/*
  * 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
  * Lookahead limiter filter
  */
 
 #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 "internal.h"
 
 typedef struct AudioLimiterContext {
     const AVClass *class;
 
     double limit;
     double attack;
     double release;
     double att;

     double level_in;
     double level_out;

     int auto_release;

     int auto_level;

     double asc;
     int asc_c;
     int asc_pos;
     double asc_coeff;
 
     double *buffer;
     int buffer_size;
     int pos;
     int *nextpos;
     double *nextdelta;
 
     double delta;
     int nextiter;
     int nextlen;
     int asc_changed;
 } AudioLimiterContext;
 
 #define OFFSET(x) offsetof(AudioLimiterContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM
 #define F AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption alimiter_options[] = {

     { "level_in",  "set input level",  OFFSET(level_in),     AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625,   64, A|F },
     { "level_out", "set output level", OFFSET(level_out),    AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625,   64, A|F },
     { "limit",     "set limit",        OFFSET(limit),        AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625,    1, A|F },
     { "attack",    "set attack",       OFFSET(attack),       AV_OPT_TYPE_DOUBLE, {.dbl=5},    0.1,   80, A|F },
     { "release",   "set release",      OFFSET(release),      AV_OPT_TYPE_DOUBLE, {.dbl=50},     1, 8000, A|F },
     { "asc",       "enable asc",       OFFSET(auto_release), AV_OPT_TYPE_BOOL,   {.i64=0},      0,    1, A|F },
     { "asc_level", "set asc level",    OFFSET(asc_coeff),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5},    0,    1, A|F },
     { "level",     "auto level",       OFFSET(auto_level),   AV_OPT_TYPE_BOOL,   {.i64=1},      0,    1, A|F },

     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(alimiter);
 
 static av_cold int init(AVFilterContext *ctx)
 {
     AudioLimiterContext *s = ctx->priv;
 
     s->attack   /= 1000.;
     s->release  /= 1000.;
     s->att       = 1.;
     s->asc_pos   = -1;
     s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
 
     return 0;
 }
 
 static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
                          double peak, double limit, double patt, int asc)
 {
     double rdelta = (1.0 - patt) / (sample_rate * release);
 
     if (asc && s->auto_release && s->asc_c > 0) {
         double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
 
         if (a_att > patt) {
             double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
 
             if (delta < rdelta)
                 rdelta = delta;
         }
     }
 
     return rdelta;
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioLimiterContext *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     const double *src = (const double *)in->data[0];
     const int channels = inlink->channels;
     const int buffer_size = s->buffer_size;
     double *dst, *buffer = s->buffer;
     const double release = s->release;
     const double limit = s->limit;
     double *nextdelta = s->nextdelta;

     double level = s->auto_level ? 1 / limit : 1;
     const double level_out = s->level_out;
     const double level_in = s->level_in;

     int *nextpos = s->nextpos;
     AVFrame *out;
     double *buf;
     int n, c, i;
 
     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];
 
     for (n = 0; n < in->nb_samples; n++) {
         double peak = 0;
 
         for (c = 0; c < channels; c++) {

             double sample = src[c] * level_in;

 
             buffer[s->pos + c] = sample;
             peak = FFMAX(peak, fabs(sample));
         }
 
         if (s->auto_release && peak > limit) {
             s->asc += peak;
             s->asc_c++;
         }
 
         if (peak > limit) {
             double patt = FFMIN(limit / peak, 1.);
             double rdelta = get_rdelta(s, release, inlink->sample_rate,
                                        peak, limit, patt, 0);
             double delta = (limit / peak - s->att) / buffer_size * channels;
             int found = 0;
 
             if (delta < s->delta) {
                 s->delta = delta;
                 nextpos[0] = s->pos;
                 nextpos[1] = -1;
                 nextdelta[0] = rdelta;
                 s->nextlen = 1;
                 s->nextiter= 0;
             } else {
                 for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
                     int j = i % buffer_size;
                     double ppeak, pdelta;
 
                     ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
                             fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
                     pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
                     if (pdelta < nextdelta[j]) {
                         nextdelta[j] = pdelta;
                         found = 1;
                         break;
                     }
                 }
                 if (found) {
                     s->nextlen = i - s->nextiter + 1;
                     nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
                     nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
                     nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
                     s->nextlen++;
                 }
             }
         }
 
         buf = &s->buffer[(s->pos + channels) % buffer_size];
         peak = 0;
         for (c = 0; c < channels; c++) {
             double sample = buf[c];
 
             peak = FFMAX(peak, fabs(sample));
         }
 
         if (s->pos == s->asc_pos && !s->asc_changed)
             s->asc_pos = -1;
 
         if (s->auto_release && s->asc_pos == -1 && peak > limit) {
             s->asc -= peak;
             s->asc_c--;
         }
 
         s->att += s->delta;
 
         for (c = 0; c < channels; c++)
             dst[c] = buf[c] * s->att;
 
         if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
             if (s->auto_release) {
                 s->delta = get_rdelta(s, release, inlink->sample_rate,
                                       peak, limit, s->att, 1);
                 if (s->nextlen > 1) {
                     int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
                     double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
                                                             fabs(buffer[pnextpos]) :
                                                             fabs(buffer[pnextpos + 1]);
                     double pdelta = (limit / ppeak - s->att) /
                                     (((buffer_size + pnextpos -
                                     ((s->pos + channels) % buffer_size)) %
                                     buffer_size) / channels);
                     if (pdelta < s->delta)
                         s->delta = pdelta;
                 }
             } else {
                 s->delta = nextdelta[s->nextiter];
                 s->att = limit / peak;
             }
 
             s->nextlen -= 1;
             nextpos[s->nextiter] = -1;
             s->nextiter = (s->nextiter + 1) % buffer_size;
         }
 
         if (s->att > 1.) {
             s->att = 1.;
             s->delta = 0.;
             s->nextiter = 0;
             s->nextlen = 0;
             nextpos[0] = -1;
         }
 
         if (s->att <= 0.) {
             s->att = 0.0000000000001;
             s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
         }
 
         if (s->att != 1. && (1. - s->att) < 0.0000000000001)
             s->att = 1.;
 
         if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
             s->delta = 0.;
 
         for (c = 0; c < channels; c++)

             dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;

 
         s->pos = (s->pos + channels) % buffer_size;
         src += channels;
         dst += channels;
     }
 
     if (in != out)
         av_frame_free(&in);
 
     return ff_filter_frame(outlink, out);
 }
 
 static int 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);
 }
 
 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioLimiterContext *s = ctx->priv;
     int obuffer_size;
 
     obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
     if (obuffer_size < inlink->channels)
         return AVERROR(EINVAL);
 
     s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
     s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
     s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
     if (!s->buffer || !s->nextdelta || !s->nextpos)
         return AVERROR(ENOMEM);
 
     memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
     s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
     s->buffer_size -= s->buffer_size % inlink->channels;
 
     return 0;
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioLimiterContext *s = ctx->priv;
 
     av_freep(&s->buffer);
     av_freep(&s->nextdelta);
     av_freep(&s->nextpos);
 }
 
 static const AVFilterPad alimiter_inputs[] = {
     {
         .name         = "main",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
         .config_props = config_input,
     },
     { NULL }
 };
 
 static const AVFilterPad alimiter_outputs[] = {
     {
         .name = "default",
         .type = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 
 AVFilter ff_af_alimiter = {
     .name           = "alimiter",

     .description    = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),

     .priv_size      = sizeof(AudioLimiterContext),
     .priv_class     = &alimiter_class,
     .init           = init,
     .uninit         = uninit,
     .query_formats  = query_formats,
     .inputs         = alimiter_inputs,
     .outputs        = alimiter_outputs,
 };