Move webrtc/{tools => rtc_tools}

webrtc/tools/event_log_visualizer/analyzer.h

Index: webrtc/tools/event_log_visualizer/analyzer.h
diff --git a/webrtc/tools/event_log_visualizer/analyzer.h b/webrtc/tools/event_log_visualizer/analyzer.h
deleted file mode 100644
index fab52b96a1b645c3524b63e1ed4dbdb1ce78a2cb..0000000000000000000000000000000000000000
--- a/webrtc/tools/event_log_visualizer/analyzer.h
+++ /dev/null
@@ -1,206 +0,0 @@
-/*
- *  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 WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
-#define WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
-
-#include <map>
-#include <memory>
-#include <set>
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "webrtc/base/function_view.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
-#include "webrtc/tools/event_log_visualizer/plot_base.h"
-
-namespace webrtc {
-namespace plotting {
-
-struct LoggedRtpPacket {
-  LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length)
-      : timestamp(timestamp), header(header), total_length(total_length) {}
-  uint64_t timestamp;
-  // TODO(terelius): This allocates space for 15 CSRCs even if none are used.
-  RTPHeader header;
-  size_t total_length;
-};
-
-struct LoggedRtcpPacket {
-  LoggedRtcpPacket(uint64_t timestamp,
-                   RTCPPacketType rtcp_type,
-                   std::unique_ptr<rtcp::RtcpPacket> rtcp_packet)
-      : timestamp(timestamp), type(rtcp_type), packet(std::move(rtcp_packet)) {}
-  uint64_t timestamp;
-  RTCPPacketType type;
-  std::unique_ptr<rtcp::RtcpPacket> packet;
-};
-
-struct LossBasedBweUpdate {
-  uint64_t timestamp;
-  int32_t new_bitrate;
-  uint8_t fraction_loss;
-  int32_t expected_packets;
-};
-
-struct AudioNetworkAdaptationEvent {
-  uint64_t timestamp;
-  AudioEncoderRuntimeConfig config;
-};
-
-class EventLogAnalyzer {
- public:
-  // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
-  // duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
-  // modified while the EventLogAnalyzer is being used.
-  explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
-
-  void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
-
-  void CreateAccumulatedPacketsGraph(PacketDirection desired_direction,
-                                     Plot* plot);
-
-  void CreatePlayoutGraph(Plot* plot);
-
-  void CreateAudioLevelGraph(Plot* plot);
-
-  void CreateSequenceNumberGraph(Plot* plot);
-
-  void CreateIncomingPacketLossGraph(Plot* plot);
-
-  void CreateDelayChangeGraph(Plot* plot);
-
-  void CreateAccumulatedDelayChangeGraph(Plot* plot);
-
-  void CreateFractionLossGraph(Plot* plot);
-
-  void CreateTotalBitrateGraph(PacketDirection desired_direction, Plot* plot);
-
-  void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
-
-  void CreateBweSimulationGraph(Plot* plot);
-
-  void CreateNetworkDelayFeedbackGraph(Plot* plot);
-  void CreateTimestampGraph(Plot* plot);
-
-  void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
-  void CreateAudioEncoderFrameLengthGraph(Plot* plot);
-  void CreateAudioEncoderUplinkPacketLossFractionGraph(Plot* plot);
-  void CreateAudioEncoderEnableFecGraph(Plot* plot);
-  void CreateAudioEncoderEnableDtxGraph(Plot* plot);
-  void CreateAudioEncoderNumChannelsGraph(Plot* plot);
-  void CreateAudioJitterBufferGraph(const std::string& replacement_file_name,
-                                    int file_sample_rate_hz,
-                                    Plot* plot);
-
-  // Returns a vector of capture and arrival timestamps for the video frames
-  // of the stream with the most number of frames.
-  std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const;
-
- private:
-  class StreamId {
-   public:
-    StreamId(uint32_t ssrc, webrtc::PacketDirection direction)
-        : ssrc_(ssrc), direction_(direction) {}
-    bool operator<(const StreamId& other) const {
-      return std::tie(ssrc_, direction_) <
-             std::tie(other.ssrc_, other.direction_);
-    }
-    bool operator==(const StreamId& other) const {
-      return std::tie(ssrc_, direction_) ==
-             std::tie(other.ssrc_, other.direction_);
-    }
-    uint32_t GetSsrc() const { return ssrc_; }
-    webrtc::PacketDirection GetDirection() const { return direction_; }
-
-   private:
-    uint32_t ssrc_;
-    webrtc::PacketDirection direction_;
-  };
-
-  template <typename T>
-  void CreateAccumulatedPacketsTimeSeries(
-      PacketDirection desired_direction,
-      Plot* plot,
-      const std::map<StreamId, std::vector<T>>& packets,
-      const std::string& label_prefix);
-
-  bool IsRtxSsrc(StreamId stream_id) const;
-
-  bool IsVideoSsrc(StreamId stream_id) const;
-
-  bool IsAudioSsrc(StreamId stream_id) const;
-
-  std::string GetStreamName(StreamId) const;
-
-  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_;
-
-  // Tracks what each stream is configured for. Note that a single SSRC can be
-  // in several sets. For example, the SSRC used for sending video over RTX
-  // will appear in both video_ssrcs_ and rtx_ssrcs_. In the unlikely case that
-  // an SSRC is reconfigured to a different media type mid-call, it will also
-  // appear in multiple sets.
-  std::set<StreamId> rtx_ssrcs_;
-  std::set<StreamId> video_ssrcs_;
-  std::set<StreamId> audio_ssrcs_;
-
-  // Maps a stream identifier consisting of ssrc and direction to the parsed
-  // RTP headers in that stream. Header extensions are parsed if the stream
-  // has been configured.
-  std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_;
-
-  std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_;
-
-  // Maps an SSRC to the timestamps of parsed audio playout events.
-  std::map<uint32_t, std::vector<uint64_t>> audio_playout_events_;
-
-  // Stores the timestamps for all log segments, in the form of associated start
-  // and end events.
-  std::vector<std::pair<uint64_t, uint64_t>> log_segments_;
-
-  // A list of all updates from the send-side loss-based bandwidth estimator.
-  std::vector<LossBasedBweUpdate> bwe_loss_updates_;
-
-  std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_;
-
-  std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent>
-      bwe_probe_cluster_created_events_;
-
-  std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_;
-
-  std::vector<ParsedRtcEventLog::BweDelayBasedUpdate> bwe_delay_updates_;
-
-  // Window and step size used for calculating moving averages, e.g. bitrate.
-  // The generated data points will be |step_| microseconds apart.
-  // Only events occuring at most |window_duration_| microseconds before the
-  // current data point will be part of the average.
-  uint64_t window_duration_;
-  uint64_t step_;
-
-  // First and last events of the log.
-  uint64_t begin_time_;
-  uint64_t end_time_;
-
-  // Duration (in seconds) of log file.
-  float call_duration_s_;
-};
-
-}  // namespace plotting
-}  // namespace webrtc
-
-#endif  // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_