Chromium Code Reviews| Index: webrtc/call/rtp_packet_sink_interface_mock.h |
| diff --git a/webrtc/call/rtp_packet_sink_interface_mock.h b/webrtc/call/rtp_packet_sink_interface_mock.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..0d30d0af9ab47e64de6f217869507904fc81f3e9 |
| --- /dev/null |
| +++ b/webrtc/call/rtp_packet_sink_interface_mock.h |
| @@ -0,0 +1,28 @@ |
| +/* |
|
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.
|
| + * Copyright (c) 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. |
| + */ |
| +#ifndef WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_MOCK_H_ |
| +#define WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_MOCK_H_ |
| + |
| +#include "webrtc/call/rtp_packet_sink_interface.h" |
| + |
| +#include "webrtc/test/gmock.h" |
| + |
| +namespace webrtc { |
| + |
| +class RtpPacketReceived; |
| + |
| +class MockRtpPacketSink : public RtpPacketSinkInterface { |
| + public: |
| + MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&)); |
| +}; |
| + |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_MOCK_H_ |