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

webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc

Index: webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index d7d48a372096e15d78f38609d7f140d70b2bec4b..f22c51b7a5f8b11fa512abfa24d56058276e93af 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -8,8 +8,9 @@
  *  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"
 
@@ -263,7 +264,7 @@ struct NetEqNetworkStatsCheck {
   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_;