| /* |
| * 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 <vector> |
| #include <map> |
| #include <memory> |
| #include <utility> |
| |
| #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::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 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<uint32_t> 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<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_; |
| |
| std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_; |
| |
| // A list of all updates from the send-side loss-based bandwidth estimator. |
| std::vector<BwePacketLossEvent> 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_ |