1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #include "webrtc/video_engine/vie_remb.h" 12 13 #include <assert.h> 14 15 #include <algorithm> 16 17 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 18 #include "webrtc/modules/utility/interface/process_thread.h" 19 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 20 #include "webrtc/system_wrappers/interface/tick_util.h" 21 #include "webrtc/system_wrappers/interface/trace.h" 22 23 namespace webrtc { 24 25 const int kRembSendIntervalMs = 200; 26 27 // % threshold for if we should send a new REMB asap. 28 const unsigned int kSendThresholdPercent = 97; 29 30 VieRemb::VieRemb() 31 : list_crit_(CriticalSectionWrapper::CreateCriticalSection()), 32 last_remb_time_(TickTime::MillisecondTimestamp()), 33 last_send_bitrate_(0), 34 bitrate_(0) {} 35 36 VieRemb::~VieRemb() {} 37 38 void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) { 39 assert(rtp_rtcp); 40 41 CriticalSectionScoped cs(list_crit_.get()); 42 if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) != 43 receive_modules_.end()) 44 return; 45 46 // The module probably doesn't have a remote SSRC yet, so don't add it to the 47 // map. 48 receive_modules_.push_back(rtp_rtcp); 49 } 50 51 void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) { 52 assert(rtp_rtcp); 53 54 CriticalSectionScoped cs(list_crit_.get()); 55 for (RtpModules::iterator it = receive_modules_.begin(); 56 it != receive_modules_.end(); ++it) { 57 if ((*it) == rtp_rtcp) { 58 receive_modules_.erase(it); 59 break; 60 } 61 } 62 } 63 64 void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) { 65 assert(rtp_rtcp); 66 67 CriticalSectionScoped cs(list_crit_.get()); 68 69 // Verify this module hasn't been added earlier. 70 if (std::find(rtcp_sender_.begin(), rtcp_sender_.end(), rtp_rtcp) != 71 rtcp_sender_.end()) 72 return; 73 rtcp_sender_.push_back(rtp_rtcp); 74 } 75 76 void VieRemb::RemoveRembSender(RtpRtcp* rtp_rtcp) { 77 assert(rtp_rtcp); 78 79 CriticalSectionScoped cs(list_crit_.get()); 80 for (RtpModules::iterator it = rtcp_sender_.begin(); 81 it != rtcp_sender_.end(); ++it) { 82 if ((*it) == rtp_rtcp) { 83 rtcp_sender_.erase(it); 84 return; 85 } 86 } 87 } 88 89 bool VieRemb::InUse() const { 90 CriticalSectionScoped cs(list_crit_.get()); 91 if (receive_modules_.empty() && rtcp_sender_.empty()) 92 return false; 93 else 94 return true; 95 } 96 97 void VieRemb::OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, 98 unsigned int bitrate) { 99 list_crit_->Enter(); 100 // If we already have an estimate, check if the new total estimate is below 101 // kSendThresholdPercent of the previous estimate. 102 if (last_send_bitrate_ > 0) { 103 unsigned int new_remb_bitrate = last_send_bitrate_ - bitrate_ + bitrate; 104 105 if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) { 106 // The new bitrate estimate is less than kSendThresholdPercent % of the 107 // last report. Send a REMB asap. 108 last_remb_time_ = TickTime::MillisecondTimestamp() - kRembSendIntervalMs; 109 } 110 } 111 bitrate_ = bitrate; 112 113 // Calculate total receive bitrate estimate. 114 int64_t now = TickTime::MillisecondTimestamp(); 115 116 if (now - last_remb_time_ < kRembSendIntervalMs) { 117 list_crit_->Leave(); 118 return; 119 } 120 last_remb_time_ = now; 121 122 if (ssrcs.empty() || receive_modules_.empty()) { 123 list_crit_->Leave(); 124 return; 125 } 126 127 // Send a REMB packet. 128 RtpRtcp* sender = NULL; 129 if (!rtcp_sender_.empty()) { 130 sender = rtcp_sender_.front(); 131 } else { 132 sender = receive_modules_.front(); 133 } 134 last_send_bitrate_ = bitrate_; 135 136 list_crit_->Leave(); 137 138 if (sender) { 139 // TODO(holmer): Change RTP module API to take a const vector reference. 140 sender->SetREMBData(bitrate_, ssrcs.size(), &ssrcs[0]); 141 } 142 } 143 144 } // namespace webrtc 145