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DSP Development Environment: Introductory Exercise for TI TMS320C54x

Module by: Mark Butala, Jason Laska. E-mail the authors

Based on: DSP Development Environment: Introductory Exercise for TI TMS320C54x (ECE 320 Specific) by Mark Butala

Summary: This exercise introduces the hardware and software used in the course. By the end of this module, you should be comfortable with the basics of testing a simple real-time DSP system with Code Composer Studio, the debugging environment we will be using throughout the semester. First you will connect the laboratory equipment and test a real-time DSP system with provided code to implement an eight-tap (eight coefficient) finite impulse response (FIR) filter. With a working system available, you will then begin to explore the debugging software used for downloading, modifying, and testing your code. Finally, you will create a filter in MATLAB and use test vectors to verify the DSP's output.


This exercise introduces the hardware and software used in testing a simple DSP system. When you complete it, you should be comfortable with the basics of testing a simple real-time DSP system with the debugging environment you will use throughout the course. First, you will connect the laboratory equipment and test a real-time DSP system with pre-written code to implement an eight-tap (eight coefficient) finite impulse response (FIR) filter. With a working system available, you will then begin to explore the debugging software used for downloading, modifying, and testing code. Finally, exercises are included to refresh your familiarity with MATLAB.

Lab Equipment

This exercise assumes you have access to a laboratory station equipped with a Texas Instruments TMS320C549 digital signal processor chip mounted on a Spectrum Digital TMS320LC54x evaluation board. The DSP evaluation module should be connected to a PC running Windows and will be controlled using the PC application Code Composer Studio, a debugger and development environment. Mounted on top of each DSP evaluation board is a Spectrum Digital surround-sound module employing a Crystal Semiconductor CS4226 codec. This board provides two analog input channels and six analog output channels at the CD sample rate of 44.1 kHz. The DSP board can also communicate with user code or a terminal emulator running on the PC via a serial data interface.

In addition to the DSP board and PC, each laboratory station should also be equipped with a function generator to provide test signals and an oscilloscope to display the processed waveforms.

Step 1: Connect cables

Use the provided BNC cables to connect the output of the function generator to input channel 1 on the DSP evaluation board. Connect output channels 1 and 2 of the board to channels 1 and 2 of the oscilloscope. The input and output connections for the DSP board are shown in Figure 1.

Figure 1
Example Hardware Setup
Example Hardware Setup (setup.png)

Note that with this configuration, you will have only one signal going into the DSP board and two signals coming out. The output on channel 1 is the filtered input signal, and the output on channel 2 is the unfiltered input signal. This allows you to view the raw input and filtered output simultaneously on the oscilloscope. Turn on the function generator and the oscilloscope.

Step 2: Log in

Use the network ID and password provided to log into the PC at your laboratory station.

When you log in, two shared networked drives should be mapped to the computer: the W: drive, which contains your own private network work directory, and the V: drive, where the necessary file for ECE 320 are stored. Be sure to save any files that you use for the course to the W: drive. Temporary file may be stored in the C:\TEMP directory; however, since files stored on the C: drive are accessible to any user, are local to each computer, and may be erased at any time, do not store course files on the C: drive. On the V: drive, the directories v:\ece420\54kx\dsplib\ and c:\ece420\54x\dsptools\ contain the files necessary to assemble and test code on the TI DSP evaluation boards.

Although you may want to work exclusively in one or the other of the lab-partners' network accounts, you should be sure that both partners have copies of the lab assignment assembly code.


Not having the assembly code during a quiz because "it's on my partner's account" is NOT a valid excuse!
For copying between partners' directory on W: or for working outside the lab, SFTP (Secure FTP) access to your files is available at

The Development Environment

The evaluation board is controlled by the PC through the JTAG interface (XDS510PP) using the application Code Composer Studio. This development environment allows the user to download, run, and debug code assembled on the PC. Work through the steps below to familiarize yourself with the debugging environment and real-time system using the provided FIR filter code (Steps 3, 4 and 5), then verify the filter's frequency response with the subsequent MATLAB exercises (Steps 6 and 7).

Step 3: Assemble filter code

Before you can execute and test the provided FIR filter code, you must assemble the source file. First, bring up a command window (Start Menu -> Run, then type cmd) and create a new directory to hold the files, and then copy them into your directory:

  • w:
  • mkdir lab0
  • cd lab0
  • copy v:\ece420\54x\dsplib\filter.asm .
  • copy v:\ece420\54x\dsplib\coef.asm .

You can also copy the files via the windows user interface. After you copy the files, you will need to add the a new path. To do this type the following into the command terminal: v:\ece420\set420path.bat

Next, assemble the filter code by typing asm filter at the DOS prompt. The assembling process first includes the FIR filter coefficients (stored in coef.asm) into the assembly file filter.asm, then compiles the result to produce an output file containing the executable binary code, filter.out.

Step 4: Verify filter execution

With your filter code assembled, double-click on the Code Composer icon to open the debugging environment. Before loading your code, you must reset the DSP board and initialize the processor mode status register (PMST). To reset the board, select the Reset option from the Debug menu in the Code Composer application. The board must be reset several times until there are no 'red' registers showing in the output window at the bottom of the screen.

Once the board is reset, select the CPU Registers option from the View menu, then select CPU Register. This will open a sub-window at the bottom of the Code Composer application window that displays several of the DSP registers. Look for the PMST register; it must be set to the hexadecimal value FFE0 to have the DSP evaluation board work correctly. If it is not set correctly, change the value of the PMST register by double-clicking on the value and making the appropriate change in the Edit Register window that comes up. What does the PMST register do (Hint: use the reference manual)? Why is it neccesary to type 0xFFE0 rather than just FFE0 in the dialog box of Code Composer?

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 (the filtered signal and raw input, respectively). Note that the "raw input" on output channel 2 may differ from the actual input on input channel 1, because of distortions introduced in converting the analog input to a digital signal and then back to an analog signal. The A/D and D/A converters on the six-channel surround board operate at a sample rate of 44.1 kHz and have an anti-aliasing filter and an anti-imaging filter, respectively, that in the ideal case would eliminate frequency content above 22.05 kHz. 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 to see 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 the frequency response of the filter by sweeping the input signal through the relevant frequency range. What is the relevant 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 channel 2 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 a different filter by including different coefficients during the assembly process. Copy a second set of FIR coefficients over to your working directory 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 the filter's -6 dB point by testing the system at various frequencies.

Now copy a third filter into your directory. The filter can be found here coef3. Find the first zero (where the signal dissapears) in the scope. Because the filter is completely canceling out the sin function, we can analyze the spectrum of the signal present to look at what distortions result from the function generator as well as noise over the wires. To do this the FFT feature of the scope is used:

  • Press the MATH button on the scope.
  • Of the buttons located below the screen, select FFT.
  • Select settings.
  • Set 'source' to channel 1.

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 the DSP to MATLAB and displaying the magnitude of the frequency response using the MATLAB command freqz. The coefficients are stored in reverse order, to prepare the convolution with the stimulus, coming in on channel 1. This is explained in more detail in two paragraphs. Once MATLAB is open, type the commands help help, help lookfor, and help freqz. What is the best way to plot the frequency response of a filter, so that the frequency access is in kHz?

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 strength='9' 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 memory locations. The numbers along the left-hand side indicate the memory locations.

In this example, the filter coefficients are placed in memory in decreasing order; that is, the last coefficient, h7 h 7 , is at location 0x1000 and the first coefficient, h0 h 0 , is stored at 0x1007.

Now that you can find the coefficients in memory, you are ready to use the MATLAB command freqz to view the filter's response. You must create a vector in MATLAB with the filter coefficients to use the freqz command. For example, if we want to view the response of the three-tap filter with coefficients -10, 20, -10 we can use the following commands in MATLAB:

  • h = [-10, 20, -10];
  • plot(abs(freqz(h)))

Note that you will have to enter eight values, the contents of memory locations 0x1000 through 0x1007, into the coefficient vector, h.

Does the MATLAB response compare with your experimental results? What might account for any differences?

Now, try three more filters. Implement them on the DSP and plot the response in MATLAB to see if filters are behaving as expected. The filters are: coef4, coef5, coef6.

Step 7: Create new filter in MATLAB and verify

MATLAB scripts will be made available to you to aid in code development. For example, one of these scripts allows you to save filter coefficients created in MATLAB in a form that can be included as part of the assembly process without having to type them in by hand (a very useful tool for long filters). These scripts may already be installed on your computer; otherwise, download the files from the links as they are introduced.

First, have MATLAB generate a "random" eight-tap filter by typing h = gen_filt; at a MATLAB prompt. Then save this vector of filter coefficients by typing save_coef('coef.asm',flipud(h)); Make sure you save the file in your own directory. (The scripts that perform these functions are available as gen_filt.m and save_coef.m)

The save_coef MATLAB script will save the coefficients of the vector h into the named file, which in this case is coef.asm. Note that the coefficient vector is "flipped" prior to being saved; this is to make the coefficients in h h fill DSP memory-locations 0x1000 through 0x1007 in reverse order, as before.

You may now re-assemble and re-run your new filter code as you did in Step 5.

Notice when you load your new filter that the contents of memory locations 0x1000 through 0x1007 update accordingly.

Step 8: Modify filter coefficients in memory

Not only can you view the contents of memory on the DSP using the debugger, you can change the contents at any memory location simply by double-clicking on the location and making the desired change in the pop-up window.

Change the contents of memory locations 0x1000 through 0x1007 such that the coefficients implement a scale and delay filter with impulse response:

hn=8192δn4 h n 8192 δ n 4
Note that the DSP interprets the integer value of 8192 as a fractional number by dividing the integer by 32,768 (the largest integer possible in a 16-bit two's complement register). The result is an output that is delayed by four samples and scaled by a factor of 14 1 4 . More information on the DSP's interpretation of numbers appears in Two's Complement and Fractional Arithmetic for 16-bit Processors.


A clear and complete understanding of how the DSP interprets numbers is absolutely necessary to effectively write programs for the DSP. Save yourself time later by learning this material now!

After you have made the changes to all eight coefficients, run your new filter and use the oscilloscope to measure the delay between the raw (input) and filtered (delayed) waveforms.

What happens to the output if you change either the scaling factor or the delay value? How many seconds long is a six-sample delay? What is the most effective way to measure the delay between channel 1 and channel 2?

Step 9: Test-vector simulation

As a final exercise, you will find the output of the DSP for an input specified by a test vector. Then you will compare that output with the output of a MATLAB simulation of the same filter processing the same input; if the DSP implementation is correct, the two outputs should be almost identical. To do this, you will generate a waveform in MATLAB and save it as a test vector. You will then run your DSP filter using the test vector as input and import the results back into MATLAB for comparison with a MATLAB simulation of the filter.

The first step in using test vectors is to generate an appropriate input signal. One way to do this is to use the MATLAB function to generate a sinusoid that sweeps across a range of frequencies. The MATLAB function save_test_vector (available as save_test_vector.m can then save the sinusoidal sweep to a file you will later include in the DSP code.

Generate a sinusoidal sweep and save it to a DSP test-vector file using the following MATLAB commands:

	  >> t=sweep(0.1*pi,0.9*pi,0.25,500);    % Generate a frequency sweep
	  >> save_test_vector('testvect.asm',t); % Save the test vector

Next, use the MATLAB conv command to generate a simulated response by filtering the sweep with the filter h h you generated using gen_filt above. Note that this operation will yield a vector of length 507 (which is n+m1 n m 1 , where n n is the length of the filter and m m is the length of the input). You should keep only the first 500 elements of the resulting vector. Type help conv to get information on the conv command.

	  >> out=conv(h,t)                  % Filter t with FIR filter h
	  >> out=out(1:500)                 % Keep first 500 elements of out

Now, modify the file filter.asm to use the alternative "test vector" core file, vectcore.asm. Rather than accepting input from the A/D converters and sending output to the D/A, this core file takes its input from, and saves its output to, memory on the DSP. The test vector is stored in a block of memory on the DSP evaluation board that will not interfere with your program code or data.


The test vector is stored in the ".etext" section. See Core File: Introduction to Six-Channel Board for TI EVM320C54 for more information on the DSP memory sections, including a memory map.
The memory block that holds the test vector is large enough to hold a vector up to 4,000 elements long. The test vector stores data for both channels of input and from all six channels of output.

To run your program with test vectors, you will need to modify filter.asm. The assembly source is simply a text file and can be edited using the editor of your preference, including WordPad, Emacs, and VI. Replace the first line of the file with two lines. Instead of:

	  .copy 	"v:\ece320\54x\dsplib\core.asm"


	  .copy 	"testvect.asm"
	  .copy	"v:\ece320\54x\dsplib\vectcore.asm"

Note that, as usual, the whitespace in front of the .copy directive is required.

These changes will copy in the test vector you created and use the alternative core file. After modifying your code, assemble it, then load and run the file using Code Composer as before. After a few seconds, halt the DSP (using the Halt command under the Debug menu) and verify that the DSP has halted at a branch statement that branches to itself. In the disassembly window, the following line should be highlighted: 0000:611F F073 B 611fh.

Next, save the test output file and load it back into MATLAB. This can be done by first saving 3,000 memory elements (six channels times 500 samples) starting with location 0x8000 in program memory. Do this by choosing File->Data->Save... in Code Composer Studio, then entering the filename output.dat and pressing Enter. Next, enter 0x8000 in the Address field of the dialog box that pops up, 3000 in the Length field, and choose Program from the drop-down menu next to Page. Always make sure that you use the correct length (six times the length of the test vector) when you save your results.

Last, use the read_vector (available as read_vector.m) function to read the saved result into MATLAB. Do this using the following MATLAB command:

	  >> [ch1, ch2] = read_vector('output.dat');

Now, the MATLAB vector ch1 corresponds to the filtered version of the test signal you generated. The MATLAB vector ch2 should be nearly identical to the test vector you generated, as it was passed from the DSP system's input to its output unchanged.


Because of quantization error introduced in saving the test vector for the 16-bit memory of the DSP, the vector ch2 will not be identical to the MATLAB generated test vector.

After loading the output of the filter into MATLAB, compare the expected output (calculated as out above) and the output of the filter (in ch1 from above). This can be done graphically by simply plotting the two curves on the same axes; for example:

	  >> plot(out,'r'); % Plot the expected curve in red 
	  >> hold on        % Plot the next plot on top of this one 
	  >> plot(ch1,'g'); % Plot the expected curve in green 
	  >> hold off 

You should also ensure that the difference between the two outputs is near zero. This can be done by plotting the difference between the two vectors:

	  >> plot(out(1:length(ch1))-ch1); % Plot error signal 

You will observe that the two sequences are not exactly the same; this is due to the fact that the DSP computes its response to 16 bits precision, while MATLAB uses 64-bit floating point numbers for its arithmetic. Truncate the floating point numbers in MATLAB so that they are rounded to 16 bits, using the round or fix command. Now how do the numbers compare?

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