Add reporting of googContentType via GetStats on send side

webrtc/call/video_send_stream.h

 /*
  *  Copyright (c) 2013 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 64 matching lines @@
     bool suspended = false;
     bool bw_limited_resolution = false;
     bool cpu_limited_resolution = false;
     bool bw_limited_framerate = false;
     bool cpu_limited_framerate = false;
     // Total number of times resolution as been requested to be changed due to
     // CPU/quality adaptation.
     int number_of_cpu_adapt_changes = 0;
     int number_of_quality_adapt_changes = 0;
     std::map<uint32_t, StreamStats> substreams;
+    webrtc::VideoContentType content_type =
+        webrtc::VideoContentType::UNSPECIFIED;
   };
 
   struct Config {
    public:
     Config() = delete;
     Config(Config&&);
     explicit Config(Transport* send_transport);
 
     Config& operator=(Config&&);
     Config& operator=(const Config&) = delete;
@@ skipping 180 matching lines @@
     EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
   }
 
  protected:
   virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_VIDEO_SEND_STREAM_H_