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
23namespace webrtc {
24
25const int kRembSendIntervalMs = 200;
26
27// % threshold for if we should send a new REMB asap.
28const unsigned int kSendThresholdPercent = 97;
29
30VieRemb::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
37VieRemb::~VieRemb() {}
38
39void 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
52void 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
65void 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
77void 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
90bool 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
98void 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