| /* |
| * 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(); |
| }; |
| 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; |
| } |
