webrtc/tools/event_log_visualizer/analyzer.cc

/*
 *  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 "webrtc/tools/event_log_visualizer/analyzer.h"

#include <algorithm>
#include <limits>
#include <map>
#include <sstream>
#include <string>
#include <utility>

#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/rate_statistics.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/audio_send_stream.h"
#include "webrtc/call/call.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"

namespace webrtc {
namespace plotting {

namespace {

class PacketFeedbackComparator {
 public:
  inline bool operator()(const webrtc::PacketFeedback& lhs,
                         const webrtc::PacketFeedback& rhs) {
    if (lhs.arrival_time_ms != rhs.arrival_time_ms)
      return lhs.arrival_time_ms < rhs.arrival_time_ms;
    if (lhs.send_time_ms != rhs.send_time_ms)
      return lhs.send_time_ms < rhs.send_time_ms;
    return lhs.sequence_number < rhs.sequence_number;
  }
};

void SortPacketFeedbackVector(std::vector<PacketFeedback>* vec) {
  auto pred = [](const PacketFeedback& packet_feedback) {
    return packet_feedback.arrival_time_ms == PacketFeedback::kNotReceived;
  };
  vec->erase(std::remove_if(vec->begin(), vec->end(), pred), vec->end());
  std::sort(vec->begin(), vec->end(), PacketFeedbackComparator());
}

std::string SsrcToString(uint32_t ssrc) {
  std::stringstream ss;
  ss << "SSRC " << ssrc;
  return ss.str();
}

// Checks whether an SSRC is contained in the list of desired SSRCs.
// Note that an empty SSRC list matches every SSRC.
bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
  if (desired_ssrc.size() == 0)
    return true;
  return std::find(desired_ssrc.begin(), desired_ssrc.end(), ssrc) !=
         desired_ssrc.end();
}

double AbsSendTimeToMicroseconds(int64_t abs_send_time) {
  // The timestamp is a fixed point representation with 6 bits for seconds
  // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the
  // time in seconds and then multiply by 1000000 to convert to microseconds.
  static constexpr double kTimestampToMicroSec =
      1000000.0 / static_cast<double>(1ul << 18);
  return abs_send_time * kTimestampToMicroSec;
}

// Computes the difference |later| - |earlier| where |later| and |earlier|
// are counters that wrap at |modulus|. The difference is chosen to have the
// least absolute value. For example if |modulus| is 8, then the difference will
// be chosen in the range [-3, 4]. If |modulus| is 9, then the difference will
// be in [-4, 4].
int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
  RTC_DCHECK_LE(1, modulus);
  RTC_DCHECK_LT(later, modulus);
  RTC_DCHECK_LT(earlier, modulus);
  int64_t difference =
      static_cast<int64_t>(later) - static_cast<int64_t>(earlier);
  int64_t max_difference = modulus / 2;
  int64_t min_difference = max_difference - modulus + 1;
  if (difference > max_difference) {
    difference -= modulus;
  }
  if (difference < min_difference) {
    difference += modulus;
  }
  if (difference > max_difference / 2 || difference < min_difference / 2) {
    LOG(LS_WARNING) << "Difference between" << later << " and " << earlier
                    << " expected to be in the range (" << min_difference / 2
                    << "," << max_difference / 2 << ") but is " << difference
                    << ". Correct unwrapping is uncertain.";
  }
  return difference;
}

// Return default values for header extensions, to use on streams without stored
// mapping data. Currently this only applies to audio streams, since the mapping
// is not stored in the event log.
// TODO(ivoc): Remove this once this mapping is stored in the event log for
//             audio streams. Tracking bug: webrtc:6399
webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
  webrtc::RtpHeaderExtensionMap default_map;
  default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
  default_map.Register<AbsoluteSendTime>(
      webrtc::RtpExtension::kAbsSendTimeDefaultId);
  return default_map;
}

constexpr float kLeftMargin = 0.01f;
constexpr float kRightMargin = 0.02f;
constexpr float kBottomMargin = 0.02f;
constexpr float kTopMargin = 0.05f;

class PacketSizeBytes {
 public:
  using DataType = LoggedRtpPacket;
  using ResultType = size_t;
  size_t operator()(const LoggedRtpPacket& packet) {
    return packet.total_length;
  }
};

class SequenceNumberDiff {
 public:
  using DataType = LoggedRtpPacket;
  using ResultType = int64_t;
  int64_t operator()(const LoggedRtpPacket& old_packet,
                     const LoggedRtpPacket& new_packet) {
    return WrappingDifference(new_packet.header.sequenceNumber,
                              old_packet.header.sequenceNumber, 1ul << 16);
  }
};

class NetworkDelayDiff {
 public:
  class AbsSendTime {
   public:
    using DataType = LoggedRtpPacket;
    using ResultType = double;
    double operator()(const LoggedRtpPacket& old_packet,
                      const LoggedRtpPacket& new_packet) {
      if (old_packet.header.extension.hasAbsoluteSendTime &&
          new_packet.header.extension.hasAbsoluteSendTime) {
        int64_t send_time_diff = WrappingDifference(
            new_packet.header.extension.absoluteSendTime,
            old_packet.header.extension.absoluteSendTime, 1ul << 24);
        int64_t recv_time_diff = new_packet.timestamp - old_packet.timestamp;
        return static_cast<double>(recv_time_diff -
                                   AbsSendTimeToMicroseconds(send_time_diff)) /
               1000;
      } else {
        return 0;
      }
    }
  };

  class CaptureTime {
   public:
    using DataType = LoggedRtpPacket;
    using ResultType = double;
    double operator()(const LoggedRtpPacket& old_packet,
                      const LoggedRtpPacket& new_packet) {
      int64_t send_time_diff = WrappingDifference(
          new_packet.header.timestamp, old_packet.header.timestamp, 1ull << 32);
      int64_t recv_time_diff = new_packet.timestamp - old_packet.timestamp;

      const double kVideoSampleRate = 90000;
      // TODO(terelius): We treat all streams as video for now, even though
      // audio might be sampled at e.g. 16kHz, because it is really difficult to
      // figure out the true sampling rate of a stream. The effect is that the
      // delay will be scaled incorrectly for non-video streams.

      double delay_change =
          static_cast<double>(recv_time_diff) / 1000 -
          static_cast<double>(send_time_diff) / kVideoSampleRate * 1000;
      if (delay_change < -10000 || 10000 < delay_change) {
        LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
        LOG(LS_WARNING) << "Old capture time " << old_packet.header.timestamp
                        << ", received time " << old_packet.timestamp;
        LOG(LS_WARNING) << "New capture time " << new_packet.header.timestamp
                        << ", received time " << new_packet.timestamp;
        LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
                        << static_cast<double>(recv_time_diff) / 1000000 << "s";
        LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
                        << static_cast<double>(send_time_diff) /
                               kVideoSampleRate
                        << "s";
      }
      return delay_change;
    }
  };
};

template <typename Extractor>
class Accumulated {
 public:
  using DataType = typename Extractor::DataType;
  using ResultType = typename Extractor::ResultType;
  ResultType operator()(const DataType& old_packet,
                        const DataType& new_packet) {
    sum += extract(old_packet, new_packet);
    return sum;
  }

 private:
  Extractor extract;
  ResultType sum = 0;
};

// For each element in data, use |Extractor| to extract a y-coordinate and
// store the result in a TimeSeries.
template <typename Extractor>
void Pointwise(const std::vector<typename Extractor::DataType>& data,
               uint64_t begin_time,
               TimeSeries* result) {
  Extractor extract;
  for (size_t i = 0; i < data.size(); i++) {
    float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
    float y = extract(data[i]);
    result->points.emplace_back(x, y);
  }
}

// For each pair of adjacent elements in |data|, use |Extractor| to extract a
// y-coordinate and store the result in a TimeSeries. Note that the x-coordinate
// will be the time of the second element in the pair.
template <typename Extractor>
void Pairwise(const std::vector<typename Extractor::DataType>& data,
              uint64_t begin_time,
              TimeSeries* result) {
  Extractor extract;
  for (size_t i = 1; i < data.size(); i++) {
    float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
    float y = extract(data[i - 1], data[i]);
    result->points.emplace_back(x, y);
  }
}

// Calculates a moving average of |data| and stores the result in a TimeSeries.
// A data point is generated every |step| microseconds from |begin_time|
// to |end_time|. The value of each data point is the average of the data
// during the preceeding |window_duration_us| microseconds.
template <typename Extractor>
void MovingAverage(const std::vector<typename Extractor::DataType>& data,
                   uint64_t begin_time,
                   uint64_t end_time,
                   uint64_t window_duration_us,
                   uint64_t step,
                   float y_scaling,
                   webrtc::plotting::TimeSeries* result) {
  size_t window_index_begin = 0;
  size_t window_index_end = 0;
  typename Extractor::ResultType sum_in_window = 0;
  Extractor extract;

  for (uint64_t t = begin_time; t < end_time + step; t += step) {
    while (window_index_end < data.size() &&
           data[window_index_end].timestamp < t) {
      sum_in_window += extract(data[window_index_end]);
      ++window_index_end;
    }
    while (window_index_begin < data.size() &&
           data[window_index_begin].timestamp < t - window_duration_us) {
      sum_in_window -= extract(data[window_index_begin]);
      ++window_index_begin;
    }
    float window_duration_s = static_cast<float>(window_duration_us) / 1000000;
    float x = static_cast<float>(t - begin_time) / 1000000;
    float y = sum_in_window / window_duration_s * y_scaling;
    result->points.emplace_back(x, y);
  }
}

}  // namespace

EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
    : parsed_log_(log), window_duration_(250000), step_(10000) {
  uint64_t first_timestamp = std::numeric_limits<uint64_t>::max();
  uint64_t last_timestamp = std::numeric_limits<uint64_t>::min();

  // Maps a stream identifier consisting of ssrc and direction
  // to the header extensions used by that stream,
  std::map<StreamId, RtpHeaderExtensionMap> extension_maps;

  PacketDirection direction;
  uint8_t header[IP_PACKET_SIZE];
  size_t header_length;
  size_t total_length;

  // Make a default extension map for streams without configuration information.
  // TODO(ivoc): Once configuration of audio streams is stored in the event log,
  //             this can be removed. Tracking bug: webrtc:6399
  RtpHeaderExtensionMap default_extension_map = GetDefaultHeaderExtensionMap();

  for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
    ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
    if (event_type != ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT &&
        event_type != ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT &&
        event_type != ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT &&
        event_type != ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT &&
        event_type != ParsedRtcEventLog::LOG_START &&
        event_type != ParsedRtcEventLog::LOG_END) {
      uint64_t timestamp = parsed_log_.GetTimestamp(i);
      first_timestamp = std::min(first_timestamp, timestamp);
      last_timestamp = std::max(last_timestamp, timestamp);
    }

    switch (parsed_log_.GetEventType(i)) {
      case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
        VideoReceiveStream::Config config(nullptr);
        parsed_log_.GetVideoReceiveConfig(i, &config);
        StreamId stream(config.rtp.remote_ssrc, kIncomingPacket);
        extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
        video_ssrcs_.insert(stream);
        StreamId rtx_stream(config.rtp.rtx_ssrc, kIncomingPacket);
        extension_maps[rtx_stream] =
            RtpHeaderExtensionMap(config.rtp.extensions);
        video_ssrcs_.insert(rtx_stream);
        rtx_ssrcs_.insert(rtx_stream);
        break;
      }
      case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
        VideoSendStream::Config config(nullptr);
        parsed_log_.GetVideoSendConfig(i, &config);
        for (auto ssrc : config.rtp.ssrcs) {
          StreamId stream(ssrc, kOutgoingPacket);
          extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
          video_ssrcs_.insert(stream);
        }
        for (auto ssrc : config.rtp.rtx.ssrcs) {
          StreamId rtx_stream(ssrc, kOutgoingPacket);
          extension_maps[rtx_stream] =
              RtpHeaderExtensionMap(config.rtp.extensions);
          video_ssrcs_.insert(rtx_stream);
          rtx_ssrcs_.insert(rtx_stream);
        }
        break;
      }
      case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
        AudioReceiveStream::Config config;
        parsed_log_.GetAudioReceiveConfig(i, &config);
        StreamId stream(config.rtp.remote_ssrc, kIncomingPacket);
        extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
        audio_ssrcs_.insert(stream);
        break;
      }
      case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
        AudioSendStream::Config config(nullptr);
        parsed_log_.GetAudioSendConfig(i, &config);
        StreamId stream(config.rtp.ssrc, kOutgoingPacket);
        extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
        audio_ssrcs_.insert(stream);
        break;
      }
      case ParsedRtcEventLog::RTP_EVENT: {
        MediaType media_type;
        parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
                                 &header_length, &total_length);
        // Parse header to get SSRC.
        RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
        RTPHeader parsed_header;
        rtp_parser.Parse(&parsed_header);
        StreamId stream(parsed_header.ssrc, direction);
        // Look up the extension_map and parse it again to get the extensions.
        if (extension_maps.count(stream) == 1) {
          RtpHeaderExtensionMap* extension_map = &extension_maps[stream];
          rtp_parser.Parse(&parsed_header, extension_map);
        } else {
          // Use the default extension map.
          // TODO(ivoc): Once configuration of audio streams is stored in the
          //             event log, this can be removed.
          //             Tracking bug: webrtc:6399
          rtp_parser.Parse(&parsed_header, &default_extension_map);
        }
        uint64_t timestamp = parsed_log_.GetTimestamp(i);
        rtp_packets_[stream].push_back(
            LoggedRtpPacket(timestamp, parsed_header, total_length));
        break;
      }
      case ParsedRtcEventLog::RTCP_EVENT: {
        uint8_t packet[IP_PACKET_SIZE];
        MediaType media_type;
        parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet,
                                  &total_length);

        // Currently feedback is logged twice, both for audio and video.
        // Only act on one of them.
        if (media_type == MediaType::AUDIO || media_type == MediaType::ANY) {
          rtcp::CommonHeader header;
          const uint8_t* packet_end = packet + total_length;
          for (const uint8_t* block = packet; block < packet_end;
               block = header.NextPacket()) {
            RTC_CHECK(header.Parse(block, packet_end - block));
            if (header.type() == rtcp::TransportFeedback::kPacketType &&
                header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
              std::unique_ptr<rtcp::TransportFeedback> rtcp_packet(
                  new rtcp::TransportFeedback());
              if (rtcp_packet->Parse(header)) {
                uint32_t ssrc = rtcp_packet->sender_ssrc();
                StreamId stream(ssrc, direction);
                uint64_t timestamp = parsed_log_.GetTimestamp(i);
                rtcp_packets_[stream].push_back(LoggedRtcpPacket(
                    timestamp, kRtcpTransportFeedback, std::move(rtcp_packet)));
              }
            } else if (header.type() == rtcp::SenderReport::kPacketType) {
              std::unique_ptr<rtcp::SenderReport> rtcp_packet(
                  new rtcp::SenderReport());
              if (rtcp_packet->Parse(header)) {
                uint32_t ssrc = rtcp_packet->sender_ssrc();
                StreamId stream(ssrc, direction);
                uint64_t timestamp = parsed_log_.GetTimestamp(i);
                rtcp_packets_[stream].push_back(LoggedRtcpPacket(
                    timestamp, kRtcpSr, std::move(rtcp_packet)));
              }
            } else if (header.type() == rtcp::ReceiverReport::kPacketType) {
              std::unique_ptr<rtcp::ReceiverReport> rtcp_packet(
                  new rtcp::ReceiverReport());
              if (rtcp_packet->Parse(header)) {
                uint32_t ssrc = rtcp_packet->sender_ssrc();
                StreamId stream(ssrc, direction);
                uint64_t timestamp = parsed_log_.GetTimestamp(i);
                rtcp_packets_[stream].push_back(LoggedRtcpPacket(
                    timestamp, kRtcpRr, std::move(rtcp_packet)));
              }
            }
          }
        }
        break;
      }
      case ParsedRtcEventLog::LOG_START: {
        break;
      }
      case ParsedRtcEventLog::LOG_END: {
        break;
      }
      case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
        break;
      }
      case ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
        LossBasedBweUpdate bwe_update;
        bwe_update.timestamp = parsed_log_.GetTimestamp(i);
        parsed_log_.GetLossBasedBweUpdate(i, &bwe_update.new_bitrate,
                                          &bwe_update.fraction_loss,
                                          &bwe_update.expected_packets);
        bwe_loss_updates_.push_back(bwe_update);
        break;
      }
      case ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
        break;
      }
      case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
        AudioNetworkAdaptationEvent ana_event;
        ana_event.timestamp = parsed_log_.GetTimestamp(i);
        parsed_log_.GetAudioNetworkAdaptation(i, &ana_event.config);
        audio_network_adaptation_events_.push_back(ana_event);
        break;
      }
      case ParsedRtcEventLog::BWE_PROBE_CLUSTER_CREATED_EVENT: {
        break;
      }
      case ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT: {
        break;
      }
      case ParsedRtcEventLog::UNKNOWN_EVENT: {
        break;
      }
    }
  }

  if (last_timestamp < first_timestamp) {
    // No useful events in the log.
    first_timestamp = last_timestamp = 0;
  }
  begin_time_ = first_timestamp;
  end_time_ = last_timestamp;
  call_duration_s_ = static_cast<float>(end_time_ - begin_time_) / 1000000;
}

class BitrateObserver : public CongestionController::Observer,
                        public RemoteBitrateObserver {
 public:
  BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}

  // TODO(minyue): remove this when old OnNetworkChanged is deprecated. See
  // https://bugs.chromium.org/p/webrtc/issues/detail?id=6796
  using CongestionController::Observer::OnNetworkChanged;

  void OnNetworkChanged(uint32_t bitrate_bps,
                        uint8_t fraction_loss,
                        int64_t rtt_ms,
                        int64_t probing_interval_ms) override {
    last_bitrate_bps_ = bitrate_bps;
    bitrate_updated_ = true;
  }

  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
                               uint32_t bitrate) override {}

  uint32_t last_bitrate_bps() const { return last_bitrate_bps_; }
  bool GetAndResetBitrateUpdated() {
    bool bitrate_updated = bitrate_updated_;
    bitrate_updated_ = false;
    return bitrate_updated;
  }

 private:
  uint32_t last_bitrate_bps_;
  bool bitrate_updated_;
};

bool EventLogAnalyzer::IsRtxSsrc(StreamId stream_id) const {
  return rtx_ssrcs_.count(stream_id) == 1;
}

bool EventLogAnalyzer::IsVideoSsrc(StreamId stream_id) const {
  return video_ssrcs_.count(stream_id) == 1;
}

bool EventLogAnalyzer::IsAudioSsrc(StreamId stream_id) const {
  return audio_ssrcs_.count(stream_id) == 1;
}

std::string EventLogAnalyzer::GetStreamName(StreamId stream_id) const {
  std::stringstream name;
  if (IsAudioSsrc(stream_id)) {
    name << "Audio ";
  } else if (IsVideoSsrc(stream_id)) {
    name << "Video ";
  } else {
    name << "Unknown ";
  }
  if (IsRtxSsrc(stream_id))
    name << "RTX ";
  if (stream_id.GetDirection() == kIncomingPacket) {
    name << "(In) ";
  } else {
    name << "(Out) ";
  }
  name << SsrcToString(stream_id.GetSsrc());
  return name.str();
}

void EventLogAnalyzer::FillAudioEncoderTimeSeries(
    Plot* plot,
    rtc::FunctionView<rtc::Optional<float>(
        const AudioNetworkAdaptationEvent& ana_event)> get_y) const {
  plot->series_list_.push_back(TimeSeries());
  plot->series_list_.back().style = LINE_DOT_GRAPH;
  for (auto& ana_event : audio_network_adaptation_events_) {
    rtc::Optional<float> y = get_y(ana_event);
    if (y) {
      float x = static_cast<float>(ana_event.timestamp - begin_time_) / 1000000;
      plot->series_list_.back().points.emplace_back(x, *y);
    }
  }
}

void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction,
                                         Plot* plot) {
  for (auto& kv : rtp_packets_) {
    StreamId stream_id = kv.first;
    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
    // Filter on direction and SSRC.
    if (stream_id.GetDirection() != desired_direction ||
        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
      continue;
    }

    TimeSeries time_series;
    time_series.label = GetStreamName(stream_id);
    time_series.style = BAR_GRAPH;
    Pointwise<PacketSizeBytes>(packet_stream, begin_time_, &time_series);
    plot->series_list_.push_back(std::move(time_series));
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
                          kTopMargin);
  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
    plot->SetTitle("Incoming RTP packets");
  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
    plot->SetTitle("Outgoing RTP packets");
  }
}

template <typename T>
void EventLogAnalyzer::CreateAccumulatedPacketsTimeSeries(
    PacketDirection desired_direction,
    Plot* plot,
    const std::map<StreamId, std::vector<T>>& packets,
    const std::string& label_prefix) {
  for (auto& kv : packets) {
    StreamId stream_id = kv.first;
    const std::vector<T>& packet_stream = kv.second;
    // Filter on direction and SSRC.
    if (stream_id.GetDirection() != desired_direction ||
        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
      continue;
    }

    TimeSeries time_series;
    time_series.label = label_prefix + " " + GetStreamName(stream_id);
    time_series.style = LINE_STEP_GRAPH;

    for (size_t i = 0; i < packet_stream.size(); i++) {
      float x = static_cast<float>(packet_stream[i].timestamp - begin_time_) /
                1000000;
      time_series.points.emplace_back(x, i + 1);
    }

    plot->series_list_.push_back(std::move(time_series));
  }
}

void EventLogAnalyzer::CreateAccumulatedPacketsGraph(
    PacketDirection desired_direction,
    Plot* plot) {
  CreateAccumulatedPacketsTimeSeries(desired_direction, plot, rtp_packets_,
                                     "RTP");
  CreateAccumulatedPacketsTimeSeries(desired_direction, plot, rtcp_packets_,
                                     "RTCP");

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
    plot->SetTitle("Accumulated Incoming RTP/RTCP packets");
  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
    plot->SetTitle("Accumulated Outgoing RTP/RTCP packets");
  }
}

// For each SSRC, plot the time between the consecutive playouts.
void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) {
  std::map<uint32_t, TimeSeries> time_series;
  std::map<uint32_t, uint64_t> last_playout;

  uint32_t ssrc;

  for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
    ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
    if (event_type == ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT) {
      parsed_log_.GetAudioPlayout(i, &ssrc);
      uint64_t timestamp = parsed_log_.GetTimestamp(i);
      if (MatchingSsrc(ssrc, desired_ssrc_)) {
        float x = static_cast<float>(timestamp - begin_time_) / 1000000;
        float y = static_cast<float>(timestamp - last_playout[ssrc]) / 1000;
        if (time_series[ssrc].points.size() == 0) {
          // There were no previusly logged playout for this SSRC.
          // Generate a point, but place it on the x-axis.
          y = 0;
        }
        time_series[ssrc].points.push_back(TimeSeriesPoint(x, y));
        last_playout[ssrc] = timestamp;
      }
    }
  }

  // Set labels and put in graph.
  for (auto& kv : time_series) {
    kv.second.label = SsrcToString(kv.first);
    kv.second.style = BAR_GRAPH;
    plot->series_list_.push_back(std::move(kv.second));
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
                          kTopMargin);
  plot->SetTitle("Audio playout");
}

// For audio SSRCs, plot the audio level.
void EventLogAnalyzer::CreateAudioLevelGraph(Plot* plot) {
  std::map<StreamId, TimeSeries> time_series;

  for (auto& kv : rtp_packets_) {
    StreamId stream_id = kv.first;
    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
    // TODO(ivoc): When audio send/receive configs are stored in the event
    //             log, a check should be added here to only process audio
    //             streams. Tracking bug: webrtc:6399
    for (auto& packet : packet_stream) {
      if (packet.header.extension.hasAudioLevel) {
        float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
        // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
        // Here we convert it to dBov.
        float y = static_cast<float>(-packet.header.extension.audioLevel);
        time_series[stream_id].points.emplace_back(TimeSeriesPoint(x, y));
      }
    }
  }

  for (auto& series : time_series) {
    series.second.label = GetStreamName(series.first);
    series.second.style = LINE_GRAPH;
    plot->series_list_.push_back(std::move(series.second));
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin,
                 kTopMargin);
  plot->SetTitle("Audio level");
}

// For each SSRC, plot the time between the consecutive playouts.
void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) {
  for (auto& kv : rtp_packets_) {
    StreamId stream_id = kv.first;
    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
    // Filter on direction and SSRC.
    if (stream_id.GetDirection() != kIncomingPacket ||
        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
      continue;
    }

    TimeSeries time_series;
    time_series.label = GetStreamName(stream_id);
    time_series.style = BAR_GRAPH;
    Pairwise<SequenceNumberDiff>(packet_stream, begin_time_, &time_series);
    plot->series_list_.push_back(std::move(time_series));
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
                          kTopMargin);
  plot->SetTitle("Sequence number");
}

void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) {
  for (auto& kv : rtp_packets_) {
    StreamId stream_id = kv.first;
    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
    // Filter on direction and SSRC.
    if (stream_id.GetDirection() != kIncomingPacket ||
        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
        packet_stream.size() == 0) {
      continue;
    }

    TimeSeries time_series;
    time_series.label = GetStreamName(stream_id);
    time_series.style = LINE_DOT_GRAPH;
    const uint64_t kWindowUs = 1000000;
    const uint64_t kStep = 1000000;
    SequenceNumberUnwrapper unwrapper_;
    SequenceNumberUnwrapper prior_unwrapper_;
    size_t window_index_begin = 0;
    size_t window_index_end = 0;
    int64_t highest_seq_number =
        unwrapper_.Unwrap(packet_stream[0].header.sequenceNumber) - 1;
    int64_t highest_prior_seq_number =
        prior_unwrapper_.Unwrap(packet_stream[0].header.sequenceNumber) - 1;

    for (uint64_t t = begin_time_; t < end_time_ + kStep; t += kStep) {
      while (window_index_end < packet_stream.size() &&
             packet_stream[window_index_end].timestamp < t) {
        int64_t sequence_number = unwrapper_.Unwrap(
            packet_stream[window_index_end].header.sequenceNumber);
        highest_seq_number = std::max(highest_seq_number, sequence_number);
        ++window_index_end;
      }
      while (window_index_begin < packet_stream.size() &&
             packet_stream[window_index_begin].timestamp < t - kWindowUs) {
        int64_t sequence_number = prior_unwrapper_.Unwrap(
            packet_stream[window_index_begin].header.sequenceNumber);
        highest_prior_seq_number =
            std::max(highest_prior_seq_number, sequence_number);
        ++window_index_begin;
      }
      float x = static_cast<float>(t - begin_time_) / 1000000;
      int64_t expected_packets = highest_seq_number - highest_prior_seq_number;
      if (expected_packets > 0) {
        int64_t received_packets = window_index_end - window_index_begin;
        int64_t lost_packets = expected_packets - received_packets;
        float y = static_cast<float>(lost_packets) / expected_packets * 100;
        time_series.points.emplace_back(x, y);
      }
    }
    plot->series_list_.push_back(std::move(time_series));
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Estimated loss rate (%)", kBottomMargin,
                          kTopMargin);
  plot->SetTitle("Estimated incoming loss rate");
}

void EventLogAnalyzer::CreateDelayChangeGraph(Plot* plot) {
  for (auto& kv : rtp_packets_) {
    StreamId stream_id = kv.first;
    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
    // Filter on direction and SSRC.
    if (stream_id.GetDirection() != kIncomingPacket ||
        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
        IsAudioSsrc(stream_id) || !IsVideoSsrc(stream_id) ||
        IsRtxSsrc(stream_id)) {
      continue;
    }

    TimeSeries capture_time_data;
    capture_time_data.label = GetStreamName(stream_id) + " capture-time";
    capture_time_data.style = BAR_GRAPH;
    Pairwise<NetworkDelayDiff::CaptureTime>(packet_stream, begin_time_,
                                            &capture_time_data);
    plot->series_list_.push_back(std::move(capture_time_data));

    TimeSeries send_time_data;
    send_time_data.label = GetStreamName(stream_id) + " abs-send-time";
    send_time_data.style = BAR_GRAPH;
    Pairwise<NetworkDelayDiff::AbsSendTime>(packet_stream, begin_time_,
                                            &send_time_data);
    plot->series_list_.push_back(std::move(send_time_data));
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
                          kTopMargin);
  plot->SetTitle("Network latency change between consecutive packets");
}

void EventLogAnalyzer::CreateAccumulatedDelayChangeGraph(Plot* plot) {
  for (auto& kv : rtp_packets_) {
    StreamId stream_id = kv.first;
    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
    // Filter on direction and SSRC.
    if (stream_id.GetDirection() != kIncomingPacket ||
        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
        IsAudioSsrc(stream_id) || !IsVideoSsrc(stream_id) ||
        IsRtxSsrc(stream_id)) {
      continue;
    }

    TimeSeries capture_time_data;
    capture_time_data.label = GetStreamName(stream_id) + " capture-time";
    capture_time_data.style = LINE_GRAPH;
    Pairwise<Accumulated<NetworkDelayDiff::CaptureTime>>(
        packet_stream, begin_time_, &capture_time_data);
    plot->series_list_.push_back(std::move(capture_time_data));

    TimeSeries send_time_data;
    send_time_data.label = GetStreamName(stream_id) + " abs-send-time";
    send_time_data.style = LINE_GRAPH;
    Pairwise<Accumulated<NetworkDelayDiff::AbsSendTime>>(
        packet_stream, begin_time_, &send_time_data);
    plot->series_list_.push_back(std::move(send_time_data));
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
                          kTopMargin);
  plot->SetTitle("Accumulated network latency change");
}

// Plot the fraction of packets lost (as perceived by the loss-based BWE).
void EventLogAnalyzer::CreateFractionLossGraph(Plot* plot) {
  plot->series_list_.push_back(TimeSeries());
  for (auto& bwe_update : bwe_loss_updates_) {
    float x = static_cast<float>(bwe_update.timestamp - begin_time_) / 1000000;
    float y = static_cast<float>(bwe_update.fraction_loss) / 255 * 100;
    plot->series_list_.back().points.emplace_back(x, y);
  }
  plot->series_list_.back().label = "Fraction lost";
  plot->series_list_.back().style = LINE_DOT_GRAPH;

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
                          kTopMargin);
  plot->SetTitle("Reported packet loss");
}

// Plot the total bandwidth used by all RTP streams.
void EventLogAnalyzer::CreateTotalBitrateGraph(
    PacketDirection desired_direction,
    Plot* plot) {
  struct TimestampSize {
    TimestampSize(uint64_t t, size_t s) : timestamp(t), size(s) {}
    uint64_t timestamp;
    size_t size;
  };
  std::vector<TimestampSize> packets;

  PacketDirection direction;
  size_t total_length;

  // Extract timestamps and sizes for the relevant packets.
  for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
    ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
    if (event_type == ParsedRtcEventLog::RTP_EVENT) {
      parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, nullptr,
                               &total_length);
      if (direction == desired_direction) {
        uint64_t timestamp = parsed_log_.GetTimestamp(i);
        packets.push_back(TimestampSize(timestamp, total_length));
      }
    }
  }

  size_t window_index_begin = 0;
  size_t window_index_end = 0;
  size_t bytes_in_window = 0;

  // Calculate a moving average of the bitrate and store in a TimeSeries.
  plot->series_list_.push_back(TimeSeries());
  for (uint64_t time = begin_time_; time < end_time_ + step_; time += step_) {
    while (window_index_end < packets.size() &&
           packets[window_index_end].timestamp < time) {
      bytes_in_window += packets[window_index_end].size;
      ++window_index_end;
    }
    while (window_index_begin < packets.size() &&
           packets[window_index_begin].timestamp < time - window_duration_) {
      RTC_DCHECK_LE(packets[window_index_begin].size, bytes_in_window);
      bytes_in_window -= packets[window_index_begin].size;
      ++window_index_begin;
    }
    float window_duration_in_seconds =
        static_cast<float>(window_duration_) / 1000000;
    float x = static_cast<float>(time - begin_time_) / 1000000;
    float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
    plot->series_list_.back().points.push_back(TimeSeriesPoint(x, y));
  }

  // Set labels.
  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
    plot->series_list_.back().label = "Incoming bitrate";
  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
    plot->series_list_.back().label = "Outgoing bitrate";
  }
  plot->series_list_.back().style = LINE_GRAPH;

  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
  if (desired_direction == kOutgoingPacket) {
    plot->series_list_.push_back(TimeSeries());
    for (auto& bwe_update : bwe_loss_updates_) {
      float x =
          static_cast<float>(bwe_update.timestamp - begin_time_) / 1000000;
      float y = static_cast<float>(bwe_update.new_bitrate) / 1000;
      plot->series_list_.back().points.emplace_back(x, y);
    }
    plot->series_list_.back().label = "Loss-based estimate";
    plot->series_list_.back().style = LINE_STEP_GRAPH;
  }
  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
    plot->SetTitle("Incoming RTP bitrate");
  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
    plot->SetTitle("Outgoing RTP bitrate");
  }
}

// For each SSRC, plot the bandwidth used by that stream.
void EventLogAnalyzer::CreateStreamBitrateGraph(
    PacketDirection desired_direction,
    Plot* plot) {
  for (auto& kv : rtp_packets_) {
    StreamId stream_id = kv.first;
    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
    // Filter on direction and SSRC.
    if (stream_id.GetDirection() != desired_direction ||
        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
      continue;
    }

    TimeSeries time_series;
    time_series.label = GetStreamName(stream_id);
    time_series.style = LINE_GRAPH;
    double bytes_to_kilobits = 8.0 / 1000;
    MovingAverage<PacketSizeBytes>(packet_stream, begin_time_, end_time_,
                                   window_duration_, step_, bytes_to_kilobits,
                                   &time_series);
    plot->series_list_.push_back(std::move(time_series));
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
    plot->SetTitle("Incoming bitrate per stream");
  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
    plot->SetTitle("Outgoing bitrate per stream");
  }
}

void EventLogAnalyzer::CreateBweSimulationGraph(Plot* plot) {
  std::map<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
  std::map<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;

  for (const auto& kv : rtp_packets_) {
    if (kv.first.GetDirection() == PacketDirection::kOutgoingPacket) {
      for (const LoggedRtpPacket& rtp_packet : kv.second)
        outgoing_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
    }
  }

  for (const auto& kv : rtcp_packets_) {
    if (kv.first.GetDirection() == PacketDirection::kIncomingPacket) {
      for (const LoggedRtcpPacket& rtcp_packet : kv.second)
        incoming_rtcp.insert(
            std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
    }
  }

  SimulatedClock clock(0);
  BitrateObserver observer;
  RtcEventLogNullImpl null_event_log;
  PacketRouter packet_router;
  CongestionController cc(&clock, &observer, &observer, &null_event_log,
                          &packet_router);
  // TODO(holmer): Log the call config and use that here instead.
  static const uint32_t kDefaultStartBitrateBps = 300000;
  cc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);

  TimeSeries time_series;
  time_series.label = "Delay-based estimate";
  time_series.style = LINE_DOT_GRAPH;
  TimeSeries acked_time_series;
  acked_time_series.label = "Acked bitrate";
  acked_time_series.style = LINE_DOT_GRAPH;

  auto rtp_iterator = outgoing_rtp.begin();
  auto rtcp_iterator = incoming_rtcp.begin();

  auto NextRtpTime = [&]() {
    if (rtp_iterator != outgoing_rtp.end())
      return static_cast<int64_t>(rtp_iterator->first);
    return std::numeric_limits<int64_t>::max();
  };

  auto NextRtcpTime = [&]() {
    if (rtcp_iterator != incoming_rtcp.end())
      return static_cast<int64_t>(rtcp_iterator->first);
    return std::numeric_limits<int64_t>::max();
  };

  auto NextProcessTime = [&]() {
    if (rtcp_iterator != incoming_rtcp.end() ||
        rtp_iterator != outgoing_rtp.end()) {
      return clock.TimeInMicroseconds() +
             std::max<int64_t>(cc.TimeUntilNextProcess() * 1000, 0);
    }
    return std::numeric_limits<int64_t>::max();
  };

  RateStatistics acked_bitrate(250, 8000);

  int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
  int64_t last_update_us = 0;
  while (time_us != std::numeric_limits<int64_t>::max()) {
    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
      const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
      if (rtcp.type == kRtcpTransportFeedback) {
        cc.OnTransportFeedback(
            *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
        std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
        SortPacketFeedbackVector(&feedback);
        rtc::Optional<uint32_t> bitrate_bps;
        if (!feedback.empty()) {
          for (const PacketFeedback& packet : feedback)
            acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
          bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
        }
        uint32_t y = 0;
        if (bitrate_bps)
          y = *bitrate_bps / 1000;
        float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
                  1000000;
        acked_time_series.points.emplace_back(x, y);
      }
      ++rtcp_iterator;
    }
    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
      const LoggedRtpPacket& rtp = *rtp_iterator->second;
      if (rtp.header.extension.hasTransportSequenceNumber) {
        RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
        cc.AddPacket(rtp.header.extension.transportSequenceNumber,
                     rtp.total_length, PacedPacketInfo());
        rtc::SentPacket sent_packet(
            rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
        cc.OnSentPacket(sent_packet);
      }
      ++rtp_iterator;
    }
    if (clock.TimeInMicroseconds() >= NextProcessTime()) {
      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
      cc.Process();
    }
    if (observer.GetAndResetBitrateUpdated() ||
        time_us - last_update_us >= 1e6) {
      uint32_t y = observer.last_bitrate_bps() / 1000;
      float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
                1000000;
      time_series.points.emplace_back(x, y);
      last_update_us = time_us;
    }
    time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
  }
  // Add the data set to the plot.
  plot->series_list_.push_back(std::move(time_series));
  plot->series_list_.push_back(std::move(acked_time_series));

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
  plot->SetTitle("Simulated BWE behavior");
}

void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
  std::map<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
  std::map<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;

  for (const auto& kv : rtp_packets_) {
    if (kv.first.GetDirection() == PacketDirection::kOutgoingPacket) {
      for (const LoggedRtpPacket& rtp_packet : kv.second)
        outgoing_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
    }
  }

  for (const auto& kv : rtcp_packets_) {
    if (kv.first.GetDirection() == PacketDirection::kIncomingPacket) {
      for (const LoggedRtcpPacket& rtcp_packet : kv.second)
        incoming_rtcp.insert(
            std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
    }
  }

  SimulatedClock clock(0);
  TransportFeedbackAdapter feedback_adapter(&clock);

  TimeSeries time_series;
  time_series.label = "Network Delay Change";
  time_series.style = LINE_DOT_GRAPH;
  int64_t estimated_base_delay_ms = std::numeric_limits<int64_t>::max();

  auto rtp_iterator = outgoing_rtp.begin();
  auto rtcp_iterator = incoming_rtcp.begin();

  auto NextRtpTime = [&]() {
    if (rtp_iterator != outgoing_rtp.end())
      return static_cast<int64_t>(rtp_iterator->first);
    return std::numeric_limits<int64_t>::max();
  };

  auto NextRtcpTime = [&]() {
    if (rtcp_iterator != incoming_rtcp.end())
      return static_cast<int64_t>(rtcp_iterator->first);
    return std::numeric_limits<int64_t>::max();
  };

  int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
  while (time_us != std::numeric_limits<int64_t>::max()) {
    clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
    if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
      const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
      if (rtcp.type == kRtcpTransportFeedback) {
        feedback_adapter.OnTransportFeedback(
            *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
        std::vector<PacketFeedback> feedback =
            feedback_adapter.GetTransportFeedbackVector();
        SortPacketFeedbackVector(&feedback);
        for (const PacketFeedback& packet : feedback) {
          int64_t y = packet.arrival_time_ms - packet.send_time_ms;
          float x =
              static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
              1000000;
          estimated_base_delay_ms = std::min(y, estimated_base_delay_ms);
          time_series.points.emplace_back(x, y);
        }
      }
      ++rtcp_iterator;
    }
    if (clock.TimeInMicroseconds() >= NextRtpTime()) {
      RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
      const LoggedRtpPacket& rtp = *rtp_iterator->second;
      if (rtp.header.extension.hasTransportSequenceNumber) {
        RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
        feedback_adapter.AddPacket(rtp.header.extension.transportSequenceNumber,
                                   rtp.total_length, PacedPacketInfo());
        feedback_adapter.OnSentPacket(
            rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
      }
      ++rtp_iterator;
    }
    time_us = std::min(NextRtpTime(), NextRtcpTime());
  }
  // We assume that the base network delay (w/o queues) is the min delay
  // observed during the call.
  for (TimeSeriesPoint& point : time_series.points)
    point.y -= estimated_base_delay_ms;
  // Add the data set to the plot.
  plot->series_list_.push_back(std::move(time_series));

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
  plot->SetTitle("Network Delay Change.");
}

std::vector<std::pair<int64_t, int64_t>> EventLogAnalyzer::GetFrameTimestamps()
    const {
  std::vector<std::pair<int64_t, int64_t>> timestamps;
  size_t largest_stream_size = 0;
  const std::vector<LoggedRtpPacket>* largest_video_stream = nullptr;
  // Find the incoming video stream with the most number of packets that is
  // not rtx.
  for (const auto& kv : rtp_packets_) {
    if (kv.first.GetDirection() == kIncomingPacket &&
        video_ssrcs_.find(kv.first) != video_ssrcs_.end() &&
        rtx_ssrcs_.find(kv.first) == rtx_ssrcs_.end() &&
        kv.second.size() > largest_stream_size) {
      largest_stream_size = kv.second.size();
      largest_video_stream = &kv.second;
    }
  }
  if (largest_video_stream == nullptr) {
    for (auto& packet : *largest_video_stream) {
      if (packet.header.markerBit) {
        int64_t capture_ms = packet.header.timestamp / 90.0;
        int64_t arrival_ms = packet.timestamp / 1000.0;
        timestamps.push_back(std::make_pair(capture_ms, arrival_ms));
      }
    }
  }
  return timestamps;
}

void EventLogAnalyzer::CreateTimestampGraph(Plot* plot) {
  for (const auto& kv : rtp_packets_) {
    const std::vector<LoggedRtpPacket>& rtp_packets = kv.second;
    StreamId stream_id = kv.first;

    {
      TimeSeries timestamp_data;
      timestamp_data.label = GetStreamName(stream_id) + " capture-time";
      timestamp_data.style = LINE_DOT_GRAPH;
      for (LoggedRtpPacket packet : rtp_packets) {
        float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
        float y = packet.header.timestamp;
        timestamp_data.points.emplace_back(x, y);
      }
      plot->series_list_.push_back(std::move(timestamp_data));
    }

    {
      auto kv = rtcp_packets_.find(stream_id);
      if (kv != rtcp_packets_.end()) {
        const auto& packets = kv->second;
        TimeSeries timestamp_data;
        timestamp_data.label = GetStreamName(stream_id) + " rtcp capture-time";
        timestamp_data.style = LINE_DOT_GRAPH;
        for (const LoggedRtcpPacket& rtcp : packets) {
          if (rtcp.type != kRtcpSr)
            continue;
          rtcp::SenderReport* sr;
          sr = static_cast<rtcp::SenderReport*>(rtcp.packet.get());
          float x = static_cast<float>(rtcp.timestamp - begin_time_) / 1000000;
          float y = sr->rtp_timestamp();
          timestamp_data.points.emplace_back(x, y);
        }
        plot->series_list_.push_back(std::move(timestamp_data));
      }
    }
  }

  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Timestamp (90khz)", kBottomMargin, kTopMargin);
  plot->SetTitle("Timestamps");
}

void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
  FillAudioEncoderTimeSeries(
      plot, [](const AudioNetworkAdaptationEvent& ana_event) {
        if (ana_event.config.bitrate_bps)
          return rtc::Optional<float>(
              static_cast<float>(*ana_event.config.bitrate_bps));
        return rtc::Optional<float>();
      });
  plot->series_list_.back().label = "Audio encoder target bitrate";
  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Bitrate (bps)", kBottomMargin, kTopMargin);
  plot->SetTitle("Reported audio encoder target bitrate");
}

void EventLogAnalyzer::CreateAudioEncoderFrameLengthGraph(Plot* plot) {
  FillAudioEncoderTimeSeries(
      plot, [](const AudioNetworkAdaptationEvent& ana_event) {
        if (ana_event.config.frame_length_ms)
          return rtc::Optional<float>(
              static_cast<float>(*ana_event.config.frame_length_ms));
        return rtc::Optional<float>();
      });
  plot->series_list_.back().label = "Audio encoder frame length";
  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Frame length (ms)", kBottomMargin, kTopMargin);
  plot->SetTitle("Reported audio encoder frame length");
}

void EventLogAnalyzer::CreateAudioEncoderUplinkPacketLossFractionGraph(
    Plot* plot) {
  FillAudioEncoderTimeSeries(
      plot, [&](const AudioNetworkAdaptationEvent& ana_event) {
        if (ana_event.config.uplink_packet_loss_fraction)
          return rtc::Optional<float>(static_cast<float>(
              *ana_event.config.uplink_packet_loss_fraction));
        return rtc::Optional<float>();
      });
  plot->series_list_.back().label = "Audio encoder uplink packet loss fraction";
  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
                          kTopMargin);
  plot->SetTitle("Reported audio encoder lost packets");
}

void EventLogAnalyzer::CreateAudioEncoderEnableFecGraph(Plot* plot) {
  FillAudioEncoderTimeSeries(
      plot, [&](const AudioNetworkAdaptationEvent& ana_event) {
        if (ana_event.config.enable_fec)
          return rtc::Optional<float>(
              static_cast<float>(*ana_event.config.enable_fec));
        return rtc::Optional<float>();
      });
  plot->series_list_.back().label = "Audio encoder FEC";
  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "FEC (false/true)", kBottomMargin, kTopMargin);
  plot->SetTitle("Reported audio encoder FEC");
}

void EventLogAnalyzer::CreateAudioEncoderEnableDtxGraph(Plot* plot) {
  FillAudioEncoderTimeSeries(
      plot, [&](const AudioNetworkAdaptationEvent& ana_event) {
        if (ana_event.config.enable_dtx)
          return rtc::Optional<float>(
              static_cast<float>(*ana_event.config.enable_dtx));
        return rtc::Optional<float>();
      });
  plot->series_list_.back().label = "Audio encoder DTX";
  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "DTX (false/true)", kBottomMargin, kTopMargin);
  plot->SetTitle("Reported audio encoder DTX");
}

void EventLogAnalyzer::CreateAudioEncoderNumChannelsGraph(Plot* plot) {
  FillAudioEncoderTimeSeries(
      plot, [&](const AudioNetworkAdaptationEvent& ana_event) {
        if (ana_event.config.num_channels)
          return rtc::Optional<float>(
              static_cast<float>(*ana_event.config.num_channels));
        return rtc::Optional<float>();
      });
  plot->series_list_.back().label = "Audio encoder number of channels";
  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
  plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
                          kBottomMargin, kTopMargin);
  plot->SetTitle("Reported audio encoder number of channels");
}
}  // namespace plotting
}  // namespace webrtc