mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00

The RTC event log analyzer would previously only plot network latency for incoming video streams. (The latency is computed from the capture time in the RTP header, and the packet receive time.) This CL adds support for audio packets, which requires estimating the RTP clock frequency for the incoming packets. Bug: None Change-Id: Idf1ff9febfdd4097976b22a61f1c5679deb6068c Reviewed-on: https://webrtc-review.googlesource.com/c/108784 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25580}
263 lines
9.4 KiB
C++
263 lines
9.4 KiB
C++
/*
|
|
* 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 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 CreateSenderAndReceiverReportPlot(
|
|
PacketDirection direction,
|
|
rtc::FunctionView<float(const rtcp::ReportBlock&)> fy,
|
|
std::string title,
|
|
std::string yaxis_label,
|
|
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(uint32_t ssrc,
|
|
const test::NetEqStatsGetter* stats_getter,
|
|
Plot* plot) const;
|
|
void CreateNetEqNetworkStatsGraph(
|
|
const NetEqStatsGetterMap& neteq_stats_getters,
|
|
rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
|
|
const std::string& plot_name,
|
|
Plot* plot) const;
|
|
void CreateNetEqLifetimeStatsGraph(
|
|
const NetEqStatsGetterMap& neteq_stats_getters,
|
|
rtc::FunctionView<float(const NetEqLifetimeStatistics&)> 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 NetEqStatsType>
|
|
void CreateNetEqStatsGraphInternal(
|
|
const NetEqStatsGetterMap& neteq_stats,
|
|
rtc::FunctionView<const std::vector<std::pair<int64_t, NetEqStatsType>>*(
|
|
const test::NetEqStatsGetter*)> data_extractor,
|
|
rtc::FunctionView<float(const NetEqStatsType&)> stats_extractor,
|
|
const std::string& plot_name,
|
|
Plot* plot) const;
|
|
|
|
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_
|