Now we will design an FIR filter using the
same LabVIEW DFD toolkit used for the IIR filters. This LabVIEW DFDtoolkit designs 3 different types of FIR filters: Kaiser-Window
filters, Equi-Ripple filters and Dolph-Chevyshev Windows filters.We will focus on the Kaiser-Window FIR filters.
Open LabVIEW (not the Embedded Edition) and create a New
Blank VI.
In the block diagram, place the All Functions»Digital Filter
Design»Filter Design»DFD Classical Filter Design Express VI.
Double click in the VI and configure a lowpass FIR filter
with the following settings:
Type: Lowpass
Passband edge frequency: 400 Hz
Stopband edge frequency: 800 Hz
Passband ripple: 1dB (doesn’t matter because stopband specs
will determine passband ripple)
Minimum stopband attenuation: 40 dB
Sampling Frequency: 8000 Hz.
Design Method: Kaiser Window
Once you have all the parameters set
correctly, press OK.
Place the DFD Save to File VI found in All Functions»Digital
Filter Design»Utilities.
Wire the filter out to the filter in of the DFD Save to File
function. Wire the error clusters.
Save the VI in Desktop \ ee 453 \<folder name>using
some meaningful title such as Filter Design.vi.
Run the VI and when prompted save the Filter Coefficients
into the same folder with a meaningful name such as LPFIR400.fds.Make sure to use the default extension (fds) when saving this
coefficient file.
Determine the predicted filter length M using the Kaiser-window design equation:
Kaiser-window Design Equation
Recall that
alpha is the desired stopband attenuation and
delta-omega is the transition region width (in radians/sample). (Show your work.)
Close LabVIEW and open LabVIEW Embedded Edition. Target the
SPEEDY-33 and open the VI we built in the System Setup section ofthis Lab (lab3setup.vi).
Add two more Waveform Graphs on the Front Panel and label
them as Time Domain–Filtered and Frequency Domain-Filtered.
Add to the DFD Filter VI (All Functions»Signal
Processing»Filters) to the Block Diagram after the Add function.Wire the output of the Add function to the DFD filter. Wire the
output of the DFD filter to the Analog Output node. Wire the outputof the DFD filter to the Time Domain – Filtered terminal.
Make a copy of the Spectral Measurements Express VI. Wire the
output of the Add function to the copy of the SpectralMeasurements. Then wire the output of the Power Spectrum to the
Frequency Domain – Filtered terminal.
Double click on the DFD Filter VI and select the path to the
Coefficient File. As soon as you load the file, you will see themagnitude, impulse and phase response graphs update.
Save the VI. Start the CD Player and then Run the VI. You
should here a heavily lowpass filtered version of your musicthrough the headphones. You can also see the effect of the
filtering by looking at the appropriate indicators.
Stop the VI and, without changing anything else, change the
Sample Rate in the Analog Input node to 18000 Hz. Save and Run theVI and listen to the output now. When done, change the Sample Rate
back to 8000 Hz. Save the VI.
What happens when the sampling frequency used to operate the filter is changed from Fs = 8000 Hz (the Fs used to design the filter) to Fs = 16000 Hz.? Explain
Stop the VI and disconnect the Analog Input node and the Add
function. Replace these objects with the Simulate Signal Express VI(Functions»Embedded Signal Generator). Double click to bring up the
properties page and configure it to generate a 200 Hz sine wavewith amplitude 10000. Set the framesize to 256. Make sure that the
sampling frequency is 8000 Hz. Save the VI with a different name.Run the VI and observe the filtered output. Use the Time-Domain
plot to carefully measure the amplitude of the outputsignal.
Repeat step 16 without changing the name of the file for the
following frequencies: 400 Hz, 600 Hz, 800 Hz.
Complete the following table based on your measured data:
Does this filter yield the gains that you expected at the frequencies above? Why or why not? Specifically comment on the gain at 400 Hz, 600 Hz and 800 Hz.
Set the Simulate Signal back to 200 Hz. Now run the VI and
stop it. Because the 200 Hz is in the passband of the filter, theoutput of the filter will look like the input, except for a time
delay. Carefully measure and record the time (in milliseconds) ofthe first zero crossing in the input signal and the corresponding
zero crossing in the output signal.
Record the time delay (in milliseconds) between the input and output signal. Convert this to a sample delay (by multiplying time delay by Fs) and compare it to the theoretically expected answer. Discuss.
