/*
  * Audio Mix Filter
  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
  *
  * This file is part of Libav.
  *
  * Libav 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.
  *
  * Libav 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 Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**
  * @file
  * Audio Mix Filter
  *
  * Mixes audio from multiple sources into a single output. The channel layout,
  * sample rate, and sample format will be the same for all inputs and the
  * output.
  */
 
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"

 #include "libavutil/channel_layout.h"

 #include "libavutil/common.h"

 #include "libavutil/float_dsp.h"

 #include "libavutil/mathematics.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 
 #include "audio.h"
 #include "avfilter.h"
 #include "formats.h"
 #include "internal.h"
 
 #define INPUT_OFF      0    /**< input has reached EOF */
 #define INPUT_ON       1    /**< input is active */
 #define INPUT_INACTIVE 2    /**< input is on, but is currently inactive */
 
 #define DURATION_LONGEST  0
 #define DURATION_SHORTEST 1
 #define DURATION_FIRST    2
 
 
 typedef struct FrameInfo {
     int nb_samples;
     int64_t pts;
     struct FrameInfo *next;
 } FrameInfo;
 
 /**
  * Linked list used to store timestamps and frame sizes of all frames in the
  * FIFO for the first input.
  *
  * This is needed to keep timestamps synchronized for the case where multiple
  * input frames are pushed to the filter for processing before a frame is
  * requested by the output link.
  */
 typedef struct FrameList {
     int nb_frames;
     int nb_samples;
     FrameInfo *list;
     FrameInfo *end;
 } FrameList;
 
 static void frame_list_clear(FrameList *frame_list)
 {
     if (frame_list) {
         while (frame_list->list) {
             FrameInfo *info = frame_list->list;
             frame_list->list = info->next;
             av_free(info);
         }
         frame_list->nb_frames  = 0;
         frame_list->nb_samples = 0;
         frame_list->end        = NULL;
     }
 }
 
 static int frame_list_next_frame_size(FrameList *frame_list)
 {
     if (!frame_list->list)
         return 0;
     return frame_list->list->nb_samples;
 }
 
 static int64_t frame_list_next_pts(FrameList *frame_list)
 {
     if (!frame_list->list)
         return AV_NOPTS_VALUE;
     return frame_list->list->pts;
 }
 
 static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
 {
     if (nb_samples >= frame_list->nb_samples) {
         frame_list_clear(frame_list);
     } else {
         int samples = nb_samples;
         while (samples > 0) {
             FrameInfo *info = frame_list->list;
             av_assert0(info != NULL);
             if (info->nb_samples <= samples) {
                 samples -= info->nb_samples;
                 frame_list->list = info->next;
                 if (!frame_list->list)
                     frame_list->end = NULL;
                 frame_list->nb_frames--;
                 frame_list->nb_samples -= info->nb_samples;
                 av_free(info);
             } else {
                 info->nb_samples       -= samples;
                 info->pts              += samples;
                 frame_list->nb_samples -= samples;
                 samples = 0;
             }
         }
     }
 }
 
 static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
 {
     FrameInfo *info = av_malloc(sizeof(*info));
     if (!info)
         return AVERROR(ENOMEM);
     info->nb_samples = nb_samples;
     info->pts        = pts;
     info->next       = NULL;
 
     if (!frame_list->list) {
         frame_list->list = info;
         frame_list->end  = info;
     } else {
         av_assert0(frame_list->end != NULL);
         frame_list->end->next = info;
         frame_list->end       = info;
     }
     frame_list->nb_frames++;
     frame_list->nb_samples += nb_samples;
 
     return 0;
 }
 
 
 typedef struct MixContext {
     const AVClass *class;       /**< class for AVOptions */

     AVFloatDSPContext fdsp;

 
     int nb_inputs;              /**< number of inputs */
     int active_inputs;          /**< number of input currently active */
     int duration_mode;          /**< mode for determining duration */
     float dropout_transition;   /**< transition time when an input drops out */
 
     int nb_channels;            /**< number of channels */
     int sample_rate;            /**< sample rate */

     int planar;

     AVAudioFifo **fifos;        /**< audio fifo for each input */
     uint8_t *input_state;       /**< current state of each input */
     float *input_scale;         /**< mixing scale factor for each input */
     float scale_norm;           /**< normalization factor for all inputs */
     int64_t next_pts;           /**< calculated pts for next output frame */
     FrameList *frame_list;      /**< list of frame info for the first input */
 } MixContext;
 
 #define OFFSET(x) offsetof(MixContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM

 #define F AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption amix_options[] = {

     { "inputs", "Number of inputs.",

             OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 32, A|F },

     { "duration", "How to determine the end-of-stream.",

             OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0,  2, A|F, "duration" },

         { "longest",  "Duration of longest input.",  0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST  }, INT_MIN, INT_MAX, A|F, "duration" },
         { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A|F, "duration" },
         { "first",    "Duration of first input.",    0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST    }, INT_MIN, INT_MAX, A|F, "duration" },

     { "dropout_transition", "Transition time, in seconds, for volume "
                             "renormalization when an input stream ends.",

             OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },

     { NULL },
 };
 

 AVFILTER_DEFINE_CLASS(amix);

 
 /**
  * Update the scaling factors to apply to each input during mixing.
  *
  * This balances the full volume range between active inputs and handles
  * volume transitions when EOF is encountered on an input but mixing continues
  * with the remaining inputs.
  */
 static void calculate_scales(MixContext *s, int nb_samples)
 {
     int i;
 
     if (s->scale_norm > s->active_inputs) {
         s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);
         s->scale_norm = FFMAX(s->scale_norm, s->active_inputs);
     }
 
     for (i = 0; i < s->nb_inputs; i++) {
         if (s->input_state[i] == INPUT_ON)
             s->input_scale[i] = 1.0f / s->scale_norm;
         else
             s->input_scale[i] = 0.0f;
     }
 }
 
 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     MixContext *s      = ctx->priv;
     int i;
     char buf[64];
 

     s->planar          = av_sample_fmt_is_planar(outlink->format);

     s->sample_rate     = outlink->sample_rate;
     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
     s->next_pts        = AV_NOPTS_VALUE;
 
     s->frame_list = av_mallocz(sizeof(*s->frame_list));
     if (!s->frame_list)
         return AVERROR(ENOMEM);
 
     s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos));
     if (!s->fifos)
         return AVERROR(ENOMEM);
 
     s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
     for (i = 0; i < s->nb_inputs; i++) {
         s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
         if (!s->fifos[i])
             return AVERROR(ENOMEM);
     }
 
     s->input_state = av_malloc(s->nb_inputs);
     if (!s->input_state)
         return AVERROR(ENOMEM);
     memset(s->input_state, INPUT_ON, s->nb_inputs);
     s->active_inputs = s->nb_inputs;
 
     s->input_scale = av_mallocz(s->nb_inputs * sizeof(*s->input_scale));
     if (!s->input_scale)
         return AVERROR(ENOMEM);
     s->scale_norm = s->active_inputs;
     calculate_scales(s, 0);
 
     av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
 
     av_log(ctx, AV_LOG_VERBOSE,

            "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,

            av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
 
     return 0;
 }
 
 /**
  * Read samples from the input FIFOs, mix, and write to the output link.
  */
 static int output_frame(AVFilterLink *outlink, int nb_samples)
 {
     AVFilterContext *ctx = outlink->src;
     MixContext      *s = ctx->priv;
     AVFilterBufferRef *out_buf, *in_buf;
     int i;
 
     calculate_scales(s, nb_samples);
 
     out_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
     if (!out_buf)
         return AVERROR(ENOMEM);
 
     in_buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);

     if (!in_buf) {
         avfilter_unref_buffer(out_buf);

         return AVERROR(ENOMEM);

     }

 
     for (i = 0; i < s->nb_inputs; i++) {
         if (s->input_state[i] == INPUT_ON) {

             int planes, plane_size, p;
 

             av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
                                nb_samples);

 
             planes     = s->planar ? s->nb_channels : 1;
             plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
             plane_size = FFALIGN(plane_size, 16);
 
             for (p = 0; p < planes; p++) {
                 s->fdsp.vector_fmac_scalar((float *)out_buf->extended_data[p],
                                            (float *) in_buf->extended_data[p],
                                            s->input_scale[i], plane_size);
             }

         }
     }
     avfilter_unref_buffer(in_buf);
 
     out_buf->pts = s->next_pts;
     if (s->next_pts != AV_NOPTS_VALUE)
         s->next_pts += nb_samples;
 

     return ff_filter_frame(outlink, out_buf);

 }
 
 /**
  * Returns the smallest number of samples available in the input FIFOs other
  * than that of the first input.
  */
 static int get_available_samples(MixContext *s)
 {
     int i;
     int available_samples = INT_MAX;
 
     av_assert0(s->nb_inputs > 1);
 
     for (i = 1; i < s->nb_inputs; i++) {
         int nb_samples;
         if (s->input_state[i] == INPUT_OFF)
             continue;
         nb_samples = av_audio_fifo_size(s->fifos[i]);
         available_samples = FFMIN(available_samples, nb_samples);
     }
     if (available_samples == INT_MAX)
         return 0;
     return available_samples;
 }
 
 /**
  * Requests a frame, if needed, from each input link other than the first.
  */
 static int request_samples(AVFilterContext *ctx, int min_samples)
 {
     MixContext *s = ctx->priv;
     int i, ret;
 
     av_assert0(s->nb_inputs > 1);
 
     for (i = 1; i < s->nb_inputs; i++) {
         ret = 0;
         if (s->input_state[i] == INPUT_OFF)
             continue;
         while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples)

             ret = ff_request_frame(ctx->inputs[i]);

         if (ret == AVERROR_EOF) {
             if (av_audio_fifo_size(s->fifos[i]) == 0) {
                 s->input_state[i] = INPUT_OFF;
                 continue;
             }

         } else if (ret < 0)

             return ret;
     }
     return 0;
 }
 
 /**
  * Calculates the number of active inputs and determines EOF based on the
  * duration option.
  *
  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
  */
 static int calc_active_inputs(MixContext *s)
 {
     int i;
     int active_inputs = 0;
     for (i = 0; i < s->nb_inputs; i++)
         active_inputs += !!(s->input_state[i] != INPUT_OFF);
     s->active_inputs = active_inputs;
 
     if (!active_inputs ||
         (s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) ||
         (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
         return AVERROR_EOF;
     return 0;
 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     MixContext      *s = ctx->priv;
     int ret;
     int wanted_samples, available_samples;
 

     ret = calc_active_inputs(s);
     if (ret < 0)
         return ret;
 

     if (s->input_state[0] == INPUT_OFF) {
         ret = request_samples(ctx, 1);
         if (ret < 0)
             return ret;
 
         ret = calc_active_inputs(s);
         if (ret < 0)
             return ret;
 
         available_samples = get_available_samples(s);
         if (!available_samples)

             return AVERROR(EAGAIN);

 
         return output_frame(outlink, available_samples);
     }
 
     if (s->frame_list->nb_frames == 0) {

         ret = ff_request_frame(ctx->inputs[0]);

         if (ret == AVERROR_EOF) {
             s->input_state[0] = INPUT_OFF;
             if (s->nb_inputs == 1)
                 return AVERROR_EOF;
             else
                 return AVERROR(EAGAIN);

         } else if (ret < 0)

             return ret;
     }
     av_assert0(s->frame_list->nb_frames > 0);
 
     wanted_samples = frame_list_next_frame_size(s->frame_list);
 
     if (s->active_inputs > 1) {

         ret = request_samples(ctx, wanted_samples);
         if (ret < 0)
             return ret;
 
         ret = calc_active_inputs(s);
         if (ret < 0)
             return ret;

     }

     if (s->active_inputs > 1) {

         available_samples = get_available_samples(s);
         if (!available_samples)

             return AVERROR(EAGAIN);

         available_samples = FFMIN(available_samples, wanted_samples);
     } else {
         available_samples = wanted_samples;
     }
 
     s->next_pts = frame_list_next_pts(s->frame_list);
     frame_list_remove_samples(s->frame_list, available_samples);
 
     return output_frame(outlink, available_samples);
 }
 

 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)

 {
     AVFilterContext  *ctx = inlink->dst;
     MixContext       *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];

     int i, ret = 0;

     for (i = 0; i < ctx->nb_inputs; i++)

         if (ctx->inputs[i] == inlink)
             break;

     if (i >= ctx->nb_inputs) {

         av_log(ctx, AV_LOG_ERROR, "unknown input link\n");

         ret = AVERROR(EINVAL);
         goto fail;

     }
 
     if (i == 0) {
         int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
                                    outlink->time_base);

         ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
         if (ret < 0)
             goto fail;

     }
 

     ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
                               buf->audio->nb_samples);

 fail:

     avfilter_unref_buffer(buf);

 
     return ret;

 }
 

 static int init(AVFilterContext *ctx, const char *args)

 {
     MixContext *s = ctx->priv;
     int i, ret;
 
     s->class = &amix_class;
     av_opt_set_defaults(s);
 

     if ((ret = av_set_options_string(s, args, "=", ":")) < 0)

         return ret;
     av_opt_free(s);
 
     for (i = 0; i < s->nb_inputs; i++) {
         char name[32];
         AVFilterPad pad = { 0 };
 
         snprintf(name, sizeof(name), "input%d", i);
         pad.type           = AVMEDIA_TYPE_AUDIO;
         pad.name           = av_strdup(name);

         pad.filter_frame   = filter_frame;

         ff_insert_inpad(ctx, i, &pad);

     }
 

     avpriv_float_dsp_init(&s->fdsp, 0);
 

     return 0;
 }
 
 static void uninit(AVFilterContext *ctx)
 {
     int i;
     MixContext *s = ctx->priv;
 
     if (s->fifos) {
         for (i = 0; i < s->nb_inputs; i++)
             av_audio_fifo_free(s->fifos[i]);
         av_freep(&s->fifos);
     }
     frame_list_clear(s->frame_list);
     av_freep(&s->frame_list);
     av_freep(&s->input_state);
     av_freep(&s->input_scale);
 

     for (i = 0; i < ctx->nb_inputs; i++)

         av_freep(&ctx->input_pads[i].name);
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats = NULL;

     ff_add_format(&formats, AV_SAMPLE_FMT_FLT);

     ff_add_format(&formats, AV_SAMPLE_FMT_FLTP);

     ff_set_common_formats(ctx, formats);

     ff_set_common_channel_layouts(ctx, ff_all_channel_layouts());
     ff_set_common_samplerates(ctx, ff_all_samplerates());
     return 0;
 }
 

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

 AVFilter avfilter_af_amix = {
     .name          = "amix",
     .description   = NULL_IF_CONFIG_SMALL("Audio mixing."),
     .priv_size     = sizeof(MixContext),
 
     .init           = init,
     .uninit         = uninit,
     .query_formats  = query_formats,
 

     .inputs    = NULL,

     .outputs   = avfilter_af_amix_outputs,

     .priv_class = &amix_class,

 };