Adding audio to video_quality_test.

webrtc/video/video_quality_test.h

 /*
  *  Copyright (c) 2015 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.
  */
 #ifndef WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_
@@ skipping 87 matching lines @@
 
   // Helper static methods.
   static VideoStream DefaultVideoStream(const Params& params);
   static std::vector<int> ParseCSV(const std::string& str);
 
   // Helper methods for setting up the call.
   void CreateCapturer(VideoCaptureInput* input);
   void SetupCommon(Transport* send_transport, Transport* recv_transport);
   void SetupScreenshare();
 
+  void CreateVoiceEngine();
+  void DestroyVoiceEngine();
+
   // We need a more general capturer than the FrameGeneratorCapturer.
   std::unique_ptr<test::VideoCapturer> capturer_;
   std::unique_ptr<test::TraceToStderr> trace_to_stderr_;
   std::unique_ptr<test::FrameGenerator> frame_generator_;
   std::unique_ptr<VideoEncoder> encoder_;
   VideoCodecUnion codec_settings_;
   Clock* const clock_;
 
+  struct VoiceEngineState {

[minyue-webrtc, 2016/07/12 10:09:36]

I think this, and CreateVoiceEngine/DestroyVoiceEngine belongs to
RunWithVideoRenderer only, therefore I moved them as static in .cc

+    VoiceEngineState()
+        : voice_engine(nullptr),
+          base(nullptr),
+          codec(nullptr),
+          send_channel_id(-1),
+          receive_channel_id(-1) {}
+
+    VoiceEngine* voice_engine;
+    VoEBase* base;
+    VoECodec* codec;
+    int send_channel_id;
+    int receive_channel_id;
+  };
+  VoiceEngineState voe_;
+  std::unique_ptr<test::FakeAudioDevice> fake_audio_device_;
+
   Params params_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_