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📄 fft.c

📁 xmms-1.2.10.tar.gz学习使用的就下吧
💻 C
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/* fft.c: Iterative implementation of a FFT * Copyright (C) 1999 Richard Boulton <richard@tartarus.org> * Convolution stuff by Ralph Loader <suckfish@ihug.co.nz> * *  This program is free software; you can redistribute it and/or modify *  it under the terms of the GNU General Public License as published by *  the Free Software Foundation; either version 2 of the License, or *  (at your option) any later version. * *  This program is distributed in the hope that it will be useful, *  but WITHOUT ANY WARRANTY; without even the implied warranty of *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the *  GNU General Public License for more details. * *  You should have received a copy of the GNU General Public License *  along with this program; if not, write to the Free Software *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. *//* * TODO * Remove compiling in of FFT_BUFFER_SIZE?  (Might slow things down, but would * be nice to be able to change size at runtime.) * Finish making / checking thread-safety. * More optimisations. */#include "config.h"#include "fft.h"#include <glib.h>#include <stdlib.h>#include <math.h>#ifndef PI #ifdef M_PI  #define PI M_PI #else  #define PI            3.14159265358979323846  /* pi */ #endif#endif/* ########### *//* # Structs # *//* ########### */struct _struct_fft_state {    /* Temporary data stores to perform FFT in. */    float real[FFT_BUFFER_SIZE];    float imag[FFT_BUFFER_SIZE];};/* ############################# *//* # Local function prototypes # *//* ############################# */static void fft_prepare(const sound_sample *input, float * re, float * im);static void fft_calculate(float * re, float * im);static void fft_output(const float *re, const float *im, float *output);static int reverseBits(unsigned int initial);/* #################### *//* # Global variables # *//* #################### *//* Table to speed up bit reverse copy */static unsigned int bitReverse[FFT_BUFFER_SIZE];/* The next two tables could be made to use less space in memory, since they * overlap hugely, but hey. */static float sintable[FFT_BUFFER_SIZE / 2];static float costable[FFT_BUFFER_SIZE / 2];/* ############################## *//* # Externally called routines # *//* ############################## *//* --------- *//* FFT stuff *//* --------- *//* * Initialisation routine - sets up tables and space to work in. * Returns a pointer to internal state, to be used when performing calls. * On error, returns NULL. * The pointer should be freed when it is finished with, by fft_close(). */fft_state *fft_init(void) {    fft_state *state;    unsigned int i;    state = (fft_state *) g_malloc (sizeof(fft_state));    if(!state) return NULL;    for(i = 0; i < FFT_BUFFER_SIZE; i++) {	bitReverse[i] = reverseBits(i);    }    for(i = 0; i < FFT_BUFFER_SIZE / 2; i++) {	float j = 2 * PI * i / FFT_BUFFER_SIZE;	costable[i] = cos(j);	sintable[i] = sin(j);    }    return state;}/* * Do all the steps of the FFT, taking as input sound data (as described in * sound.h) and returning the intensities of each frequency as floats in the * range 0 to ((FFT_BUFFER_SIZE / 2) * 32768) ^ 2 * * FIXME - the above range assumes no frequencies present have an amplitude * larger than that of the sample variation.  But this is false: we could have * a wave such that its maximums are always between samples, and it's just * inside the representable range at the places samples get taken. * Question: what _is_ the maximum value possible.  Twice that value?  Root * two times that value?  Hmmm.  Think it depends on the frequency, too. * * The input array is assumed to have FFT_BUFFER_SIZE elements, * and the output array is assumed to have (FFT_BUFFER_SIZE / 2 + 1) elements. * state is a (non-NULL) pointer returned by fft_init. */void fft_perform(const sound_sample *input, float *output, fft_state *state) {    /* Convert data from sound format to be ready for FFT */    fft_prepare(input, state->real, state->imag);    /* Do the actual FFT */    fft_calculate(state->real, state->imag);    /* Convert the FFT output into intensities */    fft_output(state->real, state->imag, output);}/* * Free the state. */void fft_close(fft_state *state) {    if(state) free(state);}/* ########################### *//* # Locally called routines # *//* ########################### *//* * Prepare data to perform an FFT on */static void fft_prepare(const sound_sample *input, float * re, float * im) {    unsigned int i;    float *realptr = re;    float *imagptr = im;        /* Get input, in reverse bit order */    for(i = 0; i < FFT_BUFFER_SIZE; i++) {	*realptr++ = input[bitReverse[i]];	*imagptr++ = 0;    }}/* * Take result of an FFT and calculate the intensities of each frequency * Note: only produces half as many data points as the input had. * This is roughly a consequence of the Nyquist sampling theorm thingy. * (FIXME - make this comment better, and helpful.) *  * The two divisions by 4 are also a consequence of this: the contributions * returned for each frequency are split into two parts, one at i in the * table, and the other at FFT_BUFFER_SIZE - i, except for i = 0 and * FFT_BUFFER_SIZE which would otherwise get float (and then 4* when squared) * the contributions. */static void fft_output(const float * re, const float * im, float *output) {    float *outputptr = output;    const float *realptr   = re;    const float *imagptr   = im;    float *endptr    = output + FFT_BUFFER_SIZE / 2; #ifdef DEBUG    unsigned int i, j;#endif     while(outputptr <= endptr) {	*outputptr = (*realptr * *realptr) + (*imagptr * *imagptr);	outputptr++; realptr++; imagptr++;    }    /* Do divisions to keep the constant and highest frequency terms in scale     * with the other terms. */    *output /= 4;    *endptr /= 4;#ifdef DEBUG    printf("Recalculated input:\n");    for(i = 0; i < FFT_BUFFER_SIZE; i++) {        float val_real = 0;        float val_imag = 0;	for(j = 0; j < FFT_BUFFER_SIZE; j++) {	    float fact_real = cos(- 2 * j * i * PI / FFT_BUFFER_SIZE);	    float fact_imag = sin(- 2 * j * i * PI / FFT_BUFFER_SIZE);	    val_real += fact_real * re[j] - fact_imag * im[j];	    val_imag += fact_real * im[j] + fact_imag * re[j];	}	printf("%5d = %8f + i * %8f\n", i,	       val_real / FFT_BUFFER_SIZE,	       val_imag / FFT_BUFFER_SIZE);    }    printf("\n");#endif}/* * Actually perform the FFT */static void fft_calculate(float * re, float * im) {    unsigned int i, j, k;    unsigned int exchanges;    float fact_real, fact_imag;    float tmp_real, tmp_imag;    unsigned int factfact;        /* Set up some variables to reduce calculation in the loops */    exchanges = 1;    factfact = FFT_BUFFER_SIZE / 2;    /* Loop through the divide and conquer steps */    for(i = FFT_BUFFER_SIZE_LOG; i != 0; i--) {	/* In this step, we have 2 ^ (i - 1) exchange groups, each with	 * 2 ^ (FFT_BUFFER_SIZE_LOG - i) exchanges	 */	/* Loop through the exchanges in a group */	for(j = 0; j != exchanges; j++) {	    /* Work out factor for this exchange	     * factor ^ (exchanges) = -1	     * So, real = cos(j * PI / exchanges),	     *     imag = sin(j * PI / exchanges)	     */	    fact_real = costable[j * factfact];	    fact_imag = sintable[j * factfact];	    	    /* Loop through all the exchange groups */	    for(k = j; k < FFT_BUFFER_SIZE; k += exchanges << 1) {		int k1 = k + exchanges;		/* newval[k]  := val[k] + factor * val[k1]		 * newval[k1] := val[k] - factor * val[k1]		 **/#ifdef DEBUG		printf("%d %d %d\n", i,j,k);		printf("Exchange %d with %d\n", k, k1);		printf("Factor %9f + i * %8f\n", fact_real, fact_imag);#endif		/* FIXME - potential scope for more optimization here? */		tmp_real = fact_real * re[k1] - fact_imag * im[k1];		tmp_imag = fact_real * im[k1] + fact_imag * re[k1];		re[k1] = re[k] - tmp_real;		im[k1] = im[k] - tmp_imag;		re[k]  += tmp_real;		im[k]  += tmp_imag;#ifdef DEBUG		for(k1 = 0; k1 < FFT_BUFFER_SIZE; k1++) {		    printf("%5d = %8f + i * %8f\n", k1, real[k1], imag[k1]);		}#endif	    }	}	exchanges <<= 1;	factfact >>= 1;    }}static int reverseBits(unsigned int initial) {    unsigned int reversed = 0, loop;    for(loop = 0; loop < FFT_BUFFER_SIZE_LOG; loop++) {	reversed <<= 1;	reversed += (initial & 1);	initial >>= 1;    }    return reversed;}

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