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575998c2da
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}
305 lines
11 KiB
C++
305 lines
11 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
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#define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
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#include <map>
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#include <memory>
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#include <set>
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#include <string>
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#include <utility>
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#include <vector>
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#include "logging/rtc_event_log/rtc_event_log_parser.h"
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#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
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#include "rtc_base/strings/string_builder.h"
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#include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
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#include "rtc_tools/rtc_event_log_visualizer/triage_notifications.h"
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namespace webrtc {
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class AnalyzerConfig {
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public:
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float GetCallTimeSec(int64_t timestamp_us) const {
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int64_t offset = normalize_time_ ? begin_time_ : 0;
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return static_cast<float>(timestamp_us - offset) / 1000000;
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}
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float CallBeginTimeSec() const { return GetCallTimeSec(begin_time_); }
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float CallEndTimeSec() const { return GetCallTimeSec(end_time_); }
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// Window and step size used for calculating moving averages, e.g. bitrate.
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// The generated data points will be |step_| microseconds apart.
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// Only events occurring at most |window_duration_| microseconds before the
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// current data point will be part of the average.
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int64_t window_duration_;
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int64_t step_;
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// First and last events of the log.
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int64_t begin_time_;
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int64_t end_time_;
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bool normalize_time_;
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};
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class EventLogAnalyzer {
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public:
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// The EventLogAnalyzer keeps a reference to the ParsedRtcEventLogNew for the
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// duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
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// modified while the EventLogAnalyzer is being used.
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EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time);
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void CreatePacketGraph(PacketDirection direction, Plot* plot);
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void CreateRtcpTypeGraph(PacketDirection direction, Plot* plot);
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void CreateAccumulatedPacketsGraph(PacketDirection direction, Plot* plot);
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void CreatePlayoutGraph(Plot* plot);
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void CreateAudioLevelGraph(PacketDirection direction, Plot* plot);
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void CreateSequenceNumberGraph(Plot* plot);
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void CreateIncomingPacketLossGraph(Plot* plot);
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void CreateIncomingDelayGraph(Plot* plot);
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void CreateFractionLossGraph(Plot* plot);
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void CreateTotalIncomingBitrateGraph(Plot* plot);
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void CreateTotalOutgoingBitrateGraph(Plot* plot,
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bool show_detector_state = false,
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bool show_alr_state = false);
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void CreateStreamBitrateGraph(PacketDirection direction, Plot* plot);
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void CreateBitrateAllocationGraph(PacketDirection direction, Plot* plot);
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void CreateGoogCcSimulationGraph(Plot* plot);
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void CreateSendSideBweSimulationGraph(Plot* plot);
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void CreateReceiveSideBweSimulationGraph(Plot* plot);
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void CreateNetworkDelayFeedbackGraph(Plot* plot);
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void CreatePacerDelayGraph(Plot* plot);
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void CreateTimestampGraph(PacketDirection direction, Plot* plot);
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void CreateSenderAndReceiverReportPlot(
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PacketDirection direction,
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rtc::FunctionView<float(const rtcp::ReportBlock&)> fy,
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std::string title,
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std::string yaxis_label,
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Plot* plot);
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void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
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void CreateAudioEncoderFrameLengthGraph(Plot* plot);
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void CreateAudioEncoderPacketLossGraph(Plot* plot);
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void CreateAudioEncoderEnableFecGraph(Plot* plot);
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void CreateAudioEncoderEnableDtxGraph(Plot* plot);
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void CreateAudioEncoderNumChannelsGraph(Plot* plot);
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using NetEqStatsGetterMap =
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std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
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NetEqStatsGetterMap SimulateNetEq(const std::string& replacement_file_name,
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int file_sample_rate_hz) const;
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void CreateAudioJitterBufferGraph(uint32_t ssrc,
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const test::NetEqStatsGetter* stats_getter,
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Plot* plot) const;
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void CreateNetEqNetworkStatsGraph(
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const NetEqStatsGetterMap& neteq_stats_getters,
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rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
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const std::string& plot_name,
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Plot* plot) const;
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void CreateNetEqLifetimeStatsGraph(
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const NetEqStatsGetterMap& neteq_stats_getters,
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rtc::FunctionView<float(const NetEqLifetimeStatistics&)> stats_extractor,
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const std::string& plot_name,
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Plot* plot) const;
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void CreateIceCandidatePairConfigGraph(Plot* plot);
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void CreateIceConnectivityCheckGraph(Plot* plot);
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void CreateDtlsTransportStateGraph(Plot* plot);
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void CreateDtlsWritableStateGraph(Plot* plot);
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void CreateTriageNotifications();
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void PrintNotifications(FILE* file);
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private:
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struct LayerDescription {
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LayerDescription(uint32_t ssrc,
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uint8_t spatial_layer,
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uint8_t temporal_layer)
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: ssrc(ssrc),
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spatial_layer(spatial_layer),
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temporal_layer(temporal_layer) {}
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bool operator<(const LayerDescription& other) const {
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if (ssrc != other.ssrc)
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return ssrc < other.ssrc;
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if (spatial_layer != other.spatial_layer)
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return spatial_layer < other.spatial_layer;
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return temporal_layer < other.temporal_layer;
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}
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uint32_t ssrc;
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uint8_t spatial_layer;
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uint8_t temporal_layer;
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};
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bool IsRtxSsrc(PacketDirection direction, uint32_t ssrc) const {
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if (direction == kIncomingPacket) {
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return parsed_log_.incoming_rtx_ssrcs().find(ssrc) !=
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parsed_log_.incoming_rtx_ssrcs().end();
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} else {
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return parsed_log_.outgoing_rtx_ssrcs().find(ssrc) !=
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parsed_log_.outgoing_rtx_ssrcs().end();
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}
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}
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bool IsVideoSsrc(PacketDirection direction, uint32_t ssrc) const {
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if (direction == kIncomingPacket) {
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return parsed_log_.incoming_video_ssrcs().find(ssrc) !=
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parsed_log_.incoming_video_ssrcs().end();
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} else {
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return parsed_log_.outgoing_video_ssrcs().find(ssrc) !=
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parsed_log_.outgoing_video_ssrcs().end();
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}
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}
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bool IsAudioSsrc(PacketDirection direction, uint32_t ssrc) const {
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if (direction == kIncomingPacket) {
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return parsed_log_.incoming_audio_ssrcs().find(ssrc) !=
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parsed_log_.incoming_audio_ssrcs().end();
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} else {
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return parsed_log_.outgoing_audio_ssrcs().find(ssrc) !=
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parsed_log_.outgoing_audio_ssrcs().end();
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}
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}
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template <typename NetEqStatsType>
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void CreateNetEqStatsGraphInternal(
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const NetEqStatsGetterMap& neteq_stats,
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rtc::FunctionView<const std::vector<std::pair<int64_t, NetEqStatsType>>*(
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const test::NetEqStatsGetter*)> data_extractor,
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rtc::FunctionView<float(const NetEqStatsType&)> stats_extractor,
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const std::string& plot_name,
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Plot* plot) const;
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template <typename IterableType>
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void CreateAccumulatedPacketsTimeSeries(Plot* plot,
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const IterableType& packets,
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const std::string& label);
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void CreateStreamGapAlerts(PacketDirection direction);
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void CreateTransmissionGapAlerts(PacketDirection direction);
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std::string GetStreamName(PacketDirection direction, uint32_t ssrc) const {
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char buffer[200];
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rtc::SimpleStringBuilder name(buffer);
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if (IsAudioSsrc(direction, ssrc)) {
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name << "Audio ";
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} else if (IsVideoSsrc(direction, ssrc)) {
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name << "Video ";
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} else {
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name << "Unknown ";
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}
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if (IsRtxSsrc(direction, ssrc)) {
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name << "RTX ";
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}
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if (direction == kIncomingPacket)
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name << "(In) ";
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else
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name << "(Out) ";
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name << "SSRC " << ssrc;
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return name.str();
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}
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std::string GetLayerName(LayerDescription layer) const {
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char buffer[100];
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rtc::SimpleStringBuilder name(buffer);
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name << "SSRC " << layer.ssrc << " sl " << layer.spatial_layer << ", tl "
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<< layer.temporal_layer;
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return name.str();
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}
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void Alert_RtpLogTimeGap(PacketDirection direction,
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float time_seconds,
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int64_t duration) {
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if (direction == kIncomingPacket) {
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incoming_rtp_recv_time_gaps_.emplace_back(time_seconds, duration);
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} else {
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outgoing_rtp_send_time_gaps_.emplace_back(time_seconds, duration);
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}
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}
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void Alert_RtcpLogTimeGap(PacketDirection direction,
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float time_seconds,
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int64_t duration) {
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if (direction == kIncomingPacket) {
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incoming_rtcp_recv_time_gaps_.emplace_back(time_seconds, duration);
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} else {
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outgoing_rtcp_send_time_gaps_.emplace_back(time_seconds, duration);
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}
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}
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void Alert_SeqNumJump(PacketDirection direction,
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float time_seconds,
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uint32_t ssrc) {
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if (direction == kIncomingPacket) {
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incoming_seq_num_jumps_.emplace_back(time_seconds, ssrc);
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} else {
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outgoing_seq_num_jumps_.emplace_back(time_seconds, ssrc);
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}
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}
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void Alert_CaptureTimeJump(PacketDirection direction,
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float time_seconds,
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uint32_t ssrc) {
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if (direction == kIncomingPacket) {
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incoming_capture_time_jumps_.emplace_back(time_seconds, ssrc);
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} else {
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outgoing_capture_time_jumps_.emplace_back(time_seconds, ssrc);
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}
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}
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void Alert_OutgoingHighLoss(double avg_loss_fraction) {
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outgoing_high_loss_alerts_.emplace_back(avg_loss_fraction);
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}
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std::string GetCandidatePairLogDescriptionFromId(uint32_t candidate_pair_id);
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const ParsedRtcEventLog& parsed_log_;
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// A list of SSRCs we are interested in analysing.
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// If left empty, all SSRCs will be considered relevant.
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std::vector<uint32_t> desired_ssrc_;
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// Stores the timestamps for all log segments, in the form of associated start
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// and end events.
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std::vector<std::pair<int64_t, int64_t>> log_segments_;
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std::vector<IncomingRtpReceiveTimeGap> incoming_rtp_recv_time_gaps_;
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std::vector<IncomingRtcpReceiveTimeGap> incoming_rtcp_recv_time_gaps_;
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std::vector<OutgoingRtpSendTimeGap> outgoing_rtp_send_time_gaps_;
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std::vector<OutgoingRtcpSendTimeGap> outgoing_rtcp_send_time_gaps_;
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std::vector<IncomingSeqNumJump> incoming_seq_num_jumps_;
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std::vector<IncomingCaptureTimeJump> incoming_capture_time_jumps_;
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std::vector<OutgoingSeqNoJump> outgoing_seq_num_jumps_;
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std::vector<OutgoingCaptureTimeJump> outgoing_capture_time_jumps_;
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std::vector<OutgoingHighLoss> outgoing_high_loss_alerts_;
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std::map<uint32_t, std::string> candidate_pair_desc_by_id_;
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AnalyzerConfig config_;
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};
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} // namespace webrtc
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#endif // RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_ANALYZER_H_
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