/*
  * Copyright (c) Markus Schmidt and Christian Holschuh
  *
  * 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/opt.h"
 #include "avfilter.h"
 #include "internal.h"
 #include "audio.h"
 
 typedef struct LFOContext {
     double freq;
     double offset;
     int srate;
     double amount;
     double pwidth;
     double phase;
 } LFOContext;
 
 typedef struct SRContext {
     double target;
     double real;
     double samples;
     double last;
 } SRContext;
 
 typedef struct ACrusherContext {
     const AVClass *class;
 
     double level_in;
     double level_out;
     double bits;
     double mix;
     int mode;
     double dc;
     double idc;
     double aa;
     double samples;
     int is_lfo;
     double lforange;
     double lforate;
 
     double sqr;
     double aa1;
     double coeff;
     int    round;
     double sov;
     double smin;
     double sdiff;
 
     LFOContext lfo;
     SRContext *sr;
 } ACrusherContext;
 
 #define OFFSET(x) offsetof(ACrusherContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption acrusher_options[] = {
     { "level_in", "set level in",         OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.015625, 64, A },
     { "level_out","set level out",        OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},    0.015625, 64, A },
     { "bits",     "set bit reduction",    OFFSET(bits),      AV_OPT_TYPE_DOUBLE, {.dbl=8},    1,        64, A },
     { "mix",      "set mix",              OFFSET(mix),       AV_OPT_TYPE_DOUBLE, {.dbl=.5},   0,         1, A },
     { "mode",     "set mode",             OFFSET(mode),      AV_OPT_TYPE_INT,    {.i64=0},    0,         1, A, "mode" },
     {   "lin",    "linear",               0,                 AV_OPT_TYPE_CONST,  {.i64=0},    0,         0, A, "mode" },
     {   "log",    "logarithmic",          0,                 AV_OPT_TYPE_CONST,  {.i64=1},    0,         0, A, "mode" },
     { "dc",       "set DC",               OFFSET(dc),        AV_OPT_TYPE_DOUBLE, {.dbl=1},  .25,         4, A },
     { "aa",       "set anti-aliasing",    OFFSET(aa),        AV_OPT_TYPE_DOUBLE, {.dbl=.5},   0,         1, A },
     { "samples",  "set sample reduction", OFFSET(samples),   AV_OPT_TYPE_DOUBLE, {.dbl=1},    1,       250, A },
     { "lfo",      "enable LFO",           OFFSET(is_lfo),    AV_OPT_TYPE_BOOL,   {.i64=0},    0,         1, A },
     { "lforange", "set LFO depth",        OFFSET(lforange),  AV_OPT_TYPE_DOUBLE, {.dbl=20},   1,       250, A },
     { "lforate",  "set LFO rate",         OFFSET(lforate),   AV_OPT_TYPE_DOUBLE, {.dbl=.3}, .01,       200, A },
     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(acrusher);
 
 static double samplereduction(ACrusherContext *s, SRContext *sr, double in)
 {
     sr->samples++;
     if (sr->samples >= s->round) {
         sr->target += s->samples;
         sr->real += s->round;
         if (sr->target + s->samples >= sr->real + 1) {
             sr->last = in;
             sr->target = 0;
             sr->real   = 0;
         }
         sr->samples = 0;
     }
     return sr->last;
 }
 
 static double add_dc(double s, double dc, double idc)
 {
     return s > 0 ? s * dc : s * idc;
 }
 
 static double remove_dc(double s, double dc, double idc)
 {
     return s > 0 ? s * idc : s * dc;
 }
 
 static inline double factor(double y, double k, double aa1, double aa)
 {
     return 0.5 * (sin(M_PI * (fabs(y - k) - aa1) / aa - M_PI_2) + 1);
 }
 
 static double bitreduction(ACrusherContext *s, double in)
 {
     const double sqr = s->sqr;
     const double coeff = s->coeff;
     const double aa = s->aa;
     const double aa1 = s->aa1;
     double y, k;
 
     // add dc
     in = add_dc(in, s->dc, s->idc);
 
     // main rounding calculation depending on mode
 
     // the idea for anti-aliasing:
     // you need a function f which brings you to the scale, where
     // you want to round and the function f_b (with f(f_b)=id) which
     // brings you back to your original scale.
     //
     // then you can use the logic below in the following way:
     // y = f(in) and k = roundf(y)
     // if (y > k + aa1)
     //      k = f_b(k) + ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
     // if (y < k + aa1)
     //      k = f_b(k) - ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
     //
     // whereas x = (fabs(f(in) - k) - aa1) * PI / aa
     // for both cases.
 
     switch (s->mode) {
     case 0:
     default:
         // linear
         y = in * coeff;
         k = roundf(y);
         if (k - aa1 <= y && y <= k + aa1) {
             k /= coeff;
         } else if (y > k + aa1) {
             k = k / coeff + ((k + 1) / coeff - k / coeff) *
                 factor(y, k, aa1, aa);
         } else {
             k = k / coeff - (k / coeff - (k - 1) / coeff) *
                 factor(y, k, aa1, aa);
         }
         break;
     case 1:
         // logarithmic
         y = sqr * log(fabs(in)) + sqr * sqr;
         k = roundf(y);
         if(!in) {
             k = 0;
         } else if (k - aa1 <= y && y <= k + aa1) {
             k = in / fabs(in) * exp(k / sqr - sqr);
         } else if (y > k + aa1) {
             double x = exp(k / sqr - sqr);
             k = FFSIGN(in) * (x + (exp((k + 1) / sqr - sqr) - x) *
                 factor(y, k, aa1, aa));
         } else {
             double x = exp(k / sqr - sqr);
             k = in / fabs(in) * (x - (x - exp((k - 1) / sqr - sqr)) *
                 factor(y, k, aa1, aa));
         }
         break;
     }
 
     // mix between dry and wet signal
     k += (in - k) * s->mix;
 
     // remove dc
     k = remove_dc(k, s->dc, s->idc);
 
     return k;
 }
 
 static double lfo_get(LFOContext *lfo)
 {
     double phs = FFMIN(100., lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
     double val;
 
     if (phs > 1)
         phs = fmod(phs, 1.);
 
     val = sin((phs * 360.) * M_PI / 180);
 
     return val * lfo->amount;
 }
 
 static void lfo_advance(LFOContext *lfo, unsigned count)
 {
     lfo->phase = fabs(lfo->phase + count * lfo->freq * (1. / lfo->srate));
     if (lfo->phase >= 1.)
         lfo->phase = fmod(lfo->phase, 1.);
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     ACrusherContext *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     AVFrame *out;
     const double *src = (const double *)in->data[0];
     double *dst;
     const double level_in = s->level_in;
     const double level_out = s->level_out;
     const double mix = s->mix;
     int n, c;
 
     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++) {
         if (s->is_lfo) {
             s->samples = s->smin + s->sdiff * (lfo_get(&s->lfo) + 0.5);
             s->round = round(s->samples);
         }
 
         for (c = 0; c < inlink->channels; c++) {
             double sample = src[c] * level_in;
 
             sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in;
             dst[c] = bitreduction(s, sample) * level_out;
         }
         src += c;
         dst += c;
 
         if (s->is_lfo)
             lfo_advance(&s->lfo, 1);
     }
 
     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 av_cold void uninit(AVFilterContext *ctx)
 {
     ACrusherContext *s = ctx->priv;
 
     av_freep(&s->sr);
 }
 
 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     ACrusherContext *s = ctx->priv;

     double rad, sunder, smax, sover;

 
     s->idc = 1. / s->dc;
     s->coeff = exp2(s->bits) - 1;
     s->sqr = sqrt(s->coeff / 2);
     s->aa1 = (1. - s->aa) / 2.;
     s->round = round(s->samples);
     rad = s->lforange / 2.;
     s->smin = FFMAX(s->samples - rad, 1.);

     sunder   = s->samples - rad - s->smin;

     smax = FFMIN(s->samples + rad, 250.);

     sover    = s->samples + rad - smax;
     smax    -= sunder;
     s->smin -= sover;

     s->sdiff = smax - s->smin;
 
     s->lfo.freq = s->lforate;
     s->lfo.pwidth = 1.;
     s->lfo.srate = inlink->sample_rate;
     s->lfo.amount = .5;
 
     s->sr = av_calloc(inlink->channels, sizeof(*s->sr));
     if (!s->sr)
         return AVERROR(ENOMEM);
 
     return 0;
 }
 
 static const AVFilterPad avfilter_af_acrusher_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
         .filter_frame = filter_frame,
     },
     { NULL }
 };
 
 static const AVFilterPad avfilter_af_acrusher_outputs[] = {
     {
         .name = "default",
         .type = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 
 AVFilter ff_af_acrusher = {
     .name          = "acrusher",
     .description   = NULL_IF_CONFIG_SMALL("Reduce audio bit resolution."),
     .priv_size     = sizeof(ACrusherContext),
     .priv_class    = &acrusher_class,
     .uninit        = uninit,
     .query_formats = query_formats,
     .inputs        = avfilter_af_acrusher_inputs,
     .outputs       = avfilter_af_acrusher_outputs,
 };