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-rw-r--r--libswresample/resample.c372
1 files changed, 372 insertions, 0 deletions
diff --git a/libswresample/resample.c b/libswresample/resample.c
new file mode 100644
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--- /dev/null
+++ b/libswresample/resample.c
@@ -0,0 +1,372 @@
+/*
+ * 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_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; 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,
+};