Support REMB in combination with send-side BWE.

webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 37 matching lines @@
       expected_packets_since_last_loss_update_(0),
       bitrate_(0),
       min_bitrate_configured_(kDefaultMinBitrateBps),
       max_bitrate_configured_(kDefaultMaxBitrateBps),
       last_low_bitrate_log_ms_(-1),
       has_decreased_since_last_fraction_loss_(false),
       time_last_receiver_block_ms_(-1),
       last_fraction_loss_(0),
       last_round_trip_time_ms_(0),
       bwe_incoming_(0),
+      delay_based_bitrate_bps_(0),
       time_last_decrease_ms_(0),
       first_report_time_ms_(-1),
       initially_lost_packets_(0),
       bitrate_at_2_seconds_kbps_(0),
       uma_update_state_(kNoUpdate),
       rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
       event_log_(nullptr) {}
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
@@ skipping 29 matching lines @@
   *loss = last_fraction_loss_;
   *rtt = last_round_trip_time_ms_;
 }
 
 void SendSideBandwidthEstimation::UpdateReceiverEstimate(
     int64_t now_ms, uint32_t bandwidth) {
   bwe_incoming_ = bandwidth;
   bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
 }
 
+void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(
+    int64_t now_ms,
+    uint32_t bitrate_bps) {
+  delay_based_bitrate_bps_ = bitrate_bps;
+  bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
+}
+
 void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
                                                       int64_t rtt,
                                                       int number_of_packets,
                                                       int64_t now_ms) {
   if (first_report_time_ms_ == -1)
     first_report_time_ms_ = now_ms;
 
   // Update RTT.
   last_round_trip_time_ms_ = rtt;
 
@@ skipping 144 matching lines @@
   }
 
   min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
 }
 
 uint32_t SendSideBandwidthEstimation::CapBitrateToThresholds(
     int64_t now_ms, uint32_t bitrate) {
   if (bwe_incoming_ > 0 && bitrate > bwe_incoming_) {
     bitrate = bwe_incoming_;
   }
+  if (delay_based_bitrate_bps_ > 0 && bitrate > delay_based_bitrate_bps_) {
+    bitrate = delay_based_bitrate_bps_;
+  }
   if (bitrate > max_bitrate_configured_) {
     bitrate = max_bitrate_configured_;
   }
   if (bitrate < min_bitrate_configured_) {
     if (last_low_bitrate_log_ms_ == -1 ||
         now_ms - last_low_bitrate_log_ms_ > kLowBitrateLogPeriodMs) {
       LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate / 1000
                       << " kbps is below configured min bitrate "
                       << min_bitrate_configured_ / 1000 << " kbps.";
       last_low_bitrate_log_ms_ = now_ms;
     }
     bitrate = min_bitrate_configured_;
   }
   return bitrate;
 }
 
 void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
   event_log_ = event_log;
 }
 
 }  // namespace webrtc