Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/

webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.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.
  */
 
+#include <memory>
+
 #include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 
 namespace webrtc {
 namespace test {
 
 using ::testing::_;
 using ::testing::SetArgPointee;
 using ::testing::Return;
 
@@ skipping 233 matching lines @@
     SetPacketLossRate(1);
     expects.stats_ref.expand_rate = 16384;
     expects.stats_ref.speech_expand_rate = 5324;
     RunTest(10, expects);  // Lost 10 * 20ms in a row.
   }
 
  private:
   MockAudioDecoder* external_decoder_;
   const int samples_per_ms_;
   const size_t frame_size_samples_;
-  rtc::scoped_ptr<test::RtpGenerator> rtp_generator_;
+  std::unique_ptr<test::RtpGenerator> rtp_generator_;
   WebRtcRTPHeader rtp_header_;
   uint32_t last_lost_time_;
   uint32_t packet_loss_interval_;
   uint8_t payload_[kPayloadSizeByte];
   int16_t output_[kMaxOutputSize];
 };
 
 TEST(NetEqNetworkStatsTest, DecodeFec) {
   MockAudioDecoder decoder(1);
   NetEqNetworkStatsTest test(NetEqDecoder::kDecoderOpus, &decoder);
@@ skipping 14 matching lines @@
   test.NoiseExpansionTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 }  // namespace test
 }  // namespace webrtc