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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 |
11 #ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ | 11 #ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ |
12 #define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ | 12 #define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ |
13 | 13 |
14 #include "testing/gmock/include/gmock/gmock.h" | 14 #include "testing/gmock/include/gmock/gmock.h" |
15 | 15 |
16 #include "webrtc/modules/interface/module.h" | 16 #include "webrtc/modules/interface/module.h" |
17 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" | 17 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" | 18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
| 19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
19 | 20 |
20 namespace webrtc { | 21 namespace webrtc { |
21 | 22 |
22 class MockRtpData : public RtpData { | 23 class MockRtpData : public RtpData { |
23 public: | 24 public: |
24 MOCK_METHOD3(OnReceivedPayloadData, | 25 MOCK_METHOD3(OnReceivedPayloadData, |
25 int32_t(const uint8_t* payloadData, | 26 int32_t(const uint8_t* payloadData, |
26 const size_t payloadSize, | 27 const size_t payloadSize, |
27 const WebRtcRTPHeader* rtpHeader)); | 28 const WebRtcRTPHeader* rtpHeader)); |
28 | 29 |
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198 int()); | 199 int()); |
199 MOCK_METHOD1(SetSelectiveRetransmissions, | 200 MOCK_METHOD1(SetSelectiveRetransmissions, |
200 int(uint8_t settings)); | 201 int(uint8_t settings)); |
201 MOCK_METHOD2(SendNACK, | 202 MOCK_METHOD2(SendNACK, |
202 int32_t(const uint16_t* nackList, const uint16_t size)); | 203 int32_t(const uint16_t* nackList, const uint16_t size)); |
203 MOCK_METHOD2(SetStorePacketsStatus, | 204 MOCK_METHOD2(SetStorePacketsStatus, |
204 void(const bool enable, const uint16_t numberToStore)); | 205 void(const bool enable, const uint16_t numberToStore)); |
205 MOCK_CONST_METHOD0(StorePackets, bool()); | 206 MOCK_CONST_METHOD0(StorePackets, bool()); |
206 MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*)); | 207 MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*)); |
207 MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*()); | 208 MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*()); |
| 209 MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet)); |
208 MOCK_METHOD1(RegisterAudioCallback, | 210 MOCK_METHOD1(RegisterAudioCallback, |
209 int32_t(RtpAudioFeedback* messagesCallback)); | 211 int32_t(RtpAudioFeedback* messagesCallback)); |
210 MOCK_METHOD1(SetAudioPacketSize, | 212 MOCK_METHOD1(SetAudioPacketSize, |
211 int32_t(const uint16_t packetSizeSamples)); | 213 int32_t(const uint16_t packetSizeSamples)); |
212 MOCK_METHOD3(SendTelephoneEventOutband, | 214 MOCK_METHOD3(SendTelephoneEventOutband, |
213 int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level)); | 215 int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level)); |
214 MOCK_METHOD1(SetSendREDPayloadType, | 216 MOCK_METHOD1(SetSendREDPayloadType, |
215 int32_t(const int8_t payloadType)); | 217 int32_t(const int8_t payloadType)); |
216 MOCK_CONST_METHOD1(SendREDPayloadType, | 218 MOCK_CONST_METHOD1(SendREDPayloadType, |
217 int32_t(int8_t& payloadType)); | 219 int32_t(int8_t& payloadType)); |
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252 void(StreamDataCountersCallback*)); | 254 void(StreamDataCountersCallback*)); |
253 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, | 255 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, |
254 StreamDataCountersCallback*(void)); | 256 StreamDataCountersCallback*(void)); |
255 // Members. | 257 // Members. |
256 unsigned int remote_ssrc_; | 258 unsigned int remote_ssrc_; |
257 }; | 259 }; |
258 | 260 |
259 } // namespace webrtc | 261 } // namespace webrtc |
260 | 262 |
261 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ | 263 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ |
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