Add reporting of googContentType via GetStats on send side

webrtc/media/engine/webrtcvideoengine.cc

 /*
  *  Copyright (c) 2014 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 1958 matching lines @@
   info.framerate_input = stats.input_frame_rate;
   info.framerate_sent = stats.encode_frame_rate;
   info.avg_encode_ms = stats.avg_encode_time_ms;
   info.encode_usage_percent = stats.encode_usage_percent;
   info.frames_encoded = stats.frames_encoded;
   info.qp_sum = stats.qp_sum;
 
   info.nominal_bitrate = stats.media_bitrate_bps;
   info.preferred_bitrate = stats.preferred_media_bitrate_bps;
 
+  info.content_type = stats.content_type;
+
   info.send_frame_width = 0;
   info.send_frame_height = 0;
   for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
            stats.substreams.begin();
        it != stats.substreams.end(); ++it) {
     // TODO(pbos): Wire up additional stats, such as padding bytes.
     webrtc::VideoSendStream::StreamStats stream_stats = it->second;
     info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
                        stream_stats.rtp_stats.transmitted.header_bytes +
                        stream_stats.rtp_stats.transmitted.padding_bytes;
@@ skipping 616 matching lines @@
     stream.temporal_layer_thresholds_bps.resize(GetDefaultVp9TemporalLayers() -
                                                 1);
   }
 
   std::vector<webrtc::VideoStream> streams;
   streams.push_back(stream);
   return streams;
 }
 
 }  // namespace cricket