Reimplemeted "Test and fix for huge bwe drop after alr state"

webrtc/modules/congestion_controller/send_side_congestion_controller.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.
  */
 
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 
 #include <algorithm>
 #include <memory>
 #include <vector>
 
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h"
+#include "webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h"
 #include "webrtc/modules/congestion_controller/probe_controller.h"
+#include "webrtc/modules/pacing/alr_detector.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/rate_limiter.h"
 #include "webrtc/rtc_base/socket.h"
+#include "webrtc/rtc_base/timeutils.h"
 
 namespace webrtc {
 namespace {
 
 static const int64_t kRetransmitWindowSizeMs = 500;
 
 // Makes sure that the bitrate and the min, max values are in valid range.
 static void ClampBitrates(int* bitrate_bps,
                           int* min_bitrate_bps,
                           int* max_bitrate_bps) {
@@ skipping 56 matching lines @@
           rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
       probe_controller_(new ProbeController(pacer_.get(), clock_)),
       retransmission_rate_limiter_(
           new RateLimiter(clock, kRetransmitWindowSizeMs)),
       transport_feedback_adapter_(clock_),
       last_reported_bitrate_bps_(0),
       last_reported_fraction_loss_(0),
       last_reported_rtt_(0),
       network_state_(kNetworkUp),
       min_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
-      delay_based_bwe_(new DelayBasedBwe(event_log_, clock_)) {
+      delay_based_bwe_(new DelayBasedBwe(event_log_, clock_)),
+      was_in_alr_(0) {
   delay_based_bwe_->SetMinBitrate(min_bitrate_bps_);
 }
 
 SendSideCongestionController::~SendSideCongestionController() {}
 
 void SendSideCongestionController::RegisterPacketFeedbackObserver(
     PacketFeedbackObserver* observer) {
   transport_feedback_adapter_.RegisterPacketFeedbackObserver(observer);
 }
 
@@ skipping 157 matching lines @@
                                         pacing_info);
 }
 
 void SendSideCongestionController::OnTransportFeedback(
     const rtcp::TransportFeedback& feedback) {
   RTC_DCHECK_RUNS_SERIALIZED(&worker_race_);
   transport_feedback_adapter_.OnTransportFeedback(feedback);
   std::vector<PacketFeedback> feedback_vector = ReceivedPacketFeedbackVector(
       transport_feedback_adapter_.GetTransportFeedbackVector());
   SortPacketFeedbackVector(&feedback_vector);
+
+  bool currently_in_alr =
+      pacer_->GetApplicationLimitedRegionStartTime().has_value();
+  if (!currently_in_alr && was_in_alr_) {
+    acknowledged_bitrate_estimator_->SetAlrEndedTimeMs(rtc::TimeMillis());
+  }
+  was_in_alr_ = currently_in_alr;
+
   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
       feedback_vector);
   DelayBasedBwe::Result result;
   {
     rtc::CritScope cs(&bwe_lock_);
     result = delay_based_bwe_->IncomingPacketFeedbackVector(
         feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps());
   }
   if (result.updated)
     bitrate_controller_->OnDelayBasedBweResult(result);
@@ skipping 57 matching lines @@
 bool SendSideCongestionController::IsSendQueueFull() const {
   return pacer_->ExpectedQueueTimeMs() > PacedSender::kMaxQueueLengthMs;
 }
 
 bool SendSideCongestionController::IsNetworkDown() const {
   rtc::CritScope cs(&network_state_lock_);
   return network_state_ == kNetworkDown;
 }
 
 }  // namespace webrtc