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/*
   This file is part of GNUnet.
   Copyright (C) 2016 GNUnet e.V.

   GNUnet is free software: you can redistribute it and/or modify it
   under the terms of the GNU Affero General Public License as published
   by the Free Software Foundation, either version 3 of the License,
   or (at your option) any later version.

   GNUnet 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
   Affero General Public License for more details.

   You should have received a copy of the GNU Affero General Public License
   along with this program.  If not, see <http://www.gnu.org/licenses/>.

     SPDX-License-Identifier: AGPL3.0-or-later
 */
/**
 * @file conversation/gnunet_gst_def.h
 * @brief FIXME
 * @author Hark
 */

#include <getopt.h>
#include <string.h>
#include <stdio.h>
#include <ctype.h>
#include <stdlib.h>
#include <unistd.h>
#include <time.h>
#include <regex.h>


#include "platform.h"
#include "gnunet_util_lib.h"
#include "gnunet_protocols.h"
// #include "gnunet/conversation.h" doesn't get installed
#include "conversation.h"
#include "gnunet_constants.h"
#include "gnunet_core_service.h"
#include "gnunet_common.h"

/*
 #include <gst/gst.h>
 #include <gst/audio/gstaudiobasesrc.h>
 #include <gst/app/gstappsrc.h>
 */

/* huh
 #include <glib-2.0/glib.h>

 #include <gstreamer-1.0/gst/gst.h>
 #include <gstreamer-1.0/gst/pbutils/pbutils.h>
 #include <gstreamer-1.0/gst/video/videooverlay.h>
 #include <gstreamer-1.0/gst/audio/gstaudiobasesrc.h>
 #include <gstreamer-1.0/gst/app/gstappsrc.h>
 */

#include <gst/gst.h>
#include <gst/audio/gstaudiobasesrc.h>
#include <gst/app/gstappsrc.h>
#include <glib.h>
#include <gst/app/gstappsink.h>

// sockets
#if HAVE_NETINET_IN_H
#include <netinet/in.h>
#endif
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netdb.h>

#include <sys/types.h>
#include <fcntl.h>


// glib stuff
// #include <glib.h>
#include <glib-2.0/glib/gprintf.h>
// #include <glib-unix.h>

// static struct AudioMessage *audio_message;


typedef struct GNUNET_gstData GNUNET_gstData;
struct GNUNET_gstData
{
  // general
  GstPipeline *pipeline;

  // things
  struct AudioMessage *audio_message;
  struct GNUNET_MessageStreamTokenizer *stdin_mst;
  GstElement *appsrc;
  GstElement *appsink;
  // settings
  int audiobackend;
  int dropsilence;
  int usertp;
  int pure_ogg;
  char *jack_pp_in;
  char *jack_pp_out;
};


#define DEBUG_READ_PURE_OGG 1
#define DEBUG_RECORD_PURE_OGG 1


/**
 * How much data to read in one go
 */
#define MAXLINE 4096

/**
 * Max number of microseconds to buffer in audiosink.
 * Default is 1000
 */
#define BUFFER_TIME 1000

/**
 * Min number of microseconds to buffer in audiosink.
 * Default is 1000
 */
#define LATENCY_TIME 1000


/**
 * Number of channels.
 * Must be one of the following (from libopusenc documentation):
 * 1, 2
 */
#define OPUS_CHANNELS 1

/**
 * Maximal size of a single opus packet.
 */
#define MAX_PAYLOAD_SIZE (1024 / OPUS_CHANNELS)

/**
 * Size of a single frame fed to the encoder, in ms.
 * Must be one of the following (from libopus documentation):
 * 2.5, 5, 10, 20, 40 or 60
 */
#define OPUS_FRAME_SIZE 40

/**
 * Expected packet loss to prepare for, in percents.
 */
#define PACKET_LOSS_PERCENTAGE 1

/**
 * Set to 1 to enable forward error correction.
 * Set to 0 to disable.
 */
#define INBAND_FEC_MODE 1

/**
 * Max number of microseconds to buffer in audiosource.
 * Default is 200000
 */
#define BUFFER_TIME 1000 /* 1ms */

/**
 * Min number of microseconds to buffer in audiosource.
 * Default is 10000
 */
#define LATENCY_TIME 1000 /* 1ms */

/**
 * Maximum delay in multiplexing streams, in ns.
 * Setting this to 0 forces page flushing, which
 * decreases delay, but increases overhead.
 */
#define OGG_MAX_DELAY 0

/**
 * Maximum delay for sending out a page, in ns.
 * Setting this to 0 forces page flushing, which
 * decreases delay, but increases overhead.
 */
#define OGG_MAX_PAGE_DELAY 0

#define SAMPLING_RATE 48000

enum
{
  AUTO,
  JACK,
  ALSA,
  FAKE,
  TEST
};

enum
{
  SOURCE,
  SINK
};

enum
{
  ENCODER,
  DECODER
};

enum
{
  FAIL,
  OK
};

enum
{
  SPEAKER,
  MICROPHONE
};