/*
  * Copyright (c) 2013 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
  * phaser audio filter
  */
 
 #include "libavutil/avassert.h"
 #include "libavutil/opt.h"
 #include "audio.h"
 #include "avfilter.h"
 #include "internal.h"

 #include "generate_wave_table.h"

 
 typedef struct AudioPhaserContext {
     const AVClass *class;
     double in_gain, out_gain;
     double delay;
     double decay;
     double speed;
 
     enum WaveType type;
 
     int delay_buffer_length;
     double *delay_buffer;
 
     int modulation_buffer_length;
     int32_t *modulation_buffer;
 
     int delay_pos, modulation_pos;
 
     void (*phaser)(struct AudioPhaserContext *p,
                    uint8_t * const *src, uint8_t **dst,
                    int nb_samples, int channels);
 } AudioPhaserContext;
 
 #define OFFSET(x) offsetof(AudioPhaserContext, x)
 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption aphaser_options[] = {
     { "in_gain",  "set input gain",            OFFSET(in_gain),  AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0,  1,   FLAGS },
     { "out_gain", "set output gain",           OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0,  1e9, FLAGS },
     { "delay",    "set delay in milliseconds", OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=3.},  0,  5,   FLAGS },
     { "decay",    "set decay",                 OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0, .99,  FLAGS },
     { "speed",    "set modulation speed",      OFFSET(speed),    AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1,  2,   FLAGS },
     { "type",     "set modulation type",       OFFSET(type),     AV_OPT_TYPE_INT,    {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
     { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
     { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
     { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
     { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },

     { NULL }

 };
 
 AVFILTER_DEFINE_CLASS(aphaser);
 

 static av_cold int init(AVFilterContext *ctx)

 {
     AudioPhaserContext *p = ctx->priv;
 
     if (p->in_gain > (1 - p->decay * p->decay))
         av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
     if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
         av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
 
     return 0;
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats;
     AVFilterChannelLayouts *layouts;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
         AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
         AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
         AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
         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;
 }
 
 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
 
 #define PHASER_PLANAR(name, type)                                      \
 static void phaser_## name ##p(AudioPhaserContext *p,                  \
                                uint8_t * const *src, uint8_t **dst,    \
                                int nb_samples, int channels)           \
 {                                                                      \
     int i, c, delay_pos, modulation_pos;                               \
                                                                        \

     av_assert0(channels > 0);                                          \

     for (c = 0; c < channels; c++) {                                   \
         type *s = (type *)src[c];                                      \
         type *d = (type *)dst[c];                                      \
         double *buffer = p->delay_buffer +                             \
                          c * p->delay_buffer_length;                   \
                                                                        \
         delay_pos      = p->delay_pos;                                 \
         modulation_pos = p->modulation_pos;                            \
                                                                        \
         for (i = 0; i < nb_samples; i++, s++, d++) {                   \
             double v = *s * p->in_gain + buffer[                       \
                        MOD(delay_pos + p->modulation_buffer[           \
                        modulation_pos],                                \
                        p->delay_buffer_length)] * p->decay;            \
                                                                        \
             modulation_pos = MOD(modulation_pos + 1,                   \
                              p->modulation_buffer_length);             \
             delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);    \
             buffer[delay_pos] = v;                                     \
                                                                        \
             *d = v * p->out_gain;                                      \
         }                                                              \
     }                                                                  \
                                                                        \
     p->delay_pos      = delay_pos;                                     \
     p->modulation_pos = modulation_pos;                                \
 }
 
 #define PHASER(name, type)                                              \
 static void phaser_## name (AudioPhaserContext *p,                      \
                             uint8_t * const *src, uint8_t **dst,        \
                             int nb_samples, int channels)               \
 {                                                                       \
     int i, c, delay_pos, modulation_pos;                                \
     type *s = (type *)src[0];                                           \
     type *d = (type *)dst[0];                                           \
     double *buffer = p->delay_buffer;                                   \
                                                                         \
     delay_pos      = p->delay_pos;                                      \
     modulation_pos = p->modulation_pos;                                 \
                                                                         \
     for (i = 0; i < nb_samples; i++) {                                  \
         int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
                    p->delay_buffer_length) * channels;                  \
         int npos;                                                       \
                                                                         \
         delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);         \
         npos = delay_pos * channels;                                    \
         for (c = 0; c < channels; c++, s++, d++) {                      \
             double v = *s * p->in_gain + buffer[pos + c] * p->decay;    \
                                                                         \
             buffer[npos + c] = v;                                       \
                                                                         \
             *d = v * p->out_gain;                                       \
         }                                                               \
                                                                         \
         modulation_pos = MOD(modulation_pos + 1,                        \
                          p->modulation_buffer_length);                  \
     }                                                                   \
                                                                         \
     p->delay_pos      = delay_pos;                                      \
     p->modulation_pos = modulation_pos;                                 \
 }
 
 PHASER_PLANAR(dbl, double)
 PHASER_PLANAR(flt, float)
 PHASER_PLANAR(s16, int16_t)
 PHASER_PLANAR(s32, int32_t)
 
 PHASER(dbl, double)
 PHASER(flt, float)
 PHASER(s16, int16_t)
 PHASER(s32, int32_t)
 
 static int config_output(AVFilterLink *outlink)
 {
     AudioPhaserContext *p = outlink->src->priv;
     AVFilterLink *inlink = outlink->src->inputs[0];
 
     p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
     p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
     p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;

     p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer));

 
     if (!p->modulation_buffer || !p->delay_buffer)
         return AVERROR(ENOMEM);
 

     ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
                            p->modulation_buffer, p->modulation_buffer_length,
                            1., p->delay_buffer_length, M_PI / 2.0);

 
     p->delay_pos = p->modulation_pos = 0;
 
     switch (inlink->format) {
     case AV_SAMPLE_FMT_DBL:  p->phaser = phaser_dbl;  break;
     case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
     case AV_SAMPLE_FMT_FLT:  p->phaser = phaser_flt;  break;
     case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
     case AV_SAMPLE_FMT_S16:  p->phaser = phaser_s16;  break;
     case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
     case AV_SAMPLE_FMT_S32:  p->phaser = phaser_s32;  break;
     case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
     default: av_assert0(0);
     }
 
     return 0;
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
 {
     AudioPhaserContext *p = inlink->dst->priv;
     AVFilterLink *outlink = inlink->dst->outputs[0];
     AVFrame *outbuf;
 
     if (av_frame_is_writable(inbuf)) {
         outbuf = inbuf;
     } else {
         outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
         if (!outbuf)
             return AVERROR(ENOMEM);
         av_frame_copy_props(outbuf, inbuf);
     }
 
     p->phaser(p, inbuf->extended_data, outbuf->extended_data,
               outbuf->nb_samples, av_frame_get_channels(outbuf));
 
     if (inbuf != outbuf)
         av_frame_free(&inbuf);
 
     return ff_filter_frame(outlink, outbuf);
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioPhaserContext *p = ctx->priv;
 
     av_freep(&p->delay_buffer);
     av_freep(&p->modulation_buffer);
 }
 
 static const AVFilterPad aphaser_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
     },
     { NULL }
 };
 
 static const AVFilterPad aphaser_outputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_output,
     },
     { NULL }
 };
 

 AVFilter ff_af_aphaser = {

     .name          = "aphaser",
     .description   = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
     .query_formats = query_formats,
     .priv_size     = sizeof(AudioPhaserContext),
     .init          = init,
     .uninit        = uninit,
     .inputs        = aphaser_inputs,
     .outputs       = aphaser_outputs,
     .priv_class    = &aphaser_class,
 };