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