Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback

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 68 matching lines @@
       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;
       // If set, default parameters will be used instead of |streams|.
       bool infer_streams;
     } ss;
     int num_thumbnails;
+    std::string rtc_event_log_name;
+    std::string rtp_dump_name;

[sprang_webrtc, 2017/07/12 13:27:06]

Could you perhaps create a new substruct here called Logging or something like
that? Then we can move these two, the |logs| boolean and encoded_frame_base_path
there.

Also, shouldn't num_thumbnails be in the Video section?

[ilnik, 2017/07/12 13:58:53]

Done. 
num_thumbnails is more of a call property, not a video. Moved it there.

   };
 
   VideoQualityTest();
   void RunWithAnalyzer(const Params& params);
   void RunWithRenderers(const Params& params);
 
   static void FillScalabilitySettings(
       Params* params,
       const std::vector<std::string>& stream_descriptors,
       int num_streams,
@@ skipping 54 matching lines @@
   int send_logs_;
 
   VideoSendStream::DegradationPreference degradation_preference_ =
       VideoSendStream::DegradationPreference::kMaintainFramerate;
   Params params_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_