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webrtc/rtc_tools/event_log_visualizer/main.cc
Steve Anton 10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00

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/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stdio.h>
#include <string.h>
#include <iostream>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "rtc_tools/event_log_visualizer/analyzer.h"
#include "rtc_tools/event_log_visualizer/plot_base.h"
#include "rtc_tools/event_log_visualizer/plot_protobuf.h"
#include "rtc_tools/event_log_visualizer/plot_python.h"
#include "system_wrappers/include/field_trial.h"
#include "test/field_trial.h"
#include "test/testsupport/file_utils.h"
WEBRTC_DEFINE_string(
plot_profile,
"default",
"A profile that selects a certain subset of the plots. Currently "
"defined profiles are \"all\", \"none\", \"sendside_bwe\","
"\"receiveside_bwe\" and \"default\"");
WEBRTC_DEFINE_bool(plot_incoming_packet_sizes,
false,
"Plot bar graph showing the size of each incoming packet.");
WEBRTC_DEFINE_bool(plot_outgoing_packet_sizes,
false,
"Plot bar graph showing the size of each outgoing packet.");
WEBRTC_DEFINE_bool(
plot_incoming_packet_count,
false,
"Plot the accumulated number of packets for each incoming stream.");
WEBRTC_DEFINE_bool(
plot_outgoing_packet_count,
false,
"Plot the accumulated number of packets for each outgoing stream.");
WEBRTC_DEFINE_bool(
plot_audio_playout,
false,
"Plot bar graph showing the time between each audio playout.");
WEBRTC_DEFINE_bool(
plot_audio_level,
false,
"Plot line graph showing the audio level of incoming audio.");
WEBRTC_DEFINE_bool(
plot_incoming_sequence_number_delta,
false,
"Plot the sequence number difference between consecutive incoming "
"packets.");
WEBRTC_DEFINE_bool(
plot_incoming_delay,
true,
"Plot the 1-way path delay for incoming packets, normalized so "
"that the first packet has delay 0.");
WEBRTC_DEFINE_bool(
plot_incoming_loss_rate,
true,
"Compute the loss rate for incoming packets using a method that's "
"similar to the one used for RTCP SR and RR fraction lost. Note "
"that the loss rate can be negative if packets are duplicated or "
"reordered.");
WEBRTC_DEFINE_bool(plot_incoming_bitrate,
true,
"Plot the total bitrate used by all incoming streams.");
WEBRTC_DEFINE_bool(plot_outgoing_bitrate,
true,
"Plot the total bitrate used by all outgoing streams.");
WEBRTC_DEFINE_bool(plot_incoming_stream_bitrate,
true,
"Plot the bitrate used by each incoming stream.");
WEBRTC_DEFINE_bool(plot_outgoing_stream_bitrate,
true,
"Plot the bitrate used by each outgoing stream.");
WEBRTC_DEFINE_bool(
plot_simulated_receiveside_bwe,
false,
"Run the receive-side bandwidth estimator with the incoming rtp "
"packets and plot the resulting estimate.");
WEBRTC_DEFINE_bool(
plot_simulated_sendside_bwe,
false,
"Run the send-side bandwidth estimator with the outgoing rtp and "
"incoming rtcp and plot the resulting estimate.");
WEBRTC_DEFINE_bool(
plot_network_delay_feedback,
true,
"Compute network delay based on sent packets and the received "
"transport feedback.");
WEBRTC_DEFINE_bool(
plot_fraction_loss_feedback,
true,
"Plot packet loss in percent for outgoing packets (as perceived by "
"the send-side bandwidth estimator).");
WEBRTC_DEFINE_bool(
plot_pacer_delay,
false,
"Plot the time each sent packet has spent in the pacer (based on "
"the difference between the RTP timestamp and the send "
"timestamp).");
WEBRTC_DEFINE_bool(
plot_timestamps,
false,
"Plot the rtp timestamps of all rtp and rtcp packets over time.");
WEBRTC_DEFINE_bool(
plot_rtcp_details,
false,
"Plot the contents of all report blocks in all sender and receiver "
"reports. This includes fraction lost, cumulative number of lost "
"packets, extended highest sequence number and time since last "
"received SR.");
WEBRTC_DEFINE_bool(plot_audio_encoder_bitrate_bps,
false,
"Plot the audio encoder target bitrate.");
WEBRTC_DEFINE_bool(plot_audio_encoder_frame_length_ms,
false,
"Plot the audio encoder frame length.");
WEBRTC_DEFINE_bool(
plot_audio_encoder_packet_loss,
false,
"Plot the uplink packet loss fraction which is sent to the audio encoder.");
WEBRTC_DEFINE_bool(plot_audio_encoder_fec,
false,
"Plot the audio encoder FEC.");
WEBRTC_DEFINE_bool(plot_audio_encoder_dtx,
false,
"Plot the audio encoder DTX.");
WEBRTC_DEFINE_bool(plot_audio_encoder_num_channels,
false,
"Plot the audio encoder number of channels.");
WEBRTC_DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics.");
WEBRTC_DEFINE_bool(plot_ice_candidate_pair_config,
false,
"Plot the ICE candidate pair config events.");
WEBRTC_DEFINE_bool(plot_ice_connectivity_check,
false,
"Plot the ICE candidate pair connectivity checks.");
WEBRTC_DEFINE_bool(plot_dtls_transport_state,
false,
"Plot DTLS transport state changes.");
WEBRTC_DEFINE_bool(plot_dtls_writable_state,
false,
"Plot DTLS writable state changes.");
WEBRTC_DEFINE_string(
force_fieldtrials,
"",
"Field trials control experimental feature code which can be forced. "
"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
" will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
"trials are separated by \"/\"");
WEBRTC_DEFINE_string(wav_filename,
"",
"Path to wav file used for simulation of jitter buffer");
WEBRTC_DEFINE_bool(help, false, "prints this message");
WEBRTC_DEFINE_bool(
show_detector_state,
false,
"Show the state of the delay based BWE detector on the total "
"bitrate graph");
WEBRTC_DEFINE_bool(show_alr_state,
false,
"Show the state ALR state on the total bitrate graph");
WEBRTC_DEFINE_bool(
parse_unconfigured_header_extensions,
true,
"Attempt to parse unconfigured header extensions using the default "
"WebRTC mapping. This can give very misleading results if the "
"application negotiates a different mapping.");
WEBRTC_DEFINE_bool(print_triage_alerts,
false,
"Print triage alerts, i.e. a list of potential problems.");
WEBRTC_DEFINE_bool(
normalize_time,
true,
"Normalize the log timestamps so that the call starts at time 0.");
WEBRTC_DEFINE_bool(protobuf_output,
false,
"Output charts as protobuf instead of python code.");
void SetAllPlotFlags(bool setting);
int main(int argc, char* argv[]) {
std::string program_name = argv[0];
std::string usage =
"A tool for visualizing WebRTC event logs.\n"
"Example usage:\n" +
program_name + " <logfile> | python\n" + "Run " + program_name +
" --help for a list of command line options\n";
// Parse command line flags without removing them. We're only interested in
// the |plot_profile| flag.
rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, false);
if (strcmp(FLAG_plot_profile, "all") == 0) {
SetAllPlotFlags(true);
} else if (strcmp(FLAG_plot_profile, "none") == 0) {
SetAllPlotFlags(false);
} else if (strcmp(FLAG_plot_profile, "sendside_bwe") == 0) {
SetAllPlotFlags(false);
FLAG_plot_outgoing_packet_sizes = true;
FLAG_plot_outgoing_bitrate = true;
FLAG_plot_outgoing_stream_bitrate = true;
FLAG_plot_simulated_sendside_bwe = true;
FLAG_plot_network_delay_feedback = true;
FLAG_plot_fraction_loss_feedback = true;
} else if (strcmp(FLAG_plot_profile, "receiveside_bwe") == 0) {
SetAllPlotFlags(false);
FLAG_plot_incoming_packet_sizes = true;
FLAG_plot_incoming_delay = true;
FLAG_plot_incoming_loss_rate = true;
FLAG_plot_incoming_bitrate = true;
FLAG_plot_incoming_stream_bitrate = true;
FLAG_plot_simulated_receiveside_bwe = true;
} else if (strcmp(FLAG_plot_profile, "default") == 0) {
// Do nothing.
} else {
rtc::Flag* plot_profile_flag = rtc::FlagList::Lookup("plot_profile");
RTC_CHECK(plot_profile_flag);
plot_profile_flag->Print(false);
}
// Parse the remaining flags. They are applied relative to the chosen profile.
rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
if (argc != 2 || FLAG_help) {
// Print usage information.
std::cout << usage;
if (FLAG_help)
rtc::FlagList::Print(nullptr, false);
return 0;
}
webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
// InitFieldTrialsFromString stores the char*, so the char array must outlive
// the application.
webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
std::string filename = argv[1];
webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions header_extensions =
webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions::kDontParse;
if (FLAG_parse_unconfigured_header_extensions) {
header_extensions = webrtc::ParsedRtcEventLog::
UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig;
}
webrtc::ParsedRtcEventLog parsed_log(header_extensions);
if (!parsed_log.ParseFile(filename)) {
std::cerr << "Could not parse the entire log file." << std::endl;
std::cerr << "Only the parsable events will be analyzed." << std::endl;
}
webrtc::EventLogAnalyzer analyzer(parsed_log, FLAG_normalize_time);
std::unique_ptr<webrtc::PlotCollection> collection;
if (FLAG_protobuf_output) {
collection.reset(new webrtc::ProtobufPlotCollection());
} else {
collection.reset(new webrtc::PythonPlotCollection());
}
if (FLAG_plot_incoming_packet_sizes) {
analyzer.CreatePacketGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_packet_sizes) {
analyzer.CreatePacketGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_incoming_packet_count) {
analyzer.CreateAccumulatedPacketsGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_packet_count) {
analyzer.CreateAccumulatedPacketsGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_audio_playout) {
analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_level) {
analyzer.CreateAudioLevelGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
analyzer.CreateAudioLevelGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_incoming_sequence_number_delta) {
analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
}
if (FLAG_plot_incoming_delay) {
analyzer.CreateIncomingDelayGraph(collection->AppendNewPlot());
}
if (FLAG_plot_incoming_loss_rate) {
analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
}
if (FLAG_plot_incoming_bitrate) {
analyzer.CreateTotalIncomingBitrateGraph(collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_bitrate) {
analyzer.CreateTotalOutgoingBitrateGraph(collection->AppendNewPlot(),
FLAG_show_detector_state,
FLAG_show_alr_state);
}
if (FLAG_plot_incoming_stream_bitrate) {
analyzer.CreateStreamBitrateGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_outgoing_stream_bitrate) {
analyzer.CreateStreamBitrateGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_simulated_receiveside_bwe) {
analyzer.CreateReceiveSideBweSimulationGraph(collection->AppendNewPlot());
}
if (FLAG_plot_simulated_sendside_bwe) {
analyzer.CreateSendSideBweSimulationGraph(collection->AppendNewPlot());
}
if (FLAG_plot_network_delay_feedback) {
analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot());
}
if (FLAG_plot_fraction_loss_feedback) {
analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
}
if (FLAG_plot_timestamps) {
analyzer.CreateTimestampGraph(webrtc::kIncomingPacket,
collection->AppendNewPlot());
analyzer.CreateTimestampGraph(webrtc::kOutgoingPacket,
collection->AppendNewPlot());
}
if (FLAG_plot_rtcp_details) {
auto GetFractionLost = [](const webrtc::rtcp::ReportBlock& block) -> float {
return static_cast<double>(block.fraction_lost()) / 256 * 100;
};
analyzer.CreateSenderAndReceiverReportPlot(
webrtc::kIncomingPacket, GetFractionLost,
"Fraction lost (incoming RTCP)", "Loss rate (percent)",
collection->AppendNewPlot());
analyzer.CreateSenderAndReceiverReportPlot(
webrtc::kOutgoingPacket, GetFractionLost,
"Fraction lost (outgoing RTCP)", "Loss rate (percent)",
collection->AppendNewPlot());
auto GetCumulativeLost =
[](const webrtc::rtcp::ReportBlock& block) -> float {
return block.cumulative_lost_signed();
};
analyzer.CreateSenderAndReceiverReportPlot(
webrtc::kIncomingPacket, GetCumulativeLost,
"Cumulative lost packets (incoming RTCP)", "Packets",
collection->AppendNewPlot());
analyzer.CreateSenderAndReceiverReportPlot(
webrtc::kOutgoingPacket, GetCumulativeLost,
"Cumulative lost packets (outgoing RTCP)", "Packets",
collection->AppendNewPlot());
auto GetHighestSeqNumber =
[](const webrtc::rtcp::ReportBlock& block) -> float {
return block.extended_high_seq_num();
};
analyzer.CreateSenderAndReceiverReportPlot(
webrtc::kIncomingPacket, GetHighestSeqNumber,
"Highest sequence number (incoming RTCP)", "Sequence number",
collection->AppendNewPlot());
analyzer.CreateSenderAndReceiverReportPlot(
webrtc::kOutgoingPacket, GetHighestSeqNumber,
"Highest sequence number (outgoing RTCP)", "Sequence number",
collection->AppendNewPlot());
auto DelaySinceLastSr =
[](const webrtc::rtcp::ReportBlock& block) -> float {
return static_cast<double>(block.delay_since_last_sr()) / 65536;
};
analyzer.CreateSenderAndReceiverReportPlot(
webrtc::kIncomingPacket, DelaySinceLastSr,
"Delay since last received sender report (incoming RTCP)", "Time (s)",
collection->AppendNewPlot());
analyzer.CreateSenderAndReceiverReportPlot(
webrtc::kOutgoingPacket, DelaySinceLastSr,
"Delay since last received sender report (outgoing RTCP)", "Time (s)",
collection->AppendNewPlot());
}
if (FLAG_plot_pacer_delay) {
analyzer.CreatePacerDelayGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_encoder_bitrate_bps) {
analyzer.CreateAudioEncoderTargetBitrateGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_encoder_frame_length_ms) {
analyzer.CreateAudioEncoderFrameLengthGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_encoder_packet_loss) {
analyzer.CreateAudioEncoderPacketLossGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_encoder_fec) {
analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_encoder_dtx) {
analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
}
if (FLAG_plot_audio_encoder_num_channels) {
analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
}
if (FLAG_plot_neteq_stats) {
std::string wav_path;
if (FLAG_wav_filename[0] != '\0') {
wav_path = FLAG_wav_filename;
} else {
wav_path = webrtc::test::ResourcePath(
"audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
}
auto neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
for (webrtc::EventLogAnalyzer::NetEqStatsGetterMap::const_iterator it =
neteq_stats.cbegin();
it != neteq_stats.cend(); ++it) {
analyzer.CreateAudioJitterBufferGraph(it->first, it->second.get(),
collection->AppendNewPlot());
}
analyzer.CreateNetEqNetworkStatsGraph(
neteq_stats,
[](const webrtc::NetEqNetworkStatistics& stats) {
return stats.expand_rate / 16384.f;
},
"Expand rate", collection->AppendNewPlot());
analyzer.CreateNetEqNetworkStatsGraph(
neteq_stats,
[](const webrtc::NetEqNetworkStatistics& stats) {
return stats.speech_expand_rate / 16384.f;
},
"Speech expand rate", collection->AppendNewPlot());
analyzer.CreateNetEqNetworkStatsGraph(
neteq_stats,
[](const webrtc::NetEqNetworkStatistics& stats) {
return stats.accelerate_rate / 16384.f;
},
"Accelerate rate", collection->AppendNewPlot());
analyzer.CreateNetEqNetworkStatsGraph(
neteq_stats,
[](const webrtc::NetEqNetworkStatistics& stats) {
return stats.packet_loss_rate / 16384.f;
},
"Packet loss rate", collection->AppendNewPlot());
analyzer.CreateNetEqLifetimeStatsGraph(
neteq_stats,
[](const webrtc::NetEqLifetimeStatistics& stats) {
return static_cast<float>(stats.concealment_events);
},
"Concealment events", collection->AppendNewPlot());
}
if (FLAG_plot_ice_candidate_pair_config) {
analyzer.CreateIceCandidatePairConfigGraph(collection->AppendNewPlot());
}
if (FLAG_plot_ice_connectivity_check) {
analyzer.CreateIceConnectivityCheckGraph(collection->AppendNewPlot());
}
if (FLAG_plot_dtls_transport_state) {
analyzer.CreateDtlsTransportStateGraph(collection->AppendNewPlot());
}
if (FLAG_plot_dtls_writable_state) {
analyzer.CreateDtlsWritableStateGraph(collection->AppendNewPlot());
}
collection->Draw();
if (FLAG_print_triage_alerts) {
analyzer.CreateTriageNotifications();
analyzer.PrintNotifications(stderr);
}
return 0;
}
void SetAllPlotFlags(bool setting) {
FLAG_plot_incoming_packet_sizes = setting;
FLAG_plot_outgoing_packet_sizes = setting;
FLAG_plot_incoming_packet_count = setting;
FLAG_plot_outgoing_packet_count = setting;
FLAG_plot_audio_playout = setting;
FLAG_plot_audio_level = setting;
FLAG_plot_incoming_sequence_number_delta = setting;
FLAG_plot_incoming_delay = setting;
FLAG_plot_incoming_loss_rate = setting;
FLAG_plot_incoming_bitrate = setting;
FLAG_plot_outgoing_bitrate = setting;
FLAG_plot_incoming_stream_bitrate = setting;
FLAG_plot_outgoing_stream_bitrate = setting;
FLAG_plot_simulated_receiveside_bwe = setting;
FLAG_plot_simulated_sendside_bwe = setting;
FLAG_plot_network_delay_feedback = setting;
FLAG_plot_fraction_loss_feedback = setting;
FLAG_plot_timestamps = setting;
FLAG_plot_rtcp_details = setting;
FLAG_plot_audio_encoder_bitrate_bps = setting;
FLAG_plot_audio_encoder_frame_length_ms = setting;
FLAG_plot_audio_encoder_packet_loss = setting;
FLAG_plot_audio_encoder_fec = setting;
FLAG_plot_audio_encoder_dtx = setting;
FLAG_plot_audio_encoder_num_channels = setting;
FLAG_plot_neteq_stats = setting;
FLAG_plot_ice_candidate_pair_config = setting;
FLAG_plot_ice_connectivity_check = setting;
}
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