/*
  * This file is part of Libav.
  *
  * Libav 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.
  *
  * Libav 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 Libav; 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,

 };