/*
 *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include <iostream>

#include "gflags/gflags.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/tools/event_log_visualizer/analyzer.h"
#include "webrtc/tools/event_log_visualizer/plot_base.h"
#include "webrtc/tools/event_log_visualizer/plot_python.h"

DEFINE_bool(incoming, true, "Plot statistics for incoming packets.");
DEFINE_bool(outgoing, true, "Plot statistics for outgoing packets.");
DEFINE_bool(plot_all, true, "Plot all different data types.");
DEFINE_bool(plot_packets,
            false,
            "Plot bar graph showing the size of each packet.");
DEFINE_bool(plot_audio_playout,
            false,
            "Plot bar graph showing the time between each audio playout.");
DEFINE_bool(plot_audio_level,
            false,
            "Plot line graph showing the audio level.");
DEFINE_bool(
    plot_sequence_number,
    false,
    "Plot the difference in sequence number between consecutive packets.");
DEFINE_bool(
    plot_delay_change,
    false,
    "Plot the difference in 1-way path delay between consecutive packets.");
DEFINE_bool(plot_accumulated_delay_change,
            false,
            "Plot the accumulated 1-way path delay change, or the path delay "
            "change compared to the first packet.");
DEFINE_bool(plot_total_bitrate,
            false,
            "Plot the total bitrate used by all streams.");
DEFINE_bool(plot_stream_bitrate,
            false,
            "Plot the bitrate used by each stream.");
DEFINE_bool(plot_bwe,
            false,
            "Run the bandwidth estimator with the logged rtp and rtcp and plot "
            "the output.");
DEFINE_bool(plot_network_delay_feedback,
            false,
            "Compute network delay based on sent packets and the received "
            "transport feedback.");
DEFINE_bool(plot_fraction_loss,
            false,
            "Plot packet loss in percent for outgoing packets (as perceived by "
            "the send-side bandwidth estimator).");

int main(int argc, char* argv[]) {
  std::string program_name = argv[0];
  std::string usage =
      "A tool for visualizing WebRTC event logs.\n"
      "Example usage:\n" +
      program_name + " <logfile> | python\n" + "Run " + program_name +
      " --help for a list of command line options\n";
  google::SetUsageMessage(usage);
  google::ParseCommandLineFlags(&argc, &argv, true);

  if (argc != 2) {
    // Print usage information.
    std::cout << google::ProgramUsage();
    return 0;
  }

  std::string filename = argv[1];

  webrtc::ParsedRtcEventLog parsed_log;

  if (!parsed_log.ParseFile(filename)) {
    std::cerr << "Could not parse the entire log file." << std::endl;
    std::cerr << "Proceeding to analyze the first "
              << parsed_log.GetNumberOfEvents() << " events in the file."
              << std::endl;
  }

  webrtc::plotting::EventLogAnalyzer analyzer(parsed_log);
  std::unique_ptr<webrtc::plotting::PlotCollection> collection(
      new webrtc::plotting::PythonPlotCollection());

  if (FLAGS_plot_all || FLAGS_plot_packets) {
    if (FLAGS_incoming) {
      analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
                                 collection->AppendNewPlot());
      analyzer.CreateAccumulatedPacketsGraph(
          webrtc::PacketDirection::kIncomingPacket,
          collection->AppendNewPlot());
    }
    if (FLAGS_outgoing) {
      analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
                                 collection->AppendNewPlot());
      analyzer.CreateAccumulatedPacketsGraph(
          webrtc::PacketDirection::kOutgoingPacket,
          collection->AppendNewPlot());
    }
  }

  if (FLAGS_plot_all || FLAGS_plot_audio_playout) {
    analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
  }

  if (FLAGS_plot_all || FLAGS_plot_audio_level) {
    analyzer.CreateAudioLevelGraph(collection->AppendNewPlot());
  }

  if (FLAGS_plot_all || FLAGS_plot_sequence_number) {
    if (FLAGS_incoming) {
      analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
    }
  }

  if (FLAGS_plot_all || FLAGS_plot_delay_change) {
    if (FLAGS_incoming) {
      analyzer.CreateDelayChangeGraph(collection->AppendNewPlot());
    }
  }

  if (FLAGS_plot_all || FLAGS_plot_accumulated_delay_change) {
    if (FLAGS_incoming) {
      analyzer.CreateAccumulatedDelayChangeGraph(collection->AppendNewPlot());
    }
  }

  if (FLAGS_plot_all || FLAGS_plot_fraction_loss) {
    analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
    analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
  }

  if (FLAGS_plot_all || FLAGS_plot_total_bitrate) {
    if (FLAGS_incoming) {
      analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
                                       collection->AppendNewPlot());
    }
    if (FLAGS_outgoing) {
      analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
                                       collection->AppendNewPlot());
    }
  }

  if (FLAGS_plot_all || FLAGS_plot_stream_bitrate) {
    if (FLAGS_incoming) {
      analyzer.CreateStreamBitrateGraph(
          webrtc::PacketDirection::kIncomingPacket,
          collection->AppendNewPlot());
    }
    if (FLAGS_outgoing) {
      analyzer.CreateStreamBitrateGraph(
          webrtc::PacketDirection::kOutgoingPacket,
          collection->AppendNewPlot());
    }
  }

  if (FLAGS_plot_all || FLAGS_plot_bwe) {
    analyzer.CreateBweSimulationGraph(collection->AppendNewPlot());
  }

  if (FLAGS_plot_all || FLAGS_plot_network_delay_feedback) {
    analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot());
  }

  collection->Draw();

  return 0;
}