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Issue 2370313002: Reland of Unify rtcp packet setters (Closed)
Patch Set: Fix breaking mistype Created 4 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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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145 rtc::CritScope cs(&lock_); 145 rtc::CritScope cs(&lock_);
146 auto it = packet_arrival_times_.lower_bound(window_start_seq_); 146 auto it = packet_arrival_times_.lower_bound(window_start_seq_);
147 if (it == packet_arrival_times_.end()) { 147 if (it == packet_arrival_times_.end()) {
148 // Feedback for all packets already sent. 148 // Feedback for all packets already sent.
149 return false; 149 return false;
150 } 150 }
151 151
152 // TODO(sprang): Measure receive times in microseconds and remove the 152 // TODO(sprang): Measure receive times in microseconds and remove the
153 // conversions below. 153 // conversions below.
154 const int64_t first_sequence = it->first; 154 const int64_t first_sequence = it->first;
155 feedback_packet->WithMediaSourceSsrc(media_ssrc_); 155 feedback_packet->SetMediaSsrc(media_ssrc_);
156 // Base sequence is the expected next (window_start_seq_). This is known, but 156 // Base sequence is the expected next (window_start_seq_). This is known, but
157 // we might not have actually received it, so the base time shall be the time 157 // we might not have actually received it, so the base time shall be the time
158 // of the first received packet in the feedback. 158 // of the first received packet in the feedback.
159 feedback_packet->WithBase(static_cast<uint16_t>(window_start_seq_ & 0xFFFF), 159 feedback_packet->SetBase(static_cast<uint16_t>(window_start_seq_ & 0xFFFF),
160 it->second * 1000); 160 it->second * 1000);
161 feedback_packet->WithFeedbackSequenceNumber(feedback_sequence_++); 161 feedback_packet->SetFeedbackSequenceNumber(feedback_sequence_++);
162 for (; it != packet_arrival_times_.end(); ++it) { 162 for (; it != packet_arrival_times_.end(); ++it) {
163 if (!feedback_packet->WithReceivedPacket( 163 if (!feedback_packet->AddReceivedPacket(
164 static_cast<uint16_t>(it->first & 0xFFFF), it->second * 1000)) { 164 static_cast<uint16_t>(it->first & 0xFFFF), it->second * 1000)) {
165 // If we can't even add the first seq to the feedback packet, we won't be 165 // If we can't even add the first seq to the feedback packet, we won't be
166 // able to build it at all. 166 // able to build it at all.
167 RTC_CHECK_NE(first_sequence, it->first); 167 RTC_CHECK_NE(first_sequence, it->first);
168 168
169 // Could not add timestamp, feedback packet might be full. Return and 169 // Could not add timestamp, feedback packet might be full. Return and
170 // try again with a fresh packet. 170 // try again with a fresh packet.
171 break; 171 break;
172 } 172 }
173 173
174 // Note: Don't erase items from packet_arrival_times_ after sending, in case 174 // Note: Don't erase items from packet_arrival_times_ after sending, in case
175 // they need to be re-sent after a reordering. Removal will be handled 175 // they need to be re-sent after a reordering. Removal will be handled
176 // by OnPacketArrival once packets are too old. 176 // by OnPacketArrival once packets are too old.
177 window_start_seq_ = it->first + 1; 177 window_start_seq_ = it->first + 1;
178 } 178 }
179 179
180 return true; 180 return true;
181 } 181 }
182 182
183 } // namespace webrtc 183 } // namespace webrtc
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