Add rtc_ prefix to the event_log_visualizer directory.

No-Try: True
Bug: None
Change-Id: Iaa2b273ddab6567321f11bf74a91751cbdf957a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146710
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28681}
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h
new file mode 100644
index 0000000..c4f7220
--- /dev/null
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h
@@ -0,0 +1,305 @@
+/*
+ *  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.
+ */
+
+#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
+#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
+
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
+#include "rtc_tools/rtc_event_log_visualizer/triage_notifications.h"
+
+namespace webrtc {
+
+class AnalyzerConfig {
+ public:
+  float GetCallTimeSec(int64_t timestamp_us) const {
+    int64_t offset = normalize_time_ ? begin_time_ : 0;
+    return static_cast<float>(timestamp_us - offset) / 1000000;
+  }
+
+  float CallBeginTimeSec() const { return GetCallTimeSec(begin_time_); }
+
+  float CallEndTimeSec() const { return GetCallTimeSec(end_time_); }
+
+  // Window and step size used for calculating moving averages, e.g. bitrate.
+  // The generated data points will be |step_| microseconds apart.
+  // Only events occurring at most |window_duration_| microseconds before the
+  // current data point will be part of the average.
+  int64_t window_duration_;
+  int64_t step_;
+
+  // First and last events of the log.
+  int64_t begin_time_;
+  int64_t end_time_;
+  bool normalize_time_;
+};
+
+class EventLogAnalyzer {
+ public:
+  // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLogNew for the
+  // duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
+  // modified while the EventLogAnalyzer is being used.
+  EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time);
+
+  void CreatePacketGraph(PacketDirection direction, Plot* plot);
+
+  void CreateRtcpTypeGraph(PacketDirection direction, Plot* plot);
+
+  void CreateAccumulatedPacketsGraph(PacketDirection direction, Plot* plot);
+
+  void CreatePlayoutGraph(Plot* plot);
+
+  void CreateAudioLevelGraph(PacketDirection direction, Plot* plot);
+
+  void CreateSequenceNumberGraph(Plot* plot);
+
+  void CreateIncomingPacketLossGraph(Plot* plot);
+
+  void CreateIncomingDelayGraph(Plot* plot);
+
+  void CreateFractionLossGraph(Plot* plot);
+
+  void CreateTotalIncomingBitrateGraph(Plot* plot);
+  void CreateTotalOutgoingBitrateGraph(Plot* plot,
+                                       bool show_detector_state = false,
+                                       bool show_alr_state = false);
+
+  void CreateStreamBitrateGraph(PacketDirection direction, Plot* plot);
+  void CreateBitrateAllocationGraph(PacketDirection direction, Plot* plot);
+
+  void CreateGoogCcSimulationGraph(Plot* plot);
+  void CreateSendSideBweSimulationGraph(Plot* plot);
+  void CreateReceiveSideBweSimulationGraph(Plot* plot);
+
+  void CreateNetworkDelayFeedbackGraph(Plot* plot);
+  void CreatePacerDelayGraph(Plot* plot);
+
+  void CreateTimestampGraph(PacketDirection direction, Plot* plot);
+  void CreateSenderAndReceiverReportPlot(
+      PacketDirection direction,
+      rtc::FunctionView<float(const rtcp::ReportBlock&)> fy,
+      std::string title,
+      std::string yaxis_label,
+      Plot* plot);
+
+  void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
+  void CreateAudioEncoderFrameLengthGraph(Plot* plot);
+  void CreateAudioEncoderPacketLossGraph(Plot* plot);
+  void CreateAudioEncoderEnableFecGraph(Plot* plot);
+  void CreateAudioEncoderEnableDtxGraph(Plot* plot);
+  void CreateAudioEncoderNumChannelsGraph(Plot* plot);
+
+  using NetEqStatsGetterMap =
+      std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
+  NetEqStatsGetterMap SimulateNetEq(const std::string& replacement_file_name,
+                                    int file_sample_rate_hz) const;
+
+  void CreateAudioJitterBufferGraph(uint32_t ssrc,
+                                    const test::NetEqStatsGetter* stats_getter,
+                                    Plot* plot) const;
+  void CreateNetEqNetworkStatsGraph(
+      const NetEqStatsGetterMap& neteq_stats_getters,
+      rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
+      const std::string& plot_name,
+      Plot* plot) const;
+  void CreateNetEqLifetimeStatsGraph(
+      const NetEqStatsGetterMap& neteq_stats_getters,
+      rtc::FunctionView<float(const NetEqLifetimeStatistics&)> stats_extractor,
+      const std::string& plot_name,
+      Plot* plot) const;
+
+  void CreateIceCandidatePairConfigGraph(Plot* plot);
+  void CreateIceConnectivityCheckGraph(Plot* plot);
+
+  void CreateDtlsTransportStateGraph(Plot* plot);
+  void CreateDtlsWritableStateGraph(Plot* plot);
+
+  void CreateTriageNotifications();
+  void PrintNotifications(FILE* file);
+
+ private:
+  struct LayerDescription {
+    LayerDescription(uint32_t ssrc,
+                     uint8_t spatial_layer,
+                     uint8_t temporal_layer)
+        : ssrc(ssrc),
+          spatial_layer(spatial_layer),
+          temporal_layer(temporal_layer) {}
+    bool operator<(const LayerDescription& other) const {
+      if (ssrc != other.ssrc)
+        return ssrc < other.ssrc;
+      if (spatial_layer != other.spatial_layer)
+        return spatial_layer < other.spatial_layer;
+      return temporal_layer < other.temporal_layer;
+    }
+    uint32_t ssrc;
+    uint8_t spatial_layer;
+    uint8_t temporal_layer;
+  };
+
+  bool IsRtxSsrc(PacketDirection direction, uint32_t ssrc) const {
+    if (direction == kIncomingPacket) {
+      return parsed_log_.incoming_rtx_ssrcs().find(ssrc) !=
+             parsed_log_.incoming_rtx_ssrcs().end();
+    } else {
+      return parsed_log_.outgoing_rtx_ssrcs().find(ssrc) !=
+             parsed_log_.outgoing_rtx_ssrcs().end();
+    }
+  }
+
+  bool IsVideoSsrc(PacketDirection direction, uint32_t ssrc) const {
+    if (direction == kIncomingPacket) {
+      return parsed_log_.incoming_video_ssrcs().find(ssrc) !=
+             parsed_log_.incoming_video_ssrcs().end();
+    } else {
+      return parsed_log_.outgoing_video_ssrcs().find(ssrc) !=
+             parsed_log_.outgoing_video_ssrcs().end();
+    }
+  }
+
+  bool IsAudioSsrc(PacketDirection direction, uint32_t ssrc) const {
+    if (direction == kIncomingPacket) {
+      return parsed_log_.incoming_audio_ssrcs().find(ssrc) !=
+             parsed_log_.incoming_audio_ssrcs().end();
+    } else {
+      return parsed_log_.outgoing_audio_ssrcs().find(ssrc) !=
+             parsed_log_.outgoing_audio_ssrcs().end();
+    }
+  }
+
+  template <typename NetEqStatsType>
+  void CreateNetEqStatsGraphInternal(
+      const NetEqStatsGetterMap& neteq_stats,
+      rtc::FunctionView<const std::vector<std::pair<int64_t, NetEqStatsType>>*(
+          const test::NetEqStatsGetter*)> data_extractor,
+      rtc::FunctionView<float(const NetEqStatsType&)> stats_extractor,
+      const std::string& plot_name,
+      Plot* plot) const;
+
+  template <typename IterableType>
+  void CreateAccumulatedPacketsTimeSeries(Plot* plot,
+                                          const IterableType& packets,
+                                          const std::string& label);
+
+  void CreateStreamGapAlerts(PacketDirection direction);
+  void CreateTransmissionGapAlerts(PacketDirection direction);
+
+  std::string GetStreamName(PacketDirection direction, uint32_t ssrc) const {
+    char buffer[200];
+    rtc::SimpleStringBuilder name(buffer);
+    if (IsAudioSsrc(direction, ssrc)) {
+      name << "Audio ";
+    } else if (IsVideoSsrc(direction, ssrc)) {
+      name << "Video ";
+    } else {
+      name << "Unknown ";
+    }
+    if (IsRtxSsrc(direction, ssrc)) {
+      name << "RTX ";
+    }
+    if (direction == kIncomingPacket)
+      name << "(In) ";
+    else
+      name << "(Out) ";
+    name << "SSRC " << ssrc;
+    return name.str();
+  }
+
+  std::string GetLayerName(LayerDescription layer) const {
+    char buffer[100];
+    rtc::SimpleStringBuilder name(buffer);
+    name << "SSRC " << layer.ssrc << " sl " << layer.spatial_layer << ", tl "
+         << layer.temporal_layer;
+    return name.str();
+  }
+
+  void Alert_RtpLogTimeGap(PacketDirection direction,
+                           float time_seconds,
+                           int64_t duration) {
+    if (direction == kIncomingPacket) {
+      incoming_rtp_recv_time_gaps_.emplace_back(time_seconds, duration);
+    } else {
+      outgoing_rtp_send_time_gaps_.emplace_back(time_seconds, duration);
+    }
+  }
+
+  void Alert_RtcpLogTimeGap(PacketDirection direction,
+                            float time_seconds,
+                            int64_t duration) {
+    if (direction == kIncomingPacket) {
+      incoming_rtcp_recv_time_gaps_.emplace_back(time_seconds, duration);
+    } else {
+      outgoing_rtcp_send_time_gaps_.emplace_back(time_seconds, duration);
+    }
+  }
+
+  void Alert_SeqNumJump(PacketDirection direction,
+                        float time_seconds,
+                        uint32_t ssrc) {
+    if (direction == kIncomingPacket) {
+      incoming_seq_num_jumps_.emplace_back(time_seconds, ssrc);
+    } else {
+      outgoing_seq_num_jumps_.emplace_back(time_seconds, ssrc);
+    }
+  }
+
+  void Alert_CaptureTimeJump(PacketDirection direction,
+                             float time_seconds,
+                             uint32_t ssrc) {
+    if (direction == kIncomingPacket) {
+      incoming_capture_time_jumps_.emplace_back(time_seconds, ssrc);
+    } else {
+      outgoing_capture_time_jumps_.emplace_back(time_seconds, ssrc);
+    }
+  }
+
+  void Alert_OutgoingHighLoss(double avg_loss_fraction) {
+    outgoing_high_loss_alerts_.emplace_back(avg_loss_fraction);
+  }
+
+  std::string GetCandidatePairLogDescriptionFromId(uint32_t candidate_pair_id);
+
+  const ParsedRtcEventLog& parsed_log_;
+
+  // A list of SSRCs we are interested in analysing.
+  // If left empty, all SSRCs will be considered relevant.
+  std::vector<uint32_t> desired_ssrc_;
+
+  // Stores the timestamps for all log segments, in the form of associated start
+  // and end events.
+  std::vector<std::pair<int64_t, int64_t>> log_segments_;
+
+  std::vector<IncomingRtpReceiveTimeGap> incoming_rtp_recv_time_gaps_;
+  std::vector<IncomingRtcpReceiveTimeGap> incoming_rtcp_recv_time_gaps_;
+  std::vector<OutgoingRtpSendTimeGap> outgoing_rtp_send_time_gaps_;
+  std::vector<OutgoingRtcpSendTimeGap> outgoing_rtcp_send_time_gaps_;
+  std::vector<IncomingSeqNumJump> incoming_seq_num_jumps_;
+  std::vector<IncomingCaptureTimeJump> incoming_capture_time_jumps_;
+  std::vector<OutgoingSeqNoJump> outgoing_seq_num_jumps_;
+  std::vector<OutgoingCaptureTimeJump> outgoing_capture_time_jumps_;
+  std::vector<OutgoingHighLoss> outgoing_high_loss_alerts_;
+
+  std::map<uint32_t, std::string> candidate_pair_desc_by_id_;
+
+  AnalyzerConfig config_;
+};
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_