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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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151 bool SendFeedbackPacket(const rtcp::TransportFeedback& packet); | 151 bool SendFeedbackPacket(const rtcp::TransportFeedback& packet); |
152 | 152 |
153 private: | 153 private: |
154 class RtcpContext; | 154 class RtcpContext; |
155 | 155 |
156 // Determine which RTCP messages should be sent and setup flags. | 156 // Determine which RTCP messages should be sent and setup flags. |
157 void PrepareReport(const std::set<RTCPPacketType>& packetTypes, | 157 void PrepareReport(const std::set<RTCPPacketType>& packetTypes, |
158 const FeedbackState& feedback_state) | 158 const FeedbackState& feedback_state) |
159 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); | 159 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
160 | 160 |
161 int32_t AddReportBlock(const RTCPReportBlock& report_block) | 161 bool AddReportBlock(const FeedbackState& feedback_state, |
| 162 uint32_t ssrc, |
| 163 StreamStatistician* statistician) |
162 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); | 164 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
163 | 165 |
164 bool PrepareReportBlock(const FeedbackState& feedback_state, | |
165 uint32_t ssrc, | |
166 StreamStatistician* statistician, | |
167 RTCPReportBlock* report_block); | |
168 | |
169 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context) | 166 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context) |
170 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); | 167 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
171 rtc::scoped_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context) | 168 rtc::scoped_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context) |
172 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); | 169 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
173 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context) | 170 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context) |
174 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); | 171 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
175 rtc::scoped_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context) | 172 rtc::scoped_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context) |
176 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); | 173 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
177 rtc::scoped_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context) | 174 rtc::scoped_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context) |
178 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); | 175 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
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296 | 293 |
297 std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_); | 294 std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_); |
298 | 295 |
299 typedef rtc::scoped_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)( | 296 typedef rtc::scoped_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)( |
300 const RtcpContext&); | 297 const RtcpContext&); |
301 std::map<RTCPPacketType, BuilderFunc> builders_; | 298 std::map<RTCPPacketType, BuilderFunc> builders_; |
302 }; | 299 }; |
303 } // namespace webrtc | 300 } // namespace webrtc |
304 | 301 |
305 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_ | 302 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_ |
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