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Now, load your assembled filter file onto the DSP by selecting Load Program from the File menu. Finally, reset the DSP again, and execute the code by selecting Run from the Debug menu.

The program you are running accepts input from input channel 1 and sends output waveforms to output channels 1 and 2 (thefiltered signal and raw input, respectively). Note that the "raw input" on output channel 2 may differ from the actualinput on input channel 1, because of distortions introduced in converting the analog input to a digital signal and thenback to an analog signal. The A/D and D/A converters on the six-channel surround board operate at a sample rate of 44.1kHz and have an anti-aliasing filter and an anti-imaging filter , respectively, that in the ideal case would eliminate frequency content above 22.05kHz. The converters on the six-channel board are also AC coupled and cannot pass DC signals. On the basis of this information, what differences do you expect tosee between the signals at input channel 1 and at output channel 2?

Set the amplitude on the function generator to 1.0 V peak-to-peak and the pulse shape to sinusoidal. Observe thefrequency response of the filter by sweeping the input signal through the relevant frequency range. What is therelevant frequency range for a DSP system with a sample rate of 44.1 kHz?

Based on the frequency response you observe, characterize the filter in terms of its type (e.g., low-pass, high-pass,band-pass) and its -6 dB (half-amplitude) cutoff frequency (or frequencies). It may help to set the trigger on channel2 of the oscilloscope since the signal on channel 1 may go to zero.

Step 5: re-assemble and re-run with new filter

Once you have determined the type of filter the DSP is implementing, you are ready to repeat the process with adifferent filter by including different coefficients during the assembly process.Copy a second set of FIR coefficients over to your workingdirectory with the following:

  • copy coef.asm coef1.asm
  • copy v:\ece420\54x\dsplib\coef2.asm coef.asm

You can now repeat the assembly and testing process with the new filter using the asm instruction at the DOS prompt and repeating the steps required to execute the code discussed in Step 4 .

Just as you did in Step 4 , determine the type of filter you are running and thefilter's -6 dB point by testing the system at various frequencies.

Step 6: check filter response in matlab

In this step, you will use MATLAB to verify the frequency response of your filter by copying the coefficients from theDSP to MATLAB and displaying the magnitude of the frequency response using the MATLAB command freqz .

The FIR filter coefficients included in the file coef.asm are stored in memory on the DSP starting at location (in hex) 0x1000 , and each filter you have assembled and run has eight coefficients.To view the filter coefficients as signed integers, select the Memory option from the View menu to bring up a Memory Window Options box. In the appropriate fields, set the starting address to 0x1000 and the format to 16-Bit Signed Int . Click "OK" to open a memory window displaying the contents of the specified memorylocations. The numbers along the left-hand side indicate the memory locations.

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Source:  OpenStax, Ece 320 - spring 2003. OpenStax CNX. Jan 22, 2004 Download for free at http://cnx.org/content/col10096/1.2
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