diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc index 28d2f63ef7..1cf7924fb7 100644 --- a/webrtc/api/webrtcsession.cc +++ b/webrtc/api/webrtcsession.cc @@ -1241,7 +1241,7 @@ void WebRtcSession::SetRawAudioSink(uint32_t ssrc, if (!voice_channel_) return; - voice_channel_->SetRawAudioSink(ssrc, rtc::ScopedToUnique(std::move(sink))); + voice_channel_->SetRawAudioSink(ssrc, std::move(sink)); } RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const { diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 61eb80a8b3..67abc5669d 100644 --- a/webrtc/call/call.cc +++ b/webrtc/call/call.cc @@ -191,9 +191,8 @@ namespace internal { Call::Call(const Call::Config& config) : clock_(Clock::GetRealTimeClock()), num_cpu_cores_(CpuInfo::DetectNumberOfCores()), - module_process_thread_( - rtc::ScopedToUnique(ProcessThread::Create("ModuleProcessThread"))), - pacer_thread_(rtc::ScopedToUnique(ProcessThread::Create("PacerThread"))), + module_process_thread_(ProcessThread::Create("ModuleProcessThread")), + pacer_thread_(ProcessThread::Create("PacerThread")), call_stats_(new CallStats(clock_)), bitrate_allocator_(new BitrateAllocator()), config_(config), diff --git a/webrtc/common_audio/lapped_transform.cc b/webrtc/common_audio/lapped_transform.cc index 0edf586d78..5ab1db1b25 100644 --- a/webrtc/common_audio/lapped_transform.cc +++ b/webrtc/common_audio/lapped_transform.cc @@ -72,8 +72,7 @@ LappedTransform::LappedTransform(size_t num_in_channels, window, shift_amount, &blocker_callback_), - fft_(rtc::ScopedToUnique( - RealFourier::Create(RealFourier::FftOrder(block_length_)))), + fft_(RealFourier::Create(RealFourier::FftOrder(block_length_))), cplx_length_(RealFourier::ComplexLength(fft_->order())), real_buf_(num_in_channels, block_length_, diff --git a/webrtc/modules/audio_device/android/audio_manager.cc b/webrtc/modules/audio_device/android/audio_manager.cc index 9174a5b7ab..01e5d5fe4f 100644 --- a/webrtc/modules/audio_device/android/audio_manager.cc +++ b/webrtc/modules/audio_device/android/audio_manager.cc @@ -66,7 +66,7 @@ bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() { // AudioManager implementation AudioManager::AudioManager() - : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())), + : j_environment_(JVM::GetInstance()->environment()), audio_layer_(AudioDeviceModule::kPlatformDefaultAudio), initialized_(false), hardware_aec_(false), @@ -80,14 +80,14 @@ AudioManager::AudioManager() {"nativeCacheAudioParameters", "(IIZZZZIIJ)V", reinterpret_cast(&webrtc::AudioManager::CacheAudioParameters)}}; - j_native_registration_ = rtc::ScopedToUnique(j_environment_->RegisterNatives( - "org/webrtc/voiceengine/WebRtcAudioManager", - native_methods, arraysize(native_methods))); + j_native_registration_ = j_environment_->RegisterNatives( + "org/webrtc/voiceengine/WebRtcAudioManager", native_methods, + arraysize(native_methods)); j_audio_manager_.reset(new JavaAudioManager( j_native_registration_.get(), - rtc::ScopedToUnique(j_native_registration_->NewObject( + j_native_registration_->NewObject( "", "(Landroid/content/Context;J)V", - JVM::GetInstance()->context(), PointerTojlong(this))))); + JVM::GetInstance()->context(), PointerTojlong(this)))); } AudioManager::~AudioManager() { diff --git a/webrtc/modules/audio_device/android/audio_record_jni.cc b/webrtc/modules/audio_device/android/audio_record_jni.cc index 5ff59971cf..8ce13861a4 100644 --- a/webrtc/modules/audio_device/android/audio_record_jni.cc +++ b/webrtc/modules/audio_device/android/audio_record_jni.cc @@ -74,7 +74,7 @@ bool AudioRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) { // AudioRecordJni implementation. AudioRecordJni::AudioRecordJni(AudioManager* audio_manager) - : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())), + : j_environment_(JVM::GetInstance()->environment()), audio_manager_(audio_manager), audio_parameters_(audio_manager->GetRecordAudioParameters()), total_delay_in_milliseconds_(0), @@ -93,14 +93,14 @@ AudioRecordJni::AudioRecordJni(AudioManager* audio_manager) &webrtc::AudioRecordJni::CacheDirectBufferAddress)}, {"nativeDataIsRecorded", "(IJ)V", reinterpret_cast(&webrtc::AudioRecordJni::DataIsRecorded)}}; - j_native_registration_ = rtc::ScopedToUnique(j_environment_->RegisterNatives( - "org/webrtc/voiceengine/WebRtcAudioRecord", - native_methods, arraysize(native_methods))); + j_native_registration_ = j_environment_->RegisterNatives( + "org/webrtc/voiceengine/WebRtcAudioRecord", native_methods, + arraysize(native_methods)); j_audio_record_.reset(new JavaAudioRecord( j_native_registration_.get(), - rtc::ScopedToUnique(j_native_registration_->NewObject( + j_native_registration_->NewObject( "", "(Landroid/content/Context;J)V", - JVM::GetInstance()->context(), PointerTojlong(this))))); + JVM::GetInstance()->context(), PointerTojlong(this)))); // Detach from this thread since we want to use the checker to verify calls // from the Java based audio thread. thread_checker_java_.DetachFromThread(); diff --git a/webrtc/modules/audio_device/android/audio_track_jni.cc b/webrtc/modules/audio_device/android/audio_track_jni.cc index 67837f51fb..fc77e32e23 100644 --- a/webrtc/modules/audio_device/android/audio_track_jni.cc +++ b/webrtc/modules/audio_device/android/audio_track_jni.cc @@ -69,7 +69,7 @@ int AudioTrackJni::JavaAudioTrack::GetStreamVolume() { // TODO(henrika): possible extend usage of AudioManager and add it as member. AudioTrackJni::AudioTrackJni(AudioManager* audio_manager) - : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())), + : j_environment_(JVM::GetInstance()->environment()), audio_parameters_(audio_manager->GetPlayoutAudioParameters()), direct_buffer_address_(nullptr), direct_buffer_capacity_in_bytes_(0), @@ -86,14 +86,14 @@ AudioTrackJni::AudioTrackJni(AudioManager* audio_manager) &webrtc::AudioTrackJni::CacheDirectBufferAddress)}, {"nativeGetPlayoutData", "(IJ)V", reinterpret_cast(&webrtc::AudioTrackJni::GetPlayoutData)}}; - j_native_registration_ = rtc::ScopedToUnique(j_environment_->RegisterNatives( - "org/webrtc/voiceengine/WebRtcAudioTrack", - native_methods, arraysize(native_methods))); + j_native_registration_ = j_environment_->RegisterNatives( + "org/webrtc/voiceengine/WebRtcAudioTrack", native_methods, + arraysize(native_methods)); j_audio_track_.reset(new JavaAudioTrack( j_native_registration_.get(), - rtc::ScopedToUnique(j_native_registration_->NewObject( + j_native_registration_->NewObject( "", "(Landroid/content/Context;J)V", - JVM::GetInstance()->context(), PointerTojlong(this))))); + JVM::GetInstance()->context(), PointerTojlong(this)))); // Detach from this thread since we want to use the checker to verify calls // from the Java based audio thread. thread_checker_java_.DetachFromThread(); diff --git a/webrtc/modules/audio_device/android/build_info.cc b/webrtc/modules/audio_device/android/build_info.cc index c6cecc96c5..455c12f7fd 100644 --- a/webrtc/modules/audio_device/android/build_info.cc +++ b/webrtc/modules/audio_device/android/build_info.cc @@ -15,10 +15,9 @@ namespace webrtc { BuildInfo::BuildInfo() - : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())), - j_build_info_(JVM::GetInstance()->GetClass( - "org/webrtc/voiceengine/BuildInfo")) { -} + : j_environment_(JVM::GetInstance()->environment()), + j_build_info_( + JVM::GetInstance()->GetClass("org/webrtc/voiceengine/BuildInfo")) {} std::string BuildInfo::GetStringFromJava(const char* name) { jmethodID id = j_build_info_.GetStaticMethodId(name, "()Ljava/lang/String;"); diff --git a/webrtc/modules/audio_device/test/audio_device_test_api.cc b/webrtc/modules/audio_device/test/audio_device_test_api.cc index f37d89cd9c..78642c3292 100644 --- a/webrtc/modules/audio_device/test/audio_device_test_api.cc +++ b/webrtc/modules/audio_device/test/audio_device_test_api.cc @@ -142,8 +142,7 @@ class AudioDeviceAPITest: public testing::Test { virtual ~AudioDeviceAPITest() {} static void SetUpTestCase() { - process_thread_ = - rtc::ScopedToUnique(ProcessThread::Create("ProcessThread")); + process_thread_ = ProcessThread::Create("ProcessThread"); process_thread_->Start(); // Windows: diff --git a/webrtc/modules/audio_device/test/func_test_manager.cc b/webrtc/modules/audio_device/test/func_test_manager.cc index bb7686c6c1..cc589653f7 100644 --- a/webrtc/modules/audio_device/test/func_test_manager.cc +++ b/webrtc/modules/audio_device/test/func_test_manager.cc @@ -594,11 +594,10 @@ FuncTestManager::~FuncTestManager() int32_t FuncTestManager::Init() { - EXPECT_TRUE((_processThread = rtc::ScopedToUnique( - ProcessThread::Create("ProcessThread"))) != NULL); - if (_processThread == NULL) - { - return -1; + EXPECT_TRUE((_processThread = ProcessThread::Create("ProcessThread")) != + NULL); + if (_processThread == NULL) { + return -1; } _processThread->Start(); @@ -832,8 +831,8 @@ int32_t FuncTestManager::TestAudioLayerSelection() // ================================================== // Next, try to make fresh start with new audio layer - EXPECT_TRUE((_processThread = rtc::ScopedToUnique( - ProcessThread::Create("ProcessThread"))) != NULL); + EXPECT_TRUE((_processThread = ProcessThread::Create("ProcessThread")) != + NULL); if (_processThread == NULL) { return -1; diff --git a/webrtc/modules/desktop_capture/desktop_frame.cc b/webrtc/modules/desktop_capture/desktop_frame.cc index 6bc7b2e38f..3278ed46dc 100644 --- a/webrtc/modules/desktop_capture/desktop_frame.cc +++ b/webrtc/modules/desktop_capture/desktop_frame.cc @@ -84,8 +84,7 @@ std::unique_ptr SharedMemoryDesktopFrame::Create( size_t buffer_size = size.width() * size.height() * DesktopFrame::kBytesPerPixel; std::unique_ptr shared_memory; - shared_memory = rtc::ScopedToUnique( - shared_memory_factory->CreateSharedMemory(buffer_size)); + shared_memory = shared_memory_factory->CreateSharedMemory(buffer_size); if (!shared_memory) return nullptr; diff --git a/webrtc/modules/desktop_capture/desktop_frame_win.cc b/webrtc/modules/desktop_capture/desktop_frame_win.cc index e91e37eb5a..624b729203 100644 --- a/webrtc/modules/desktop_capture/desktop_frame_win.cc +++ b/webrtc/modules/desktop_capture/desktop_frame_win.cc @@ -49,8 +49,7 @@ DesktopFrameWin* DesktopFrameWin::Create( std::unique_ptr shared_memory; HANDLE section_handle = nullptr; if (shared_memory_factory) { - shared_memory = rtc::ScopedToUnique( - shared_memory_factory->CreateSharedMemory(buffer_size)); + shared_memory = shared_memory_factory->CreateSharedMemory(buffer_size); section_handle = shared_memory->handle(); } void* data = nullptr; diff --git a/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc index 022e1ce9e6..5a494f424a 100644 --- a/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc +++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc @@ -75,8 +75,7 @@ ScreenCapturerWinGdi::~ScreenCapturerWinGdi() { void ScreenCapturerWinGdi::SetSharedMemoryFactory( rtc::scoped_ptr shared_memory_factory) { - shared_memory_factory_ = - rtc::ScopedToUnique(std::move(shared_memory_factory)); + shared_memory_factory_ = std::move(shared_memory_factory); } void ScreenCapturerWinGdi::Capture(const DesktopRegion& region) { diff --git a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc index 4bcd4d1918..053a0a398d 100644 --- a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc +++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc @@ -83,8 +83,7 @@ void ScreenCapturerWinMagnifier::Start(Callback* callback) { void ScreenCapturerWinMagnifier::SetSharedMemoryFactory( rtc::scoped_ptr shared_memory_factory) { - shared_memory_factory_ = - rtc::ScopedToUnique(std::move(shared_memory_factory)); + shared_memory_factory_ = std::move(shared_memory_factory); } void ScreenCapturerWinMagnifier::Capture(const DesktopRegion& region) { diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc index f3be09206e..0111571262 100644 --- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc +++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc @@ -207,8 +207,8 @@ TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) { packets[i].sequence_number, packets[i].arrival_time_ms * 1000)); rtc::Buffer raw_packet = feedback->Build(); - feedback = rtc::ScopedToUnique(rtcp::TransportFeedback::ParseFrom( - raw_packet.data(), raw_packet.size())); + feedback = rtcp::TransportFeedback::ParseFrom(raw_packet.data(), + raw_packet.size()); std::vector expected_packets; expected_packets.push_back(packets[i]); @@ -276,8 +276,8 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) { info.arrival_time_ms * 1000)); rtc::Buffer raw_packet = feedback->Build(); - feedback = rtc::ScopedToUnique( - rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size())); + feedback = + rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size()); std::vector received_feedback; @@ -297,8 +297,8 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) { EXPECT_TRUE(feedback->WithReceivedPacket(info.sequence_number, info.arrival_time_ms * 1000)); raw_packet = feedback->Build(); - feedback = rtc::ScopedToUnique( - rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size())); + feedback = + rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size()); EXPECT_TRUE(feedback.get() != nullptr); EXPECT_CALL(*bitrate_estimator_, IncomingPacketFeedbackVector(_)) diff --git a/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/webrtc/modules/utility/source/process_thread_impl_unittest.cc index 9fa9edfa24..16f3b501f8 100644 --- a/webrtc/modules/utility/source/process_thread_impl_unittest.cc +++ b/webrtc/modules/utility/source/process_thread_impl_unittest.cc @@ -297,7 +297,7 @@ TEST(ProcessThreadImpl, PostTask) { std::unique_ptr task_ran(EventWrapper::Create()); std::unique_ptr task(new RaiseEventTask(task_ran.get())); thread.Start(); - thread.PostTask(rtc::UniqueToScoped(std::move(task))); + thread.PostTask(std::move(task)); EXPECT_EQ(kEventSignaled, task_ran->Wait(100)); thread.Stop(); } diff --git a/webrtc/modules/video_capture/test/video_capture_unittest.cc b/webrtc/modules/video_capture/test/video_capture_unittest.cc index 7ab33ffeab..e75ad038c8 100644 --- a/webrtc/modules/video_capture/test/video_capture_unittest.cc +++ b/webrtc/modules/video_capture/test/video_capture_unittest.cc @@ -434,8 +434,7 @@ class VideoCaptureExternalTest : public testing::Test { public: void SetUp() { capture_module_ = VideoCaptureFactory::Create(0, capture_input_interface_); - process_module_ = - rtc::ScopedToUnique(webrtc::ProcessThread::Create("ProcessThread")); + process_module_ = webrtc::ProcessThread::Create("ProcessThread"); process_module_->Start(); process_module_->RegisterModule(capture_module_); diff --git a/webrtc/voice_engine/shared_data.cc b/webrtc/voice_engine/shared_data.cc index 997f51b439..7a67561d1e 100644 --- a/webrtc/voice_engine/shared_data.cc +++ b/webrtc/voice_engine/shared_data.cc @@ -28,7 +28,7 @@ SharedData::SharedData(const Config& config) _engineStatistics(_gInstanceCounter), _audioDevicePtr(NULL), _moduleProcessThreadPtr( - rtc::ScopedToUnique(ProcessThread::Create("VoiceProcessThread"))) { + ProcessThread::Create("VoiceProcessThread")) { Trace::CreateTrace(); if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0) {