 dbm -20 dbm -30 dbm -40 dbm -50 dbm -60 dbm -70 dbm -80 dbm -90 dbm -100 dbm Output Versus Input Power for Traveling-Wave Amplifier -20 dbm -10 dbm 0 dbm 10 dbm 20 dbm fundamental second third input Figure 5.2: The power in the first three harmonics at the output of the traveling-wave amplifier versus the input power. power in the fundamental of the output (using transient and Fourier analysis) while sweeping the input power. This generally requires many simulation runs. The actual compression point is found by drawing a figure similar to Figure 5.3 and determining the proper value graphically. The intercept points are defined in terms of the power levels of the various harmonics as extrapolated from their asymptotic small-signal behavior. When the input signal is small, a doubling of the input power doubles the output power at the fundamental, multiplies the output power of the second harmonic by 4 (2 2 ), and multiplies the output power of the third harmonic by 8 (2 3 ). Thus, the asymptotic slope of the fundamental is 1 db/db, the asymptotic slope of the second harmonic is 2 db/db, and the asymptotic slope of the third harmonic is 3 db/db. The second-order intercept point (IP 2 )

5.1. Introduction 255 1 db Compression Point for Traveling Wave Amplifier 30 dbm Small Signal Fundamental Output Power 20 dbm 10 dbm 1 db Compression 0 dbm Small Signal 1 db 10 dbm 20 dbm 10 dbm 0 dbm 10 dbm 20 dbm Input Power Figure 5.3: The 1 db compression point is the value of the fundamental output power where it crosses the line that represents the output power extrapolated from small signal conditions minus 1 db. In this example, the compression point is 22.3 dbm. is where the asymptotes for the second harmonic and the fundamental cross. Similarly, the third-order intercept point (IP 3 )iswhere the asymptotes for the third harmonic and the fundamental cross. The definitions of IP 2 and IP 3 are illustrated in Figure 5.4 on the following page. The intercept point is easier to measure with a time-domain circuit simulator than is the compression point because it requires just one transient and one Fourier analysis. Simply choose an input power level that you are confident puts the circuit in the small-signal regime and then measure level of the first, second, and third har-

256 Chapter 5. Fourier Analysis Intercept Points for Traveling Wave Amplifier 50 dbm IP3 Output Power 0 dbm 50 dbm Fundamental 2nd IP2 3rd 100 dbm 20 dbm 10 dbm 0 dbm 10 dbm 20 dbm 30 dbm 40 dbm Input Power Figure 5.4: The second-order intercept point (IP 2 ) is the output power where the second-order term as extrapolated from small-signal conditions crosses the extrapolated power of the fundamental. Similarly, the third-order intercept point (IP 3 ) is the output power where the third-order term as extrapolated from small-signal conditions crosses the extrapolated power of the fundamental. In this example, IP 2 is 47.6 dbm and IP 3 is 39.4 dbm.

5.2. Fourier Analysis Theory 257 monics. Let P i be the power of the input test signal and P 1, P 2, and P 3 be the power in the first, second, and third harmonics at the output. Assume that the power is measured in your favorite flavor of decibels (dbm, dbw, dbv, etc). Then, IP 2 =2P 1 P 2 (5.2) IP 3 =(3P 1 P 3 )/2 (5.3) These calculations are only accurate if P i is small enough so that the output power levels are in the asymptotic region for all harmonics, and if P i is large enough so that the Fourier analysis accurately computes P 1, P 2,andP 3. In the intercept point definitions just given, harmonic distortion was used for illustrative purposes only. The same definitions are just as easily be applied to intermodulation distortion. In this case, the second-order distortion terms play the role of the second harmonic, and the third-order distortion terms play the role of the third harmonic. 5.2 Fourier Analysis Theory It is a fundamental tenet in the theory of communications systems that waveforms can be decomposed into a possibly infinite sum of sinusoids or complex exponentials. In a circuit simulator, Fourier analysis is what is used to perform the decomposition. Fourier analysis takes a waveform, which is either a continuous or discrete real function of time, and transforms it into a spectrum, which is either a continuous or discrete complex function of frequency. The value of the spectrum at frequency f is the coefficient of the complex exponential of the same frequency. 5.2.1 The Fourier Series 5.2.1.1 Periodic Waveforms Periodic waveforms are generated by circuits in steady-state undergoing autonomous oscillation or responding to periodic stimuli (assuming that all stimulus have the same period T ). The frequency of

258 Chapter 5. Fourier Analysis the response is usually equal to the frequency of the stimulus, though occasionally the frequencies of the two are multiples of some common frequency. If a waveform is T -periodic and continuous, then the waveform can be written as a Fourier series. [ ( ) ( )] 2πkt 2πkt x(t) = a k cos + b k sin T T k=0 (5.4) Awaveformx(t) ist-periodicifx(t) = x(t T ) for all t. The sequence of Fourier coefficients (a k and b k ) is called the spectrum of x. Periodic waveforms consist of a sum sinusoids at integer multiples of the fundamental frequency f = 1/T. The signals at frequencies above the fundamental are called harmonics. Thus, the k th harmonic is the signal at kf. Given a T-periodic waveform x(t), you compute the Fourier coefficients by evaluating the following Fourier integrals. a k = 2 T b k = 2 T a 0 = 1 T t+t t t+t t t+t t x(τ)dτ (5.5) ( 2πkτ x(τ)cos T ( 2πkτ x(τ)sin T ) dτ (5.6) ) dτ (5.7) The Fourier Integral as a Bank of Receivers In order to understand how the Fourier integral works, it is useful to view it as a bank of receivers, one for each coefficient that is to be computed, as shown in Figure 5.5 on the next page. Each receiver translates the frequencies of the input signal so that the frequency of interest is moved to DC, and then filters the results in order to eliminate the signals at other frequencies. A comb filter is used that has zeros at the frequency of the fundamental and every harmonic. The comb filter completely eliminates all frequencies that exist in the signal assuming that it is precisely T -periodic except the one at DC. For