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| 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/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 | |
| 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 sender->SetREMBData(bitrate_, ssrcs); | |
| 140 } | |
| 141 } | |
| 142 | |
| 143 } // namespace webrtc | |
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Chromium Code Reviews
Issue 1506773002:
Merge webrtc/video_engine/ into webrtc/video/ (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master