| Index: webrtc/tools/event_log_visualizer/generate_timeseries.cc
|
| diff --git a/webrtc/tools/event_log_visualizer/generate_timeseries.cc b/webrtc/tools/event_log_visualizer/generate_timeseries.cc
|
| deleted file mode 100644
|
| index 27dc59c04259d64a85f4f03e6aa18c242e207d6d..0000000000000000000000000000000000000000
|
| --- a/webrtc/tools/event_log_visualizer/generate_timeseries.cc
|
| +++ /dev/null
|
| @@ -1,162 +0,0 @@
|
| -/*
|
| - * 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 <iostream>
|
| -
|
| -#include "gflags/gflags.h"
|
| -#include "webrtc/call/rtc_event_log_parser.h"
|
| -#include "webrtc/tools/event_log_visualizer/analyzer.h"
|
| -#include "webrtc/tools/event_log_visualizer/plot_base.h"
|
| -#include "webrtc/tools/event_log_visualizer/plot_python.h"
|
| -
|
| -DEFINE_bool(incoming, true, "Plot statistics for incoming packets.");
|
| -DEFINE_bool(outgoing, true, "Plot statistics for outgoing packets.");
|
| -DEFINE_bool(plot_all, true, "Plot all different data types.");
|
| -DEFINE_bool(plot_packets,
|
| - false,
|
| - "Plot bar graph showing the size of each packet.");
|
| -DEFINE_bool(plot_audio_playout,
|
| - false,
|
| - "Plot bar graph showing the time between each audio playout.");
|
| -DEFINE_bool(
|
| - plot_sequence_number,
|
| - false,
|
| - "Plot the difference in sequence number between consecutive packets.");
|
| -DEFINE_bool(
|
| - plot_delay_change,
|
| - false,
|
| - "Plot the difference in 1-way path delay between consecutive packets.");
|
| -DEFINE_bool(plot_accumulated_delay_change,
|
| - false,
|
| - "Plot the accumulated 1-way path delay change, or the path delay "
|
| - "change compared to the first packet.");
|
| -DEFINE_bool(plot_total_bitrate,
|
| - false,
|
| - "Plot the total bitrate used by all streams.");
|
| -DEFINE_bool(plot_stream_bitrate,
|
| - false,
|
| - "Plot the bitrate used by each stream.");
|
| -DEFINE_bool(plot_bwe,
|
| - false,
|
| - "Run the bandwidth estimator with the logged rtp and rtcp and plot "
|
| - "the output.");
|
| -DEFINE_bool(plot_network_delay_feedback,
|
| - false,
|
| - "Compute network delay based on sent packets and the received "
|
| - "transport feedback.");
|
| -DEFINE_bool(plot_fraction_loss,
|
| - false,
|
| - "Plot packet loss in percent for outgoing packets (as perceived by "
|
| - "the send-side bandwidth estimator).");
|
| -
|
| -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";
|
| - google::SetUsageMessage(usage);
|
| - google::ParseCommandLineFlags(&argc, &argv, true);
|
| -
|
| - if (argc != 2) {
|
| - // Print usage information.
|
| - std::cout << google::ProgramUsage();
|
| - return 0;
|
| - }
|
| -
|
| - std::string filename = argv[1];
|
| -
|
| - webrtc::ParsedRtcEventLog parsed_log;
|
| -
|
| - if (!parsed_log.ParseFile(filename)) {
|
| - std::cerr << "Could not parse the entire log file." << std::endl;
|
| - std::cerr << "Proceeding to analyze the first "
|
| - << parsed_log.GetNumberOfEvents() << " events in the file."
|
| - << std::endl;
|
| - }
|
| -
|
| - webrtc::plotting::EventLogAnalyzer analyzer(parsed_log);
|
| - std::unique_ptr<webrtc::plotting::PlotCollection> collection(
|
| - new webrtc::plotting::PythonPlotCollection());
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_packets) {
|
| - if (FLAGS_incoming) {
|
| - analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
|
| - collection->AppendNewPlot());
|
| - }
|
| - if (FLAGS_outgoing) {
|
| - analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
|
| - collection->AppendNewPlot());
|
| - }
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_audio_playout) {
|
| - analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_sequence_number) {
|
| - if (FLAGS_incoming) {
|
| - analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
|
| - }
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_delay_change) {
|
| - if (FLAGS_incoming) {
|
| - analyzer.CreateDelayChangeGraph(collection->AppendNewPlot());
|
| - }
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_accumulated_delay_change) {
|
| - if (FLAGS_incoming) {
|
| - analyzer.CreateAccumulatedDelayChangeGraph(collection->AppendNewPlot());
|
| - }
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_fraction_loss) {
|
| - analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_total_bitrate) {
|
| - if (FLAGS_incoming) {
|
| - analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
|
| - collection->AppendNewPlot());
|
| - }
|
| - if (FLAGS_outgoing) {
|
| - analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
|
| - collection->AppendNewPlot());
|
| - }
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_stream_bitrate) {
|
| - if (FLAGS_incoming) {
|
| - analyzer.CreateStreamBitrateGraph(
|
| - webrtc::PacketDirection::kIncomingPacket,
|
| - collection->AppendNewPlot());
|
| - }
|
| - if (FLAGS_outgoing) {
|
| - analyzer.CreateStreamBitrateGraph(
|
| - webrtc::PacketDirection::kOutgoingPacket,
|
| - collection->AppendNewPlot());
|
| - }
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_bwe) {
|
| - analyzer.CreateBweGraph(collection->AppendNewPlot());
|
| - }
|
| -
|
| - if (FLAGS_plot_all || FLAGS_plot_network_delay_feedback) {
|
| - analyzer.CreateNetworkDelayFeebackGraph(collection->AppendNewPlot());
|
| - }
|
| -
|
| - collection->Draw();
|
| -
|
| - return 0;
|
| -}
|
|
|