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 52 matching lines @@
     FakeNetworkPipe::Config pipe;
     bool logs;
     struct {                             // Spatial scalability.
       std::vector<VideoStream> streams;  // If empty, one stream is assumed.
       size_t selected_stream;
       int num_spatial_layers;
       int selected_sl;
       // If empty, bitrates are generated in VP9Impl automatically.
       std::vector<SpatialLayer> spatial_layers;
     } ss;
+    bool audio;
+    bool audio_video_sync;
   };
   // (*) Set to -1.1 if generating graph data for simulcast or SVC and the
   // selected stream/layer doesn't have the same resolution as the largest
   // stream/layer (to ignore the PSNR and SSIM calculation errors).
 
   VideoQualityTest();
   void RunWithAnalyzer(const Params& params);
   void RunWithVideoRenderer(const Params& params);
 
   static void FillScalabilitySettings(
@@ skipping 29 matching lines @@
   std::unique_ptr<VideoEncoder> encoder_;
   VideoCodecUnion codec_settings_;
   Clock* const clock_;
 
   Params params_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_