Index: webrtc/video/rtp_video_stream_receiver_unittest.cc |
diff --git a/webrtc/video/rtp_video_stream_receiver_unittest.cc b/webrtc/video/rtp_video_stream_receiver_unittest.cc |
index d309397dd471d32fcc40adb7fbc391905feb3652..37697d71b3673bb0214a390265d96fe6c414d771 100644 |
--- a/webrtc/video/rtp_video_stream_receiver_unittest.cc |
+++ b/webrtc/video/rtp_video_stream_receiver_unittest.cc |
@@ -11,6 +11,7 @@ |
#include "webrtc/test/gtest.h" |
#include "webrtc/test/gmock.h" |
+#include "webrtc/call/test/mock_rtp_packet_sink_interface.h" |
#include "webrtc/common_video/h264/h264_common.h" |
#include "webrtc/media/base/mediaconstants.h" |
#include "webrtc/modules/pacing/packet_router.h" |
@@ -95,11 +96,6 @@ class MockOnCompleteFrameCallback |
rtc::ByteBufferWriter buffer_; |
}; |
-class MockRtpPacketSink : public RtpPacketSinkInterface { |
- public: |
- MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&)); |
-}; |
- |
constexpr uint32_t kSsrc = 111; |
constexpr uint16_t kSequenceNumber = 222; |
std::unique_ptr<RtpPacketReceived> CreateRtpPacketReceived( |