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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/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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