Add content type information to encoded images and corresponding rtp extension header

webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc

 /*
  *  Copyright (c) 2012 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 72 matching lines @@
   }
 
   rtp_header->type.Video.is_first_packet_in_frame = is_first_packet;
   RtpDepacketizer::ParsedPayload parsed_payload;
   if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
     return -1;
 
   rtp_header->frameType = parsed_payload.frame_type;
   rtp_header->type = parsed_payload.type;
   rtp_header->type.Video.rotation = kVideoRotation_0;
+  rtp_header->type.Video.content_type = VideoContentType::kDefault;
 
   // Retrieve the video rotation information.
   if (rtp_header->header.extension.hasVideoRotation) {
     rtp_header->type.Video.rotation =
         rtp_header->header.extension.videoRotation;
   }
 
+  if (rtp_header->header.extension.hasVideoContentType) {
+    rtp_header->type.Video.content_type =
+        rtp_header->header.extension.videoContentType;
+  }
+
   rtp_header->type.Video.playout_delay =
       rtp_header->header.extension.playout_delay;
 
   return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
                                                parsed_payload.payload_length,
                                                rtp_header) == 0
              ? 0
              : -1;
 }
 
 RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
     uint16_t last_payload_length) const {
   return kRtpDead;
 }
 
 int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
     RtpFeedback* callback,
     int8_t payload_type,
     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     const PayloadUnion& specific_payload) const {
   // TODO(pbos): Remove as soon as audio can handle a changing payload type
   // without this callback.
   return 0;
 }
 
 }  // namespace webrtc