Remove use of RtpHeaderExtension and clean up

webrtc/call/rtc_event_log.cc

 /*
  *  Copyright (c) 2015 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 250 matching lines @@
   for (const auto& kv : config.rtp.rtx) {
     rtclog::RtxMap* rtx = receiver_config->add_rtx_map();
     rtx->set_payload_type(kv.first);
     rtx->mutable_config()->set_rtx_ssrc(kv.second.ssrc);
     rtx->mutable_config()->set_rtx_payload_type(kv.second.payload_type);
   }
 
   for (const auto& e : config.rtp.extensions) {
     rtclog::RtpHeaderExtension* extension =
         receiver_config->add_header_extensions();
-    extension->set_name(e.name);
+    extension->set_name(e.uri);
     extension->set_id(e.id);
   }
 
   for (const auto& d : config.decoders) {
     rtclog::DecoderConfig* decoder = receiver_config->add_decoders();
     decoder->set_name(d.payload_name);
     decoder->set_payload_type(d.payload_type);
   }
   if (!event_queue_.Insert(&event)) {
     LOG(LS_WARNING) << "Config queue full. Not logging config event.";
   }
 }
 
 void RtcEventLogImpl::LogVideoSendStreamConfig(
     const VideoSendStream::Config& config) {
   std::unique_ptr<rtclog::Event> event(new rtclog::Event());
   event->set_timestamp_us(clock_->TimeInMicroseconds());
   event->set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
 
   rtclog::VideoSendConfig* sender_config = event->mutable_video_sender_config();
 
   for (const auto& ssrc : config.rtp.ssrcs) {
     sender_config->add_ssrcs(ssrc);
   }
 
   for (const auto& e : config.rtp.extensions) {
     rtclog::RtpHeaderExtension* extension =
         sender_config->add_header_extensions();
-    extension->set_name(e.name);
+    extension->set_name(e.uri);
     extension->set_id(e.id);
   }
 
   for (const auto& rtx_ssrc : config.rtp.rtx.ssrcs) {
     sender_config->add_rtx_ssrcs(rtx_ssrc);
   }
   sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type);
 
   rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
   encoder->set_name(config.encoder_settings.payload_name);
@@ skipping 142 matching lines @@
 // RtcEventLog member functions.
 std::unique_ptr<RtcEventLog> RtcEventLog::Create(const Clock* clock) {
 #ifdef ENABLE_RTC_EVENT_LOG
   return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl(clock));
 #else
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 #endif  // ENABLE_RTC_EVENT_LOG
 }
 
 }  // namespace webrtc