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/*
* 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.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
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#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 AnalyzerConfig {
public:
float GetCallTimeSec(int64_t timestamp_us) const {
int64_t offset = normalize_time_ ? begin_time_ : 0;
return static_cast<float>(timestamp_us - offset) / 1000000;
}
float CallBeginTimeSec() const { return GetCallTimeSec(begin_time_); }
float CallEndTimeSec() const { return GetCallTimeSec(end_time_); }
// 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_;
bool normalize_time_;
};
class EventLogAnalyzer {
public:
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// 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 ParsedRtcEventLog& log, bool normalize_time);
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void CreatePacketGraph(PacketDirection direction, Plot* plot);
void CreateRtcpTypeGraph(PacketDirection direction, Plot* plot);
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void CreateAccumulatedPacketsGraph(PacketDirection direction, Plot* plot);
void CreatePlayoutGraph(Plot* plot);
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void CreateAudioLevelGraph(PacketDirection direction, Plot* plot);
void CreateSequenceNumberGraph(Plot* plot);
void CreateIncomingPacketLossGraph(Plot* plot);
void CreateIncomingDelayGraph(Plot* plot);
void CreateFractionLossGraph(Plot* plot);
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void CreateTotalIncomingBitrateGraph(Plot* plot);
void CreateTotalOutgoingBitrateGraph(Plot* plot,
bool show_detector_state = false,
bool show_alr_state = false);
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void CreateStreamBitrateGraph(PacketDirection direction, Plot* plot);
void CreateBitrateAllocationGraph(PacketDirection direction, Plot* plot);
void CreateGoogCcSimulationGraph(Plot* plot);
void CreateSendSideBweSimulationGraph(Plot* plot);
void CreateReceiveSideBweSimulationGraph(Plot* plot);
void CreateNetworkDelayFeedbackGraph(Plot* plot);
void CreatePacerDelayGraph(Plot* plot);
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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 CreateDtlsTransportStateGraph(Plot* plot);
void CreateDtlsWritableStateGraph(Plot* plot);
void CreateTriageNotifications();
void PrintNotifications(FILE* file);
private:
struct LayerDescription {
LayerDescription(uint32_t ssrc,
uint8_t spatial_layer,
uint8_t temporal_layer)
: ssrc(ssrc),
spatial_layer(spatial_layer),
temporal_layer(temporal_layer) {}
bool operator<(const LayerDescription& other) const {
if (ssrc != other.ssrc)
return ssrc < other.ssrc;
if (spatial_layer != other.spatial_layer)
return spatial_layer < other.spatial_layer;
return temporal_layer < other.temporal_layer;
}
uint32_t ssrc;
uint8_t spatial_layer;
uint8_t temporal_layer;
};
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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();
}
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}
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();
}
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}
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;
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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();
}
Revert "Create new API for RtcEventLogParser." This reverts commit 9e336ec0b8a77c3461d13677cff3563c11c88daa. Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming. Original change's description: > Create new API for RtcEventLogParser. > > The new API stores events gathered by event type. For example, it is > possible to ask fo 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: I8e7168302beb69b2e163a097a2a142b86dd4a26b > Reviewed-on: https://webrtc-review.googlesource.com/60865 > Reviewed-by: Minyue Li <minyue@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23015} TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8111 Reviewed-on: https://webrtc-review.googlesource.com/72500 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23026}
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std::string GetLayerName(LayerDescription layer) const {
char buffer[100];
rtc::SimpleStringBuilder name(buffer);
name << "SSRC " << layer.ssrc << " sl " << layer.spatial_layer << ", tl "
<< layer.temporal_layer;
return name.str();
}
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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_;
// 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.
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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_;
AnalyzerConfig config_;
};
} // namespace webrtc
#endif // RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_