Lecture 16: Introduction to Linear time-invariant filters; response of FIR filters to sinusoidal and exponential inputs: frequency response and system.

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Lecture 16: Introduction to Linear time-invariant filters; response of FIR filters to sinusoidal and exponential inputs: frequency response and system function Sections 4.2.2, 4.3, 4.4.1, Sections 2.2.3, 2.3

In developing the DFT, we considered discrete-time signals which are either vectors (i.e., consisted of finitely many samples) or periodic extensions thereof. We now turn our attention to general discrete-time signals, namely sequences such as ▫x = x[ · ]= {x[n],n ∈ Z} If x is a linear combination of (not necessarily periodic) sinusoids, then it also has a spectrum. Its spectrum consists of the coe ffi cients of these sinusoids given (or plotted) as a function of frequency.

Linear filters can be used to alter the spectra of sequences in an immediate, “real time”, fashion. A linear filter H acts as a (linear) transformation of an input sequence x to an output sequence y. (Since the input and output spaces are infinite- dimensional, this linear transformation is not represented by a finite-dimensional matrix.) y = H(x)

Analytically, the simplest discrete-time filter is the so- called finite impulse response (FIR) filter. At each sampling instant n, a new input sample is read in and stored in a bu ff er containing the M + 1 most recent input samples, i.e., x[n − M : n]. These samples are linearly combined using a fixed vector b of coe ffi cients to produce an output sample y[n]. This procedure is described by a single formula:  y[n]= b 0 x[n]+ b 1 x[n − 1] + ··· + b M x[n − M],n ∈ Z, known as the filter input-output relationship.

An FIR filter has two notable properties: ▫Linearity:  If three identical filters are used on input sequences x (1), x (2) and x (3) = c 1 x (1) + c 2 x (2), then the output sequence of the third filter is the same (in terms of coe ffi cients) linear combination of the output sequences of the first two filters, i.e., y (3) =c 1 y (1) + c 2 y (2) ▫Time invariance:  If two identical filters are used on two input sequences which are time-delayed versions of each other, then the observed output sequences will also be time-delayed versions of each other (with the same delay).

An FIR filter modifies the spectrum of an input sequence x, i.e, it changes the amounts (coe ffi cients) of the sinusoidal components of x. To illustrate this point, we take a single complex sinusoid of frequency ω: x[n]= e jωn,n ∈ Z and put it through a filter with coe ffi cient vector b =[ ] T. The output sequence y is given by y[n]= x[n]+2x[n − 1] + 2x[n − 2] + x[n − 3] = e jωn + 2e jω(n−1) + 2e jω(n−2) +e jω(n−3) = (1+2e −jω +2e -j2ω +e -j3ω ) e jωn = (1+2e −jω +2e -j2ω +e −j3ω ) · x[n] Thus the output is a complex sinusoid of the same frequency; the filter merely scales the input by a complex factor which depends on the frequency ω. This is true for any FIR filter.

The scaling factor above is known as the frequency response of the filter and is denoted by H(e jω ): H(e jω ) = 1+2e −jω +2e -j2ω +e -j3ω It is a polynomial in (negative) powers of e jω. The expression for H(e jω ) can be simplified by noting the (non-circular) symmetry of the coe ffi cient vector b = b 0:3 = � [ ] T about the “middle” index 3/2. Factoring out e −j3ω/2, we obtain H(e jω ) = e −j3ω/2 (e j3ω/2 +2e jω/2 +2e -jω/2 +e -j3ω/2 ) = e −j3ω/2 · (4 cos(ω/2) + 2cos(3ω/2))

The modulus |H(e jω )| of the frequency response is known as the amplitude, or magnitude, response of the filter. In this case, noting that |e jθ | = 1, we have |H(e jω )| = |4 cos(ω/2) + 2cos(3ω/2)| This function of ω is symmetric about ω = π and has three zeros in the interval [0, 2π): at ω =2π/3, π and 4π/3. Thus for any of the three input sequences x (1) [n]=e j2πn/3, x (2) [n]=(−1) n, x (3) [n]=e j4πn/3,n ∈ Z, the filter output equals 0 for all n. By linearity, the same is true about any linear combination of these three input sequences; in particular, the sequence x[n] = cos(2πn/3+φ0). This means that those frequencies have been filtered from the original input

The angle ∠ H(e jω ) of the frequency response is known as the phase response of the filter. In this case, recalling that ∠ z 1 z 2 = ∠ z 1 + ∠ z 2, we have ∠ H(e jω )= −(3ω/2) + ∠ (4 cos(ω/2) + 2cos(3ω/2)) The first term is linear in ω, while the second equals 0 or ±π, depending on whether the real number 4 cos(ω/2) + 2cos(3ω/2) is positive or negative. This means that ∠ H(e jω ) is piecewise linear with constant slope (= −3/2) and discontinuities of size π occurring wherever 4 cos(ω/2) + 2cos(3ω/2) changes sign, i.e., at ω =2π/3, ω = π and ω =4π/3. Note: The phase response ∠ H(e jω ) is piecewise linear for all FIR filters whose coe ffi cients have even or odd symmetry about the middle index (M/2).

The expression for the frequency response H(e jω ) is always periodic with period (in ω) equal to 2π; as a result, H(e jω ) is (automatically) periodically extended outside the interval [0, 2π). The amplitude response |H(e jω )| is symmetric (even) about ω = 0 and π, while the phase response ∠ H(e jω ) is antisymmetric (odd) about the same frequencies. This is true for all FIR filters with real-valued coe ffi cients b 0,...,b M, since H(e jω )= b 0 + b 1 e -jω + ··· + b M e -jωM and H(e -jω )= H(e j(2π-ω) )= b 0 + b 1 e jω + ··· + b M e jωM = H ∗ (e jω ) (Recall that |z ∗ | = |z| and ∠ z* = − ∠ z.)

Plotting |H(e jω )| and ∠ H(e jω ) requires computing H(e jω ) at a su ffi ciently dense set of frequencies ω in [0, 2π), i.e., ω = 0, 2π/N, 4π/N,..., 1 − 2π/N, where N >> M These are the Fourier frequencies for a vector of length N. Provided N ≥ M +1, the resulting vector of values of H(e jω ) equals the DFT of [b ; 0 N−M−1 ]

Example The following MATLAB script computes the amplitude and phase responses of the FIR filter with input-output relationship y[n]= x[n]+2x[n − 1] + 2x[n − 2] + x[n − 3] at N = 512 uniformly spaced frequencies in [0, 2π). b=[ ].’ ; H = fft(b,512) ; A = abs(H) ; % amplitude response q = angle(H) ; % phase response w = (0: 1/512 : 1-1/512).’*(2*pi) ; plot(w,A) ; plot(w,q) ;

Similar conclusions can be drawn when the input signal is a complex exponential, i.e., x[n]= z n, n ∈ Z where z is an arbitrary complex number. Using the same filter as previously, we have, for all n, n−2 + z n−3 y[n]= z n +2z n−1 + 2z n−2 + z n−3 = (1+2z −1 +2z −2 + z −3 ) z n = (1+2z −1 +2z −2 + z −3 ) x[n] The complex scaling factor H(z) = 1+2z −1 +2z −2 + z −3 is known as the system function of the filter. The frequency response H(e jω ) of an FIR filter is the restriction of its system function to the unit circle z = e jω.

Problems: 4.2, 4.4