chasing indexing error

1 files changed, 0 insertions, 372 deletions

diff --git a/ffmpeg/libswresample/resample.c b/ffmpeg/libswresample/resample.c
deleted file mode 100644
index 8b1b6ca..0000000
--- a/ffmpeg/libswresample/resample.c
+++ /dev/null
@@ -1,372 +0,0 @@
-/*
- * 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_clipl_int32(llrint(tab[i] * scale / norm));
-            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;
-}
-
-static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
-                                    double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta,
-                                    double precision, int cheby){
-    double cutoff = cutoff0? cutoff0 : 0.97;
-    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");
-            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<<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_freep(&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; 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;
-}
-
-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; i<a->ch_count; i++){
-        for(j=0; j<s->in_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,
-};