Remove use of RtpHeaderExtension and clean up

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index c0a709e3c6296765df1cf6c0ab8ec6146010a636..e9659c67e8c48ecd3bf09369d27562322b1e5be3 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -79,11 +79,11 @@ AudioSendStream::AudioSendStream(
   channel_proxy_->RegisterExternalTransport(config.send_transport);
 
   for (const auto& extension : config.rtp.extensions) {
-    if (extension.name == RtpExtension::kAbsSendTime) {
+    if (extension.uri == RtpExtension::kAbsSendTimeUri) {
       channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
-    } else if (extension.name == RtpExtension::kAudioLevel) {
+    } else if (extension.uri == RtpExtension::kAudioLevelUri) {
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
-    } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";