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Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/rtc_tools/rtc_event_log_visualizer/main.cc | 653 |
1 files changed, 653 insertions, 0 deletions
diff --git a/third_party/libwebrtc/rtc_tools/rtc_event_log_visualizer/main.cc b/third_party/libwebrtc/rtc_tools/rtc_event_log_visualizer/main.cc new file mode 100644 index 0000000000..4988808cab --- /dev/null +++ b/third_party/libwebrtc/rtc_tools/rtc_event_log_visualizer/main.cc @@ -0,0 +1,653 @@ +/* + * 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 <cstdio> +#include <fstream> +#include <iostream> +#include <map> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/algorithm/container.h" +#include "absl/flags/flag.h" +#include "absl/flags/parse.h" +#include "absl/flags/usage.h" +#include "absl/flags/usage_config.h" +#include "absl/strings/match.h" +#include "api/neteq/neteq.h" +#include "api/rtc_event_log/rtc_event_log.h" +#include "logging/rtc_event_log/rtc_event_log_parser.h" +#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_tools/rtc_event_log_visualizer/alerts.h" +#include "rtc_tools/rtc_event_log_visualizer/analyze_audio.h" +#include "rtc_tools/rtc_event_log_visualizer/analyzer.h" +#include "rtc_tools/rtc_event_log_visualizer/conversational_speech_en.h" +#include "rtc_tools/rtc_event_log_visualizer/plot_base.h" +#include "system_wrappers/include/field_trial.h" + +ABSL_FLAG(std::string, + plot, + "default", + "A comma separated list of plot names. See --list_plots for valid " + "options."); + +ABSL_FLAG( + std::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 \"/\""); +ABSL_FLAG(std::string, + wav_filename, + "", + "Path to wav file used for simulation of jitter buffer"); + +ABSL_FLAG(bool, + show_detector_state, + false, + "Show the state of the delay based BWE detector on the total " + "bitrate graph"); + +ABSL_FLAG(bool, + show_alr_state, + false, + "Show the state ALR state on the total bitrate graph"); + +ABSL_FLAG(bool, + show_link_capacity, + true, + "Show the lower and upper link capacity on the outgoing bitrate " + "graph"); + +ABSL_FLAG(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."); + +ABSL_FLAG(bool, + print_triage_alerts, + true, + "Print triage alerts, i.e. a list of potential problems."); + +ABSL_FLAG(bool, + normalize_time, + true, + "Normalize the log timestamps so that the call starts at time 0."); + +ABSL_FLAG(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."); + +ABSL_FLAG(bool, + protobuf_output, + false, + "Output charts as protobuf instead of python code."); + +ABSL_FLAG(bool, + list_plots, + false, + "List of registered plots (for use with the --plot flag)"); + +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_; +}; + +bool ContainsHelppackageFlags(absl::string_view filename) { + return absl::EndsWith(filename, "main.cc"); +} + +} // namespace + +int main(int argc, char* argv[]) { + absl::SetProgramUsageMessage( + "A tool for visualizing WebRTC event logs.\n" + "Example usage:\n" + "./event_log_visualizer <logfile> | python\n"); + absl::FlagsUsageConfig flag_config; + flag_config.contains_help_flags = &ContainsHelppackageFlags; + absl::SetFlagsUsageConfig(flag_config); + std::vector<char*> args = absl::ParseCommandLine(argc, argv); + + // Print RTC_LOG warnings and errors even in release builds. + if (rtc::LogMessage::GetLogToDebug() > rtc::LS_WARNING) { + rtc::LogMessage::LogToDebug(rtc::LS_WARNING); + } + rtc::LogMessage::SetLogToStderr(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_preemptive_rate", + "simulated_neteq_accelerate_rate", "simulated_neteq_speech_expand_rate", + "simulated_neteq_expand_rate"}}}; + + std::vector<std::string> plot_flags = + StrSplit(absl::GetFlag(FLAGS_plot), ","); + + // InitFieldTrialsFromString stores the char*, so the char array must outlive + // the application. + const std::string field_trials = absl::GetFlag(FLAGS_force_fieldtrials); + webrtc::field_trial::InitFieldTrialsFromString(field_trials.c_str()); + + webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions header_extensions = + webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions::kDontParse; + if (absl::GetFlag(FLAGS_parse_unconfigured_header_extensions)) { + header_extensions = webrtc::ParsedRtcEventLog:: + UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig; + } + webrtc::ParsedRtcEventLog parsed_log(header_extensions, + /*allow_incomplete_logs*/ true); + + if (args.size() == 2) { + std::string filename = args[1]; + auto status = parsed_log.ParseFile(filename); + if (!status.ok()) { + std::cerr << "Failed to parse " << filename << ": " << status.message() + << std::endl; + return -1; + } + } + + webrtc::AnalyzerConfig config; + config.window_duration_ = webrtc::TimeDelta::Millis(250); + config.step_ = webrtc::TimeDelta::Millis(10); + if (!parsed_log.start_log_events().empty()) { + config.rtc_to_utc_offset_ = parsed_log.start_log_events()[0].utc_time() - + parsed_log.start_log_events()[0].log_time(); + } + config.normalize_time_ = absl::GetFlag(FLAGS_normalize_time); + config.begin_time_ = parsed_log.first_timestamp(); + config.end_time_ = parsed_log.last_timestamp(); + if (config.end_time_ < config.begin_time_) { + RTC_LOG(LS_WARNING) << "Log end time " << config.end_time_ + << " not after begin time " << config.begin_time_ + << ". Nothing to analyze. Is the log broken?"; + return -1; + } + + webrtc::EventLogAnalyzer analyzer(parsed_log, config); + webrtc::PlotCollection collection; + collection.SetCallTimeToUtcOffsetMs(config.CallTimeToUtcOffsetMs()); + + 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("incoming_packet_rate", [&](Plot* plot) { + analyzer.CreatePacketRateGraph(webrtc::kIncomingPacket, plot); + }); + plots.RegisterPlot("outgoing_packet_rate", [&](Plot* plot) { + analyzer.CreatePacketRateGraph(webrtc::kOutgoingPacket, plot); + }); + plots.RegisterPlot("total_incoming_packet_rate", [&](Plot* plot) { + analyzer.CreateTotalPacketRateGraph(webrtc::kIncomingPacket, plot); + }); + plots.RegisterPlot("total_outgoing_packet_rate", [&](Plot* plot) { + analyzer.CreateTotalPacketRateGraph(webrtc::kOutgoingPacket, plot); + }); + plots.RegisterPlot("audio_playout", + [&](Plot* plot) { analyzer.CreatePlayoutGraph(plot); }); + + plots.RegisterPlot("neteq_set_minimum_delay", [&](Plot* plot) { + analyzer.CreateNetEqSetMinimumDelay(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, absl::GetFlag(FLAGS_show_detector_state), + absl::GetFlag(FLAGS_show_alr_state), + absl::GetFlag(FLAGS_show_link_capacity)); + }); + 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) { + CreateAudioEncoderTargetBitrateGraph(parsed_log, config, plot); + }); + plots.RegisterPlot("audio_encoder_frame_length", [&](Plot* plot) { + CreateAudioEncoderFrameLengthGraph(parsed_log, config, plot); + }); + plots.RegisterPlot("audio_encoder_packet_loss", [&](Plot* plot) { + CreateAudioEncoderPacketLossGraph(parsed_log, config, plot); + }); + plots.RegisterPlot("audio_encoder_fec", [&](Plot* plot) { + CreateAudioEncoderEnableFecGraph(parsed_log, config, plot); + }); + plots.RegisterPlot("audio_encoder_dtx", [&](Plot* plot) { + CreateAudioEncoderEnableDtxGraph(parsed_log, config, plot); + }); + plots.RegisterPlot("audio_encoder_num_channels", [&](Plot* plot) { + CreateAudioEncoderNumChannelsGraph(parsed_log, config, 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; + bool has_generated_wav_file = false; + if (!absl::GetFlag(FLAGS_wav_filename).empty()) { + wav_path = absl::GetFlag(FLAGS_wav_filename); + } else { + // TODO(bugs.webrtc.org/14248): Remove the need to generate a file + // and read the file directly from memory. + wav_path = std::tmpnam(nullptr); + std::ofstream out_wav_file(wav_path); + out_wav_file.write( + reinterpret_cast<char*>(&webrtc::conversational_speech_en_wav[0]), + webrtc::conversational_speech_en_wav_len); + has_generated_wav_file = true; + } + absl::optional<webrtc::NetEqStatsGetterMap> neteq_stats; + + plots.RegisterPlot("simulated_neteq_expand_rate", [&](Plot* plot) { + if (!neteq_stats) { + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + } + webrtc::CreateNetEqNetworkStatsGraph( + parsed_log, config, *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 = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + } + webrtc::CreateNetEqNetworkStatsGraph( + parsed_log, config, *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 = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + } + webrtc::CreateNetEqNetworkStatsGraph( + parsed_log, config, *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 = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + } + webrtc::CreateNetEqNetworkStatsGraph( + parsed_log, config, *neteq_stats, + [](const webrtc::NetEqNetworkStatistics& stats) { + return stats.preemptive_rate / 16384.f; + }, + "Preemptive rate", plot); + }); + + plots.RegisterPlot("simulated_neteq_concealment_events", [&](Plot* plot) { + if (!neteq_stats) { + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + } + webrtc::CreateNetEqLifetimeStatsGraph( + parsed_log, config, *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 = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + } + webrtc::CreateNetEqNetworkStatsGraph( + parsed_log, config, *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 (absl::GetFlag(FLAGS_list_plots)) { + 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; + std::cerr << " all = every registered plot" << 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; + } + if (args.size() != 2) { + // Print usage information. + std::cerr << absl::ProgramUsageMessage(); + return 1; + } + + 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 = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + } + for (webrtc::NetEqStatsGetterMap::const_iterator it = neteq_stats->cbegin(); + it != neteq_stats->cend(); ++it) { + webrtc::CreateAudioJitterBufferGraph(parsed_log, config, it->first, + it->second.get(), + collection.AppendNewPlot()); + } + } + + if (absl::GetFlag(FLAGS_protobuf_output)) { + webrtc::analytics::ChartCollection proto_charts; + collection.ExportProtobuf(&proto_charts); + std::cout << proto_charts.SerializeAsString(); + } else { + collection.PrintPythonCode(absl::GetFlag(FLAGS_shared_xaxis)); + } + + if (absl::GetFlag(FLAGS_print_triage_alerts)) { + webrtc::TriageHelper triage_alerts(config); + triage_alerts.AnalyzeLog(parsed_log); + triage_alerts.Print(stderr); + } + + // TODO(bugs.webrtc.org/14248): Remove the need to generate a file + // and read the file directly from memory. + if (has_generated_wav_file) { + RTC_CHECK_EQ(std::remove(wav_path.c_str()), 0) + << "Failed to remove " << wav_path; + } + return 0; +} |