/* * 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 * audio resampling * @author Michael Niedermayer <michaelni@gmx.at> */ #include "libavutil/log.h" #include "swresample_internal.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 ResampleContext { const AVClass *av_class; 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; double factor; } ResampleContext; /** * 0th order modified bessel function of the first kind. */ static double bessel(double x){ double v=1; double lastv=0; double t=1; int i; static const double inv[100]={ 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10), 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20), 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30), 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40), 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50), 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60), 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70), 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80), 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90), 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000) }; x= x*x/4; for(i=0; v != lastv; i++){ lastv=v; t *= x*inv[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 * @return 0 on success, negative on error */ static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){ int ph, i; double x, y, w; double *tab = av_malloc(tap_count * sizeof(*tab)); const int center= (tap_count-1)/2; if (!tab) return AVERROR(ENOMEM); /* 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 av_free(tab); return 0; } ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){ double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); int phase_count= 1<<phase_shift; if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) { c = av_mallocz(sizeof(*c)); if (!c) return NULL; c->phase_shift = phase_shift; c->phase_mask = phase_count - 1; c->linear = linear; c->factor = factor; c->filter_length = FFMAX((int)ceil(filter_size/factor), 1); c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM)); if (!c->filter_bank) goto error; if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE)) goto error; 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->compensation_distance= 0; if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2)) goto error; c->ideal_dst_incr= c->dst_incr; c->index= -phase_count*((c->filter_length-1)/2); c->frac= 0; return c; error: av_free(c->filter_bank); av_free(c); return NULL; } void swri_resample_free(ResampleContext **c){ if(!*c) return; av_freep(&(*c)->filter_bank); av_freep(c); } int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ ResampleContext *c; int ret; if (!s || compensation_distance < 0) return AVERROR(EINVAL); if (!compensation_distance && sample_delta) return AVERROR(EINVAL); if (!s->resample) { s->flags |= SWR_FLAG_RESAMPLE; ret = swr_init(s); if (ret < 0) return ret; } c= s->resample; c->compensation_distance= compensation_distance; if (compensation_distance) c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; else c->dst_incr = c->ideal_dst_incr; return 0; } int swri_resample(ResampleContext *c, int16_t *dst, const int16_t *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; } index += dst_index * dst_incr; index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr; frac = (frac + dst_index * (int64_t)dst_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 + c->filter_length > src_size || -sample_index >= src_size){ break; }else if(sample_index < 0){ for(i=0; i<c->filter_length; i++) val += src[FFABS(sample_index + i)] * filter[i]; }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_int16(lrintf(val)); #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; } int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ int i, ret= -1; for(i=0; i<dst->ch_count; i++){ ret= swri_resample(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); } return ret; }