Initial version of the local visualization tool for WebrtcEventLogs.

Plot graphs to python code. Generate packet, playout, sequence number and total bitrate graphs.

Add bitrate and latency plots. Generate multiple figures, and control which ones are used through command line flags.

Committed: https://crrev.com/a478786ef1513790194792010f766324a469db4d
Review-Url: https://codereview.webrtc.org/1995523002
Cr-Original-Commit-Position: refs/heads/master@{#13443}
Cr-Commit-Position: refs/heads/master@{#13463}
diff --git a/webrtc/tools/event_log_visualizer/generate_timeseries.cc b/webrtc/tools/event_log_visualizer/generate_timeseries.cc
new file mode 100644
index 0000000..d213947
--- /dev/null
+++ b/webrtc/tools/event_log_visualizer/generate_timeseries.cc
@@ -0,0 +1,138 @@
+/*
+ *  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.");
+
+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->append_new_plot());
+    }
+    if (FLAGS_outgoing) {
+      analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
+                                 collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_audio_playout) {
+    analyzer.CreatePlayoutGraph(collection->append_new_plot());
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_sequence_number) {
+    if (FLAGS_incoming) {
+      analyzer.CreateSequenceNumberGraph(collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_delay_change) {
+    if (FLAGS_incoming) {
+      analyzer.CreateDelayChangeGraph(collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_accumulated_delay_change) {
+    if (FLAGS_incoming) {
+      analyzer.CreateAccumulatedDelayChangeGraph(collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_total_bitrate) {
+    if (FLAGS_incoming) {
+      analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
+                                       collection->append_new_plot());
+    }
+    if (FLAGS_outgoing) {
+      analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
+                                       collection->append_new_plot());
+    }
+  }
+
+  if (FLAGS_plot_all || FLAGS_plot_stream_bitrate) {
+    if (FLAGS_incoming) {
+      analyzer.CreateStreamBitrateGraph(
+          webrtc::PacketDirection::kIncomingPacket,
+          collection->append_new_plot());
+    }
+    if (FLAGS_outgoing) {
+      analyzer.CreateStreamBitrateGraph(
+          webrtc::PacketDirection::kOutgoingPacket,
+          collection->append_new_plot());
+    }
+  }
+
+  collection->draw();
+
+  return 0;
+}