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  1. /*
  2.  * audio resampling
  3.  * Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. /**
  23.  * @file
  24.  * audio resampling
  25.  * @author Michael Niedermayer <michaelni@gmx.at>
  26.  */
  27.  
  28. #include "libavutil/log.h"
  29. #include "libavutil/avassert.h"
  30. #include "swresample_internal.h"
  31.  
  32.  
  33. typedef struct ResampleContext {
  34.     const AVClass *av_class;
  35.     uint8_t *filter_bank;
  36.     int filter_length;
  37.     int filter_alloc;
  38.     int ideal_dst_incr;
  39.     int dst_incr;
  40.     int index;
  41.     int frac;
  42.     int src_incr;
  43.     int compensation_distance;
  44.     int phase_shift;
  45.     int phase_mask;
  46.     int linear;
  47.     enum SwrFilterType filter_type;
  48.     int kaiser_beta;
  49.     double factor;
  50.     enum AVSampleFormat format;
  51.     int felem_size;
  52.     int filter_shift;
  53. } ResampleContext;
  54.  
  55. /**
  56.  * 0th order modified bessel function of the first kind.
  57.  */
  58. static double bessel(double x){
  59.     double v=1;
  60.     double lastv=0;
  61.     double t=1;
  62.     int i;
  63.     static const double inv[100]={
  64.  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),
  65.  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),
  66.  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),
  67.  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),
  68.  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),
  69.  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),
  70.  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),
  71.  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),
  72.  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),
  73.  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)
  74.     };
  75.  
  76.     x= x*x/4;
  77.     for(i=0; v != lastv; i++){
  78.         lastv=v;
  79.         t *= x*inv[i];
  80.         v += t;
  81.         av_assert2(i<99);
  82.     }
  83.     return v;
  84. }
  85.  
  86. /**
  87.  * builds a polyphase filterbank.
  88.  * @param factor resampling factor
  89.  * @param scale wanted sum of coefficients for each filter
  90.  * @param filter_type  filter type
  91.  * @param kaiser_beta  kaiser window beta
  92.  * @return 0 on success, negative on error
  93.  */
  94. static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
  95.                         int filter_type, int kaiser_beta){
  96.     int ph, i;
  97.     double x, y, w;
  98.     double *tab = av_malloc(tap_count * sizeof(*tab));
  99.     const int center= (tap_count-1)/2;
  100.  
  101.     if (!tab)
  102.         return AVERROR(ENOMEM);
  103.  
  104.     /* if upsampling, only need to interpolate, no filter */
  105.     if (factor > 1.0)
  106.         factor = 1.0;
  107.  
  108.     for(ph=0;ph<phase_count;ph++) {
  109.         double norm = 0;
  110.         for(i=0;i<tap_count;i++) {
  111.             x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
  112.             if (x == 0) y = 1.0;
  113.             else        y = sin(x) / x;
  114.             switch(filter_type){
  115.             case SWR_FILTER_TYPE_CUBIC:{
  116.                 const float d= -0.5; //first order derivative = -0.5
  117.                 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
  118.                 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*(            -x*x + x*x*x);
  119.                 else      y=                       d*(-4 + 8*x - 5*x*x + x*x*x);
  120.                 break;}
  121.             case SWR_FILTER_TYPE_BLACKMAN_NUTTALL:
  122.                 w = 2.0*x / (factor*tap_count) + M_PI;
  123.                 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
  124.                 break;
  125.             case SWR_FILTER_TYPE_KAISER:
  126.                 w = 2.0*x / (factor*tap_count*M_PI);
  127.                 y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
  128.                 break;
  129.             default:
  130.                 av_assert0(0);
  131.             }
  132.  
  133.             tab[i] = y;
  134.             norm += y;
  135.         }
  136.  
  137.         /* normalize so that an uniform color remains the same */
  138.         switch(c->format){
  139.         case AV_SAMPLE_FMT_S16P:
  140.             for(i=0;i<tap_count;i++)
  141.                 ((int16_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX);
  142.             break;
  143.         case AV_SAMPLE_FMT_S32P:
  144.             for(i=0;i<tap_count;i++)
  145.                 ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));
  146.             break;
  147.         case AV_SAMPLE_FMT_FLTP:
  148.             for(i=0;i<tap_count;i++)
  149.                 ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
  150.             break;
  151.         case AV_SAMPLE_FMT_DBLP:
  152.             for(i=0;i<tap_count;i++)
  153.                 ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
  154.             break;
  155.         }
  156.     }
  157. #if 0
  158.     {
  159. #define LEN 1024
  160.         int j,k;
  161.         double sine[LEN + tap_count];
  162.         double filtered[LEN];
  163.         double maxff=-2, minff=2, maxsf=-2, minsf=2;
  164.         for(i=0; i<LEN; i++){
  165.             double ss=0, sf=0, ff=0;
  166.             for(j=0; j<LEN+tap_count; j++)
  167.                 sine[j]= cos(i*j*M_PI/LEN);
  168.             for(j=0; j<LEN; j++){
  169.                 double sum=0;
  170.                 ph=0;
  171.                 for(k=0; k<tap_count; k++)
  172.                     sum += filter[ph * tap_count + k] * sine[k+j];
  173.                 filtered[j]= sum / (1<<FILTER_SHIFT);
  174.                 ss+= sine[j + center] * sine[j + center];
  175.                 ff+= filtered[j] * filtered[j];
  176.                 sf+= sine[j + center] * filtered[j];
  177.             }
  178.             ss= sqrt(2*ss/LEN);
  179.             ff= sqrt(2*ff/LEN);
  180.             sf= 2*sf/LEN;
  181.             maxff= FFMAX(maxff, ff);
  182.             minff= FFMIN(minff, ff);
  183.             maxsf= FFMAX(maxsf, sf);
  184.             minsf= FFMIN(minsf, sf);
  185.             if(i%11==0){
  186.                 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);
  187.                 minff=minsf= 2;
  188.                 maxff=maxsf= -2;
  189.             }
  190.         }
  191.     }
  192. #endif
  193.  
  194.     av_free(tab);
  195.     return 0;
  196. }
  197.  
  198. static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
  199.                                     double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta,
  200.                                     double precision, int cheby){
  201.     double cutoff = cutoff0? cutoff0 : 0.97;
  202.     double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
  203.     int phase_count= 1<<phase_shift;
  204.  
  205.     if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
  206.            || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format
  207.            || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
  208.         c = av_mallocz(sizeof(*c));
  209.         if (!c)
  210.             return NULL;
  211.  
  212.         c->format= format;
  213.  
  214.         c->felem_size= av_get_bytes_per_sample(c->format);
  215.  
  216.         switch(c->format){
  217.         case AV_SAMPLE_FMT_S16P:
  218.             c->filter_shift = 15;
  219.             break;
  220.         case AV_SAMPLE_FMT_S32P:
  221.             c->filter_shift = 30;
  222.             break;
  223.         case AV_SAMPLE_FMT_FLTP:
  224.         case AV_SAMPLE_FMT_DBLP:
  225.             c->filter_shift = 0;
  226.             break;
  227.         default:
  228.             av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
  229.             av_assert0(0);
  230.         }
  231.  
  232.         c->phase_shift   = phase_shift;
  233.         c->phase_mask    = phase_count - 1;
  234.         c->linear        = linear;
  235.         c->factor        = factor;
  236.         c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
  237.         c->filter_alloc  = FFALIGN(c->filter_length, 8);
  238.         c->filter_bank   = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size);
  239.         c->filter_type   = filter_type;
  240.         c->kaiser_beta   = kaiser_beta;
  241.         if (!c->filter_bank)
  242.             goto error;
  243.         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))
  244.             goto error;
  245.         memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);
  246.         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);
  247.     }
  248.  
  249.     c->compensation_distance= 0;
  250.     if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
  251.         goto error;
  252.     c->ideal_dst_incr= c->dst_incr;
  253.  
  254.     c->index= -phase_count*((c->filter_length-1)/2);
  255.     c->frac= 0;
  256.  
  257.     return c;
  258. error:
  259.     av_freep(&c->filter_bank);
  260.     av_free(c);
  261.     return NULL;
  262. }
  263.  
  264. static void resample_free(ResampleContext **c){
  265.     if(!*c)
  266.         return;
  267.     av_freep(&(*c)->filter_bank);
  268.     av_freep(c);
  269. }
  270.  
  271. static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){
  272.     c->compensation_distance= compensation_distance;
  273.     if (compensation_distance)
  274.         c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
  275.     else
  276.         c->dst_incr = c->ideal_dst_incr;
  277.     return 0;
  278. }
  279.  
  280. #define TEMPLATE_RESAMPLE_S16
  281. #include "resample_template.c"
  282. #undef TEMPLATE_RESAMPLE_S16
  283.  
  284. #define TEMPLATE_RESAMPLE_S32
  285. #include "resample_template.c"
  286. #undef TEMPLATE_RESAMPLE_S32
  287.  
  288. #define TEMPLATE_RESAMPLE_FLT
  289. #include "resample_template.c"
  290. #undef TEMPLATE_RESAMPLE_FLT
  291.  
  292. #define TEMPLATE_RESAMPLE_DBL
  293. #include "resample_template.c"
  294. #undef TEMPLATE_RESAMPLE_DBL
  295.  
  296. // XXX FIXME the whole C loop should be written in asm so this x86 specific code here isnt needed
  297. #if HAVE_MMXEXT_INLINE
  298.  
  299. #include "x86/resample_mmx.h"
  300.  
  301. #define TEMPLATE_RESAMPLE_S16_MMX2
  302. #include "resample_template.c"
  303. #undef TEMPLATE_RESAMPLE_S16_MMX2
  304.  
  305. #if HAVE_SSSE3_INLINE
  306. #define TEMPLATE_RESAMPLE_S16_SSSE3
  307. #include "resample_template.c"
  308. #undef TEMPLATE_RESAMPLE_S16_SSSE3
  309. #endif
  310.  
  311. #endif // HAVE_MMXEXT_INLINE
  312.  
  313. static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
  314.     int i, ret= -1;
  315.     int av_unused mm_flags = av_get_cpu_flags();
  316.     int need_emms= 0;
  317.  
  318.     for(i=0; i<dst->ch_count; i++){
  319. #if HAVE_MMXEXT_INLINE
  320. #if HAVE_SSSE3_INLINE
  321.              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);
  322.         else
  323. #endif
  324.              if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_MMX2 )){
  325.                  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);
  326.                  need_emms= 1;
  327.              } else
  328. #endif
  329.              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);
  330.         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);
  331.         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);
  332.         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);
  333.     }
  334.     if(need_emms)
  335.         emms_c();
  336.     return ret;
  337. }
  338.  
  339. static int64_t get_delay(struct SwrContext *s, int64_t base){
  340.     ResampleContext *c = s->resample;
  341.     int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
  342.     num <<= c->phase_shift;
  343.     num -= c->index;
  344.     num *= c->src_incr;
  345.     num -= c->frac;
  346.     return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift);
  347. }
  348.  
  349. static int resample_flush(struct SwrContext *s) {
  350.     AudioData *a= &s->in_buffer;
  351.     int i, j, ret;
  352.     if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
  353.         return ret;
  354.     av_assert0(a->planar);
  355.     for(i=0; i<a->ch_count; i++){
  356.         for(j=0; j<s->in_buffer_count; j++){
  357.             memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j  )*a->bps,
  358.                 a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
  359.         }
  360.     }
  361.     s->in_buffer_count += (s->in_buffer_count+1)/2;
  362.     return 0;
  363. }
  364.  
  365. struct Resampler const swri_resampler={
  366.   resample_init,
  367.   resample_free,
  368.   multiple_resample,
  369.   resample_flush,
  370.   set_compensation,
  371.   get_delay,
  372. };
  373.