Remove use of RtpHeaderExtension and clean up

webrtc/audio/audio_receive_stream.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 20 matching lines @@
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 namespace {
 
 bool UseSendSideBwe(const webrtc::AudioReceiveStream::Config& config) {
   if (!config.rtp.transport_cc) {
     return false;
   }
   for (const auto& extension : config.rtp.extensions) {
-    if (extension.name == RtpExtension::kTransportSequenceNumber) {
+    if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       return true;
     }
   }
   return false;
 }
 }  // namespace
 
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
@@ skipping 38 matching lines @@
   RTC_DCHECK(congestion_controller);
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
 
   channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
 
   for (const auto& extension : config.rtp.extensions) {
-    if (extension.name == RtpExtension::kAudioLevel) {
+    if (extension.uri == RtpExtension::kAudioLevelUri) {
       channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, extension.id);
       RTC_DCHECK(registered);
-    } else if (extension.name == RtpExtension::kAbsSendTime) {
+    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
       channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, extension.id);
       RTC_DCHECK(registered);
-    } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
   // Configure bandwidth estimation.
   channel_proxy_->RegisterReceiverCongestionControlObjects(
@@ skipping 122 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc