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/eval.h"
  #include "libavcodec/avcodec.h"
  #include "avfilter.h"
  #include "internal.h"
  
  typedef struct {
      struct AVResampleContext *resample;
      int out_rate;
      double ratio;
      AVFilterBufferRef *outsamplesref;
      int unconsumed_nb_samples,
          max_cached_nb_samples;
      int16_t *cached_data[8],
              *resampled_data[8];
  } AResampleContext;
  
  static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
  {
      AResampleContext *aresample = ctx->priv;
      int ret;
  
      if (args) {
          if ((ret = ff_parse_sample_rate(&aresample->out_rate, args, ctx)) < 0)
              return ret;
      } else {
          aresample->out_rate = -1;
      }
  
      return 0;
  }
  
  static av_cold void uninit(AVFilterContext *ctx)
  {
      AResampleContext *aresample = ctx->priv;
      if (aresample->outsamplesref) {
          int nb_channels =
              av_get_channel_layout_nb_channels(
                  aresample->outsamplesref->audio->channel_layout);
          avfilter_unref_buffer(aresample->outsamplesref);
          while (nb_channels--) {
              av_freep(&(aresample->cached_data[nb_channels]));
              av_freep(&(aresample->resampled_data[nb_channels]));
          }
      }
  
      if (aresample->resample)
          av_resample_close(aresample->resample);
  }
  
  static int config_output(AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      AVFilterLink *inlink = ctx->inputs[0];
      AResampleContext *aresample = ctx->priv;
  
      if (aresample->out_rate == -1)
          aresample->out_rate = outlink->sample_rate;
      else
          outlink->sample_rate = aresample->out_rate;
      outlink->time_base = (AVRational) {1, aresample->out_rate};
  
      //TODO: make the resampling parameters configurable
      aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
                                             16, 10, 0, 0.8);
  
      aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
  
      av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
             inlink->sample_rate, outlink->sample_rate);
      return 0;
  }
  
  static int query_formats(AVFilterContext *ctx)
  {
      AVFilterFormats *formats = NULL;
  
      avfilter_add_format(&formats, AV_SAMPLE_FMT_S16);
      if (!formats)
          return AVERROR(ENOMEM);
      avfilter_set_common_sample_formats(ctx, formats);
  
      formats = avfilter_make_all_channel_layouts();
      if (!formats)
          return AVERROR(ENOMEM);
      avfilter_set_common_channel_layouts(ctx, formats);
  
      formats = avfilter_make_all_packing_formats();
      if (!formats)
          return AVERROR(ENOMEM);
      avfilter_set_common_packing_formats(ctx, formats);
  
      return 0;
  }
  
  static void deinterleave(int16_t **outp, int16_t *in,
                           int nb_channels, int nb_samples)
  {
      int16_t *out[8];
      memcpy(out, outp, nb_channels * sizeof(int16_t*));
  
      switch (nb_channels) {
      case 2:
          while (nb_samples--) {
              *out[0]++ = *in++;
              *out[1]++ = *in++;
          }
          break;
      case 3:
          while (nb_samples--) {
              *out[0]++ = *in++;
              *out[1]++ = *in++;
              *out[2]++ = *in++;
          }
          break;
      case 4:
          while (nb_samples--) {
              *out[0]++ = *in++;
              *out[1]++ = *in++;
              *out[2]++ = *in++;
              *out[3]++ = *in++;
          }
          break;
      case 5:
          while (nb_samples--) {
              *out[0]++ = *in++;
              *out[1]++ = *in++;
              *out[2]++ = *in++;
              *out[3]++ = *in++;
              *out[4]++ = *in++;
          }
          break;
      case 6:
          while (nb_samples--) {
              *out[0]++ = *in++;
              *out[1]++ = *in++;
              *out[2]++ = *in++;
              *out[3]++ = *in++;
              *out[4]++ = *in++;
              *out[5]++ = *in++;
          }
          break;
      case 8:
          while (nb_samples--) {
              *out[0]++ = *in++;
              *out[1]++ = *in++;
              *out[2]++ = *in++;
              *out[3]++ = *in++;
              *out[4]++ = *in++;
              *out[5]++ = *in++;
              *out[6]++ = *in++;
              *out[7]++ = *in++;
          }
          break;
      }
  }
  
  static void interleave(int16_t *out, int16_t **inp,
          int nb_channels, int nb_samples)
  {
      int16_t *in[8];
      memcpy(in, inp, nb_channels * sizeof(int16_t*));
  
      switch (nb_channels) {
      case 2:
          while (nb_samples--) {
              *out++ = *in[0]++;
              *out++ = *in[1]++;
          }
          break;
      case 3:
          while (nb_samples--) {
              *out++ = *in[0]++;
              *out++ = *in[1]++;
              *out++ = *in[2]++;
          }
          break;
      case 4:
          while (nb_samples--) {
              *out++ = *in[0]++;
              *out++ = *in[1]++;
              *out++ = *in[2]++;
              *out++ = *in[3]++;
          }
          break;
      case 5:
          while (nb_samples--) {
              *out++ = *in[0]++;
              *out++ = *in[1]++;
              *out++ = *in[2]++;
              *out++ = *in[3]++;
              *out++ = *in[4]++;
          }
          break;
      case 6:
          while (nb_samples--) {
              *out++ = *in[0]++;
              *out++ = *in[1]++;
              *out++ = *in[2]++;
              *out++ = *in[3]++;
              *out++ = *in[4]++;
              *out++ = *in[5]++;
          }
          break;
      case 8:
          while (nb_samples--) {
              *out++ = *in[0]++;
              *out++ = *in[1]++;
              *out++ = *in[2]++;
              *out++ = *in[3]++;
              *out++ = *in[4]++;
              *out++ = *in[5]++;
              *out++ = *in[6]++;
              *out++ = *in[7]++;
          }
          break;
      }
  }
  
  static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
  {
      AResampleContext *aresample  = inlink->dst->priv;
      AVFilterLink * const outlink = inlink->dst->outputs[0];
      int i,
          in_nb_samples            = insamplesref->audio->nb_samples,
          cached_nb_samples        = in_nb_samples + aresample->unconsumed_nb_samples,
          requested_out_nb_samples = aresample->ratio * cached_nb_samples,
          nb_channels              =
              av_get_channel_layout_nb_channels(inlink->channel_layout);
  
      if (cached_nb_samples > aresample->max_cached_nb_samples) {
          for (i = 0; i < nb_channels; i++) {
              aresample->cached_data[i]    =
                  av_realloc(aresample->cached_data[i], cached_nb_samples * sizeof(int16_t));
              aresample->resampled_data[i] =
                  av_realloc(aresample->resampled_data[i],
                             FFALIGN(sizeof(int16_t) * requested_out_nb_samples, 16));
  
              if (aresample->cached_data[i] == NULL || aresample->resampled_data[i] == NULL)
                  return;
          }
          aresample->max_cached_nb_samples = cached_nb_samples;
  
          if (aresample->outsamplesref)
              avfilter_unref_buffer(aresample->outsamplesref);
  
          aresample->outsamplesref =
              avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, requested_out_nb_samples);
          outlink->out_buf = aresample->outsamplesref;
      }
  
      avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
      aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
      aresample->outsamplesref->pts =
          av_rescale(outlink->sample_rate, insamplesref->pts, inlink->sample_rate);
  
      /* av_resample() works with planar audio buffers */
      if (!inlink->planar && nb_channels > 1) {
          int16_t *out[8];
          for (i = 0; i < nb_channels; i++)
              out[i] = aresample->cached_data[i] + aresample->unconsumed_nb_samples;
  
          deinterleave(out, (int16_t *)insamplesref->data[0],
                       nb_channels, in_nb_samples);
      } else {
          for (i = 0; i < nb_channels; i++)
              memcpy(aresample->cached_data[i] + aresample->unconsumed_nb_samples,
                     insamplesref->data[i],
                     in_nb_samples * sizeof(int16_t));
      }
  
      for (i = 0; i < nb_channels; i++) {
          int consumed_nb_samples;
          const int is_last = i+1 == nb_channels;
  
          aresample->outsamplesref->audio->nb_samples =
              av_resample(aresample->resample,
                          aresample->resampled_data[i], aresample->cached_data[i],
                          &consumed_nb_samples,
                          cached_nb_samples,
                          requested_out_nb_samples, is_last);
  
          /* move unconsumed data back to the beginning of the cache */
          aresample->unconsumed_nb_samples = cached_nb_samples - consumed_nb_samples;
          memmove(aresample->cached_data[i],
                  aresample->cached_data[i] + consumed_nb_samples,
                  aresample->unconsumed_nb_samples * sizeof(int16_t));
      }
  
  
      /* copy resampled data to the output samplesref */
      if (!inlink->planar && nb_channels > 1) {
          interleave((int16_t *)aresample->outsamplesref->data[0],
                     aresample->resampled_data,
                     nb_channels, aresample->outsamplesref->audio->nb_samples);
      } else {
          for (i = 0; i < nb_channels; i++)
              memcpy(aresample->outsamplesref->data[i], aresample->resampled_data[i],
                     aresample->outsamplesref->audio->nb_samples * sizeof(int16_t));
      }
  
      avfilter_filter_samples(outlink, avfilter_ref_buffer(aresample->outsamplesref, ~0));
      avfilter_unref_buffer(insamplesref);
  }
  
  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    = (const AVFilterPad[]) {{ .name      = "default",
                                      .type            = AVMEDIA_TYPE_AUDIO,
                                      .filter_samples  = filter_samples,
                                      .min_perms       = AV_PERM_READ, },
                                    { .name = NULL}},
      .outputs   = (const AVFilterPad[]) {{ .name      = "default",
                                      .config_props    = config_output,
                                      .type            = AVMEDIA_TYPE_AUDIO, },
                                    { .name = NULL}},
  };
  