Rename main file for visualization tool.

Review-Url: https://codereview.webrtc.org/2226273002 Cr-Commit-Position: refs/heads/master@{#13712}
2016-08-10 07:12:38 -07:00
parent c4ac700781
commit d49a37b1d9
3 changed files with 2 additions and 2 deletions
--- a/webrtc/tools/event_log_visualizer/main.cc
+++ b/webrtc/tools/event_log_visualizer/main.cc
@ -0,0 +1,162 @@
+/*
+ *  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;
+}