| /* |
| * audio resampling |
| * Copyright (c) 2004-2012 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 "libavutil/avassert.h" |
| #include "swresample_internal.h" |
| |
| #define WINDOW_TYPE 9 |
| |
| |
| |
| typedef struct ResampleContext { |
| const AVClass *av_class; |
| uint8_t *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; |
| enum AVSampleFormat format; |
| int felem_size; |
| int filter_shift; |
| } 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(ResampleContext *c, void *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 */ |
| switch(c->format){ |
| case AV_SAMPLE_FMT_S16P: |
| for(i=0;i<tap_count;i++) |
| ((int16_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX); |
| break; |
| case AV_SAMPLE_FMT_S32P: |
| for(i=0;i<tap_count;i++) |
| ((int32_t*)filter)[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX); |
| break; |
| case AV_SAMPLE_FMT_FLTP: |
| for(i=0;i<tap_count;i++) |
| ((float*)filter)[ph * tap_count + i] = tab[i] * scale / norm; |
| break; |
| case AV_SAMPLE_FMT_DBLP: |
| for(i=0;i<tap_count;i++) |
| ((double*)filter)[ph * tap_count + i] = tab[i] * scale / norm; |
| break; |
| } |
| } |
| #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, enum AVSampleFormat format){ |
| 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->format != format) { |
| c = av_mallocz(sizeof(*c)); |
| if (!c) |
| return NULL; |
| |
| c->format= format; |
| |
| c->felem_size= av_get_bytes_per_sample(c->format); |
| |
| switch(c->format){ |
| case AV_SAMPLE_FMT_S16P: |
| c->filter_shift = 15; |
| break; |
| case AV_SAMPLE_FMT_S32P: |
| c->filter_shift = 30; |
| break; |
| case AV_SAMPLE_FMT_FLTP: |
| case AV_SAMPLE_FMT_DBLP: |
| c->filter_shift = 0; |
| break; |
| default: |
| av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n"); |
| 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)*c->felem_size); |
| if (!c->filter_bank) |
| goto error; |
| if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, phase_count, 1<<c->filter_shift, WINDOW_TYPE)) |
| goto error; |
| memcpy(c->filter_bank + (c->filter_length*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_length-1)*c->felem_size); |
| memcpy(c->filter_bank + (c->filter_length*phase_count )*c->felem_size, c->filter_bank + (c->filter_length - 1)*c->felem_size, c->felem_size); |
| } |
| |
| 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; |
| } |
| |
| #define RENAME(N) N ## _int16 |
| #define FILTER_SHIFT 15 |
| #define DELEM int16_t |
| #define FELEM int16_t |
| #define FELEM2 int32_t |
| #define FELEML int64_t |
| #define FELEM_MAX INT16_MAX |
| #define FELEM_MIN INT16_MIN |
| #define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\ |
| d = (unsigned)(v + 32768) > 65535 ? (v>>31) ^ 32767 : v |
| #include "resample_template.c" |
| |
| #undef RENAME |
| #undef FELEM |
| #undef FELEM2 |
| #undef DELEM |
| #undef FELEML |
| #undef OUT |
| #undef FELEM_MIN |
| #undef FELEM_MAX |
| #undef FILTER_SHIFT |
| |
| |
| #define RENAME(N) N ## _int32 |
| #define FILTER_SHIFT 30 |
| #define DELEM int32_t |
| #define FELEM int32_t |
| #define FELEM2 int64_t |
| #define FELEML int64_t |
| #define FELEM_MAX INT32_MAX |
| #define FELEM_MIN INT32_MIN |
| #define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\ |
| d = (uint64_t)(v + 0x80000000) > 0xFFFFFFFF ? (v>>63) ^ 0x7FFFFFFF : v |
| #include "resample_template.c" |
| |
| #undef RENAME |
| #undef FELEM |
| #undef FELEM2 |
| #undef DELEM |
| #undef FELEML |
| #undef OUT |
| #undef FELEM_MIN |
| #undef FELEM_MAX |
| #undef FILTER_SHIFT |
| |
| |
| #define RENAME(N) N ## _float |
| #define FILTER_SHIFT 0 |
| #define DELEM float |
| #define FELEM float |
| #define FELEM2 float |
| #define FELEML float |
| #define OUT(d, v) d = v |
| #include "resample_template.c" |
| |
| #undef RENAME |
| #undef FELEM |
| #undef FELEM2 |
| #undef DELEM |
| #undef FELEML |
| #undef OUT |
| #undef FELEM_MIN |
| #undef FELEM_MAX |
| #undef FILTER_SHIFT |
| |
| |
| #define RENAME(N) N ## _double |
| #define FILTER_SHIFT 0 |
| #define DELEM double |
| #define FELEM double |
| #define FELEM2 double |
| #define FELEML double |
| #define OUT(d, v) d = v |
| #include "resample_template.c" |
| |
| |
| 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++){ |
| 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); |
| 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); |
| 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); |
| 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); |
| } |
| |
| return ret; |
| } |
| |
| int64_t swr_get_delay(struct SwrContext *s, int64_t base){ |
| ResampleContext *c = s->resample; |
| if(c){ |
| int64_t num = s->in_buffer_count - (c->filter_length-1)/2; |
| num <<= c->phase_shift; |
| num -= c->index; |
| num *= c->src_incr; |
| num -= c->frac; |
| |
| return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift); |
| }else{ |
| return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; |
| } |
| } |
