Only one implementation of MockRtpPacketSink once

webrtc/video/rtp_video_stream_receiver_unittest.cc

 /*
  *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #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"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/frame_object.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/modules/video_coding/packet.h"
 #include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
 #include "webrtc/modules/video_coding/timing.h"
@@ skipping 64 matching lines @@
     }
     DoOnCompleteFrame(frame.get());
   }
   void AppendExpectedBitstream(const uint8_t data[], size_t size_in_bytes) {
     // TODO(Johan): Let rtc::ByteBuffer handle uint8_t* instead of char*.
     buffer_.WriteBytes(reinterpret_cast<const char*>(data), size_in_bytes);
   }
   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(
     uint32_t ssrc = kSsrc,
     uint16_t sequence_number = kSequenceNumber) {
   auto packet = rtc::MakeUnique<RtpPacketReceived>();
   packet->SetSsrc(ssrc);
   packet->SetSequenceNumber(sequence_number);
   return packet;
 }
@@ skipping 339 matching lines @@
   rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
   EXPECT_DEATH(rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink),
                "");
 
   // Test tear-down.
   rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink);
 }
 #endif
 
 }  // namespace webrtc