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eladalon
2017/07/26 08:21:39
Please note that I've suffixed "mock" rather than
danilchap
2017/07/26 08:38:13
in webrtc mocks usually go into own folder and use
eladalon
2017/07/26 09:01:05
Done.
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| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 #ifndef WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_MOCK_H_ | |
| 11 #define WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_MOCK_H_ | |
| 12 | |
| 13 #include "webrtc/call/rtp_packet_sink_interface.h" | |
| 14 | |
| 15 #include "webrtc/test/gmock.h" | |
| 16 | |
| 17 namespace webrtc { | |
| 18 | |
| 19 class RtpPacketReceived; | |
| 20 | |
| 21 class MockRtpPacketSink : public RtpPacketSinkInterface { | |
| 22 public: | |
| 23 MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&)); | |
| 24 }; | |
| 25 | |
| 26 } // namespace webrtc | |
| 27 | |
| 28 #endif // WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_MOCK_H_ | |
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