/*
  * Copyright (c) 1998 Juergen Mueller And Sundry Contributors
  * This source code is freely redistributable and may be used for
  * any purpose.  This copyright notice must be maintained.
  * Juergen Mueller And Sundry Contributors are not responsible for
  * the consequences of using this software.
  *
  * 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
  * chorus audio filter
  */
 
 #include "libavutil/avstring.h"
 #include "libavutil/opt.h"
 #include "audio.h"
 #include "avfilter.h"
 #include "internal.h"
 #include "generate_wave_table.h"
 
 typedef struct ChorusContext {
     const AVClass *class;
     float in_gain, out_gain;
     char *delays_str;
     char *decays_str;
     char *speeds_str;
     char *depths_str;
     float *delays;
     float *decays;
     float *speeds;
     float *depths;
     uint8_t **chorusbuf;
     int **phase;
     int *length;
     int32_t **lookup_table;
     int *counter;
     int num_chorus;
     int max_samples;
     int channels;
     int modulation;
     int fade_out;
     int64_t next_pts;
 } ChorusContext;
 
 #define OFFSET(x) offsetof(ChorusContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption chorus_options[] = {
     { "in_gain",  "set input gain",  OFFSET(in_gain),    AV_OPT_TYPE_FLOAT,  {.dbl=.4}, 0, 1, A },
     { "out_gain", "set output gain", OFFSET(out_gain),   AV_OPT_TYPE_FLOAT,  {.dbl=.4}, 0, 1, A },
     { "delays",   "set delays",      OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
     { "decays",   "set decays",      OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
     { "speeds",   "set speeds",      OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
     { "depths",   "set depths",      OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(chorus);
 
 static void count_items(char *item_str, int *nb_items)
 {
     char *p;
 
     *nb_items = 1;
     for (p = item_str; *p; p++) {
         if (*p == '|')
             (*nb_items)++;
     }
 
 }
 
 static void fill_items(char *item_str, int *nb_items, float *items)
 {
     char *p, *saveptr = NULL;
     int i, new_nb_items = 0;
 
     p = item_str;
     for (i = 0; i < *nb_items; i++) {
         char *tstr = av_strtok(p, "|", &saveptr);
         p = NULL;

         if (tstr)
             new_nb_items += sscanf(tstr, "%f", &items[new_nb_items]) == 1;

     }
 
     *nb_items = new_nb_items;
 }
 
 static av_cold int init(AVFilterContext *ctx)
 {
     ChorusContext *s = ctx->priv;
     int nb_delays, nb_decays, nb_speeds, nb_depths;
 
     if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
         av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
         return AVERROR(EINVAL);
     }
 
     count_items(s->delays_str, &nb_delays);
     count_items(s->decays_str, &nb_decays);
     count_items(s->speeds_str, &nb_speeds);
     count_items(s->depths_str, &nb_depths);
 
     s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
     s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
     s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
     s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
 
     if (!s->delays || !s->decays || !s->speeds || !s->depths)
         return AVERROR(ENOMEM);
 
     fill_items(s->delays_str, &nb_delays, s->delays);
     fill_items(s->decays_str, &nb_decays, s->decays);
     fill_items(s->speeds_str, &nb_speeds, s->speeds);
     fill_items(s->depths_str, &nb_depths, s->depths);
 
     if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
         av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
         return AVERROR(EINVAL);
     }
 
     s->num_chorus = nb_delays;
 
     if (s->num_chorus < 1) {
         av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
         return AVERROR(EINVAL);
     }
 
     s->length = av_calloc(s->num_chorus, sizeof(*s->length));
     s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
 
     if (!s->length || !s->lookup_table)
         return AVERROR(ENOMEM);
 
     s->next_pts = AV_NOPTS_VALUE;
 
     return 0;
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats;
     AVFilterChannelLayouts *layouts;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_FLTP, 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_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     ChorusContext *s = ctx->priv;
     float sum_in_volume = 1.0;
     int n;
 
     s->channels = outlink->channels;
 
     for (n = 0; n < s->num_chorus; n++) {
         int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
         int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
 
         s->length[n] = outlink->sample_rate / s->speeds[n];
 
         s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
         if (!s->lookup_table[n])
             return AVERROR(ENOMEM);
 
         ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
                                s->length[n], 0., depth_samples, 0);
         s->max_samples = FFMAX(s->max_samples, samples);
     }
 
     for (n = 0; n < s->num_chorus; n++)
         sum_in_volume += s->decays[n];
 
     if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
         av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
 
     s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
     if (!s->counter)
         return AVERROR(ENOMEM);
 
     s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
     if (!s->phase)
         return AVERROR(ENOMEM);
 
     for (n = 0; n < outlink->channels; n++) {
         s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
         if (!s->phase[n])
             return AVERROR(ENOMEM);
     }
 
     s->fade_out = s->max_samples;
 
     return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
                                               outlink->channels,
                                               s->max_samples,
                                               outlink->format, 0);
 }
 
 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
 {
     AVFilterContext *ctx = inlink->dst;
     ChorusContext *s = ctx->priv;
     AVFrame *out_frame;
     int c, i, n;
 
     if (av_frame_is_writable(frame)) {
         out_frame = frame;
     } else {
         out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);

         if (!out_frame) {
             av_frame_free(&frame);

             return AVERROR(ENOMEM);

         }

         av_frame_copy_props(out_frame, frame);
     }
 
     for (c = 0; c < inlink->channels; c++) {
         const float *src = (const float *)frame->extended_data[c];
         float *dst = (float *)out_frame->extended_data[c];
         float *chorusbuf = (float *)s->chorusbuf[c];
         int *phase = s->phase[c];
 
         for (i = 0; i < frame->nb_samples; i++) {
             float out, in = src[i];
 
             out = in * s->in_gain;
 
             for (n = 0; n < s->num_chorus; n++) {
                 out += chorusbuf[MOD(s->max_samples + s->counter[c] -
                                      s->lookup_table[n][phase[n]],
                                      s->max_samples)] * s->decays[n];
                 phase[n] = MOD(phase[n] + 1, s->length[n]);
             }
 
             out *= s->out_gain;
 
             dst[i] = out;
 
             chorusbuf[s->counter[c]] = in;
             s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
         }
     }
 
     s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
 
     if (frame != out_frame)
         av_frame_free(&frame);
 
     return ff_filter_frame(ctx->outputs[0], out_frame);
 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     ChorusContext *s = ctx->priv;
     int ret;
 
     ret = ff_request_frame(ctx->inputs[0]);
 
     if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
         int nb_samples = FFMIN(s->fade_out, 2048);
         AVFrame *frame;
 
         frame = ff_get_audio_buffer(outlink, nb_samples);
         if (!frame)
             return AVERROR(ENOMEM);
         s->fade_out -= nb_samples;
 
         av_samples_set_silence(frame->extended_data, 0,
                                frame->nb_samples,
                                outlink->channels,
                                frame->format);
 
         frame->pts = s->next_pts;
         if (s->next_pts != AV_NOPTS_VALUE)
             s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
 
         ret = filter_frame(ctx->inputs[0], frame);
     }
 
     return ret;
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     ChorusContext *s = ctx->priv;
     int n;
 
     av_freep(&s->delays);
     av_freep(&s->decays);
     av_freep(&s->speeds);
     av_freep(&s->depths);
 
     if (s->chorusbuf)
         av_freep(&s->chorusbuf[0]);
     av_freep(&s->chorusbuf);
 
     if (s->phase)
         for (n = 0; n < s->channels; n++)
             av_freep(&s->phase[n]);
     av_freep(&s->phase);
 
     av_freep(&s->counter);
     av_freep(&s->length);
 
     if (s->lookup_table)
         for (n = 0; n < s->num_chorus; n++)
             av_freep(&s->lookup_table[n]);
     av_freep(&s->lookup_table);
 }
 
 static const AVFilterPad chorus_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
     },
     { NULL }
 };
 
 static const AVFilterPad chorus_outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .request_frame = request_frame,
         .config_props  = config_output,
     },
     { NULL }
 };
 
 AVFilter ff_af_chorus = {
     .name          = "chorus",
     .description   = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
     .query_formats = query_formats,
     .priv_size     = sizeof(ChorusContext),
     .priv_class    = &chorus_class,
     .init          = init,
     .uninit        = uninit,
     .inputs        = chorus_inputs,
     .outputs       = chorus_outputs,
 };