Digital Signal Processing Laboratory 7: IIR Notch Filters Using the TMS320C6711

Transcription

1 Digital Signal Processing Laboratory 7: IIR Notch Filters Using the TMS320C6711 PreLab due Wednesday, 3 November 2010 Objective: To implement a simple filter using a digital signal processing microprocessor using the TI Code Composer development tools. To compare the performance of different filter designs using inputs from a signal generator and viewing outputs on an oscilloscope To filter audio signals. Introduction: A signal processing microprocessor is a special purpose processor with an architecture designed to efficiently implement DSP operations in real time for embedded applications. For this lab and the following labs we will use a development kit board with a TI tms320c6416 and ADCs and DACs for real time signal processing. A block diagram taken from the second reference is shown here. The AIC23 Codec includes the conversions between digital and analog signals.

2 The Code Composer Studio development tools allow you to create programs that can be downloaded to the dsk6416 board. The compiler interprets the C code instructions and creates a sequence of assembly language instructions which will accomplish the desired operations. The.asm file, which is created by the compiler, is then used by the linker to create a corresponding set of binary machine instructions. When these instructions are loaded into the DSK processor, executing them implements the original C code instructions. In this laboratory we will use interrupts to respond to the analog interface ADC and DAC at 8000 samples per second. At each interrupt a data sample is read and added to a queue. The queue, which is implemented using a circular buffer, is used to compute the filtered output for an FIR filter. An IIR filter will have a queue for the input and a separate queue for the output. The frequency selective behavior of the filter is determined by its coefficients. The filter will be implemented using floating point arithmetic, but its output will be converted to a short integer before it is sent to the DAC. For the prelab, we will explore and modify a program written in C. The two files shown below will be provided in the laboratory. Based on these files, answer the questions below. File fir.c // program fir.c #include "DSK6416_AIC23.h" // codec support Uint32 fs=dsk6416_aic23_freq_8khz; //set sampling rate #define DSK6416_AIC23_INPUT_MIC 0x0015 #define DSK6416_AIC23_INPUT_LINE 0x0011 Uint16 inputsource=dsk6416_aic23_input_line; // select line in

3 #include "test.cof" float x[n]; //filter coefficient file //filter delay line interrupt void c_int11() //ISR - AIC23 codec interrupts at 8kHz { short i; float yn = 0.0; } x[0] = (float)(input_left_sample()); //get new input into delay line for (i=0 ; i<n ; i++) //calculate filter output yn += h[i]*x[i]; for (i=(n-1) ; i>0 ; i--) //shuffle delay line contents x[i] = x[i-1]; output_left_sample((short)(yn)); //output to codec return; void main() //main body of program does nothing { comm_intr(); //initialise DSK while(1); //infinite loop } File ave5.cof //ave5.cof Coefficient file for use with programs fir.c and firprn.c //Implements five point moving average filter #define N 5 float h[n] = {0.2, 0.2, 0.2, 0.2, 0.2}; Questions: 1. What does the main program do? Why is it set up to be in an infinite loop? 2. When an interrupt from the ADC indicates that a new sample of data is ready, the c_int11 function is called. a. How does this function have access to the values of N and the coefficients h? b. What is the purpose of the first for loop? c. What is the purpose of the second for loop? 3. If the first 10 data values that cause interrupts are [ ] indicate the value that would be output for y and the values that would be stored in the x array at the end of each call to the c_int11 function by an interrupt. Assume that when the program starts, the x array is initialized to have all values equal to 0. A table is provided below. 4. Repeat (3) for the case where h[n]={ } instead of the values shown. What would this filter do? 5. ± Modify the C code to implement a notch filter with two poles at ρ jω 0 z = e for ρ = 0. 9 and ± jω two zeros at z = e 0 where the frequency value is selected to put the notch at 1000 Hz. Specifically indicate the changes to the coefficient file and the interrupt processing file. Table for Question 3:

4 Interrupt 1 yn x[0] x[1] x[2] x[3] x[4] Interrupt 2 Interrupt 3 Interrupt 4 Interrupt 5 Interrupt 6 Interrupt 7 Interrupt 8 Interrupt 9 Interrupt 10 Table for Question 4: Interrupt 1 yn x[0] x[1] x[2] x[3] x[4] Interrupt 2 Interrupt 3 Interrupt 4 Interrupt 5 Interrupt 6 Interrupt 7 Interrupt 8 Interrupt 9 Interrupt 10 References: Download the following references:

5 For the Code Composer tools, download the document at For the DSK board, download the document at For the processor, download the document at Submit the answers to the questions, the two tables, and the modified C code. (c) Copyright Sally Wood