/*
  * 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
  */
 

 #include "libavutil/avassert.h"

 #include "libavutil/avstring.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "internal.h"
 
 typedef struct AudioEchoContext {
     const AVClass *class;
     float in_gain, out_gain;
     char *delays, *decays;
     float *delay, *decay;
     int nb_echoes;
     int delay_index;
     uint8_t **delayptrs;
     int max_samples, fade_out;
     int *samples;
     int64_t next_pts;
 
     void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
                          uint8_t * const *src, uint8_t **dst,
                          int nb_samples, int channels);
 } AudioEchoContext;
 
 #define OFFSET(x) offsetof(AudioEchoContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption aecho_options[] = {
     { "in_gain",  "set signal input gain",  OFFSET(in_gain),  AV_OPT_TYPE_FLOAT,  {.dbl=0.6}, 0, 1, A },
     { "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT,  {.dbl=0.3}, 0, 1, A },
     { "delays",   "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
     { "decays",   "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },

     { NULL }

 };
 
 AVFILTER_DEFINE_CLASS(aecho);
 
 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 void uninit(AVFilterContext *ctx)
 {
     AudioEchoContext *s = ctx->priv;
 
     av_freep(&s->delay);
     av_freep(&s->decay);
     av_freep(&s->samples);
 
     if (s->delayptrs)

         av_freep(&s->delayptrs[0]);

     av_freep(&s->delayptrs);
 }
 
 static av_cold int init(AVFilterContext *ctx)
 {
     AudioEchoContext *s = ctx->priv;
     int nb_delays, nb_decays, i;
 
     if (!s->delays || !s->decays) {
         av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
         return AVERROR(EINVAL);
     }
 
     count_items(s->delays, &nb_delays);
     count_items(s->decays, &nb_decays);
 
     s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
     s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
     if (!s->delay || !s->decay)
         return AVERROR(ENOMEM);
 
     fill_items(s->delays, &nb_delays, s->delay);
     fill_items(s->decays, &nb_decays, s->decay);
 
     if (nb_delays != nb_decays) {
         av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
         return AVERROR(EINVAL);
     }
 
     s->nb_echoes = nb_delays;
     if (!s->nb_echoes) {
         av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
         return AVERROR(EINVAL);
     }
 
     s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
     if (!s->samples)
         return AVERROR(ENOMEM);
 
     for (i = 0; i < nb_delays; i++) {
         if (s->delay[i] <= 0 || s->delay[i] > 90000) {
             av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
             return AVERROR(EINVAL);
         }
         if (s->decay[i] <= 0 || s->decay[i] > 1) {
             av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
             return AVERROR(EINVAL);
         }
     }
 
     s->next_pts = AV_NOPTS_VALUE;
 
     av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
     return 0;
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterChannelLayouts *layouts;
     AVFilterFormats *formats;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
         AV_SAMPLE_FMT_FLTP, 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);

 }
 
 #define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
 
 #define ECHO(name, type, min, max)                                          \
 static void echo_samples_## name ##p(AudioEchoContext *ctx,                 \
                                      uint8_t **delayptrs,                   \
                                      uint8_t * const *src, uint8_t **dst,   \
                                      int nb_samples, int channels)          \
 {                                                                           \
     const double out_gain = ctx->out_gain;                                  \
     const double in_gain = ctx->in_gain;                                    \
     const int nb_echoes = ctx->nb_echoes;                                   \
     const int max_samples = ctx->max_samples;                               \

     int i, j, chan, av_uninit(index);                                       \
                                                                             \
     av_assert1(channels > 0); /* would corrupt delay_index */               \

                                                                             \
     for (chan = 0; chan < channels; chan++) {                               \
         const type *s = (type *)src[chan];                                  \
         type *d = (type *)dst[chan];                                        \
         type *dbuf = (type *)delayptrs[chan];                               \
                                                                             \
         index = ctx->delay_index;                                           \
         for (i = 0; i < nb_samples; i++, s++, d++) {                        \
             double out, in;                                                 \
                                                                             \
             in = *s;                                                        \
             out = in * in_gain;                                             \
             for (j = 0; j < nb_echoes; j++) {                               \
                 int ix = index + max_samples - ctx->samples[j];             \
                 ix = MOD(ix, max_samples);                                  \
                 out += dbuf[ix] * ctx->decay[j];                            \
             }                                                               \
             out *= out_gain;                                                \
                                                                             \
             *d = av_clipd(out, min, max);                                   \
             dbuf[index] = in;                                               \
                                                                             \
             index = MOD(index + 1, max_samples);                            \
         }                                                                   \
     }                                                                       \
     ctx->delay_index = index;                                               \
 }
 
 ECHO(dbl, double,  -1.0,      1.0      )
 ECHO(flt, float,   -1.0,      1.0      )
 ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
 ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
 
 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AudioEchoContext *s = ctx->priv;
     float volume = 1.0;
     int i;
 
     for (i = 0; i < s->nb_echoes; i++) {
         s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
         s->max_samples = FFMAX(s->max_samples, s->samples[i]);
         volume += s->decay[i];
     }
 
     if (s->max_samples <= 0) {
         av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
         return AVERROR(EINVAL);
     }
     s->fade_out = s->max_samples;
 
     if (volume * s->in_gain * s->out_gain > 1.0)
         av_log(ctx, AV_LOG_WARNING,
                "out_gain %f can cause saturation of output\n", s->out_gain);
 
     switch (outlink->format) {
     case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
     case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
     case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
     case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
     }
 
 
     if (s->delayptrs)

         av_freep(&s->delayptrs[0]);

     av_freep(&s->delayptrs);
 
     return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
                                               outlink->channels,
                                               s->max_samples,
                                               outlink->format, 0);
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioEchoContext *s = ctx->priv;
     AVFrame *out_frame;
 
     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);
     }
 

     s->echo_samples(s, s->delayptrs, frame->extended_data, out_frame->extended_data,

                     frame->nb_samples, inlink->channels);
 

     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;
     AudioEchoContext *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);
 

         s->echo_samples(s, s->delayptrs, frame->extended_data, frame->extended_data,

                         frame->nb_samples, outlink->channels);
 
         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);
 
         return ff_filter_frame(outlink, frame);
     }
 
     return ret;
 }
 
 static const AVFilterPad aecho_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
     },

     { NULL }

 };
 
 static const AVFilterPad aecho_outputs[] = {
     {
         .name          = "default",
         .request_frame = request_frame,
         .config_props  = config_output,
         .type          = AVMEDIA_TYPE_AUDIO,
     },

     { NULL }

 };
 

 AVFilter ff_af_aecho = {

     .name          = "aecho",
     .description   = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
     .query_formats = query_formats,
     .priv_size     = sizeof(AudioEchoContext),
     .priv_class    = &aecho_class,
     .init          = init,
     .uninit        = uninit,
     .inputs        = aecho_inputs,
     .outputs       = aecho_outputs,
 };