/*
 *  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 <stdio.h>
#include <string.h>

#include <iostream>
#include <map>
#include <memory>
#include <string>
#include <utility>

#include "absl/algorithm/container.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "rtc_tools/event_log_visualizer/analyzer.h"
#include "rtc_tools/event_log_visualizer/plot_base.h"
#include "rtc_tools/event_log_visualizer/plot_protobuf.h"
#include "rtc_tools/event_log_visualizer/plot_python.h"
#include "system_wrappers/include/field_trial.h"
#include "test/field_trial.h"
#include "test/testsupport/file_utils.h"

WEBRTC_DEFINE_string(
    plot,
    "default",
    "A comma separated list of plot names. See below for valid options.");

WEBRTC_DEFINE_string(
    force_fieldtrials,
    "",
    "Field trials control experimental feature code which can be forced. "
    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
    " will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
    "trials are separated by \"/\"");
WEBRTC_DEFINE_string(wav_filename,
                     "",
                     "Path to wav file used for simulation of jitter buffer");
WEBRTC_DEFINE_bool(help, false, "prints this message");

WEBRTC_DEFINE_bool(
    show_detector_state,
    false,
    "Show the state of the delay based BWE detector on the total "
    "bitrate graph");

WEBRTC_DEFINE_bool(show_alr_state,
                   false,
                   "Show the state ALR state on the total bitrate graph");

WEBRTC_DEFINE_bool(
    parse_unconfigured_header_extensions,
    true,
    "Attempt to parse unconfigured header extensions using the default "
    "WebRTC mapping. This can give very misleading results if the "
    "application negotiates a different mapping.");

WEBRTC_DEFINE_bool(print_triage_alerts,
                   false,
                   "Print triage alerts, i.e. a list of potential problems.");

WEBRTC_DEFINE_bool(
    normalize_time,
    true,
    "Normalize the log timestamps so that the call starts at time 0.");

WEBRTC_DEFINE_bool(shared_xaxis,
                   false,
                   "Share x-axis between all plots so that zooming in one plot "
                   "updates all the others too. A downside is that certain "
                   "operations like panning become much slower.");

WEBRTC_DEFINE_bool(protobuf_output,
                   false,
                   "Output charts as protobuf instead of python code.");

using webrtc::Plot;

namespace {
std::vector<std::string> StrSplit(const std::string& s,
                                  const std::string& delimiter) {
  std::vector<std::string> v;
  size_t pos = 0;
  while (pos < s.length()) {
    const std::string token = s.substr(pos, s.find(delimiter, pos) - pos);
    pos += token.length() + delimiter.length();
    v.push_back(token);
  }
  return v;
}

struct PlotDeclaration {
  PlotDeclaration(const std::string& label, std::function<void(Plot*)> f)
      : label(label), enabled(false), plot_func(f) {}
  const std::string label;
  bool enabled;
  // TODO(terelius): Add a help text/explanation.
  const std::function<void(Plot*)> plot_func;
};

class PlotMap {
 public:
  void RegisterPlot(const std::string& label, std::function<void(Plot*)> f) {
    for (const auto& plot : plots_) {
      RTC_DCHECK(plot.label != label)
          << "Can't use the same label for multiple plots";
    }
    plots_.push_back({label, f});
  }

  bool EnablePlotsByFlags(
      const std::vector<std::string>& flags,
      const std::map<std::string, std::vector<std::string>>& flag_aliases) {
    bool status = true;
    for (const std::string& flag : flags) {
      auto alias_it = flag_aliases.find(flag);
      if (alias_it != flag_aliases.end()) {
        const auto& replacements = alias_it->second;
        for (const auto& replacement : replacements) {
          status &= EnablePlotByFlag(replacement);
        }
      } else {
        status &= EnablePlotByFlag(flag);
      }
    }
    return status;
  }

  void EnableAllPlots() {
    for (auto& plot : plots_) {
      plot.enabled = true;
    }
  }

  std::vector<PlotDeclaration>::iterator begin() { return plots_.begin(); }
  std::vector<PlotDeclaration>::iterator end() { return plots_.end(); }

 private:
  bool EnablePlotByFlag(const std::string& flag) {
    for (auto& plot : plots_) {
      if (plot.label == flag) {
        plot.enabled = true;
        return true;
      }
    }
    if (flag == "simulated_neteq_jitter_buffer_delay") {
      // This flag is handled separately.
      return true;
    }
    std::cerr << "Unrecognized plot name \'" << flag << "\'. Aborting."
              << std::endl;
    return false;
  }

  std::vector<PlotDeclaration> plots_;
};
}  // namespace

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";

  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);

  // Flag replacements
  std::map<std::string, std::vector<std::string>> flag_aliases = {
      {"default",
       {"incoming_delay", "incoming_loss_rate", "incoming_bitrate",
        "outgoing_bitrate", "incoming_stream_bitrate",
        "outgoing_stream_bitrate", "network_delay_feedback",
        "fraction_loss_feedback"}},
      {"sendside_bwe",
       {"outgoing_packet_sizes", "outgoing_bitrate", "outgoing_stream_bitrate",
        "simulated_sendside_bwe", "network_delay_feedback",
        "fraction_loss_feedback"}},
      {"receiveside_bwe",
       {"incoming_packet_sizes", "incoming_delay", "incoming_loss_rate",
        "incoming_bitrate", "incoming_stream_bitrate",
        "simulated_receiveside_bwe"}},
      {"rtcp_details",
       {"incoming_rtcp_fraction_lost", "outgoing_rtcp_fraction_lost",
        "incoming_rtcp_cumulative_lost", "outgoing_rtcp_cumulative_lost",
        "incoming_rtcp_highest_seq_number", "outgoing_rtcp_highest_seq_number",
        "incoming_rtcp_delay_since_last_sr",
        "outgoing_rtcp_delay_since_last_sr"}},
      {"simulated_neteq_stats",
       {"simulated_neteq_jitter_buffer_delay",
        "simulated_neteq_preferred_buffer_size",
        "simulated_neteq_concealment_events",
        "simulated_neteq_packet_loss_rate", "simulated_neteq_preemptive_rate",
        "simulated_neteq_accelerate_rate", "simulated_neteq_speech_expand_rate",
        "simulated_neteq_expand_rate"}}};

  std::vector<std::string> plot_flags = StrSplit(FLAG_plot, ",");

  // InitFieldTrialsFromString stores the char*, so the char array must outlive
  // the application.
  webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);

  webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions header_extensions =
      webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions::kDontParse;
  if (FLAG_parse_unconfigured_header_extensions) {
    header_extensions = webrtc::ParsedRtcEventLog::
        UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig;
  }
  webrtc::ParsedRtcEventLog parsed_log(header_extensions);

  if (argc == 2) {
    std::string filename = argv[1];
    if (!parsed_log.ParseFile(filename)) {
      std::cerr << "Could not parse the entire log file." << std::endl;
      std::cerr << "Only the parsable events will be analyzed." << std::endl;
    }
  }

  webrtc::EventLogAnalyzer analyzer(parsed_log, FLAG_normalize_time);
  std::unique_ptr<webrtc::PlotCollection> collection;
  if (FLAG_protobuf_output) {
    collection.reset(new webrtc::ProtobufPlotCollection());
  } else {
    collection.reset(new webrtc::PythonPlotCollection(FLAG_shared_xaxis));
  }

  PlotMap plots;
  plots.RegisterPlot("incoming_packet_sizes", [&](Plot* plot) {
    analyzer.CreatePacketGraph(webrtc::kIncomingPacket, plot);
  });

  plots.RegisterPlot("outgoing_packet_sizes", [&](Plot* plot) {
    analyzer.CreatePacketGraph(webrtc::kOutgoingPacket, plot);
  });
  plots.RegisterPlot("incoming_rtcp_types", [&](Plot* plot) {
    analyzer.CreateRtcpTypeGraph(webrtc::kIncomingPacket, plot);
  });
  plots.RegisterPlot("outgoing_rtcp_types", [&](Plot* plot) {
    analyzer.CreateRtcpTypeGraph(webrtc::kOutgoingPacket, plot);
  });
  plots.RegisterPlot("incoming_packet_count", [&](Plot* plot) {
    analyzer.CreateAccumulatedPacketsGraph(webrtc::kIncomingPacket, plot);
  });
  plots.RegisterPlot("outgoing_packet_count", [&](Plot* plot) {
    analyzer.CreateAccumulatedPacketsGraph(webrtc::kOutgoingPacket, plot);
  });
  plots.RegisterPlot("audio_playout",
                     [&](Plot* plot) { analyzer.CreatePlayoutGraph(plot); });
  plots.RegisterPlot("incoming_audio_level", [&](Plot* plot) {
    analyzer.CreateAudioLevelGraph(webrtc::kIncomingPacket, plot);
  });
  plots.RegisterPlot("outgoing_audio_level", [&](Plot* plot) {
    analyzer.CreateAudioLevelGraph(webrtc::kOutgoingPacket, plot);
  });
  plots.RegisterPlot("incoming_sequence_number_delta", [&](Plot* plot) {
    analyzer.CreateSequenceNumberGraph(plot);
  });
  plots.RegisterPlot("incoming_delay", [&](Plot* plot) {
    analyzer.CreateIncomingDelayGraph(plot);
  });
  plots.RegisterPlot("incoming_loss_rate", [&](Plot* plot) {
    analyzer.CreateIncomingPacketLossGraph(plot);
  });
  plots.RegisterPlot("incoming_bitrate", [&](Plot* plot) {
    analyzer.CreateTotalIncomingBitrateGraph(plot);
  });
  plots.RegisterPlot("outgoing_bitrate", [&](Plot* plot) {
    analyzer.CreateTotalOutgoingBitrateGraph(plot, FLAG_show_detector_state,
                                             FLAG_show_alr_state);
  });
  plots.RegisterPlot("incoming_stream_bitrate", [&](Plot* plot) {
    analyzer.CreateStreamBitrateGraph(webrtc::kIncomingPacket, plot);
  });
  plots.RegisterPlot("outgoing_stream_bitrate", [&](Plot* plot) {
    analyzer.CreateStreamBitrateGraph(webrtc::kOutgoingPacket, plot);
  });
  plots.RegisterPlot("incoming_layer_bitrate_allocation", [&](Plot* plot) {
    analyzer.CreateBitrateAllocationGraph(webrtc::kIncomingPacket, plot);
  });
  plots.RegisterPlot("outgoing_layer_bitrate_allocation", [&](Plot* plot) {
    analyzer.CreateBitrateAllocationGraph(webrtc::kOutgoingPacket, plot);
  });
  plots.RegisterPlot("simulated_receiveside_bwe", [&](Plot* plot) {
    analyzer.CreateReceiveSideBweSimulationGraph(plot);
  });
  plots.RegisterPlot("simulated_sendside_bwe", [&](Plot* plot) {
    analyzer.CreateSendSideBweSimulationGraph(plot);
  });
  plots.RegisterPlot("simulated_goog_cc", [&](Plot* plot) {
    analyzer.CreateGoogCcSimulationGraph(plot);
  });
  plots.RegisterPlot("network_delay_feedback", [&](Plot* plot) {
    analyzer.CreateNetworkDelayFeedbackGraph(plot);
  });
  plots.RegisterPlot("fraction_loss_feedback", [&](Plot* plot) {
    analyzer.CreateFractionLossGraph(plot);
  });
  plots.RegisterPlot("incoming_timestamps", [&](Plot* plot) {
    analyzer.CreateTimestampGraph(webrtc::kIncomingPacket, plot);
  });
  plots.RegisterPlot("outgoing_timestamps", [&](Plot* plot) {
    analyzer.CreateTimestampGraph(webrtc::kOutgoingPacket, plot);
  });

  auto GetFractionLost = [](const webrtc::rtcp::ReportBlock& block) -> float {
    return static_cast<double>(block.fraction_lost()) / 256 * 100;
  };
  plots.RegisterPlot("incoming_rtcp_fraction_lost", [&](Plot* plot) {
    analyzer.CreateSenderAndReceiverReportPlot(
        webrtc::kIncomingPacket, GetFractionLost,
        "Fraction lost (incoming RTCP)", "Loss rate (percent)", plot);
  });
  plots.RegisterPlot("outgoing_rtcp_fraction_lost", [&](Plot* plot) {
    analyzer.CreateSenderAndReceiverReportPlot(
        webrtc::kOutgoingPacket, GetFractionLost,
        "Fraction lost (outgoing RTCP)", "Loss rate (percent)", plot);
  });
  auto GetCumulativeLost = [](const webrtc::rtcp::ReportBlock& block) -> float {
    return block.cumulative_lost_signed();
  };
  plots.RegisterPlot("incoming_rtcp_cumulative_lost", [&](Plot* plot) {
    analyzer.CreateSenderAndReceiverReportPlot(
        webrtc::kIncomingPacket, GetCumulativeLost,
        "Cumulative lost packets (incoming RTCP)", "Packets", plot);
  });
  plots.RegisterPlot("outgoing_rtcp_cumulative_lost", [&](Plot* plot) {
    analyzer.CreateSenderAndReceiverReportPlot(
        webrtc::kOutgoingPacket, GetCumulativeLost,
        "Cumulative lost packets (outgoing RTCP)", "Packets", plot);
  });

  auto GetHighestSeqNumber =
      [](const webrtc::rtcp::ReportBlock& block) -> float {
    return block.extended_high_seq_num();
  };
  plots.RegisterPlot("incoming_rtcp_highest_seq_number", [&](Plot* plot) {
    analyzer.CreateSenderAndReceiverReportPlot(
        webrtc::kIncomingPacket, GetHighestSeqNumber,
        "Highest sequence number (incoming RTCP)", "Sequence number", plot);
  });
  plots.RegisterPlot("outgoing_rtcp_highest_seq_number", [&](Plot* plot) {
    analyzer.CreateSenderAndReceiverReportPlot(
        webrtc::kOutgoingPacket, GetHighestSeqNumber,
        "Highest sequence number (outgoing RTCP)", "Sequence number", plot);
  });

  auto DelaySinceLastSr = [](const webrtc::rtcp::ReportBlock& block) -> float {
    return static_cast<double>(block.delay_since_last_sr()) / 65536;
  };
  plots.RegisterPlot("incoming_rtcp_delay_since_last_sr", [&](Plot* plot) {
    analyzer.CreateSenderAndReceiverReportPlot(
        webrtc::kIncomingPacket, DelaySinceLastSr,
        "Delay since last received sender report (incoming RTCP)", "Time (s)",
        plot);
  });
  plots.RegisterPlot("outgoing_rtcp_delay_since_last_sr", [&](Plot* plot) {
    analyzer.CreateSenderAndReceiverReportPlot(
        webrtc::kOutgoingPacket, DelaySinceLastSr,
        "Delay since last received sender report (outgoing RTCP)", "Time (s)",
        plot);
  });

  plots.RegisterPlot("pacer_delay",
                     [&](Plot* plot) { analyzer.CreatePacerDelayGraph(plot); });
  plots.RegisterPlot("audio_encoder_bitrate", [&](Plot* plot) {
    analyzer.CreateAudioEncoderTargetBitrateGraph(plot);
  });
  plots.RegisterPlot("audio_encoder_frame_length", [&](Plot* plot) {
    analyzer.CreateAudioEncoderFrameLengthGraph(plot);
  });
  plots.RegisterPlot("audio_encoder_packet_loss", [&](Plot* plot) {
    analyzer.CreateAudioEncoderPacketLossGraph(plot);
  });
  plots.RegisterPlot("audio_encoder_fec", [&](Plot* plot) {
    analyzer.CreateAudioEncoderEnableFecGraph(plot);
  });
  plots.RegisterPlot("audio_encoder_dtx", [&](Plot* plot) {
    analyzer.CreateAudioEncoderEnableDtxGraph(plot);
  });
  plots.RegisterPlot("audio_encoder_num_channels", [&](Plot* plot) {
    analyzer.CreateAudioEncoderNumChannelsGraph(plot);
  });

  plots.RegisterPlot("ice_candidate_pair_config", [&](Plot* plot) {
    analyzer.CreateIceCandidatePairConfigGraph(plot);
  });
  plots.RegisterPlot("ice_connectivity_check", [&](Plot* plot) {
    analyzer.CreateIceConnectivityCheckGraph(plot);
  });
  plots.RegisterPlot("dtls_transport_state", [&](Plot* plot) {
    analyzer.CreateDtlsTransportStateGraph(plot);
  });
  plots.RegisterPlot("dtls_writable_state", [&](Plot* plot) {
    analyzer.CreateDtlsWritableStateGraph(plot);
  });

  std::string wav_path;
  if (FLAG_wav_filename[0] != '\0') {
    wav_path = FLAG_wav_filename;
  } else {
    wav_path = webrtc::test::ResourcePath(
        "audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
  }
  absl::optional<webrtc::EventLogAnalyzer::NetEqStatsGetterMap> neteq_stats;

  plots.RegisterPlot("simulated_neteq_expand_rate", [&](Plot* plot) {
    if (!neteq_stats) {
      neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
    }
    analyzer.CreateNetEqNetworkStatsGraph(
        *neteq_stats,
        [](const webrtc::NetEqNetworkStatistics& stats) {
          return stats.expand_rate / 16384.f;
        },
        "Expand rate", plot);
  });

  plots.RegisterPlot("simulated_neteq_speech_expand_rate", [&](Plot* plot) {
    if (!neteq_stats) {
      neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
    }
    analyzer.CreateNetEqNetworkStatsGraph(
        *neteq_stats,
        [](const webrtc::NetEqNetworkStatistics& stats) {
          return stats.speech_expand_rate / 16384.f;
        },
        "Speech expand rate", plot);
  });

  plots.RegisterPlot("simulated_neteq_accelerate_rate", [&](Plot* plot) {
    if (!neteq_stats) {
      neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
    }
    analyzer.CreateNetEqNetworkStatsGraph(
        *neteq_stats,
        [](const webrtc::NetEqNetworkStatistics& stats) {
          return stats.accelerate_rate / 16384.f;
        },
        "Accelerate rate", plot);
  });

  plots.RegisterPlot("simulated_neteq_preemptive_rate", [&](Plot* plot) {
    if (!neteq_stats) {
      neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
    }
    analyzer.CreateNetEqNetworkStatsGraph(
        *neteq_stats,
        [](const webrtc::NetEqNetworkStatistics& stats) {
          return stats.preemptive_rate / 16384.f;
        },
        "Preemptive rate", plot);
  });

  plots.RegisterPlot("simulated_neteq_packet_loss_rate", [&](Plot* plot) {
    if (!neteq_stats) {
      neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
    }
    analyzer.CreateNetEqNetworkStatsGraph(
        *neteq_stats,
        [](const webrtc::NetEqNetworkStatistics& stats) {
          return stats.packet_loss_rate / 16384.f;
        },
        "Packet loss rate", plot);
  });

  plots.RegisterPlot("simulated_neteq_concealment_events", [&](Plot* plot) {
    if (!neteq_stats) {
      neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
    }
    analyzer.CreateNetEqLifetimeStatsGraph(
        *neteq_stats,
        [](const webrtc::NetEqLifetimeStatistics& stats) {
          return static_cast<float>(stats.concealment_events);
        },
        "Concealment events", plot);
  });

  plots.RegisterPlot("simulated_neteq_preferred_buffer_size", [&](Plot* plot) {
    if (!neteq_stats) {
      neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
    }
    analyzer.CreateNetEqNetworkStatsGraph(
        *neteq_stats,
        [](const webrtc::NetEqNetworkStatistics& stats) {
          return stats.preferred_buffer_size_ms;
        },
        "Preferred buffer size (ms)", plot);
  });

  if (absl::c_find(plot_flags, "all") != plot_flags.end()) {
    plots.EnableAllPlots();
    // Treated separately since it isn't registered like the other plots.
    plot_flags.push_back("simulated_neteq_jitter_buffer_delay");
  } else {
    bool success = plots.EnablePlotsByFlags(plot_flags, flag_aliases);
    if (!success) {
      return 1;
    }
  }

  if (argc != 2 || FLAG_help) {
    // Print usage information.
    std::cerr << usage;
    if (FLAG_help) {
      rtc::FlagList::Print(nullptr, false);
      std::cerr << "List of registered plots (for use with the --plot flag):"
                << std::endl;
      for (const auto& plot : plots) {
        // TODO(terelius): Also print a help text.
        std::cerr << "  " << plot.label << std::endl;
      }
      // The following flag doesn't fit the model used for the other plots.
      std::cerr << "simulated_neteq_jitter_buffer_delay" << std::endl;
      std::cerr << "List of plot aliases (for use with the --plot flag):"
                << std::endl;
      std::cerr << "  all = every registered plot" << std::endl;
      for (const auto& alias : flag_aliases) {
        std::cerr << "  " << alias.first << " = ";
        for (const auto& replacement : alias.second) {
          std::cerr << replacement << ",";
        }
        std::cerr << std::endl;
      }
    }
    return 0;
  }

  for (const auto& plot : plots) {
    if (plot.enabled) {
      Plot* output = collection->AppendNewPlot();
      plot.plot_func(output);
      output->SetId(plot.label);
    }
  }

  // The model we use for registering plots assumes that the each plot label
  // can be mapped to a lambda that will produce exactly one plot. The
  // simulated_neteq_jitter_buffer_delay plot doesn't fit this model since it
  // creates multiple plots, and would need some state kept between the lambda
  // calls.
  if (absl::c_find(plot_flags, "simulated_neteq_jitter_buffer_delay") !=
      plot_flags.end()) {
    if (!neteq_stats) {
      neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
    }
    for (webrtc::EventLogAnalyzer::NetEqStatsGetterMap::const_iterator it =
             neteq_stats->cbegin();
         it != neteq_stats->cend(); ++it) {
      analyzer.CreateAudioJitterBufferGraph(it->first, it->second.get(),
                                            collection->AppendNewPlot());
    }
  }

  collection->Draw();

  if (FLAG_print_triage_alerts) {
    analyzer.CreateTriageNotifications();
    analyzer.PrintNotifications(stderr);
  }

  return 0;
}