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 16 matching lines @@
   // Parameters are grouped into smaller structs to make it easier to set
   // the desired elements and skip unused, using aggregate initialization.
   // Unfortunately, C++11 (as opposed to C11) doesn't support unnamed structs,
   // which makes the implementation of VideoQualityTest a bit uglier.
   struct Params {
     Params();
     ~Params();
     struct CallConfig {
       bool send_side_bwe;
       Call::Config::BitrateConfig call_bitrate_config;
+      int num_thumbnails;
     } call;
     struct Video {
       bool enabled;
       size_t width;
       size_t height;
       int32_t fps;
       int min_bitrate_bps;
       int target_bitrate_bps;
       int max_bitrate_bps;
       bool suspend_below_min_bitrate;
       std::string codec;
       int num_temporal_layers;
       int selected_tl;
       int min_transmit_bps;
       bool ulpfec;
       bool flexfec;
-      std::string encoded_frame_base_path;
       std::string clip_name;  // "Generator" to generate frames instead.
       size_t capture_device_index;
     } video;
     struct Audio {
       bool enabled;
       bool sync_video;
       bool dtx;
     } audio;
     struct Screenshare {
       bool enabled;
       int32_t slide_change_interval;
       int32_t scroll_duration;
       std::vector<std::string> slides;
     } screenshare;
     struct Analyzer {
       std::string test_label;
       double avg_psnr_threshold;  // (*)
       double avg_ssim_threshold;  // (*)
       int test_durations_secs;
       std::string graph_data_output_filename;
       std::string graph_title;
     } analyzer;
     FakeNetworkPipe::Config pipe;
-    bool logs;
     struct SS {                          // 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;
       // If set, default parameters will be used instead of |streams|.
       bool infer_streams;
     } ss;
-    int num_thumbnails;
+    struct Logging {
+      bool logs;
+      std::string rtc_event_log_name;
+      std::string rtp_dump_name;
+      std::string encoded_frame_base_path;
+    } logging;
   };
 
   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_