Remove use of RtpHeaderExtension and clean up

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index a94034c6496079a40a24968586c502948f38ed5b..730cd39f9b47ccab4604b888aac59944aad7a35b 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -99,11 +99,11 @@ struct ConfigHelper {
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+        RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+        RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
-        RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+        RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
@@ -171,7 +171,7 @@ TEST(AudioSendStreamTest, ConfigToString) {
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = kSsrc;
   config.rtp.extensions.push_back(
-      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+      RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
   config.rtp.c_name = kCName;
   config.voe_channel_id = kChannelId;
   config.cng_payload_type = 42;