/*
  * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
  *
  * 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/avstring.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "internal.h"
 #include "generate_wave_table.h"
 
 #define INTERPOLATION_LINEAR    0
 #define INTERPOLATION_QUADRATIC 1
 
 typedef struct FlangerContext {
     const AVClass *class;
     double delay_min;
     double delay_depth;
     double feedback_gain;
     double delay_gain;
     double speed;
     int wave_shape;
     double channel_phase;
     int interpolation;
     double in_gain;
     int max_samples;
     uint8_t **delay_buffer;
     int delay_buf_pos;
     double *delay_last;
     float *lfo;
     int lfo_length;
     int lfo_pos;
 } FlangerContext;
 
 #define OFFSET(x) offsetof(FlangerContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption flanger_options[] = {
     { "delay", "base delay in milliseconds",        OFFSET(delay_min),   AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
     { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
     { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
     { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
     { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
     { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
     { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, A, "type" },
     { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, A, "type" },
     { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, A, "type" },
     { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, A, "type" },
     { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
     { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
     { "linear",     NULL, 0, AV_OPT_TYPE_CONST,  {.i64=INTERPOLATION_LINEAR},    0, 0, A, "itype" },
     { "quadratic",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(flanger);
 
 static int init(AVFilterContext *ctx)
 {
     FlangerContext *s = ctx->priv;
 
     s->feedback_gain /= 100;
     s->delay_gain    /= 100;
     s->channel_phase /= 100;
     s->delay_min     /= 1000;
     s->delay_depth   /= 1000;
     s->in_gain        = 1 / (1 + s->delay_gain);
     s->delay_gain    /= 1 + s->delay_gain;
     s->delay_gain    *= 1 - fabs(s->feedback_gain);
 
     return 0;
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterChannelLayouts *layouts;
     AVFilterFormats *formats;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE
     };
 
     layouts = ff_all_channel_layouts();
     if (!layouts)
         return AVERROR(ENOMEM);
     ff_set_common_channel_layouts(ctx, layouts);
 
     formats = ff_make_format_list(sample_fmts);
     if (!formats)
         return AVERROR(ENOMEM);
     ff_set_common_formats(ctx, formats);
 
     formats = ff_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);
     ff_set_common_samplerates(ctx, formats);
 
     return 0;
 }
 
 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     FlangerContext *s = ctx->priv;
 
     s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
     s->lfo_length  = inlink->sample_rate / s->speed;
     s->delay_last  = av_calloc(inlink->channels, sizeof(*s->delay_last));
     s->lfo         = av_calloc(s->lfo_length, sizeof(*s->lfo));
     if (!s->lfo || !s->delay_last)
         return AVERROR(ENOMEM);
 
     ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
                            floor(s->delay_min * inlink->sample_rate + 0.5),
                            s->max_samples - 2., 3 * M_PI_2);
 
     return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
                                               inlink->channels, s->max_samples,
                                               inlink->format, 0);
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
 {
     AVFilterContext *ctx = inlink->dst;
     FlangerContext *s = ctx->priv;
     AVFrame *out_frame;
     int chan, i;
 
     if (av_frame_is_writable(frame)) {
         out_frame = frame;
     } else {
         out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
         if (!out_frame)
             return AVERROR(ENOMEM);
         av_frame_copy_props(out_frame, frame);
     }
 
     for (i = 0; i < frame->nb_samples; i++) {
 
         s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
 
         for (chan = 0; chan < inlink->channels; chan++) {
             double *src = (double *)frame->extended_data[chan];
             double *dst = (double *)out_frame->extended_data[chan];
             double delayed_0, delayed_1;
             double delayed;
             double in, out;
             int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
             double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
             int int_delay = (int)delay;
             double frac_delay = modf(delay, &delay);
             double *delay_buffer = (double *)s->delay_buffer[chan];
 
             in = src[i];
             delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
                                                            s->feedback_gain;
             delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
             delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
 
             if (s->interpolation == INTERPOLATION_LINEAR) {
                 delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
             } else {
                 double a, b;
                 double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
                 delayed_2 -= delayed_0;
                 delayed_1 -= delayed_0;
                 a = delayed_2 * .5 - delayed_1;
                 b = delayed_1 *  2 - delayed_2 *.5;
                 delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
             }
 
             s->delay_last[chan] = delayed;
             out = in * s->in_gain + delayed * s->delay_gain;
             dst[i] = out;
         }
         s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
     }
 
     if (frame != out_frame)
         av_frame_free(&frame);
 
     return ff_filter_frame(ctx->outputs[0], out_frame);
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     FlangerContext *s = ctx->priv;
 
     av_freep(&s->lfo);
     av_freep(&s->delay_last);
 
     if (s->delay_buffer)
         av_freep(&s->delay_buffer[0]);
     av_freep(&s->delay_buffer);
 }
 
 static const AVFilterPad flanger_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
         .filter_frame = filter_frame,
     },
     { NULL }
 };
 
 static const AVFilterPad flanger_outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 
 AVFilter ff_af_flanger = {
     .name          = "flanger",
     .description   = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
     .query_formats = query_formats,
     .priv_size     = sizeof(FlangerContext),
     .priv_class    = &flanger_class,
     .init          = init,
     .uninit        = uninit,
     .inputs        = flanger_inputs,
     .outputs       = flanger_outputs,
 };