Chromium Code Reviews| 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_; |
| }; |