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/*

 * Copyright (c) 2012 Pavel Koshevoy 

 *

 * 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

 * tempo scaling audio filter -- an implementation of WSOLA algorithm

 *

 * Based on MIT licensed yaeAudioTempoFilter.h and yaeAudioFragment.h

 * from Apprentice Video player by Pavel Koshevoy.

 * https://sourceforge.net/projects/apprenticevideo/

 *

 * An explanation of SOLA algorithm is available at

 * http://www.surina.net/article/time-and-pitch-scaling.html

 *

 * WSOLA is very similar to SOLA, only one major difference exists between

 * these algorithms.  SOLA shifts audio fragments along the output stream,

 * where as WSOLA shifts audio fragments along the input stream.

 *

 * The advantage of WSOLA algorithm is that the overlap region size is

 * always the same, therefore the blending function is constant and

 * can be precomputed.

 */

#include 

#include "libavcodec/avfft.h"

#include "libavutil/avassert.h"

#include "libavutil/avstring.h"

#include "libavutil/channel_layout.h"

#include "libavutil/eval.h"

#include "libavutil/opt.h"

#include "libavutil/samplefmt.h"

#include "avfilter.h"

#include "audio.h"

#include "internal.h"

/**

 * A fragment of audio waveform

 */

typedef struct {

    // index of the first sample of this fragment in the overall waveform;

    // 0: input sample position

    // 1: output sample position

    int64_t position[2];

    // original packed multi-channel samples:

    uint8_t *data;

    // number of samples in this fragment:

    int nsamples;

    // rDFT transform of the down-mixed mono fragment, used for

    // fast waveform alignment via correlation in frequency domain:

    FFTSample *xdat;

} AudioFragment;

/**

 * Filter state machine states

 */

typedef enum {

    YAE_LOAD_FRAGMENT,

    YAE_ADJUST_POSITION,

    YAE_RELOAD_FRAGMENT,

    YAE_OUTPUT_OVERLAP_ADD,

    YAE_FLUSH_OUTPUT,

} FilterState;

/**

 * Filter state machine

 */

typedef struct {

    const AVClass *class;

    // ring-buffer of input samples, necessary because some times

    // input fragment position may be adjusted backwards:

    uint8_t *buffer;

    // ring-buffer maximum capacity, expressed in sample rate time base:

    int ring;

    // ring-buffer house keeping:

    int size;

    int head;

    int tail;

    // 0: input sample position corresponding to the ring buffer tail

    // 1: output sample position

    int64_t position[2];

    // sample format:

    enum AVSampleFormat format;

    // number of channels:

    int channels;

    // row of bytes to skip from one sample to next, across multple channels;

    // stride = (number-of-channels * bits-per-sample-per-channel) / 8

    int stride;

    // fragment window size, power-of-two integer:

    int window;

    // Hann window coefficients, for feathering

    // (blending) the overlapping fragment region:

    float *hann;

    // tempo scaling factor:

    double tempo;

    // a snapshot of previous fragment input and output position values

    // captured when the tempo scale factor was set most recently:

    int64_t origin[2];

    // current/previous fragment ring-buffer:

    AudioFragment frag[2];

    // current fragment index:

    uint64_t nfrag;

    // current state:

    FilterState state;

    // for fast correlation calculation in frequency domain:

    RDFTContext *real_to_complex;

    RDFTContext *complex_to_real;

    FFTSample *correlation;

    // for managing AVFilterPad.request_frame and AVFilterPad.filter_frame

    AVFrame *dst_buffer;

    uint8_t *dst;

    uint8_t *dst_end;

    uint64_t nsamples_in;

    uint64_t nsamples_out;

} ATempoContext;

#define OFFSET(x) offsetof(ATempoContext, x)

static const AVOption atempo_options[] = {

    { "tempo", "set tempo scale factor",

      OFFSET(tempo), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0.5, 2.0,

      AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM },

    { NULL }

};

AVFILTER_DEFINE_CLASS(atempo);

inline static AudioFragment *yae_curr_frag(ATempoContext *atempo)

{

    return &atempo->frag[atempo->nfrag % 2];

}

inline static AudioFragment *yae_prev_frag(ATempoContext *atempo)

{

    return &atempo->frag[(atempo->nfrag + 1) % 2];

}

/**

 * Reset filter to initial state, do not deallocate existing local buffers.

 */

static void yae_clear(ATempoContext *atempo)

{

    atempo->size = 0;

    atempo->head = 0;

    atempo->tail = 0;

    atempo->nfrag = 0;

    atempo->state = YAE_LOAD_FRAGMENT;

    atempo->position[0] = 0;

    atempo->position[1] = 0;

    atempo->origin[0] = 0;

    atempo->origin[1] = 0;

    atempo->frag[0].position[0] = 0;

    atempo->frag[0].position[1] = 0;

    atempo->frag[0].nsamples    = 0;

    atempo->frag[1].position[0] = 0;

    atempo->frag[1].position[1] = 0;

    atempo->frag[1].nsamples    = 0;

    // shift left position of 1st fragment by half a window

    // so that no re-normalization would be required for

    // the left half of the 1st fragment:

    atempo->frag[0].position[0] = -(int64_t)(atempo->window / 2);

    atempo->frag[0].position[1] = -(int64_t)(atempo->window / 2);

    av_frame_free(&atempo->dst_buffer);

    atempo->dst     = NULL;

    atempo->dst_end = NULL;

    atempo->nsamples_in       = 0;

    atempo->nsamples_out      = 0;

}

/**

 * Reset filter to initial state and deallocate all buffers.

 */

static void yae_release_buffers(ATempoContext *atempo)

{

    yae_clear(atempo);

    av_freep(&atempo->frag[0].data);

    av_freep(&atempo->frag[1].data);

    av_freep(&atempo->frag[0].xdat);

    av_freep(&atempo->frag[1].xdat);

    av_freep(&atempo->buffer);

    av_freep(&atempo->hann);

    av_freep(&atempo->correlation);

    av_rdft_end(atempo->real_to_complex);

    atempo->real_to_complex = NULL;

    av_rdft_end(atempo->complex_to_real);

    atempo->complex_to_real = NULL;

}

/* av_realloc is not aligned enough; fortunately, the data does not need to

 * be preserved */

#define RE_MALLOC_OR_FAIL(field, field_size)                    \

    do {                                                        \

        av_freep(&field);                                       \

        field = av_malloc(field_size);                          \

        if (!field) {                                           \

            yae_release_buffers(atempo);                        \

            return AVERROR(ENOMEM);                             \

        }                                                       \

    } while (0)

/**

 * Prepare filter for processing audio data of given format,

 * sample rate and number of channels.

 */

static int yae_reset(ATempoContext *atempo,

                     enum AVSampleFormat format,

                     int sample_rate,

                     int channels)

{

    const int sample_size = av_get_bytes_per_sample(format);

    uint32_t nlevels  = 0;

    uint32_t pot;

    int i;

    atempo->format   = format;

    atempo->channels = channels;

    atempo->stride   = sample_size * channels;

    // pick a segment window size:

    atempo->window = sample_rate / 24;

    // adjust window size to be a power-of-two integer:

    nlevels = av_log2(atempo->window);

    pot = 1 << nlevels;

    av_assert0(pot <= atempo->window);

    if (pot < atempo->window) {

        atempo->window = pot * 2;

        nlevels++;

    }

    // initialize audio fragment buffers:

    RE_MALLOC_OR_FAIL(atempo->frag[0].data, atempo->window * atempo->stride);

    RE_MALLOC_OR_FAIL(atempo->frag[1].data, atempo->window * atempo->stride);

    RE_MALLOC_OR_FAIL(atempo->frag[0].xdat, atempo->window * sizeof(FFTComplex));

    RE_MALLOC_OR_FAIL(atempo->frag[1].xdat, atempo->window * sizeof(FFTComplex));

    // initialize rDFT contexts:

    av_rdft_end(atempo->real_to_complex);

    atempo->real_to_complex = NULL;

    av_rdft_end(atempo->complex_to_real);

    atempo->complex_to_real = NULL;

    atempo->real_to_complex = av_rdft_init(nlevels + 1, DFT_R2C);

    if (!atempo->real_to_complex) {

        yae_release_buffers(atempo);

        return AVERROR(ENOMEM);

    }

    atempo->complex_to_real = av_rdft_init(nlevels + 1, IDFT_C2R);

    if (!atempo->complex_to_real) {

        yae_release_buffers(atempo);

        return AVERROR(ENOMEM);

    }

    RE_MALLOC_OR_FAIL(atempo->correlation, atempo->window * sizeof(FFTComplex));

    atempo->ring = atempo->window * 3;

    RE_MALLOC_OR_FAIL(atempo->buffer, atempo->ring * atempo->stride);

    // initialize the Hann window function:

    RE_MALLOC_OR_FAIL(atempo->hann, atempo->window * sizeof(float));

    for (i = 0; i < atempo->window; i++) {

        double t = (double)i / (double)(atempo->window - 1);

        double h = 0.5 * (1.0 - cos(2.0 * M_PI * t));

        atempo->hann[i] = (float)h;

    }

    yae_clear(atempo);

    return 0;

}

static int yae_set_tempo(AVFilterContext *ctx, const char *arg_tempo)

{

    const AudioFragment *prev;

    ATempoContext *atempo = ctx->priv;

    char   *tail = NULL;

    double tempo = av_strtod(arg_tempo, &tail);

    if (tail && *tail) {

        av_log(ctx, AV_LOG_ERROR, "Invalid tempo value '%s'\n", arg_tempo);

        return AVERROR(EINVAL);

    }

    if (tempo < 0.5 || tempo > 2.0) {

        av_log(ctx, AV_LOG_ERROR, "Tempo value %f exceeds [0.5, 2.0] range\n",

               tempo);

        return AVERROR(EINVAL);

    }

    prev = yae_prev_frag(atempo);

    atempo->origin[0] = prev->position[0] + atempo->window / 2;

    atempo->origin[1] = prev->position[1] + atempo->window / 2;

    atempo->tempo = tempo;

    return 0;

}

/**

 * A helper macro for initializing complex data buffer with scalar data

 * of a given type.

 */

#define yae_init_xdat(scalar_type, scalar_max)                          \

    do {                                                                \

        const uint8_t *src_end = src +                                  \

            frag->nsamples * atempo->channels * sizeof(scalar_type);    \

                                                                        \

        FFTSample *xdat = frag->xdat;                                   \

        scalar_type tmp;                                                \

                                                                        \

        if (atempo->channels == 1) {                                    \

            for (; src < src_end; xdat++) {                             \

                tmp = *(const scalar_type *)src;                        \

                src += sizeof(scalar_type);                             \

                                                                        \

                *xdat = (FFTSample)tmp;                                 \

            }                                                           \

        } else {                                                        \

            FFTSample s, max, ti, si;                                   \

            int i;                                                      \

                                                                        \

            for (; src < src_end; xdat++) {                             \

                tmp = *(const scalar_type *)src;                        \

                src += sizeof(scalar_type);                             \

                                                                        \

                max = (FFTSample)tmp;                                   \

                s = FFMIN((FFTSample)scalar_max,                        \

                          (FFTSample)fabsf(max));                       \

                                                                        \

                for (i = 1; i < atempo->channels; i++) {                \

                    tmp = *(const scalar_type *)src;                    \

                    src += sizeof(scalar_type);                         \

                                                                        \

                    ti = (FFTSample)tmp;                                \

                    si = FFMIN((FFTSample)scalar_max,                   \

                               (FFTSample)fabsf(ti));                   \

                                                                        \

                    if (s < si) {                                       \

                        s   = si;                                       \

                        max = ti;                                       \

                    }                                                   \

                }                                                       \

                                                                        \

                *xdat = max;                                            \

            }                                                           \

        }                                                               \

    } while (0)

/**

 * Initialize complex data buffer of a given audio fragment

 * with down-mixed mono data of appropriate scalar type.

 */

static void yae_downmix(ATempoContext *atempo, AudioFragment *frag)

{

    // shortcuts:

    const uint8_t *src = frag->data;

    // init complex data buffer used for FFT and Correlation:

    memset(frag->xdat, 0, sizeof(FFTComplex) * atempo->window);

    if (atempo->format == AV_SAMPLE_FMT_U8) {

        yae_init_xdat(uint8_t, 127);

    } else if (atempo->format == AV_SAMPLE_FMT_S16) {

        yae_init_xdat(int16_t, 32767);

    } else if (atempo->format == AV_SAMPLE_FMT_S32) {

        yae_init_xdat(int, 2147483647);

    } else if (atempo->format == AV_SAMPLE_FMT_FLT) {

        yae_init_xdat(float, 1);

    } else if (atempo->format == AV_SAMPLE_FMT_DBL) {

        yae_init_xdat(double, 1);

    }

}

/**

 * Populate the internal data buffer on as-needed basis.

 *

 * @return

 *   0 if requested data was already available or was successfully loaded,

 *   AVERROR(EAGAIN) if more input data is required.

 */

static int yae_load_data(ATempoContext *atempo,

                         const uint8_t **src_ref,

                         const uint8_t *src_end,

                         int64_t stop_here)

{

    // shortcut:

    const uint8_t *src = *src_ref;

    const int read_size = stop_here - atempo->position[0];

    if (stop_here <= atempo->position[0]) {

        return 0;

    }

    // samples are not expected to be skipped:

    av_assert0(read_size <= atempo->ring);

    while (atempo->position[0] < stop_here && src < src_end) {

        int src_samples = (src_end - src) / atempo->stride;

        // load data piece-wise, in order to avoid complicating the logic:

        int nsamples = FFMIN(read_size, src_samples);

        int na;

        int nb;

        nsamples = FFMIN(nsamples, atempo->ring);

        na = FFMIN(nsamples, atempo->ring - atempo->tail);

        nb = FFMIN(nsamples - na, atempo->ring);

        if (na) {

            uint8_t *a = atempo->buffer + atempo->tail * atempo->stride;

            memcpy(a, src, na * atempo->stride);

            src += na * atempo->stride;

            atempo->position[0] += na;

            atempo->size = FFMIN(atempo->size + na, atempo->ring);

            atempo->tail = (atempo->tail + na) % atempo->ring;

            atempo->head =

                atempo->size < atempo->ring ?

                atempo->tail - atempo->size :

                atempo->tail;

        }

        if (nb) {

            uint8_t *b = atempo->buffer;

            memcpy(b, src, nb * atempo->stride);

            src += nb * atempo->stride;

            atempo->position[0] += nb;

            atempo->size = FFMIN(atempo->size + nb, atempo->ring);

            atempo->tail = (atempo->tail + nb) % atempo->ring;

            atempo->head =

                atempo->size < atempo->ring ?

                atempo->tail - atempo->size :

                atempo->tail;

        }

    }

    // pass back the updated source buffer pointer:

    *src_ref = src;

    // sanity check:

    av_assert0(atempo->position[0] <= stop_here);

    return atempo->position[0] == stop_here ? 0 : AVERROR(EAGAIN);

}

/**

 * Populate current audio fragment data buffer.

 *

 * @return

 *   0 when the fragment is ready,

 *   AVERROR(EAGAIN) if more input data is required.

 */

static int yae_load_frag(ATempoContext *atempo,

                         const uint8_t **src_ref,

                         const uint8_t *src_end)

{

    // shortcuts:

    AudioFragment *frag = yae_curr_frag(atempo);

    uint8_t *dst;

    int64_t missing, start, zeros;

    uint32_t nsamples;

    const uint8_t *a, *b;

    int i0, i1, n0, n1, na, nb;

    int64_t stop_here = frag->position[0] + atempo->window;

    if (src_ref && yae_load_data(atempo, src_ref, src_end, stop_here) != 0) {

        return AVERROR(EAGAIN);

    }

    // calculate the number of samples we don't have:

    missing =

        stop_here > atempo->position[0] ?

        stop_here - atempo->position[0] : 0;

    nsamples =

        missing < (int64_t)atempo->window ?

        (uint32_t)(atempo->window - missing) : 0;

    // setup the output buffer:

    frag->nsamples = nsamples;

    dst = frag->data;

    start = atempo->position[0] - atempo->size;

    zeros = 0;

    if (frag->position[0] < start) {

        // what we don't have we substitute with zeros:

        zeros = FFMIN(start - frag->position[0], (int64_t)nsamples);

        av_assert0(zeros != nsamples);

        memset(dst, 0, zeros * atempo->stride);

        dst += zeros * atempo->stride;

    }

    if (zeros == nsamples) {

        return 0;

    }

    // get the remaining data from the ring buffer:

    na = (atempo->head < atempo->tail ?

          atempo->tail - atempo->head :

          atempo->ring - atempo->head);

    nb = atempo->head < atempo->tail ? 0 : atempo->tail;

    // sanity check:

    av_assert0(nsamples <= zeros + na + nb);

    a = atempo->buffer + atempo->head * atempo->stride;

    b = atempo->buffer;

    i0 = frag->position[0] + zeros - start;

    i1 = i0 < na ? 0 : i0 - na;

    n0 = i0 < na ? FFMIN(na - i0, (int)(nsamples - zeros)) : 0;

    n1 = nsamples - zeros - n0;

    if (n0) {

        memcpy(dst, a + i0 * atempo->stride, n0 * atempo->stride);

        dst += n0 * atempo->stride;

    }

    if (n1) {

        memcpy(dst, b + i1 * atempo->stride, n1 * atempo->stride);

    }

    return 0;

}

/**

 * Prepare for loading next audio fragment.

 */

static void yae_advance_to_next_frag(ATempoContext *atempo)

{

    const double fragment_step = atempo->tempo * (double)(atempo->window / 2);

    const AudioFragment *prev;

    AudioFragment       *frag;

    atempo->nfrag++;

    prev = yae_prev_frag(atempo);

    frag = yae_curr_frag(atempo);

    frag->position[0] = prev->position[0] + (int64_t)fragment_step;

    frag->position[1] = prev->position[1] + atempo->window / 2;

    frag->nsamples    = 0;

}

/**

 * Calculate cross-correlation via rDFT.

 *

 * Multiply two vectors of complex numbers (result of real_to_complex rDFT)

 * and transform back via complex_to_real rDFT.

 */

static void yae_xcorr_via_rdft(FFTSample *xcorr,

                               RDFTContext *complex_to_real,

                               const FFTComplex *xa,

                               const FFTComplex *xb,

                               const int window)

{

    FFTComplex *xc = (FFTComplex *)xcorr;

    int i;

    // NOTE: first element requires special care -- Given Y = rDFT(X),

    // Im(Y[0]) and Im(Y[N/2]) are always zero, therefore av_rdft_calc

    // stores Re(Y[N/2]) in place of Im(Y[0]).

    xc->re = xa->re * xb->re;

    xc->im = xa->im * xb->im;

    xa++;

    xb++;

    xc++;

    for (i = 1; i < window; i++, xa++, xb++, xc++) {

        xc->re = (xa->re * xb->re + xa->im * xb->im);

        xc->im = (xa->im * xb->re - xa->re * xb->im);

    }

    // apply inverse rDFT:

    av_rdft_calc(complex_to_real, xcorr);

}

/**

 * Calculate alignment offset for given fragment

 * relative to the previous fragment.

 *

 * @return alignment offset of current fragment relative to previous.

 */

static int yae_align(AudioFragment *frag,

                     const AudioFragment *prev,

                     const int window,

                     const int delta_max,

                     const int drift,

                     FFTSample *correlation,

                     RDFTContext *complex_to_real)

{

    int       best_offset = -drift;

    FFTSample best_metric = -FLT_MAX;

    FFTSample *xcorr;

    int i0;

    int i1;

    int i;

    yae_xcorr_via_rdft(correlation,

                       complex_to_real,

                       (const FFTComplex *)prev->xdat,

                       (const FFTComplex *)frag->xdat,

                       window);

    // identify search window boundaries:

    i0 = FFMAX(window / 2 - delta_max - drift, 0);

    i0 = FFMIN(i0, window);

    i1 = FFMIN(window / 2 + delta_max - drift, window - window / 16);

    i1 = FFMAX(i1, 0);

    // identify cross-correlation peaks within search window:

    xcorr = correlation + i0;

    for (i = i0; i < i1; i++, xcorr++) {

        FFTSample metric = *xcorr;

        // normalize:

        FFTSample drifti = (FFTSample)(drift + i);

        metric *= drifti * (FFTSample)(i - i0) * (FFTSample)(i1 - i);

        if (metric > best_metric) {

            best_metric = metric;

            best_offset = i - window / 2;

        }

    }

    return best_offset;

}

/**

 * Adjust current fragment position for better alignment

 * with previous fragment.

 *

 * @return alignment correction.

 */

static int yae_adjust_position(ATempoContext *atempo)

{

    const AudioFragment *prev = yae_prev_frag(atempo);

    AudioFragment       *frag = yae_curr_frag(atempo);

    const double prev_output_position =

        (double)(prev->position[1] - atempo->origin[1] + atempo->window / 2);

    const double ideal_output_position =

        (double)(prev->position[0] - atempo->origin[0] + atempo->window / 2) /

        atempo->tempo;

    const int drift = (int)(prev_output_position - ideal_output_position);

    const int delta_max  = atempo->window / 2;

    const int correction = yae_align(frag,

                                     prev,

                                     atempo->window,

                                     delta_max,

                                     drift,

                                     atempo->correlation,

                                     atempo->complex_to_real);

    if (correction) {

        // adjust fragment position:

        frag->position[0] -= correction;

        // clear so that the fragment can be reloaded:

        frag->nsamples = 0;

    }

    return correction;

}

/**

 * A helper macro for blending the overlap region of previous

 * and current audio fragment.

 */

#define yae_blend(scalar_type)                                          \

    do {                                                                \

        const scalar_type *aaa = (const scalar_type *)a;                \

        const scalar_type *bbb = (const scalar_type *)b;                \

                                                                        \

        scalar_type *out     = (scalar_type *)dst;                      \

        scalar_type *out_end = (scalar_type *)dst_end;                  \

        int64_t i;                                                      \

                                                                        \

        for (i = 0; i < overlap && out < out_end;                       \

             i++, atempo->position[1]++, wa++, wb++) {                  \

            float w0 = *wa;                                             \

            float w1 = *wb;                                             \

            int j;                                                      \

                                                                        \

            for (j = 0; j < atempo->channels;                           \

                 j++, aaa++, bbb++, out++) {                            \

                float t0 = (float)*aaa;                                 \

                float t1 = (float)*bbb;                                 \

                                                                        \

                *out =                                                  \

                    frag->position[0] + i < 0 ?                         \

                    *aaa :                                              \

                    (scalar_type)(t0 * w0 + t1 * w1);                   \

            }                                                           \

        }                                                               \

        dst = (uint8_t *)out;                                           \

    } while (0)

/**

 * Blend the overlap region of previous and current audio fragment

 * and output the results to the given destination buffer.

 *

 * @return

 *   0 if the overlap region was completely stored in the dst buffer,

 *   AVERROR(EAGAIN) if more destination buffer space is required.

 */

static int yae_overlap_add(ATempoContext *atempo,

                           uint8_t **dst_ref,

                           uint8_t *dst_end)

{

    // shortcuts:

    const AudioFragment *prev = yae_prev_frag(atempo);

    const AudioFragment *frag = yae_curr_frag(atempo);

    const int64_t start_here = FFMAX(atempo->position[1],

                                     frag->position[1]);

    const int64_t stop_here = FFMIN(prev->position[1] + prev->nsamples,

                                    frag->position[1] + frag->nsamples);

    const int64_t overlap = stop_here - start_here;

    const int64_t ia = start_here - prev->position[1];

    const int64_t ib = start_here - frag->position[1];

    const float *wa = atempo->hann + ia;

    const float *wb = atempo->hann + ib;

    const uint8_t *a = prev->data + ia * atempo->stride;

    const uint8_t *b = frag->data + ib * atempo->stride;

    uint8_t *dst = *dst_ref;

    av_assert0(start_here <= stop_here &&

               frag->position[1] <= start_here &&

               overlap <= frag->nsamples);

    if (atempo->format == AV_SAMPLE_FMT_U8) {

        yae_blend(uint8_t);

    } else if (atempo->format == AV_SAMPLE_FMT_S16) {

        yae_blend(int16_t);

    } else if (atempo->format == AV_SAMPLE_FMT_S32) {

        yae_blend(int);

    } else if (atempo->format == AV_SAMPLE_FMT_FLT) {

        yae_blend(float);

    } else if (atempo->format == AV_SAMPLE_FMT_DBL) {

        yae_blend(double);

    }

    // pass-back the updated destination buffer pointer:

    *dst_ref = dst;

    return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);

}

/**

 * Feed as much data to the filter as it is able to consume

 * and receive as much processed data in the destination buffer

 * as it is able to produce or store.

 */

static void

yae_apply(ATempoContext *atempo,

          const uint8_t **src_ref,

          const uint8_t *src_end,

          uint8_t **dst_ref,

          uint8_t *dst_end)

{

    while (1) {

        if (atempo->state == YAE_LOAD_FRAGMENT) {

            // load additional data for the current fragment:

            if (yae_load_frag(atempo, src_ref, src_end) != 0) {

                break;

            }

            // down-mix to mono:

            yae_downmix(atempo, yae_curr_frag(atempo));

            // apply rDFT:

            av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);

            // must load the second fragment before alignment can start:

            if (!atempo->nfrag) {

                yae_advance_to_next_frag(atempo);

                continue;

            }

            atempo->state = YAE_ADJUST_POSITION;

        }

        if (atempo->state == YAE_ADJUST_POSITION) {

            // adjust position for better alignment:

            if (yae_adjust_position(atempo)) {

                // reload the fragment at the corrected position, so that the

                // Hann window blending would not require normalization:

                atempo->state = YAE_RELOAD_FRAGMENT;

            } else {

                atempo->state = YAE_OUTPUT_OVERLAP_ADD;

            }

        }

        if (atempo->state == YAE_RELOAD_FRAGMENT) {

            // load additional data if necessary due to position adjustment:

            if (yae_load_frag(atempo, src_ref, src_end) != 0) {

                break;

            }

            // down-mix to mono:

            yae_downmix(atempo, yae_curr_frag(atempo));

            // apply rDFT:

            av_rdft_calc(atempo->real_to_complex, yae_curr_frag(atempo)->xdat);

            atempo->state = YAE_OUTPUT_OVERLAP_ADD;

        }

        if (atempo->state == YAE_OUTPUT_OVERLAP_ADD) {

            // overlap-add and output the result:

            if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {

                break;

            }

            // advance to the next fragment, repeat:

            yae_advance_to_next_frag(atempo);

            atempo->state = YAE_LOAD_FRAGMENT;

        }

    }

}

/**

 * Flush any buffered data from the filter.

 *

 * @return

 *   0 if all data was completely stored in the dst buffer,

 *   AVERROR(EAGAIN) if more destination buffer space is required.

 */

static int yae_flush(ATempoContext *atempo,

                     uint8_t **dst_ref,

                     uint8_t *dst_end)

{

    AudioFragment *frag = yae_curr_frag(atempo);

    int64_t overlap_end;

    int64_t start_here;

    int64_t stop_here;

    int64_t offset;

    const uint8_t *src;

    uint8_t *dst;

    int src_size;

    int dst_size;

    int nbytes;

    atempo->state = YAE_FLUSH_OUTPUT;

    if (atempo->position[0] == frag->position[0] + frag->nsamples &&

        atempo->position[1] == frag->position[1] + frag->nsamples) {

        // the current fragment is already flushed:

        return 0;

    }

    if (frag->position[0] + frag->nsamples < atempo->position[0]) {

        // finish loading the current (possibly partial) fragment:

        yae_load_frag(atempo, NULL, NULL);

        if (atempo->nfrag) {

            // down-mix to mono:

            yae_downmix(atempo, frag);

            // apply rDFT:

            av_rdft_calc(atempo->real_to_complex, frag->xdat);

            // align current fragment to previous fragment:

            if (yae_adjust_position(atempo)) {

                // reload the current fragment due to adjusted position:

                yae_load_frag(atempo, NULL, NULL);

            }

        }

    }

    // flush the overlap region:

    overlap_end = frag->position[1] + FFMIN(atempo->window / 2,

                                            frag->nsamples);

    while (atempo->position[1] < overlap_end) {

        if (yae_overlap_add(atempo, dst_ref, dst_end) != 0) {

            return AVERROR(EAGAIN);

        }

    }

    // flush the remaininder of the current fragment:

    start_here = FFMAX(atempo->position[1], overlap_end);

    stop_here  = frag->position[1] + frag->nsamples;

    offset     = start_here - frag->position[1];

    av_assert0(start_here <= stop_here && frag->position[1] <= start_here);

    src = frag->data + offset * atempo->stride;

    dst = (uint8_t *)*dst_ref;

    src_size = (int)(stop_here - start_here) * atempo->stride;

    dst_size = dst_end - dst;

    nbytes = FFMIN(src_size, dst_size);

    memcpy(dst, src, nbytes);

    dst += nbytes;

    atempo->position[1] += (nbytes / atempo->stride);

    // pass-back the updated destination buffer pointer:

    *dst_ref = (uint8_t *)dst;

    return atempo->position[1] == stop_here ? 0 : AVERROR(EAGAIN);

}

static av_cold int init(AVFilterContext *ctx)

{

    ATempoContext *atempo = ctx->priv;

    atempo->format = AV_SAMPLE_FMT_NONE;

    atempo->state  = YAE_LOAD_FRAGMENT;

    return 0;

}

static av_cold void uninit(AVFilterContext *ctx)

{

    ATempoContext *atempo = ctx->priv;

    yae_release_buffers(atempo);

}

static int query_formats(AVFilterContext *ctx)

{

    AVFilterChannelLayouts *layouts = NULL;

    AVFilterFormats        *formats = NULL;

    // WSOLA necessitates an internal sliding window ring buffer

    // for incoming audio stream.

    //

    // Planar sample formats are too cumbersome to store in a ring buffer,

    // therefore planar sample formats are not supported.

    //

    static const enum AVSampleFormat sample_fmts[] = {

        AV_SAMPLE_FMT_U8,

        AV_SAMPLE_FMT_S16,

        AV_SAMPLE_FMT_S32,

        AV_SAMPLE_FMT_FLT,

        AV_SAMPLE_FMT_DBL,

        AV_SAMPLE_FMT_NONE

    };

    layouts = ff_all_channel_layouts();

    if (!layouts) {

        return AVERROR(ENOMEM);

    }

    ff_set_common_channel_layouts(ctx, layouts);

    formats = ff_make_format_list(sample_fmts);

    if (!formats) {

        return AVERROR(ENOMEM);

    }

    ff_set_common_formats(ctx, formats);

    formats = ff_all_samplerates();

    if (!formats) {

        return AVERROR(ENOMEM);

    }

    ff_set_common_samplerates(ctx, formats);

    return 0;

}

static int config_props(AVFilterLink *inlink)

{

    AVFilterContext  *ctx = inlink->dst;

    ATempoContext *atempo = ctx->priv;

    enum AVSampleFormat format = inlink->format;

    int sample_rate = (int)inlink->sample_rate;

    int channels = av_get_channel_layout_nb_channels(inlink->channel_layout);

    ctx->outputs[0]->flags |= FF_LINK_FLAG_REQUEST_LOOP;

    return yae_reset(atempo, format, sample_rate, channels);

}

static int push_samples(ATempoContext *atempo,

                        AVFilterLink *outlink,

                        int n_out)

{

    int ret;

    atempo->dst_buffer->sample_rate = outlink->sample_rate;

    atempo->dst_buffer->nb_samples  = n_out;

    // adjust the PTS:

    atempo->dst_buffer->pts =

        av_rescale_q(atempo->nsamples_out,

                     (AVRational){ 1, outlink->sample_rate },

                     outlink->time_base);

    ret = ff_filter_frame(outlink, atempo->dst_buffer);

    if (ret < 0)

        return ret;

    atempo->dst_buffer = NULL;

    atempo->dst        = NULL;

    atempo->dst_end    = NULL;

    atempo->nsamples_out += n_out;

    return 0;

}

static int filter_frame(AVFilterLink *inlink, AVFrame *src_buffer)

{

    AVFilterContext  *ctx = inlink->dst;

    ATempoContext *atempo = ctx->priv;

    AVFilterLink *outlink = ctx->outputs[0];

    int ret = 0;

    int n_in = src_buffer->nb_samples;

    int n_out = (int)(0.5 + ((double)n_in) / atempo->tempo);

    const uint8_t *src = src_buffer->data[0];

    const uint8_t *src_end = src + n_in * atempo->stride;

    while (src < src_end) {

        if (!atempo->dst_buffer) {

            atempo->dst_buffer = ff_get_audio_buffer(outlink, n_out);

            if (!atempo->dst_buffer)

                return AVERROR(ENOMEM);

            av_frame_copy_props(atempo->dst_buffer, src_buffer);

            atempo->dst = atempo->dst_buffer->data[0];

            atempo->dst_end = atempo->dst + n_out * atempo->stride;

        }

        yae_apply(atempo, &src, src_end, &atempo->dst, atempo->dst_end);

        if (atempo->dst == atempo->dst_end) {

            int n_samples = ((atempo->dst - atempo->dst_buffer->data[0]) /

                             atempo->stride);

            ret = push_samples(atempo, outlink, n_samples);

            if (ret < 0)

                goto end;

        }

    }

    atempo->nsamples_in += n_in;

end:

    av_frame_free(&src_buffer);

    return ret;

}

static int request_frame(AVFilterLink *outlink)

{

    AVFilterContext  *ctx = outlink->src;

    ATempoContext *atempo = ctx->priv;

    int ret;

    ret = ff_request_frame(ctx->inputs[0]);

    if (ret == AVERROR_EOF) {

        // flush the filter:

        int n_max = atempo->ring;

        int n_out;

        int err = AVERROR(EAGAIN);

        while (err == AVERROR(EAGAIN)) {

            if (!atempo->dst_buffer) {

                atempo->dst_buffer = ff_get_audio_buffer(outlink, n_max);

                if (!atempo->dst_buffer)