Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

rtc_tools/event_log_visualizer/analyzer.h

@@ -0,0 +1,207 @@
+/*
+ *  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_RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
+#define WEBRTC_RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
+
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#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/rtc_base/function_view.h"
+#include "webrtc/rtc_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 CreateIncomingDelayDeltaGraph(Plot* plot);
+  void CreateIncomingDelayGraph(Plot* plot);
+
+  void CreateFractionLossGraph(Plot* plot);
+
+  void CreateTotalBitrateGraph(PacketDirection desired_direction,
+                               Plot* plot,
+                               bool show_detector_state = false);
+
+  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 CreateAudioEncoderPacketLossGraph(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_RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_