Add NetEq delay plotting to event_log_visualizer

webrtc/tools/event_log_visualizer/main.cc

 /*
  *  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 "webrtc/base/flags.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/test/field_trial.h"
+#include "webrtc/test/testsupport/fileutils.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.");
@@ skipping 44 matching lines @@
             "Plot the audio encoder frame length.");
 DEFINE_bool(
     audio_encoder_uplink_packet_loss_fraction,
     false,
     "Plot the uplink packet loss fraction which is send to the audio encoder.");
 DEFINE_bool(audio_encoder_fec, false, "Plot the audio encoder FEC.");
 DEFINE_bool(audio_encoder_dtx, false, "Plot the audio encoder DTX.");
 DEFINE_bool(audio_encoder_num_channels,
             false,
             "Plot the audio encoder number of channels.");
+DEFINE_bool(plot_audio_jitter_buffer,
+            false,
+            "Plot the audio jitter buffer delay profile.");
 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 \"/\"");
 DEFINE_bool(help, false, "prints this message");
 
 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";
   rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
   if (FLAG_help) {
     rtc::FlagList::Print(nullptr, false);
     return 0;
   }
 
   if (argc != 2) {
     // Print usage information.
     std::cout << usage;
     return 0;
   }
 
+  webrtc::test::SetExecutablePath(argv[0]);

[hlundin-webrtc, 2017/06/08 09:54:55]

This is needed to make webrtc::test::ResourcePath work.

   webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
 
   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."
@@ skipping 106 matching lines @@
   }
 
   if (FLAG_plot_all || FLAG_audio_encoder_dtx) {
     analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
   }
 
   if (FLAG_plot_all || FLAG_audio_encoder_num_channels) {
     analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
   }
 
+  if (FLAG_plot_all || FLAG_plot_audio_jitter_buffer) {
+    analyzer.CreateAudioJitterBufferGraph(
+        webrtc::test::ResourcePath(
+            "audio_processing/conversational_speech/EN_script2_F_sp2_B1",
+            "wav"),
+        48000, collection->AppendNewPlot());
+  }
+
   collection->Draw();
 
   return 0;
 }