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
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(Clock * clock)30 VieRemb::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
~VieRemb()37 VieRemb::~VieRemb() {}
38
AddReceiveChannel(RtpRtcp * rtp_rtcp)39 void 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
RemoveReceiveChannel(RtpRtcp * rtp_rtcp)52 void 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
AddRembSender(RtpRtcp * rtp_rtcp)65 void 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
RemoveRembSender(RtpRtcp * rtp_rtcp)77 void 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
InUse() const90 bool 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
OnReceiveBitrateChanged(const std::vector<unsigned int> & ssrcs,unsigned int bitrate)98 void 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