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/*

 * Copyright (c) 2004 Michael Niedermayer 

 * Copyright (c) 2012 Justin Ruggles 

 *

 * 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"

struct ResampleContext {

    AVAudioResampleContext *avr;

    AudioData *buffer;

    uint8_t *filter_bank;

    int filter_length;

    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 AVResampleFilterType filter_type;

    int kaiser_beta;

    double factor;

    void (*set_filter)(void *filter, double *tab, int phase, int tap_count);

    void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,

                         int dst_index, const void *src0, int src_size,

                         int index, int frac);

};

/* 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)

{

    int ph, i;

    double x, y, w, factor;

    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);

    /* if upsampling, only need to interpolate, no filter */

    factor = FFMIN(c->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 (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->factor        = factor;

    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  = resample_one_dbl;

        c->set_filter    = set_filter_dbl;

        break;

    case AV_SAMPLE_FMT_FLTP:

        c->resample_one  = resample_one_flt;

        c->set_filter    = set_filter_flt;

        break;

    case AV_SAMPLE_FMT_S32P:

        c->resample_one  = resample_one_s32;

        c->set_filter    = set_filter_s32;

        break;

    case AV_SAMPLE_FMT_S16P:

        c->resample_one  = resample_one_s16;

        c->set_filter    = set_filter_s16;

        break;

    }

    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) < 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->index = -phase_count * ((c->filter_length - 1) / 2);

    c->frac  = 0;

    /* allocate internal buffer */

    c->buffer = ff_audio_data_alloc(avr->resample_channels, 0,

                                    avr->internal_sample_fmt,

                                    "resample buffer");

    if (!c->buffer)

        goto error;

    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;

    AudioData *fifo_buf = NULL;

    int ret = 0;

    if (compensation_distance < 0)

        return AVERROR(EINVAL);

    if (!compensation_distance && sample_delta)

        return AVERROR(EINVAL);

    if (!avr->resample_needed) {

#if FF_API_RESAMPLE_CLOSE_OPEN

        /* if resampling was not enabled previously, re-initialize the

           AVAudioResampleContext and force resampling */

        int fifo_samples;

        int restore_matrix = 0;

        double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };

        /* buffer any remaining samples in the output FIFO before closing */

        fifo_samples = av_audio_fifo_size(avr->out_fifo);

        if (fifo_samples > 0) {

            fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,

                                           avr->out_sample_fmt, NULL);

            if (!fifo_buf)

                return AVERROR(EINVAL);

            ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,

                                               fifo_samples);

            if (ret < 0)

                goto reinit_fail;

        }

        /* save the channel mixing matrix */

        if (avr->am) {

            ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);

            if (ret < 0)

                goto reinit_fail;

            restore_matrix = 1;

        }

        /* close the AVAudioResampleContext */

        avresample_close(avr);

        avr->force_resampling = 1;

        /* restore the channel mixing matrix */

        if (restore_matrix) {

            ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);

            if (ret < 0)

                goto reinit_fail;

        }

        /* re-open the AVAudioResampleContext */

        ret = avresample_open(avr);

        if (ret < 0)

            goto reinit_fail;

        /* restore buffered samples to the output FIFO */

        if (fifo_samples > 0) {

            ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,

                                            fifo_samples);

            if (ret < 0)

                goto reinit_fail;

            ff_audio_data_free(&fifo_buf);

        }

#else

        av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");

        return AVERROR(EINVAL);

#endif

    }

    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;

reinit_fail:

    ff_audio_data_free(&fifo_buf);

    return ret;

}

static int resample(ResampleContext *c, void *dst, const void *src,

                    int *consumed, int src_size, int dst_size, int update_ctx)

{

    int dst_index;

    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 (compensation_distance == 0 && c->filter_length == 1 &&

        c->phase_shift == 0) {

        int64_t index2 = ((int64_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_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);

                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 ||

                -sample_index >= src_size)

                break;

            if (dst)

                c->resample_one(c, 0, dst, dst_index, src, src_size, 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 = FFMAX(index, 0) >> c->phase_shift;

    if (update_ctx) {

        if (index >= 0)

            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);

    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) {

        /* no remaining samples to flush */

        return 0;

    } else {

        /* TODO: pad buffer to flush completely */

    }

    /* 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);

        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);

    }

    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);

    av_dlog(c->avr, "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)

{

    if (!avr->resample_needed || !avr->resample)

        return 0;

    return avr->resample->buffer->nb_samples;

}