Guide Digital Signal Processing: Laboratory Manual

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DIGITAL SIGNAL PROCESSING LAB MANUAL. P. M. 1 SATHYABAMA UNIVERSITY (Established under Section 3 of the UGC Act, ) (A CHRISTIAN.
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Some features of this site may not work without it. We apologize for the inconvenience. Development of experiments for the digital signal processing teaching laboratory. Date Author Jen, Kwang-Suz. Metadata Show full item record. To represent complex exponential as a function of real and imaginary. To determine impulse and step response of two vectors using. To perform linear convolution of two sequence using DFT using.

To determine z-transform from the given transfer function and its ROC.

Digital Signal Processing

To determine rational z-transform from the given poles and zeros using. To determine partial fraction expansion of rational z-transform using. To generate rectangular ,hamming ,hanning ,blackman and kaier. To study coefficient quantization effects on the frequency response of a.

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AIM : To represent basic signals like. From this experiment we can conclude that we can observe the waveforms. How circular convolution is used to calculate the Z-transform of a signal? List few Applications of circular convolution 7. What are the different methods used to calculate circular convolution?


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Explain properties of circular convolution? Explain modulo N operation What is the importance of circular convolution to realization of digital systems or digital filters? An FIR transversal filter structure can be obtained directly from the equation for discrete-time convolution. FIR — filter is a finite impulse response filter. Order of the filter should be specified. Infinite response is truncated to get finite impulse response. Placing a window of finite length does this.

What is a filter? Differentiate analog filter and digital filter. Define FIR filter.

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What are the differences between recursive and non recursive systems? List a few Applications of FIR filters. Explain limitations of FIR filters. What is the different method to design FIR filters? Explain different window functions. Differentiate rectangular, triangular and Kaiser windows.

IIR filters can be expanded as infinite impulse response filters. In designing IIR filters, cutoff frequencies of the filters should be mentioned. The order of the filter can be estimated using Butterworth polynomial. Filter coefficients can be found and the response can be plotted. List some advantages of digital filters over analog filters.

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What are the different methods to design IIR filters? Why IIR filters are not reliable? What are different applications of IIR filters? What are advantages of IIR filters? What are disadvantages of IIR filters? Differentiate Butterworth and Chebyshev approximations. What is meant by impulse response? The above equation is the mathematical representation of the DFT. As the number of computations involved in transforming an N point time domain signal into its corresponding frequency domain signal was found to be N2 complex multiplications, an alternative algorithm involving lesser number of computations is opted.

Define transform. What is the need for transform? Differentiate Fourier transform and discrete Fourier transform. What is meant by radix? What is meant by twiddle factor and give its properties? How FFT is useful to represent a signal? How the original signal is reconstructed from the FFT of a signal?


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The effects are easiest to observe when only one of these is varied between the two sinusoids being added. In the simplest case, when two sinusoids with the same frequency and phase but with different amplitudes are added together the result is a sinusoid who's amplitude is the sum of the originals and who's frequency and phase remain unchanged. When the two sinusoids have different frequencies the result is more complicated.

The new signal is no longer a sinusoid since it doesn't follow the simple up and down pattern. The frequency of the resulting signal will be the lower of the two original frequencies. In the figure we can see that the resulting signal goes through two and a half cycles that is it repeats itself this many times just like the second sinusoid. The amplitude of the resulting signal is the sum of the originals and the phase is unchanged. Adding together sinusoids with different phases can have surprising results. The phase of the resulting sinusoid is the sum of the phases of the constituents.

When a number of different frequencies are combined the frequency of the result will in general be that of the lowest frequency component. This is often called the fundamental frequency of the signal. What is meant by ringing effect? Why do we need to represent a signal in frequency domain? Why Fourier Series converges only for periodic signals? How the ringing effect can be rectified? What are the different forms of Fourier series?

What are the limitations of Fourier series? Write few applications of Fourier series. The frequency — magnitude plot gives the frequency response of the filter. LPF 2. What are the filter specifications required to design the analog filters? What is meant by frequency response of filter? What is meant by magnitude response? What is meant by phase response? Differentiate ideal filter and practical filter responses.

What are the different types of analog filter approximations? Define order of the filter and explain important role it plays in designing of a filter. Explain advantages and disadvantages of Butterworth filter 9.

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Explain advantages and disadvantages of Chebyshev filter Why Chebyshev is better than Butterworth filter? Note: The integral definition of the Fourier transform is only well-defined when the function is integrable. It is not sufficient for a function to be simply square-integrable. In this case one would need to use the Plancherel theorem. An often more useful alternative is the power spectral density PSD , which describes how the power of a signal or time series is distributed with frequency.

Here power can be the actual physical power, or more often, for convenience with abstract signals, can be defined as the squared value of the signal, that is, as the actual power dissipated in a load if the signal were a voltage applied across it.

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Define power signal. Define energy signal. Define power spectral density of a signal. How the energy of a signal can be calculated? Explain difference between energy spectral density and power spectral density. Explain the PSD plot.