Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/ (This is a re-land of https://codereview.webrtc.org/1921233002, which got reverted for breaking Chromium.) BUG=webrtc:5520 Review URL: https://codereview.webrtc.org/1923133002 Cr-Commit-Position: refs/heads/master@{#12522}
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h index 381e35e..eb9d5b8 100644 --- a/webrtc/modules/audio_coding/include/audio_coding_module.h +++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -11,6 +11,7 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ #define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ +#include <memory> #include <string> #include <vector>
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.h b/webrtc/modules/audio_coding/neteq/decoder_database.h index 8dbec22..ec8a5d6 100644 --- a/webrtc/modules/audio_coding/neteq/decoder_database.h +++ b/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -12,6 +12,7 @@ #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_ #include <map> +#include <memory> #include <string> #include "webrtc/base/constructormagic.h"
diff --git a/webrtc/modules/audio_coding/neteq/neteq.cc b/webrtc/modules/audio_coding/neteq/neteq.cc index c2a0cb6..edc29da 100644 --- a/webrtc/modules/audio_coding/neteq/neteq.cc +++ b/webrtc/modules/audio_coding/neteq/neteq.cc
@@ -10,6 +10,7 @@ #include "webrtc/modules/audio_coding/neteq/include/neteq.h" +#include <memory> #include <sstream> #include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc index e9ada1a..66b2b8b 100644 --- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> + #include "webrtc/modules/audio_coding/neteq/include/neteq.h" #include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
diff --git a/webrtc/modules/audio_coding/neteq/tick_timer_unittest.cc b/webrtc/modules/audio_coding/neteq/tick_timer_unittest.cc index 465ce3f..55edcf5 100644 --- a/webrtc/modules/audio_coding/neteq/tick_timer_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/tick_timer_unittest.cc
@@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> + #include "webrtc/modules/audio_coding/neteq/tick_timer.h" #include "testing/gmock/include/gmock/gmock.h"
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc index f68c175..b61b7b3 100644 --- a/webrtc/modules/audio_processing/test/debug_dump_test.cc +++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -10,6 +10,7 @@ #include <stddef.h> // size_t +#include <memory> #include <string> #include <vector>
diff --git a/webrtc/modules/congestion_controller/include/congestion_controller.h b/webrtc/modules/congestion_controller/include/congestion_controller.h index 13614f5..7f18150 100644 --- a/webrtc/modules/congestion_controller/include/congestion_controller.h +++ b/webrtc/modules/congestion_controller/include/congestion_controller.h
@@ -11,6 +11,8 @@ #ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_CONGESTION_CONTROLLER_H_ #define WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_CONGESTION_CONTROLLER_H_ +#include <memory> + #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/include/module.h" @@ -69,9 +71,9 @@ private: Clock* const clock_; - const rtc::scoped_ptr<PacedSender> pacer_; - const rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; - const rtc::scoped_ptr<BitrateController> bitrate_controller_; + const std::unique_ptr<PacedSender> pacer_; + const std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; + const std::unique_ptr<BitrateController> bitrate_controller_; PacketRouter packet_router_; RemoteEstimatorProxy remote_estimator_proxy_; TransportFeedbackAdapter transport_feedback_adapter_;
diff --git a/webrtc/modules/desktop_capture/cropping_window_capturer.cc b/webrtc/modules/desktop_capture/cropping_window_capturer.cc index 0dd564f..cbe7d96 100644 --- a/webrtc/modules/desktop_capture/cropping_window_capturer.cc +++ b/webrtc/modules/desktop_capture/cropping_window_capturer.cc
@@ -32,7 +32,7 @@ } void CroppingWindowCapturer::SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) { + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) { window_capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory)); }
diff --git a/webrtc/modules/desktop_capture/cropping_window_capturer.h b/webrtc/modules/desktop_capture/cropping_window_capturer.h index 177b544..27957ad 100644 --- a/webrtc/modules/desktop_capture/cropping_window_capturer.h +++ b/webrtc/modules/desktop_capture/cropping_window_capturer.h
@@ -32,7 +32,7 @@ // DesktopCapturer implementation. void Start(DesktopCapturer::Callback* callback) override; void SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) override; + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override; void Capture(const DesktopRegion& region) override; void SetExcludedWindow(WindowId window) override;
diff --git a/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc b/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc index 52b111c..4c6e27e 100644 --- a/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc +++ b/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
@@ -138,7 +138,7 @@ } void DesktopAndCursorComposer::SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) { + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) { desktop_capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory)); }
diff --git a/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h b/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h index bcf345e..971943b 100644 --- a/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h +++ b/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
@@ -37,7 +37,7 @@ // DesktopCapturer interface. void Start(DesktopCapturer::Callback* callback) override; void SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) override; + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override; void Capture(const DesktopRegion& region) override; void SetExcludedWindow(WindowId window) override;
diff --git a/webrtc/modules/desktop_capture/desktop_capturer.h b/webrtc/modules/desktop_capture/desktop_capturer.h index 9c2b8c3..103740a 100644 --- a/webrtc/modules/desktop_capture/desktop_capturer.h +++ b/webrtc/modules/desktop_capture/desktop_capturer.h
@@ -13,6 +13,8 @@ #include <stddef.h> +#include <memory> + #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/desktop_capture/desktop_capture_types.h" #include "webrtc/modules/desktop_capture/shared_memory.h" @@ -48,7 +50,7 @@ // where Capture() is called. It will be destroyed on the same thread. Shared // memory is currently supported only by some DesktopCapturer implementations. virtual void SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) {} + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {} // Captures next frame. |region| specifies region of the capture target that // should be fresh in the resulting frame. The frame may also include fresh
diff --git a/webrtc/modules/desktop_capture/screen_capturer_unittest.cc b/webrtc/modules/desktop_capture/screen_capturer_unittest.cc index 6f4963e..bc87ed3 100644 --- a/webrtc/modules/desktop_capture/screen_capturer_unittest.cc +++ b/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
@@ -60,8 +60,8 @@ FakeSharedMemoryFactory() {} ~FakeSharedMemoryFactory() override {} - rtc::scoped_ptr<SharedMemory> CreateSharedMemory(size_t size) override { - return rtc::scoped_ptr<SharedMemory>( + std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) override { + return std::unique_ptr<SharedMemory>( new FakeSharedMemory(new char[size], size)); } @@ -118,7 +118,7 @@ capturer_->Start(&callback_); capturer_->SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory>(new FakeSharedMemoryFactory())); + std::unique_ptr<SharedMemoryFactory>(new FakeSharedMemoryFactory())); capturer_->Capture(DesktopRegion()); ASSERT_TRUE(frame);
diff --git a/webrtc/modules/desktop_capture/shared_memory.h b/webrtc/modules/desktop_capture/shared_memory.h index 45f531e..e1d1e7c 100644 --- a/webrtc/modules/desktop_capture/shared_memory.h +++ b/webrtc/modules/desktop_capture/shared_memory.h
@@ -17,6 +17,8 @@ #include <windows.h> #endif +#include <memory> + #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/typedefs.h" @@ -69,7 +71,7 @@ SharedMemoryFactory() {} virtual ~SharedMemoryFactory() {} - virtual rtc::scoped_ptr<SharedMemory> CreateSharedMemory(size_t size) = 0; + virtual std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) = 0; private: RTC_DISALLOW_COPY_AND_ASSIGN(SharedMemoryFactory);
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 5a494f4..2a5a87e 100644 --- a/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc +++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
@@ -74,7 +74,7 @@ } void ScreenCapturerWinGdi::SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) { + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) { shared_memory_factory_ = std::move(shared_memory_factory); }
diff --git a/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h b/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h index 7fb674d..f43aa0d 100644 --- a/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h +++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
@@ -39,7 +39,7 @@ // Overridden from ScreenCapturer: void Start(Callback* callback) override; void SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) override; + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override; void Capture(const DesktopRegion& region) override; bool GetScreenList(ScreenList* screens) override; bool SelectScreen(ScreenId id) override;
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 053a0a3..e3a5f25 100644 --- a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc +++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
@@ -82,7 +82,7 @@ } void ScreenCapturerWinMagnifier::SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) { + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) { shared_memory_factory_ = std::move(shared_memory_factory); }
diff --git a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h index ad3ddb1..82ef528 100644 --- a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h +++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
@@ -48,7 +48,7 @@ // Overridden from ScreenCapturer: void Start(Callback* callback) override; void SetSharedMemoryFactory( - rtc::scoped_ptr<SharedMemoryFactory> shared_memory_factory) override; + std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override; void Capture(const DesktopRegion& region) override; bool GetScreenList(ScreenList* screens) override; bool SelectScreen(ScreenId id) override;
diff --git a/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h index a76e5e0..d57518a 100644 --- a/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h +++ b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -11,6 +11,8 @@ #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ +#include <memory> + #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/system_wrappers/include/rtp_to_ntp.h" @@ -41,7 +43,7 @@ private: Clock* clock_; - rtc::scoped_ptr<TimestampExtrapolator> ts_extrapolator_; + std::unique_ptr<TimestampExtrapolator> ts_extrapolator_; RtcpList rtcp_list_; int64_t last_timing_log_ms_; RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h index 6ce6434..06ef2fc 100644 --- a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h +++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -12,6 +12,7 @@ #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ #include <map> +#include <memory> #include "webrtc/base/criticalsection.h" #include "webrtc/base/scoped_ptr.h" @@ -181,7 +182,7 @@ rtc::CriticalSection crit_sect_; RtpUtility::PayloadTypeMap payload_type_map_; - rtc::scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_; + std::unique_ptr<RTPPayloadStrategy> rtp_payload_strategy_; int8_t red_payload_type_; int8_t ulpfec_payload_type_; int8_t incoming_payload_type_;
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc index 28e98ba..83bd284 100644 --- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
@@ -12,8 +12,9 @@ #include <assert.h> +#include <memory> + #include "webrtc/base/logging.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h" @@ -87,7 +88,7 @@ // Add to list without RED header, aka a virtual RTP packet // we remove the RED header - rtc::scoped_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet( + std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet( new ForwardErrorCorrection::ReceivedPacket); received_packet->pkt = new ForwardErrorCorrection::Packet; @@ -135,7 +136,7 @@ } ++packet_counter_.num_packets; - rtc::scoped_ptr<ForwardErrorCorrection::ReceivedPacket> + std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> second_received_packet; if (blockLength > 0) { // handle block length, split into 2 packets
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc index ee8f408..cd60d9b 100644 --- a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
@@ -11,10 +11,10 @@ #include <string.h> #include <list> +#include <memory> #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" @@ -92,9 +92,9 @@ uint8_t ulpfec_payload_type); MockRtpData rtp_data_callback_; - rtc::scoped_ptr<ForwardErrorCorrection> fec_; - rtc::scoped_ptr<FecReceiver> receiver_fec_; - rtc::scoped_ptr<FrameGenerator> generator_; + std::unique_ptr<ForwardErrorCorrection> fec_; + std::unique_ptr<FecReceiver> receiver_fec_; + std::unique_ptr<FrameGenerator> generator_; }; void DeletePackets(std::list<Packet*>* packets) { @@ -415,12 +415,12 @@ size_t length, uint8_t ulpfec_payload_type) { webrtc::RTPHeader header; - rtc::scoped_ptr<webrtc::RtpHeaderParser> parser( + std::unique_ptr<webrtc::RtpHeaderParser> parser( webrtc::RtpHeaderParser::Create()); ASSERT_TRUE(parser->Parse(data, length, &header)); webrtc::NullRtpData null_callback; - rtc::scoped_ptr<webrtc::FecReceiver> receiver_fec( + std::unique_ptr<webrtc::FecReceiver> receiver_fec( webrtc::FecReceiver::Create(&null_callback)); receiver_fec->AddReceivedRedPacket(header, data, length, ulpfec_payload_type);
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc index b85d813..623c658 100644 --- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc +++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -15,6 +15,7 @@ #include <algorithm> #include <iterator> +#include <memory> #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" @@ -163,7 +164,7 @@ // -- Generate packet masks -- // Always allocate space for a large mask. - rtc::scoped_ptr<uint8_t[]> packet_mask( + std::unique_ptr<uint8_t[]> packet_mask( new uint8_t[num_fec_packets * kMaskSizeLBitSet]); memset(packet_mask.get(), 0, num_fec_packets * num_mask_bytes); internal::GeneratePacketMasks(num_media_packets, num_fec_packets,
diff --git a/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc b/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc index 6d8b407..e23a3fa 100644 --- a/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc +++ b/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.cc
@@ -9,13 +9,13 @@ */ #include "webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h" +#include <memory> #include <vector> #include "webrtc/base/bitbuffer.h" #include "webrtc/base/bytebuffer.h" #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" -#include "webrtc/base/scoped_ptr.h" namespace webrtc { namespace { @@ -103,7 +103,7 @@ sps_parsed_ = false; // Parse out the SPS RBSP. It should be small, so it's ok that we create a // copy. We'll eventually write this back. - rtc::scoped_ptr<rtc::ByteBufferWriter> sps_rbsp( + std::unique_ptr<rtc::ByteBufferWriter> sps_rbsp( ParseRbsp(sps + kNaluHeaderAndTypeSize, length - kNaluHeaderAndTypeSize)); rtc::BitBuffer sps_parser(reinterpret_cast<const uint8_t*>(sps_rbsp->Data()), sps_rbsp->Length()); @@ -209,7 +209,7 @@ // We're starting a new stream, so reset picture type rewriting values. pps_ = PpsState(); pps_parsed_ = false; - rtc::scoped_ptr<rtc::ByteBufferWriter> buffer( + std::unique_ptr<rtc::ByteBufferWriter> buffer( ParseRbsp(pps + kNaluHeaderAndTypeSize, length - kNaluHeaderAndTypeSize)); rtc::BitBuffer parser(reinterpret_cast<const uint8_t*>(buffer->Data()), buffer->Length()); @@ -317,7 +317,7 @@ RTC_CHECK(sps_parsed_); RTC_CHECK(pps_parsed_); last_slice_qp_delta_parsed_ = false; - rtc::scoped_ptr<rtc::ByteBufferWriter> slice_rbsp(ParseRbsp( + std::unique_ptr<rtc::ByteBufferWriter> slice_rbsp(ParseRbsp( source + kNaluHeaderAndTypeSize, source_length - kNaluHeaderAndTypeSize)); rtc::BitBuffer slice_reader( reinterpret_cast<const uint8_t*>(slice_rbsp->Data()),
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc index e19c31b..a73d4ed 100644 --- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -11,10 +11,10 @@ #include <algorithm> #include <iterator> #include <list> +#include <memory> #include <set> #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" @@ -105,7 +105,7 @@ size_t packet_length = len; uint8_t restored_packet[1500]; RTPHeader header; - rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); + std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); if (!parser->Parse(ptr, len, &header)) { return false; } @@ -279,11 +279,11 @@ void TearDown() override { delete rtp_rtcp_module_; } - rtc::scoped_ptr<ReceiveStatistics> receive_statistics_; + std::unique_ptr<ReceiveStatistics> receive_statistics_; RTPPayloadRegistry rtp_payload_registry_; - rtc::scoped_ptr<RtpReceiver> rtp_receiver_; + std::unique_ptr<RtpReceiver> rtp_receiver_; RtpRtcp* rtp_rtcp_module_; - rtc::scoped_ptr<TestRtpFeedback> rtp_feedback_; + std::unique_ptr<TestRtpFeedback> rtp_feedback_; RtxLoopBackTransport transport_; VerifyingRtxReceiver receiver_; uint8_t payload_data[65000];
diff --git a/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc b/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc index 96c564a..ec5228a 100644 --- a/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
@@ -9,6 +9,7 @@ */ #include <list> +#include <memory> #include <vector> #include "testing/gtest/include/gtest/gtest.h" @@ -188,7 +189,7 @@ TEST_F(ProducerFecTest, BuildRedPacket) { generator_->NewFrame(1); test::RawRtpPacket* packet = generator_->NextPacket(0, 10); - rtc::scoped_ptr<RedPacket> red_packet(producer_->BuildRedPacket( + std::unique_ptr<RedPacket> red_packet(producer_->BuildRedPacket( packet->data, packet->length - kRtpHeaderSize, kRtpHeaderSize, kRedPayloadType)); EXPECT_EQ(packet->length + 1, red_packet->length());
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc index c265c17..898ec02 100644 --- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -8,9 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> + #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" #include "webrtc/system_wrappers/include/clock.h" @@ -36,7 +37,7 @@ protected: SimulatedClock clock_; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics_; + std::unique_ptr<ReceiveStatistics> receive_statistics_; RTPHeader header1_; RTPHeader header2_; };
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc index 183076f..bbfb52c 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> + #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common_types.h" @@ -74,13 +76,13 @@ OverUseDetectorOptions over_use_detector_options_; Clock* system_clock_; ModuleRtpRtcpImpl* dummy_rtp_rtcp_impl_; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics_; + std::unique_ptr<ReceiveStatistics> receive_statistics_; RTCPSender* rtcp_sender_; RTCPReceiver* rtcp_receiver_; TestTransport* test_transport_; test::NullTransport null_transport_; MockRemoteBitrateObserver remote_bitrate_observer_; - rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; + std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; }; void RtcpFormatRembTest::SetUp() {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc index 4ad49561..5cdaa3a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
@@ -651,10 +651,10 @@ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // De-serialize packet. -rtc::scoped_ptr<TransportFeedback> TransportFeedback::ParseFrom( +std::unique_ptr<TransportFeedback> TransportFeedback::ParseFrom( const uint8_t* buffer, size_t length) { - rtc::scoped_ptr<TransportFeedback> packet(new TransportFeedback()); + std::unique_ptr<TransportFeedback> packet(new TransportFeedback()); if (length < kMinSizeBytes) { LOG(LS_WARNING) << "Buffer too small (" << length
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h index ad6fd16..7a74d7f 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
@@ -12,6 +12,7 @@ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TRANSPORT_FEEDBACK_H_ #include <deque> +#include <memory> #include <vector> #include "webrtc/base/constructormagic.h" @@ -58,7 +59,7 @@ static const uint8_t kFeedbackMessageType = 15; // TODO(sprang): IANA reg? static const uint8_t kPayloadType = 205; // RTPFB, see RFC4585. - static rtc::scoped_ptr<TransportFeedback> ParseFrom(const uint8_t* buffer, + static std::unique_ptr<TransportFeedback> ParseFrom(const uint8_t* buffer, size_t length); protected:
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc index 3615065..203d70f 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
@@ -11,6 +11,7 @@ #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" #include <limits> +#include <memory> #include "testing/gtest/include/gtest/gtest.h" @@ -43,7 +44,7 @@ void WithInput(const uint16_t received_seq[], const int64_t received_ts[], uint16_t length) { - rtc::scoped_ptr<int64_t[]> temp_deltas; + std::unique_ptr<int64_t[]> temp_deltas; if (received_ts == nullptr) { temp_deltas.reset(new int64_t[length]); GenerateDeltas(received_seq, length, temp_deltas.get()); @@ -136,7 +137,7 @@ std::vector<int64_t> expected_deltas_; size_t expected_size_; int64_t default_delta_; - rtc::scoped_ptr<TransportFeedback> feedback_; + std::unique_ptr<TransportFeedback> feedback_; rtc::Buffer serialized_; }; @@ -356,7 +357,7 @@ TEST(RtcpPacketTest, TransportFeedback_Limits) { // Sequence number wrap above 0x8000. - rtc::scoped_ptr<TransportFeedback> packet(new TransportFeedback()); + std::unique_ptr<TransportFeedback> packet(new TransportFeedback()); packet->WithBase(0, 0); EXPECT_TRUE(packet->WithReceivedPacket(0x8000, 1000)); @@ -446,7 +447,7 @@ &mod_buffer[2], ByteReader<uint16_t>::ReadBigEndian(&mod_buffer[2]) + ((kPaddingBytes + 3) / 4)); - rtc::scoped_ptr<TransportFeedback> parsed_packet( + std::unique_ptr<TransportFeedback> parsed_packet( TransportFeedback::ParseFrom(mod_buffer, kExpectedSizeWithPadding)); ASSERT_TRUE(parsed_packet.get() != nullptr); EXPECT_EQ(kExpectedSizeWords * 4, packet.size()); // Padding not included. @@ -468,7 +469,7 @@ feedback.WithReceivedPacket(deltas, deltas * 1000 + kLargeTimeDelta); rtc::Buffer serialized_packet = feedback.Build(); - rtc::scoped_ptr<TransportFeedback> deserialized_packet = + std::unique_ptr<TransportFeedback> deserialized_packet = TransportFeedback::ParseFrom(serialized_packet.data(), serialized_packet.size()); EXPECT_TRUE(deserialized_packet.get() != nullptr);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h index da578c7..9a9c73d 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -90,7 +90,7 @@ bool xr_dlrr_item; std::unique_ptr<RTCPVoIPMetric> VoIPMetric; - rtc::scoped_ptr<rtcp::TransportFeedback> transport_feedback_; + std::unique_ptr<rtcp::TransportFeedback> transport_feedback_; private: RTC_DISALLOW_COPY_AND_ASSIGN(RTCPPacketInformation);
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc index b48fed6..924d009 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> + #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" @@ -145,7 +147,7 @@ TestTransport* test_transport_; RTCPHelp::RTCPPacketInformation rtcp_packet_info_; MockRemoteBitrateObserver remote_bitrate_observer_; - rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; + std::unique_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; };
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc index 9a5d471..8817d4d 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> + #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" @@ -256,9 +258,9 @@ SimulatedClock clock_; TestTransport test_transport_; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics_; - rtc::scoped_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_; - rtc::scoped_ptr<RTCPSender> rtcp_sender_; + std::unique_ptr<ReceiveStatistics> receive_statistics_; + std::unique_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_; + std::unique_ptr<RTCPSender> rtcp_sender_; }; TEST_F(RtcpSenderTest, SetRtcpStatus) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h index 4067a40..fedd1dc 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
@@ -13,6 +13,8 @@ #include <stddef.h> // size_t, ptrdiff_t +#include <memory> + #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" @@ -468,7 +470,7 @@ RTCPPacketTypes _packetType; RTCPPacket _packet; - rtc::scoped_ptr<webrtc::rtcp::RtcpPacket> rtcp_packet_; + std::unique_ptr<webrtc::rtcp::RtcpPacket> rtcp_packet_; }; class RTCPPacketIterator {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc index d29e3d4..12c2db5 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> #include <vector> #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" @@ -72,7 +72,7 @@ void TestFua(size_t frame_size, size_t max_payload_size, const std::vector<size_t>& expected_sizes) { - rtc::scoped_ptr<uint8_t[]> frame; + std::unique_ptr<uint8_t[]> frame; frame.reset(new uint8_t[frame_size]); frame[0] = 0x05; // F=0, NRI=0, Type=5. for (size_t i = 0; i < frame_size - kNalHeaderSize; ++i) { @@ -82,11 +82,11 @@ fragmentation.VerifyAndAllocateFragmentationHeader(1); fragmentation.fragmentationOffset[0] = 0; fragmentation.fragmentationLength[0] = frame_size; - rtc::scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create( + std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create( kRtpVideoH264, max_payload_size, NULL, kEmptyFrame)); packetizer->SetPayloadData(frame.get(), frame_size, &fragmentation); - rtc::scoped_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]); + std::unique_ptr<uint8_t[]> packet(new uint8_t[max_payload_size]); size_t length = 0; bool last = false; size_t offset = kNalHeaderSize; @@ -156,7 +156,7 @@ fragmentation.VerifyAndAllocateFragmentationHeader(1); fragmentation.fragmentationOffset[0] = 0; fragmentation.fragmentationLength[0] = sizeof(frame); - rtc::scoped_ptr<RtpPacketizer> packetizer( + std::unique_ptr<RtpPacketizer> packetizer( RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame)); packetizer->SetPayloadData(frame, sizeof(frame), &fragmentation); uint8_t packet[kMaxPayloadSize] = {0}; @@ -185,7 +185,7 @@ frame[fragmentation.fragmentationOffset[0]] = 0x01; frame[fragmentation.fragmentationOffset[1]] = 0x01; - rtc::scoped_ptr<RtpPacketizer> packetizer( + std::unique_ptr<RtpPacketizer> packetizer( RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame)); packetizer->SetPayloadData(frame, kFrameSize, &fragmentation); @@ -222,7 +222,7 @@ fragmentation.fragmentationOffset[2] = 4; fragmentation.fragmentationLength[2] = kNalHeaderSize + kFrameSize - kPayloadOffset; - rtc::scoped_ptr<RtpPacketizer> packetizer( + std::unique_ptr<RtpPacketizer> packetizer( RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame)); packetizer->SetPayloadData(frame, kFrameSize, &fragmentation); @@ -257,7 +257,7 @@ fragmentation.fragmentationOffset[2] = 4; fragmentation.fragmentationLength[2] = kNalHeaderSize + kFrameSize - kPayloadOffset; - rtc::scoped_ptr<RtpPacketizer> packetizer( + std::unique_ptr<RtpPacketizer> packetizer( RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame)); packetizer->SetPayloadData(frame, kFrameSize, &fragmentation); @@ -305,7 +305,7 @@ frame[nalu_offset + j] = i + j; } } - rtc::scoped_ptr<RtpPacketizer> packetizer( + std::unique_ptr<RtpPacketizer> packetizer( RtpPacketizer::Create(kRtpVideoH264, kMaxPayloadSize, NULL, kEmptyFrame)); packetizer->SetPayloadData(frame, kFrameSize, &fragmentation); @@ -394,7 +394,7 @@ ::testing::ElementsAreArray(data, length)); } - rtc::scoped_ptr<RtpDepacketizer> depacketizer_; + std::unique_ptr<RtpDepacketizer> depacketizer_; }; TEST_F(RtpDepacketizerH264Test, TestSingleNalu) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc index b956447..079d964 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> + #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h" @@ -417,7 +419,7 @@ ::testing::ElementsAreArray(data, length)); } - rtc::scoped_ptr<RtpDepacketizer> depacketizer_; + std::unique_ptr<RtpDepacketizer> depacketizer_; }; TEST_F(RtpDepacketizerVp8Test, BasicHeader) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc index 5bbafe4..f9514ad 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
@@ -8,6 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> #include <vector> #include "testing/gmock/include/gmock/gmock.h" @@ -76,7 +77,7 @@ const RTPVideoHeaderVP9& expected, size_t expected_hdr_length, size_t expected_length) { - rtc::scoped_ptr<RtpDepacketizer> depacketizer(new RtpDepacketizerVp9()); + std::unique_ptr<RtpDepacketizer> depacketizer(new RtpDepacketizerVp9()); RtpDepacketizer::ParsedPayload parsed; ASSERT_TRUE(depacketizer->Parse(&parsed, packet, expected_length)); EXPECT_EQ(kRtpVideoVp9, parsed.type.Video.codec); @@ -127,12 +128,12 @@ expected_.InitRTPVideoHeaderVP9(); } - rtc::scoped_ptr<uint8_t[]> packet_; - rtc::scoped_ptr<uint8_t[]> payload_; + std::unique_ptr<uint8_t[]> packet_; + std::unique_ptr<uint8_t[]> payload_; size_t payload_size_; size_t payload_pos_; RTPVideoHeaderVP9 expected_; - rtc::scoped_ptr<RtpPacketizerVp9> packetizer_; + std::unique_ptr<RtpPacketizerVp9> packetizer_; void Init(size_t payload_size, size_t packet_size) { payload_.reset(new uint8_t[payload_size]); @@ -469,7 +470,7 @@ } RTPVideoHeaderVP9 expected_; - rtc::scoped_ptr<RtpDepacketizer> depacketizer_; + std::unique_ptr<RtpDepacketizer> depacketizer_; }; TEST_F(RtpDepacketizerVp9Test, ParseBasicHeader) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc index cbded68..5bbe97a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -8,11 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include <memory> + #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h" @@ -58,7 +59,7 @@ return returned_payload_on_heap; } - rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_; + std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_; testing::NiceMock<MockRTPPayloadStrategy>* mock_payload_strategy_; }; @@ -296,9 +297,9 @@ uint16_t original_sequence_number = 1234; uint32_t original_ssrc = 500; - rtc::scoped_ptr<const uint8_t[]> packet(GenerateRtxPacket( + std::unique_ptr<const uint8_t[]> packet(GenerateRtxPacket( header_length, payload_length, original_sequence_number)); - rtc::scoped_ptr<uint8_t[]> restored_packet( + std::unique_ptr<uint8_t[]> restored_packet( new uint8_t[header_length + payload_length]); size_t length = original_length; bool success = rtp_payload_registry->RestoreOriginalPacket( @@ -312,7 +313,7 @@ EXPECT_EQ(original_length - kRtxHeaderSize, length) << "The restored packet should be exactly kRtxHeaderSize smaller."; - rtc::scoped_ptr<RtpHeaderParser> header_parser(RtpHeaderParser::Create()); + std::unique_ptr<RtpHeaderParser> header_parser(RtpHeaderParser::Create()); RTPHeader restored_header; ASSERT_TRUE( header_parser->Parse(restored_packet.get(), length, &restored_header));
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h index 7c6287c..dca1978 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -11,6 +11,8 @@ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ +#include <memory> + #include "webrtc/base/criticalsection.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" @@ -75,7 +77,7 @@ Clock* clock_; RTPPayloadRegistry* rtp_payload_registry_; - rtc::scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_; + std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_; RtpFeedback* cb_rtp_feedback_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc index f53f55a..9d76c1a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -13,6 +13,8 @@ #include <assert.h> #include <string.h> +#include <memory> + #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/trace_event.h" @@ -74,7 +76,7 @@ } // We are not allowed to hold a critical section when calling below functions. - rtc::scoped_ptr<RtpDepacketizer> depacketizer( + std::unique_ptr<RtpDepacketizer> depacketizer( RtpDepacketizer::Create(rtp_header->type.Video.codec)); if (depacketizer.get() == NULL) { LOG(LS_ERROR) << "Failed to create depacketizer.";
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc index 708b9af..7e0ac31 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -9,6 +9,7 @@ */ #include <map> +#include <memory> #include <set> #include "testing/gmock/include/gmock/gmock.h" @@ -68,7 +69,7 @@ size_t len, const PacketOptions& options) override { RTPHeader header; - rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); + std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); EXPECT_TRUE(parser->Parse(static_cast<const uint8_t*>(data), len, &header)); ++rtp_packets_sent_; last_rtp_header_ = header; @@ -115,10 +116,10 @@ RtcpPacketTypeCounter packets_sent_; RtcpPacketTypeCounter packets_received_; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics_; + std::unique_ptr<ReceiveStatistics> receive_statistics_; SendTransport transport_; RtcpRttStatsTestImpl rtt_stats_; - rtc::scoped_ptr<ModuleRtpRtcpImpl> impl_; + std::unique_ptr<ModuleRtpRtcpImpl> impl_; uint32_t remote_ssrc_; void SetRemoteSsrc(uint32_t ssrc) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h index 99465c6..f9d5df6 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -13,6 +13,7 @@ #include <list> #include <map> +#include <memory> #include <utility> #include <vector> @@ -423,8 +424,8 @@ Bitrate total_bitrate_sent_; const bool audio_configured_; - const rtc::scoped_ptr<RTPSenderAudio> audio_; - const rtc::scoped_ptr<RTPSenderVideo> video_; + const std::unique_ptr<RTPSenderAudio> audio_; + const std::unique_ptr<RTPSenderVideo> video_; RtpPacketSender* const paced_sender_; TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc index f9dc8f1..f350eff 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -9,12 +9,12 @@ */ #include <list> +#include <memory> #include <vector> #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/buffer.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/call/mock/mock_rtc_event_log.h" #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" @@ -148,7 +148,7 @@ MockRtcEventLog mock_rtc_event_log_; MockRtpPacketSender mock_paced_sender_; MockTransportSequenceNumberAllocator seq_num_allocator_; - rtc::scoped_ptr<RTPSender> rtp_sender_; + std::unique_ptr<RTPSender> rtp_sender_; int payload_; LoopbackTransportTest transport_; const bool kMarkerBit; @@ -202,7 +202,7 @@ rtp_sender_video_.reset( new RTPSenderVideo(&fake_clock_, rtp_sender_.get())); } - rtc::scoped_ptr<RTPSenderVideo> rtp_sender_video_; + std::unique_ptr<RTPSenderVideo> rtp_sender_video_; void VerifyCVOPacket(uint8_t* data, size_t len, @@ -849,7 +849,7 @@ rtp_header_len += 4; // 4 extra bytes common to all extension headers. // Create and set up parser. - rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser( + std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser( webrtc::RtpHeaderParser::Create()); ASSERT_TRUE(rtp_parser.get() != nullptr); rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, @@ -968,7 +968,7 @@ rtp_sender_->SetRtxSsrc(1234); // Create and set up parser. - rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser( + std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser( webrtc::RtpHeaderParser::Create()); ASSERT_TRUE(rtp_parser.get() != nullptr); rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, @@ -1402,7 +1402,7 @@ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type, capture_time_ms + 2000, 0, nullptr, 0, nullptr)); - rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser( + std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser( webrtc::RtpHeaderParser::Create()); ASSERT_TRUE(rtp_parser.get() != nullptr); webrtc::RTPHeader rtp_header;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc index d617f10..0bf95b7 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -13,6 +13,7 @@ #include <stdlib.h> #include <string.h> +#include <memory> #include <vector> #include "webrtc/base/checks.h" @@ -111,7 +112,7 @@ int64_t capture_time_ms, StorageType media_packet_storage, bool protect) { - rtc::scoped_ptr<RedPacket> red_packet; + std::unique_ptr<RedPacket> red_packet; std::vector<RedPacket*> fec_packets; StorageType fec_storage = kDontRetransmit; uint16_t next_fec_sequence_number = 0; @@ -224,7 +225,7 @@ return -1; } - rtc::scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create( + std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create( videoType, _rtpSender.MaxDataPayloadLength(), video_header ? &(video_header->codecHeader) : nullptr, frameType));
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc index 67e8a65..ea2d98b 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -11,6 +11,7 @@ #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h" #include <algorithm> +#include <memory> #include <vector> #include "webrtc/test/null_transport.h" @@ -41,7 +42,7 @@ } } RTPHeader header; - rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); + std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) { return false; } @@ -100,9 +101,9 @@ &fake_clock_, NULL, NULL, rtp_payload_registry_.get())); } - rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_; - rtc::scoped_ptr<RtpReceiver> rtp_receiver_; - rtc::scoped_ptr<RtpRtcp> module_; + std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_; + std::unique_ptr<RtpReceiver> rtp_receiver_; + std::unique_ptr<RtpRtcp> module_; uint32_t test_ssrc_; uint32_t test_timestamp_; uint16_t test_sequence_number_;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc index 9b44c4f..8069b09 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -9,6 +9,7 @@ */ #include <algorithm> +#include <memory> #include <vector> #include "testing/gtest/include/gtest/gtest.h" @@ -135,12 +136,12 @@ RtpRtcp* module1; RtpRtcp* module2; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics2_; - rtc::scoped_ptr<RtpReceiver> rtp_receiver1_; - rtc::scoped_ptr<RtpReceiver> rtp_receiver2_; - rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry1_; - rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry2_; + std::unique_ptr<ReceiveStatistics> receive_statistics1_; + std::unique_ptr<ReceiveStatistics> receive_statistics2_; + std::unique_ptr<RtpReceiver> rtp_receiver1_; + std::unique_ptr<RtpReceiver> rtp_receiver2_; + std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_; + std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_; VerifyingAudioReceiver* data_receiver1; VerifyingAudioReceiver* data_receiver2; LoopBackTransport* transport1;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc index d4b3641..c1359df 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -9,6 +9,7 @@ */ #include <algorithm> +#include <memory> #include <vector> #include "testing/gmock/include/gmock/gmock.h" @@ -175,14 +176,14 @@ delete receiver; } - rtc::scoped_ptr<TestRtpFeedback> rtp_feedback1_; - rtc::scoped_ptr<TestRtpFeedback> rtp_feedback2_; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics1_; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics2_; - rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry1_; - rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry2_; - rtc::scoped_ptr<RtpReceiver> rtp_receiver1_; - rtc::scoped_ptr<RtpReceiver> rtp_receiver2_; + std::unique_ptr<TestRtpFeedback> rtp_feedback1_; + std::unique_ptr<TestRtpFeedback> rtp_feedback2_; + std::unique_ptr<ReceiveStatistics> receive_statistics1_; + std::unique_ptr<ReceiveStatistics> receive_statistics2_; + std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_; + std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_; + std::unique_ptr<RtpReceiver> rtp_receiver1_; + std::unique_ptr<RtpReceiver> rtp_receiver2_; RtpRtcp* module1; RtpRtcp* module2; TestRtpReceiver* receiver;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc index 16ea540..74daba9 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -11,6 +11,7 @@ #include <stdlib.h> #include <algorithm> +#include <memory> #include <vector> #include "testing/gtest/include/gtest/gtest.h" @@ -127,9 +128,9 @@ } int test_id_; - rtc::scoped_ptr<ReceiveStatistics> receive_statistics_; + std::unique_ptr<ReceiveStatistics> receive_statistics_; RTPPayloadRegistry rtp_payload_registry_; - rtc::scoped_ptr<RtpReceiver> rtp_receiver_; + std::unique_ptr<RtpReceiver> rtp_receiver_; RtpRtcp* video_module_; LoopBackTransport* transport_; TestRtpReceiver* receiver_; @@ -170,7 +171,7 @@ kPadSize); ++seq_num; RTPHeader header; - rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); + std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create()); EXPECT_TRUE(parser->Parse(padding_packet, packet_size, &header)); PayloadUnion payload_specific; EXPECT_TRUE(rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
diff --git a/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc index 466214c..b7c4ef5 100644 --- a/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc +++ b/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
@@ -45,8 +45,9 @@ #include <math.h> +#include <memory> + #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h" #include "webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h" #include "webrtc/test/testsupport/fileutils.h" @@ -191,7 +192,7 @@ int RecoveredMediaPackets(int num_media_packets, int num_fec_packets, uint8_t* state) { - rtc::scoped_ptr<uint8_t[]> state_tmp( + std::unique_ptr<uint8_t[]> state_tmp( new uint8_t[num_media_packets + num_fec_packets]); memcpy(state_tmp.get(), state, num_media_packets + num_fec_packets); int num_recovered_packets = 0; @@ -385,7 +386,7 @@ // (which containes the code size parameters/protection length). void ComputeMetricsForCode(CodeType code_type, int code_index) { - rtc::scoped_ptr<double[]> prob_weight(new double[kNumLossModels]); + std::unique_ptr<double[]> prob_weight(new double[kNumLossModels]); memset(prob_weight.get() , 0, sizeof(double) * kNumLossModels); MetricsFecCode metrics_code; SetMetricsZero(&metrics_code); @@ -393,7 +394,7 @@ int num_media_packets = code_params_[code_index].num_media_packets; int num_fec_packets = code_params_[code_index].num_fec_packets; int tot_num_packets = num_media_packets + num_fec_packets; - rtc::scoped_ptr<uint8_t[]> state(new uint8_t[tot_num_packets]); + std::unique_ptr<uint8_t[]> state(new uint8_t[tot_num_packets]); memset(state.get() , 0, tot_num_packets); int num_loss_configurations = static_cast<int>(pow(2.0f, tot_num_packets));
diff --git a/webrtc/modules/utility/include/jvm_android.h b/webrtc/modules/utility/include/jvm_android.h index f527dff..305e7cf 100644 --- a/webrtc/modules/utility/include/jvm_android.h +++ b/webrtc/modules/utility/include/jvm_android.h
@@ -12,6 +12,8 @@ #define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_ #include <jni.h> + +#include <memory> #include <string> #include "webrtc/base/scoped_ptr.h" @@ -76,7 +78,7 @@ NativeRegistration(JNIEnv* jni, jclass clazz); ~NativeRegistration(); - rtc::scoped_ptr<GlobalRef> NewObject( + std::unique_ptr<GlobalRef> NewObject( const char* name, const char* signature, ...); private: @@ -96,7 +98,7 @@ // Note that the class name must be one of the names in the static // |loaded_classes| array defined in jvm_android.cc. // This method must be called on the construction thread. - rtc::scoped_ptr<NativeRegistration> RegisterNatives( + std::unique_ptr<NativeRegistration> RegisterNatives( const char* name, const JNINativeMethod *methods, int num_methods); // Converts from Java string to std::string. @@ -120,9 +122,9 @@ // webrtc::JVM::Initialize(jvm, context); // // // Header (.h) file of example class called User. -// rtc::scoped_ptr<JNIEnvironment> env; -// rtc::scoped_ptr<NativeRegistration> reg; -// rtc::scoped_ptr<GlobalRef> obj; +// std::unique_ptr<JNIEnvironment> env; +// std::unique_ptr<NativeRegistration> reg; +// std::unique_ptr<GlobalRef> obj; // // // Construction (in .cc file) of User class. // User::User() { @@ -156,7 +158,7 @@ // Creates a JNIEnvironment object. // This method returns a NULL pointer if AttachCurrentThread() has not been // called successfully. Use the AttachCurrentThreadIfNeeded class if needed. - rtc::scoped_ptr<JNIEnvironment> environment(); + std::unique_ptr<JNIEnvironment> environment(); // Returns a JavaClass object given class |name|. // Note that the class name must be one of the names in the static
diff --git a/webrtc/modules/utility/include/mock/mock_process_thread.h b/webrtc/modules/utility/include/mock/mock_process_thread.h index 9560e40..3d39307 100644 --- a/webrtc/modules/utility/include/mock/mock_process_thread.h +++ b/webrtc/modules/utility/include/mock/mock_process_thread.h
@@ -31,7 +31,7 @@ // MOCK_METHOD1 gets confused with mocking this method, so we work around it // by overriding the method from the interface and forwarding the call to a // mocked, simpler method. - void PostTask(rtc::scoped_ptr<ProcessTask> task) override { + void PostTask(std::unique_ptr<ProcessTask> task) override { PostTask(task.get()); } };
diff --git a/webrtc/modules/utility/include/process_thread.h b/webrtc/modules/utility/include/process_thread.h index 285a5ea..4d77452 100644 --- a/webrtc/modules/utility/include/process_thread.h +++ b/webrtc/modules/utility/include/process_thread.h
@@ -11,6 +11,8 @@ #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_ #define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_ +#include <memory> + #include "webrtc/typedefs.h" #include "webrtc/base/scoped_ptr.h" @@ -29,7 +31,7 @@ public: virtual ~ProcessThread(); - static rtc::scoped_ptr<ProcessThread> Create(const char* thread_name); + static std::unique_ptr<ProcessThread> Create(const char* thread_name); // Starts the worker thread. Must be called from the construction thread. virtual void Start() = 0; @@ -50,7 +52,7 @@ // construction thread of the ProcessThread instance, if the task did not // get a chance to run (e.g. posting the task while shutting down or when // the thread never runs). - virtual void PostTask(rtc::scoped_ptr<ProcessTask> task) = 0; + virtual void PostTask(std::unique_ptr<ProcessTask> task) = 0; // Adds a module that will start to receive callbacks on the worker thread. // Can be called from any thread.
diff --git a/webrtc/modules/utility/source/jvm_android.cc b/webrtc/modules/utility/source/jvm_android.cc index eb37fda..d53d1b5 100644 --- a/webrtc/modules/utility/source/jvm_android.cc +++ b/webrtc/modules/utility/source/jvm_android.cc
@@ -10,6 +10,8 @@ #include <android/log.h> +#include <memory> + #include "webrtc/modules/utility/include/jvm_android.h" #include "webrtc/base/checks.h" @@ -139,7 +141,7 @@ CHECK_EXCEPTION(jni_) << "Error during UnregisterNatives"; } -rtc::scoped_ptr<GlobalRef> NativeRegistration::NewObject( +std::unique_ptr<GlobalRef> NativeRegistration::NewObject( const char* name, const char* signature, ...) { ALOGD("NativeRegistration::NewObject%s", GetThreadInfo().c_str()); va_list args; @@ -149,7 +151,7 @@ args); CHECK_EXCEPTION(jni_) << "Error during NewObjectV"; va_end(args); - return rtc::scoped_ptr<GlobalRef>(new GlobalRef(jni_, obj)); + return std::unique_ptr<GlobalRef>(new GlobalRef(jni_, obj)); } // JavaClass implementation. @@ -181,14 +183,14 @@ RTC_DCHECK(thread_checker_.CalledOnValidThread()); } -rtc::scoped_ptr<NativeRegistration> JNIEnvironment::RegisterNatives( +std::unique_ptr<NativeRegistration> JNIEnvironment::RegisterNatives( const char* name, const JNINativeMethod *methods, int num_methods) { ALOGD("JNIEnvironment::RegisterNatives(%s)", name); RTC_DCHECK(thread_checker_.CalledOnValidThread()); jclass clazz = LookUpClass(name); jni_->RegisterNatives(clazz, methods, num_methods); CHECK_EXCEPTION(jni_) << "Error during RegisterNatives"; - return rtc::scoped_ptr<NativeRegistration>( + return std::unique_ptr<NativeRegistration>( new NativeRegistration(jni_, clazz)); } @@ -240,7 +242,7 @@ DeleteGlobalRef(jni(), context_); } -rtc::scoped_ptr<JNIEnvironment> JVM::environment() { +std::unique_ptr<JNIEnvironment> JVM::environment() { ALOGD("JVM::environment%s", GetThreadInfo().c_str()); // The JNIEnv is used for thread-local storage. For this reason, we cannot // share a JNIEnv between threads. If a piece of code has no other way to get @@ -250,9 +252,9 @@ JNIEnv* jni = GetEnv(jvm_); if (!jni) { ALOGE("AttachCurrentThread() has not been called on this thread."); - return rtc::scoped_ptr<JNIEnvironment>(); + return std::unique_ptr<JNIEnvironment>(); } - return rtc::scoped_ptr<JNIEnvironment>(new JNIEnvironment(jni)); + return std::unique_ptr<JNIEnvironment>(new JNIEnvironment(jni)); } JavaClass JVM::GetClass(const char* name) {
diff --git a/webrtc/modules/utility/source/process_thread_impl.cc b/webrtc/modules/utility/source/process_thread_impl.cc index 8cdf016..68c7ab6 100644 --- a/webrtc/modules/utility/source/process_thread_impl.cc +++ b/webrtc/modules/utility/source/process_thread_impl.cc
@@ -36,9 +36,9 @@ ProcessThread::~ProcessThread() {} // static -rtc::scoped_ptr<ProcessThread> ProcessThread::Create( +std::unique_ptr<ProcessThread> ProcessThread::Create( const char* thread_name) { - return rtc::scoped_ptr<ProcessThread>(new ProcessThreadImpl(thread_name)); + return std::unique_ptr<ProcessThread>(new ProcessThreadImpl(thread_name)); } ProcessThreadImpl::ProcessThreadImpl(const char* thread_name) @@ -119,7 +119,7 @@ wake_up_->Set(); } -void ProcessThreadImpl::PostTask(rtc::scoped_ptr<ProcessTask> task) { +void ProcessThreadImpl::PostTask(std::unique_ptr<ProcessTask> task) { // Allowed to be called on any thread. { rtc::CritScope lock(&lock_);
diff --git a/webrtc/modules/utility/source/process_thread_impl.h b/webrtc/modules/utility/source/process_thread_impl.h index 2855ed9..330aec9 100644 --- a/webrtc/modules/utility/source/process_thread_impl.h +++ b/webrtc/modules/utility/source/process_thread_impl.h
@@ -33,7 +33,7 @@ void Stop() override; void WakeUp(Module* module) override; - void PostTask(rtc::scoped_ptr<ProcessTask> task) override; + void PostTask(std::unique_ptr<ProcessTask> task) override; void RegisterModule(Module* module) override; void DeRegisterModule(Module* module) override;
diff --git a/webrtc/modules/video_coding/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/jitter_buffer_unittest.cc index df70ea9..eb7d78b 100644 --- a/webrtc/modules/video_coding/jitter_buffer_unittest.cc +++ b/webrtc/modules/video_coding/jitter_buffer_unittest.cc
@@ -195,7 +195,7 @@ MOCK_METHOD1(WakeUp, void(Module* module)); MOCK_METHOD1(RegisterModule, void(Module* module)); MOCK_METHOD1(DeRegisterModule, void(Module* module)); - void PostTask(rtc::scoped_ptr<ProcessTask> task) {} + void PostTask(std::unique_ptr<ProcessTask> task) {} }; class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
diff --git a/webrtc/modules/video_coding/packet_buffer.h b/webrtc/modules/video_coding/packet_buffer.h index caa81f6..75049b3 100644 --- a/webrtc/modules/video_coding/packet_buffer.h +++ b/webrtc/modules/video_coding/packet_buffer.h
@@ -14,6 +14,7 @@ #include <array> #include <vector> #include <map> +#include <memory> #include <set> #include <queue>
diff --git a/webrtc/modules/video_coding/packet_buffer_unittest.cc b/webrtc/modules/video_coding/packet_buffer_unittest.cc index 28c62c2..eb0a03b 100644 --- a/webrtc/modules/video_coding/packet_buffer_unittest.cc +++ b/webrtc/modules/video_coding/packet_buffer_unittest.cc
@@ -10,6 +10,7 @@ #include <cstring> #include <limits> +#include <memory> #include "webrtc/modules/video_coding/frame_object.h" #include "webrtc/modules/video_coding/packet_buffer.h"
diff --git a/webrtc/modules/video_processing/util/noise_estimation.h b/webrtc/modules/video_processing/util/noise_estimation.h index 5299b31..fa5b522 100644 --- a/webrtc/modules/video_processing/util/noise_estimation.h +++ b/webrtc/modules/video_processing/util/noise_estimation.h
@@ -11,6 +11,8 @@ #ifndef WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_NOISE_ESTIMATION_H_ #define WEBRTC_MODULES_VIDEO_PROCESSING_UTIL_NOISE_ESTIMATION_H_ +#include <memory> + #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_processing/include/video_processing_defines.h"