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}
diff --git a/webrtc/modules/audio_coding/test/APITest.h b/webrtc/modules/audio_coding/test/APITest.h
index a1937c2..af2a3a1 100644
--- a/webrtc/modules/audio_coding/test/APITest.h
+++ b/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 @@
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;
diff --git a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
index ba3c8d9..e063224 100644
--- a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/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 @@
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 @@
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");
diff --git a/webrtc/modules/audio_coding/test/PacketLossTest.cc b/webrtc/modules/audio_coding/test/PacketLossTest.cc
index ad3e834..891471d 100644
--- a/webrtc/modules/audio_coding/test/PacketLossTest.cc
+++ b/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 @@
#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_);
diff --git a/webrtc/modules/audio_coding/test/PacketLossTest.h b/webrtc/modules/audio_coding/test/PacketLossTest.h
index f3570ae..705fe73 100644
--- a/webrtc/modules/audio_coding/test/PacketLossTest.h
+++ b/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 @@
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_;
diff --git a/webrtc/modules/audio_coding/test/SpatialAudio.h b/webrtc/modules/audio_coding/test/SpatialAudio.h
index 3548cc9..270c370 100644
--- a/webrtc/modules/audio_coding/test/SpatialAudio.h
+++ b/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 @@
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;
diff --git a/webrtc/modules/audio_coding/test/TestAllCodecs.h b/webrtc/modules/audio_coding/test/TestAllCodecs.h
index e79bd69..6d6f380 100644
--- a/webrtc/modules/audio_coding/test/TestAllCodecs.h
+++ b/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 @@
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_;
diff --git a/webrtc/modules/audio_coding/test/TestRedFec.h b/webrtc/modules/audio_coding/test/TestRedFec.h
index 6343d8e..e936f75 100644
--- a/webrtc/modules/audio_coding/test/TestRedFec.h
+++ 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 @@
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;
diff --git a/webrtc/modules/audio_coding/test/TestStereo.h b/webrtc/modules/audio_coding/test/TestStereo.h
index 4526be6..3489421 100644
--- a/webrtc/modules/audio_coding/test/TestStereo.h
+++ b/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 @@
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_;
diff --git a/webrtc/modules/audio_coding/test/TestVADDTX.h b/webrtc/modules/audio_coding/test/TestVADDTX.h
index 1e7f0ef..893babc 100644
--- a/webrtc/modules/audio_coding/test/TestVADDTX.h
+++ b/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 @@
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.
diff --git a/webrtc/modules/audio_coding/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
index 56e136b..3ca7fd2 100644
--- a/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/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 @@
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");
diff --git a/webrtc/modules/audio_coding/test/TwoWayCommunication.h b/webrtc/modules/audio_coding/test/TwoWayCommunication.h
index 7763993..f9d37f7 100644
--- a/webrtc/modules/audio_coding/test/TwoWayCommunication.h
+++ b/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 @@
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;
diff --git a/webrtc/modules/audio_coding/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc
index a8c137f..7288d50 100644
--- a/webrtc/modules/audio_coding/test/delay_test.cc
+++ b/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 @@
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_;
diff --git a/webrtc/modules/audio_coding/test/iSACTest.h b/webrtc/modules/audio_coding/test/iSACTest.h
index c5bb515..7d3a77e 100644
--- a/webrtc/modules/audio_coding/test/iSACTest.h
+++ b/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 @@
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;
diff --git a/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
index 481df55..966f4c6 100644
--- a/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/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 @@
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.
diff --git a/webrtc/modules/audio_coding/test/opus_test.h b/webrtc/modules/audio_coding/test/opus_test.h
index 93c9ffb..ce570f6 100644
--- a/webrtc/modules/audio_coding/test/opus_test.h
+++ b/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 @@
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_;
diff --git a/webrtc/modules/audio_coding/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
index 195e9d8..99c1c2d 100644
--- a/webrtc/modules/audio_coding/test/target_delay_unittest.cc
+++ b/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 @@
return acm_->LeastRequiredDelayMs();
}
- rtc::scoped_ptr<AudioCodingModule> acm_;
+ std::unique_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_info_;
uint8_t payload_[kPayloadLenBytes];
};