text: /*
   * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
   * Copyright (c) 2015 Paul B Mahol
   *
   * 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
   * Audio (Sidechain) Compressor filter
   */
  
  #include "libavutil/audio_fifo.h"
  #include "libavutil/avassert.h"
  #include "libavutil/channel_layout.h"
  #include "libavutil/common.h"
  #include "libavutil/opt.h"
  
  #include "audio.h"
  #include "avfilter.h"
  #include "formats.h"
  #include "hermite.h"
  #include "internal.h"
  
  typedef struct SidechainCompressContext {
      const AVClass *class;
  
      double level_in;
      double level_sc;
      double attack, attack_coeff;
      double release, release_coeff;
      double lin_slope;
      double ratio;
      double threshold;
      double makeup;
      double mix;
      double thres;
      double knee;
      double knee_start;
      double knee_stop;
      double lin_knee_start;
      double adj_knee_start;
      double compressed_knee_stop;
      int link;
      int detection;
  
      AVAudioFifo *fifo[2];
      int64_t pts;
  } SidechainCompressContext;
  
  #define OFFSET(x) offsetof(SidechainCompressContext, x)
  #define A AV_OPT_FLAG_AUDIO_PARAM
  #define F AV_OPT_FLAG_FILTERING_PARAM
  
  static const AVOption options[] = {
      { "level_in",  "set input gain",     OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},        0.015625,   64, A|F },
      { "threshold", "set threshold",      OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0.000976563,    1, A|F },
      { "ratio",     "set ratio",          OFFSET(ratio),     AV_OPT_TYPE_DOUBLE, {.dbl=2},               1,   20, A|F },
      { "attack",    "set attack",         OFFSET(attack),    AV_OPT_TYPE_DOUBLE, {.dbl=20},           0.01, 2000, A|F },
      { "release",   "set release",        OFFSET(release),   AV_OPT_TYPE_DOUBLE, {.dbl=250},          0.01, 9000, A|F },
      { "makeup",    "set make up gain",   OFFSET(makeup),    AV_OPT_TYPE_DOUBLE, {.dbl=1},               1,   64, A|F },
      { "knee",      "set knee",           OFFSET(knee),      AV_OPT_TYPE_DOUBLE, {.dbl=2.82843},         1,    8, A|F },
      { "link",      "set link type",      OFFSET(link),      AV_OPT_TYPE_INT,    {.i64=0},               0,    1, A|F, "link" },
      {   "average", 0,                    0,                 AV_OPT_TYPE_CONST,  {.i64=0},               0,    0, A|F, "link" },
      {   "maximum", 0,                    0,                 AV_OPT_TYPE_CONST,  {.i64=1},               0,    0, A|F, "link" },
      { "detection", "set detection",      OFFSET(detection), AV_OPT_TYPE_INT,    {.i64=1},               0,    1, A|F, "detection" },
      {   "peak",    0,                    0,                 AV_OPT_TYPE_CONST,  {.i64=0},               0,    0, A|F, "detection" },
      {   "rms",     0,                    0,                 AV_OPT_TYPE_CONST,  {.i64=1},               0,    0, A|F, "detection" },
      { "level_sc",  "set sidechain gain", OFFSET(level_sc),  AV_OPT_TYPE_DOUBLE, {.dbl=1},        0.015625,   64, A|F },
      { "mix",       "set mix",            OFFSET(mix),       AV_OPT_TYPE_DOUBLE, {.dbl=1},               0,    1, A|F },
      { NULL }
  };
  
  #define sidechaincompress_options options
  AVFILTER_DEFINE_CLASS(sidechaincompress);
  
  // A fake infinity value (because real infinity may break some hosts)
  #define FAKE_INFINITY (65536.0 * 65536.0)
  
  // Check for infinity (with appropriate-ish tolerance)
  #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
  
  static double output_gain(double lin_slope, double ratio, double thres,
                            double knee, double knee_start, double knee_stop,
                            double compressed_knee_stop, int detection)
  {
      double slope = log(lin_slope);
      double gain = 0.0;
      double delta = 0.0;
  
      if (detection)
          slope *= 0.5;
  
      if (IS_FAKE_INFINITY(ratio)) {
          gain = thres;
          delta = 0.0;
      } else {
          gain = (slope - thres) / ratio + thres;
          delta = 1.0 / ratio;
      }
  
      if (knee > 1.0 && slope < knee_stop)
          gain = hermite_interpolation(slope, knee_start, knee_stop,
                                       knee_start, compressed_knee_stop,
                                       1.0, delta);
  
      return exp(gain - slope);
  }
  
  static int compressor_config_output(AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      SidechainCompressContext *s = ctx->priv;
  
      s->thres = log(s->threshold);
      s->lin_knee_start = s->threshold / sqrt(s->knee);
      s->adj_knee_start = s->lin_knee_start * s->lin_knee_start;
      s->knee_start = log(s->lin_knee_start);
      s->knee_stop = log(s->threshold * sqrt(s->knee));
      s->compressed_knee_stop = (s->knee_stop - s->thres) / s->ratio + s->thres;
  
      s->attack_coeff = FFMIN(1., 1. / (s->attack * outlink->sample_rate / 4000.));
      s->release_coeff = FFMIN(1., 1. / (s->release * outlink->sample_rate / 4000.));
  
      return 0;
  }
  
  static void compressor(SidechainCompressContext *s,
                         const double *src, double *dst, const double *scsrc, int nb_samples,
                         double level_in, double level_sc,
                         AVFilterLink *inlink, AVFilterLink *sclink)
  {
      const double makeup = s->makeup;
      const double mix = s->mix;
      int i, c;
  
      for (i = 0; i < nb_samples; i++) {
          double abs_sample, gain = 1.0;
  
          abs_sample = fabs(scsrc[0] * level_sc);
  
          if (s->link == 1) {
              for (c = 1; c < sclink->channels; c++)
                  abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
          } else {
              for (c = 1; c < sclink->channels; c++)
                  abs_sample += fabs(scsrc[c] * level_sc);
  
              abs_sample /= sclink->channels;
          }
  
          if (s->detection)
              abs_sample *= abs_sample;
  
          s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? s->attack_coeff : s->release_coeff);
  
          if (s->lin_slope > 0.0 && s->lin_slope > (s->detection ? s->adj_knee_start : s->lin_knee_start))
              gain = output_gain(s->lin_slope, s->ratio, s->thres, s->knee,
                                 s->knee_start, s->knee_stop,
                                 s->compressed_knee_stop, s->detection);
  
          for (c = 0; c < inlink->channels; c++)
              dst[c] = src[c] * level_in * (gain * makeup * mix + (1. - mix));
  
          src += inlink->channels;
          dst += inlink->channels;
          scsrc += sclink->channels;
      }
  }
  
  #if CONFIG_SIDECHAINCOMPRESS_FILTER
  static int filter_frame(AVFilterLink *link, AVFrame *frame)
  {
      AVFilterContext *ctx = link->dst;
      SidechainCompressContext *s = ctx->priv;
      AVFilterLink *outlink = ctx->outputs[0];
      AVFrame *out = NULL, *in[2] = { NULL };
      double *dst;
      int nb_samples;
      int i;
  
      for (i = 0; i < 2; i++)
          if (link == ctx->inputs[i])
              break;
      av_assert0(i < 2);
      av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,
                          frame->nb_samples);
      av_frame_free(&frame);
  
      nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
      if (!nb_samples)
          return 0;
  
      out = ff_get_audio_buffer(outlink, nb_samples);
      if (!out)
          return AVERROR(ENOMEM);
      for (i = 0; i < 2; i++) {
          in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
          if (!in[i]) {
              av_frame_free(&in[0]);
              av_frame_free(&in[1]);
              av_frame_free(&out);
              return AVERROR(ENOMEM);
          }
          av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
      }
  
      dst = (double *)out->data[0];
      out->pts = s->pts;
      s->pts += nb_samples;
  
      compressor(s, (double *)in[0]->data[0], dst,
                 (double *)in[1]->data[0], nb_samples,
                 s->level_in, s->level_sc,
                 ctx->inputs[0], ctx->inputs[1]);
  
      av_frame_free(&in[0]);
      av_frame_free(&in[1]);
  
      return ff_filter_frame(outlink, out);
  }
  
  static int request_frame(AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      SidechainCompressContext *s = ctx->priv;
      int i;
  
      /* get a frame on each input */
      for (i = 0; i < 2; i++) {
          AVFilterLink *inlink = ctx->inputs[i];
          if (!av_audio_fifo_size(s->fifo[i]))
              return ff_request_frame(inlink);
      }
  
      return 0;
  }
  
  static int query_formats(AVFilterContext *ctx)
  {
      AVFilterFormats *formats;
      AVFilterChannelLayouts *layouts = NULL;
      static const enum AVSampleFormat sample_fmts[] = {
          AV_SAMPLE_FMT_DBL,
          AV_SAMPLE_FMT_NONE
      };
      int ret, i;
  
      if (!ctx->inputs[0]->in_channel_layouts ||
          !ctx->inputs[0]->in_channel_layouts->nb_channel_layouts) {
          av_log(ctx, AV_LOG_WARNING,
                 "No channel layout for input 1\n");
              return AVERROR(EAGAIN);
      }
  
      if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 ||
          (ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
          return ret;
  
      for (i = 0; i < 2; i++) {
          layouts = ff_all_channel_counts();
          if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
              return ret;
      }
  
      formats = ff_make_format_list(sample_fmts);
      if ((ret = ff_set_common_formats(ctx, formats)) < 0)
          return ret;
  
      formats = ff_all_samplerates();
      return ff_set_common_samplerates(ctx, formats);
  }
  
  static int config_output(AVFilterLink *outlink)
  {
      AVFilterContext *ctx = outlink->src;
      SidechainCompressContext *s = ctx->priv;
  
      if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
          av_log(ctx, AV_LOG_ERROR,
                 "Inputs must have the same sample rate "
                 "%d for in0 vs %d for in1\n",
                 ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
          return AVERROR(EINVAL);
      }
  
      outlink->sample_rate = ctx->inputs[0]->sample_rate;
      outlink->time_base   = ctx->inputs[0]->time_base;
      outlink->channel_layout = ctx->inputs[0]->channel_layout;
      outlink->channels = ctx->inputs[0]->channels;
  
      s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
      s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
      if (!s->fifo[0] || !s->fifo[1])
          return AVERROR(ENOMEM);
  
      compressor_config_output(outlink);
  
      return 0;
  }
  
  static av_cold void uninit(AVFilterContext *ctx)
  {
      SidechainCompressContext *s = ctx->priv;
  
      av_audio_fifo_free(s->fifo[0]);
      av_audio_fifo_free(s->fifo[1]);
  }
  
  static const AVFilterPad sidechaincompress_inputs[] = {
      {
          .name           = "main",
          .type           = AVMEDIA_TYPE_AUDIO,
          .filter_frame   = filter_frame,
      },{
          .name           = "sidechain",
          .type           = AVMEDIA_TYPE_AUDIO,
          .filter_frame   = filter_frame,
      },
      { NULL }
  };
  
  static const AVFilterPad sidechaincompress_outputs[] = {
      {
          .name          = "default",
          .type          = AVMEDIA_TYPE_AUDIO,
          .config_props  = config_output,
          .request_frame = request_frame,
      },
      { NULL }
  };
  
  AVFilter ff_af_sidechaincompress = {
      .name           = "sidechaincompress",
      .description    = NULL_IF_CONFIG_SMALL("Sidechain compressor."),
      .priv_size      = sizeof(SidechainCompressContext),
      .priv_class     = &sidechaincompress_class,
      .query_formats  = query_formats,
      .uninit         = uninit,
      .inputs         = sidechaincompress_inputs,
      .outputs        = sidechaincompress_outputs,
  };
  #endif  /* CONFIG_SIDECHAINCOMPRESS_FILTER */
  
  #if CONFIG_ACOMPRESSOR_FILTER
  static int acompressor_filter_frame(AVFilterLink *inlink, AVFrame *in)
  {
      const double *src = (const double *)in->data[0];
      AVFilterContext *ctx = inlink->dst;
      SidechainCompressContext *s = ctx->priv;
      AVFilterLink *outlink = ctx->outputs[0];
      AVFrame *out;
      double *dst;
  
      if (av_frame_is_writable(in)) {
          out = in;
      } else {
          out = ff_get_audio_buffer(inlink, in->nb_samples);
          if (!out) {
              av_frame_free(&in);
              return AVERROR(ENOMEM);
          }
          av_frame_copy_props(out, in);
      }
      dst = (double *)out->data[0];
  
      compressor(s, src, dst, src, in->nb_samples,
                 s->level_in, s->level_in,
                 inlink, inlink);
  
      if (out != in)
          av_frame_free(&in);
      return ff_filter_frame(outlink, out);
  }
  
  static int acompressor_query_formats(AVFilterContext *ctx)
  {
      AVFilterFormats *formats;
      AVFilterChannelLayouts *layouts;
      static const enum AVSampleFormat sample_fmts[] = {
          AV_SAMPLE_FMT_DBL,
          AV_SAMPLE_FMT_NONE
      };
      int ret;
  
      layouts = ff_all_channel_counts();
      if (!layouts)
          return AVERROR(ENOMEM);
      ret = ff_set_common_channel_layouts(ctx, layouts);
      if (ret < 0)
          return ret;
  
      formats = ff_make_format_list(sample_fmts);
      if (!formats)
          return AVERROR(ENOMEM);
      ret = ff_set_common_formats(ctx, formats);
      if (ret < 0)
          return ret;
  
      formats = ff_all_samplerates();
      if (!formats)
          return AVERROR(ENOMEM);
      return ff_set_common_samplerates(ctx, formats);
  }
  
  #define acompressor_options options
  AVFILTER_DEFINE_CLASS(acompressor);
  
  static const AVFilterPad acompressor_inputs[] = {
      {
          .name           = "default",
          .type           = AVMEDIA_TYPE_AUDIO,
          .filter_frame   = acompressor_filter_frame,
      },
      { NULL }
  };
  
  static const AVFilterPad acompressor_outputs[] = {
      {
          .name          = "default",
          .type          = AVMEDIA_TYPE_AUDIO,
          .config_props  = compressor_config_output,
      },
      { NULL }
  };
  
  AVFilter ff_af_acompressor = {
      .name           = "acompressor",
      .description    = NULL_IF_CONFIG_SMALL("Audio compressor."),
      .priv_size      = sizeof(SidechainCompressContext),
      .priv_class     = &acompressor_class,
      .query_formats  = acompressor_query_formats,
      .inputs         = acompressor_inputs,
      .outputs        = acompressor_outputs,
  };
  #endif  /* CONFIG_ACOMPRESSOR_FILTER */
  