Piggybacking simulcast id and ALR experiment id into video content type extension.

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::UNSPECIFIED;
+  rtp_header->type.Video.content_type = VideoContentType::Unspecified();
   rtp_header->type.Video.video_timing.flags = TimingFrameFlags::kInvalid;
 
   // 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 =
@@ skipping 24 matching lines @@
     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