/*
  * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit
  *
  * 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) Gate filter
  */
 

 #include "libavutil/audio_fifo.h"

 #include "libavutil/avassert.h"

 #include "libavutil/channel_layout.h"
 #include "libavutil/opt.h"
 #include "avfilter.h"
 #include "audio.h"

 #include "filters.h"

 #include "formats.h"
 #include "hermite.h"
 
 typedef struct AudioGateContext {
     const AVClass *class;
 
     double level_in;

     double level_sc;

     double attack;
     double release;
     double threshold;
     double ratio;
     double knee;
     double makeup;
     double range;
     int link;
     int detection;
 
     double thres;
     double knee_start;
     double lin_knee_stop;
     double knee_stop;
     double lin_slope;
     double attack_coeff;
     double release_coeff;

     AVAudioFifo *fifo[2];
     int64_t pts;

 } AudioGateContext;
 
 #define OFFSET(x) offsetof(AudioGateContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 

 static const AVOption options[] = {

     { "level_in",  "set input level",        OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},           0.015625,   64, A },
     { "range",     "set max gain reduction", OFFSET(range),     AV_OPT_TYPE_DOUBLE, {.dbl=0.06125},     0, 1, A },
     { "threshold", "set threshold",          OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125},       0, 1, A },
     { "ratio",     "set ratio",              OFFSET(ratio),     AV_OPT_TYPE_DOUBLE, {.dbl=2},           1,  9000, A },
     { "attack",    "set attack",             OFFSET(attack),    AV_OPT_TYPE_DOUBLE, {.dbl=20},          0.01, 9000, A },
     { "release",   "set release",            OFFSET(release),   AV_OPT_TYPE_DOUBLE, {.dbl=250},         0.01, 9000, A },
     { "makeup",    "set makeup gain",        OFFSET(makeup),    AV_OPT_TYPE_DOUBLE, {.dbl=1},           1,   64, A },
     { "knee",      "set knee",               OFFSET(knee),      AV_OPT_TYPE_DOUBLE, {.dbl=2.828427125}, 1,    8, A },

     { "detection", "set detection",          OFFSET(detection), AV_OPT_TYPE_INT,    {.i64=1},           0,    1, A, "detection" },

     {   "peak",    0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=0},           0,    0, A, "detection" },
     {   "rms",     0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=1},           0,    0, A, "detection" },
     { "link",      "set link",               OFFSET(link),      AV_OPT_TYPE_INT,    {.i64=0},           0,    1, A, "link" },
     {   "average", 0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=0},           0,    0, A, "link" },
     {   "maximum", 0,                        0,                 AV_OPT_TYPE_CONST,  {.i64=1},           0,    0, A, "link" },

     { "level_sc",  "set sidechain gain",     OFFSET(level_sc),  AV_OPT_TYPE_DOUBLE, {.dbl=1},           0.015625,   64, A },

     { NULL }
 };
 

 static int agate_config_input(AVFilterLink *inlink)

 {
     AVFilterContext *ctx = inlink->dst;
     AudioGateContext *s = ctx->priv;
     double lin_threshold = s->threshold;
     double lin_knee_sqrt = sqrt(s->knee);
     double lin_knee_start;
 
     if (s->detection)
         lin_threshold *= lin_threshold;
 
     s->attack_coeff  = FFMIN(1., 1. / (s->attack * inlink->sample_rate / 4000.));
     s->release_coeff = FFMIN(1., 1. / (s->release * inlink->sample_rate / 4000.));
     s->lin_knee_stop = lin_threshold * lin_knee_sqrt;
     lin_knee_start = lin_threshold / lin_knee_sqrt;
     s->thres = log(lin_threshold);
     s->knee_start = log(lin_knee_start);
     s->knee_stop = log(s->lin_knee_stop);
 
     return 0;
 }
 
 // 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 lin_knee_stop, double range)
 {
     if (lin_slope < lin_knee_stop) {
         double slope = log(lin_slope);
         double tratio = ratio;
         double gain = 0.;
         double delta = 0.;
 
         if (IS_FAKE_INFINITY(ratio))
             tratio = 1000.;
         gain = (slope - thres) * tratio + thres;
         delta = tratio;
 
         if (knee > 1. && slope > knee_start) {
             gain = hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio  + thres), knee_stop, delta, 1.);
         }
         return FFMAX(range, exp(gain - slope));
     }
 
     return 1.;
 }
 

 static void gate(AudioGateContext *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 attack_coeff = s->attack_coeff;
     const double release_coeff = s->release_coeff;
     int n, c;
 

     for (n = 0; n < nb_samples; n++, src += inlink->channels, dst += inlink->channels, scsrc += sclink->channels) {

         double abs_sample = fabs(scsrc[0] * level_sc), gain = 1.0;

 
         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 ? attack_coeff : release_coeff);
         if (s->lin_slope > 0.0)
             gain = output_gain(s->lin_slope, s->ratio, s->thres,
                                s->knee, s->knee_start, s->knee_stop,
                                s->lin_knee_stop, s->range);
 
         for (c = 0; c < inlink->channels; c++)

             dst[c] = src[c] * level_in * gain * makeup;

     }

 }
 

 #if CONFIG_AGATE_FILTER
 
 #define agate_options options
 AVFILTER_DEFINE_CLASS(agate);
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats = NULL;
     AVFilterChannelLayouts *layouts;
     int ret;
 
     if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBL)) < 0)
         return ret;
     ret = ff_set_common_formats(ctx, formats);
     if (ret < 0)
         return 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_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);
 
     return ff_set_common_samplerates(ctx, formats);
 }
 

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     const double *src = (const double *)in->data[0];
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
     AudioGateContext *s = ctx->priv;
     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];
 

     gate(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 const AVFilterPad inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,

         .config_props = agate_config_input,

     },
     { NULL }
 };
 
 static const AVFilterPad outputs[] = {
     {
         .name = "default",
         .type = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 
 AVFilter ff_af_agate = {
     .name           = "agate",
     .description    = NULL_IF_CONFIG_SMALL("Audio gate."),
     .query_formats  = query_formats,
     .priv_size      = sizeof(AudioGateContext),
     .priv_class     = &agate_class,
     .inputs         = inputs,
     .outputs        = outputs,
 };

 #endif /* CONFIG_AGATE_FILTER */
 

 #if CONFIG_SIDECHAINGATE_FILTER
 
 #define sidechaingate_options options
 AVFILTER_DEFINE_CLASS(sidechaingate);
 

 static int activate(AVFilterContext *ctx)

 {
     AudioGateContext *s = ctx->priv;

     AVFrame *out = NULL, *in[2] = { NULL };

     int ret, i, nb_samples;

     double *dst;

     FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);

     if ((ret = ff_inlink_consume_frame(ctx->inputs[0], &in[0])) > 0) {
         av_audio_fifo_write(s->fifo[0], (void **)in[0]->extended_data,
                             in[0]->nb_samples);
         av_frame_free(&in[0]);
     }
     if (ret < 0)
         return ret;
     if ((ret = ff_inlink_consume_frame(ctx->inputs[1], &in[1])) > 0) {
         av_audio_fifo_write(s->fifo[1], (void **)in[1]->extended_data,
                             in[1]->nb_samples);
         av_frame_free(&in[1]);
     }
     if (ret < 0)
         return ret;

     nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));

     if (nb_samples) {
         out = ff_get_audio_buffer(ctx->outputs[0], 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;

         gate(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]);

         ret = ff_filter_frame(ctx->outputs[0], out);
         if (ret < 0)
             return ret;
     }

     FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
     FF_FILTER_FORWARD_STATUS(ctx->inputs[1], ctx->outputs[0]);

     if (ff_outlink_frame_wanted(ctx->outputs[0])) {
         if (!av_audio_fifo_size(s->fifo[0]))
             ff_inlink_request_frame(ctx->inputs[0]);
         if (!av_audio_fifo_size(s->fifo[1]))
             ff_inlink_request_frame(ctx->inputs[1]);
     }
     return 0;

 }
 
 static int scquery_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 scconfig_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;

     AudioGateContext *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);
 
 

     agate_config_input(ctx->inputs[0]);
 
     return 0;
 }
 

 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioGateContext *s = ctx->priv;
 
     av_audio_fifo_free(s->fifo[0]);
     av_audio_fifo_free(s->fifo[1]);
 }
 

 static const AVFilterPad sidechaingate_inputs[] = {
     {
         .name           = "main",
         .type           = AVMEDIA_TYPE_AUDIO,
     },{
         .name           = "sidechain",
         .type           = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 
 static const AVFilterPad sidechaingate_outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .config_props  = scconfig_output,
     },
     { NULL }
 };
 
 AVFilter ff_af_sidechaingate = {
     .name           = "sidechaingate",
     .description    = NULL_IF_CONFIG_SMALL("Audio sidechain gate."),
     .priv_size      = sizeof(AudioGateContext),
     .priv_class     = &sidechaingate_class,
     .query_formats  = scquery_formats,

     .activate       = activate,

     .uninit         = uninit,

     .inputs         = sidechaingate_inputs,
     .outputs        = sidechaingate_outputs,
 };
 #endif  /* CONFIG_SIDECHAINGATE_FILTER */