From 741fb4b9e135cfb161a749db88713229038577bb Mon Sep 17 00:00:00 2001 From: Tim Redfern Date: Thu, 5 Sep 2013 17:55:35 +0100 Subject: making act segmenter --- ffmpeg1/libswresample/resample.c | 372 --------------------------------------- 1 file changed, 372 deletions(-) delete mode 100644 ffmpeg1/libswresample/resample.c (limited to 'ffmpeg1/libswresample/resample.c') diff --git a/ffmpeg1/libswresample/resample.c b/ffmpeg1/libswresample/resample.c deleted file mode 100644 index fb9da7c..0000000 --- a/ffmpeg1/libswresample/resample.c +++ /dev/null @@ -1,372 +0,0 @@ -/* - * audio resampling - * Copyright (c) 2004-2012 Michael Niedermayer - * - * 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 - */ - -#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;phformat){ - case AV_SAMPLE_FMT_S16P: - for(i=0;iphase_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"); - av_assert0(0); - } - - 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_calloc(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<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; -} - -static void resample_free(ResampleContext **c){ - if(!*c) - return; - av_freep(&(*c)->filter_bank); - av_freep(c); -} - -static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){ - 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 TEMPLATE_RESAMPLE_S16 -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_S16 - -#define TEMPLATE_RESAMPLE_S32 -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_S32 - -#define TEMPLATE_RESAMPLE_FLT -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_FLT - -#define TEMPLATE_RESAMPLE_DBL -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_DBL - -// 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 TEMPLATE_RESAMPLE_S16_MMX2 -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_S16_MMX2 - -#if HAVE_SSSE3_INLINE -#define TEMPLATE_RESAMPLE_S16_SSSE3 -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_S16_SSSE3 -#endif - -#endif // HAVE_MMXEXT_INLINE - -static int 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; ich_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; -} - -static int64_t get_delay(struct SwrContext *s, int64_t base){ - ResampleContext *c = s->resample; - 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); -} - -static int resample_flush(struct SwrContext *s) { - AudioData *a= &s->in_buffer; - int i, j, ret; - if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0) - return ret; - av_assert0(a->planar); - for(i=0; ich_count; i++){ - for(j=0; jin_buffer_count; j++){ - memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps, - a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps); - } - } - s->in_buffer_count += (s->in_buffer_count+1)/2; - return 0; -} - -struct Resampler const swri_resampler={ - resample_init, - resample_free, - multiple_resample, - resample_flush, - set_compensation, - get_delay, -}; -- cgit v1.2.3