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Side by Side Diff: webrtc/modules/pacing/packet_router.cc

Issue 1962303002: Added cluster id to PacedSender::Callback::TimeToSendPacket. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addded probing cluster unittest. Created 4 years, 7 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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36 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) { 36 void PacketRouter::RemoveRtpModule(RtpRtcp* rtp_module) {
37 rtc::CritScope cs(&modules_crit_); 37 rtc::CritScope cs(&modules_crit_);
38 RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) != 38 RTC_DCHECK(std::find(rtp_modules_.begin(), rtp_modules_.end(), rtp_module) !=
39 rtp_modules_.end()); 39 rtp_modules_.end());
40 rtp_modules_.remove(rtp_module); 40 rtp_modules_.remove(rtp_module);
41 } 41 }
42 42
43 bool PacketRouter::TimeToSendPacket(uint32_t ssrc, 43 bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
44 uint16_t sequence_number, 44 uint16_t sequence_number,
45 int64_t capture_timestamp, 45 int64_t capture_timestamp,
46 bool retransmission) { 46 bool retransmission,
47 int probe_cluster_id) {
47 RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread()); 48 RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
48 rtc::CritScope cs(&modules_crit_); 49 rtc::CritScope cs(&modules_crit_);
49 for (auto* rtp_module : rtp_modules_) { 50 for (auto* rtp_module : rtp_modules_) {
50 if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) { 51 if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
51 return rtp_module->TimeToSendPacket(ssrc, sequence_number, 52 return rtp_module->TimeToSendPacket(ssrc, sequence_number,
52 capture_timestamp, retransmission); 53 capture_timestamp, retransmission);
53 } 54 }
54 } 55 }
55 return true; 56 return true;
56 } 57 }
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97 rtc::CritScope cs(&modules_crit_); 98 rtc::CritScope cs(&modules_crit_);
98 for (auto* rtp_module : rtp_modules_) { 99 for (auto* rtp_module : rtp_modules_) {
99 packet->WithPacketSenderSsrc(rtp_module->SSRC()); 100 packet->WithPacketSenderSsrc(rtp_module->SSRC());
100 if (rtp_module->SendFeedbackPacket(*packet)) 101 if (rtp_module->SendFeedbackPacket(*packet))
101 return true; 102 return true;
102 } 103 }
103 return false; 104 return false;
104 } 105 }
105 106
106 } // namespace webrtc 107 } // namespace webrtc
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