Make the BWE threshold adaptive.

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc

 /*
  *  Copyright (c) 2013 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 278 matching lines @@
   if (!inter_arrival_.get()) {
     inter_arrival_.reset(
         new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
                          kTimestampToMs, true));
   }
   if (inter_arrival_->ComputeDeltas(timestamp, arrival_time_ms, payload_size,
                                     &ts_delta, &t_delta, &size_delta)) {
     double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
     estimator_.Update(t_delta, ts_delta_ms, size_delta, detector_.State());
     detector_.Detect(estimator_.offset(), ts_delta_ms,
-                     estimator_.num_of_deltas(), now_ms);
+                     estimator_.num_of_deltas(), arrival_time_ms);
     UpdateStats(static_cast<int>(t_delta - ts_delta_ms), now_ms);
   }
   if (detector_.State() == kBwOverusing) {
-    unsigned int incoming_bitrate = incoming_bitrate_.Rate(now_ms);
+    uint32_t incoming_bitrate_bps = incoming_bitrate_.Rate(now_ms);
     if (prior_state != kBwOverusing ||
-        remote_rate_.TimeToReduceFurther(now_ms, incoming_bitrate)) {
+        remote_rate_.TimeToReduceFurther(now_ms, incoming_bitrate_bps)) {
       // The first overuse should immediately trigger a new estimate.
       // We also have to update the estimate immediately if we are overusing
       // and the target bitrate is too high compared to what we are receiving.
       UpdateEstimate(now_ms);
     }
   }
 }
 
 int32_t RemoteBitrateEstimatorAbsSendTime::Process() {
   if (TimeUntilNextProcess() > 0) {
@@ skipping 34 matching lines @@
     // We can't update the estimate if we don't have any active streams.
     inter_arrival_.reset();
     // We deliberately don't reset the first_packet_time_ms_ here for now since
     // we only probe for bandwidth in the beginning of a call right now.
     return;
   }
 
   const RateControlInput input(detector_.State(),
                                incoming_bitrate_.Rate(now_ms),
                                estimator_.var_noise());
-  const RateControlRegion region = remote_rate_.Update(&input, now_ms);
+  remote_rate_.Update(&input, now_ms);
   unsigned int target_bitrate = remote_rate_.UpdateBandwidthEstimate(now_ms);
   if (remote_rate_.ValidEstimate()) {
     process_interval_ms_ = remote_rate_.GetFeedbackInterval();
     observer_->OnReceiveBitrateChanged(Keys(ssrcs_), target_bitrate);
   }
-  detector_.SetRateControlRegion(region);
 }
 
 void RemoteBitrateEstimatorAbsSendTime::OnRttUpdate(int64_t rtt) {
   CriticalSectionScoped cs(crit_sect_.get());
   remote_rate_.SetRtt(rtt);
 }
 
 void RemoteBitrateEstimatorAbsSendTime::RemoveStream(unsigned int ssrc) {
   CriticalSectionScoped cs(crit_sect_.get());
   ssrcs_.erase(ssrc);
@@ skipping 47 matching lines @@
 
   RemoveStaleEntries(
       &recent_update_time_ms_,
       &recent_propagation_delta_ms_,
       now_ms - kPropagationDeltaQueueMaxTimeMs);
 
   total_propagation_delta_ms_ =
       std::max(total_propagation_delta_ms_ + propagation_delta_ms, 0);
 }
 }  // namespace webrtc