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Unified Diff: webrtc/test/call_test.cc

Issue 1984983002: Remove use of RtpHeaderExtension and clean up (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed nit Created 4 years, 7 months ago
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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) {

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