Remove calls to ScopedToUnique and UniqueToScoped
They're just no-ops now, and will soon go away.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1914153002
Cr-Commit-Position: refs/heads/master@{#12510}
diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
index 28d2f63..1cf7924 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/api/webrtcsession.cc
@@ -1241,7 +1241,7 @@
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 61eb80a..67abc56 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -191,9 +191,8 @@
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 0edf586..5ab1db1 100644
--- a/webrtc/common_audio/lapped_transform.cc
+++ b/webrtc/common_audio/lapped_transform.cc
@@ -72,8 +72,7 @@
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 9174a5b..01e5d5f 100644
--- a/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/webrtc/modules/audio_device/android/audio_manager.cc
@@ -66,7 +66,7 @@
// 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 @@
{"nativeCacheAudioParameters",
"(IIZZZZIIJ)V",
reinterpret_cast<void*>(&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(
"<init>", "(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 5ff5997..8ce1386 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 @@
// 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 @@
&webrtc::AudioRecordJni::CacheDirectBufferAddress)},
{"nativeDataIsRecorded", "(IJ)V",
reinterpret_cast<void*>(&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(
"<init>", "(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 67837f5..fc77e32 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 @@
// 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 @@
&webrtc::AudioTrackJni::CacheDirectBufferAddress)},
{"nativeGetPlayoutData", "(IJ)V",
reinterpret_cast<void*>(&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(
"<init>", "(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 c6cecc9..455c12f 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 f37d89c..78642c3 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 @@
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 bb7686c..cc58965 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 @@
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 @@
// ==================================================
// 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 6bc7b2e..3278ed4 100644
--- a/webrtc/modules/desktop_capture/desktop_frame.cc
+++ b/webrtc/modules/desktop_capture/desktop_frame.cc
@@ -84,8 +84,7 @@
size_t buffer_size =
size.width() * size.height() * DesktopFrame::kBytesPerPixel;
std::unique_ptr<SharedMemory> 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 e91e37e..624b729 100644
--- a/webrtc/modules/desktop_capture/desktop_frame_win.cc
+++ b/webrtc/modules/desktop_capture/desktop_frame_win.cc
@@ -49,8 +49,7 @@
std::unique_ptr<SharedMemory> 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 022e1ce..5a494f4 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 @@
void ScreenCapturerWinGdi::SetSharedMemoryFactory(
rtc::scoped_ptr<SharedMemoryFactory> 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 4bcd4d1..053a0a3 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::SetSharedMemoryFactory(
rtc::scoped_ptr<SharedMemoryFactory> 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 f3be092..0111571 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 @@
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<PacketInfo> expected_packets;
expected_packets.push_back(packets[i]);
@@ -276,8 +276,8 @@
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<PacketInfo> received_feedback;
@@ -297,8 +297,8 @@
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 9fa9edf..16f3b50 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 @@
std::unique_ptr<EventWrapper> task_ran(EventWrapper::Create());
std::unique_ptr<RaiseEventTask> 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 7ab33ff..e75ad03 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 @@
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 997f51b..7a67561 100644
--- a/webrtc/voice_engine/shared_data.cc
+++ b/webrtc/voice_engine/shared_data.cc
@@ -28,7 +28,7 @@
_engineStatistics(_gInstanceCounter),
_audioDevicePtr(NULL),
_moduleProcessThreadPtr(
- rtc::ScopedToUnique(ProcessThread::Create("VoiceProcessThread"))) {
+ ProcessThread::Create("VoiceProcessThread")) {
Trace::CreateTrace();
if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
{