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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 1877253002: Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: git cl format dtmf_queue.cc Created 4 years, 8 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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105 last_process_time_(configuration.clock->TimeInMilliseconds()), 105 last_process_time_(configuration.clock->TimeInMilliseconds()),
106 last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()), 106 last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
107 last_rtt_process_time_(configuration.clock->TimeInMilliseconds()), 107 last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
108 packet_overhead_(28), // IPV4 UDP. 108 packet_overhead_(28), // IPV4 UDP.
109 nack_last_time_sent_full_(0), 109 nack_last_time_sent_full_(0),
110 nack_last_time_sent_full_prev_(0), 110 nack_last_time_sent_full_prev_(0),
111 nack_last_seq_number_sent_(0), 111 nack_last_seq_number_sent_(0),
112 key_frame_req_method_(kKeyFrameReqPliRtcp), 112 key_frame_req_method_(kKeyFrameReqPliRtcp),
113 remote_bitrate_(configuration.remote_bitrate_estimator), 113 remote_bitrate_(configuration.remote_bitrate_estimator),
114 rtt_stats_(configuration.rtt_stats), 114 rtt_stats_(configuration.rtt_stats),
115 critical_section_rtt_(CriticalSectionWrapper::CreateCriticalSection()),
116 rtt_ms_(0) { 115 rtt_ms_(0) {
117 // Make sure that RTCP objects are aware of our SSRC. 116 // Make sure that RTCP objects are aware of our SSRC.
118 uint32_t SSRC = rtp_sender_.SSRC(); 117 uint32_t SSRC = rtp_sender_.SSRC();
119 rtcp_sender_.SetSSRC(SSRC); 118 rtcp_sender_.SetSSRC(SSRC);
120 SetRtcpReceiverSsrcs(SSRC); 119 SetRtcpReceiverSsrcs(SSRC);
121 SetMaxTransferUnit(IP_PACKET_SIZE); 120 SetMaxTransferUnit(IP_PACKET_SIZE);
122 } 121 }
123 122
124 // Returns the number of milliseconds until the module want a worker thread 123 // Returns the number of milliseconds until the module want a worker thread
125 // to call Process. 124 // to call Process.
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975 974
976 void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) { 975 void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
977 std::set<uint32_t> ssrcs; 976 std::set<uint32_t> ssrcs;
978 ssrcs.insert(main_ssrc); 977 ssrcs.insert(main_ssrc);
979 if (rtp_sender_.RtxStatus() != kRtxOff) 978 if (rtp_sender_.RtxStatus() != kRtxOff)
980 ssrcs.insert(rtp_sender_.RtxSsrc()); 979 ssrcs.insert(rtp_sender_.RtxSsrc());
981 rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs); 980 rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs);
982 } 981 }
983 982
984 void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) { 983 void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
985 CriticalSectionScoped cs(critical_section_rtt_.get()); 984 rtc::CritScope cs(&critical_section_rtt_);
986 rtt_ms_ = rtt_ms; 985 rtt_ms_ = rtt_ms;
987 } 986 }
988 987
989 int64_t ModuleRtpRtcpImpl::rtt_ms() const { 988 int64_t ModuleRtpRtcpImpl::rtt_ms() const {
990 CriticalSectionScoped cs(critical_section_rtt_.get()); 989 rtc::CritScope cs(&critical_section_rtt_);
991 return rtt_ms_; 990 return rtt_ms_;
992 } 991 }
993 992
994 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 993 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
995 StreamDataCountersCallback* callback) { 994 StreamDataCountersCallback* callback) {
996 rtp_sender_.RegisterRtpStatisticsCallback(callback); 995 rtp_sender_.RegisterRtpStatisticsCallback(callback);
997 } 996 }
998 997
999 StreamDataCountersCallback* 998 StreamDataCountersCallback*
1000 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 999 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
1001 return rtp_sender_.GetRtpStatisticsCallback(); 1000 return rtp_sender_.GetRtpStatisticsCallback();
1002 } 1001 }
1003 } // namespace webrtc 1002 } // namespace webrtc
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