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Side by Side Diff: webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

Issue 1803923003: Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 9 months ago
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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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212 int(uint8_t settings)); 212 int(uint8_t settings));
213 MOCK_METHOD2(SendNACK, 213 MOCK_METHOD2(SendNACK,
214 int32_t(const uint16_t* nackList, const uint16_t size)); 214 int32_t(const uint16_t* nackList, const uint16_t size));
215 MOCK_METHOD1(SendNack, void(const std::vector<uint16_t>& sequence_numbers)); 215 MOCK_METHOD1(SendNack, void(const std::vector<uint16_t>& sequence_numbers));
216 MOCK_METHOD2(SetStorePacketsStatus, 216 MOCK_METHOD2(SetStorePacketsStatus,
217 void(const bool enable, const uint16_t numberToStore)); 217 void(const bool enable, const uint16_t numberToStore));
218 MOCK_CONST_METHOD0(StorePackets, bool()); 218 MOCK_CONST_METHOD0(StorePackets, bool());
219 MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*)); 219 MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
220 MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*()); 220 MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
221 MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet)); 221 MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
222 MOCK_METHOD1(RegisterAudioCallback,
223 int32_t(RtpAudioFeedback* messagesCallback));
224 MOCK_METHOD1(SetAudioPacketSize, 222 MOCK_METHOD1(SetAudioPacketSize,
225 int32_t(const uint16_t packetSizeSamples)); 223 int32_t(const uint16_t packetSizeSamples));
226 MOCK_METHOD3(SendTelephoneEventOutband, 224 MOCK_METHOD3(SendTelephoneEventOutband,
227 int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level)); 225 int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level));
228 MOCK_METHOD1(SetSendREDPayloadType, 226 MOCK_METHOD1(SetSendREDPayloadType,
229 int32_t(const int8_t payloadType)); 227 int32_t(const int8_t payloadType));
230 MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t* payloadType)); 228 MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t* payloadType));
231 MOCK_METHOD2(SetRTPAudioLevelIndicationStatus, 229 MOCK_METHOD2(SetRTPAudioLevelIndicationStatus,
232 int32_t(const bool enable, const uint8_t ID)); 230 int32_t(const bool enable, const uint8_t ID));
233 MOCK_METHOD1(SetAudioLevel, 231 MOCK_METHOD1(SetAudioLevel,
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261 void(StreamDataCountersCallback*)); 259 void(StreamDataCountersCallback*));
262 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, 260 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
263 StreamDataCountersCallback*(void)); 261 StreamDataCountersCallback*(void));
264 // Members. 262 // Members.
265 unsigned int remote_ssrc_; 263 unsigned int remote_ssrc_;
266 }; 264 };
267 265
268 } // namespace webrtc 266 } // namespace webrtc
269 267
270 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ 268 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
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