blob: 9fdc554cb53d2aeea1ed77ec092cf39c006a44cf [file] [log] [blame]
/*
* 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;
}