Introduction In this exercise, you will program in the DSP's assembly language to create FIR filters. Begin by studying the assembly code for the basic FIR filter filter.asm.

filter.asm

filter.asm applies an FIR filter to the signal from input channel 1 and sends the resulting output to output channel 1. It also sends the original signal to output channel 2. First, create a work directory on your network drive for the files in this exercise, and copy filter.asm and core.asm into it. Then use MATLAB to generate two 20-tap FIR filters. The first filter should pass signals from 4 kHz to 8 kHz; the second filter should pass from 8 kHz to 12 kHz. For both filters, allow a 1 kHz transition band on each edge of the filter passband. To create these filters, first convert these band edges to digital frequencies based on the 44.1 kHz sample rate of the system, then use the MATLAB command remez to generate this filter; you can type help remez for more information. Use the save_coef command to save each of these filters into different files. (Make sure you reverse the vectors of filter coefficients before you save them.) Also save your filters as a MATLAB matrix, since you will need them later to generate test vectors. This can be done using the MATLAB save command. Once this is done, use the freqz command to plot the frequency response of each filter.

Part 1: Single-Channel FIR Filter For now, you will implement only the filter with a 4 kHz to 8 kHz passband. Edit filter.asm to use the coefficients for this filter by making several changes. First, the length of the FIR filter for this exercise is 20, not 8. Therefore, you need to change FIR_len to 20. FIR_len is set using the .set directive, which assigns a number to a symbolic name. You will need to change this to FIR_len .set 20. Second, you will need to ensure that the .copy directive brings in the correct coefficients. Change the filename to point to the file that contains the coefficients for your first filter. Third, you will need to modify the .align and .space directives appropriately. The TI TMS320C54x DSP requires that circular buffers, which are used for the FIR filter coefficient and state buffers, be aligned so that they begin at an address that is a multiple of a power of two greater than the length of the buffer. Since you are using a 20-tap filter (which uses 20-element state and coefficient buffers), the next greater power of two is 32. Therefore, you will need to align both the state and coefficient buffers to an address that is a multiple of 32. (16-element buffers would also require alignment to a multiple of 32.) This is done with the .align command. In addition, memory must be reserved for the state buffer. This is done using the .space directive, which takes as its input the number of bits of space to allocate. Therefore, to allocate 20 words of storage, use the directive .space 16*20 as shown below:

Assemble your code, set PMST to 0xFFE0, reset the DSP, and run. Ensure that it is has the correct frequency response. After you have verified that this code works properly, proceed to the next step.

Part 2: Dual-Channel FIR Filters First, make a copy of your modified filter.asm file from Part 1. Work from this copy; do not modify your working filter from the previous part. You will use that code again later. Next, modify your code so that in addition to sending the output of your first filter (with a 4 kHz to 8 kHz passband) to output channel 1 and the unfiltered input to output channel 2, it sends the output of your second filter (with a 8 kHz to 12 kHz passband) to output channel 3. To do this, you will need to use the .align and .copy directives to load the second set of coefficients into data memory. You will also need to add instructions to initialize a pointer to the second set of coefficients and to perform the calculations for the second filter. Challenge Problem Can you implement the dual-channel system without using the auxiliary registers AR4 and AR5? Why is this more difficult? Renaming AR4 and AR5 using the .asg directive does not count! Using the techniques introduced in DSP Development Environment: Introductory Exercise for TI TMS320C54x, generate an appropriate test vector and expected outputs in MATLAB. Then, using the test-vector core file also introduced in DSP Development Environment: Introductory Exercise for TI TMS320C54x, find the system's output given this test vector. In MATLAB, plot the expected and actual outputs of the both filters and the difference between the expected and actual outputs. Why is the output from the DSP system not exactly the same as the output from MATLAB?

Part 3: Alternative Single-Channel FIR Implementation An alternative method of implementing symmetric FIR filters uses the firs instruction. Modify your code from Part 1 to implement the filter with a 4 kHz to 8 kHz passband using the firs. Two differences in implementation between your code from Part 1 and the code you will write for this part are that (1) the firs instruction expects coefficients to be located in program memory instead of data memory, and (2) firs requires the states to be broken up into two separate circular buffers. Refer to the firs instruction on page 4-59 in the Mnemonic Instruction Set manual, as well as a description and example of its use on pages 4-5 through 4-8 of the Applications Guide for more information (Volumes 2 and 4 respectively of the TMS320C54x DSP Reference Set). AR0 needs to be set to -1 for this code to work properly. Why? COEFF is a label to the coefficients now expected to be in program memory. Refer to the firs description for more information).

Because states and coefficients are now treated differently than in your previous FIR implementation, you will need to modify the pointer initializations to

Use the test-vector core file to find the output of this system given the same test vector you used to test the two-filter system. Compare the output of this code against the output of the same filter implemented using the mac instruction. Are the results the same? Why or why not? Ensure that the filtered output is sent to output channel 1, and that the unmodified output is still sent to output channel 2.