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A digital filter is determined by the following difference equation. $$y(n)=y(n-1)+x(n)-x(n-4)$$ a) Find this filter's unit sample response. b) What is the filter's transfer function? How would you characterize this filter (lowpass, highpass, special purpose, \(\ldots) ?\) c) Find the filter's output when the input is the sinusoid \(\sin \left(\frac{\pi n}{2}\right)\). d) In another case, the input sequence is zero for \(n<0,\) then becomes nonzero. Sammy measures the output to be \(y(n)=\delta(n)+\delta(n-1)\). Can his measurement be correct? In other words, is there an input that can yield this output? If so, find the input \(x(n)\) that gives rise to this output. If not, why not?

Short Answer

Expert verified
a) Unit sample response is \( h(n) = \delta(n) + \delta(n-1) + \delta(n-2) + \delta(n-3) \). b) Transfer function is \( H(z) = \frac{1 - z^{-4}}{1 - z^{-1}} \); it's a high-pass filter. c) Output is periodic like input. d) Measurement is incorrect; no such \( x(n) \) exists.

Step by step solution

01

Find the Unit Sample Response

To find the unit sample response, we compute the output to an impulse input, i.e., \( x(n) = \delta(n) \). The difference equation is \( y(n) = y(n-1) + \delta(n) - \delta(n-4) \). For \( n < 0 \), \( y(n) = 0 \). For \( n = 0 \), \( y(0) = \delta(0) = 1 \). For \( n = 1 \), \( y(1) = y(0) + \delta(1) - \delta(-3) = 1 \). For \( n = 2 \), \( y(2) = y(1) + \delta(2) - \delta(-2) = 1 \). Similarly, \( y(3) = 1 \), but for \( n = 4 \), we have \( y(4) = y(3) + \delta(4) - \delta(0) = 0 \). Therefore, for \( n \geq 4 \), \( y(n) = 0 \). The unit sample response is \( h(n) = \delta(n) + \delta(n-1) + \delta(n-2) + \delta(n-3) \).
02

Determine the Transfer Function

The transfer function \( H(z) \) is found by taking the Z-transform of the difference equation. Using \( Y(z) = Y(z)z^{-1} + X(z) - X(z)z^{-4} \), solving for \( H(z) = \frac{Y(z)}{X(z)} \): \[ H(z) = \frac{1 - z^{-4}}{1 - z^{-1}} \]. This function has a zero at \( z = 1 \) and poles at \( z = 0 \). The transfer function corresponds to an ideal discrete-time differentiator, making it a high-pass filter.
03

Compute the Filter's Output for Sinusoidal Input

For the input \( x(n) = \sin\left(\frac{\pi n}{2}\right) \), we note this is a combination of phase-shifted complex exponentials. Given \( y(n) = y(n-1) + \sin\left(\frac{\pi n}{2}\right) - \sin\left(\frac{\pi (n-4)}{2}\right) \), the output sequence is a repeating pattern as \( \sin \left(\frac{\pi}{2}(n-4)\right) \) shifts \( \sin\left(\frac{\pi n}{2} \right) \) by multiples of periods. Because the input is periodic with period 4, so is the output.
04

Verify if Sammy's Measurement is Correct

The given output is \( y(n) = \delta(n) + \delta(n-1) \), suggesting an initial impulse and its delayed version. From Step 1, we found the unit impulse response has a different pattern, and given its finite duration, it cannot sum to this outcome. Therefore, there is no initial input \( x(n) \) satisfying this exact condition in the original difference equation. Sammy's measurement is not correct as the output doesn't align with achievable responses from zero initial conditions.

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Key Concepts

These are the key concepts you need to understand to accurately answer the question.

Difference Equation
In digital signal processing, a difference equation plays a crucial role in describing the relationship between the input and output of a digital filter. It's a mathematical formulation derived from discrete signals. In our exercise, the difference equation is given by \( y(n) = y(n-1) + x(n) - x(n-4) \). This provides a recursive formula that computes the current output \( y(n) \) using the previous output \( y(n-1) \) and current and delayed inputs \( x(n) \) and \( x(n-4) \) respectively.
This particular equation models how the current output depends not just on the immediate input but on inputs and outputs from several previous time steps as well. Such difference equations are indispensable in defining digital filters, which apply algorithmic changes to input signals over time.
Understanding how to manipulate and solve these equations is fundamental to working with digital filters in signal processing.
Unit Sample Response
The unit sample response, also known as the impulse response, is the output of a system when the input is an impulse signal \( \delta(n) \). It's a key concept as it characterizes the complete behavior of a filter or system.
For the digital filter described by the difference equation, the unit sample response is computed by substituting \( x(n) = \delta(n) \) into the equation. Practically, this involves finding \( y(n) \) at various points by propagating the impulse through the system.
In our example, for the given difference equation \( y(n) = y(n-1) + \delta(n) - \delta(n-4) \), the response \( h(n) \) was found to be \( h(n) = \delta(n) + \delta(n-1) + \delta(n-2) + \delta(n-3) \). This means the system produces a sequence of four impulses spanning from \( n = 0 \) to \( n = 3 \), and no response for \( n \geq 4 \). This knowledge is vital for understanding how the filter affects incoming signals.
Transfer Function
The transfer function of a digital filter provides a powerful tool for analyzing and understanding how the filter modifies signals in the frequency domain. It's typically represented in the Z-domain and in our case, is derived by taking the Z-transform of the difference equation.
Transforming the equation \( y(n) = y(n-1) + x(n) - x(n-4) \) gives the transfer function \( H(z) = \frac{1 - z^{-4}}{1 - z^{-1}} \). This expression helps to understand the impact of the filter in terms of frequencies.
Importantly, the structure of \( H(z) \) shows the placement of poles and zeros, which indicate frequencies that are suppressed or amplified. Our filter has zeros at \( z = 1 \) and poles at \( z = 0 \). This denotes a specific type of filter, known as the high-pass filter, as it allows high-frequency components to pass while attenuating low-frequency components.
High-Pass Filter
High-pass filters are an essential tool in digital signal processing. As the name suggests, they allow signals with a frequency higher than a certain cutoff frequency to pass through while reducing the amplitude of any signal with a lower frequency.
From the transfer function \( H(z) = \frac{1 - z^{-4}}{1 - z^{-1}} \), we identified our digital filter as a high-pass filter. This is because the equation zeroes out or minimizes low-frequency components while permitting high-frequency data.
High-pass filters are extensively used in applications like audio processing and communication systems where it's crucial to block unwanted low-frequency noise or interference, thereby enhancing the clarity and quality of the signal.
The high-pass behavior of the filter can significantly affect the characteristics of the output signal, making it vital to understand its function in designing various digital signal processing systems.

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Most popular questions from this chapter

Sampling Signals If a signal is bandlimited to \(W \mathrm{~Hz},\) we can sample it at any rate \(\frac{1}{T_{n}}>2 W\) and recover the waveform exactly. This statement of the Sampling Theorem can be taken to mean that all information about the original signal can be extracted from the samples. While true in principle, you do have to be careful how you do so. In addition to the rms value of a signal, an important aspect of a signal is its peak value, which equals \(\max \\{|s(t)|\\}\). a) Let \(s(t)\) be a sinusoid having frequency \(W \mathrm{~Hz}\). If we sample it at precisely the Nyquist rate, how accurately do the samples convey the sinusoid's amplitude? In other words, find the worst case example. b) How fast would you need to sample for the amplitude estimate to be within \(5 \%\) of the true value? c) Another issue in sampling is the inherent amplitude quantization produced by A/D converters. Assume the maximum voltage allowed by the converter is \(V_{\text {max }}\) volts and that it quantizes amplitudes to \(b\) bits. We can express the quantized sample \(Q\left(s\left(n T_{s}\right)\right)\) as \(s\left(n T_{s}\right)+\epsilon(t),\) where \(\epsilon(t)\) represents the quantization error at the \(n^{\text {th }}\) sample. Assuming the converter rounds, how large is maximum quantization error? d) We can describe the quantization error as noise, with a power proportional to the square of the maximum error. What is the signal-to-noise ratio of the quantization error for a full-range sinusoid? Express your result in decibels.

Sampling and Filtering The signal \(s(t)\) is bandlimited to \(4 \mathrm{kHz}\). We want to sample it, but it has been subjected to various signal processing manipulations. a) What sampling frequency (if any works) can be used to sample the result of passing \(s(t)\) through an \(\mathrm{RC}\) highpass filter with \(R=10 \mathrm{k} \Omega\) and \(C=8 \mathrm{nF} ?\) b) What sampling frequency (if any works) can be used to sample the derivative of \(s(t) ?\) c) The signal \(s(t)\) has been modulated by an \(8 \mathrm{kHz}\) sinusoid having an unknown phase: the resulting signal is \(s(t) \sin \left(2 \pi f_{0} t+\phi\right)\), with \(f_{0}=8 \mathrm{kHz}\) and \(\phi=?\) Can the modulated signal be sampled so that the original signal can be recovered from the modulated signal regardless of the phase value \(\phi ?\) If so, show how and find the smallest sampling rate that can be used; if not, show why not.

A problem on Samantha's homework asks for the 8 -point DFT of the discrete- time signal \(\delta(n-1)+\) \(\delta(n-7)\) a) What answer should Samantha obtain? b) As a check, her group partner Sammy says that he computed the inverse DFT of her answer and got \(\delta(n+1)+\delta(n-1) .\) Does Sammy's result mean that Samantha's answer is wrong? c) The homework problem says to low pass-filter the sequence by multiplying its DFT by $$H(k)=\left\\{\begin{array}{ll} 1 & \text { if } k=\\{0,1,7\\} \\ 0 & \text { otherwise } \end{array}\right.$$ and then computing the inverse DFT. Will this filtering algorithm work? If so, find the filtered output; if not, why not?

Find the Fourier transforms of the following sequences, where \(s(n)\) is some sequence having Fourier transform \(S\left(e^{j 2 \pi f}\right)\) a) \((-1)^{n} s(n)\) b) \(s(n) \cos \left(2 \pi f_{0} n\right)\) c) \(x(n)=\left\\{\begin{array}{l}s\left(\frac{n}{2}\right) \text { if } n(\text { even }) \\ 0 \text { if } n(\text { odd })\end{array}\right.\) d) \(n s(n)\)

The signal \(x(n)\) equals \(\delta(n)-\delta(n-1)\). a) Find the length-8 DFT (discrete Fourier transform) of this signal. b) You are told that when \(x(n)\) served as the input to a linear FIR (finite impulse response) filter, the output was \(y(n)=\delta(n)-\delta(n-1)+2 \delta(n-2) .\) Is this statement true? If so, indicate why and find the system's unit sample response; if not, show why not.

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