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0295a967c0
Bug: webrtc:8450 Change-Id: Idea093854a644f3018a565168425583dc4783ce9 Reviewed-on: https://webrtc-review.googlesource.com/15480 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20442}
339 lines
13 KiB
C++
339 lines
13 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <iostream>
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#include "logging/rtc_event_log/rtc_event_log_parser.h"
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#include "rtc_base/flags.h"
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#include "rtc_tools/event_log_visualizer/analyzer.h"
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#include "rtc_tools/event_log_visualizer/plot_base.h"
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#include "rtc_tools/event_log_visualizer/plot_python.h"
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#include "test/field_trial.h"
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#include "test/testsupport/fileutils.h"
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DEFINE_string(plot_profile,
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"default",
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"A profile that selects a certain subset of the plots. Currently "
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"defined profiles are \"all\", \"none\", \"sendside_bwe\","
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"\"receiveside_bwe\" and \"default\"");
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DEFINE_bool(plot_incoming_packet_sizes,
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false,
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"Plot bar graph showing the size of each incoming packet.");
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DEFINE_bool(plot_outgoing_packet_sizes,
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false,
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"Plot bar graph showing the size of each outgoing packet.");
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DEFINE_bool(plot_incoming_packet_count,
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false,
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"Plot the accumulated number of packets for each incoming stream.");
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DEFINE_bool(plot_outgoing_packet_count,
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false,
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"Plot the accumulated number of packets for each outgoing stream.");
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DEFINE_bool(plot_audio_playout,
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false,
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"Plot bar graph showing the time between each audio playout.");
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DEFINE_bool(plot_audio_level,
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false,
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"Plot line graph showing the audio level of incoming audio.");
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DEFINE_bool(plot_incoming_sequence_number_delta,
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false,
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"Plot the sequence number difference between consecutive incoming "
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"packets.");
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DEFINE_bool(
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plot_incoming_delay_delta,
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false,
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"Plot the difference in 1-way path delay between consecutive packets.");
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DEFINE_bool(plot_incoming_delay,
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true,
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"Plot the 1-way path delay for incoming packets, normalized so "
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"that the first packet has delay 0.");
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DEFINE_bool(plot_incoming_loss_rate,
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true,
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"Compute the loss rate for incoming packets using a method that's "
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"similar to the one used for RTCP SR and RR fraction lost. Note "
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"that the loss rate can be negative if packets are duplicated or "
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"reordered.");
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DEFINE_bool(plot_incoming_bitrate,
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true,
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"Plot the total bitrate used by all incoming streams.");
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DEFINE_bool(plot_outgoing_bitrate,
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true,
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"Plot the total bitrate used by all outgoing streams.");
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DEFINE_bool(plot_incoming_stream_bitrate,
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true,
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"Plot the bitrate used by each incoming stream.");
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DEFINE_bool(plot_outgoing_stream_bitrate,
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true,
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"Plot the bitrate used by each outgoing stream.");
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DEFINE_bool(plot_simulated_receiveside_bwe,
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false,
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"Run the receive-side bandwidth estimator with the incoming rtp "
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"packets and plot the resulting estimate.");
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DEFINE_bool(plot_simulated_sendside_bwe,
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false,
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"Run the send-side bandwidth estimator with the outgoing rtp and "
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"incoming rtcp and plot the resulting estimate.");
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DEFINE_bool(plot_network_delay_feedback,
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true,
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"Compute network delay based on sent packets and the received "
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"transport feedback.");
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DEFINE_bool(plot_fraction_loss_feedback,
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true,
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"Plot packet loss in percent for outgoing packets (as perceived by "
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"the send-side bandwidth estimator).");
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DEFINE_bool(plot_pacer_delay,
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false,
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"Plot the time each sent packet has spent in the pacer (based on "
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"the difference between the RTP timestamp and the send "
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"timestamp).");
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DEFINE_bool(plot_timestamps,
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false,
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"Plot the rtp timestamps of all rtp and rtcp packets over time.");
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DEFINE_bool(plot_audio_encoder_bitrate_bps,
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false,
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"Plot the audio encoder target bitrate.");
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DEFINE_bool(plot_audio_encoder_frame_length_ms,
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false,
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"Plot the audio encoder frame length.");
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DEFINE_bool(
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plot_audio_encoder_packet_loss,
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false,
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"Plot the uplink packet loss fraction which is sent to the audio encoder.");
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DEFINE_bool(plot_audio_encoder_fec, false, "Plot the audio encoder FEC.");
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DEFINE_bool(plot_audio_encoder_dtx, false, "Plot the audio encoder DTX.");
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DEFINE_bool(plot_audio_encoder_num_channels,
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false,
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"Plot the audio encoder number of channels.");
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DEFINE_bool(plot_audio_jitter_buffer,
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false,
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"Plot the audio jitter buffer delay profile.");
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DEFINE_string(
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force_fieldtrials,
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"",
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"Field trials control experimental feature code which can be forced. "
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"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
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" will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
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"trials are separated by \"/\"");
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DEFINE_string(wav_filename,
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"",
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"Path to wav file used for simulation of jitter buffer");
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DEFINE_bool(help, false, "prints this message");
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DEFINE_bool(show_detector_state,
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false,
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"Show the state of the delay based BWE detector on the total "
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"bitrate graph");
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void SetAllPlotFlags(bool setting);
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int main(int argc, char* argv[]) {
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std::string program_name = argv[0];
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std::string usage =
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"A tool for visualizing WebRTC event logs.\n"
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"Example usage:\n" +
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program_name + " <logfile> | python\n" + "Run " + program_name +
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" --help for a list of command line options\n";
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// Parse command line flags without removing them. We're only interested in
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// the |plot_profile| flag.
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rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, false);
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if (strcmp(FLAG_plot_profile, "all") == 0) {
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SetAllPlotFlags(true);
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} else if (strcmp(FLAG_plot_profile, "none") == 0) {
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SetAllPlotFlags(false);
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} else if (strcmp(FLAG_plot_profile, "sendside_bwe") == 0) {
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SetAllPlotFlags(false);
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FLAG_plot_outgoing_packet_sizes = true;
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FLAG_plot_outgoing_bitrate = true;
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FLAG_plot_outgoing_stream_bitrate = true;
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FLAG_plot_simulated_sendside_bwe = true;
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FLAG_plot_network_delay_feedback = true;
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FLAG_plot_fraction_loss_feedback = true;
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} else if (strcmp(FLAG_plot_profile, "receiveside_bwe") == 0) {
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SetAllPlotFlags(false);
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FLAG_plot_incoming_packet_sizes = true;
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FLAG_plot_incoming_delay_delta = true;
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FLAG_plot_incoming_delay = true;
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FLAG_plot_incoming_loss_rate = true;
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FLAG_plot_incoming_bitrate = true;
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FLAG_plot_incoming_stream_bitrate = true;
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FLAG_plot_simulated_receiveside_bwe = true;
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} else if (strcmp(FLAG_plot_profile, "default") == 0) {
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// Do nothing.
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} else {
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rtc::Flag* plot_profile_flag = rtc::FlagList::Lookup("plot_profile");
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RTC_CHECK(plot_profile_flag);
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plot_profile_flag->Print(false);
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}
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// Parse the remaining flags. They are applied relative to the chosen profile.
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rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
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if (argc != 2 || FLAG_help) {
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// Print usage information.
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std::cout << usage;
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if (FLAG_help)
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rtc::FlagList::Print(nullptr, false);
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return 0;
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}
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webrtc::test::SetExecutablePath(argv[0]);
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webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
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std::string filename = argv[1];
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webrtc::ParsedRtcEventLog parsed_log;
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if (!parsed_log.ParseFile(filename)) {
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std::cerr << "Could not parse the entire log file." << std::endl;
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std::cerr << "Proceeding to analyze the first "
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<< parsed_log.GetNumberOfEvents() << " events in the file."
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<< std::endl;
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}
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webrtc::plotting::EventLogAnalyzer analyzer(parsed_log);
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std::unique_ptr<webrtc::plotting::PlotCollection> collection(
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new webrtc::plotting::PythonPlotCollection());
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if (FLAG_plot_incoming_packet_sizes) {
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analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
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collection->AppendNewPlot());
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}
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if (FLAG_plot_outgoing_packet_sizes) {
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analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
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collection->AppendNewPlot());
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}
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if (FLAG_plot_incoming_packet_count) {
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analyzer.CreateAccumulatedPacketsGraph(
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webrtc::PacketDirection::kIncomingPacket, collection->AppendNewPlot());
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}
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if (FLAG_plot_outgoing_packet_count) {
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analyzer.CreateAccumulatedPacketsGraph(
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webrtc::PacketDirection::kOutgoingPacket, collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_playout) {
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analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_level) {
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analyzer.CreateAudioLevelGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_incoming_sequence_number_delta) {
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analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_incoming_delay_delta) {
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analyzer.CreateIncomingDelayDeltaGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_incoming_delay) {
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analyzer.CreateIncomingDelayGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_incoming_loss_rate) {
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analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_incoming_bitrate) {
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analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
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collection->AppendNewPlot(),
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FLAG_show_detector_state);
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}
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if (FLAG_plot_outgoing_bitrate) {
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analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
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collection->AppendNewPlot(),
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FLAG_show_detector_state);
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}
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if (FLAG_plot_incoming_stream_bitrate) {
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analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
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collection->AppendNewPlot());
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}
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if (FLAG_plot_outgoing_stream_bitrate) {
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analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
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collection->AppendNewPlot());
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}
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if (FLAG_plot_simulated_receiveside_bwe) {
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analyzer.CreateReceiveSideBweSimulationGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_simulated_sendside_bwe) {
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analyzer.CreateSendSideBweSimulationGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_network_delay_feedback) {
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analyzer.CreateNetworkDelayFeedbackGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_fraction_loss_feedback) {
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analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_timestamps) {
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analyzer.CreateTimestampGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_pacer_delay) {
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analyzer.CreatePacerDelayGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_encoder_bitrate_bps) {
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analyzer.CreateAudioEncoderTargetBitrateGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_encoder_frame_length_ms) {
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analyzer.CreateAudioEncoderFrameLengthGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_encoder_packet_loss) {
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analyzer.CreateAudioEncoderPacketLossGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_encoder_fec) {
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analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_encoder_dtx) {
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analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_encoder_num_channels) {
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analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
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}
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if (FLAG_plot_audio_jitter_buffer) {
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std::string wav_path;
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if (FLAG_wav_filename[0] != '\0') {
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wav_path = FLAG_wav_filename;
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} else {
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wav_path = webrtc::test::ResourcePath(
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"audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
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}
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analyzer.CreateAudioJitterBufferGraph(wav_path, 48000,
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collection->AppendNewPlot());
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}
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collection->Draw();
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return 0;
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}
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void SetAllPlotFlags(bool setting) {
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FLAG_plot_incoming_packet_sizes = setting;
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FLAG_plot_outgoing_packet_sizes = setting;
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FLAG_plot_incoming_packet_count = setting;
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FLAG_plot_outgoing_packet_count = setting;
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FLAG_plot_audio_playout = setting;
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FLAG_plot_audio_level = setting;
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FLAG_plot_incoming_sequence_number_delta = setting;
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FLAG_plot_incoming_delay_delta = setting;
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FLAG_plot_incoming_delay = setting;
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FLAG_plot_incoming_loss_rate = setting;
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FLAG_plot_incoming_bitrate = setting;
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FLAG_plot_outgoing_bitrate = setting;
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FLAG_plot_incoming_stream_bitrate = setting;
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FLAG_plot_outgoing_stream_bitrate = setting;
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FLAG_plot_simulated_receiveside_bwe = setting;
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FLAG_plot_simulated_sendside_bwe = setting;
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FLAG_plot_network_delay_feedback = setting;
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FLAG_plot_fraction_loss_feedback = setting;
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FLAG_plot_timestamps = setting;
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FLAG_plot_audio_encoder_bitrate_bps = setting;
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FLAG_plot_audio_encoder_frame_length_ms = setting;
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FLAG_plot_audio_encoder_packet_loss = setting;
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FLAG_plot_audio_encoder_fec = setting;
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FLAG_plot_audio_encoder_dtx = setting;
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FLAG_plot_audio_encoder_num_channels = setting;
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FLAG_plot_audio_jitter_buffer = setting;
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}
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