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/*

 * 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 "libavresample/avresample.h"

#include "libavutil/attributes.h"

#include "libavutil/audio_fifo.h"

#include "libavutil/common.h"

#include "libavutil/mathematics.h"

#include "libavutil/opt.h"

#include "libavutil/samplefmt.h"

#include "audio.h"

#include "avfilter.h"

#include "internal.h"

typedef struct ASyncContext {

    const AVClass *class;

    AVAudioResampleContext *avr;

    int64_t pts;            ///< timestamp in samples of the first sample in fifo

    int min_delta;          ///< pad/trim min threshold in samples

    int first_frame;        ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE

    int64_t first_pts;      ///< user-specified first expected pts, in samples

    int comp;               ///< current resample compensation

    /* options */

    int resample;

    float min_delta_sec;

    int max_comp;

    /* set by filter_frame() to signal an output frame to request_frame() */

    int got_output;

} ASyncContext;

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

#define A AV_OPT_FLAG_AUDIO_PARAM

#define F AV_OPT_FLAG_FILTERING_PARAM

static const AVOption asyncts_options[] = {

    { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample),      AV_OPT_TYPE_INT,   { .i64 = 0 },   0, 1,       A|F },

    { "min_delta",  "Minimum difference between timestamps and audio data "

                    "(in seconds) to trigger padding/trimmin the data.",        OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { .dbl = 0.1 }, 0, INT_MAX, A|F },

    { "max_comp",   "Maximum compensation in samples per second.",              OFFSET(max_comp),      AV_OPT_TYPE_INT,   { .i64 = 500 }, 0, INT_MAX, A|F },

    { "first_pts",  "Assume the first pts should be this value.",               OFFSET(first_pts),     AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_MAX, A|F },

    { NULL }

};

AVFILTER_DEFINE_CLASS(asyncts);

static av_cold int init(AVFilterContext *ctx)

{

    ASyncContext *s = ctx->priv;

    s->pts         = AV_NOPTS_VALUE;

    s->first_frame = 1;

    return 0;

}

static av_cold void uninit(AVFilterContext *ctx)

{

    ASyncContext *s = ctx->priv;

    if (s->avr) {

        avresample_close(s->avr);

        avresample_free(&s->avr);

    }

}

static int config_props(AVFilterLink *link)

{

    ASyncContext *s = link->src->priv;

    int ret;

    s->min_delta = s->min_delta_sec * link->sample_rate;

    link->time_base = (AVRational){1, link->sample_rate};

    s->avr = avresample_alloc_context();

    if (!s->avr)

        return AVERROR(ENOMEM);

    av_opt_set_int(s->avr,  "in_channel_layout", link->channel_layout, 0);

    av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0);

    av_opt_set_int(s->avr,  "in_sample_fmt",     link->format,         0);

    av_opt_set_int(s->avr, "out_sample_fmt",     link->format,         0);

    av_opt_set_int(s->avr,  "in_sample_rate",    link->sample_rate,    0);

    av_opt_set_int(s->avr, "out_sample_rate",    link->sample_rate,    0);

    if (s->resample)

        av_opt_set_int(s->avr, "force_resampling", 1, 0);

    if ((ret = avresample_open(s->avr)) < 0)

        return ret;

    return 0;

}

/* get amount of data currently buffered, in samples */

static int64_t get_delay(ASyncContext *s)

{

    return avresample_available(s->avr) + avresample_get_delay(s->avr);

}

static void handle_trimming(AVFilterContext *ctx)

{

    ASyncContext *s = ctx->priv;

    if (s->pts < s->first_pts) {

        int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr));

        av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n",

               delta);

        avresample_read(s->avr, NULL, delta);

        s->pts += delta;

    } else if (s->first_frame)

        s->pts = s->first_pts;

}

static int request_frame(AVFilterLink *link)

{

    AVFilterContext *ctx = link->src;

    ASyncContext      *s = ctx->priv;

    int ret = 0;

    int nb_samples;

    s->got_output = 0;

    while (ret >= 0 && !s->got_output)

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

    /* flush the fifo */

    if (ret == AVERROR_EOF) {

        if (s->first_pts != AV_NOPTS_VALUE)

            handle_trimming(ctx);

        if (nb_samples = get_delay(s)) {

            AVFrame *buf = ff_get_audio_buffer(link, nb_samples);

            if (!buf)

                return AVERROR(ENOMEM);

            ret = avresample_convert(s->avr, buf->extended_data,

                                     buf->linesize[0], nb_samples, NULL, 0, 0);

            if (ret <= 0) {

                av_frame_free(&buf);

                return (ret < 0) ? ret : AVERROR_EOF;

            }

            buf->pts = s->pts;

            return ff_filter_frame(link, buf);

        }

    }

    return ret;

}

static int write_to_fifo(ASyncContext *s, AVFrame *buf)

{

    int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,

                                 buf->linesize[0], buf->nb_samples);

    av_frame_free(&buf);

    return ret;

}

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

{

    AVFilterContext  *ctx = inlink->dst;

    ASyncContext       *s = ctx->priv;

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

    int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout);

    int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :

                  av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);

    int out_size, ret;

    int64_t delta;

    int64_t new_pts;

    /* buffer data until we get the next timestamp */

    if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) {

        if (pts != AV_NOPTS_VALUE) {

            s->pts = pts - get_delay(s);

        }

        return write_to_fifo(s, buf);

    }

    if (s->first_pts != AV_NOPTS_VALUE) {

        handle_trimming(ctx);

        if (!avresample_available(s->avr))

            return write_to_fifo(s, buf);

    }

    /* when we have two timestamps, compute how many samples would we have

     * to add/remove to get proper sync between data and timestamps */

    delta    = pts - s->pts - get_delay(s);

    out_size = avresample_available(s->avr);

    if (labs(delta) > s->min_delta ||

        (s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) {

        av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);

        out_size = av_clipl_int32((int64_t)out_size + delta);

    } else {

        if (s->resample) {

            // adjust the compensation if delta is non-zero

            int delay = get_delay(s);

            int comp = s->comp + av_clip(delta * inlink->sample_rate / delay,

                                         -s->max_comp, s->max_comp);

            if (comp != s->comp) {

                av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);

                if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) {

                    s->comp = comp;

                }

            }

        }

        // adjust PTS to avoid monotonicity errors with input PTS jitter

        pts -= delta;

        delta = 0;

    }

    if (out_size > 0) {

        AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size);

        if (!buf_out) {

            ret = AVERROR(ENOMEM);

            goto fail;

        }

        if (s->first_frame && delta > 0) {

            int planar = av_sample_fmt_is_planar(buf_out->format);

            int planes = planar ?  nb_channels : 1;

            int block_size = av_get_bytes_per_sample(buf_out->format) *

                             (planar ? 1 : nb_channels);

            int ch;

            av_samples_set_silence(buf_out->extended_data, 0, delta,

                                   nb_channels, buf->format);

            for (ch = 0; ch < planes; ch++)

                buf_out->extended_data[ch] += delta * block_size;

            avresample_read(s->avr, buf_out->extended_data, out_size);

            for (ch = 0; ch < planes; ch++)

                buf_out->extended_data[ch] -= delta * block_size;

        } else {

            avresample_read(s->avr, buf_out->extended_data, out_size);

            if (delta > 0) {

                av_samples_set_silence(buf_out->extended_data, out_size - delta,

                                       delta, nb_channels, buf->format);

            }

        }

        buf_out->pts = s->pts;

        ret = ff_filter_frame(outlink, buf_out);

        if (ret < 0)

            goto fail;

        s->got_output = 1;

    } else if (avresample_available(s->avr)) {

        av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "

               "whole buffer.\n");

    }

    /* drain any remaining buffered data */

    avresample_read(s->avr, NULL, avresample_available(s->avr));

    new_pts = pts - avresample_get_delay(s->avr);

    /* check for s->pts monotonicity */

    if (new_pts > s->pts) {

        s->pts = new_pts;

        ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,

                                 buf->linesize[0], buf->nb_samples);

    } else {

        av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "

               "whole buffer.\n");

        ret = 0;

    }

    s->first_frame = 0;

fail:

    av_frame_free(&buf);

    return ret;

}

static const AVFilterPad avfilter_af_asyncts_inputs[] = {

    {

        .name          = "default",

        .type          = AVMEDIA_TYPE_AUDIO,

        .filter_frame  = filter_frame

    },

    { NULL }

};

static const AVFilterPad avfilter_af_asyncts_outputs[] = {

    {

        .name          = "default",

        .type          = AVMEDIA_TYPE_AUDIO,

        .config_props  = config_props,

        .request_frame = request_frame

    },

    { NULL }

};

AVFilter avfilter_af_asyncts = {

    .name        = "asyncts",

    .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),

    .init        = init,

    .uninit      = uninit,

    .priv_size   = sizeof(ASyncContext),

    .priv_class  = &asyncts_class,

    .inputs      = avfilter_af_asyncts_inputs,

    .outputs     = avfilter_af_asyncts_outputs,

};