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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 verifiedthat 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 previouspart. 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 to12 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 initializea pointer to the second set of coefficients and to perform the
calculations for the second filter.
One extra credit point will be awarded to you and your
partner if you can implement the dual-channel systemwithout 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 TITMS320C54x , generate an appropriate test vector and expected outputs in MATLAB. Then, using the test-vector corefile also introduced in DSP Development Environment: Introductory Exercise for TITMS320C54x , find the system's output given this test vector. In MATLAB, plot the expected and actual outputs ofthe both filters and the difference between the expected and actual outputs. Why is the output from the DSP system notexactly 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 insteadof data memory, and (2)
firs requires the states
to be broken up into two separate circular buffers. Refer tothe
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 programmemory. Refer to the
firs description for more
information).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 tomodify 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 thetwo-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 outputchannel 1, and that the unmodified output is still sent to
output channel 2.
The quiz for Lab 1 is broken down as follows:
.asm file
ready to
demonstrate each. Of the 4 points, you get 0.5 points for asingle 20-tap filter, 2 points for the two-filter system,
and 1.5 points for the system using the
firs opcode.The oral quiz may cover signal processing material relating to FIR filters, including, but not limited to, the delay throughFIR filters, generalized linear phase, and the differences between ideal FIR filters and realizable FIR filters. You mayalso be asked questions about digital sampling theory, including, but not limited to, the Nyquist sampling theoremand the relationship between the analog frequency spectrum and the digital frequency spectrum of a continuous-time signalthat has been sampled.
The oral quiz will cover the code that you have written during the lab. You are expected tounderstand, in detail, all of the code in the files you have worked on, even if your partner or a TA wrote it. (You arenot expected to understand the core file in detail). The TA will ask you to explain various lines of code as part of thequiz. The TAs may also ask questions about 2's complement fractional arithmetic, circular buffers, alignment, and themechanics of either of the two FIR filter implementations. You could be ready to trace through any of the code on paperand explain what each line of code does.
Use the TI documentation, specifically the Mnemonic Instruction Set manual. Hard-copies of this manual can also be found in the lab. Also, feel free to ask the TAs tohelp explain the code that you have been given.
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