dsp_firlms2_c.c
来自「dm642函数库」· C语言 代码 · 共 117 行
C
117 行
/* ======================================================================== */
/* TEXAS INSTRUMENTS, INC. */
/* */
/* DSPLIB DSP Signal Processing Library */
/* */
/* This library contains proprietary intellectual property of Texas */
/* Instruments, Inc. The library and its source code are protected by */
/* various copyrights, and portions may also be protected by patents or */
/* other legal protections. */
/* */
/* This software is licensed for use with Texas Instruments TMS320 */
/* family DSPs. This license was provided to you prior to installing */
/* the software. You may review this license by consulting the file */
/* TI_license.PDF which accompanies the files in this library. */
/* ------------------------------------------------------------------------ */
/* Copyright (C) 2003 Texas Instruments, Incorporated. */
/* All Rights Reserved. */
/* ======================================================================== */
/* ======================================================================== */
/* NAME */
/* DSP_firlms2 -- Least Mean Square Adaptive Filter */
/* Natural C Implementation */
/* */
/* REVISION DATE */
/* 13-Mar-2002 */
/* */
/* USAGE */
/* */
/* This routine has the following C prototype: */
/* */
/* long DSP_firlms2 */
/* ( */
/* short *restrict h, // Filter Coefficients */
/* const short *restrict x, // Input Data */
/* short b, // Error from previous FIR */
/* int nh, // Number of Coefficients */
/* ) */
/* */
/* The DSP_firlms2 routine accepts a list of 'nh' input data and 'nh' */
/* adaptive filter coefficients and updates the coefficients by */
/* adding weighted error times the inputs to the original */
/* coefficients. This assumes single sample input followed by the */
/* last nh-1 inputs and nh coefficients. */
/* */
/* DESCRIPTION */
/* */
/* The algorithm is simple and is implemented as follows: */
/* */
/* long DSP_firlms2 */
/* ( */
/* short *restrict h, // Filter Coefficients */
/* const short *restrict x, // Input Data */
/* short b, // Error from previous FIR */
/* int nh, // Number of Coefficients */
/* ) */
/* */
/* { */
/* int i; */
/* long r=0; */
/* */
/* for(i = 0; i < nh; i++) */
/* { */
/* * Update the filter coefficients * */
/* */
/* h[i] += (x[i] * b) >> 15; */
/* */
/* * Calculate the filter output * */
/* */
/* r += x[i + 1] * h[i]; */
/* } */
/* return r; */
/* } */
/* */
/* ASSUMPTIONS */
/* */
/* - Assumes 16-bit input data, error and filter coefficients. */
/* - nh > = 4 and is a multiple of 4 */
/* */
/* SOURCE */
/* */
/* DSPLIB */
/* ------------------------------------------------------------------------ */
/* Copyright (c) 2003 Texas Instruments, Incorporated. */
/* All Rights Reserved. */
/* ======================================================================== */
long DSP_firlms2_c
(
short *restrict h, // h[nh] = Coefficient Array
const short *restrict x, // x[nh] = Input Array
short b, // b = Error from Previous FIR
int nh // nh = Number of Coefficients
)
{
int i;
long r = 0; // r is output of the filter
for (i = 0; i < nh; i++)
{
h[i] += (x[i] * b) >> 15;
r += x[i+1] * h[i];
}
return r;
}
/* ======================================================================== */
/* End of file: dsp_firlms2.c */
/* ------------------------------------------------------------------------ */
/* Copyright (c) 2003 Texas Instruments, Incorporated. */
/* All Rights Reserved. */
/* ======================================================================== */
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