Migrated GN target :neteq_opus_quality_test

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

 /*
  *  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 33 matching lines @@
   bool Lost() override;
   void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
 
  private:
   double loss_rate_;
 };
 
 class GilbertElliotLoss : public LossModel {
  public:
   GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
+  ~GilbertElliotLoss() override;
   bool Lost() override;
 
  private:
   // Prob. of losing current packet, when previous packet is lost.
   double prob_trans_11_;
   // Prob. of losing current packet, when previous packet is not lost.
   double prob_trans_01_;
   bool lost_last_;
   std::unique_ptr<UniformLoss> uniform_loss_model_;
 };
 
 class NetEqQualityTest : public ::testing::Test {
  protected:
   NetEqQualityTest(int block_duration_ms,
                    int in_sampling_khz,
                    int out_sampling_khz,
                    NetEqDecoder decoder_type);
-  virtual ~NetEqQualityTest();
+  ~NetEqQualityTest() override;
 
   void SetUp() override;
 
   // EncodeBlock(...) does the following:
   // 1. encodes a block of audio, saved in |in_data| and has a length of
   // |block_size_samples| (samples per channel),
   // 2. save the bit stream to |payload| of |max_bytes| bytes in size,
   // 3. returns the length of the payload (in bytes),
   virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
                           rtc::Buffer* payload, size_t max_bytes) = 0;
@@ skipping 48 matching lines @@
   AudioFrame out_frame_;
   WebRtcRTPHeader rtp_header_;
 
   size_t total_payload_size_bytes_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_