Mercurial > libavcodec.hg
view resample2.c @ 5479:943c732c905d libavcodec
use dsputil for float to signed 16-bit sample conversion
author | jbr |
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date | Sat, 04 Aug 2007 20:59:22 +0000 |
parents | 2b72f9bc4f06 |
children | c2ab2ac31edb |
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/* * audio resampling * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> * * This file is part of FFmpeg. * * FFmpeg 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 2.1 of the License, or (at your option) any later version. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file resample2.c * audio resampling * @author Michael Niedermayer <michaelni@gmx.at> */ #include "avcodec.h" #include "dsputil.h" #ifndef CONFIG_RESAMPLE_HP #define FILTER_SHIFT 15 #define FELEM int16_t #define FELEM2 int32_t #define FELEML int64_t #define FELEM_MAX INT16_MAX #define FELEM_MIN INT16_MIN #define WINDOW_TYPE 9 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE) #define FILTER_SHIFT 30 #define FELEM int32_t #define FELEM2 int64_t #define FELEML int64_t #define FELEM_MAX INT32_MAX #define FELEM_MIN INT32_MIN #define WINDOW_TYPE 12 #else #define FILTER_SHIFT 0 #define FELEM double #define FELEM2 double #define FELEML double #define WINDOW_TYPE 24 #endif typedef struct AVResampleContext{ FELEM *filter_bank; int filter_length; int ideal_dst_incr; int dst_incr; int index; int frac; int src_incr; int compensation_distance; int phase_shift; int phase_mask; int linear; }AVResampleContext; /** * 0th order modified bessel function of the first kind. */ static double bessel(double x){ double v=1; double t=1; int i; x= x*x/4; for(i=1; i<50; i++){ t *= x/(i*i); v += 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..16->kaiser windowed sinc beta=2..16 */ void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ int ph, i; 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; 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; default: w = 2.0*x / (factor*tap_count*M_PI); y *= bessel(type*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++) { #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE filter[ph * tap_count + i] = tab[i] / norm; #else filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX); #endif } } #if 0 { #define LEN 1024 int j,k; double sine[LEN + tap_count]; double filtered[LEN]; double maxff=-2, minff=2, maxsf=-2, minsf=2; for(i=0; i<LEN; i++){ double ss=0, sf=0, ff=0; for(j=0; j<LEN+tap_count; j++) sine[j]= cos(i*j*M_PI/LEN); for(j=0; j<LEN; j++){ double sum=0; ph=0; for(k=0; k<tap_count; k++) sum += filter[ph * tap_count + k] * sine[k+j]; filtered[j]= sum / (1<<FILTER_SHIFT); ss+= sine[j + center] * sine[j + center]; ff+= filtered[j] * filtered[j]; sf+= sine[j + center] * filtered[j]; } ss= sqrt(2*ss/LEN); ff= sqrt(2*ff/LEN); sf= 2*sf/LEN; maxff= FFMAX(maxff, ff); minff= FFMIN(minff, ff); maxsf= FFMAX(maxsf, sf); minsf= FFMIN(minsf, sf); if(i%11==0){ av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf); minff=minsf= 2; maxff=maxsf= -2; } } } #endif } /** * Initializes an audio resampler. * Note, if either rate is not an integer then simply scale both rates up so they are. */ AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ AVResampleContext *c= av_mallocz(sizeof(AVResampleContext)); double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); int phase_count= 1<<phase_shift; c->phase_shift= phase_shift; c->phase_mask= phase_count-1; c->linear= linear; c->filter_length= FFMAX((int)ceil(filter_size/factor), 1); c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE); memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM)); c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 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); } /** * Compensates samplerate/timestamp drift. The compensation is done by changing * the resampler parameters, so no audible clicks or similar distortions ocur * @param compensation_distance distance in output samples over which the compensation should be performed * @param sample_delta number of output samples which should be output less * * example: av_resample_compensate(c, 10, 500) * here instead of 510 samples only 500 samples would be output * * note, due to rounding the actual compensation might be slightly different, * especially if the compensation_distance is large and the in_rate used during init is small */ 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; int compensation_distance= c->compensation_distance; if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ int64_t index2= ((int64_t)index)<<32; int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr); for(dst_index=0; dst_index < dst_size; dst_index++){ dst[dst_index] = src[index2>>32]; index2 += incr; } frac += dst_index * dst_incr_frac; index += dst_index * dst_incr; index += frac / c->src_incr; frac %= c->src_incr; }else{ for(dst_index=0; dst_index < dst_size; dst_index++){ FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask); int sample_index= index >> c->phase_shift; FELEM2 val=0; if(sample_index < 0){ for(i=0; i<c->filter_length; i++) val += src[FFABS(sample_index + i) % src_size] * filter[i]; }else if(sample_index + c->filter_length > src_size){ break; }else if(c->linear){ FELEM2 v2=0; for(i=0; i<c->filter_length; i++){ val += src[sample_index + i] * (FELEM2)filter[i]; v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; } val+=(v2-val)*(FELEML)frac / c->src_incr; }else{ for(i=0; i<c->filter_length; i++){ val += src[sample_index + i] * (FELEM2)filter[i]; } } #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE dst[dst_index] = av_clip(lrintf(val), -32768, 32767); #else val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT; dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val; #endif frac += dst_incr_frac; index += dst_incr; if(frac >= c->src_incr){ frac -= c->src_incr; index++; } if(dst_index + 1 == compensation_distance){ compensation_distance= 0; dst_incr_frac= c->ideal_dst_incr % c->src_incr; dst_incr= c->ideal_dst_incr / c->src_incr; } } } *consumed= FFMAX(index, 0) >> c->phase_shift; if(index>=0) index &= c->phase_mask; if(compensation_distance){ compensation_distance -= dst_index; assert(compensation_distance > 0); } if(update_ctx){ c->frac= frac; c->index= index; c->dst_incr= dst_incr_frac + c->src_incr*dst_incr; c->compensation_distance= compensation_distance; } #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; }