text: /*
   * Copyright (c) 2011 Stefano Sabatini
   * Copyright (c) 2011 Mina Nagy Zaki
   *
   * 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
   * resampling audio filter
   */
  
  #include "libavutil/avstring.h"
  #include "libavutil/channel_layout.h"
  #include "libavutil/opt.h"
  #include "libavutil/samplefmt.h"
  #include "libavutil/avassert.h"
  #include "libswresample/swresample.h"
  #include "avfilter.h"
  #include "audio.h"
  #include "internal.h"
  
  typedef struct {
      double ratio;
      struct SwrContext *swr;
      int64_t next_pts;
      int req_fullfilled;
  } AResampleContext;
  
  static av_cold int init(AVFilterContext *ctx, const char *args)
  {
      AResampleContext *aresample = ctx->priv;
      int ret = 0;
      char *argd = av_strdup(args);
  
      aresample->next_pts = AV_NOPTS_VALUE;
      aresample->swr = swr_alloc();
      if (!aresample->swr) {
          ret = AVERROR(ENOMEM);
          goto end;
      }
  
      if (args) {
          char *ptr = argd, *token;
  
          while (token = av_strtok(ptr, ":", &ptr)) {
              char *value;
              av_strtok(token, "=", &value);
  
              if (value) {
                  if ((ret = av_opt_set(aresample->swr, token, value, 0)) < 0)
                      goto end;
              } else {
                  int out_rate;
                  if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
                      goto end;
                  if ((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
                      goto end;
              }
          }
      }
  end:
      av_free(argd);
      return ret;
  }
  
  static av_cold void uninit(AVFilterContext *ctx)
  {
      AResampleContext *aresample = ctx->priv;
      swr_free(&aresample->swr);
  }
  
  static int query_formats(AVFilterContext *ctx)
  {
      AResampleContext *aresample = ctx->priv;
      int out_rate                   = av_get_int(aresample->swr, "osr", NULL);
      uint64_t out_layout            = av_get_int(aresample->swr, "ocl", NULL);
      enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
  
      AVFilterLink *inlink  = ctx->inputs[0];
      AVFilterLink *outlink = ctx->outputs[0];
  
      AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
      AVFilterFormats        *out_formats;
      AVFilterFormats        *in_samplerates  = ff_all_samplerates();
      AVFilterFormats        *out_samplerates;
      AVFilterChannelLayouts *in_layouts      = ff_all_channel_counts();
      AVFilterChannelLayouts *out_layouts;
  
      ff_formats_ref  (in_formats,      &inlink->out_formats);
      ff_formats_ref  (in_samplerates,  &inlink->out_samplerates);
      ff_channel_layouts_ref(in_layouts,      &inlink->out_channel_layouts);
  
      if(out_rate > 0) {
          out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
      } else {
          out_samplerates = ff_all_samplerates();
      }
      ff_formats_ref(out_samplerates, &outlink->in_samplerates);
  
      if(out_format != AV_SAMPLE_FMT_NONE) {
          out_formats = ff_make_format_list((int[]){ out_format, -1 });
      } else
          out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
      ff_formats_ref(out_formats, &outlink->in_formats);
  
      if(out_layout) {
          out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
      } else
          out_layouts = ff_all_channel_counts();
      ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
  
      return 0;
  }
  
  
  static int config_output(AVFilterLink *outlink)
  {
      int ret;
      AVFilterContext *ctx = outlink->src;
      AVFilterLink *inlink = ctx->inputs[0];
      AResampleContext *aresample = ctx->priv;
      int out_rate;
      uint64_t out_layout;
      enum AVSampleFormat out_format;
      char inchl_buf[128], outchl_buf[128];
  
      aresample->swr = swr_alloc_set_opts(aresample->swr,
                                          outlink->channel_layout, outlink->format, outlink->sample_rate,
                                          inlink->channel_layout, inlink->format, inlink->sample_rate,
                                          0, ctx);
      if (!aresample->swr)
          return AVERROR(ENOMEM);
      if (!inlink->channel_layout)
          av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
      if (!outlink->channel_layout)
          av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
  
      ret = swr_init(aresample->swr);
      if (ret < 0)
          return ret;
  
      out_rate   = av_get_int(aresample->swr, "osr", NULL);
      out_layout = av_get_int(aresample->swr, "ocl", NULL);
      out_format = av_get_int(aresample->swr, "osf", NULL);
      outlink->time_base = (AVRational) {1, out_rate};
  
      av_assert0(outlink->sample_rate == out_rate);
      av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
      av_assert0(outlink->format == out_format);
  
      aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
  
      av_get_channel_layout_string(inchl_buf,  sizeof(inchl_buf),  inlink ->channels, inlink ->channel_layout);
      av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
  
      av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
             inlink ->channels, inchl_buf,  av_get_sample_fmt_name(inlink->format),  inlink->sample_rate,
             outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
      return 0;
  }
  
  static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
  {
      AResampleContext *aresample = inlink->dst->priv;
      const int n_in  = insamplesref->audio->nb_samples;
      int n_out       = n_in * aresample->ratio * 2 + 256;
      AVFilterLink *const outlink = inlink->dst->outputs[0];
      AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
      int ret;
  
      if(!outsamplesref)
          return AVERROR(ENOMEM);
  
      avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
      outsamplesref->format                = outlink->format;
      outsamplesref->audio->channels       = outlink->channels;
      outsamplesref->audio->channel_layout = outlink->channel_layout;
      outsamplesref->audio->sample_rate    = outlink->sample_rate;
  
      if(insamplesref->pts != AV_NOPTS_VALUE) {
          int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
          int64_t outpts= swr_next_pts(aresample->swr, inpts);
          aresample->next_pts =
          outsamplesref->pts  = ROUNDED_DIV(outpts, inlink->sample_rate);
      } else {
          outsamplesref->pts  = AV_NOPTS_VALUE;
      }
      n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
                                   (void *)insamplesref->extended_data, n_in);
      if (n_out <= 0) {
          avfilter_unref_buffer(outsamplesref);
          avfilter_unref_buffer(insamplesref);
          return 0;
      }
  
      outsamplesref->audio->nb_samples  = n_out;
  
      ret = ff_filter_frame(outlink, outsamplesref);
      aresample->req_fullfilled= 1;
      avfilter_unref_buffer(insamplesref);
      return ret;
  }
  
  static int request_frame(AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      AResampleContext *aresample = ctx->priv;
      AVFilterLink *const inlink = outlink->src->inputs[0];
      int ret;
  
      aresample->req_fullfilled = 0;
      do{
          ret = ff_request_frame(ctx->inputs[0]);
      }while(!aresample->req_fullfilled && ret>=0);
  
      if (ret == AVERROR_EOF) {
          AVFilterBufferRef *outsamplesref;
          int n_out = 4096;
  
          outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
          if (!outsamplesref)
              return AVERROR(ENOMEM);
          n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
          if (n_out <= 0) {
              avfilter_unref_buffer(outsamplesref);
              return (n_out == 0) ? AVERROR_EOF : n_out;
          }
  
          outsamplesref->audio->sample_rate = outlink->sample_rate;
          outsamplesref->audio->nb_samples  = n_out;
  #if 0
          outsamplesref->pts = aresample->next_pts;
          if(aresample->next_pts != AV_NOPTS_VALUE)
              aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
  #else
          outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN);
          outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
  #endif
  
          ff_filter_frame(outlink, outsamplesref);
          return 0;
      }
      return ret;
  }
  
  static const AVFilterPad aresample_inputs[] = {
      {
          .name         = "default",
          .type         = AVMEDIA_TYPE_AUDIO,
          .filter_frame = filter_frame,
          .min_perms    = AV_PERM_READ,
      },
      { NULL },
  };
  
  static const AVFilterPad aresample_outputs[] = {
      {
          .name          = "default",
          .config_props  = config_output,
          .request_frame = request_frame,
          .type          = AVMEDIA_TYPE_AUDIO,
      },
      { NULL },
  };
  
  AVFilter avfilter_af_aresample = {
      .name          = "aresample",
      .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
      .init          = init,
      .uninit        = uninit,
      .query_formats = query_formats,
      .priv_size     = sizeof(AResampleContext),
      .inputs        = aresample_inputs,
      .outputs       = aresample_outputs,
  };
  