Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.

webrtc/rtc_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.
  */
 
@@ skipping 93 matching lines @@
 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_string(wav_filename,
+              "",
+              "Path to wav file used for simulation of jitter buffer");
 DEFINE_bool(help, false, "prints this message");
 
 DEFINE_bool(show_detector_state,
             false,
             "Show the state of the delay based BWE detector on the total "
             "bitrate graph");
 
 void SetAllPlotFlags(bool setting);
 
 
@@ skipping 124 matching lines @@
   if (FLAG_plot_audio_encoder_fec) {
     analyzer.CreateAudioEncoderEnableFecGraph(collection->AppendNewPlot());
   }
   if (FLAG_plot_audio_encoder_dtx) {
     analyzer.CreateAudioEncoderEnableDtxGraph(collection->AppendNewPlot());
   }
   if (FLAG_plot_audio_encoder_num_channels) {
     analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
   }
   if (FLAG_plot_audio_jitter_buffer) {
-    analyzer.CreateAudioJitterBufferGraph(
-        webrtc::test::ResourcePath(
-            "audio_processing/conversational_speech/EN_script2_F_sp2_B1",
-            "wav"),
-        48000, collection->AppendNewPlot());
+    std::string wav_path;
+    if (FLAG_wav_filename[0] != '\0') {
+      wav_path = FLAG_wav_filename;
+    } else {
+      wav_path = webrtc::test::ResourcePath(
+          "audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
+    }
+    analyzer.CreateAudioJitterBufferGraph(wav_path, 48000,
+                                          collection->AppendNewPlot());
   }
 
   collection->Draw();
 
   return 0;
 }
 
 
 void SetAllPlotFlags(bool setting) {
   FLAG_plot_incoming_packet_sizes = setting;
@@ skipping 15 matching lines @@
   FLAG_plot_fraction_loss_feedback = setting;
   FLAG_plot_timestamps = setting;
   FLAG_plot_audio_encoder_bitrate_bps = setting;
   FLAG_plot_audio_encoder_frame_length_ms = setting;
   FLAG_plot_audio_encoder_packet_loss = setting;
   FLAG_plot_audio_encoder_fec = setting;
   FLAG_plot_audio_encoder_dtx = setting;
   FLAG_plot_audio_encoder_num_channels = setting;
   FLAG_plot_audio_jitter_buffer = setting;
 }