Let’s now go back to the original DSP program and add the newly designed filter. Switch to the Block Diagram and Double-click on the DFD Filter Express VI you added on the Block Diagram in step 1. In “Path” section of the “Configure Filter” prompt, click the Browse button and navigate to the filter design you saved earlier, click “OK” to accept. Run the program.
Because the signal frequency is significantly lower than the passband cutoff frequency, the signal should pass through the filter with very little attenuation. Change the frequency plot to manual scale. Sweep the input signal frequency through the range 500 Hz. – 3000 Hz. and note the resulting time-domain and frequency-domain plots.
Set the input sine wave to the following frequencies: 1000 Hz, 1200 Hz, 1500 Hz, 2000 Hz, and 3000 Hz. For each frequency input, observe the corresponding filter output (both displays as well as the actual sound) and measure/record the exact amplitude of the output signal for each frequency. The easiest way to measure the amplitudes is to stop the simulation and then place your mouse pointer at the appropriate position on the graph. Note: You will need to change the scale of the graphs to read accurately. Call your lab assistant for help on how to change the scale.
Answer these questions
Record the amplitude and corresponding attenuation (in dB) of the filtered sinusoid for each of the frequencies below.
The equation for attenuation is:
attenuation = -20log(output amplitude/(maximum amplitude.
Take special note of the attenuations at 1000 Hz and 1500 Hz. What do you expect these values to be theoretically? (show work) How do your actual values compare to the theoretical values?
Put the input frequency to 800 Hz and run the program again. Without changing anything else, change the sampling frequency in the Analog Input and Analog Output elements to 8000 Hz and run the program. Observe what happens to the output signal. When done, set all sampling frequencies back to 16000 Hz.
Answer these questions
Explain what happens to the filter’s performance when the sampling frequency is changed from Fs = 16000 Hz to 8000 Hz. Specifically, what happens to the 800 Hz signal when it is passed through the filter now? Explain what this implies about the importance of operating a digital filter using the same sampling frequency that you used to design it.
Now deactivate the Line-In input in Surround Mixer and run the program again using your CD input instead. Listen to and observe the output of the filter (in both the time and frequency domain).
Answer these questions
Discuss the effect of the lowpass filter on the music signal.
To really see the effect of the filter more graphically, remove the Analog Input element and Add function from the Block Diagram and add a EMB Uniform White Noise Waveform node instead (Functions»Embedded Signal Generation»EMB Uniform White Noise Waveform.vi). This block generates white noise. White noise has energy spread equally across the frequency spectrum. For this reason, a white noise generator is often a good test of a filter’s performance. Connect this block to the input of the IIR Filter. Double click on the Noise block and set the Amplitude = 10000, save and close it.
Run the program and observe the sound of the noise and take note of both its spectrum and time-domain shape. Now remove the lowpass filter block completely from the worksheet and connect the output of the Noise block directly to the Analog Output element. Re-run the program and observe the signal again. This is what unfiltered white noise looks/sounds like.
Program with Noise Generator
Answer these questions
Discuss how the shape of the white noise signal changes when it is passed through the lowpass filter. Also, describe how the sound changes.
Re-add an IIR filter block back into your system and connect it to the system the same way you did the previous IIR block. Load your new filter coefficient file into the IIR filter block. Keeping the Noise Generator as your input signal, run the program again and note the difference between how the noise gets filtered by the bandpass and lowpass filters.
Remove the Noise block and re-add the Analog Input node and the Add Function. Connect these to the rest of the system as you did earlier and run the system using your CD input. Observe the output signal.
Add a second IIR filter block to your system in parallel with the first block. Load the original lowpass filter coefficient file into this block. Double-click on the Analog Output Element and select 2 channels multiple samples. Now send the output of one filter to one of the channels of the Analog Output element and the output of the other filter to the other channel. (Note: Only one of the filter outputs can be connected to the displays. It doesn’t matter which one you have connected). Run the program using your CD input, and listen to the filtered output. You should hear a different signal in each ear.
As a final experiment, break the parallel filter connections and connect the 2 filter blocks in cascade rather than parallel by passing the output of one filter through the other filter and sending the output of the 2nd filter to the Analog Output element. Run the program using your CD input.
Answer these questions
When the lowpass and bandpass filters are cascaded together, what happens to the output signal? Why?
Call your TA over to verify that you have completed this last part of the lab. If time permits, feel free to design a few other types of IIR filters and connecting them in various ways.
Questions & Answers
A golfer on a fairway is 70 m away from the green, which sits below the level of the fairway by 20 m. If the golfer hits the ball at an angle of 40° with an initial speed of 20 m/s, how close to the green does she come?
A mouse of mass 200 g falls 100 m down a vertical mine shaft and lands at the bottom with a speed of 8.0 m/s. During its fall, how much work is done on the mouse by air resistance
Chemistry is a branch of science that deals with the study of matter,it composition,it structure and the changes it undergoes
Adjei
please, I'm a physics student and I need help in physics
Adjanou
chemistry could also be understood like the sexual attraction/repulsion of the male and female elements. the reaction varies depending on the energy differences of each given gender. + masculine -female.
Pedro
A ball is thrown straight up.it passes a 2.0m high window 7.50 m off the ground on it path up and takes 1.30 s to go past the window.what was the ball initial velocity
2. A sled plus passenger with total mass 50 kg is pulled 20 m across the snow (0.20) at constant velocity by a force directed 25° above the horizontal. Calculate (a) the work of the applied force, (b) the work of friction, and (c) the total work.
you have been hired as an espert witness in a court case involving an automobile accident. the accident involved car A of mass 1500kg which crashed into stationary car B of mass 1100kg. the driver of car A applied his brakes 15 m before he skidded and crashed into car B. after the collision, car A s
can someone explain to me, an ignorant high school student, why the trend of the graph doesn't follow the fact that the higher frequency a sound wave is, the more power it is, hence, making me think the phons output would follow this general trend?
Nevermind i just realied that the graph is the phons output for a person with normal hearing and not just the phons output of the sound waves power, I should read the entire thing next time
Joseph
Follow up question, does anyone know where I can find a graph that accuretly depicts the actual relative "power" output of sound over its frequency instead of just humans hearing
Joseph
"Generation of electrical energy from sound energy | IEEE Conference Publication | IEEE Xplore" ***ieeexplore.ieee.org/document/7150687?reload=true
A string is 3.00 m long with a mass of 5.00 g. The string is held taut with a tension of 500.00 N applied to the string. A pulse is sent down the string. How long does it take the pulse to travel the 3.00 m of the string?
