Test and fix for huge bwe drop after alr state.

webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc

 /*
  *  Copyright (c) 2016 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/delay_based_bwe_unittest_helper.h"
@@ skipping 133 matching lines @@
     ++i;
   }
   it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
   return std::max((*it)->next_rtp_time(), time_now_us);
 }
 }  // namespace test
 
 DelayBasedBweTest::DelayBasedBweTest()
     : clock_(100000000),
       acknowledged_bitrate_estimator_(
-          rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
+          rtc::MakeUnique<AcknowledgedBitrateEstimator>(
+              rtc::MakeUnique<BitrateEstimatorCreator>())),
       bitrate_estimator_(new DelayBasedBwe(nullptr, &clock_)),
       stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
                                                   clock_.TimeInMicroseconds())),
       arrival_time_offset_ms_(0),
       first_update_(true) {}
 
 DelayBasedBweTest::~DelayBasedBweTest() {}
 
 void DelayBasedBweTest::AddDefaultStream() {
   stream_generator_->AddStream(new test::RtpStream(30, 3e5));
@@ skipping 12 matching lines @@
 void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
                                          int64_t send_time_ms,
                                          uint16_t sequence_number,
                                          size_t payload_size,
                                          const PacedPacketInfo& pacing_info) {
   RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
   PacketFeedback packet(arrival_time_ms + arrival_time_offset_ms_, send_time_ms,
                         sequence_number, payload_size, pacing_info);
   std::vector<PacketFeedback> packets;
   packets.push_back(packet);
-  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets, false);
   DelayBasedBwe::Result result =
       bitrate_estimator_->IncomingPacketFeedbackVector(
           packets, acknowledged_bitrate_estimator_->bitrate_bps());
   const uint32_t kDummySsrc = 0;
   if (result.updated) {
     bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
                                               result.target_bitrate_bps);
   }
 }
 
@@ skipping 14 matching lines @@
 
   bool overuse = false;
   bitrate_observer_.Reset();
   clock_.AdvanceTimeMicroseconds(1000 * packets.back().arrival_time_ms -
                                  clock_.TimeInMicroseconds());
   for (auto& packet : packets) {
     RTC_CHECK_GE(packet.arrival_time_ms + arrival_time_offset_ms_, 0);
     packet.arrival_time_ms += arrival_time_offset_ms_;
   }
 
-  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets, false);
   DelayBasedBwe::Result result =
       bitrate_estimator_->IncomingPacketFeedbackVector(
           packets, acknowledged_bitrate_estimator_->bitrate_bps());
   const uint32_t kDummySsrc = 0;
   if (result.updated) {
     bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
                                               result.target_bitrate_bps);
     if (!first_update_ && result.target_bitrate_bps < bitrate_bps)
       overuse = true;
     first_update_ = false;
@@ skipping 271 matching lines @@
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
                      sequence_number++, 1000);
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   uint32_t bitrate_after = 0;
   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
   EXPECT_LT(bitrate_after, bitrate_before);
 }
 }  // namespace webrtc