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

webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc

 /*
  *  Copyright (c) 2013 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 <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
@@ skipping 609 matching lines @@
 
   // Set bitrate.
   EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
                                      channels_ == 1 ? 32000 : 64000));
 
   // Check number of channels for decoder.
   EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
 
   // Encode & decode.
   int16_t audio_type;
-  rtc::scoped_ptr<int16_t[]> output_data_decode(
+  std::unique_ptr<int16_t[]> output_data_decode(
       new int16_t[kPackets * kOpus20msFrameSamples * channels_]);
   OpusRepacketizer* rp = opus_repacketizer_create();
 
   for (int idx = 0; idx < kPackets; idx++) {
     auto speech_block = speech_data_.GetNextBlock();
     encoded_bytes_ =
         WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
                           rtc::CheckedDivExact(speech_block.size(), channels_),
                           kMaxBytes, bitstream_);
     EXPECT_EQ(OPUS_OK, opus_repacketizer_cat(rp, bitstream_, encoded_bytes_));
@@ skipping 15 matching lines @@
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
 }
 
 INSTANTIATE_TEST_CASE_P(VariousMode,
                         OpusTest,
                         Combine(Values(1, 2), Values(0, 1)));
 
 
 }  // namespace webrtc