Remove use of RtpHeaderExtension and clean up

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.
  */
 
@@ skipping 402 matching lines @@
 
   // RTP/RTCP initialization.
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     module_process_thread_->RegisterModule(rtp_rtcp);
     congestion_controller_->packet_router()->AddRtpModule(rtp_rtcp);
   }
 
   video_sender_->RegisterProtectionCallback(this);
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
-    const std::string& extension = config_.rtp.extensions[i].name;
+    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));
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
       RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
                           StringToRtpExtensionType(extension), id));
     }
   }
@@ skipping 367 matching lines @@
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
 }  // namespace internal
 }  // namespace webrtc