text: /*
   * Copyright (c) 2016 Kyle Swanson <k@ylo.ph>.
   *
   * 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
   */
  
  /* http://k.ylo.ph/2016/04/04/loudnorm.html */
  
  #include "libavutil/opt.h"
  #include "avfilter.h"
  #include "internal.h"
  #include "audio.h"
  #include "ebur128.h"
  
  enum FrameType {
      FIRST_FRAME,
      INNER_FRAME,
      FINAL_FRAME,
      LINEAR_MODE,
      FRAME_NB
  };
  
  enum LimiterState {
      OUT,
      ATTACK,
      SUSTAIN,
      RELEASE,
      STATE_NB
  };
  
  enum PrintFormat {
      NONE,
      JSON,
      SUMMARY,
      PF_NB
  };
  
  typedef struct LoudNormContext {
      const AVClass *class;
      double target_i;
      double target_lra;
      double target_tp;
      double measured_i;
      double measured_lra;
      double measured_tp;
      double measured_thresh;
      double offset;
      int linear;
      int dual_mono;
      enum PrintFormat print_format;
  
      double *buf;
      int buf_size;
      int buf_index;
      int prev_buf_index;
  
      double delta[30];
      double weights[21];
      double prev_delta;
      int index;
  
      double gain_reduction[2];
      double *limiter_buf;
      double *prev_smp;
      int limiter_buf_index;
      int limiter_buf_size;
      enum LimiterState limiter_state;
      int peak_index;
      int env_index;
      int env_cnt;
      int attack_length;
      int release_length;
  
      int64_t pts;
      enum FrameType frame_type;
      int above_threshold;
      int prev_nb_samples;
      int channels;
  
      FFEBUR128State *r128_in;
      FFEBUR128State *r128_out;
  } LoudNormContext;
  
  #define OFFSET(x) offsetof(LoudNormContext, x)
  #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  
  static const AVOption loudnorm_options[] = {
      { "I",                "set integrated loudness target",    OFFSET(target_i),         AV_OPT_TYPE_DOUBLE,  {.dbl = -24.},   -70.,       -5.,  FLAGS },
      { "i",                "set integrated loudness target",    OFFSET(target_i),         AV_OPT_TYPE_DOUBLE,  {.dbl = -24.},   -70.,       -5.,  FLAGS },
      { "LRA",              "set loudness range target",         OFFSET(target_lra),       AV_OPT_TYPE_DOUBLE,  {.dbl =  7.},     1.,        20.,  FLAGS },
      { "lra",              "set loudness range target",         OFFSET(target_lra),       AV_OPT_TYPE_DOUBLE,  {.dbl =  7.},     1.,        20.,  FLAGS },
      { "TP",               "set maximum true peak",             OFFSET(target_tp),        AV_OPT_TYPE_DOUBLE,  {.dbl = -2.},    -9.,         0.,  FLAGS },
      { "tp",               "set maximum true peak",             OFFSET(target_tp),        AV_OPT_TYPE_DOUBLE,  {.dbl = -2.},    -9.,         0.,  FLAGS },
      { "measured_I",       "measured IL of input file",         OFFSET(measured_i),       AV_OPT_TYPE_DOUBLE,  {.dbl =  0.},    -99.,        0.,  FLAGS },
      { "measured_i",       "measured IL of input file",         OFFSET(measured_i),       AV_OPT_TYPE_DOUBLE,  {.dbl =  0.},    -99.,        0.,  FLAGS },
      { "measured_LRA",     "measured LRA of input file",        OFFSET(measured_lra),     AV_OPT_TYPE_DOUBLE,  {.dbl =  0.},     0.,        99.,  FLAGS },
      { "measured_lra",     "measured LRA of input file",        OFFSET(measured_lra),     AV_OPT_TYPE_DOUBLE,  {.dbl =  0.},     0.,        99.,  FLAGS },
      { "measured_TP",      "measured true peak of input file",  OFFSET(measured_tp),      AV_OPT_TYPE_DOUBLE,  {.dbl =  99.},   -99.,       99.,  FLAGS },
      { "measured_tp",      "measured true peak of input file",  OFFSET(measured_tp),      AV_OPT_TYPE_DOUBLE,  {.dbl =  99.},   -99.,       99.,  FLAGS },
      { "measured_thresh",  "measured threshold of input file",  OFFSET(measured_thresh),  AV_OPT_TYPE_DOUBLE,  {.dbl = -70.},   -99.,        0.,  FLAGS },
      { "offset",           "set offset gain",                   OFFSET(offset),           AV_OPT_TYPE_DOUBLE,  {.dbl =  0.},    -99.,       99.,  FLAGS },
      { "linear",           "normalize linearly if possible",    OFFSET(linear),           AV_OPT_TYPE_BOOL,    {.i64 =  1},        0,         1,  FLAGS },
      { "dual_mono",        "treat mono input as dual-mono",     OFFSET(dual_mono),        AV_OPT_TYPE_BOOL,    {.i64 =  0},        0,         1,  FLAGS },
      { "print_format",     "set print format for stats",        OFFSET(print_format),     AV_OPT_TYPE_INT,     {.i64 =  NONE},  NONE,  PF_NB -1,  FLAGS, "print_format" },
      {     "none",         0,                                   0,                        AV_OPT_TYPE_CONST,   {.i64 =  NONE},     0,         0,  FLAGS, "print_format" },
      {     "json",         0,                                   0,                        AV_OPT_TYPE_CONST,   {.i64 =  JSON},     0,         0,  FLAGS, "print_format" },
      {     "summary",      0,                                   0,                        AV_OPT_TYPE_CONST,   {.i64 =  SUMMARY},  0,         0,  FLAGS, "print_format" },
      { NULL }
  };
  
  AVFILTER_DEFINE_CLASS(loudnorm);
  
  static inline int frame_size(int sample_rate, int frame_len_msec)
  {
      const int frame_size = round((double)sample_rate * (frame_len_msec / 1000.0));
      return frame_size + (frame_size % 2);
  }
  
  static void init_gaussian_filter(LoudNormContext *s)
  {
      double total_weight = 0.0;
      const double sigma = 3.5;
      double adjust;
      int i;
  
      const int offset = 21 / 2;
      const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI));
      const double c2 = 2.0 * pow(sigma, 2.0);
  
      for (i = 0; i < 21; i++) {
          const int x = i - offset;
          s->weights[i] = c1 * exp(-(pow(x, 2.0) / c2));
          total_weight += s->weights[i];
      }
  
      adjust = 1.0 / total_weight;
      for (i = 0; i < 21; i++)
          s->weights[i] *= adjust;
  }
  
  static double gaussian_filter(LoudNormContext *s, int index)
  {
      double result = 0.;
      int i;
  
      index = index - 10 > 0 ? index - 10 : index + 20;
      for (i = 0; i < 21; i++)
          result += s->delta[((index + i) < 30) ? (index + i) : (index + i - 30)] * s->weights[i];
  
      return result;
  }
  
  static void detect_peak(LoudNormContext *s, int offset, int nb_samples, int channels, int *peak_delta, double *peak_value)
  {
      int n, c, i, index;
      double ceiling;
      double *buf;
  
      *peak_delta = -1;
      buf = s->limiter_buf;
      ceiling = s->target_tp;
  
      index = s->limiter_buf_index + (offset * channels) + (1920 * channels);
      if (index >= s->limiter_buf_size)
          index -= s->limiter_buf_size;
  
      if (s->frame_type == FIRST_FRAME) {
          for (c = 0; c < channels; c++)
              s->prev_smp[c] = fabs(buf[index + c - channels]);
      }
  
      for (n = 0; n < nb_samples; n++) {
          for (c = 0; c < channels; c++) {
              double this, next, max_peak;
  
              this = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]);
              next = fabs(buf[(index + c + channels) < s->limiter_buf_size ? (index + c + channels) : (index + c + channels - s->limiter_buf_size)]);
  
              if ((s->prev_smp[c] <= this) && (next <= this) && (this > ceiling) && (n > 0)) {
                  int detected;
  
                  detected = 1;
                  for (i = 2; i < 12; i++) {
                      next = fabs(buf[(index + c + (i * channels)) < s->limiter_buf_size ? (index + c + (i * channels)) : (index + c + (i * channels) - s->limiter_buf_size)]);
                      if (next > this) {
                          detected = 0;
                          break;
                      }
                  }
  
                  if (!detected)
                      continue;
  
                  for (c = 0; c < channels; c++) {
                      if (c == 0 || fabs(buf[index + c]) > max_peak)
                          max_peak = fabs(buf[index + c]);
  
                      s->prev_smp[c] = fabs(buf[(index + c) < s->limiter_buf_size ? (index + c) : (index + c - s->limiter_buf_size)]);
                  }
  
                  *peak_delta = n;
                  s->peak_index = index;
                  *peak_value = max_peak;
                  return;
              }
  
              s->prev_smp[c] = this;
          }
  
          index += channels;
          if (index >= s->limiter_buf_size)
              index -= s->limiter_buf_size;
      }
  }
  
  static void true_peak_limiter(LoudNormContext *s, double *out, int nb_samples, int channels)
  {
      int n, c, index, peak_delta, smp_cnt;
      double ceiling, peak_value;
      double *buf;
  
      buf = s->limiter_buf;
      ceiling = s->target_tp;
      index = s->limiter_buf_index;
      smp_cnt = 0;
  
      if (s->frame_type == FIRST_FRAME) {
          double max;
  
          max = 0.;
          for (n = 0; n < 1920; n++) {
              for (c = 0; c < channels; c++) {
                max = fabs(buf[c]) > max ? fabs(buf[c]) : max;
              }
              buf += channels;
          }
  
          if (max > ceiling) {
              s->gain_reduction[1] = ceiling / max;
              s->limiter_state = SUSTAIN;
              buf = s->limiter_buf;
  
              for (n = 0; n < 1920; n++) {
                  for (c = 0; c < channels; c++) {
                      double env;
                      env = s->gain_reduction[1];
                      buf[c] *= env;
                  }
                  buf += channels;
              }
          }
  
          buf = s->limiter_buf;
      }
  
      do {
  
          switch(s->limiter_state) {
          case OUT:
              detect_peak(s, smp_cnt, nb_samples - smp_cnt, channels, &peak_delta, &peak_value);
              if (peak_delta != -1) {
                  s->env_cnt = 0;
                  smp_cnt += (peak_delta - s->attack_length);
                  s->gain_reduction[0] = 1.;
                  s->gain_reduction[1] = ceiling / peak_value;
                  s->limiter_state = ATTACK;
  
                  s->env_index = s->peak_index - (s->attack_length * channels);
                  if (s->env_index < 0)
                      s->env_index += s->limiter_buf_size;
  
                  s->env_index += (s->env_cnt * channels);
                  if (s->env_index > s->limiter_buf_size)
                      s->env_index -= s->limiter_buf_size;
  
              } else {
                  smp_cnt = nb_samples;
              }
              break;
  
          case ATTACK:
              for (; s->env_cnt < s->attack_length; s->env_cnt++) {
                  for (c = 0; c < channels; c++) {
                      double env;
                      env = s->gain_reduction[0] - ((double) s->env_cnt / (s->attack_length - 1) * (s->gain_reduction[0] - s->gain_reduction[1]));
                      buf[s->env_index + c] *= env;
                  }
  
                  s->env_index += channels;
                  if (s->env_index >= s->limiter_buf_size)
                      s->env_index -= s->limiter_buf_size;
  
                  smp_cnt++;
                  if (smp_cnt >= nb_samples) {
                      s->env_cnt++;
                      break;
                  }
              }
  
              if (smp_cnt < nb_samples) {
                  s->env_cnt = 0;
                  s->attack_length = 1920;
                  s->limiter_state = SUSTAIN;
              }
              break;
  
          case SUSTAIN:
              detect_peak(s, smp_cnt, nb_samples, channels, &peak_delta, &peak_value);
              if (peak_delta == -1) {
                  s->limiter_state = RELEASE;
                  s->gain_reduction[0] = s->gain_reduction[1];
                  s->gain_reduction[1] = 1.;
                  s->env_cnt = 0;
                  break;
              } else {
                  double gain_reduction;
                  gain_reduction = ceiling / peak_value;
  
                  if (gain_reduction < s->gain_reduction[1]) {
                      s->limiter_state = ATTACK;
  
                      s->attack_length = peak_delta;
                      if (s->attack_length <= 1)
                          s->attack_length =  2;
  
                      s->gain_reduction[0] = s->gain_reduction[1];
                      s->gain_reduction[1] = gain_reduction;
                      s->env_cnt = 0;
                      break;
                  }
  
                  for (s->env_cnt = 0; s->env_cnt < peak_delta; s->env_cnt++) {
                      for (c = 0; c < channels; c++) {
                          double env;
                          env = s->gain_reduction[1];
                          buf[s->env_index + c] *= env;
                      }
  
                      s->env_index += channels;
                      if (s->env_index >= s->limiter_buf_size)
                          s->env_index -= s->limiter_buf_size;
  
                      smp_cnt++;
                      if (smp_cnt >= nb_samples) {
                          s->env_cnt++;
                          break;
                      }
                  }
              }
              break;
  
          case RELEASE:
              for (; s->env_cnt < s->release_length; s->env_cnt++) {
                  for (c = 0; c < channels; c++) {
                      double env;
                      env = s->gain_reduction[0] + (((double) s->env_cnt / (s->release_length - 1)) * (s->gain_reduction[1] - s->gain_reduction[0]));
                      buf[s->env_index + c] *= env;
                  }
  
                  s->env_index += channels;
                  if (s->env_index >= s->limiter_buf_size)
                      s->env_index -= s->limiter_buf_size;
  
                  smp_cnt++;
                  if (smp_cnt >= nb_samples) {
                      s->env_cnt++;
                      break;
                  }
              }
  
              if (smp_cnt < nb_samples) {
                  s->env_cnt = 0;
                  s->limiter_state = OUT;
              }
  
              break;
          }
  
      } while (smp_cnt < nb_samples);
  
      for (n = 0; n < nb_samples; n++) {
          for (c = 0; c < channels; c++) {
              out[c] = buf[index + c];
              if (fabs(out[c]) > ceiling) {
                  out[c] = ceiling * (out[c] < 0 ? -1 : 1);
              }
          }
          out += channels;
          index += channels;
          if (index >= s->limiter_buf_size)
              index -= s->limiter_buf_size;
      }
  }
  
  static int filter_frame(AVFilterLink *inlink, AVFrame *in)
  {
      AVFilterContext *ctx = inlink->dst;
      LoudNormContext *s = ctx->priv;
      AVFilterLink *outlink = ctx->outputs[0];
      AVFrame *out;
      const double *src;
      double *dst;
      double *buf;
      double *limiter_buf;
      int i, n, c, subframe_length, src_index;
      double gain, gain_next, env_global, env_shortterm,
      global, shortterm, lra, relative_threshold;
  
      if (av_frame_is_writable(in)) {
          out = in;
      } else {
          out = ff_get_audio_buffer(outlink, in->nb_samples);
          if (!out) {
              av_frame_free(&in);
              return AVERROR(ENOMEM);
          }
          av_frame_copy_props(out, in);
      }
  
      if (s->pts == AV_NOPTS_VALUE)
          s->pts = in->pts;
  
      out->pts = s->pts;
      src = (const double *)in->data[0];
      dst = (double *)out->data[0];
      buf = s->buf;
      limiter_buf = s->limiter_buf;
  
      ff_ebur128_add_frames_double(s->r128_in, src, in->nb_samples);
  
      if (s->frame_type == FIRST_FRAME && in->nb_samples < frame_size(inlink->sample_rate, 3000)) {
          double offset, offset_tp, true_peak;
  
          ff_ebur128_loudness_global(s->r128_in, &global);
          for (c = 0; c < inlink->channels; c++) {
              double tmp;
              ff_ebur128_sample_peak(s->r128_in, c, &tmp);
              if (c == 0 || tmp > true_peak)
                  true_peak = tmp;
          }
  
          offset    = pow(10., (s->target_i - global) / 20.);
          offset_tp = true_peak * offset;
          s->offset = offset_tp < s->target_tp ? offset : s->target_tp - true_peak;
          s->frame_type = LINEAR_MODE;
      }
  
      switch (s->frame_type) {
      case FIRST_FRAME:
          for (n = 0; n < in->nb_samples; n++) {
              for (c = 0; c < inlink->channels; c++) {
                  buf[s->buf_index + c] = src[c];
              }
              src += inlink->channels;
              s->buf_index += inlink->channels;
          }
  
          ff_ebur128_loudness_shortterm(s->r128_in, &shortterm);
  
          if (shortterm < s->measured_thresh) {
              s->above_threshold = 0;
              env_shortterm = shortterm <= -70. ? 0. : s->target_i - s->measured_i;
          } else {
              s->above_threshold = 1;
              env_shortterm = shortterm <= -70. ? 0. : s->target_i - shortterm;
          }
  
          for (n = 0; n < 30; n++)
              s->delta[n] = pow(10., env_shortterm / 20.);
          s->prev_delta = s->delta[s->index];
  
          s->buf_index =
          s->limiter_buf_index = 0;
  
          for (n = 0; n < (s->limiter_buf_size / inlink->channels); n++) {
              for (c = 0; c < inlink->channels; c++) {
                  limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * s->delta[s->index] * s->offset;
              }
              s->limiter_buf_index += inlink->channels;
              if (s->limiter_buf_index >= s->limiter_buf_size)
                  s->limiter_buf_index -= s->limiter_buf_size;
  
              s->buf_index += inlink->channels;
          }
  
          subframe_length = frame_size(inlink->sample_rate, 100);
          true_peak_limiter(s, dst, subframe_length, inlink->channels);
          ff_ebur128_add_frames_double(s->r128_out, dst, subframe_length);
  
          s->pts +=
          out->nb_samples =
          inlink->min_samples =
          inlink->max_samples = subframe_length;
  
          s->frame_type = INNER_FRAME;
          break;
  
      case INNER_FRAME:
          gain      = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30);
          gain_next = gaussian_filter(s, s->index + 11 < 30 ? s->index + 11 : s->index + 11 - 30);
  
          for (n = 0; n < in->nb_samples; n++) {
              for (c = 0; c < inlink->channels; c++) {
                  buf[s->prev_buf_index + c] = src[c];
                  limiter_buf[s->limiter_buf_index + c] = buf[s->buf_index + c] * (gain + (((double) n / in->nb_samples) * (gain_next - gain))) * s->offset;
              }
              src += inlink->channels;
  
              s->limiter_buf_index += inlink->channels;
              if (s->limiter_buf_index >= s->limiter_buf_size)
                  s->limiter_buf_index -= s->limiter_buf_size;
  
              s->prev_buf_index += inlink->channels;
              if (s->prev_buf_index >= s->buf_size)
                  s->prev_buf_index -= s->buf_size;
  
              s->buf_index += inlink->channels;
              if (s->buf_index >= s->buf_size)
                  s->buf_index -= s->buf_size;
          }
  
          subframe_length = (frame_size(inlink->sample_rate, 100) - in->nb_samples) * inlink->channels;
          s->limiter_buf_index = s->limiter_buf_index + subframe_length < s->limiter_buf_size ? s->limiter_buf_index + subframe_length : s->limiter_buf_index + subframe_length - s->limiter_buf_size;
  
          true_peak_limiter(s, dst, in->nb_samples, inlink->channels);
          ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
  
          ff_ebur128_loudness_range(s->r128_in, &lra);
          ff_ebur128_loudness_global(s->r128_in, &global);
          ff_ebur128_loudness_shortterm(s->r128_in, &shortterm);
          ff_ebur128_relative_threshold(s->r128_in, &relative_threshold);
  
          if (s->above_threshold == 0) {
              double shortterm_out;
  
              if (shortterm > s->measured_thresh)
                  s->prev_delta *= 1.0058;
  
              ff_ebur128_loudness_shortterm(s->r128_out, &shortterm_out);
              if (shortterm_out >= s->target_i)
                  s->above_threshold = 1;
          }
  
          if (shortterm < relative_threshold || shortterm <= -70. || s->above_threshold == 0) {
              s->delta[s->index] = s->prev_delta;
          } else {
              env_global = fabs(shortterm - global) < (s->target_lra / 2.) ? shortterm - global : (s->target_lra / 2.) * ((shortterm - global) < 0 ? -1 : 1);
              env_shortterm = s->target_i - shortterm;
              s->delta[s->index] = pow(10., (env_global + env_shortterm) / 20.);
          }
  
          s->prev_delta = s->delta[s->index];
          s->index++;
          if (s->index >= 30)
              s->index -= 30;
          s->prev_nb_samples = in->nb_samples;
          s->pts += in->nb_samples;
          break;
  
      case FINAL_FRAME:
          gain = gaussian_filter(s, s->index + 10 < 30 ? s->index + 10 : s->index + 10 - 30);
          s->limiter_buf_index = 0;
          src_index = 0;
  
          for (n = 0; n < s->limiter_buf_size / inlink->channels; n++) {
              for (c = 0; c < inlink->channels; c++) {
                  s->limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset;
              }
              src_index += inlink->channels;
  
              s->limiter_buf_index += inlink->channels;
              if (s->limiter_buf_index >= s->limiter_buf_size)
                  s->limiter_buf_index -= s->limiter_buf_size;
          }
  
          subframe_length = frame_size(inlink->sample_rate, 100);
          for (i = 0; i < in->nb_samples / subframe_length; i++) {
              true_peak_limiter(s, dst, subframe_length, inlink->channels);
  
              for (n = 0; n < subframe_length; n++) {
                  for (c = 0; c < inlink->channels; c++) {
                      if (src_index < (in->nb_samples * inlink->channels)) {
                          limiter_buf[s->limiter_buf_index + c] = src[src_index + c] * gain * s->offset;
                      } else {
                          limiter_buf[s->limiter_buf_index + c] = 0.;
                      }
                  }
  
                  if (src_index < (in->nb_samples * inlink->channels))
                      src_index += inlink->channels;
  
                  s->limiter_buf_index += inlink->channels;
                  if (s->limiter_buf_index >= s->limiter_buf_size)
                      s->limiter_buf_index -= s->limiter_buf_size;
              }
  
              dst += (subframe_length * inlink->channels);
          }
  
          dst = (double *)out->data[0];
          ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
          break;
  
      case LINEAR_MODE:
          for (n = 0; n < in->nb_samples; n++) {
              for (c = 0; c < inlink->channels; c++) {
                  dst[c] = src[c] * s->offset;
              }
              src += inlink->channels;
              dst += inlink->channels;
          }
  
          dst = (double *)out->data[0];
          ff_ebur128_add_frames_double(s->r128_out, dst, in->nb_samples);
          s->pts += in->nb_samples;
          break;
      }
  
      if (in != out)
          av_frame_free(&in);
  
      return ff_filter_frame(outlink, out);
  }
  
  static int request_frame(AVFilterLink *outlink)
  {
      int ret;
      AVFilterContext *ctx = outlink->src;
      AVFilterLink *inlink = ctx->inputs[0];
      LoudNormContext *s = ctx->priv;
  
      ret = ff_request_frame(inlink);
      if (ret == AVERROR_EOF && s->frame_type == INNER_FRAME) {
          double *src;
          double *buf;
          int nb_samples, n, c, offset;
          AVFrame *frame;
  
          nb_samples  = (s->buf_size / inlink->channels) - s->prev_nb_samples;
          nb_samples -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples);
  
          frame = ff_get_audio_buffer(outlink, nb_samples);
          if (!frame)
              return AVERROR(ENOMEM);
          frame->nb_samples = nb_samples;
  
          buf = s->buf;
          src = (double *)frame->data[0];
  
          offset  = ((s->limiter_buf_size / inlink->channels) - s->prev_nb_samples) * inlink->channels;
          offset -= (frame_size(inlink->sample_rate, 100) - s->prev_nb_samples) * inlink->channels;
          s->buf_index = s->buf_index - offset < 0 ? s->buf_index - offset + s->buf_size : s->buf_index - offset;
  
          for (n = 0; n < nb_samples; n++) {
              for (c = 0; c < inlink->channels; c++) {
                  src[c] = buf[s->buf_index + c];
              }
              src += inlink->channels;
              s->buf_index += inlink->channels;
              if (s->buf_index >= s->buf_size)
                  s->buf_index -= s->buf_size;
          }
  
          s->frame_type = FINAL_FRAME;
          ret = filter_frame(inlink, frame);
      }
      return ret;
  }
  
  static int query_formats(AVFilterContext *ctx)
  {
      LoudNormContext *s = ctx->priv;
      AVFilterFormats *formats;
      AVFilterLink *inlink = ctx->inputs[0];
      AVFilterLink *outlink = ctx->outputs[0];
      static const int input_srate[] = {192000, -1};
      static const enum AVSampleFormat sample_fmts[] = {
          AV_SAMPLE_FMT_DBL,
          AV_SAMPLE_FMT_NONE
      };
      int ret = ff_set_common_all_channel_counts(ctx);
      if (ret < 0)
          return ret;
  
      ret = ff_set_common_formats_from_list(ctx, sample_fmts);
      if (ret < 0)
          return ret;
  
      if (s->frame_type != LINEAR_MODE) {
          formats = ff_make_format_list(input_srate);
          if (!formats)
              return AVERROR(ENOMEM);
          ret = ff_formats_ref(formats, &inlink->outcfg.samplerates);
          if (ret < 0)
              return ret;
          ret = ff_formats_ref(formats, &outlink->incfg.samplerates);
          if (ret < 0)
              return ret;
      }
  
      return 0;
  }
  
  static int config_input(AVFilterLink *inlink)
  {
      AVFilterContext *ctx = inlink->dst;
      LoudNormContext *s = ctx->priv;
  
      s->r128_in = ff_ebur128_init(inlink->channels, inlink->sample_rate, 0, FF_EBUR128_MODE_I | FF_EBUR128_MODE_S | FF_EBUR128_MODE_LRA | FF_EBUR128_MODE_SAMPLE_PEAK);
      if (!s->r128_in)
          return AVERROR(ENOMEM);
  
      s->r128_out = ff_ebur128_init(inlink->channels, inlink->sample_rate, 0, FF_EBUR128_MODE_I | FF_EBUR128_MODE_S | FF_EBUR128_MODE_LRA | FF_EBUR128_MODE_SAMPLE_PEAK);
      if (!s->r128_out)
          return AVERROR(ENOMEM);
  
      if (inlink->channels == 1 && s->dual_mono) {
          ff_ebur128_set_channel(s->r128_in,  0, FF_EBUR128_DUAL_MONO);
          ff_ebur128_set_channel(s->r128_out, 0, FF_EBUR128_DUAL_MONO);
      }
  
      s->buf_size = frame_size(inlink->sample_rate, 3000) * inlink->channels;
      s->buf = av_malloc_array(s->buf_size, sizeof(*s->buf));
      if (!s->buf)
          return AVERROR(ENOMEM);
  
      s->limiter_buf_size = frame_size(inlink->sample_rate, 210) * inlink->channels;
      s->limiter_buf = av_malloc_array(s->buf_size, sizeof(*s->limiter_buf));
      if (!s->limiter_buf)
          return AVERROR(ENOMEM);
  
      s->prev_smp = av_malloc_array(inlink->channels, sizeof(*s->prev_smp));
      if (!s->prev_smp)
          return AVERROR(ENOMEM);
  
      init_gaussian_filter(s);
  
      if (s->frame_type != LINEAR_MODE) {
          inlink->min_samples =
          inlink->max_samples = frame_size(inlink->sample_rate, 3000);
      }
  
      s->pts = AV_NOPTS_VALUE;
      s->buf_index =
      s->prev_buf_index =
      s->limiter_buf_index = 0;
      s->channels = inlink->channels;
      s->index = 1;
      s->limiter_state = OUT;
      s->offset = pow(10., s->offset / 20.);
      s->target_tp = pow(10., s->target_tp / 20.);
      s->attack_length = frame_size(inlink->sample_rate, 10);
      s->release_length = frame_size(inlink->sample_rate, 100);
  
      return 0;
  }
  
  static av_cold int init(AVFilterContext *ctx)
  {
      LoudNormContext *s = ctx->priv;
      s->frame_type = FIRST_FRAME;
  
      if (s->linear) {
          double offset, offset_tp;
          offset    = s->target_i - s->measured_i;
          offset_tp = s->measured_tp + offset;
  
          if (s->measured_tp != 99 && s->measured_thresh != -70 && s->measured_lra != 0 && s->measured_i != 0) {
              if ((offset_tp <= s->target_tp) && (s->measured_lra <= s->target_lra)) {
                  s->frame_type = LINEAR_MODE;
                  s->offset = offset;
              }
          }
      }
  
      return 0;
  }
  
  static av_cold void uninit(AVFilterContext *ctx)
  {
      LoudNormContext *s = ctx->priv;
      double i_in, i_out, lra_in, lra_out, thresh_in, thresh_out, tp_in, tp_out;
      int c;
  
      if (!s->r128_in || !s->r128_out)
          goto end;
  
      ff_ebur128_loudness_range(s->r128_in, &lra_in);
      ff_ebur128_loudness_global(s->r128_in, &i_in);
      ff_ebur128_relative_threshold(s->r128_in, &thresh_in);
      for (c = 0; c < s->channels; c++) {
          double tmp;
          ff_ebur128_sample_peak(s->r128_in, c, &tmp);
          if ((c == 0) || (tmp > tp_in))
              tp_in = tmp;
      }
  
      ff_ebur128_loudness_range(s->r128_out, &lra_out);
      ff_ebur128_loudness_global(s->r128_out, &i_out);
      ff_ebur128_relative_threshold(s->r128_out, &thresh_out);
      for (c = 0; c < s->channels; c++) {
          double tmp;
          ff_ebur128_sample_peak(s->r128_out, c, &tmp);
          if ((c == 0) || (tmp > tp_out))
              tp_out = tmp;
      }
  
      switch(s->print_format) {
      case NONE:
          break;
  
      case JSON:
          av_log(ctx, AV_LOG_INFO,
              "\n{\n"
              "\t\"input_i\" : \"%.2f\",\n"
              "\t\"input_tp\" : \"%.2f\",\n"
              "\t\"input_lra\" : \"%.2f\",\n"
              "\t\"input_thresh\" : \"%.2f\",\n"
              "\t\"output_i\" : \"%.2f\",\n"
              "\t\"output_tp\" : \"%+.2f\",\n"
              "\t\"output_lra\" : \"%.2f\",\n"
              "\t\"output_thresh\" : \"%.2f\",\n"
              "\t\"normalization_type\" : \"%s\",\n"
              "\t\"target_offset\" : \"%.2f\"\n"
              "}\n",
              i_in,
              20. * log10(tp_in),
              lra_in,
              thresh_in,
              i_out,
              20. * log10(tp_out),
              lra_out,
              thresh_out,
              s->frame_type == LINEAR_MODE ? "linear" : "dynamic",
              s->target_i - i_out
          );
          break;
  
      case SUMMARY:
          av_log(ctx, AV_LOG_INFO,
              "\n"
              "Input Integrated:   %+6.1f LUFS\n"
              "Input True Peak:    %+6.1f dBTP\n"
              "Input LRA:          %6.1f LU\n"
              "Input Threshold:    %+6.1f LUFS\n"
              "\n"
              "Output Integrated:  %+6.1f LUFS\n"
              "Output True Peak:   %+6.1f dBTP\n"
              "Output LRA:         %6.1f LU\n"
              "Output Threshold:   %+6.1f LUFS\n"
              "\n"
              "Normalization Type:   %s\n"
              "Target Offset:      %+6.1f LU\n",
              i_in,
              20. * log10(tp_in),
              lra_in,
              thresh_in,
              i_out,
              20. * log10(tp_out),
              lra_out,
              thresh_out,
              s->frame_type == LINEAR_MODE ? "Linear" : "Dynamic",
              s->target_i - i_out
          );
          break;
      }
  
  end:
      if (s->r128_in)
          ff_ebur128_destroy(&s->r128_in);
      if (s->r128_out)
          ff_ebur128_destroy(&s->r128_out);
      av_freep(&s->limiter_buf);
      av_freep(&s->prev_smp);
      av_freep(&s->buf);
  }
  
  static const AVFilterPad avfilter_af_loudnorm_inputs[] = {
      {
          .name         = "default",
          .type         = AVMEDIA_TYPE_AUDIO,
          .config_props = config_input,
          .filter_frame = filter_frame,
      },
  };
  
  static const AVFilterPad avfilter_af_loudnorm_outputs[] = {
      {
          .name          = "default",
          .request_frame = request_frame,
          .type          = AVMEDIA_TYPE_AUDIO,
      },
  };
  
  const AVFilter ff_af_loudnorm = {
      .name          = "loudnorm",
      .description   = NULL_IF_CONFIG_SMALL("EBU R128 loudness normalization"),
      .priv_size     = sizeof(LoudNormContext),
      .priv_class    = &loudnorm_class,
      .query_formats = query_formats,
      .init          = init,
      .uninit        = uninit,
      FILTER_INPUTS(avfilter_af_loudnorm_inputs),
      FILTER_OUTPUTS(avfilter_af_loudnorm_outputs),
  };
  