blob: d4a7fc3078238bd2be69bfc3aceb18cd47809867 [file] [log] [blame]
/*
* 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 RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
#define RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_tools/event_log_visualizer/plot_base.h"
#include "rtc_tools/event_log_visualizer/triage_notifications.h"
namespace webrtc {
class EventLogAnalyzer {
public:
// The EventLogAnalyzer keeps a reference to the ParsedRtcEventLogNew for the
// duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
// modified while the EventLogAnalyzer is being used.
EventLogAnalyzer(const ParsedRtcEventLogNew& log, bool normalize_time);
void CreatePacketGraph(PacketDirection direction, Plot* plot);
void CreateAccumulatedPacketsGraph(PacketDirection direction, Plot* plot);
void CreatePlayoutGraph(Plot* plot);
void CreateAudioLevelGraph(PacketDirection direction, Plot* plot);
void CreateSequenceNumberGraph(Plot* plot);
void CreateIncomingPacketLossGraph(Plot* plot);
void CreateIncomingDelayDeltaGraph(Plot* plot);
void CreateIncomingDelayGraph(Plot* plot);
void CreateFractionLossGraph(Plot* plot);
void CreateTotalIncomingBitrateGraph(Plot* plot);
void CreateTotalOutgoingBitrateGraph(Plot* plot,
bool show_detector_state = false,
bool show_alr_state = false);
void CreateStreamBitrateGraph(PacketDirection direction, Plot* plot);
void CreateSendSideBweSimulationGraph(Plot* plot);
void CreateReceiveSideBweSimulationGraph(Plot* plot);
void CreateNetworkDelayFeedbackGraph(Plot* plot);
void CreatePacerDelayGraph(Plot* plot);
void CreateTimestampGraph(PacketDirection direction, 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);
using NetEqStatsGetterMap =
std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
NetEqStatsGetterMap SimulateNetEq(const std::string& replacement_file_name,
int file_sample_rate_hz) const;
void CreateAudioJitterBufferGraph(
const NetEqStatsGetterMap& neteq_stats_getters,
Plot* plot) const;
void CreateNetEqStatsGraph(
const NetEqStatsGetterMap& neteq_stats_getters,
rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
const std::string& plot_name,
Plot* plot) const;
void CreateIceCandidatePairConfigGraph(Plot* plot);
void CreateIceConnectivityCheckGraph(Plot* plot);
void CreateTriageNotifications();
void PrintNotifications(FILE* file);
private:
bool IsRtxSsrc(PacketDirection direction, uint32_t ssrc) const {
if (direction == kIncomingPacket) {
return parsed_log_.incoming_rtx_ssrcs().find(ssrc) !=
parsed_log_.incoming_rtx_ssrcs().end();
} else {
return parsed_log_.outgoing_rtx_ssrcs().find(ssrc) !=
parsed_log_.outgoing_rtx_ssrcs().end();
}
}
bool IsVideoSsrc(PacketDirection direction, uint32_t ssrc) const {
if (direction == kIncomingPacket) {
return parsed_log_.incoming_video_ssrcs().find(ssrc) !=
parsed_log_.incoming_video_ssrcs().end();
} else {
return parsed_log_.outgoing_video_ssrcs().find(ssrc) !=
parsed_log_.outgoing_video_ssrcs().end();
}
}
bool IsAudioSsrc(PacketDirection direction, uint32_t ssrc) const {
if (direction == kIncomingPacket) {
return parsed_log_.incoming_audio_ssrcs().find(ssrc) !=
parsed_log_.incoming_audio_ssrcs().end();
} else {
return parsed_log_.outgoing_audio_ssrcs().find(ssrc) !=
parsed_log_.outgoing_audio_ssrcs().end();
}
}
template <typename IterableType>
void CreateAccumulatedPacketsTimeSeries(Plot* plot,
const IterableType& packets,
const std::string& label);
void CreateStreamGapAlerts(PacketDirection direction);
void CreateTransmissionGapAlerts(PacketDirection direction);
std::string GetStreamName(PacketDirection direction, uint32_t ssrc) const {
char buffer[200];
rtc::SimpleStringBuilder name(buffer);
if (IsAudioSsrc(direction, ssrc)) {
name << "Audio ";
} else if (IsVideoSsrc(direction, ssrc)) {
name << "Video ";
} else {
name << "Unknown ";
}
if (IsRtxSsrc(direction, ssrc)) {
name << "RTX ";
}
if (direction == kIncomingPacket)
name << "(In) ";
else
name << "(Out) ";
name << "SSRC " << ssrc;
return name.str();
}
int64_t ToCallTimeUs(int64_t timestamp) const;
float ToCallTimeSec(int64_t timestamp) const;
void Alert_RtpLogTimeGap(PacketDirection direction,
float time_seconds,
int64_t duration) {
if (direction == kIncomingPacket) {
incoming_rtp_recv_time_gaps_.emplace_back(time_seconds, duration);
} else {
outgoing_rtp_send_time_gaps_.emplace_back(time_seconds, duration);
}
}
void Alert_RtcpLogTimeGap(PacketDirection direction,
float time_seconds,
int64_t duration) {
if (direction == kIncomingPacket) {
incoming_rtcp_recv_time_gaps_.emplace_back(time_seconds, duration);
} else {
outgoing_rtcp_send_time_gaps_.emplace_back(time_seconds, duration);
}
}
void Alert_SeqNumJump(PacketDirection direction,
float time_seconds,
uint32_t ssrc) {
if (direction == kIncomingPacket) {
incoming_seq_num_jumps_.emplace_back(time_seconds, ssrc);
} else {
outgoing_seq_num_jumps_.emplace_back(time_seconds, ssrc);
}
}
void Alert_CaptureTimeJump(PacketDirection direction,
float time_seconds,
uint32_t ssrc) {
if (direction == kIncomingPacket) {
incoming_capture_time_jumps_.emplace_back(time_seconds, ssrc);
} else {
outgoing_capture_time_jumps_.emplace_back(time_seconds, ssrc);
}
}
void Alert_OutgoingHighLoss(double avg_loss_fraction) {
outgoing_high_loss_alerts_.emplace_back(avg_loss_fraction);
}
std::string GetCandidatePairLogDescriptionFromId(uint32_t candidate_pair_id);
const ParsedRtcEventLogNew& 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_;
// Stores the timestamps for all log segments, in the form of associated start
// and end events.
std::vector<std::pair<int64_t, int64_t>> log_segments_;
std::vector<IncomingRtpReceiveTimeGap> incoming_rtp_recv_time_gaps_;
std::vector<IncomingRtcpReceiveTimeGap> incoming_rtcp_recv_time_gaps_;
std::vector<OutgoingRtpSendTimeGap> outgoing_rtp_send_time_gaps_;
std::vector<OutgoingRtcpSendTimeGap> outgoing_rtcp_send_time_gaps_;
std::vector<IncomingSeqNumJump> incoming_seq_num_jumps_;
std::vector<IncomingCaptureTimeJump> incoming_capture_time_jumps_;
std::vector<OutgoingSeqNoJump> outgoing_seq_num_jumps_;
std::vector<OutgoingCaptureTimeJump> outgoing_capture_time_jumps_;
std::vector<OutgoingHighLoss> outgoing_high_loss_alerts_;
std::map<uint32_t, std::string> candidate_pair_desc_by_id_;
// 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.
int64_t window_duration_;
int64_t step_;
// First and last events of the log.
int64_t begin_time_;
int64_t end_time_;
const bool normalize_time_;
// Duration (in seconds) of log file.
float call_duration_s_;
};
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_