text: /*
   * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
   *
   * 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/avstring.h"
  #include "libavutil/opt.h"
  #include "libavutil/samplefmt.h"
  #include "avfilter.h"
  #include "audio.h"
  #include "internal.h"
  #include "generate_wave_table.h"
  
  #define INTERPOLATION_LINEAR    0
  #define INTERPOLATION_QUADRATIC 1
  
  typedef struct FlangerContext {
      const AVClass *class;
      double delay_min;
      double delay_depth;
      double feedback_gain;
      double delay_gain;
      double speed;
      int wave_shape;
      double channel_phase;
      int interpolation;
      double in_gain;
      int max_samples;
      uint8_t **delay_buffer;
      int delay_buf_pos;
      double *delay_last;
      float *lfo;
      int lfo_length;
      int lfo_pos;
  } FlangerContext;
  
  #define OFFSET(x) offsetof(FlangerContext, x)
  #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
  
  static const AVOption flanger_options[] = {
      { "delay", "base delay in milliseconds",        OFFSET(delay_min),   AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
      { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
      { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
      { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
      { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
      { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
      { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, A, "type" },
      { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, A, "type" },
      { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, A, "type" },
      { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, A, "type" },
      { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
      { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
      { "linear",     NULL, 0, AV_OPT_TYPE_CONST,  {.i64=INTERPOLATION_LINEAR},    0, 0, A, "itype" },
      { "quadratic",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
      { NULL }
  };
  
  AVFILTER_DEFINE_CLASS(flanger);
  
  static int init(AVFilterContext *ctx)
  {
      FlangerContext *s = ctx->priv;
  
      s->feedback_gain /= 100;
      s->delay_gain    /= 100;
      s->channel_phase /= 100;
      s->delay_min     /= 1000;
      s->delay_depth   /= 1000;
      s->in_gain        = 1 / (1 + s->delay_gain);
      s->delay_gain    /= 1 + s->delay_gain;
      s->delay_gain    *= 1 - fabs(s->feedback_gain);
  
      return 0;
  }
  
  static int query_formats(AVFilterContext *ctx)
  {
      AVFilterChannelLayouts *layouts;
      AVFilterFormats *formats;
      static const enum AVSampleFormat sample_fmts[] = {
          AV_SAMPLE_FMT_DBLP, 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);
  }
  
  static int config_input(AVFilterLink *inlink)
  {
      AVFilterContext *ctx = inlink->dst;
      FlangerContext *s = ctx->priv;
  
      s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
      s->lfo_length  = inlink->sample_rate / s->speed;
      s->delay_last  = av_calloc(inlink->channels, sizeof(*s->delay_last));
      s->lfo         = av_calloc(s->lfo_length, sizeof(*s->lfo));
      if (!s->lfo || !s->delay_last)
          return AVERROR(ENOMEM);
  
      ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
                             rint(s->delay_min * inlink->sample_rate),
                             s->max_samples - 2., 3 * M_PI_2);
  
      return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
                                                inlink->channels, s->max_samples,
                                                inlink->format, 0);
  }
  
  static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
  {
      AVFilterContext *ctx = inlink->dst;
      FlangerContext *s = ctx->priv;
      AVFrame *out_frame;
      int chan, i;
  
      if (av_frame_is_writable(frame)) {
          out_frame = frame;
      } else {
          out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
          if (!out_frame) {
              av_frame_free(&frame);
              return AVERROR(ENOMEM);
          }
          av_frame_copy_props(out_frame, frame);
      }
  
      for (i = 0; i < frame->nb_samples; i++) {
  
          s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
  
          for (chan = 0; chan < inlink->channels; chan++) {
              double *src = (double *)frame->extended_data[chan];
              double *dst = (double *)out_frame->extended_data[chan];
              double delayed_0, delayed_1;
              double delayed;
              double in, out;
              int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
              double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
              int int_delay = (int)delay;
              double frac_delay = modf(delay, &delay);
              double *delay_buffer = (double *)s->delay_buffer[chan];
  
              in = src[i];
              delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
                                                             s->feedback_gain;
              delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
              delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
  
              if (s->interpolation == INTERPOLATION_LINEAR) {
                  delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
              } else {
                  double a, b;
                  double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
                  delayed_2 -= delayed_0;
                  delayed_1 -= delayed_0;
                  a = delayed_2 * .5 - delayed_1;
                  b = delayed_1 *  2 - delayed_2 *.5;
                  delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
              }
  
              s->delay_last[chan] = delayed;
              out = in * s->in_gain + delayed * s->delay_gain;
              dst[i] = out;
          }
          s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
      }
  
      if (frame != out_frame)
          av_frame_free(&frame);
  
      return ff_filter_frame(ctx->outputs[0], out_frame);
  }
  
  static av_cold void uninit(AVFilterContext *ctx)
  {
      FlangerContext *s = ctx->priv;
  
      av_freep(&s->lfo);
      av_freep(&s->delay_last);
  
      if (s->delay_buffer)
          av_freep(&s->delay_buffer[0]);
      av_freep(&s->delay_buffer);
  }
  
  static const AVFilterPad flanger_inputs[] = {
      {
          .name         = "default",
          .type         = AVMEDIA_TYPE_AUDIO,
          .config_props = config_input,
          .filter_frame = filter_frame,
      },
      { NULL }
  };
  
  static const AVFilterPad flanger_outputs[] = {
      {
          .name          = "default",
          .type          = AVMEDIA_TYPE_AUDIO,
      },
      { NULL }
  };
  
  AVFilter ff_af_flanger = {
      .name          = "flanger",
      .description   = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
      .query_formats = query_formats,
      .priv_size     = sizeof(FlangerContext),
      .priv_class    = &flanger_class,
      .init          = init,
      .uninit        = uninit,
      .inputs        = flanger_inputs,
      .outputs       = flanger_outputs,
  };
  