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372 lines
14 KiB
372 lines
14 KiB
/* |
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* audio resampling |
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* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at> |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/** |
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* @file |
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* audio resampling |
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* @author Michael Niedermayer <michaelni@gmx.at> |
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*/ |
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#include "libavutil/log.h" |
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#include "libavutil/avassert.h" |
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#include "swresample_internal.h" |
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typedef struct ResampleContext { |
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const AVClass *av_class; |
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uint8_t *filter_bank; |
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int filter_length; |
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int filter_alloc; |
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int ideal_dst_incr; |
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int dst_incr; |
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int index; |
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int frac; |
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int src_incr; |
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int compensation_distance; |
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int phase_shift; |
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int phase_mask; |
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int linear; |
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enum SwrFilterType filter_type; |
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int kaiser_beta; |
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double factor; |
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enum AVSampleFormat format; |
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int felem_size; |
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int filter_shift; |
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} ResampleContext; |
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/** |
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* 0th order modified bessel function of the first kind. |
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*/ |
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static double bessel(double x){ |
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double v=1; |
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double lastv=0; |
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double t=1; |
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int i; |
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static const double inv[100]={ |
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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), |
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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), |
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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), |
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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), |
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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), |
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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), |
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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), |
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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), |
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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), |
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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) |
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}; |
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x= x*x/4; |
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for(i=0; v != lastv; i++){ |
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lastv=v; |
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t *= x*inv[i]; |
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v += t; |
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av_assert2(i<99); |
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} |
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return v; |
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} |
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/** |
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* builds a polyphase filterbank. |
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* @param factor resampling factor |
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* @param scale wanted sum of coefficients for each filter |
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* @param filter_type filter type |
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* @param kaiser_beta kaiser window beta |
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* @return 0 on success, negative on error |
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*/ |
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static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale, |
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int filter_type, int kaiser_beta){ |
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int ph, i; |
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double x, y, w; |
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double *tab = av_malloc(tap_count * sizeof(*tab)); |
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const int center= (tap_count-1)/2; |
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if (!tab) |
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return AVERROR(ENOMEM); |
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/* if upsampling, only need to interpolate, no filter */ |
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if (factor > 1.0) |
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factor = 1.0; |
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for(ph=0;ph<phase_count;ph++) { |
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double norm = 0; |
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for(i=0;i<tap_count;i++) { |
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x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; |
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if (x == 0) y = 1.0; |
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else y = sin(x) / x; |
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switch(filter_type){ |
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case SWR_FILTER_TYPE_CUBIC:{ |
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const float d= -0.5; //first order derivative = -0.5 |
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x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); |
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if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); |
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else y= d*(-4 + 8*x - 5*x*x + x*x*x); |
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break;} |
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case SWR_FILTER_TYPE_BLACKMAN_NUTTALL: |
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w = 2.0*x / (factor*tap_count) + M_PI; |
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y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); |
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break; |
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case SWR_FILTER_TYPE_KAISER: |
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w = 2.0*x / (factor*tap_count*M_PI); |
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y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0))); |
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break; |
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default: |
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av_assert0(0); |
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} |
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tab[i] = y; |
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norm += y; |
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} |
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/* normalize so that an uniform color remains the same */ |
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switch(c->format){ |
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case AV_SAMPLE_FMT_S16P: |
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for(i=0;i<tap_count;i++) |
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((int16_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX); |
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break; |
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case AV_SAMPLE_FMT_S32P: |
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for(i=0;i<tap_count;i++) |
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((int32_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX); |
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break; |
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case AV_SAMPLE_FMT_FLTP: |
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for(i=0;i<tap_count;i++) |
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((float*)filter)[ph * alloc + i] = tab[i] * scale / norm; |
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break; |
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case AV_SAMPLE_FMT_DBLP: |
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for(i=0;i<tap_count;i++) |
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((double*)filter)[ph * alloc + i] = tab[i] * scale / norm; |
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break; |
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} |
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} |
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#if 0 |
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{ |
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#define LEN 1024 |
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int j,k; |
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double sine[LEN + tap_count]; |
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double filtered[LEN]; |
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double maxff=-2, minff=2, maxsf=-2, minsf=2; |
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for(i=0; i<LEN; i++){ |
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double ss=0, sf=0, ff=0; |
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for(j=0; j<LEN+tap_count; j++) |
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sine[j]= cos(i*j*M_PI/LEN); |
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for(j=0; j<LEN; j++){ |
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double sum=0; |
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ph=0; |
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for(k=0; k<tap_count; k++) |
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sum += filter[ph * tap_count + k] * sine[k+j]; |
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filtered[j]= sum / (1<<FILTER_SHIFT); |
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ss+= sine[j + center] * sine[j + center]; |
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ff+= filtered[j] * filtered[j]; |
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sf+= sine[j + center] * filtered[j]; |
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} |
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ss= sqrt(2*ss/LEN); |
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ff= sqrt(2*ff/LEN); |
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sf= 2*sf/LEN; |
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maxff= FFMAX(maxff, ff); |
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minff= FFMIN(minff, ff); |
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maxsf= FFMAX(maxsf, sf); |
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minsf= FFMIN(minsf, sf); |
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if(i%11==0){ |
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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); |
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minff=minsf= 2; |
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maxff=maxsf= -2; |
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} |
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} |
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} |
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#endif |
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av_free(tab); |
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return 0; |
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} |
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static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, |
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double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, |
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double precision, int cheby){ |
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double cutoff = cutoff0? cutoff0 : 0.97; |
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double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); |
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int phase_count= 1<<phase_shift; |
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if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor |
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|| c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format |
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|| c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) { |
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c = av_mallocz(sizeof(*c)); |
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if (!c) |
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return NULL; |
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c->format= format; |
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c->felem_size= av_get_bytes_per_sample(c->format); |
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switch(c->format){ |
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case AV_SAMPLE_FMT_S16P: |
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c->filter_shift = 15; |
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break; |
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case AV_SAMPLE_FMT_S32P: |
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c->filter_shift = 30; |
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break; |
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case AV_SAMPLE_FMT_FLTP: |
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case AV_SAMPLE_FMT_DBLP: |
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c->filter_shift = 0; |
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break; |
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default: |
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av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n"); |
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av_assert0(0); |
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} |
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c->phase_shift = phase_shift; |
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c->phase_mask = phase_count - 1; |
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c->linear = linear; |
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c->factor = factor; |
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c->filter_length = FFMAX((int)ceil(filter_size/factor), 1); |
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c->filter_alloc = FFALIGN(c->filter_length, 8); |
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c->filter_bank = av_mallocz(c->filter_alloc*(phase_count+1)*c->felem_size); |
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c->filter_type = filter_type; |
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c->kaiser_beta = kaiser_beta; |
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if (!c->filter_bank) |
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goto error; |
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if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta)) |
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goto error; |
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memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size); |
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memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size); |
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} |
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c->compensation_distance= 0; |
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if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2)) |
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goto error; |
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c->ideal_dst_incr= c->dst_incr; |
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c->index= -phase_count*((c->filter_length-1)/2); |
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c->frac= 0; |
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return c; |
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error: |
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av_free(c->filter_bank); |
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av_free(c); |
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return NULL; |
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} |
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static void resample_free(ResampleContext **c){ |
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if(!*c) |
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return; |
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av_freep(&(*c)->filter_bank); |
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av_freep(c); |
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} |
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static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){ |
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c->compensation_distance= compensation_distance; |
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if (compensation_distance) |
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c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; |
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else |
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c->dst_incr = c->ideal_dst_incr; |
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return 0; |
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} |
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#define TEMPLATE_RESAMPLE_S16 |
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#include "resample_template.c" |
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#undef TEMPLATE_RESAMPLE_S16 |
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#define TEMPLATE_RESAMPLE_S32 |
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#include "resample_template.c" |
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#undef TEMPLATE_RESAMPLE_S32 |
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#define TEMPLATE_RESAMPLE_FLT |
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#include "resample_template.c" |
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#undef TEMPLATE_RESAMPLE_FLT |
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#define TEMPLATE_RESAMPLE_DBL |
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#include "resample_template.c" |
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#undef TEMPLATE_RESAMPLE_DBL |
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// XXX FIXME the whole C loop should be written in asm so this x86 specific code here isnt needed |
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#if HAVE_MMXEXT_INLINE |
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#include "x86/resample_mmx.h" |
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#define TEMPLATE_RESAMPLE_S16_MMX2 |
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#include "resample_template.c" |
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#undef TEMPLATE_RESAMPLE_S16_MMX2 |
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#if HAVE_SSSE3_INLINE |
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#define TEMPLATE_RESAMPLE_S16_SSSE3 |
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#include "resample_template.c" |
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#undef TEMPLATE_RESAMPLE_S16_SSSE3 |
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#endif |
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#endif // HAVE_MMXEXT_INLINE |
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static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ |
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int i, ret= -1; |
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int av_unused mm_flags = av_get_cpu_flags(); |
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int need_emms= 0; |
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for(i=0; i<dst->ch_count; i++){ |
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#if HAVE_MMXEXT_INLINE |
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#if HAVE_SSSE3_INLINE |
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if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_SSSE3)) ret= swri_resample_int16_ssse3(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); |
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else |
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#endif |
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if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_MMX2 )){ |
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ret= swri_resample_int16_mmx2 (c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); |
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need_emms= 1; |
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} else |
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#endif |
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if(c->format == AV_SAMPLE_FMT_S16P) ret= swri_resample_int16(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); |
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else if(c->format == AV_SAMPLE_FMT_S32P) ret= swri_resample_int32(c, (int32_t*)dst->ch[i], (const int32_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); |
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else if(c->format == AV_SAMPLE_FMT_FLTP) ret= swri_resample_float(c, (float *)dst->ch[i], (const float *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); |
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else if(c->format == AV_SAMPLE_FMT_DBLP) ret= swri_resample_double(c,(double *)dst->ch[i], (const double *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); |
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} |
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if(need_emms) |
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emms_c(); |
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return ret; |
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} |
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static int64_t get_delay(struct SwrContext *s, int64_t base){ |
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ResampleContext *c = s->resample; |
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int64_t num = s->in_buffer_count - (c->filter_length-1)/2; |
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num <<= c->phase_shift; |
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num -= c->index; |
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num *= c->src_incr; |
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num -= c->frac; |
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return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift); |
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} |
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static int resample_flush(struct SwrContext *s) { |
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AudioData *a= &s->in_buffer; |
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int i, j, ret; |
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if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0) |
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return ret; |
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av_assert0(a->planar); |
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for(i=0; i<a->ch_count; i++){ |
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for(j=0; j<s->in_buffer_count; j++){ |
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memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps, |
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a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps); |
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} |
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} |
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s->in_buffer_count += (s->in_buffer_count+1)/2; |
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return 0; |
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} |
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struct Resampler const swri_resampler={ |
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resample_init, |
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resample_free, |
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multiple_resample, |
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resample_flush, |
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set_compensation, |
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get_delay, |
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};
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