Migrated GN target :neteq_opus_quality_test

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc

 /*
  *  Copyright (c) 2014 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 258 matching lines @@
   return (drop_this < loss_rate_ * RAND_MAX);
 }
 
 GilbertElliotLoss::GilbertElliotLoss(double prob_trans_11, double prob_trans_01)
     : prob_trans_11_(prob_trans_11),
       prob_trans_01_(prob_trans_01),
       lost_last_(false),
       uniform_loss_model_(new UniformLoss(0)) {
 }
 
+GilbertElliotLoss::~GilbertElliotLoss() {}
+
 bool GilbertElliotLoss::Lost() {
   // Simulate bursty channel (Gilbert model).
   // (1st order) Markov chain model with memory of the previous/last
   // packet state (lost or received).
   if (lost_last_) {
     // Previous packet was not received.
     uniform_loss_model_->set_loss_rate(prob_trans_11_);
     return lost_last_ = uniform_loss_model_->Lost();
   } else {
     uniform_loss_model_->set_loss_rate(prob_trans_01_);
@@ skipping 141 matching lines @@
     }
   }
   Log() << "Average bit rate was "
         << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms
         << " kbps"
         << std::endl;
 }
 
 }  // namespace test
 }  // namespace webrtc