Remove Absolute Send Time from list of supported header extensions for audio streams.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 6f0ff6334bd229989c7e7ba181abb3c2dc952b84..01a2db1844967a1e7b351561c9f50ad0ae5e0aa4 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -2050,12 +2050,14 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
   TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri);
 }
 
-// Test support for absolute send time header extension.
-TEST_F(WebRtcVoiceEngineTestFake, SendAbsoluteSendTimeHeaderExtensions) {
-  TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri);
-}
-TEST_F(WebRtcVoiceEngineTestFake, RecvAbsoluteSendTimeHeaderExtensions) {
-  TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAbsSendTimeUri);
+// Test support for transport sequence number header extension.
+TEST_F(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) {
+  TestSetSendRtpHeaderExtensions(
+      webrtc::RtpExtension::kTransportSequenceNumberUri);
+}
+TEST_F(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) {
+  TestSetRecvRtpHeaderExtensions(
+      webrtc::RtpExtension::kTransportSequenceNumberUri);
 }
 
 // Test that we can create a channel and start sending on it.