/* * 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" typedef struct ResampleContext { const AVClass *av_class; uint8_t *filter_bank; int filter_length; int filter_alloc; 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; enum SwrFilterType filter_type; int kaiser_beta; 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; av_assert2(i<99); } return v; } /** * builds a polyphase filterbank. * @param factor resampling factor * @param scale wanted sum of coefficients for each filter * @param filter_type filter type * @param kaiser_beta kaiser window beta * @return 0 on success, negative on error */ static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale, int filter_type, int kaiser_beta){ 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(filter_type){ case SWR_FILTER_TYPE_CUBIC:{ 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 SWR_FILTER_TYPE_BLACKMAN_NUTTALL: 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 SWR_FILTER_TYPE_KAISER: w = 2.0*x / (factor*tap_count*M_PI); y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0))); break; default: av_assert0(0); } 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 * alloc + 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 * alloc + 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 * alloc + i] = tab[i] * scale / norm; break; case AV_SAMPLE_FMT_DBLP: for(i=0;i<tap_count;i++) ((double*)filter)[ph * alloc + 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, enum SwrFilterType filter_type, int kaiser_beta){ 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->filter_type != filter_type || c->kaiser_beta != kaiser_beta) { 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_alloc = FFALIGN(c->filter_length, 8); c->filter_bank = av_mallocz(c->filter_alloc*(phase_count+1)*c->felem_size); c->filter_type = filter_type; c->kaiser_beta = kaiser_beta; if (!c->filter_bank) goto error; 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)) goto error; memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size); 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); } 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" #undef RENAME #undef FELEM #undef FELEM2 #undef DELEM #undef FELEML #undef OUT #undef FELEM_MIN #undef FELEM_MAX #undef FILTER_SHIFT // XXX FIXME the whole C loop should be written in asm so this x86 specific code here isnt needed #if HAVE_MMXEXT_INLINE #include "x86/resample_mmx.h" #define COMMON_CORE COMMON_CORE_INT16_MMX2 #define RENAME(N) N ## _int16_mmx2 #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 COMMON_CORE #undef RENAME #undef FELEM #undef FELEM2 #undef DELEM #undef FELEML #undef OUT #undef FELEM_MIN #undef FELEM_MAX #undef FILTER_SHIFT #if HAVE_SSSE3_INLINE #define COMMON_CORE COMMON_CORE_INT16_SSSE3 #define RENAME(N) N ## _int16_ssse3 #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" #endif #endif // HAVE_MMXEXT_INLINE int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ int i, ret= -1; int av_unused mm_flags = av_get_cpu_flags(); int need_emms= 0; for(i=0; i<dst->ch_count; i++){ #if HAVE_MMXEXT_INLINE #if HAVE_SSSE3_INLINE 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); else #endif if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_MMX2 )){ 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); need_emms= 1; } else #endif 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); 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); 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); 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); } if(need_emms) emms_c(); 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; } }