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/*

 * audio resampling

 * Copyright (c) 2004-2012 Michael Niedermayer 

 *

 * 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 

 */

#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

        double norm = 0;

        for(i=0;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

                ((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

                ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));

            break;

        case AV_SAMPLE_FMT_FLTP:

            for(i=0;i

                ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;

            break;

        case AV_SAMPLE_FMT_DBLP:

            for(i=0;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

            double ss=0, sf=0, ff=0;

            for(j=0; j

                sine[j]= cos(i*j*M_PI/LEN);

            for(j=0; j

                double sum=0;

                ph=0;

                for(k=0; k

                    sum += filter[ph * tap_count + k] * sine[k+j];

                filtered[j]= sum / (1<

                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<

    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<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; ich_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; ich_count; i++){

        for(j=0; jin_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,

};