Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc |
index a68f06e6848dcd3c3ad5433ad74aceeeb2d788ab..ca844f7df5feb25cef579ddef45cb9e1b21a457b 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc |
@@ -15,6 +15,7 @@ |
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h" |
#include "webrtc/test/gmock.h" |
#include "webrtc/test/gtest.h" |
@@ -46,7 +47,7 @@ class RtpReceiverTest : public ::testing::Test { |
: fake_clock_(123456), |
rtp_receiver_( |
RtpReceiver::CreateAudioReceiver(&fake_clock_, |
- nullptr, |
+ &mock_rtp_data_, |
nullptr, |
&rtp_payload_registry_)) { |
CodecInst voice_codec = {}; |
@@ -73,6 +74,7 @@ class RtpReceiverTest : public ::testing::Test { |
} |
SimulatedClock fake_clock_; |
+ ::testing::NiceMock<MockRtpData> mock_rtp_data_; |
danilchap
2017/05/31 14:57:33
to make this line look nicer
nisse-webrtc
2017/06/01 06:59:56
Done.
|
RTPPayloadRegistry rtp_payload_registry_; |
std::unique_ptr<RtpReceiver> rtp_receiver_; |
}; |