/*
  * 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,
 };