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

webrtc/video/video_send_stream.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.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 840 matching lines @@
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
     transport->packet_router()->AddSendRtpModule(rtp_rtcp);
 
   for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) {
     const std::string& extension = config_->rtp.extensions[i].uri;
     int id = config_->rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(id, 1);
     RTC_DCHECK_LE(id, 14);
     RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
+    // TODO(ilnik): Remove if statement once experiment is not needed.
     if (StringToRtpExtensionType(extension) == kRtpExtensionVideoContentType &&
         !field_trial::IsEnabled("WebRTC-VideoContentTypeExtension")) {
       continue;
     }
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
       RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
                           StringToRtpExtensionType(extension), id));
     }
   }
 
@@ skipping 499 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc