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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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