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. |