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
VieRemb()30 VieRemb::VieRemb()
31 : list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
32 last_remb_time_(TickTime::MillisecondTimestamp()),
33 last_send_bitrate_(0),
34 bitrate_(0) {}
35
~VieRemb()36 VieRemb::~VieRemb() {}
37
AddReceiveChannel(RtpRtcp * rtp_rtcp)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
RemoveReceiveChannel(RtpRtcp * rtp_rtcp)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
AddRembSender(RtpRtcp * rtp_rtcp)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
RemoveRembSender(RtpRtcp * rtp_rtcp)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
InUse() const89 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
OnReceiveBitrateChanged(const std::vector<unsigned int> & ssrcs,unsigned int bitrate)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