Fix an error in Audio Network Adaptor: time constant passed wrong.

webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.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.
  */
 
@@ skipping 51 matching lines @@
       FecController::Config::Threshold(
           fec_enabling_threshold.low_bandwidth_bps(),
           fec_enabling_threshold.low_bandwidth_packet_loss(),
           fec_enabling_threshold.high_bandwidth_bps(),
           fec_enabling_threshold.high_bandwidth_packet_loss()),
       FecController::Config::Threshold(
           fec_disabling_threshold.low_bandwidth_bps(),
           fec_disabling_threshold.low_bandwidth_packet_loss(),
           fec_disabling_threshold.high_bandwidth_bps(),
           fec_disabling_threshold.high_bandwidth_packet_loss()),
-      config.has_time_constant_ms(), clock)));
+      config.time_constant_ms(), clock)));
 }
 
 std::unique_ptr<FrameLengthController> CreateFrameLengthController(
     const audio_network_adaptor::config::FrameLengthController& config,
     rtc::ArrayView<const int> encoder_frame_lengths_ms,
     int initial_frame_length_ms) {
   RTC_CHECK(config.has_fl_increasing_packet_loss_fraction());
   RTC_CHECK(config.has_fl_decreasing_packet_loss_fraction());
   RTC_CHECK(config.has_fl_20ms_to_60ms_bandwidth_bps());
   RTC_CHECK(config.has_fl_60ms_to_20ms_bandwidth_bps());
@@ skipping 242 matching lines @@
       NormalizeUplinkBandwidth(scoring_point.uplink_bandwidth_bps) -
       NormalizeUplinkBandwidth(uplink_bandwidth_bps);
   float diff_normalized_packet_loss =
       NormalizePacketLossFraction(scoring_point.uplink_packet_loss_fraction) -
       NormalizePacketLossFraction(uplink_packet_loss_fraction);
   return std::pow(diff_normalized_bitrate_bps, 2) +
          std::pow(diff_normalized_packet_loss, 2);
 }
 
 }  // namespace webrtc