/*
  * 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/avstring.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "internal.h"
 
 typedef struct ChanDelay {
     int delay;
     unsigned delay_index;
     unsigned index;
     uint8_t *samples;
 } ChanDelay;
 
 typedef struct AudioDelayContext {
     const AVClass *class;
     char *delays;
     ChanDelay *chandelay;
     int nb_delays;
     int block_align;
     unsigned max_delay;
     int64_t next_pts;
 
     void (*delay_channel)(ChanDelay *d, int nb_samples,
                           const uint8_t *src, uint8_t *dst);
 } AudioDelayContext;
 
 #define OFFSET(x) offsetof(AudioDelayContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption adelay_options[] = {
     { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(adelay);
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterChannelLayouts *layouts;
     AVFilterFormats *formats;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_U8P, 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 DELAY(name, type, fill)                                           \
 static void delay_channel_## name ##p(ChanDelay *d, int nb_samples,       \
                                       const uint8_t *ssrc, uint8_t *ddst) \
 {                                                                         \
     const type *src = (type *)ssrc;                                       \
     type *dst = (type *)ddst;                                             \
     type *samples = (type *)d->samples;                                   \
                                                                           \
     while (nb_samples) {                                                  \
         if (d->delay_index < d->delay) {                                  \
             const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
                                                                           \
             memcpy(&samples[d->delay_index], src, len * sizeof(type));    \
             memset(dst, fill, len * sizeof(type));                        \
             d->delay_index += len;                                        \
             src += len;                                                   \
             dst += len;                                                   \
             nb_samples -= len;                                            \
         } else {                                                          \
             *dst = samples[d->index];                                     \
             samples[d->index] = *src;                                     \
             nb_samples--;                                                 \
             d->index++;                                                   \
             src++, dst++;                                                 \
             d->index = d->index >= d->delay ? 0 : d->index;               \
         }                                                                 \
     }                                                                     \
 }
 
 DELAY(u8,  uint8_t, 0x80)
 DELAY(s16, int16_t, 0)
 DELAY(s32, int32_t, 0)
 DELAY(flt, float,   0)
 DELAY(dbl, double,  0)
 
 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioDelayContext *s = ctx->priv;
     char *p, *arg, *saveptr = NULL;
     int i;
 
     s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
     if (!s->chandelay)
         return AVERROR(ENOMEM);
     s->nb_delays = inlink->channels;
     s->block_align = av_get_bytes_per_sample(inlink->format);
 
     p = s->delays;
     for (i = 0; i < s->nb_delays; i++) {
         ChanDelay *d = &s->chandelay[i];
         float delay;

         char type = 0;
         int ret;

 
         if (!(arg = av_strtok(p, "|", &saveptr)))
             break;
 
         p = NULL;
 

         ret = sscanf(arg, "%d%c", &d->delay, &type);
         if (ret != 2 || type != 'S') {
             sscanf(arg, "%f", &delay);
             d->delay = delay * inlink->sample_rate / 1000.0;
         }
 

         if (d->delay < 0) {
             av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
             return AVERROR(EINVAL);
         }
     }
 
     for (i = 0; i < s->nb_delays; i++) {
         ChanDelay *d = &s->chandelay[i];
 
         if (!d->delay)
             continue;
 
         d->samples = av_malloc_array(d->delay, s->block_align);
         if (!d->samples)
             return AVERROR(ENOMEM);
 
         s->max_delay = FFMAX(s->max_delay, d->delay);
     }
 
     switch (inlink->format) {
     case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
     case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
     case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
     case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
     case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
     }
 
     return 0;
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioDelayContext *s = ctx->priv;
     AVFrame *out_frame;
     int i;
 
     if (ctx->is_disabled || !s->delays)
         return ff_filter_frame(ctx->outputs[0], frame);
 
     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 (i = 0; i < s->nb_delays; i++) {
         ChanDelay *d = &s->chandelay[i];
         const uint8_t *src = frame->extended_data[i];
         uint8_t *dst = out_frame->extended_data[i];
 
         if (!d->delay)
             memcpy(dst, src, frame->nb_samples * s->block_align);
         else
             s->delay_channel(d, frame->nb_samples, src, dst);
     }
 
     s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
     av_frame_free(&frame);
     return ff_filter_frame(ctx->outputs[0], out_frame);
 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AudioDelayContext *s = ctx->priv;
     int ret;
 
     ret = ff_request_frame(ctx->inputs[0]);
     if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {
         int nb_samples = FFMIN(s->max_delay, 2048);
         AVFrame *frame;
 
         frame = ff_get_audio_buffer(outlink, nb_samples);
         if (!frame)
             return AVERROR(ENOMEM);
         s->max_delay -= 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)
 {
     AudioDelayContext *s = ctx->priv;
     int i;
 
     for (i = 0; i < s->nb_delays; i++)

         av_freep(&s->chandelay[i].samples);

     av_freep(&s->chandelay);
 }
 
 static const AVFilterPad adelay_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
         .filter_frame = filter_frame,
     },
     { NULL }
 };
 
 static const AVFilterPad adelay_outputs[] = {
     {
         .name          = "default",
         .request_frame = request_frame,
         .type          = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 

 AVFilter ff_af_adelay = {

     .name          = "adelay",
     .description   = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
     .query_formats = query_formats,
     .priv_size     = sizeof(AudioDelayContext),
     .priv_class    = &adelay_class,
     .uninit        = uninit,
     .inputs        = adelay_inputs,
     .outputs       = adelay_outputs,
     .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
 };