1. (50pt) NOTE: To get full mark, you are required to: (1) Plot sampled signals and...
36. Sampling a low-pass signal. A signal x(t) = sin( 1,000.71) is sampled at the rate of F, and sent through a unity-gain ideal low-pass filter with the cutoff frequency at F,/2. Find and plot the Fourier transform of the reconstructed signal z(t) at filter's output if a. F=20 kHz b. Fs =800 Hz
21. The signal x(t) = cos(1,8001t – 1/6) is sampled uniformly at the rate of 1 kHz and passed through an ideal low-pass filter with a DC gain of 0.001 and a cutoff frequency of 500 Hz. Find the filter's output.
3. (50 points] Consider the signal (t= cos(27 (100)+]: 1) Let's take samples of x(t) at a sampling rate fs = 180 Hz. Sketch the spectrum X (f) of the sampled signal x (t). Properly label x-axis and y-axis. 2) Now suppose we will use an ideal lowpass filter of gain 1/fs with a cutoff frequency 90 Hz for the sampled signal xs(t). What is the output of the filter x,(t)? 3) Now let's take samples of x(t) at sampling...
Q1) Given an analog signal X(t) = 3 cos (2π . 2000t) + 2 cos (2π . 5500t) sampled at a rate of 10,000 Hz, a. Sketch the spectrum of the sampled signal up to 20 kHz; b. Sketch the recovered analog signal spectrum if an ideal lowpass filter with a cutoff frequency of 4 kHz is used to filter the sampled signal in order to recover the original signal ; c. Determine the frequency/frequencies of aliasing noise . Q2)...
[MATLAB Scriptfile task] Design N-band tone vocoder with a given figure (below) in MATLAB implementing the given script file(bands_cutoff). This program should be able to process any sound(.wav) file. Then, graph the band-passed signals and amplitude envelopes extracted(after rectification and low-pass filtering) and waveforms of the original sound and vocoded sound. Additionally, using the output of the script file, make spectrograms of the original sound and the synthesized sound. Bandpass filter Modulation Band-limiting Envelope detection BPF RECT LPF BPF sine...
[MATLAB Scriptfile task] Design N-band tone vocoder with a given figure (below) in MATLAB implementing the given script file(bands_cutoff). This program should be able to process any sound(.wav) file. Then, graph the band-passed signals and amplitude envelopes extracted(after rectification and low-pass filtering) and waveforms of the original sound and vocoded sound. Additionally, using the output of the script file, make spectrograms of the original sound and the synthesized sound. Bandpass filter Modulation Band-limiting Envelope detection BPF RECT LPF BPF sine...
ON MATLAB: ii. Using FIR low-pass filter, remove signal S2, considering fc = 20 Hz as a cut-off frequency and consider two sets of filter coefficients: 11 and 301. Plot the time and frequency domain of the filtered signal, and comment. the process x(n) is sum of two signals S1 and S2; mathematically be described as: ?(?) = ?1 + ?2 where ?1 = ?1cos(2??1??? ) and ?2 = ?2cos(2??2??? ), A1 = A2 = 1; f1 = 10Hz; f2 =...
please select all the right answers QUESTION 1 Analyzer's task is to 0 A. Perform required computation with digital signals. O8. Convert analog time-domain signals into digital frequency domain information. Use digital Fourier transform to perform computation. DConvert digital frequency domain information into analog time-domain signals. QUESTION 2 When performing digital signal analysis O A- Sampling frequency needs to be at least 4 times the slowest frequency of the signal. Aliasing can occur whe n the sampling rate is too...
1. Read the laboratory supplement entitled “Frequency Response". 2. Read the remainder of this handout. 3. In Multisim, build the circuit shown in Figure 1 with C=0.22 uF and R = 2.2 k12. This circuit looks like a simple voltage divider except that one of the resistors is replaced by a capacitor. Il Figure 1: RC network. F Set up Vin to be a 1 Vpp sinusoid with 0 VDC Offset using a function generator. 2. Connect the oscilloscope in...
There is a system that can multiply and / or add several input signals (constants or variables) in the time domain; the input signals are f1 (t), f2 (t) and f3; the output signals are f4 (t), f5 (t) and f6 (t). The substime lowpass filter (FPBj) is a system that has a transfer function h (t) (H (w)), with unit gain in the operating area and cutoff frequency of + - 400 Hz. (The filter gain passes abruptly from...