| /* |
| * 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. |
| */ |
| |
| #include <stdio.h> |
| #include <string.h> |
| |
| #include <iostream> |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "logging/rtc_event_log/rtc_event_log_parser.h" |
| #include "modules/audio_coding/neteq/include/neteq.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/flags.h" |
| #include "rtc_tools/event_log_visualizer/analyzer.h" |
| #include "rtc_tools/event_log_visualizer/plot_base.h" |
| #include "rtc_tools/event_log_visualizer/plot_protobuf.h" |
| #include "rtc_tools/event_log_visualizer/plot_python.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "test/field_trial.h" |
| #include "test/testsupport/file_utils.h" |
| |
| WEBRTC_DEFINE_string( |
| plot, |
| "default", |
| "A comma separated list of plot names. See below for valid options."); |
| |
| WEBRTC_DEFINE_string( |
| force_fieldtrials, |
| "", |
| "Field trials control experimental feature code which can be forced. " |
| "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/" |
| " will assign the group Enabled to field trial WebRTC-FooFeature. Multiple " |
| "trials are separated by \"/\""); |
| WEBRTC_DEFINE_string(wav_filename, |
| "", |
| "Path to wav file used for simulation of jitter buffer"); |
| WEBRTC_DEFINE_bool(help, false, "prints this message"); |
| |
| WEBRTC_DEFINE_bool( |
| show_detector_state, |
| false, |
| "Show the state of the delay based BWE detector on the total " |
| "bitrate graph"); |
| |
| WEBRTC_DEFINE_bool(show_alr_state, |
| false, |
| "Show the state ALR state on the total bitrate graph"); |
| |
| WEBRTC_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."); |
| |
| WEBRTC_DEFINE_bool(print_triage_alerts, |
| false, |
| "Print triage alerts, i.e. a list of potential problems."); |
| |
| WEBRTC_DEFINE_bool( |
| normalize_time, |
| true, |
| "Normalize the log timestamps so that the call starts at time 0."); |
| |
| WEBRTC_DEFINE_bool(shared_xaxis, |
| false, |
| "Share x-axis between all plots so that zooming in one plot " |
| "updates all the others too. A downside is that certain " |
| "operations like panning become much slower."); |
| |
| WEBRTC_DEFINE_bool(protobuf_output, |
| false, |
| "Output charts as protobuf instead of python code."); |
| |
| using webrtc::Plot; |
| |
| namespace { |
| std::vector<std::string> StrSplit(const std::string& s, |
| const std::string& delimiter) { |
| std::vector<std::string> v; |
| size_t pos = 0; |
| while (pos < s.length()) { |
| const std::string token = s.substr(pos, s.find(delimiter, pos) - pos); |
| pos += token.length() + delimiter.length(); |
| v.push_back(token); |
| } |
| return v; |
| } |
| |
| struct PlotDeclaration { |
| PlotDeclaration(const std::string& label, std::function<void(Plot*)> f) |
| : label(label), enabled(false), plot_func(f) {} |
| const std::string label; |
| bool enabled; |
| // TODO(terelius): Add a help text/explanation. |
| const std::function<void(Plot*)> plot_func; |
| }; |
| |
| class PlotMap { |
| public: |
| void RegisterPlot(const std::string& label, std::function<void(Plot*)> f) { |
| for (const auto& plot : plots_) { |
| RTC_DCHECK(plot.label != label) |
| << "Can't use the same label for multiple plots"; |
| } |
| plots_.push_back({label, f}); |
| } |
| |
| bool EnablePlotsByFlags( |
| const std::vector<std::string>& flags, |
| const std::map<std::string, std::vector<std::string>>& flag_aliases) { |
| bool status = true; |
| for (const std::string& flag : flags) { |
| auto alias_it = flag_aliases.find(flag); |
| if (alias_it != flag_aliases.end()) { |
| const auto& replacements = alias_it->second; |
| for (const auto& replacement : replacements) { |
| status &= EnablePlotByFlag(replacement); |
| } |
| } else { |
| status &= EnablePlotByFlag(flag); |
| } |
| } |
| return status; |
| } |
| |
| void EnableAllPlots() { |
| for (auto& plot : plots_) { |
| plot.enabled = true; |
| } |
| } |
| |
| std::vector<PlotDeclaration>::iterator begin() { return plots_.begin(); } |
| std::vector<PlotDeclaration>::iterator end() { return plots_.end(); } |
| |
| private: |
| bool EnablePlotByFlag(const std::string& flag) { |
| for (auto& plot : plots_) { |
| if (plot.label == flag) { |
| plot.enabled = true; |
| return true; |
| } |
| } |
| if (flag == "simulated_neteq_jitter_buffer_delay") { |
| // This flag is handled separately. |
| return true; |
| } |
| std::cerr << "Unrecognized plot name \'" << flag << "\'. Aborting." |
| << std::endl; |
| return false; |
| } |
| |
| std::vector<PlotDeclaration> plots_; |
| }; |
| } // namespace |
| |
| int main(int argc, char* argv[]) { |
| std::string program_name = argv[0]; |
| std::string usage = |
| "A tool for visualizing WebRTC event logs.\n" |
| "Example usage:\n" + |
| program_name + " <logfile> | python\n" + "Run " + program_name + |
| " --help for a list of command line options\n"; |
| |
| rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true); |
| |
| // Flag replacements |
| std::map<std::string, std::vector<std::string>> flag_aliases = { |
| {"default", |
| {"incoming_delay", "incoming_loss_rate", "incoming_bitrate", |
| "outgoing_bitrate," |
| "incoming_stream_bitrate", |
| "outgoing_stream_bitrate", "network_delay_feedback", |
| "fraction_loss_feedback"}}, |
| {"sendside_bwe", |
| {"outgoing_packet_sizes", "outgoing_bitrate", "outgoing_stream_bitrate", |
| "simulated_sendside_bwe", "network_delay_feedback", |
| "fraction_loss_feedback"}}, |
| {"receiveside_bwe", |
| {"incoming_packet_sizes", "incoming_delay", "incoming_loss_rate", |
| "incoming_" |
| "bitrate", |
| "incoming_stream_bitrate", "simulated_receiveside_bwe"}}, |
| {"rtcp_details", |
| {"incoming_rtcp_fraction_lost", "outgoing_rtcp_fraction_lost", |
| "incoming_rtcp_" |
| "cumulative_lost", |
| "outgoing_rtcp_cumulative_lost", |
| "incoming_rtcp_highest_" |
| "seq_number", |
| "outgoing_rtcp_highest_seq_number", |
| "incoming_rtcp_delay_since_" |
| "last_sr", |
| "outgoing_rtcp_delay_since_last_sr"}}, |
| {"simulated_neteq_stats", |
| {"simulated_neteq_jitter_buffer_delay", |
| "simulated_neteq_preferred_buffer_" |
| "size", |
| "simulated_neteq_" |
| "concealment_events", |
| "simulated_neteq_packet_loss_rate", |
| "simulated_neteq_" |
| "preemptive_rate", |
| "simulated_neteq_" |
| "accelerate_rate", |
| "simulated_neteq_speech_expand_rate", |
| "simulated_neteq_" |
| "expand_rate"}}}; |
| |
| std::vector<std::string> plot_flags = StrSplit(FLAG_plot, ","); |
| |
| // InitFieldTrialsFromString stores the char*, so the char array must outlive |
| // the application. |
| webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials); |
| |
| webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions header_extensions = |
| webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions::kDontParse; |
| if (FLAG_parse_unconfigured_header_extensions) { |
| header_extensions = webrtc::ParsedRtcEventLog:: |
| UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig; |
| } |
| webrtc::ParsedRtcEventLog parsed_log(header_extensions); |
| |
| if (argc == 2) { |
| std::string filename = argv[1]; |
| if (!parsed_log.ParseFile(filename)) { |
| std::cerr << "Could not parse the entire log file." << std::endl; |
| std::cerr << "Only the parsable events will be analyzed." << std::endl; |
| } |
| } |
| |
| webrtc::EventLogAnalyzer analyzer(parsed_log, FLAG_normalize_time); |
| std::unique_ptr<webrtc::PlotCollection> collection; |
| if (FLAG_protobuf_output) { |
| collection.reset(new webrtc::ProtobufPlotCollection()); |
| } else { |
| collection.reset(new webrtc::PythonPlotCollection(FLAG_shared_xaxis)); |
| } |
| |
| PlotMap plots; |
| plots.RegisterPlot("incoming_packet_sizes", [&](Plot* plot) { |
| analyzer.CreatePacketGraph(webrtc::kIncomingPacket, plot); |
| }); |
| |
| plots.RegisterPlot("outgoing_packet_sizes", [&](Plot* plot) { |
| analyzer.CreatePacketGraph(webrtc::kOutgoingPacket, plot); |
| }); |
| plots.RegisterPlot("incoming_rtcp_types", [&](Plot* plot) { |
| analyzer.CreateRtcpTypeGraph(webrtc::kIncomingPacket, plot); |
| }); |
| plots.RegisterPlot("outgoing_rtcp_types", [&](Plot* plot) { |
| analyzer.CreateRtcpTypeGraph(webrtc::kOutgoingPacket, plot); |
| }); |
| plots.RegisterPlot("incoming_packet_count", [&](Plot* plot) { |
| analyzer.CreateAccumulatedPacketsGraph(webrtc::kIncomingPacket, plot); |
| }); |
| plots.RegisterPlot("outgoing_packet_count", [&](Plot* plot) { |
| analyzer.CreateAccumulatedPacketsGraph(webrtc::kOutgoingPacket, plot); |
| }); |
| plots.RegisterPlot("audio_playout", |
| [&](Plot* plot) { analyzer.CreatePlayoutGraph(plot); }); |
| plots.RegisterPlot("incoming_audio_level", [&](Plot* plot) { |
| analyzer.CreateAudioLevelGraph(webrtc::kIncomingPacket, plot); |
| }); |
| plots.RegisterPlot("outgoing_audio_level", [&](Plot* plot) { |
| analyzer.CreateAudioLevelGraph(webrtc::kOutgoingPacket, plot); |
| }); |
| plots.RegisterPlot("incoming_sequence_number_delta", [&](Plot* plot) { |
| analyzer.CreateSequenceNumberGraph(plot); |
| }); |
| plots.RegisterPlot("incoming_delay", [&](Plot* plot) { |
| analyzer.CreateIncomingDelayGraph(plot); |
| }); |
| plots.RegisterPlot("incoming_loss_rate", [&](Plot* plot) { |
| analyzer.CreateIncomingPacketLossGraph(plot); |
| }); |
| plots.RegisterPlot("incoming_bitrate", [&](Plot* plot) { |
| analyzer.CreateTotalIncomingBitrateGraph(plot); |
| }); |
| plots.RegisterPlot("outgoing_bitrate", [&](Plot* plot) { |
| analyzer.CreateTotalOutgoingBitrateGraph(plot, FLAG_show_detector_state, |
| FLAG_show_alr_state); |
| }); |
| plots.RegisterPlot("incoming_stream_bitrate", [&](Plot* plot) { |
| analyzer.CreateStreamBitrateGraph(webrtc::kIncomingPacket, plot); |
| }); |
| plots.RegisterPlot("outgoing_stream_bitrate", [&](Plot* plot) { |
| analyzer.CreateStreamBitrateGraph(webrtc::kOutgoingPacket, plot); |
| }); |
| plots.RegisterPlot("incoming_layer_bitrate_allocation", [&](Plot* plot) { |
| analyzer.CreateBitrateAllocationGraph(webrtc::kIncomingPacket, plot); |
| }); |
| plots.RegisterPlot("outgoing_layer_bitrate_allocation", [&](Plot* plot) { |
| analyzer.CreateBitrateAllocationGraph(webrtc::kOutgoingPacket, plot); |
| }); |
| plots.RegisterPlot("simulated_receiveside_bwe", [&](Plot* plot) { |
| analyzer.CreateReceiveSideBweSimulationGraph(plot); |
| }); |
| plots.RegisterPlot("simulated_sendside_bwe", [&](Plot* plot) { |
| analyzer.CreateSendSideBweSimulationGraph(plot); |
| }); |
| plots.RegisterPlot("simulated_goog_cc", [&](Plot* plot) { |
| analyzer.CreateGoogCcSimulationGraph(plot); |
| }); |
| plots.RegisterPlot("network_delay_feedback", [&](Plot* plot) { |
| analyzer.CreateNetworkDelayFeedbackGraph(plot); |
| }); |
| plots.RegisterPlot("fraction_loss_feedback", [&](Plot* plot) { |
| analyzer.CreateFractionLossGraph(plot); |
| }); |
| plots.RegisterPlot("incoming_timestamps", [&](Plot* plot) { |
| analyzer.CreateTimestampGraph(webrtc::kIncomingPacket, plot); |
| }); |
| plots.RegisterPlot("outgoing_timestamps", [&](Plot* plot) { |
| analyzer.CreateTimestampGraph(webrtc::kOutgoingPacket, plot); |
| }); |
| |
| auto GetFractionLost = [](const webrtc::rtcp::ReportBlock& block) -> float { |
| return static_cast<double>(block.fraction_lost()) / 256 * 100; |
| }; |
| plots.RegisterPlot("incoming_rtcp_fraction_lost", [&](Plot* plot) { |
| analyzer.CreateSenderAndReceiverReportPlot( |
| webrtc::kIncomingPacket, GetFractionLost, |
| "Fraction lost (incoming RTCP)", "Loss rate (percent)", plot); |
| }); |
| plots.RegisterPlot("outgoing_rtcp_fraction_lost", [&](Plot* plot) { |
| analyzer.CreateSenderAndReceiverReportPlot( |
| webrtc::kOutgoingPacket, GetFractionLost, |
| "Fraction lost (outgoing RTCP)", "Loss rate (percent)", plot); |
| }); |
| auto GetCumulativeLost = [](const webrtc::rtcp::ReportBlock& block) -> float { |
| return block.cumulative_lost_signed(); |
| }; |
| plots.RegisterPlot("incoming_rtcp_cumulative_lost", [&](Plot* plot) { |
| analyzer.CreateSenderAndReceiverReportPlot( |
| webrtc::kIncomingPacket, GetCumulativeLost, |
| "Cumulative lost packets (incoming RTCP)", "Packets", plot); |
| }); |
| plots.RegisterPlot("outgoing_rtcp_cumulative_lost", [&](Plot* plot) { |
| analyzer.CreateSenderAndReceiverReportPlot( |
| webrtc::kOutgoingPacket, GetCumulativeLost, |
| "Cumulative lost packets (outgoing RTCP)", "Packets", plot); |
| }); |
| |
| auto GetHighestSeqNumber = |
| [](const webrtc::rtcp::ReportBlock& block) -> float { |
| return block.extended_high_seq_num(); |
| }; |
| plots.RegisterPlot("incoming_rtcp_highest_seq_number", [&](Plot* plot) { |
| analyzer.CreateSenderAndReceiverReportPlot( |
| webrtc::kIncomingPacket, GetHighestSeqNumber, |
| "Highest sequence number (incoming RTCP)", "Sequence number", plot); |
| }); |
| plots.RegisterPlot("outgoing_rtcp_highest_seq_number", [&](Plot* plot) { |
| analyzer.CreateSenderAndReceiverReportPlot( |
| webrtc::kOutgoingPacket, GetHighestSeqNumber, |
| "Highest sequence number (outgoing RTCP)", "Sequence number", plot); |
| }); |
| |
| auto DelaySinceLastSr = [](const webrtc::rtcp::ReportBlock& block) -> float { |
| return static_cast<double>(block.delay_since_last_sr()) / 65536; |
| }; |
| plots.RegisterPlot("incoming_rtcp_delay_since_last_sr", [&](Plot* plot) { |
| analyzer.CreateSenderAndReceiverReportPlot( |
| webrtc::kIncomingPacket, DelaySinceLastSr, |
| "Delay since last received sender report (incoming RTCP)", "Time (s)", |
| plot); |
| }); |
| plots.RegisterPlot("outgoing_rtcp_delay_since_last_sr", [&](Plot* plot) { |
| analyzer.CreateSenderAndReceiverReportPlot( |
| webrtc::kOutgoingPacket, DelaySinceLastSr, |
| "Delay since last received sender report (outgoing RTCP)", "Time (s)", |
| plot); |
| }); |
| |
| plots.RegisterPlot("pacer_delay", |
| [&](Plot* plot) { analyzer.CreatePacerDelayGraph(plot); }); |
| plots.RegisterPlot("audio_encoder_bitrate", [&](Plot* plot) { |
| analyzer.CreateAudioEncoderTargetBitrateGraph(plot); |
| }); |
| plots.RegisterPlot("audio_encoder_frame_length", [&](Plot* plot) { |
| analyzer.CreateAudioEncoderFrameLengthGraph(plot); |
| }); |
| plots.RegisterPlot("audio_encoder_packet_loss", [&](Plot* plot) { |
| analyzer.CreateAudioEncoderPacketLossGraph(plot); |
| }); |
| plots.RegisterPlot("audio_encoder_fec", [&](Plot* plot) { |
| analyzer.CreateAudioEncoderEnableFecGraph(plot); |
| }); |
| plots.RegisterPlot("audio_encoder_dtx", [&](Plot* plot) { |
| analyzer.CreateAudioEncoderEnableDtxGraph(plot); |
| }); |
| plots.RegisterPlot("audio_encoder_num_channels", [&](Plot* plot) { |
| analyzer.CreateAudioEncoderNumChannelsGraph(plot); |
| }); |
| |
| plots.RegisterPlot("ice_candidate_pair_config", [&](Plot* plot) { |
| analyzer.CreateIceCandidatePairConfigGraph(plot); |
| }); |
| plots.RegisterPlot("ice_connectivity_check", [&](Plot* plot) { |
| analyzer.CreateIceConnectivityCheckGraph(plot); |
| }); |
| plots.RegisterPlot("dtls_transport_state", [&](Plot* plot) { |
| analyzer.CreateDtlsTransportStateGraph(plot); |
| }); |
| plots.RegisterPlot("dtls_writable_state", [&](Plot* plot) { |
| analyzer.CreateDtlsWritableStateGraph(plot); |
| }); |
| |
| std::string wav_path; |
| if (FLAG_wav_filename[0] != '\0') { |
| wav_path = FLAG_wav_filename; |
| } else { |
| wav_path = webrtc::test::ResourcePath( |
| "audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav"); |
| } |
| absl::optional<webrtc::EventLogAnalyzer::NetEqStatsGetterMap> neteq_stats; |
| |
| plots.RegisterPlot("simulated_neteq_expand_rate", [&](Plot* plot) { |
| if (!neteq_stats) { |
| neteq_stats = analyzer.SimulateNetEq(wav_path, 48000); |
| } |
| analyzer.CreateNetEqNetworkStatsGraph( |
| *neteq_stats, |
| [](const webrtc::NetEqNetworkStatistics& stats) { |
| return stats.expand_rate / 16384.f; |
| }, |
| "Expand rate", plot); |
| }); |
| |
| plots.RegisterPlot("simulated_neteq_speech_expand_rate", [&](Plot* plot) { |
| if (!neteq_stats) { |
| neteq_stats = analyzer.SimulateNetEq(wav_path, 48000); |
| } |
| analyzer.CreateNetEqNetworkStatsGraph( |
| *neteq_stats, |
| [](const webrtc::NetEqNetworkStatistics& stats) { |
| return stats.speech_expand_rate / 16384.f; |
| }, |
| "Speech expand rate", plot); |
| }); |
| |
| plots.RegisterPlot("simulated_neteq_accelerate_rate", [&](Plot* plot) { |
| if (!neteq_stats) { |
| neteq_stats = analyzer.SimulateNetEq(wav_path, 48000); |
| } |
| analyzer.CreateNetEqNetworkStatsGraph( |
| *neteq_stats, |
| [](const webrtc::NetEqNetworkStatistics& stats) { |
| return stats.accelerate_rate / 16384.f; |
| }, |
| "Accelerate rate", plot); |
| }); |
| |
| plots.RegisterPlot("simulated_neteq_preemptive_rate", [&](Plot* plot) { |
| if (!neteq_stats) { |
| neteq_stats = analyzer.SimulateNetEq(wav_path, 48000); |
| } |
| analyzer.CreateNetEqNetworkStatsGraph( |
| *neteq_stats, |
| [](const webrtc::NetEqNetworkStatistics& stats) { |
| return stats.preemptive_rate / 16384.f; |
| }, |
| "Preemptive rate", plot); |
| }); |
| |
| plots.RegisterPlot("simulated_neteq_packet_loss_rate", [&](Plot* plot) { |
| if (!neteq_stats) { |
| neteq_stats = analyzer.SimulateNetEq(wav_path, 48000); |
| } |
| analyzer.CreateNetEqNetworkStatsGraph( |
| *neteq_stats, |
| [](const webrtc::NetEqNetworkStatistics& stats) { |
| return stats.packet_loss_rate / 16384.f; |
| }, |
| "Packet loss rate", plot); |
| }); |
| |
| plots.RegisterPlot("simulated_neteq_concealment_events", [&](Plot* plot) { |
| if (!neteq_stats) { |
| neteq_stats = analyzer.SimulateNetEq(wav_path, 48000); |
| } |
| analyzer.CreateNetEqLifetimeStatsGraph( |
| *neteq_stats, |
| [](const webrtc::NetEqLifetimeStatistics& stats) { |
| return static_cast<float>(stats.concealment_events); |
| }, |
| "Concealment events", plot); |
| }); |
| |
| plots.RegisterPlot("simulated_neteq_preferred_buffer_size", [&](Plot* plot) { |
| if (!neteq_stats) { |
| neteq_stats = analyzer.SimulateNetEq(wav_path, 48000); |
| } |
| analyzer.CreateNetEqNetworkStatsGraph( |
| *neteq_stats, |
| [](const webrtc::NetEqNetworkStatistics& stats) { |
| return stats.preferred_buffer_size_ms; |
| }, |
| "Preferred buffer size (ms)", plot); |
| }); |
| |
| if (absl::c_find(plot_flags, "all") != plot_flags.end()) { |
| plots.EnableAllPlots(); |
| // Treated separately since it isn't registered like the other plots. |
| plot_flags.push_back("simulated_neteq_jitter_buffer_delay"); |
| } else { |
| bool success = plots.EnablePlotsByFlags(plot_flags, flag_aliases); |
| if (!success) { |
| return 1; |
| } |
| } |
| |
| if (argc != 2 || FLAG_help) { |
| // Print usage information. |
| std::cerr << usage; |
| if (FLAG_help) { |
| rtc::FlagList::Print(nullptr, false); |
| std::cerr << "List of registered plots (for use with the --plot flag):" |
| << std::endl; |
| for (const auto& plot : plots) { |
| // TODO(terelius): Also print a help text. |
| std::cerr << " " << plot.label << std::endl; |
| } |
| // The following flag doesn't fit the model used for the other plots. |
| std::cerr << "simulated_neteq_jitter_buffer_delay" << std::endl; |
| std::cerr << "List of plot aliases (for use with the --plot flag):" |
| << std::endl; |
| for (const auto& alias : flag_aliases) { |
| std::cerr << " " << alias.first << " = "; |
| for (const auto& replacement : alias.second) { |
| std::cerr << replacement << ","; |
| } |
| std::cerr << std::endl; |
| } |
| } |
| return 0; |
| } |
| |
| for (const auto& plot : plots) { |
| if (plot.enabled) { |
| Plot* output = collection->AppendNewPlot(); |
| plot.plot_func(output); |
| output->SetId(plot.label); |
| } |
| } |
| |
| // The model we use for registering plots assumes that the each plot label |
| // can be mapped to a lambda that will produce exactly one plot. The |
| // simulated_neteq_jitter_buffer_delay plot doesn't fit this model since it |
| // creates multiple plots, and would need some state kept between the lambda |
| // calls. |
| if (absl::c_find(plot_flags, "simulated_neteq_jitter_buffer_delay") != |
| plot_flags.end()) { |
| if (!neteq_stats) { |
| neteq_stats = analyzer.SimulateNetEq(wav_path, 48000); |
| } |
| for (webrtc::EventLogAnalyzer::NetEqStatsGetterMap::const_iterator it = |
| neteq_stats->cbegin(); |
| it != neteq_stats->cend(); ++it) { |
| analyzer.CreateAudioJitterBufferGraph(it->first, it->second.get(), |
| collection->AppendNewPlot()); |
| } |
| } |
| |
| collection->Draw(); |
| |
| if (FLAG_print_triage_alerts) { |
| analyzer.CreateTriageNotifications(); |
| analyzer.PrintNotifications(stderr); |
| } |
| |
| return 0; |
| } |
| |
