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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
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+++ 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;
+}