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Step 2: log in

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

The development environment

The evaluation board is controlled by the PC through the JTAG interface (XDS510PP) using the application CodeComposer 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 theprovided 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 upa DOS prompt window and create a new directory to hold the files, then copy filter.asm , coef1.asm , coef2.asm , core.asm , and vectcore.asm into your directory.

Next, make a copy of coef1.asm called "coef.asm" and assemble the filter code by typing asm filter at the DOS prompt. The assembling process first includes the FIR filtercoefficients (stored in coef.asm ) into the assembly file filter.asm , then compiles the result to produce an output file containing the executablebinary 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. Beforeloading 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.

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 applicationwindow 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 setcorrectly, 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.

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 1and 2 (the filtered signal and raw input, respectively). Note that the "raw input" on output channel 2 may differfrom the actual input on input channel 1, because of distortions introduced in converting the analog input to adigital signal and then back to an analog signal. The A/D and D/A converters on the six-channel surround boardoperate 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 above22.05 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 youexpect to see between the signals at input channel 1 and at output channel 2?

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Source:  OpenStax, Dsp laboratory with ti tms320c54x. OpenStax CNX. Jan 22, 2004 Download for free at http://cnx.org/content/col10078/1.2
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