Digital Filters

Aim of the Experiment

A goal of this experiment is to implement discrete-time (digital) filters in real time.

In this experiment you will see the special architectural features of digital signal processors such as circular buffers that make the processors well suited for signal processing. You will implement different digital filters using in two different programming approaches:

In this experiment, you will construct finite impulse response (FIR) filter and infinite impulse response (IIR) filters on the DSP. For the FIR filter implementation, you will use the C-based convolution routine as well as the two assembly routines contained in the lab manual.

The response of these filters will be measured in two ways:

Equipment to be checked out

All the above equipment can be checked out from the checkout in the second floor. The above list of equipment is the equipment required per work station.

Directions

Downloads

Recitation slides Part 1 and Part 2 by Ms. Debarati Kundu, The University of Texas at Austin

Lab #3 Handout explaining milestones for lab #3. Updated Sept. 26, 2011

Assembly code files convol1.sa and convolve.sa.

TI C6748DSP files for lab #3 as a single zip file (7-zip free zip program). Updated Oct. 4, 2011

Overview Slides by Prof. Steven Tretter, University of Maryland (from Jan. 2008 lab manual)

OLD: LabVIEW Virtual instrument for test and measurement of results on the C6713 DSK.

Lab Report

All measurements and findings should be in lab report (cycle counts, frequency response, phase responses, etc.). Your report must contain the following:

  1. Theoretical and experimental amplitude response of the FIR filter.
  2. Profiler results for different optimization levels for FIR/C and FIR/Assembly.
  3. Number of taps before you run out of time for FIR/C.
  4. Why use circular buffers?
  5. What restrictions does the C6713 impose on the use of hardware circular buffers and why?
  6. Theoretical and experimental amplitude response of the IIR filter.
  7. Theoretical and experimental phase response of the IIR filter.
  8. What happens if you do not multiply by the scale factor in the IIR filter implementation?
  9. Advantages and disadvantages of FIR vs. IIR filters.

Milestones to Demonstrate to the TA

After each one of these parts, show them to the TA:

Coding Guidelines

For this lab, it is especially important to realize that the addition of an element into a circular array and the actual convolution calculation are two entirely separate operations.

Labs 5, 6, and 7 will directly involve performing multiple convolutions for FIR filters, so it makes sense to go ahead and put the Lab #3 convolution routine in a function.

As mentioned on the code grading page, the use of the C modulo operator "%" is discouraged and will be penalized.

Debugging and Troubleshooting Tips

Here are general tips for debugging and troubleshooting:

  1. Remez filter is just "equiripple" in fdatool.
  2. The "Lab_3_Block_FIR" folder on the website of lab 3 is only used for task 4. For tasks 2 and 3, please use the two main files from "Lab_3_FIR" folder and for other files please use project files in lab 1 or lab 2. We recommend lab 1 because the interrupt is already set.
  3. Please notice which channel are you using when reading samples from signal generator. You can also use T connector to feed both channels with signal.

When seeing an incorrect output on the oscilloscope, the cause may be one of the following:

  1. Changed the sampling rate in Interrupt Service Routine (isr) file but forgot to change it in the configuration file. The sampling frequency specified in isr file is only used for computation.
  2. The interrupt is not working. Even though the interrupt is not working, the oscilloscope could still show sinusoidal-like noise. This can be easily checked by measuring the magnitude of the waveform, and knowing that noise should have a very small magnitude.
  3. Using the single-channel versions of ReadSample(data) and WriteSample(data) defined in the files for lab 3. These functions have bugs in them. Instead, use the two-channel versions ReadSample(left, right) and WriteSample(left, right).

LabVIEW Implementation: (not for fall 2011)

Correction: The LabVIEW implementation will be provided to you for this lab.

The advantage of using LabVIEW to analyze the output as this is testing can be automated with fairly complicated analysis techniques (e.g. FFT) and these methods are not affected by non-ideal hardware (such as a faulty/poor data converter, bad signal generator, etc.)

The LabVIEW portion will generate a random signal (that has frequency content throughout the spectrum), send it to the DSP, the DSP will filter it (FIR or IIR), and LabVIEW will read the result. Then, by knowing the input signal and the output signal, the transfer function of the filter can be found. Conceptually, by knowing the z-transforms (or Fourier transforms) of both the input and the output, the transfer function can be found by dividing the two. The LabVIEW VIs for generating random noise and finding the transfer function work with arrays of data rather than individual samples. Thus, your LabVIEW VI will need to use a "for" loop to read and write samples to the DSP. The following LabVIEW VI's may be of use:

  1. Read & Write RTDX VI's (the Write VI is used in the lab2.vi)
  2. "PeriodicRandom" - Generates random samples
  3. "TransferFunction" - Finds frequency response of filter
  4. "UnwrapPhase" -- unwraps phase (especially useful for viewing FIR filter phase responses)

Assignment

Submitting this assignment is optional, but doing it would be useful with your QUIZ preparations

Assignment

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