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.

	   
	  1	.copy "core.asm"  	; Copy in core file
	  2   					; This initializes DSP and jumps to "main"
	  3	
	  4  FIR_len .set   8			; This is an 8-tap filter.
	  5	
	  6          .sect ".data"		; Flag following as data declarations
	  7   
	  8    	    .align 16			; Align to a multiple of 16
	  9  coef					; assign label "coeff"
	  10	    .copy "coef.asm"		; Copy in coefficients
	  11
	  12	    .align 16
	  13 firstate
	  14 	    .space 16*8			; Allocate 8 words of storage for
	  15					; filter state.
	  16
	  17	    .sect ".text"		; Flag the following as program code
	  18  main
	  19      ; Initialize various pointers
	  20	    stm    #FIR_len,BK		; initialize circular buffer length
	  21	    stm    #coef,AR2    	; initialize coefficient pointer
	  22	    stm	   #firstate,AR3	; initialize state pointer
	  23	    stm    #1,AR0		; initialize AR0 for pointer increment
	  24	
	  25  loop
	  26      ; Wait for a new block of 64 samples to come in
	  27  	    WAITDATA
	  28	
	  29      ; BlockLen = the number of samples that come from WAITDATA (64)
	  30    	    stm     #BlockLen-1, BRC	; Put repeat count into repeat counter
	  31    	    rptb    endblock-1		; Repeat between here and 'endblock' 
	  32
	  33    	    ld	    *AR6,16, A		; Receive ch1 into A accumulator
	  34    	    mar     *+AR6(2)            ; Rcv data is in every other channel
	  35    	    ld      *AR6,16, B		; Receive ch2 into B accumulator
	  36    	    mar     *+AR6(2)            ; Rcv data is in every other channel
	  37    
	  38    	    ld	    A,B			; Transfer A into B for safekeeping
	  39
	  40      ; The following code executes a single FIR filter.
	  41	
	  42    	    sth     A,*AR3+%		; store current input into state buffer
	  43    	    rptz    A,(FIR_len-1)	; clear A and repeat
	  44    	    mac     *AR2+0%,*AR3+0%,A	; multiply coef. by state & accumulate
	  45    
	  46    	    rnd     A			; Round off value in 'A' to 16 bits                            
	  47    
	  48     ; end of FIR filter code. Output is in the high part of 'A.'
	  49
	  50    	    sth     A, *AR7+		; Store filter output (from A) into ch1
	  51    	    sth     B, *AR7+		; Store saved input (from B) into ch2
	  52    
	  53    	    sth     B, *AR7+		; Store saved input to ch3...ch6 also
	  54    	    sth     B, *AR7+		; ch4
	  55    	    sth     B, *AR7+		; ch5 
	  56    	    sth     B, *AR7+		; ch6
	  57   
	  58  endblock:
	  59      b loop
	

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 from v:\ece320\54x\dsplib to your work directory (this is thesame file you worked with in Lab 0). 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.

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:

	
	1         .align 32             % Align to a multiple of 32
	2  coef   .copy  "filter1.asm"  % Copy FIR filter coefficients
	3
	4         .align 32             % Align to a multiple of 32
	5  state  .space 16*20          % Allocate 20 words of data space
      

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.

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.

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?

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 [link] manual, as well as a description and example of its use on pages 4-5 through 4-8 of the Applications Guide [link] 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?

	
	 
	1	mvdd	*AR2,*AR3+0%		; write x(-N/2) over x(-N)
	2	sth	A,*AR2			; write x(0) over x(-N/2)
	3	add	*AR2+0%,*AR3+0%,A 	; add x(0) and x(-(N-1))
	4					;   (prepare for first multiply)
	5
	6	rptz	B,#(FIR_len/2-1)  	  
	7	firs	*AR2+0%,*AR3+0%,COEFF
	8	mar	??????? 		; Fill in these two instructions
	9	mar	???????			; They modify AR2 and AR3.
	10
	11					; note that the result is now in the
	12					;  B accumulator
      

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

	
	1	stm	#(FIR_len/2),BK		; initialize circular buffer length
	2	stm	#firstate_,AR2		; initialize location containing first 
	3					;   half of states
	4
	5	stm	#-1,AR0			; Initialize AR0 to -1
	6
	7	stm	#firstate2_,AR3	        ; initialize location containing last half
	
      

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.

The quiz for Lab 1 is broken down as follows:

The oral quiz may cover signal processing material relating to FIR filters, including, but not limited to, the delay through FIR filters, generalized linear phase, and the differences between ideal FIR filters and realizable FIR filters. You may also be asked questions about digital sampling theory, including, but not limited to, the Nyquist sampling theorem and the relationship between the analog frequency spectrum and the digital frequency spectrum of a continuous-time signal that has been sampled.

The oral quiz will cover the code that you have written during the lab. You are expected to understand, in detail, all of the code in the files you have worked on, even if your partner or a TA wrote it. (You are not expected to understand the core file in detail). The TA will ask you to explain various lines of code as part of the quiz. The TAs may also ask questions about 2's complement fractional arithmetic, circular buffers, alignment, and the mechanics of either of the two FIR filter implementations. You could be ready to trace through any of the code on paper and explain what each line of code does.

Use the TI documentation, specifically the Mnemonic Instruction Set [link] manual. Hard-copies of this manual can also be found in the lab. Also, feel free to ask the TAs to help explain the code that you have been given.