Implement googContentType GetStats metric reported on receive side.

webrtc/video/receive_statistics_proxy.cc

 /*
  *  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 481 matching lines @@
   // Get current frame rates here, as only updating them on new frames prevents
   // us from ever correctly displaying frame rate of 0.
   int64_t now_ms = clock_->TimeInMilliseconds();
   UpdateFramerate(now_ms);
   stats_.render_frame_rate = renders_fps_estimator_.Rate(now_ms).value_or(0);
   stats_.decode_frame_rate = decode_fps_estimator_.Rate(now_ms).value_or(0);
   stats_.total_bitrate_bps =
       static_cast<int>(total_byte_tracker_.ComputeRate() * 8);
   stats_.interframe_delay_max_ms =
       interframe_delay_max_moving_.Max(now_ms).value_or(-1);
+  if (videocontenttypehelpers::IsScreenshare(last_content_type_)) {
+    stats_.content_type = "screen";
+  } else {
+    stats_.content_type = "realtime";

[tommi, 2017/08/30 15:03:28]

fyi - the code generation for this statement generates code for two calls to the
assignment operator. An alternative could be:

stats_.content_type = videocontenttypehelpers::IsScreenshare(last_content_type_)
: "screen" : "realtime";

[ilnik, 2017/08/30 15:30:24]

Done.

+  }
   return stats_;
 }
 
 rtc::Optional<TimingFrameInfo>
 ReceiveStatisticsProxy::GetAndResetTimingFrameInfo() {
   rtc::CritScope lock(&crit_);
   rtc::Optional<TimingFrameInfo> info = timing_frame_info_;
   // Reset reported value to empty, so it will be always
   // overwritten in |OnTimingFrameInfoUpdated|, thus allowing to store new
   // value instead of reported one.
@@ skipping 295 matching lines @@
   flow_duration_ms += other.flow_duration_ms;
   total_media_bytes += other.total_media_bytes;
   received_height.Add(other.received_height);
   received_width.Add(other.received_width);
   qp_counter.Add(other.qp_counter);
   frame_counts.key_frames += other.frame_counts.key_frames;
   frame_counts.delta_frames += other.frame_counts.delta_frames;
 }
 
 }  // namespace webrtc