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

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1695763004

Cr-Commit-Position: refs/heads/master@{#11618}

webrtc/modules/audio_coding/test/APITest.h

@@ -11,7 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
@@ -82,8 +83,8 @@ class APITest : public ACMTest {
   bool APIRunB();
 
   //--- ACMs
-  rtc::scoped_ptr<AudioCodingModule> _acmA;
-  rtc::scoped_ptr<AudioCodingModule> _acmB;
+  std::unique_ptr<AudioCodingModule> _acmA;
+  std::unique_ptr<AudioCodingModule> _acmB;
 
   //--- Channels
   Channel* _channel_A2B;

webrtc/modules/audio_coding/test/EncodeDecodeTest.cc

@@ -10,12 +10,12 @@
 
 #include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
+#include <memory>
 #include <sstream>
 #include <stdio.h>
 #include <stdlib.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
@@ -275,7 +275,7 @@ void EncodeDecodeTest::Perform() {
   codePars[1] = 0;
   codePars[2] = 0;
 
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
   struct CodecInst sendCodecTmp;
   numCodecs = acm->NumberOfCodecs();
 
@@ -331,7 +331,7 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
                                            int codeId,
                                            int* codePars,
                                            int testMode) {
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
   RTPFile rtpFile;
   std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
                                                     "encode_decode_rtp");

webrtc/modules/audio_coding/test/PacketLossTest.cc

@@ -10,6 +10,8 @@
 
 #include "webrtc/modules/audio_coding/test/PacketLossTest.h"
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -126,7 +128,7 @@ void PacketLossTest::Perform() {
 #ifndef WEBRTC_CODEC_OPUS
   return;
 #else
-  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
 
   int codec_id = acm->Codec("opus", 48000, channels_);
 

webrtc/modules/audio_coding/test/PacketLossTest.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
 
+#include <memory>
 #include <string>
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
 
 namespace webrtc {
@@ -55,8 +55,8 @@ class PacketLossTest : public ACMTest {
   int channels_;
   std::string in_file_name_;
   int sample_rate_hz_;
-  rtc::scoped_ptr<SenderWithFEC> sender_;
-  rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_;
+  std::unique_ptr<SenderWithFEC> sender_;
+  std::unique_ptr<ReceiverWithPacketLoss> receiver_;
   int expected_loss_rate_;
   int actual_loss_rate_;
   int burst_length_;

webrtc/modules/audio_coding/test/SpatialAudio.h

@@ -11,7 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
@@ -33,9 +34,9 @@ class SpatialAudio : public ACMTest {
   void EncodeDecode(double leftPanning, double rightPanning);
   void EncodeDecode();
 
-  rtc::scoped_ptr<AudioCodingModule> _acmLeft;
-  rtc::scoped_ptr<AudioCodingModule> _acmRight;
-  rtc::scoped_ptr<AudioCodingModule> _acmReceiver;
+  std::unique_ptr<AudioCodingModule> _acmLeft;
+  std::unique_ptr<AudioCodingModule> _acmRight;
+  std::unique_ptr<AudioCodingModule> _acmReceiver;
   Channel* _channel;
   PCMFile _inFile;
   PCMFile _outFile;

webrtc/modules/audio_coding/test/TestAllCodecs.h

@@ -11,7 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
@@ -69,8 +70,8 @@ class TestAllCodecs : public ACMTest {
   void DisplaySendReceiveCodec();
 
   int test_mode_;
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+  std::unique_ptr<AudioCodingModule> acm_a_;
+  std::unique_ptr<AudioCodingModule> acm_b_;
   TestPack* channel_a_to_b_;
   PCMFile infile_a_;
   PCMFile outfile_b_;

webrtc/modules/audio_coding/test/TestRedFec.h

@@ -11,8 +11,9 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
 
+#include <memory>
 #include <string>
-#include "webrtc/base/scoped_ptr.h"
+
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
@@ -36,8 +37,8 @@ class TestRedFec : public ACMTest {
   void Run();
   void OpenOutFile(int16_t testNumber);
   int32_t SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
-  rtc::scoped_ptr<AudioCodingModule> _acmA;
-  rtc::scoped_ptr<AudioCodingModule> _acmB;
+  std::unique_ptr<AudioCodingModule> _acmA;
+  std::unique_ptr<AudioCodingModule> _acmB;
 
   Channel* _channelA2B;
 

webrtc/modules/audio_coding/test/TestStereo.h

@@ -13,7 +13,8 @@
 
 #include <math.h>
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
@@ -82,8 +83,8 @@ class TestStereo : public ACMTest {
 
   int test_mode_;
 
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+  std::unique_ptr<AudioCodingModule> acm_a_;
+  std::unique_ptr<AudioCodingModule> acm_b_;
 
   TestPackStereo* channel_a2b_;
 

webrtc/modules/audio_coding/test/TestVADDTX.h

@@ -11,8 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
 
+#include <memory>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
@@ -68,10 +68,10 @@ class TestVadDtx : public ACMTest {
   void Run(std::string in_filename, int frequency, int channels,
            std::string out_filename, bool append, const int* expects);
 
-  rtc::scoped_ptr<AudioCodingModule> acm_send_;
-  rtc::scoped_ptr<AudioCodingModule> acm_receive_;
-  rtc::scoped_ptr<Channel> channel_;
-  rtc::scoped_ptr<ActivityMonitor> monitor_;
+  std::unique_ptr<AudioCodingModule> acm_send_;
+  std::unique_ptr<AudioCodingModule> acm_receive_;
+  std::unique_ptr<Channel> channel_;
+  std::unique_ptr<ActivityMonitor> monitor_;
 };
 
 // TestWebRtcVadDtx is to verify that the WebRTC VAD/DTX perform as they should.

webrtc/modules/audio_coding/test/TwoWayCommunication.cc

@@ -14,6 +14,8 @@
 #include <stdio.h>
 #include <string.h>
 
+#include <memory>
+
 #ifdef WIN32
 #include <Windows.h>
 #endif
@@ -66,7 +68,7 @@ TwoWayCommunication::~TwoWayCommunication() {
 
 void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
                                       uint8_t* codecID_B) {
-  rtc::scoped_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
+  std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
   uint8_t noCodec = tmpACM->NumberOfCodecs();
   CodecInst codecInst;
   printf("List of Supported Codecs\n");

webrtc/modules/audio_coding/test/TwoWayCommunication.h

@@ -11,7 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
 #define WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
@@ -31,11 +32,11 @@ class TwoWayCommunication : public ACMTest {
   void SetUp();
   void SetUpAutotest();
 
-  rtc::scoped_ptr<AudioCodingModule> _acmA;
-  rtc::scoped_ptr<AudioCodingModule> _acmB;
+  std::unique_ptr<AudioCodingModule> _acmA;
+  std::unique_ptr<AudioCodingModule> _acmB;
 
-  rtc::scoped_ptr<AudioCodingModule> _acmRefA;
-  rtc::scoped_ptr<AudioCodingModule> _acmRefB;
+  std::unique_ptr<AudioCodingModule> _acmRefA;
+  std::unique_ptr<AudioCodingModule> _acmRefB;
 
   Channel* _channel_A2B;
   Channel* _channel_B2A;

webrtc/modules/audio_coding/test/delay_test.cc

@@ -12,10 +12,10 @@
 #include <math.h>
 
 #include <iostream>
+#include <memory>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
@@ -223,8 +223,8 @@ class DelayTest {
     out_file_b_.Close();
   }
 
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+  std::unique_ptr<AudioCodingModule> acm_a_;
+  std::unique_ptr<AudioCodingModule> acm_b_;
 
   Channel* channel_a2b_;
 

webrtc/modules/audio_coding/test/iSACTest.h

@@ -13,7 +13,8 @@
 
 #include <string.h>
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
@@ -51,11 +52,11 @@ class ISACTest : public ACMTest {
 
   void SwitchingSamplingRate(int testNr, int maxSampRateChange);
 
-  rtc::scoped_ptr<AudioCodingModule> _acmA;
-  rtc::scoped_ptr<AudioCodingModule> _acmB;
+  std::unique_ptr<AudioCodingModule> _acmA;
+  std::unique_ptr<AudioCodingModule> _acmB;
 
-  rtc::scoped_ptr<Channel> _channel_A2B;
-  rtc::scoped_ptr<Channel> _channel_B2A;
+  std::unique_ptr<Channel> _channel_A2B;
+  std::unique_ptr<Channel> _channel_B2A;
 
   PCMFile _inFileA;
   PCMFile _inFileB;

webrtc/modules/audio_coding/test/insert_packet_with_timing.cc

@@ -10,9 +10,10 @@
 
 #include <stdio.h>
 
+#include <memory>
+
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
@@ -241,8 +242,8 @@ class InsertPacketWithTiming {
   SimulatedClock* sender_clock_;
   SimulatedClock* receiver_clock_;
 
-  rtc::scoped_ptr<AudioCodingModule> send_acm_;
-  rtc::scoped_ptr<AudioCodingModule> receive_acm_;
+  std::unique_ptr<AudioCodingModule> send_acm_;
+  std::unique_ptr<AudioCodingModule> receive_acm_;
   Channel* channel_;
 
   FILE* seq_num_fid_;  // Input (text), one sequence number per line.

webrtc/modules/audio_coding/test/opus_test.h

@@ -13,7 +13,8 @@
 
 #include <math.h>
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/test/ACMTest.h"
@@ -39,7 +40,7 @@ class OpusTest : public ACMTest {
 
   void OpenOutFile(int test_number);
 
-  rtc::scoped_ptr<AudioCodingModule> acm_receiver_;
+  std::unique_ptr<AudioCodingModule> acm_receiver_;
   TestPackStereo* channel_a2b_;
   PCMFile in_file_stereo_;
   PCMFile in_file_mono_;

webrtc/modules/audio_coding/test/target_delay_unittest.cc

@@ -8,8 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
@@ -193,7 +194,7 @@ class TargetDelayTest : public ::testing::Test {
     return acm_->LeastRequiredDelayMs();
   }
 
-  rtc::scoped_ptr<AudioCodingModule> acm_;
+  std::unique_ptr<AudioCodingModule> acm_;
   WebRtcRTPHeader rtp_info_;
   uint8_t payload_[kPayloadLenBytes];
 };