Mercurial > libavcodec.hg
view resample2.c @ 2222:ef568cc0972c libavcodec
10l (gcc 2.95 compilation)
author | michael |
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date | Sun, 12 Sep 2004 17:35:54 +0000 |
parents | 3f52c129d00f |
children | b88e05b9b445 |
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/* * audio resampling * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ /** * @file resample2.c * audio resampling * @author Michael Niedermayer <michaelni@gmx.at> */ #include "avcodec.h" #include "common.h" #include "dsputil.h" #define PHASE_SHIFT 10 #define PHASE_COUNT (1<<PHASE_SHIFT) #define PHASE_MASK (PHASE_COUNT-1) #define FILTER_SHIFT 15 typedef struct AVResampleContext{ short *filter_bank; int filter_length; int ideal_dst_incr; int dst_incr; int index; int frac; int src_incr; int compensation_distance; }AVResampleContext; /** * 0th order modified bessel function of the first kind. */ double bessel(double x){ double v=1; double t=1; int i; for(i=1; i<50; i++){ t *= i; v += pow(x*x/4, i)/(t*t); } return v; } /** * builds a polyphase filterbank. * @param factor resampling factor * @param scale wanted sum of coefficients for each filter * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16 */ void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){ int ph, i, v; double x, y, w, tab[tap_count]; const int center= (tap_count-1)/2; /* if upsampling, only need to interpolate, no filter */ if (factor > 1.0) factor = 1.0; for(ph=0;ph<phase_count;ph++) { double norm = 0; double e= 0; for(i=0;i<tap_count;i++) { x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; if (x == 0) y = 1.0; else y = sin(x) / x; switch(type){ case 0:{ const float d= -0.5; //first order derivative = -0.5 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); else y= d*(-4 + 8*x - 5*x*x + x*x*x); break;} case 1: w = 2.0*x / (factor*tap_count) + M_PI; y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); break; case 2: w = 2.0*x / (factor*tap_count*M_PI); y *= bessel(16*sqrt(FFMAX(1-w*w, 0))); break; } tab[i] = y; norm += y; } /* normalize so that an uniform color remains the same */ for(i=0;i<tap_count;i++) { v = clip(lrintf(tab[i] * scale / norm + e), -32768, 32767); filter[ph * tap_count + i] = v; e += tab[i] * scale / norm - v; } } } /** * initalizes a audio resampler. * note, if either rate is not a integer then simply scale both rates up so they are */ AVResampleContext *av_resample_init(int out_rate, int in_rate){ AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); double factor= FFMIN(out_rate / (double)in_rate, 1.0); memset(c, 0, sizeof(AVResampleContext)); c->filter_length= ceil(16.0/factor); c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short)); av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1); c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1; c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 2]= 1; c->src_incr= out_rate; c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT; c->index= -PHASE_COUNT*((c->filter_length-1)/2); return c; } void av_resample_close(AVResampleContext *c){ av_freep(&c->filter_bank); av_freep(&c); } void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){ // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr; c->compensation_distance= compensation_distance; c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; } /** * resamples. * @param src an array of unconsumed samples * @param consumed the number of samples of src which have been consumed are returned here * @param src_size the number of unconsumed samples available * @param dst_size the amount of space in samples available in dst * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context * @return the number of samples written in dst or -1 if an error occured */ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){ int dst_index, i; int index= c->index; int frac= c->frac; int dst_incr_frac= c->dst_incr % c->src_incr; int dst_incr= c->dst_incr / c->src_incr; if(c->compensation_distance && c->compensation_distance < dst_size) dst_size= c->compensation_distance; for(dst_index=0; dst_index < dst_size; dst_index++){ short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK); int sample_index= index >> PHASE_SHIFT; int val=0; if(sample_index < 0){ for(i=0; i<c->filter_length; i++) val += src[ABS(sample_index + i) % src_size] * filter[i]; }else if(sample_index + c->filter_length > src_size){ break; }else{ #if 0 int64_t v=0; int sub_phase= (frac<<12) / c->src_incr; for(i=0; i<c->filter_length; i++){ int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase; v += src[sample_index + i] * coeff; } val= v>>12; #else for(i=0; i<c->filter_length; i++){ val += src[sample_index + i] * filter[i]; } #endif } val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; frac += dst_incr_frac; index += dst_incr; if(frac >= c->src_incr){ frac -= c->src_incr; index++; } } *consumed= FFMAX(index, 0) >> PHASE_SHIFT; index= FFMIN(index, 0); if(update_ctx){ if(c->compensation_distance){ c->compensation_distance -= dst_index; if(!c->compensation_distance) c->dst_incr= c->ideal_dst_incr; } c->frac= frac; c->index= index; } #if 0 if(update_ctx && !c->compensation_distance){ #undef rand av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2); av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance); } #endif return dst_index; }