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Reland "Create new API for RtcEventLogParser."

Browse Source 
The new API stores events gathered by event type. For example, it is
possible to ask for a list of all incoming RTCP messages or all audio
playout events.

The new API is experimental and may change over next few weeks. Once
it has stabilized and all unit tests and existing tools have been
ported to the new API, the old one will be removed.

This CL also updates the event_log_visualizer tool to use the new
parser API. This is not a funcional change except for:
- Incoming and outgoing audio level are now drawn in two separate plots.
- Incoming and outgoing timstamps are now drawn in two separate plots.
- RTCP count is no longer split into Video and Audio. It also counts
  all RTCP packets rather than only specific message types.
- Slight timing difference in sendside BWE simulation due to only
  iterating over transport feedbacks and not over all RTCP packets.
  This timing changes are not visible in the plots.


Media type for RTCP messages might not be identified correctly by
rtc_event_log2text anymore. On the other hand, assigning a specific
media type to an RTCP packet was a bit hacky to begin with.

Bug: webrtc:8111
Change-Id: Ib244338c86a2c1a010c668a7aba440482023b512
Reviewed-on: https://webrtc-review.googlesource.com/73140
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23056}
This commit is contained in:
Bjorn Terelius
2018-04-27 14:33:34 +02:00
committed by Commit Bot
parent ebd9abc1a2
commit c4ca1d3f37
23 changed files with 3402 additions and 1539 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1911
rtc_tools/event_log_visualizer/analyzer.cc
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File diff suppressed because it is too large Load Diff

265
rtc_tools/event_log_visualizer/analyzer.h
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@ -18,70 +18,27 @@
#include <utility>
#include <vector>
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "rtc_base/function_view.h"
#include "logging/rtc_event_log/rtc_event_log_parser_new.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 {
namespace plotting {
struct LoggedRtpPacket {
LoggedRtpPacket(uint64_t timestamp,
RTPHeader header,
size_t header_length,
size_t total_length)
: timestamp(timestamp),
header(header),
header_length(header_length),
total_length(total_length) {}
uint64_t timestamp;
// TODO(terelius): This allocates space for 15 CSRCs even if none are used.
RTPHeader header;
size_t header_length;
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 LossBasedBweUpdate {
uint64_t timestamp;
int32_t new_bitrate;
uint8_t fraction_loss;
int32_t expected_packets;
};
struct AudioNetworkAdaptationEvent {
uint64_t timestamp;
AudioEncoderRuntimeConfig config;
};
class EventLogAnalyzer {
public:
// The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
// duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
// 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.
explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
explicit EventLogAnalyzer(const ParsedRtcEventLogNew& log);
void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
void CreatePacketGraph(PacketDirection direction, Plot* plot);
void CreateAccumulatedPacketsGraph(PacketDirection desired_direction,
Plot* plot);
void CreateAccumulatedPacketsGraph(PacketDirection direction, Plot* plot);
void CreatePlayoutGraph(Plot* plot);
void CreateAudioLevelGraph(Plot* plot);
void CreateAudioLevelGraph(PacketDirection direction, Plot* plot);
void CreateSequenceNumberGraph(Plot* plot);
@ -92,19 +49,20 @@ class EventLogAnalyzer {
void CreateFractionLossGraph(Plot* plot);
void CreateTotalBitrateGraph(PacketDirection desired_direction,
Plot* plot,
bool show_detector_state = false,
bool show_alr_state = false);
void CreateTotalIncomingBitrateGraph(Plot* plot);
void CreateTotalOutgoingBitrateGraph(Plot* plot,
bool show_detector_state = false,
bool show_alr_state = false);
void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
void CreateStreamBitrateGraph(PacketDirection direction, Plot* plot);
void CreateSendSideBweSimulationGraph(Plot* plot);
void CreateReceiveSideBweSimulationGraph(Plot* plot);
void CreateNetworkDelayFeedbackGraph(Plot* plot);
void CreatePacerDelayGraph(Plot* plot);
void CreateTimestampGraph(Plot* plot);
void CreateTimestampGraph(PacketDirection direction, Plot* plot);
void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
void CreateAudioEncoderFrameLengthGraph(Plot* plot);
@ -119,108 +77,136 @@ class EventLogAnalyzer {
void CreateIceCandidatePairConfigGraph(Plot* plot);
void CreateIceConnectivityCheckGraph(Plot* plot);
// Returns a vector of capture and arrival timestamps for the video frames
// of the stream with the most number of frames.
std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const;
void CreateTriageNotifications();
void PrintNotifications(FILE* file);
private:
class StreamId {
public:
StreamId(uint32_t ssrc, webrtc::PacketDirection direction)
: ssrc_(ssrc), direction_(direction) {}
bool operator<(const StreamId& other) const {
return std::tie(ssrc_, direction_) <
std::tie(other.ssrc_, other.direction_);
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 operator==(const StreamId& other) const {
return std::tie(ssrc_, direction_) ==
std::tie(other.ssrc_, other.direction_);
}
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();
}
uint32_t GetSsrc() const { return ssrc_; }
webrtc::PacketDirection GetDirection() const { return direction_; }
}
private:
uint32_t ssrc_;
webrtc::PacketDirection direction_;
};
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 T>
void CreateAccumulatedPacketsTimeSeries(
PacketDirection desired_direction,
Plot* plot,
const std::map<StreamId, std::vector<T>>& packets,
const std::string& label_prefix);
template <typename IterableType>
void CreateAccumulatedPacketsTimeSeries(Plot* plot,
const IterableType& packets,
const std::string& label);
bool IsRtxSsrc(StreamId stream_id) const;
void CreateStreamGapAlerts(PacketDirection direction);
void CreateTransmissionGapAlerts(PacketDirection direction);
bool IsVideoSsrc(StreamId stream_id) const;
bool IsAudioSsrc(StreamId stream_id) const;
std::string GetStreamName(StreamId stream_id) const;
rtc::Optional<uint32_t> EstimateRtpClockFrequency(
const std::vector<LoggedRtpPacket>& packets) const;
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();
}
float ToCallTime(int64_t timestamp) const;
void Notification(std::unique_ptr<TriageNotification> notification);
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 ParsedRtcEventLog& parsed_log_;
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_;
// Tracks what each stream is configured for. Note that a single SSRC can be
// in several sets. For example, the SSRC used for sending video over RTX
// will appear in both video_ssrcs_ and rtx_ssrcs_. In the unlikely case that
// an SSRC is reconfigured to a different media type mid-call, it will also
// appear in multiple sets.
std::set<StreamId> rtx_ssrcs_;
std::set<StreamId> video_ssrcs_;
std::set<StreamId> audio_ssrcs_;
// Maps a stream identifier consisting of ssrc and direction 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_;
// Maps an SSRC to the timestamps of parsed audio playout events.
std::map<uint32_t, std::vector<uint64_t>> audio_playout_events_;
// Stores the timestamps for all log segments, in the form of associated start
// and end events.
std::vector<std::pair<uint64_t, uint64_t>> log_segments_;
std::vector<std::pair<int64_t, int64_t>> log_segments_;
// A list of all updates from the send-side loss-based bandwidth estimator.
std::vector<LossBasedBweUpdate> bwe_loss_updates_;
std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_;
std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent>
bwe_probe_cluster_created_events_;
std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_;
std::vector<ParsedRtcEventLog::BweDelayBasedUpdate> bwe_delay_updates_;
std::vector<std::unique_ptr<TriageNotification>> notifications_;
std::vector<ParsedRtcEventLog::AlrStateEvent> alr_state_events_;
std::vector<ParsedRtcEventLog::IceCandidatePairConfig>
ice_candidate_pair_configs_;
std::vector<ParsedRtcEventLog::IceCandidatePairEvent>
ice_candidate_pair_events_;
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_;
@ -228,18 +214,17 @@ class EventLogAnalyzer {
// 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_;
int64_t window_duration_;
int64_t step_;
// First and last events of the log.
uint64_t begin_time_;
uint64_t end_time_;
int64_t begin_time_;
int64_t end_time_;
// Duration (in seconds) of log file.
float call_duration_s_;
};
} // namespace plotting
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_

69
rtc_tools/event_log_visualizer/main.cc
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@ -10,7 +10,7 @@
#include <iostream>
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
#include "rtc_base/flags.h"
#include "rtc_tools/event_log_visualizer/analyzer.h"
#include "rtc_tools/event_log_visualizer/plot_base.h"
@ -143,10 +143,15 @@ DEFINE_bool(show_alr_state,
false,
"Show the state ALR state on the total bitrate graph");
DEFINE_bool(
print_triage_notifications,
false,
"Print triage notifications, i.e. a list of suspicious looking events.");
DEFINE_bool(parse_unconfigured_header_extensions,
true,
"Attempt to parse unconfigured header extensions using the default "
"WebRTC mapping. This can give very misleading results if the "
"application negotiates a different mapping.");
DEFINE_bool(print_triage_alerts,
false,
"Print triage alerts, i.e. a list of potential problems.");
void SetAllPlotFlags(bool setting);
@ -209,7 +214,13 @@ int main(int argc, char* argv[]) {
std::string filename = argv[1];
webrtc::ParsedRtcEventLog parsed_log;
webrtc::ParsedRtcEventLogNew::UnconfiguredHeaderExtensions header_extensions =
webrtc::ParsedRtcEventLogNew::UnconfiguredHeaderExtensions::kDontParse;
if (FLAG_parse_unconfigured_header_extensions) {
header_extensions = webrtc::ParsedRtcEventLogNew::
UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig;
}
webrtc::ParsedRtcEventLogNew parsed_log(header_extensions);
if (!parsed_log.ParseFile(filename)) {
std::cerr << "Could not parse the entire log file." << std::endl;
@ -218,31 +229,34 @@ int main(int argc, char* argv[]) {
<< std::endl;
}
webrtc::plotting::EventLogAnalyzer analyzer(parsed_log);
std::unique_ptr<webrtc::plotting::PlotCollection> collection(
new webrtc::plotting::PythonPlotCollection());
webrtc::EventLogAnalyzer analyzer(parsed_log);
std::unique_ptr<webrtc::PlotCollection> collection(
new webrtc::PythonPlotCollection());
if (FLAG_plot_incoming_packet_sizes) {
analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
analyzer.CreatePacketGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_packet_sizes) {
analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
analyzer.CreatePacketGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_incoming_packet_count) {
analyzer.CreateAccumulatedPacketsGraph(
webrtc::PacketDirection::kIncomingPacket, collection->AppendNewPlot());
analyzer.CreateAccumulatedPacketsGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_packet_count) {
analyzer.CreateAccumulatedPacketsGraph(
webrtc::PacketDirection::kOutgoingPacket, collection->AppendNewPlot());
analyzer.CreateAccumulatedPacketsGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_audio_playout) {
analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_level) {
analyzer.CreateAudioLevelGraph(collection->AppendNewPlot());
analyzer.CreateAudioLevelGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
analyzer.CreateAudioLevelGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_incoming_sequence_number_delta) {
analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
@ -257,23 +271,19 @@ int main(int argc, char* argv[]) {
analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
}
if (FLAG_plot_incoming_bitrate) {
analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
collection->AppendNewPlot(),
FLAG_show_detector_state,
FLAG_show_alr_state);
analyzer.CreateTotalIncomingBitrateGraph(collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_bitrate) {
analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
collection->AppendNewPlot(),
FLAG_show_detector_state,
FLAG_show_alr_state);
analyzer.CreateTotalOutgoingBitrateGraph(collection->AppendNewPlot(),
FLAG_show_detector_state,
FLAG_show_alr_state);
}
if (FLAG_plot_incoming_stream_bitrate) {
analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
analyzer.CreateStreamBitrateGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_stream_bitrate) {
analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
analyzer.CreateStreamBitrateGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_simulated_receiveside_bwe) {
@ -289,7 +299,10 @@ int main(int argc, char* argv[]) {
analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
}
if (FLAG_plot_timestamps) {
analyzer.CreateTimestampGraph(collection->AppendNewPlot());
analyzer.CreateTimestampGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
analyzer.CreateTimestampGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_pacer_delay) {
analyzer.CreatePacerDelayGraph(collection->AppendNewPlot());
@ -333,7 +346,7 @@ int main(int argc, char* argv[]) {
collection->Draw();
if (FLAG_print_triage_notifications) {
if (FLAG_print_triage_alerts) {
analyzer.CreateTriageNotifications();
analyzer.PrintNotifications(stderr);
}

2
rtc_tools/event_log_visualizer/plot_base.cc
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@ -15,7 +15,6 @@
#include "rtc_base/checks.h"
namespace webrtc {
namespace plotting {
void Plot::SetXAxis(float min_value,
float max_value,
@ -85,5 +84,4 @@ void Plot::AppendTimeSeriesIfNotEmpty(TimeSeries&& time_series) {
}
}
} // namespace plotting
} // namespace webrtc

2
rtc_tools/event_log_visualizer/plot_base.h
View File

@ -16,7 +16,6 @@
#include <vector>
namespace webrtc {
namespace plotting {
enum class LineStyle {
kNone, // No line connecting the points. Used to create scatter plots.
@ -173,7 +172,6 @@ class PlotCollection {
std::vector<std::unique_ptr<Plot> > plots_;
};
} // namespace plotting
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_BASE_H_

2
rtc_tools/event_log_visualizer/plot_protobuf.cc
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@ -13,7 +13,6 @@
#include <memory>
namespace webrtc {
namespace plotting {
ProtobufPlot::ProtobufPlot() {}
@ -83,5 +82,4 @@ Plot* ProtobufPlotCollection::AppendNewPlot() {
return plot;
}
} // namespace plotting
} // namespace webrtc

2
rtc_tools/event_log_visualizer/plot_protobuf.h
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@ -17,7 +17,6 @@ RTC_POP_IGNORING_WUNDEF()
#include "rtc_tools/event_log_visualizer/plot_base.h"
namespace webrtc {
namespace plotting {
class ProtobufPlot final : public Plot {
public:
@ -36,7 +35,6 @@ class ProtobufPlotCollection final : public PlotCollection {
void ExportProtobuf(webrtc::analytics::ChartCollection* collection);
};
} // namespace plotting
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PROTOBUF_H_

2
rtc_tools/event_log_visualizer/plot_python.cc
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@ -17,7 +17,6 @@
#include "rtc_base/checks.h"
namespace webrtc {
namespace plotting {
PythonPlot::PythonPlot() {}
@ -180,5 +179,4 @@ Plot* PythonPlotCollection::AppendNewPlot() {
return plot;
}
} // namespace plotting
} // namespace webrtc

2
rtc_tools/event_log_visualizer/plot_python.h
View File

@ -13,7 +13,6 @@
#include "rtc_tools/event_log_visualizer/plot_base.h"
namespace webrtc {
namespace plotting {
class PythonPlot final : public Plot {
public:
@ -30,7 +29,6 @@ class PythonPlotCollection final : public PlotCollection {
Plot* AppendNewPlot() override;
};
} // namespace plotting
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_

89
rtc_tools/event_log_visualizer/triage_notifications.h
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@ -14,130 +14,136 @@
#include <string>
namespace webrtc {
namespace plotting {
class TriageNotification {
public:
TriageNotification() : time_seconds_() {}
explicit TriageNotification(float time_seconds)
: time_seconds_(time_seconds) {}
virtual ~TriageNotification() = default;
virtual std::string ToString() = 0;
rtc::Optional<float> Time() { return time_seconds_; }
private:
rtc::Optional<float> time_seconds_;
};
class IncomingRtpReceiveTimeGap : public TriageNotification {
class IncomingRtpReceiveTimeGap {
public:
IncomingRtpReceiveTimeGap(float time_seconds, int64_t duration)
: TriageNotification(time_seconds), duration_(duration) {}
std::string ToString() {
: time_seconds_(time_seconds), duration_(duration) {}
float Time() const { return time_seconds_; }
std::string ToString() const {
return std::string("No RTP packets received for ") +
std::to_string(duration_) + std::string(" ms");
}
private:
float time_seconds_;
int64_t duration_;
};
class IncomingRtcpReceiveTimeGap : public TriageNotification {
class IncomingRtcpReceiveTimeGap {
public:
IncomingRtcpReceiveTimeGap(float time_seconds, int64_t duration)
: TriageNotification(time_seconds), duration_(duration) {}
std::string ToString() {
: time_seconds_(time_seconds), duration_(duration) {}
float Time() const { return time_seconds_; }
std::string ToString() const {
return std::string("No RTCP packets received for ") +
std::to_string(duration_) + std::string(" ms");
}
private:
float time_seconds_;
int64_t duration_;
};
class OutgoingRtpSendTimeGap : public TriageNotification {
class OutgoingRtpSendTimeGap {
public:
OutgoingRtpSendTimeGap(float time_seconds, int64_t duration)
: TriageNotification(time_seconds), duration_(duration) {}
std::string ToString() {
: time_seconds_(time_seconds), duration_(duration) {}
float Time() const { return time_seconds_; }
std::string ToString() const {
return std::string("No RTP packets sent for ") + std::to_string(duration_) +
std::string(" ms");
}
private:
float time_seconds_;
int64_t duration_;
};
class OutgoingRtcpSendTimeGap : public TriageNotification {
class OutgoingRtcpSendTimeGap {
public:
OutgoingRtcpSendTimeGap(float time_seconds, int64_t duration)
: TriageNotification(time_seconds), duration_(duration) {}
std::string ToString() {
: time_seconds_(time_seconds), duration_(duration) {}
float Time() const { return time_seconds_; }
std::string ToString() const {
return std::string("No RTCP packets sent for ") +
std::to_string(duration_) + std::string(" ms");
}
private:
float time_seconds_;
int64_t duration_;
};
class IncomingSeqNoJump : public TriageNotification {
class IncomingSeqNumJump {
public:
IncomingSeqNoJump(float time_seconds, uint32_t ssrc)
: TriageNotification(time_seconds), ssrc_(ssrc) {}
std::string ToString() {
IncomingSeqNumJump(float time_seconds, uint32_t ssrc)
: time_seconds_(time_seconds), ssrc_(ssrc) {}
float Time() const { return time_seconds_; }
std::string ToString() const {
return std::string("Sequence number jumps on incoming SSRC ") +
std::to_string(ssrc_);
}
private:
float time_seconds_;
uint32_t ssrc_;
};
class IncomingCaptureTimeJump : public TriageNotification {
class IncomingCaptureTimeJump {
public:
IncomingCaptureTimeJump(float time_seconds, uint32_t ssrc)
: TriageNotification(time_seconds), ssrc_(ssrc) {}
std::string ToString() {
: time_seconds_(time_seconds), ssrc_(ssrc) {}
float Time() const { return time_seconds_; }
std::string ToString() const {
return std::string("Capture timestamp jumps on incoming SSRC ") +
std::to_string(ssrc_);
}
private:
float time_seconds_;
uint32_t ssrc_;
};
class OutgoingSeqNoJump : public TriageNotification {
class OutgoingSeqNoJump {
public:
OutgoingSeqNoJump(float time_seconds, uint32_t ssrc)
: TriageNotification(time_seconds), ssrc_(ssrc) {}
std::string ToString() {
: time_seconds_(time_seconds), ssrc_(ssrc) {}
float Time() const { return time_seconds_; }
std::string ToString() const {
return std::string("Sequence number jumps on outgoing SSRC ") +
std::to_string(ssrc_);
}
private:
float time_seconds_;
uint32_t ssrc_;
};
class OutgoingCaptureTimeJump : public TriageNotification {
class OutgoingCaptureTimeJump {
public:
OutgoingCaptureTimeJump(float time_seconds, uint32_t ssrc)
: TriageNotification(time_seconds), ssrc_(ssrc) {}
std::string ToString() {
: time_seconds_(time_seconds), ssrc_(ssrc) {}
float Time() const { return time_seconds_; }
std::string ToString() const {
return std::string("Capture timestamp jumps on outgoing SSRC ") +
std::to_string(ssrc_);
}
private:
float time_seconds_;
uint32_t ssrc_;
};
class OutgoingHighLoss : public TriageNotification {
class OutgoingHighLoss {
public:
explicit OutgoingHighLoss(double avg_loss_fraction)
: avg_loss_fraction_(avg_loss_fraction) {}
std::string ToString() {
std::string ToString() const {
return std::string("High average loss (") +
std::to_string(avg_loss_fraction_ * 100) +
std::string("%) across the call.");
@ -147,7 +153,6 @@ class OutgoingHighLoss : public TriageNotification {
double avg_loss_fraction_;
};
} // namespace plotting
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_TRIAGE_NOTIFICATIONS_H_