Index: webrtc/test/call_test.cc |
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc |
index 768c007c3cbed2f5a3f19f21830ff93bbe2893d9..8da747a830bffe775427f840d5f035e1b1350e1e 100644 |
--- a/webrtc/test/call_test.cc |
+++ b/webrtc/test/call_test.cc |
@@ -184,12 +184,12 @@ void CallTest::CreateSendConfig(size_t num_video_streams, |
video_send_config_.encoder_settings.payload_type = |
kFakeVideoSendPayloadType; |
video_send_config_.rtp.extensions.push_back( |
- RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); |
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); |
video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams); |
for (size_t i = 0; i < num_video_streams; ++i) |
video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); |
video_send_config_.rtp.extensions.push_back(RtpExtension( |
- RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId)); |
+ RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); |
} |
if (num_audio_streams > 0) { |