/* * 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 #include #include #include #include "webrtc/call/rtc_event_log_parser.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 { 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 CreatePlayoutGraph(Plot* plot); void CreateSequenceNumberGraph(Plot* plot); void CreateDelayChangeGraph(Plot* plot); void CreateAccumulatedDelayChangeGraph(Plot* plot); void CreateTotalBitrateGraph(PacketDirection desired_direction, Plot* plot); void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot); void CreateBweGraph(Plot* plot); void CreateNetworkDelayFeebackGraph(Plot* plot); private: class StreamId { public: StreamId(uint32_t ssrc, webrtc::PacketDirection direction) : ssrc_(ssrc), direction_(direction) {} bool operator<(const StreamId& other) const; bool operator==(const StreamId& other) const; uint32_t GetSsrc() const { return ssrc_; } webrtc::PacketDirection GetDirection() const { return direction_; } private: uint32_t ssrc_; webrtc::PacketDirection direction_; }; struct LoggedRtpPacket { LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length) : timestamp(timestamp), header(header), total_length(total_length) {} uint64_t timestamp; RTPHeader header; size_t total_length; }; struct LoggedRtcpPacket { LoggedRtcpPacket(uint64_t timestamp, RTCPPacketType rtcp_type, std::unique_ptr rtcp_packet) : timestamp(timestamp), type(rtcp_type), packet(std::move(rtcp_packet)) {} uint64_t timestamp; RTCPPacketType type; std::unique_ptr packet; }; struct BwePacketLossEvent { uint64_t timestamp; int32_t new_bitrate; uint8_t fraction_loss; int32_t expected_packets; }; const ParsedRtcEventLog& parsed_log_; // A list of SSRCs we are interested in analysing. // If left empty, all SSRCs will be considered relevant. std::vector desired_ssrc_; // Maps a stream identifier consisting of ssrc, direction and MediaType // to the parsed RTP headers in that stream. Header extensions are parsed // if the stream has been configured. std::map> rtp_packets_; std::map> rtcp_packets_; // A list of all updates from the send-side loss-based bandwidth estimator. std::vector bwe_loss_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_