/*
  * 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
  * Crossover filter
  *
  * Split an audio stream into several bands.
  */
 
 #include "libavutil/attributes.h"
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"

 #include "libavutil/eval.h"

 #include "libavutil/internal.h"
 #include "libavutil/opt.h"
 
 #include "audio.h"
 #include "avfilter.h"
 #include "formats.h"
 #include "internal.h"
 
 #define MAX_SPLITS 16
 #define MAX_BANDS MAX_SPLITS + 1
 
 typedef struct BiquadContext {
     double a0, a1, a2;
     double b1, b2;
     double i1, i2;
     double o1, o2;
 } BiquadContext;
 
 typedef struct CrossoverChannel {
     BiquadContext lp[MAX_BANDS][4];

     BiquadContext hp[MAX_BANDS][4];

 } CrossoverChannel;
 
 typedef struct AudioCrossoverContext {
     const AVClass *class;
 
     char *splits_str;
     int order;
 
     int filter_count;
     int nb_splits;
     float *splits;
 
     CrossoverChannel *xover;

 
     AVFrame *input_frame;
     AVFrame *frames[MAX_BANDS];

 } AudioCrossoverContext;
 
 #define OFFSET(x) offsetof(AudioCrossoverContext, x)
 #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption acrossover_options[] = {
     { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
     { "order", "set order",             OFFSET(order),      AV_OPT_TYPE_INT,    {.i64=1},     0, 2, AF, "m" },
     { "2nd",   "2nd order",             0,                  AV_OPT_TYPE_CONST,  {.i64=0},     0, 0, AF, "m" },
     { "4th",   "4th order",             0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0, 0, AF, "m" },
     { "8th",   "8th order",             0,                  AV_OPT_TYPE_CONST,  {.i64=2},     0, 0, AF, "m" },
     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(acrossover);
 
 static av_cold int init(AVFilterContext *ctx)
 {
     AudioCrossoverContext *s = ctx->priv;
     char *p, *arg, *saveptr = NULL;
     int i, ret = 0;
 
     s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
     if (!s->splits)
         return AVERROR(ENOMEM);
 
     p = s->splits_str;
     for (i = 0; i < MAX_SPLITS; i++) {
         float freq;
 
         if (!(arg = av_strtok(p, " |", &saveptr)))
             break;
 
         p = NULL;
 

         if (av_sscanf(arg, "%f", &freq) != 1) {
             av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
             return AVERROR(EINVAL);
         }

         if (freq <= 0) {
             av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
             return AVERROR(EINVAL);
         }
 
         if (i > 0 && freq <= s->splits[i-1]) {
             av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
             return AVERROR(EINVAL);
         }
 
         s->splits[i] = freq;
     }
 
     s->nb_splits = i;
 
     for (i = 0; i <= s->nb_splits; i++) {
         AVFilterPad pad  = { 0 };
         char *name;
 
         pad.type = AVMEDIA_TYPE_AUDIO;
         name = av_asprintf("out%d", ctx->nb_outputs);
         if (!name)
             return AVERROR(ENOMEM);
         pad.name = name;
 
         if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
             av_freep(&pad.name);
             return ret;
         }
     }
 
     return ret;
 }
 

 static void set_lp(BiquadContext *b, double fc, double q, double sr)

 {

     double omega = 2.0 * M_PI * fc / sr;

     double sn = sin(omega);
     double cs = cos(omega);

     double alpha = sn / (2. * q);

     double inv = 1.0 / (1.0 + alpha);
 

     b->a0 = (1. - cs) * 0.5 * inv;
     b->a1 = (1. - cs) * inv;

     b->a2 = b->a0;
     b->b1 = -2. * cs * inv;
     b->b2 = (1. - alpha) * inv;
 }
 

 static void set_hp(BiquadContext *b, double fc, double q, double sr)
 {
     double omega = 2 * M_PI * fc / sr;
     double sn = sin(omega);
     double cs = cos(omega);
     double alpha = sn / (2 * q);
     double inv = 1.0 / (1.0 + alpha);
 
     b->a0 = inv * (1. + cs) / 2.;
     b->a1 = -2. * b->a0;
     b->a2 = b->a0;
     b->b1 = -2. * cs * inv;
     b->b2 = (1. - alpha) * inv;
 }
 

 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioCrossoverContext *s = ctx->priv;
     int ch, band, sample_rate = inlink->sample_rate;
     double q;
 
     s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
     if (!s->xover)
         return AVERROR(ENOMEM);
 
     switch (s->order) {
     case 0:
         q = 0.5;
         s->filter_count = 1;
         break;
     case 1:
         q = M_SQRT1_2;
         s->filter_count = 2;
         break;
     case 2:
         q = 0.54;
         s->filter_count = 4;
         break;
     }
 
     for (ch = 0; ch < inlink->channels; ch++) {
         for (band = 0; band <= s->nb_splits; band++) {
             set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);

             set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);

 
             if (s->order > 1) {
                 set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);

                 set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);

                 set_lp(&s->xover[ch].lp[band][2], s->splits[band],    q, sample_rate);

                 set_hp(&s->xover[ch].hp[band][2], s->splits[band],    q, sample_rate);

                 set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);

                 set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);

             } else {
                 set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);

                 set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);

             }
         }
     }
 
     return 0;
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats;
     AVFilterChannelLayouts *layouts;
     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 double biquad_process(BiquadContext *b, double in)
 {
     double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;
 
     b->i2 = b->i1;
     b->o2 = b->o1;
     b->i1 = in;
     b->o1 = out;
 
     return out;
 }
 

 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)

 {
     AudioCrossoverContext *s = ctx->priv;

     AVFrame *in = s->input_frame;
     AVFrame **frames = s->frames;
     const int start = (in->channels * jobnr) / nb_jobs;
     const int end = (in->channels * (jobnr+1)) / nb_jobs;
     int f, band;

     for (int ch = start; ch < end; ch++) {

         const double *src = (const double *)in->extended_data[ch];
         CrossoverChannel *xover = &s->xover[ch];
 

         for (int i = 0; i < in->nb_samples; i++) {

             double sample = src[i], lo, hi;

             for (band = 0; band < ctx->nb_outputs; band++) {
                 double *dst = (double *)frames[band]->extended_data[ch];

                 lo = sample;

                 hi = sample;

                 for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
                     BiquadContext *lp = &xover->lp[band][f];
                     lo = biquad_process(lp, lo);

                 }

                 for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
                     BiquadContext *hp = &xover->hp[band][f];
                     hi = biquad_process(hp, hi);
                 }

 
                 dst[i] = lo;
 
                 sample = hi;

             }
         }
     }
 

     return 0;
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioCrossoverContext *s = ctx->priv;
     AVFrame **frames = s->frames;
     int i, ret = 0;
 
     for (i = 0; i < ctx->nb_outputs; i++) {
         frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
 
         if (!frames[i]) {
             ret = AVERROR(ENOMEM);
             break;
         }
 
         frames[i]->pts = in->pts;
     }
 
     if (ret < 0)
         goto fail;
 
     s->input_frame = in;
     ctx->internal->execute(ctx, filter_channels, NULL, NULL, FFMIN(inlink->channels,
                                                                    ff_filter_get_nb_threads(ctx)));
 

     for (i = 0; i < ctx->nb_outputs; i++) {
         ret = ff_filter_frame(ctx->outputs[i], frames[i]);

         frames[i] = NULL;

         if (ret < 0)
             break;
     }
 
 fail:

     for (i = 0; i < ctx->nb_outputs; i++)
         av_frame_free(&frames[i]);

     av_frame_free(&in);

     s->input_frame = NULL;

 
     return ret;
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioCrossoverContext *s = ctx->priv;
     int i;
 
     av_freep(&s->splits);

     av_freep(&s->xover);

 
     for (i = 0; i < ctx->nb_outputs; i++)
         av_freep(&ctx->output_pads[i].name);
 }
 
 static const AVFilterPad inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
         .config_props = config_input,
     },
     { NULL }
 };
 
 AVFilter ff_af_acrossover = {
     .name           = "acrossover",
     .description    = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
     .priv_size      = sizeof(AudioCrossoverContext),
     .priv_class     = &acrossover_class,
     .init           = init,
     .uninit         = uninit,
     .query_formats  = query_formats,
     .inputs         = inputs,
     .outputs        = NULL,

     .flags          = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
                       AVFILTER_FLAG_SLICE_THREADS,

 };