/* * audio resampling * Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at> * bessel function: Copyright (c) 2006 Xiaogang Zhang * * 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/avassert.h"#include "libavutil/mem.h"#include "resample.h" /** * 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, double kaiser_beta){    int ph, i;    int ph_nb = phase_count % 2 ? phase_count : phase_count / 2 + 1;    double x, y, w, t, s;    double *tab = av_malloc_array(tap_count+1,  sizeof(*tab));    double *sin_lut = av_malloc_array(ph_nb, sizeof(*sin_lut));    const int center= (tap_count-1)/2;    double norm = 0;    int ret = AVERROR(ENOMEM);     if (!tab || !sin_lut)        goto fail;     av_assert0(tap_count == 1 || tap_count % 2 == 0);     /* if upsampling, only need to interpolate, no filter */    if (factor > 1.0)        factor = 1.0;     if (factor == 1.0) {        for (ph = 0; ph < ph_nb; ph++)            sin_lut[ph] = sin(M_PI * ph / phase_count) * (center & 1 ? 1 : -1);    }    for(ph = 0; ph < ph_nb; ph++) {        s = sin_lut[ph];        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 if (factor == 1.0)                y = s / x;            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);                t = -cos(w);                y *= 0.3635819 - 0.4891775 * t + 0.1365995 * (2*t*t-1) - 0.0106411 * (4*t*t*t - 3*t);                break;            case SWR_FILTER_TYPE_KAISER:                w = 2.0*x / (factor*tap_count*M_PI);                y *= av_bessel_i0(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));                break;            default:                av_assert0(0);            }             tab[i] = y;            s = -s;            if (!ph)                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_int16(lrintf(tab[i] * scale / norm));            if (phase_count % 2) break;            for (i = 0; i < tap_count; i++)                ((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i];            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));            if (phase_count % 2) break;            for (i = 0; i < tap_count; i++)                ((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i];            break;        case AV_SAMPLE_FMT_FLTP:            for(i=0;i<tap_count;i++)                ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;            if (phase_count % 2) break;            for (i = 0; i < tap_count; i++)                ((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i];            break;        case AV_SAMPLE_FMT_DBLP:            for(i=0;i<tap_count;i++)                ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;            if (phase_count % 2) break;            for (i = 0; i < tap_count; i++)                ((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i];            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     ret = 0;fail:    av_free(tab);    av_free(sin_lut);    return ret;} static void resample_free(ResampleContext **cc){    ResampleContext *c = *cc;    if(!c)        return;    av_freep(&c->filter_bank);    av_freep(cc);} 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, double kaiser_beta,                                    double precision, int cheby, int exact_rational){    double cutoff = cutoff0? cutoff0 : 0.97;    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);    int phase_count= 1<<phase_shift;    int phase_count_compensation = phase_count;    int filter_length = FFMAX((int)ceil(filter_size/factor), 1);     if (filter_length > 1)        filter_length = FFALIGN(filter_length, 2);     if (exact_rational) {        int phase_count_exact, phase_count_exact_den;         av_reduce(&phase_count_exact, &phase_count_exact_den, out_rate, in_rate, INT_MAX);        if (phase_count_exact <= phase_count) {            phase_count_compensation = phase_count_exact * (phase_count / phase_count_exact);            phase_count = phase_count_exact;        }    }     if (!c || c->phase_count != phase_count || c->linear!=linear || c->factor != factor           || c->filter_length != filter_length || c->format != format           || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {        resample_free(&c);        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);        }         if (filter_size/factor > INT32_MAX/256) {            av_log(NULL, AV_LOG_ERROR, "Filter length too large\n");            goto error;        }         c->phase_count   = phase_count;        c->linear        = linear;        c->factor        = factor;        c->filter_length = filter_length;        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;        c->phase_count_compensation = phase_count_compensation;        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;    while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {        c->dst_incr *= 2;        c->src_incr *= 2;    }    c->ideal_dst_incr = c->dst_incr;    c->dst_incr_div   = c->dst_incr / c->src_incr;    c->dst_incr_mod   = c->dst_incr % c->src_incr;     c->index= -phase_count*((c->filter_length-1)/2);    c->frac= 0;     swri_resample_dsp_init(c);     return c;error:    av_freep(&c->filter_bank);    av_free(c);    return NULL;} static int rebuild_filter_bank_with_compensation(ResampleContext *c){    uint8_t *new_filter_bank;    int new_src_incr, new_dst_incr;    int phase_count = c->phase_count_compensation;    int ret;     if (phase_count == c->phase_count)        return 0;     av_assert0(!c->frac && !c->dst_incr_mod);     new_filter_bank = av_calloc(c->filter_alloc, (phase_count + 1) * c->felem_size);    if (!new_filter_bank)        return AVERROR(ENOMEM);     ret = build_filter(c, new_filter_bank, c->factor, c->filter_length, c->filter_alloc,                       phase_count, 1 << c->filter_shift, c->filter_type, c->kaiser_beta);    if (ret < 0) {        av_freep(&new_filter_bank);        return ret;    }    memcpy(new_filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, new_filter_bank, (c->filter_alloc-1)*c->felem_size);    memcpy(new_filter_bank + (c->filter_alloc*phase_count  )*c->felem_size, new_filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);     if (!av_reduce(&new_src_incr, &new_dst_incr, c->src_incr,                   c->dst_incr * (int64_t)(phase_count/c->phase_count), INT32_MAX/2))    {        av_freep(&new_filter_bank);        return AVERROR(EINVAL);    }     c->src_incr = new_src_incr;    c->dst_incr = new_dst_incr;    while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {        c->dst_incr *= 2;        c->src_incr *= 2;    }    c->ideal_dst_incr = c->dst_incr;    c->dst_incr_div   = c->dst_incr / c->src_incr;    c->dst_incr_mod   = c->dst_incr % c->src_incr;    c->index         *= phase_count / c->phase_count;    c->phase_count    = phase_count;    av_freep(&c->filter_bank);    c->filter_bank = new_filter_bank;    return 0;} static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){    int ret;     if (compensation_distance && sample_delta) {        ret = rebuild_filter_bank_with_compensation(c);        if (ret < 0)            return ret;    }     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;     c->dst_incr_div   = c->dst_incr / c->src_incr;    c->dst_incr_mod   = c->dst_incr % c->src_incr;     return 0;} static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){    int i;    int64_t max_src_size = (INT64_MAX/2 / c->phase_count) / c->src_incr;     if (c->compensation_distance)        dst_size = FFMIN(dst_size, c->compensation_distance);    src_size = FFMIN(src_size, max_src_size);     *consumed = 0;     if (c->filter_length == 1 && c->phase_count == 1) {        int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*c->index + 1;        int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr + 1;        int new_size = (src_size * (int64_t)c->src_incr - c->frac + c->dst_incr - 1) / c->dst_incr;         dst_size = FFMAX(FFMIN(dst_size, new_size), 0);        if (dst_size > 0) {            for (i = 0; i < dst->ch_count; i++) {                c->dsp.resample_one(dst->ch[i], src->ch[i], dst_size, index2, incr);                if (i+1 == dst->ch_count) {                    c->index += dst_size * c->dst_incr_div;                    c->index += (c->frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr;                    av_assert2(c->index >= 0);                    *consumed = c->index;                    c->frac   = (c->frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr;                    c->index = 0;                }            }        }    } else {        int64_t end_index = (1LL + src_size - c->filter_length) * c->phase_count;        int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;        int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;        int (*resample_func)(struct ResampleContext *c, void *dst,                             const void *src, int n, int update_ctx);         dst_size = FFMAX(FFMIN(dst_size, delta_n), 0);        if (dst_size > 0) {            /* resample_linear and resample_common should have same behavior             * when frac and dst_incr_mod are zero */            resample_func = (c->linear && (c->frac || c->dst_incr_mod)) ?                            c->dsp.resample_linear : c->dsp.resample_common;            for (i = 0; i < dst->ch_count; i++)                *consumed = resample_func(c, dst->ch[i], src->ch[i], dst_size, i+1 == dst->ch_count);        }    }     if (c->compensation_distance) {        c->compensation_distance -= dst_size;        if (!c->compensation_distance) {            c->dst_incr     = c->ideal_dst_incr;            c->dst_incr_div = c->dst_incr / c->src_incr;            c->dst_incr_mod = c->dst_incr % c->src_incr;        }    }     return dst_size;} 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_count;    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_count);} static int64_t get_out_samples(struct SwrContext *s, int in_samples) {    ResampleContext *c = s->resample;    // The + 2 are added to allow implementations to be slightly inaccurate, they should not be needed currently.    // They also make it easier to proof that changes and optimizations do not    // break the upper bound.    int64_t num = s->in_buffer_count + 2LL + in_samples;    num *= c->phase_count;    num -= c->index;    num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) * c->phase_count, AV_ROUND_UP) + 2;     if (c->compensation_distance) {        if (num > INT_MAX)            return AVERROR(EINVAL);         num = FFMAX(num, (num * c->ideal_dst_incr - 1) / c->dst_incr + 1);    }    return num;} static int resample_flush(struct SwrContext *s) {    ResampleContext *c = s->resample;    AudioData *a= &s->in_buffer;    int i, j, ret;    int reflection = (FFMIN(s->in_buffer_count, c->filter_length) + 1) / 2;     if((ret = swri_realloc_audio(a, s->in_buffer_index + s->in_buffer_count + reflection)) < 0)        return ret;    av_assert0(a->planar);    for(i=0; i<a->ch_count; i++){        for(j=0; j<reflection; 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 += reflection;    return 0;} // in fact the whole handle multiple ridiculously small buffers might need more thinking...static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src,                                 int in_count, int *out_idx, int *out_sz){    int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res;     if (c->index >= 0)        return 0;     if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0)        return res;     // copy    for (n = *out_sz; n < num; n++) {        for (ch = 0; ch < src->ch_count; ch++) {            memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size),                   src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size);        }    }     // if not enough data is in, return and wait for more    if (num < c->filter_length + 1) {        *out_sz = num;        *out_idx = c->filter_length;        return INT_MAX;    }     // else invert    for (n = 1; n <= c->filter_length; n++) {        for (ch = 0; ch < src->ch_count; ch++) {            memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size),                   dst->ch[ch] + ((c->filter_length + n) * c->felem_size),                   c->felem_size);        }    }     res = num - *out_sz;    *out_idx = c->filter_length;    while (c->index < 0) {        --*out_idx;        c->index += c->phase_count;    }    *out_sz = FFMAX(*out_sz + c->filter_length,                    1 + c->filter_length * 2) - *out_idx; 

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