/* * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> * * 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 */ #include "libavutil/common.h"#include "libavutil/libm.h"#include "libavutil/log.h"#include "internal.h"#include "resample.h"#include "audio_data.h"  /* double template */#define CONFIG_RESAMPLE_DBL#include "resample_template.c"#undef CONFIG_RESAMPLE_DBL /* float template */#define CONFIG_RESAMPLE_FLT#include "resample_template.c"#undef CONFIG_RESAMPLE_FLT /* s32 template */#define CONFIG_RESAMPLE_S32#include "resample_template.c"#undef CONFIG_RESAMPLE_S32 /* s16 template */#include "resample_template.c"  /* 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;     x = x * x / 4;    for (i = 1; v != lastv; i++) {        lastv = v;        t    *= x / (i * i);        v    += t;    }    return v;} /* Build a polyphase filterbank. */static int build_filter(ResampleContext *c, double factor){    int ph, i;    double x, y, w;    double *tab;    int tap_count    = c->filter_length;    int phase_count  = 1 << c->phase_shift;    const int center = (tap_count - 1) / 2;     tab = av_malloc(tap_count * sizeof(*tab));    if (!tab)        return AVERROR(ENOMEM);     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 (c->filter_type) {            case AV_RESAMPLE_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 AV_RESAMPLE_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 AV_RESAMPLE_FILTER_TYPE_KAISER:                w  = 2.0 * x / (factor * tap_count * M_PI);                y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));                break;            }             tab[i] = y;            norm  += y;        }        /* normalize so that an uniform color remains the same */        for (i = 0; i < tap_count; i++)            tab[i] = tab[i] / norm;         c->set_filter(c->filter_bank, tab, ph, tap_count);    }     av_free(tab);    return 0;} ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr){    ResampleContext *c;    int out_rate    = avr->out_sample_rate;    int in_rate     = avr->in_sample_rate;    double factor   = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);    int phase_count = 1 << avr->phase_shift;    int felem_size;     if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&        avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&        avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&        avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {        av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "               "resampling: %s\n",               av_get_sample_fmt_name(avr->internal_sample_fmt));        return NULL;    }    c = av_mallocz(sizeof(*c));    if (!c)        return NULL;     c->avr           = avr;    c->phase_shift   = avr->phase_shift;    c->phase_mask    = phase_count - 1;    c->linear        = avr->linear_interp;    c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);    c->filter_type   = avr->filter_type;    c->kaiser_beta   = avr->kaiser_beta;     switch (avr->internal_sample_fmt) {    case AV_SAMPLE_FMT_DBLP:        c->resample_one  = c->linear ? resample_linear_dbl : resample_one_dbl;        c->resample_nearest = resample_nearest_dbl;        c->set_filter    = set_filter_dbl;        break;    case AV_SAMPLE_FMT_FLTP:        c->resample_one  = c->linear ? resample_linear_flt : resample_one_flt;        c->resample_nearest = resample_nearest_flt;        c->set_filter    = set_filter_flt;        break;    case AV_SAMPLE_FMT_S32P:        c->resample_one  = c->linear ? resample_linear_s32 : resample_one_s32;        c->resample_nearest = resample_nearest_s32;        c->set_filter    = set_filter_s32;        break;    case AV_SAMPLE_FMT_S16P:        c->resample_one  = c->linear ? resample_linear_s16 : resample_one_s16;        c->resample_nearest = resample_nearest_s16;        c->set_filter    = set_filter_s16;        break;    }     if (ARCH_AARCH64)        ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);    if (ARCH_ARM)        ff_audio_resample_init_arm(c, avr->internal_sample_fmt);     felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);    c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);    if (!c->filter_bank)        goto error;     if (build_filter(c, factor) < 0)        goto error;     memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],           c->filter_bank, (c->filter_length - 1) * felem_size);    memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],           &c->filter_bank[(c->filter_length - 1) * felem_size], 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->padding_size   = (c->filter_length - 1) / 2;    c->initial_padding_filled = 0;    c->index = 0;    c->frac  = 0;     /* allocate internal buffer */    c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,                                    avr->internal_sample_fmt,                                    "resample buffer");    if (!c->buffer)        goto error;    c->buffer->nb_samples      = c->padding_size;    c->initial_padding_samples = c->padding_size;     av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",           av_get_sample_fmt_name(avr->internal_sample_fmt),           avr->in_sample_rate, avr->out_sample_rate);     return c; error:    ff_audio_data_free(&c->buffer);    av_free(c->filter_bank);    av_free(c);    return NULL;} void ff_audio_resample_free(ResampleContext **c){    if (!*c)        return;    ff_audio_data_free(&(*c)->buffer);    av_free((*c)->filter_bank);    av_freep(c);} int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,                                int compensation_distance){    ResampleContext *c;     if (compensation_distance < 0)        return AVERROR(EINVAL);    if (!compensation_distance && sample_delta)        return AVERROR(EINVAL);     if (!avr->resample_needed) {        av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");        return AVERROR(EINVAL);    }    c = avr->resample;    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;} static int resample(ResampleContext *c, void *dst, const void *src,                    int *consumed, int src_size, int dst_size, int update_ctx,                    int nearest_neighbour){    int dst_index;    unsigned int index = c->index;    int frac          = c->frac;    int dst_incr_frac = c->dst_incr % c->src_incr;    int dst_incr      = c->dst_incr / c->src_incr;    int compensation_distance = c->compensation_distance;     if (!dst != !src)        return AVERROR(EINVAL);     if (nearest_neighbour) {        uint64_t index2 = ((uint64_t)index) << 32;        int64_t incr   = (1LL << 32) * c->dst_incr / c->src_incr;        dst_size       = FFMIN(dst_size,                               (src_size-1-index) * (int64_t)c->src_incr /                               c->dst_incr);         if (dst) {            for(dst_index = 0; dst_index < dst_size; dst_index++) {                c->resample_nearest(dst, dst_index, src, index2 >> 32);                index2 += incr;            }        } else {            dst_index = dst_size;        }        index += dst_index * dst_incr;        index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;        frac   = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;    } else {        for (dst_index = 0; dst_index < dst_size; dst_index++) {            int sample_index = index >> c->phase_shift;             if (sample_index + c->filter_length > src_size)                break;             if (dst)                c->resample_one(c, dst, dst_index, src, index, frac);             frac  += dst_incr_frac;            index += dst_incr;            if (frac >= c->src_incr) {                frac -= c->src_incr;                index++;            }            if (dst_index + 1 == compensation_distance) {                compensation_distance = 0;                dst_incr_frac = c->ideal_dst_incr % c->src_incr;                dst_incr      = c->ideal_dst_incr / c->src_incr;            }        }    }    if (consumed)        *consumed = index >> c->phase_shift;     if (update_ctx) {        index &= c->phase_mask;         if (compensation_distance) {            compensation_distance -= dst_index;            if (compensation_distance <= 0)                return AVERROR_BUG;        }        c->frac     = frac;        c->index    = index;        c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;        c->compensation_distance = compensation_distance;    }     return dst_index;} int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src){    int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;    int ret = AVERROR(EINVAL);    int nearest_neighbour = (c->compensation_distance == 0 &&                             c->filter_length == 1 &&                             c->phase_shift == 0);     in_samples  = src ? src->nb_samples : 0;    in_leftover = c->buffer->nb_samples;     /* add input samples to the internal buffer */    if (src) {        ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);        if (ret < 0)            return ret;    } else if (in_leftover <= c->final_padding_samples) {        /* no remaining samples to flush */        return 0;    }     if (!c->initial_padding_filled) {        int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);        int i;         if (src && c->buffer->nb_samples < 2 * c->padding_size)            return 0;         for (i = 0; i < c->padding_size; i++)            for (ch = 0; ch < c->buffer->channels; ch++) {                if (c->buffer->nb_samples > 2 * c->padding_size - i) {                    memcpy(c->buffer->data[ch] + bps * i,                           c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);                } else {                    memset(c->buffer->data[ch] + bps * i, 0, bps);                }            }        c->initial_padding_filled = 1;    }     if (!src && !c->final_padding_filled) {        int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);        int i;         ret = ff_audio_data_realloc(c->buffer,                                    FFMAX(in_samples, in_leftover) +                                    c->padding_size);        if (ret < 0) {            av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");            return AVERROR(ENOMEM);        }         for (i = 0; i < c->padding_size; i++)            for (ch = 0; ch < c->buffer->channels; ch++) {                if (in_leftover > i) {                    memcpy(c->buffer->data[ch] + bps * (in_leftover + i),                           c->buffer->data[ch] + bps * (in_leftover - i - 1),                           bps);                } else {                    memset(c->buffer->data[ch] + bps * (in_leftover + i),                           0, bps);                }            }        c->buffer->nb_samples   += c->padding_size;        c->final_padding_samples = c->padding_size;        c->final_padding_filled  = 1;    }      /* calculate output size and reallocate output buffer if needed */    /* TODO: try to calculate this without the dummy resample() run */    if (!dst->read_only && dst->allow_realloc) {        out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,                               INT_MAX, 0, nearest_neighbour);        ret = ff_audio_data_realloc(dst, out_samples);        if (ret < 0) {            av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");            return ret;        }    }     /* resample each channel plane */    for (ch = 0; ch < c->buffer->channels; ch++) {        out_samples = resample(c, (void *)dst->data[ch],                               (const void *)c->buffer->data[ch], &consumed,                               c->buffer->nb_samples, dst->allocated_samples,                               ch + 1 == c->buffer->channels, nearest_neighbour);    }    if (out_samples < 0) {        av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");        return out_samples;    }     /* drain consumed samples from the internal buffer */    ff_audio_data_drain(c->buffer, consumed);    c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);     av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",            in_samples, in_leftover, out_samples, c->buffer->nb_samples);     dst->nb_samples = out_samples;    return 0;} int avresample_get_delay(AVAudioResampleContext *avr){    ResampleContext *c = avr->resample;     if (!avr->resample_needed || !avr->resample)        return 0;     return FFMAX(c->buffer->nb_samples - c->padding_size, 0);}