Fix simulator issue where chokes didn't apply to non-congested packets.

webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc

 /*
  *  Copyright (c) 2015 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.
  *
  */
 
 //  Implementation of Network-Assisted Dynamic Adaptation's (NADA's) proposal.
 //  Version according to Draft Document (mentioned in references)
 //  http://tools.ietf.org/html/draft-zhu-rmcat-nada-06
 //  From March 26, 2015.
 
 #include <math.h>
 #include <algorithm>
 #include <vector>
 #include <iostream>
 
+#include "webrtc/base/common.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
 
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
-const int NadaBweReceiver::kMedian;
-const int NadaBweSender::kMinRefRateKbps;
-const int NadaBweSender::kMaxRefRateKbps;
-const int64_t NadaBweReceiver::kReceivingRateTimeWindowMs;
+const int NadaBweSender::kMinRefRateKbps = 150;
+const int NadaBweSender::kMaxRefRateKbps = 1500;
+const int64_t NadaBweReceiver::kReceivingRateTimeWindowMs = 500;
 
 NadaBweReceiver::NadaBweReceiver(int flow_id)
     : BweReceiver(flow_id),
       clock_(0),
       last_feedback_ms_(0),
       recv_stats_(ReceiveStatistics::Create(&clock_)),
       baseline_delay_ms_(0),
       delay_signal_ms_(0),
       last_congestion_signal_ms_(0),
       last_delays_index_(0),
@@ skipping 13 matching lines @@
   clock_.AdvanceTimeMilliseconds(arrival_time_ms - clock_.TimeInMilliseconds());
   recv_stats_->IncomingPacket(media_packet.header(),
                               media_packet.payload_size(), false);
   int64_t delay_ms = arrival_time_ms -
                      media_packet.creation_time_us() / 1000;  // Refered as x_n.
   // The min should be updated within the first 10 minutes.
   if (clock_.TimeInMilliseconds() < 10 * 60 * 1000) {
     baseline_delay_ms_ = std::min(baseline_delay_ms_, delay_ms);
   }
   delay_signal_ms_ = delay_ms - baseline_delay_ms_;  // Refered as d_n.
+  const int kMedian = ARRAY_SIZE(last_delays_ms_);
   last_delays_ms_[(last_delays_index_++) % kMedian] = delay_signal_ms_;
   int size = std::min(last_delays_index_, kMedian);
   int64_t median_filtered_delay_ms_ = MedianFilter(last_delays_ms_, size);
   exp_smoothed_delay_ms_ = ExponentialSmoothingFilter(
       median_filtered_delay_ms_, exp_smoothed_delay_ms_, kAlpha);
 
   if (exp_smoothed_delay_ms_ < kDelayLowThresholdMs) {
     est_queuing_delay_signal_ms_ = exp_smoothed_delay_ms_;
   } else if (exp_smoothed_delay_ms_ < kDelayMaxThresholdMs) {
     est_queuing_delay_signal_ms_ = static_cast<int64_t>(
@@ skipping 241 matching lines @@
                         (kTheta - (bitrate_kbps_ - kMinRefRateKbps) * x_hat)) /
                            (kTauOMs * kTauOMs) +
                        0.5f);
 
   bitrate_kbps_ = bitrate_kbps_ + smoothing_factor * original_increase;
 }
 
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc