text: /*
   * Copyright (c) 2013 Nicolas George
   *
   * 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/opt.h"
  #include "avfilter.h"
  #include "internal.h"
  
  typedef struct {
      const AVClass *class;
      int sample_rate;
      int rescale_pts;
  } ASetRateContext;
  
  #define CONTEXT ASetRateContext
  #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  
  #define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
      { name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type,          \
        { .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
  
  #define OPT_INT(name, field, def, min, max, descr, ...) \
      OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
  
  static const AVOption asetrate_options[] = {
      OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
      OPT_INT("r",           sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
      {NULL},
  };
  
  AVFILTER_DEFINE_CLASS(asetrate);
  
  static av_cold int query_formats(AVFilterContext *ctx)
  {
      ASetRateContext *sr = ctx->priv;
      int sample_rates[] = { sr->sample_rate, -1 };
  
      return ff_formats_ref(ff_make_format_list(sample_rates),
                     &ctx->outputs[0]->in_samplerates);
  }
  
  static av_cold int config_props(AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      ASetRateContext *sr = ctx->priv;
      AVFilterLink *inlink = ctx->inputs[0];
      AVRational intb = ctx->inputs[0]->time_base;
      int inrate = inlink->sample_rate;
  
      if (intb.num == 1 && intb.den == inrate) {
          outlink->time_base.num = 1;
          outlink->time_base.den = outlink->sample_rate;
      } else {
          outlink->time_base = intb;
          sr->rescale_pts = 1;
          if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))
              av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");
      }
      return 0;
  }
  
  static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
  {
      AVFilterContext *ctx = inlink->dst;
      ASetRateContext *sr = ctx->priv;
      AVFilterLink *outlink = ctx->outputs[0];
  
      frame->sample_rate = outlink->sample_rate;
      if (sr->rescale_pts)
          frame->pts = av_rescale(frame->pts, inlink->sample_rate,
                                             outlink->sample_rate);
      return ff_filter_frame(outlink, frame);
  }
  
  static const AVFilterPad asetrate_inputs[] = {
      {
          .name         = "default",
          .type         = AVMEDIA_TYPE_AUDIO,
          .filter_frame = filter_frame,
      },
      { NULL }
  };
  
  static const AVFilterPad asetrate_outputs[] = {
      {
          .name         = "default",
          .type         = AVMEDIA_TYPE_AUDIO,
          .config_props = config_props,
      },
      { NULL }
  };
  
  AVFilter ff_af_asetrate = {
      .name          = "asetrate",
      .description   = NULL_IF_CONFIG_SMALL("Change the sample rate without "
                                            "altering the data."),
      .query_formats = query_formats,
      .priv_size     = sizeof(ASetRateContext),
      .inputs        = asetrate_inputs,
      .outputs       = asetrate_outputs,
      .priv_class    = &asetrate_class,
  };
  