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Exp4 - Phase2 Report Problems

The document outlines a lab report for EEE 312, detailing experiments involving signal processing techniques such as windowing, spectral analysis, and the use of DFTs. It includes exercises on plotting data sequences, justifying claims about spectral resolution, and analyzing FDM systems. Additionally, it discusses the importance of short-time analysis in speech signal processing and provides a step-by-step process for visualizing the effects of frequency-domain techniques.

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0% found this document useful (0 votes)
13 views2 pages

Exp4 - Phase2 Report Problems

The document outlines a lab report for EEE 312, detailing experiments involving signal processing techniques such as windowing, spectral analysis, and the use of DFTs. It includes exercises on plotting data sequences, justifying claims about spectral resolution, and analyzing FDM systems. Additionally, it discusses the importance of short-time analysis in speech signal processing and provides a step-by-step process for visualizing the effects of frequency-domain techniques.

Uploaded by

ta1612642434
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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EEE 312 Lab Report: Experiment 4 Phase 2

Exercise 4 (4.1 – 4.3):

1. 64 samples 125µs apart are available for x(t) = sin (2𝜋f1t) + 0.05sin (2𝜋f2t), where f1 and f2
are 1062.5 Hz and 1625 Hz respectively.

(a) Plot the data sequence and magnitude spectrum. (use stem()).

(b) Perform windowing (Hann) and again plot the data sequence and magnitude spectrum
Comment on your result.

2. “Windowing not only distorts the spectral estimate due to the leakage effects; it also reduces
spectral resolution.” ---Justify this claim by explanation and simulated example. [See Proakis]

3. “Ability to resolve closely spaced spectral lines of different frequencies in limited by the
window main lobe width” --- Justify this statement by proper explanation and results of the
following problem.

Let, 𝑥[𝑛] = 𝑐𝑜𝑠𝜔𝑛 +𝑐𝑜𝑠𝜔𝑛+𝑐𝑜𝑠𝜔𝑛 where 𝜔 = 0.2𝜋, 𝜔 = 0.22𝜋 and 𝜔 = 0.6𝜋. Consider, window
lengths L = 25, 50, 100. Comment on your result.

Exercise 5.2:
Consider the following FDM system.

𝑠𝑖𝑛𝑐 (100𝑡), |𝑡| ≤ 𝑡 𝑠𝑖𝑛𝑐 (100𝑡), |𝑡| ≤ 𝑡


𝑚 (𝑡) = , 𝑚 (𝑡) =
0, 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒 0, 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
Where 𝑡 =0.1𝑠. Let, carrier 𝑐1(𝑡) = cos (2𝜋𝑓1 𝑡), where 𝑓1 = 250 𝐻𝑧. and carrier 𝑐2 (𝑡) =
cos(2𝜋𝑓2𝑡), where 𝑓2= 750 𝐻𝑧. Obtain magnitude spectra at all different stages of the system.
Regenerate the final demodulated message signal as well.
Insightful Exercise:

2. Present a technique to obtain N-point DFTs of two real sequences using a single N-
point DFT. Say x[n] = {1, 2, 0, 1} and y[n]= {2, 2, 1, 1}.

Obtain X[k] and Y[k] by using 4-point DFT only once.

3. Observe the output spectrum of an up-sampler. Consider the bandlimited input sequence used in
Appendix. If L is the upsampling factor then

>> y=zeros(1,L*length(x));

>> y(1:L:length(y))=x; % up-sampled sequence

Explain your answer on mathematical basis. [Hint: Find z-transform expression of up-
sampler output. Take help from Experiment 2 –Exercise 1.3]

Additional Task:

Speech signals are real-world examples of non-stationary signals — their frequency content
evolves over time due to changes in articulation, pitch, and phonemes. To analyze such signals
effectively, we often use short-time analysis by segmenting the signal into small frames (e.g., 20–
50 ms), within which the signal can be treated as approximately stationary. This allows us to apply
frequency-domain techniques such as the Discrete-Time Fourier Transform (DTFT) and the
Discrete Fourier Transform (DFT). Follow this step-by-step process to visualize the importance of
DTFT and DFT in speech signal processing.
 Load an audio file and extract a short frame
 Plot the time-domain signal
 Compute and plot the DTFT (via freqz())
 Compute and plot the DFT using fft()
 Perform frequency-domain low-pass filtering (Zero the high frequency components)
 Visualize filtering effect (Plot the frequency spectrum of the LPF)
 Reconstruct the time-domain signal
 Compare original and filtered signals

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