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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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49 MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc)); | 49 MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc)); |
50 MOCK_METHOD4(IncomingAudioNTP, | 50 MOCK_METHOD4(IncomingAudioNTP, |
51 int32_t(uint32_t audio_received_ntp_secs, | 51 int32_t(uint32_t audio_received_ntp_secs, |
52 uint32_t audio_received_ntp_frac, | 52 uint32_t audio_received_ntp_frac, |
53 uint32_t audio_rtcp_arrival_time_secs, | 53 uint32_t audio_rtcp_arrival_time_secs, |
54 uint32_t audio_rtcp_arrival_time_frac)); | 54 uint32_t audio_rtcp_arrival_time_frac)); |
55 MOCK_METHOD0(InitSender, int32_t()); | 55 MOCK_METHOD0(InitSender, int32_t()); |
56 MOCK_METHOD1(RegisterSendTransport, int32_t(Transport* outgoing_transport)); | 56 MOCK_METHOD1(RegisterSendTransport, int32_t(Transport* outgoing_transport)); |
57 MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size)); | 57 MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size)); |
58 MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet)); | 58 MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet)); |
59 MOCK_CONST_METHOD0(MaxPayloadSize, size_t()); | 59 MOCK_CONST_METHOD1(MaxPayloadSize, |
danilchap
2017/05/09 14:45:22
remove
erikvarga1
2017/05/09 15:19:41
Done.
| |
60 size_t(rtc::ArrayView<const RtpExtensionSize>)); | |
60 MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t()); | 61 MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t()); |
61 MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst& voice_codec)); | 62 MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst& voice_codec)); |
62 MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec& video_codec)); | 63 MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec& video_codec)); |
63 MOCK_METHOD2(RegisterVideoSendPayload, | 64 MOCK_METHOD2(RegisterVideoSendPayload, |
64 void(int payload_type, const char* payload_name)); | 65 void(int payload_type, const char* payload_name)); |
65 MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type)); | 66 MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type)); |
66 MOCK_METHOD2(RegisterSendRtpHeaderExtension, | 67 MOCK_METHOD2(RegisterSendRtpHeaderExtension, |
67 int32_t(RTPExtensionType type, uint8_t id)); | 68 int32_t(RTPExtensionType type, uint8_t id)); |
68 MOCK_METHOD1(DeregisterSendRtpHeaderExtension, | 69 MOCK_METHOD1(DeregisterSendRtpHeaderExtension, |
69 int32_t(RTPExtensionType type)); | 70 int32_t(RTPExtensionType type)); |
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209 private: | 210 private: |
210 // Mocking this method is currently not required and having a default | 211 // Mocking this method is currently not required and having a default |
211 // implementation like MOCK_METHOD0(TimeUntilNextProcess, int64_t()) | 212 // implementation like MOCK_METHOD0(TimeUntilNextProcess, int64_t()) |
212 // can be dangerous since it can cause a tight loop on a process thread. | 213 // can be dangerous since it can cause a tight loop on a process thread. |
213 virtual int64_t TimeUntilNextProcess() { return 0xffffffff; } | 214 virtual int64_t TimeUntilNextProcess() { return 0xffffffff; } |
214 }; | 215 }; |
215 | 216 |
216 } // namespace webrtc | 217 } // namespace webrtc |
217 | 218 |
218 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ | 219 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ |
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