E E 2 7 5 Lab

June 30, 2006

FIR Filters in Matlab
Lab 5. FIR Filter Design in Matlab
Digital filters with finite-duration impulse reponse (all-zero, or FIR filters) have both advantages and disadvantages when compared to infinite-duration impulse response (IIR) filters. FIR filters have the following primary advantages: • They can have exactly linear phase. • They are always stable, even when quantized. • The design methods are generally linear. • They can be realized efficiently in hardware. • The filter startup transients have finite duration. The primary disadvantage of FIR filters is that they often require a much higher filter order than IIR filters to achieve a given level of performance. Correspondingly, the delay of these filters is often much greater than for an equal performance IIR filter.

FIR Methods
The Signal Processing Toolbox supports a variety of methods for the design of FIR filters. F ilterM ethod Windowing Multiband with Transition Bands Constrained Least Squares Arbitrary Response Raised Cosine Description Apply window to truncated inverse Fourier transform of desired filter Equiripple or least squares approach over frequency subbands Minimize squared integral error over entire frequency range subject to maximum error constraints Arbitrary responses, including nonlinear phase and complex filters Lowpass response with smooth, sinusoidal transition c 2006GM F ilterF unctions fir1, fir2, kaiserord firls, firpm, firpmord fircls, fircls1

cfirpm firrcos

The FIR filter coefficients give the impulse response. Try this: b = [-1 0 2 -3 1]. .Impulse Response Revisited FIR filters are described by difference equations of the form M y(n) = m=0 bm x(n − m) where the filtered signal y(n) is just a linear combination of current and previous values of the input signal x(n). The denominator of the transfer function will always be a = 1. impz(b.1) Why are the x-axis scales different? Linear Phase Filters A filter whose impulse response is symmetric about its midpoint is called a (generalized) linear phase filter. The order of the filter is n = length(b) − 1. • The magnitude of the DFT is scaled by the filter’s magnitude response (there is no amplitude distortion).etc.. The coefficients b are the numerator coefficients of the transfer function. % no need to specify the a coefficients stem(b) figure..]. For an order n linear phase FIR filter. • The phase shift φ of a filtered signal will vary linearly with frequency ω (pure time delay with no phase distortion).. the phase delay and group delay is n/2. • The DFT of the impulse response will be either purely real or purely imaginary. For such filters. • The phase delay −φ(ω)/ω and group delay −dφ(ω)/d(ω) will be equal and constant. If the input signal is the unit impulse x = [1 0 0 0 . then the corresponding impulse response y(n) = h(n) is identical to b(n): h(0) = b0 x(0) = b0 h(1) = b0 x(1) + b1 x(0) = b1 h(2) = b0 x(2) + b1 x(1) + b2 x(0) = b2 ..

The absence of either amplitude distortion or phase distortion preserves the waveform of signals in the passband. even symmetry Even length. . n + 1 H(0) = 0 b(k) = −b(n + 2 − k).0. n + 1 No restriction b(k) = b(n + 2 − k). n + 1 H(0) = 0 Symmetry ResponseH(1) No restriction H(1) = 0 H(1) = 0 No restriction The functions fir1.. Try: a = 1.5)... .. k = 1. Except for cfirpm. fircls.. odd symmetry Depending on the filter type. FIR Filter Types The symmetric impulse response of a linear phase filter can have an odd or an even number of points. . odd symmetry Even length.. The function cfirpm can design any type of linear or nonlinear phase filter. fvtool(b. all the FIR filter design functions in the Signal Processing Toolbox design linear phase filters only. Both firls and firpm design type III and IV linear phase FIR filters given a ’hilbert’ or ’differentiator’ flag. certain restrictions apply: F ilter T ype Type I Type II Type III Type IV F ilter Order Even Odd Even Odd ResponseH(0) (N yquist) b(k) = b(n + 2 − k). firls. k = 1. fircls1... . n + 1 No restriction b(k) = −b(n + 2 − k). and firrcos all design type I and II linear phase FIR filters by default. even symmetry Odd length. k = 1.. leading to four filter types: • • • • Type Type Type Type I: II: III: IV: Odd length. b = fir1(5. k = 1. firpm. . fir2...and can have an odd or even symmetry about the midpoint.a) Look at the phase delay and the group delay..

For odd-valued n in these cases. the abrupt truncation leads to overshoot (Gibb’s phenomenon) and ripples in the spectrum. The approximation to the ideal filter is “best” in a mean square sense. • h(n) is symmetrically truncated (multiplied by a finite. Windowing does not explicitly impose amplitude response constraints. However. Window-Based Design Windowing is a common design method for FIR filters. The undesirable effects of truncation are reduced or eliminated by the use of tapered windows. fir1 does not design type II highpass and bandstop filters. compared to other approximations of the same length. symmetric window) to create a linear phase finite impulse response. Typing help window provides a list of available functions: . such as passband ripple or stopband attenuation. by Parseval’s theorem. • The corresponding ideal impulse response h(n) is determined by the inverse Fourier transform. this response cannot be implemented in a digital filter because it is infinite and noncausal. In general.Because the frequency response of a type II filter is zero at the Nyquist frequency (“high” frequency). 50. fir1 adds 1 to the order and returns a type I filter. 100) specify? How does the filter change with the argument? Windowing Functions The Signal Processing Toolbox supports a variety of windows commonly used in FIR filter design. • The ideal frequency response H(f ) is sampled. In this method. It must be used iteratively to produce designs that meet such specifications. Try: edit windemo windemo(20) windemo(50) windemo(100) What do the arguments (20.

64.’m’) . figure. w = gausswin(64.alpha) and w = window(@gausswin.w) n=linspace(0.bartlett barthannwin blackman blackmanharris bohmanwin chebwin flattopwin gausswin hamming hann kaiser nuttallwin parzenwin rectwin triang tukeywin Bartlett window Modified Bartlett-Hanning window Blackman window Minimum 4-term Blackman-Harris window Bohman window Chebyshev window Flat Top window Gaussian window Hamming window Hann window Kaiser window Nuttall defined minimum 4-term Blackman-Harris window Parzen (de la Valle-Poussin) window Rectangular window Triangular window Tukey window Individual functions take inputs for a window length n and window parameters and return the window w in a column vector of length n. stem(n. wy=w’.alpha) both return a Gaussian window of length 64 with standard deviation equal to 1/alpha.3). Try: help window n=15. figure.wy. w=gausswin(n.*y. w=W(length(y)+1:end).6*pi.100). stem(n. stem(-7:7. y=sinc(n).y) W=gausswin(2*length(y).) The window function serves as a gateway to the individual functions. (Note: Use w’ for array products with row vector impulse responses.3).

By windowing the truncated signal in the time domain.5 0. Try: edit windft windft Tuncated signal and DFT: Truncated Signal 1 0.4 0.Explain the y and wy plots. high-frequency components are introduced that are visible in the DFT. endpoints are assigned a reduced weight.2 0.1 0.7 0. windowed signal and DFT: .8 0. but increase the width of the main lobe. Windowing and Spectra When a signal is truncated.3 0.5 0 −0. The effect on the DFT is to reduce the height of the side lobes.5 −1 0 0.6 0.9 1 DFT of Truncated Signal 50 40 Magnitude 30 20 10 0 0 1 2 3 4 5 6 7 8 9 10 Tuncated.

Windowed Signal 1 0. which can be any window in the Signal Processing Toolbox.3 0. Ideally. .7 0.8 0. The main lobe should be as narrow as possible and the side lobes should contain as little energy as possible. Several windows can be given as input arguments for comparative display.2 0. adjustable by changing the filter order. wvtool(windowname(n)) opens WVTool with time and frequency domain plots of the n-length window specified in windowname. Windowed Signal 25 20 Magnitude 15 10 5 0 0 1 2 3 4 5 6 7 8 9 10 Window Visualization Tool The transition bandwidth of a window-based FIR filter is determined by the width of the main lobe of the DFT of the window function.4 0.6 0. The Window Visualization Tool (WVTool) allows you to investigate the tradeoffs among different windows and filter orders. The actual approximation error is scaled by the amount of the passband magnitude response. Passband and stopband ripples are determined by the magnitude of the side lobe of the DFT of the window function.9 1 DFT of Truncated. and are usually not adjustable by changing the filter order. the spectrum of a window should approximate an impulse.1 0.Truncated.5 0.5 −1 0 0.5 0 −0.

For example. Some windows are combinations of simpler windows. The Hamming window minimizes side lobe peaks (at the expense of slower high-frequency decay). the Hann window is the sum of a rectangular and a cosine window. Can you think of an advantage one would have over the other? . Other windows are based on simple mathematical formulas for easy application.Try: wvtool(hamming(32). The Kaiser window has a parameter that can be tuned to control side lobe levels. wintool opens WinTool with a default 64-point Hamming window.experiment with different window designs and export them to the workspace. The Hann window is easy to use as a convolution in the frequency domain. • Use WinTool to interactively design windows with certain specifications and export them to the Matlab workspace. • Use WVTool for displaying and comparing existing windows created in the Matlab workspace. Most window types satisfy some optimality criterion. the Kaiser window gives the best approximation to such an optimal window.2. Try it . The Hann window improves high-frequency decay (at the expense of larger peaks in the side lobes). In the discrete domain. An optimal time-limited window maximizes energy in its spectrum over a given frequency band.kaiser(32.5).1). Comment on the Chebyshev compared to the Blackman-Harris window.kaiser(32.5).flattopwin(32)) wvtool(kaiser(32. Other windows emphasize certain desirable features.kaiser(32.10)) Window Design and Analysis Tool The Window Design and Analysis Tool (WinTool) is used in conjunction with the Window Visualization Tool. and the Bartlett window is the convolution of two rectangular windows.

4*(-25:25)). type fvtool(b. but a nonrectangular window reduces its magnitude. The window applied here is a simple rectangular window. This is at the expense of transition width (the windowed version takes longer to ramp from passband to stopband) and optimality (the windowed version does not minimize the integrated least squared error).4*sinc(0. fvtool(b. To display the filter’s frequency response in FVTool. digital low-pass filter with a cutoff frequency of ω0 rad/s.*hamming(51)’. Its impulse response sequence h(n) is 1 2π π −π H(ω)ejωn dω = 1 2π ω0 −ω0 ejωn dω = ω0 ω0 sinc n π π This filter is not implementable since its impulse response is infinite and noncausal. This “Gibb’s effect” does not vanish as the filter length increases. in the integrated least squares sense.*hamming(51)’. a length 51 filter with a lowpass cutoff frequency ω0 of 0. and magnitude 0 at frequencies between ω0 and π. Try: b=0.1) Ringing and ripples occur in the response. especially near the band edge. turncate it by applying a window. fvtool(bw.1) Using a Hamming window greatly reduces the ringing. Multiplication by a window in the time domain causes a convolution or smoothing in the frequency domain.4π rad/s is b=0. fvtool(b.Example: Lowpass Filter Consider the ideal.1) Right-click the y-axis label in FVTool and choose Magnitude squared on both plots. Retain the central section of impulse response in the turncation to obtain a linear phase FIR filter. For example.4*(-25:25)). By Parseval’s theorem. or “brick wall”.1) bw=b. This filter has magnitude 1 at all frequencies less than ω0 . To create a finite-duration impulse response. this is the length 51 filter that best approximates the ideal lowpass filter. Apply a length 51 Hamming window to the filter and display the result using FVTool: bw=b. .4*sinc(0.

If you do not specify a window. The commands n=50. The vector window must be n + 1 elements long. b=fir1(n. For a highpass filter. Given a filter order and a description of an ideal filter. fir1 applies a Hamming window. create a row vector b containing the coefficients of the order n Hamming-windowed filter. but they accept any windowing function. and bandstop. Both use Hamming windows by default. uses the window specified in column vector window for the design. The kaiserord function estimates the filter order.window). kaiserord returns appropriate input parameters for the fir1 function.Wn. Wn=0. half the sampling frequency. linear phase FIR filter with cutoff frequency W n. and a maximum allowable ripple.Where in the plot is the ringing reduced by the window? Standard Band FIR Design The Signal Processing Toolbox functions fir1 and fir2 are both based on the windowing method.Wn). bandpass. This is a lowpass. cutoff frequency. where 1 corresponds to the Nyquist frequency. Try: edit fir1demo fir1demo Exercise . W n is a number between 0 and 1. a vector of magnitude. b=fir1(n. append the string ’stop’ for the bandstop configuration. Given a vector of frequency band edges. these functions return a windowed inverse Fourier transform of the ideal filter. fir1 resembles the IIR filter design functions in that it is formulated to design filters in standard band configurations: lowpass. simply append the string ’high’ to the function’s parameter list. For a bandpass or bandstop filter.4. and Kaiser window β parameter needed to meet a given set of specifications. specify Wn as a two-element vector containing the passband edge frequencies. highpass.

4. Arbitrary Response FIR Filters The fir2 function also designs windowed FIR filters. The vector window must be n + 1 elements long. Try: edit directstop2 directstop2(10) directstop2(100) directstop2(500) edit nlinphase nlinphase . Filter the signal y with the designed filter. 3. The function cfirpm is used to design complex and nonlinear-phase equiripple FIR filters. (The IIR counterpart of this function is yulewalk). Compare signals and spectra before and after filtering.f. Use a sampling frequency of 8192 Hz. fir2 applies a Hamming window. The frequencymagnitude characteristics of this filter match those given by vectors f and m.1. b=fir2(n. Plot the response. 2. If you do not specify a window. b=fir2(n. uses the window specified in column vector window for the design. returns row vector b containing the n + 1 coefficients of an order n FIR filter.m. Design a windowed FIR bandstop filter to remove the 300 Hz component from the threetone signal with noise y from Lab IIR Filters in Matlab.f.m).window). It allows arbitrary frequency-domain constraints. but with an arbitrarily shaped piecewise linear frequency response.

integrated squared error between an ideal piecewise linear function and the magnitude response of the filter over a set of desired frequency bands. b=firls(n. n 2 sinc 1 − (2R ns ) ns 0≤R≤1 R is called the rolloff factor. which is the impulse response of an ideal lowpass filter. The function firls allows you to introduce constraints by defining upper and lower bounds for the frequency response in each band. forms the basis for several other interpolating functions of the form n . where f (0) = 1 ns One commonly-used form is the raised cosine function: h(n) = f (n)sinc hrc (n) = n cos(πR ns ) n . The function fircls1 is used specifically to design lowpass and highpass linear phase FIR filters using constrained least squares. Try: edit firlsdemo firlsdemo edit firclsdemo firclsdemo Raised Cosine Filters The sinc function. This results in .Multiband Filters The function firls designs linear-phase FIR filters that minimize the weighted. Like the sinc function.a) returns row vector b containing the n+1 coefficients of the order n FIR filter whose frequencyamplitude characteristics approximately match those given by vectors f and a. the raised cosine function is 1 at n = 0 and 0 at all other sampling instances n = ns . In contrast to the sinc function.f. the raised cosine has faster decaying oscillations on either side of the origin for R > 0.

improved reconstruction if samples are not acquired at exactly the sampling instants (i. and return an order n lowpass linear-phase FIR filter with a raised cosine transition band. The partial convolutions of the signal are returned to the time domain with the IFFT. r must be in the range [0. b=firrcos(n.F0.’rolloff’) interprets the third argument.25. Try: b = firrcos(20. freqz(b) Describe what you see.fs.’bandwidth’) are equivalent. all in hertz. Frequency Domain Filtering Often. The filter coefficients (impulse response) and each block of data are transformed to the frequency domain using the FFT. 1]. an input signal x(n) is partitioned into equal length data blocks. df . It also uses fewer past and future values in the reconstruction.. as compared to the sinc function. where they can be efficiently convolved using multiplication. and total attenuation at high frequencies. as the rolloff factor instead of the transition bandwidth.e. In the overlap-add method. b=firrcos(n. if there is jitter). The width of the transition band is determined by the rolloff factor.df.fs) b=firrcos(n. . b is normalized so that the nominal passband gain is always equal to 1. The cutoff frequency is F 0. a raised cosine function in the middle.0.F0. and the processed result constructed from the processed pieces.0.F0.fs. and sampling frequency is f s. df must be small enough so that F 0 ± df /2 is between 0 and f s/2. r. where they are shifted and summed using superposition.df. The shape of the function’s spectrum is the “raised cosine”. a long (perhaps continuous) stream of data must be processed by a system with only a finite length buffer for storage. The data must be processed in pieces. The ideal raised cosine lowpass filter frequency response consists of unity gain at low frequencies.r.2). the transition bandwidth df .25.

3). y=conv(h. y1=conv(h. edit filttimes filttimes Above what order can you say that fftfilt is always faster? (The material in this lab was put together by Paul Beliveau and handout derives principally from the MathWorks training document “MATLAB for Signal Processing”.x) uses an FFT length of nfft = 2nextpow2(n) and a data block length of nfft-length (b)+1 (ensures circular convolution). y=fftfilt(b. firfilt incurs an “offline” startup cost when converting the coefficients b to the frequency domain. zeros(1. fftfilt outperforms filter. After that.x1) y2=conv(h. Y=Y1+Y2 Describe what is happening in this code.x) x1=[1 2 3]. 2006.) c 2006GM . Try: x=[1 2 3 4 5 6].x2) Y1=[y1. h=[1 1 1].3)]. (Multiplications are a good measure of performance.fftfilt implements the overlap-add method for FIR filters. x2=[4 5 6]. The following script takes a few moments to run. Y2=[zeros(1. since they are typically expensive on hardware.) The net result is that for filters of high order.y2]. the number of multiplications fftfilt performs relative to filter (which implements the filter in direct form) is ≈ log2 (L)/N . where L is the block length and N is the filter length.