嵌入式系统设计报告英文含代码.doc

1.System Design SchemaThis system is called speech spectrum analyzer based on the development tools called TM320C6416. The software is tested on code composer studio 6416(CCS6416). The system can realize the function that It can analyse the spectrum of the speech signals inputted by fast fourier transform(FFT) algorithm, and then print the frequency by ergodic algorithm, which can help find the number of the maxmum magnitude location. In addition, the system also can paly the voice which was recoded before. The C6X is used to designate a member of Texas Instruments (TI) TMS320C6000 family of digital signal processors. The architecture of the C6x digital signal processor is very well suited for numerically intensive calculations. The FFT with the radix-2 algorithm is a efficient algorithm that is used for converting a time-domain signal into an equivalent frequency-domine signal , based on the discrete frouier transform(DFT), The FFT reduces considerably the computational requirements of the DFT. The DFT of a discrete-time signal x(nT) isThe FFT algorithm takes advantage of the periodicity and symmetry of the twiddle constants to reduce the computational requirements of the FFT. From the periodicity of W,and from the symmetry of WAn eight-point FFT is illustrated through the following exercise. We will see that flow graphs for higher-order FFT (larger N) can readily be obtained.figure1. Eight-point FFT flow graph using DITFor the reason that the speech signals recognition is quite hard ,so 512-point FFT has been adopted. After FFT algorithm finished, 512 magnitudes will be acquired. Then the maximum value will be found by the ergodic algorithm, and we can get the accurate frequency by the equation of , and then print it in CCS. Besides, I have set many flags to identify whether the frequency belongs to high frequency, or low frequency by the way of the LED turning on.2.Goal and Test Level of the system AchievedThe system can record the voice in surroundings, and then play what it records, besides, the system can also analyse the frequency of the recording voice by FFT algorithm. And t the analysed speech signal spectrum by C language can be a consistent with what is analysed by the built-in algorithm of CCS6416. I have compared the result produced by C language and the result of the CCS6416s built-in GUI FFT algorithm, and I can get the perfect accordance with just 0.002% deviation. So the system can reach to the purpose accurately. 3.Block Diagram and Flow Chart of System ImplementationThis system consists of three parts, one is speech signal acquisition module, and another is signal processing module, which is finished by STM320C6416 and FFT algorithm, and the last is speech sognal output module.Block Diagram is showed in figure2: figure2 block biagram of speech signal spectrum analysis systemFlow chat of system is shown in figure3:figure3 flow chart of speech signal spectrum analysis systemAfter burnprocess has been finished, the system is waiting. In the case of testing the switch has been pressed, the record function starts, and the record will finish in the condition that switch three has been uplifted or the buffer is been fulfilled with that can be observed if the LED three is off. And then if switch zero has been pressed, the system will begin to play what has record before, and playing is finished in the condition of switch zero is uplifted, the the FFT begin after LED one turning on. The frequency will be printed in the CCS6416, and while LED one turns off ,it means the FFT is over, and the whole function has been finished.4.the Key Point and Difficulties of System ImplementationThe key point of this system is the implement of FFT algorithm. FFT algorithm has to analyze the spectrum of speech signal accurately, which is hard for C language to implement. The difficulties of the system implementation lies on the FFT algorithms operatoring speed and accuracy. I adopted 512-point FFT analysis, and then I find the maximum magnitude by ergodic algorithm, which has to compare all 512 points with each other to find the maxmum magnitude. There is no doubt that it is time-consuming. So the system can not fulfilled the real-time capability of computing magnitude. 5.Core Code and Explanation of SystemThe code of the system is listed as follows: /record.c record/play input using external memory#include #include math.h#include dsk6416_aic23.h/codec supportUint32 fs=DSK6416_AIC23_FREQ_8KHZ;/set sampling rateUint16 inputsource=DSK6416_AIC23_INPUT_MIC; / select input unsigned long i,count,s;short bufferN;short buffer1SAMPLENUMBER;long m;float max;void InitForFFT();int INPUTSAMPLENUMBER,DATASAMPLENUMBER;float fWaveRSAMPLENUMBER,fWaveISAMPLENUMBER,wSAMPLENUMBER;float sin_tabSAMPLENUMBER,cos_tabSAMPLENUMBER;#pragma DATA_SECTION(buffer,.EXT_RAM) /buffer -external memoryvoid main() int q,t,n,m,l,tt=0; short playing = 0; DSK6416_DIP_init(); DSK6416_LED_init(); comm_poll(); /init DSK, codec, McBSP InitForFFT(); count=0; while(1) /infinite loop if(DSK6416_DIP_get(3) = 0) /if SW#3 is pressed DSK6416_LED_on(3); /turn on LED#3 buffercount+ = input_left_sample(); if(DSK6416_DIP_get(3) =1|(count=N-1) /if SW#3 is pressed DSK6416_LED_off(3); if(DSK6416_DIP_get(0)=0) /if SW#0 pressed count=0; playing = 1; while (playing = 1) DSK6416_LED_on(0); /turn on LED#0 output_left_sample(buffercount+); if(count=N-1) DSK6416_LED_off(0); for(s=50;sN/SAMPLENUMBER;s+) n=0; for(l=SAMPLENUMBER*(s-1);ls*512;l+) n=l-SAMPLENUMBER*(s-1); buffer1n=bufferl; for(q=0;qSAMPLENUMBER;q+) fWaveRq=buffer1q; fWaveIq=0.0f; FFT(fWaveR,fWaveI); max=w0; m=0; for(t=0;tSAMPLENUMBER/2;t+) if(maxwt) max=wt;m=t; /for(l=SAMPLENUMBER*(s-1);l=1046&tt513) DSK6416_LED_on(2); else if(tt0) DSK6416_LED_on(1); else DSK6416_LED_on(3); DSK6416_LED_on(1); DSK6416_LED_on(2); if(tt25) printf(%dn,tt); DSK6416_LED_off(1); DSK6416_LED_off(2); DSK6416_LED_off(3); /for(s=1;s=N-1) / while (playing = 1) playing=0; DSK6416_LED_on(0); /if(DSK6416_DIP_get(0)=0) /while(1)/main void FFT(float dataRSAMPLENUMBER,float dataISAMPLENUMBER)int x0,x1,x2,x3,x4,x5,x6,x7,x8,xx;int i,j,k,b,p,L;float TR,TI,temp;/* following code invert sequence */for ( i=0;iSAMPLENUMBER;i+ )x0=x1=x2=x3=x4=x5=x6=x7=x8=0;x0=i&0x01; x1=(i/2)&0x01; x2=(i/4)&0x01; x3=(i/8)&0x01;x4=(i/16)&0x01; x5=(i/32)&0x01; x6=(i/64)&0x01;x7=(i/128)&0x01;x8=(i/256)&0x01;/x=x0*1024+x1*512+x2*256+x3*128+x4*64+x5*32+x6*16+x7*8+x8*4+x9*2+x10;xx=x0*256+x1*128+x2*64+x3*32+x4*16+x5*8+x6*4+x7*2+x8;dataIxx=dataRi;for ( i=0;iSAMPLENUMBER;i+ )dataRi=dataIi; dataIi=0; /* following code FFT */for ( L=1;L0 ) b=b*2; i-; /* b= 2(L-1) */for ( j=0;j0 ) /* p=pow(2,7-L)*j; */p=p*2; i-;p=p*j;for ( k=j;kSAMPLENUMBER;k=k+2*b ) /* for (3) */TR=dataRk; TI=dataIk; temp=dataRk+b;dataRk=dataRk+dataRk+b*cos_tabp+dataIk+b*sin_tabp;dataIk=dataIk-dataRk+b*sin_tabp+dataIk+b*cos_tabp;dataRk+b=TR-dataRk+b*cos_tabp-dataIk+b*sin_tabp;dataIk+b=TI+temp*sin_tabp-dataIk+b*cos_tabp; /* END for (3) */ /* END for (2) */ /* END for (1) */for ( i=0;iSAMPLENUMBER;i+ ) wi=sqrt(dataRi*dataRi+dataIi*dataIi); /* END FFT */Firstly, I set the sampling fruquency as 8 Khz, and the total buffer number is 512000 which means that it can record the voice as long as 64 seconds. The core of the code includes three parts. One is the recording part, and the second is playing part, and the third is the FFT analysis part. While the DSP chip detect the switch three was pressed, then recording starts. Recording is accomplished by the fuction called input_sample., which can complete the AD sampling and store the sampled value into buffer. LED three will turn off while the buffer has been filled or the switch three is uplifted. Then if you press switch zero, then playing starts, which can paly what has been recorded. And the playing part is accomplished by the function called output_sample, which can read the discrete value in the buffer and turn in to analogy speech signal. After the palying is over, LED zero will turn off, and LED one will turn on indicating that FFT with the radix-2 analysis begins. FFT algorithm computing is accomplished by looking up to table, and the table is made up by the sine function and cosine fuction, we will get the real and imaginary parts of magnitudes. And then by the way of ergodic algorithm to find the accurate number of location of the maxmum magnitude. We will get the frequency by equation of, After the FFT finished, LED one will turn off, and the speech signal spectrum analyzer has fulfilled all function.6.Debugging Recording of System and Testing IllustrationEach time, FFT analyze 512 points. So I divide the whole buffer into lots of little buffer containing 512 points. The debugging processing mainly depends on the location of breakpoints. I set the breakpoint in the location of FFT analyze of each little buffer called buffer1, and when the code execute the location of breakpoint, I open the view of FFT analyze of CCS6416 built-in, and compared the result printed by the FFT algorithm realized by C language and the result of the view