基于tms320vc5402的自适应fir滤波器的设计毕业论文外文翻译

基于tms320vc5402的自适应fir滤波器的设计毕业论文外文翻译内容摘要：

s range 48000h–48FFFh and 16K external data memory in the address range 40000h–7FFFFh. Software system based on TMS320VC5402 The Integrates DSP development environment is CCS (Code Compo ser Studio) that is provided by TI. CCS cooperates with simulator XDS510. The adaptive FIR filter software design uses the assembly language to plete. Fig 5 shows the low chart of software program design. BK is FIR circular buffer size, and blockrepeat counter (BRC) is set 99. The initialization of hardware contains: clock, interrupt, RAM, McBSP0, AC01 and timer on chip, then endless loop is running in the main program until interrupt of timer es. Initialization of AC01 uses the synchronous serial port and skillfully applies mand idle 1. After the initialization data is transmitted then the system waits for the transmission interrupt of the serial port to guarantee safe transmission of data. The LMS, ST||MPY, and RPTBD instructions on the ’54x can reduce the execution time of code for adaptive filtering. LMS algorithm is given by LMS Xmem , Ymem. 5 The LMS instruction can be explained by MACXmem+ ,Ymem+,B。 BB+(*coeff+ *input +)|| { ADD Xmem ,16, A。 A(*coeff+A) 16} Where Xmem points to coefficient, and Ymem points to signal. The LMS algorithm calculates the filter output and updates each coefficient in the filter in parallel. Due to TMS320VC5402 is fixedpoint DSP, so input data, refer data and stepsize need to take format , then these data must be changed fixedpoint format, respectively multiply by 152 . Circular addressing and overflow would be noted in assemble program. Fig 5 Flow chat Using MATLAB verify algorithm, and obtaining sample values of input signal and desired signal, copying these data to assemble program, pile assemble file, parts of assemble codes given by: loop: LD 0,B STM 8,BRC。 Repeat 256 times RPTBD lms end1 MPY *input,A。 Au*e*x LMS *coeff,*input+。 BB+coeff*input, AA+0A16 ST A,*coeff+0%。 load coefficient to coff_buffer ||MPY *input,A LMS *coeff,*input+。 filtering, Baccumulate filtered output y, Afinal coefficient lms end: STH A,*coeff+0%。 store final coefficient STH B,*output。 store the filtered output 6 LD *refer+,A。 load refer data SUB *output+,A。 pute error STL A,*error LD @delta,T MPY *error+,A。 pute u*e(n) STH A,@deltau LD @deltau,T。 T=u*e MAR *+input(9)。 modify input buffer pointer, BANZ loop,*AR6 plete: nop Then piling, assembling and linking to create the COFF files for TMS320VC5402, and loading the files to DSP target system to run by simulator, using debugger to debug. Measuring wave has be gain as follows Comparing Fig 6(b) with Fig 6(c), we get that it is automatic adjustment through about 5 seconds after system starts, their error reduces gradually, the system converges, convergence rate is dependent on the appropriate choice of �, and we can obtain from Fig 6(d) that the longer the response time is, the smaller the error is. Accor。