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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(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 | |
37 VieRemb::~VieRemb() {} | |
38 | |
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 | |
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 | |
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 | |
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 | |
90 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 | |
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 | |
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