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Diffstat (limited to 'libswresample/resample.c')
| -rw-r--r-- | libswresample/resample.c | 372 |
1 files changed, 372 insertions, 0 deletions
diff --git a/libswresample/resample.c b/libswresample/resample.c new file mode 100644 index 0000000000..fb9da7c354 --- /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, +}; |