From 27f982bbcb88cbe5d47ff7e67a2a70063974554d Mon Sep 17 00:00:00 2001 From: kwiberg Date: Tue, 1 Mar 2016 11:52:33 -0800 Subject: [PATCH] Replace scoped_ptr with unique_ptr in webrtc/video/ BUG=webrtc:5520 Review URL: https://codereview.webrtc.org/1751903002 Cr-Commit-Position: refs/heads/master@{#11833} --- webrtc/video/call_stats.h | 4 +- webrtc/video/call_stats_unittest.cc | 5 ++- webrtc/video/end_to_end_tests.cc | 44 +++++++++---------- webrtc/video/overuse_frame_detector.cc | 4 +- webrtc/video/overuse_frame_detector.h | 4 +- .../video/overuse_frame_detector_unittest.cc | 9 ++-- webrtc/video/payload_router.h | 1 - webrtc/video/payload_router_unittest.cc | 5 ++- webrtc/video/replay.cc | 12 ++--- webrtc/video/send_statistics_proxy.h | 4 +- .../video/send_statistics_proxy_unittest.cc | 3 +- webrtc/video/video_capture_input_unittest.cc | 8 ++-- webrtc/video/video_quality_test.cc | 7 ++- webrtc/video/video_quality_test.h | 9 ++-- webrtc/video/video_receive_stream.h | 4 +- webrtc/video/video_send_stream_tests.cc | 24 +++++----- webrtc/video/vie_channel.h | 8 ++-- webrtc/video/vie_encoder.h | 10 ++--- webrtc/video/vie_receiver.h | 14 +++--- webrtc/video/vie_remb.h | 1 - webrtc/video/vie_remb_unittest.cc | 6 +-- webrtc/video/vie_sync_module.h | 5 ++- 22 files changed, 97 insertions(+), 94 deletions(-) diff --git a/webrtc/video/call_stats.h b/webrtc/video/call_stats.h index bb3670c39d..9a5967e648 100644 --- a/webrtc/video/call_stats.h +++ b/webrtc/video/call_stats.h @@ -12,10 +12,10 @@ #define WEBRTC_VIDEO_CALL_STATS_H_ #include +#include #include "webrtc/base/constructormagic.h" #include "webrtc/base/criticalsection.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/include/module.h" #include "webrtc/system_wrappers/include/clock.h" @@ -64,7 +64,7 @@ class CallStats : public Module { // Protecting all members. rtc::CriticalSection crit_; // Observer receiving statistics updates. - rtc::scoped_ptr rtcp_rtt_stats_; + std::unique_ptr rtcp_rtt_stats_; // The last time 'Process' resulted in statistic update. int64_t last_process_time_; // The last RTT in the statistics update (zero if there is no valid estimate). diff --git a/webrtc/video/call_stats_unittest.cc b/webrtc/video/call_stats_unittest.cc index 2421cc7148..6e2e1bca78 100644 --- a/webrtc/video/call_stats_unittest.cc +++ b/webrtc/video/call_stats_unittest.cc @@ -8,10 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include + #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_rtcp_defines.h" #include "webrtc/system_wrappers/include/metrics.h" #include "webrtc/system_wrappers/include/tick_util.h" @@ -39,7 +40,7 @@ class CallStatsTest : public ::testing::Test { protected: virtual void SetUp() { call_stats_.reset(new CallStats(&fake_clock_)); } SimulatedClock fake_clock_; - rtc::scoped_ptr call_stats_; + std::unique_ptr call_stats_; }; TEST_F(CallStatsTest, AddAndTriggerCallback) { diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc index 2a88e28836..9e8ead9927 100644 --- a/webrtc/video/end_to_end_tests.cc +++ b/webrtc/video/end_to_end_tests.cc @@ -10,6 +10,7 @@ #include #include #include +#include #include #include @@ -17,7 +18,6 @@ #include "webrtc/base/checks.h" #include "webrtc/base/event.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/timeutils.h" #include "webrtc/call.h" #include "webrtc/call/transport_adapter.h" @@ -173,7 +173,7 @@ TEST_F(EndToEndTest, RendersSingleDelayedFrame) { // Create frames that are smaller than the send width/height, this is done to // check that the callbacks are done after processing video. - rtc::scoped_ptr frame_generator( + std::unique_ptr frame_generator( test::FrameGenerator::CreateChromaGenerator(kWidth, kHeight)); video_send_stream_->Input()->IncomingCapturedFrame( *frame_generator->NextFrame()); @@ -220,7 +220,7 @@ TEST_F(EndToEndTest, TransmitsFirstFrame) { CreateVideoStreams(); Start(); - rtc::scoped_ptr frame_generator( + std::unique_ptr frame_generator( test::FrameGenerator::CreateChromaGenerator( video_encoder_config_.streams[0].width, video_encoder_config_.streams[0].height)); @@ -282,8 +282,8 @@ TEST_F(EndToEndTest, SendsAndReceivesVP9) { bool IsTextureSupported() const override { return false; } private: - rtc::scoped_ptr encoder_; - rtc::scoped_ptr decoder_; + std::unique_ptr encoder_; + std::unique_ptr decoder_; int frame_counter_; } test; @@ -338,8 +338,8 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) { bool IsTextureSupported() const override { return false; } private: - rtc::scoped_ptr encoder_; - rtc::scoped_ptr decoder_; + std::unique_ptr encoder_; + std::unique_ptr decoder_; int frame_counter_; } test; @@ -816,7 +816,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool enable_rtx, bool enable_red) { const int payload_type_; const uint32_t retransmission_ssrc_; const int retransmission_payload_type_; - rtc::scoped_ptr encoder_; + std::unique_ptr encoder_; const std::string payload_name_; int marker_bits_observed_; uint32_t retransmitted_timestamp_ GUARDED_BY(&crit_); @@ -908,7 +908,7 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) { receiver_transport.SetReceiver(sender_call_->Receiver()); CreateSendConfig(1, 0, &sender_transport); - rtc::scoped_ptr encoder( + std::unique_ptr encoder( VideoEncoder::Create(VideoEncoder::kVp8)); video_send_config_.encoder_settings.encoder = encoder.get(); video_send_config_.encoder_settings.payload_name = "VP8"; @@ -926,7 +926,7 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) { // Create frames that are smaller than the send width/height, this is done to // check that the callbacks are done after processing video. - rtc::scoped_ptr frame_generator( + std::unique_ptr frame_generator( test::FrameGenerator::CreateChromaGenerator(kWidth / 2, kHeight / 2)); video_send_stream_->Input()->IncomingCapturedFrame( *frame_generator->NextFrame()); @@ -1213,16 +1213,16 @@ class MultiStreamTest { virtual ~MultiStreamTest() {} void RunTest() { - rtc::scoped_ptr sender_call(Call::Create(Call::Config())); - rtc::scoped_ptr receiver_call(Call::Create(Call::Config())); - rtc::scoped_ptr sender_transport( + std::unique_ptr sender_call(Call::Create(Call::Config())); + std::unique_ptr receiver_call(Call::Create(Call::Config())); + std::unique_ptr sender_transport( CreateSendTransport(sender_call.get())); - rtc::scoped_ptr receiver_transport( + std::unique_ptr receiver_transport( CreateReceiveTransport(receiver_call.get())); sender_transport->SetReceiver(receiver_call->Receiver()); receiver_transport->SetReceiver(sender_call->Receiver()); - rtc::scoped_ptr encoders[kNumStreams]; + std::unique_ptr encoders[kNumStreams]; for (size_t i = 0; i < kNumStreams; ++i) encoders[i].reset(VideoEncoder::Create(VideoEncoder::kVp8)); @@ -1374,7 +1374,7 @@ TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) { } private: - rtc::scoped_ptr observers_[kNumStreams]; + std::unique_ptr observers_[kNumStreams]; } tester; tester.RunTest(); @@ -1492,7 +1492,7 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) { rtc::CriticalSection lock_; rtc::Event done_; - rtc::scoped_ptr parser_; + std::unique_ptr parser_; SequenceNumberUnwrapper unwrapper_; std::set received_packed_ids_; std::set streams_observed_; @@ -1706,7 +1706,7 @@ TEST_F(EndToEndTest, ObserversEncodedFrames) { } private: - rtc::scoped_ptr buffer_; + std::unique_ptr buffer_; size_t length_; FrameType frame_type_; rtc::Event called_; @@ -1730,7 +1730,7 @@ TEST_F(EndToEndTest, ObserversEncodedFrames) { CreateVideoStreams(); Start(); - rtc::scoped_ptr frame_generator( + std::unique_ptr frame_generator( test::FrameGenerator::CreateChromaGenerator( video_encoder_config_.streams[0].width, video_encoder_config_.streams[0].height)); @@ -1960,7 +1960,7 @@ TEST_F(EndToEndTest, RembWithSendSideBwe) { Clock* const clock_; uint32_t sender_ssrc_; int remb_bitrate_bps_; - rtc::scoped_ptr rtp_rtcp_; + std::unique_ptr rtp_rtcp_; test::PacketTransport* receive_transport_; rtc::Event event_; rtc::PlatformThread poller_thread_; @@ -1986,7 +1986,7 @@ TEST_F(EndToEndTest, VerifyNackStats) { Action OnSendRtp(const uint8_t* packet, size_t length) override { rtc::CritScope lock(&crit_); if (++sent_rtp_packets_ == kPacketNumberToDrop) { - rtc::scoped_ptr parser(RtpHeaderParser::Create()); + std::unique_ptr parser(RtpHeaderParser::Create()); RTPHeader header; EXPECT_TRUE(parser->Parse(packet, length, &header)); dropped_rtp_packet_ = header.sequenceNumber; @@ -2162,7 +2162,7 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx, const bool use_rtx_; const bool use_red_; const bool screenshare_; - const rtc::scoped_ptr vp8_encoder_; + const std::unique_ptr vp8_encoder_; Call* sender_call_; Call* receiver_call_; int64_t start_runtime_ms_; diff --git a/webrtc/video/overuse_frame_detector.cc b/webrtc/video/overuse_frame_detector.cc index 18c6b9e7ed..522a505594 100644 --- a/webrtc/video/overuse_frame_detector.cc +++ b/webrtc/video/overuse_frame_detector.cc @@ -166,8 +166,8 @@ class OveruseFrameDetector::SendProcessingUsage { const float kMaxSampleDiffMs; uint64_t count_; const CpuOveruseOptions options_; - rtc::scoped_ptr filtered_processing_ms_; - rtc::scoped_ptr filtered_frame_diff_ms_; + std::unique_ptr filtered_processing_ms_; + std::unique_ptr filtered_frame_diff_ms_; }; OveruseFrameDetector::OveruseFrameDetector( diff --git a/webrtc/video/overuse_frame_detector.h b/webrtc/video/overuse_frame_detector.h index 43c9e28459..9f78c6c0ff 100644 --- a/webrtc/video/overuse_frame_detector.h +++ b/webrtc/video/overuse_frame_detector.h @@ -12,11 +12,11 @@ #define WEBRTC_VIDEO_OVERUSE_FRAME_DETECTOR_H_ #include +#include #include "webrtc/base/constructormagic.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/optional.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/exp_filter.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/base/thread_checker.h" @@ -154,7 +154,7 @@ class OveruseFrameDetector : public Module { // TODO(asapersson): Can these be regular members (avoid separate heap // allocs)? - const rtc::scoped_ptr usage_ GUARDED_BY(crit_); + const std::unique_ptr usage_ GUARDED_BY(crit_); std::list frame_timing_ GUARDED_BY(crit_); rtc::ThreadChecker processing_thread_; diff --git a/webrtc/video/overuse_frame_detector_unittest.cc b/webrtc/video/overuse_frame_detector_unittest.cc index 1a6384ca8a..06cff38bf6 100644 --- a/webrtc/video/overuse_frame_detector_unittest.cc +++ b/webrtc/video/overuse_frame_detector_unittest.cc @@ -8,12 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include + #include "webrtc/video/overuse_frame_detector.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/video_frame.h" @@ -121,9 +122,9 @@ class OveruseFrameDetectorTest : public ::testing::Test, int UsagePercent() { return metrics_.encode_usage_percent; } CpuOveruseOptions options_; - rtc::scoped_ptr clock_; - rtc::scoped_ptr observer_; - rtc::scoped_ptr overuse_detector_; + std::unique_ptr clock_; + std::unique_ptr observer_; + std::unique_ptr overuse_detector_; CpuOveruseMetrics metrics_; }; diff --git a/webrtc/video/payload_router.h b/webrtc/video/payload_router.h index 661856d2a5..9eaf716322 100644 --- a/webrtc/video/payload_router.h +++ b/webrtc/video/payload_router.h @@ -15,7 +15,6 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/criticalsection.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/system_wrappers/include/atomic32.h" diff --git a/webrtc/video/payload_router_unittest.cc b/webrtc/video/payload_router_unittest.cc index 9b831a3ef7..5fe478f904 100644 --- a/webrtc/video/payload_router_unittest.cc +++ b/webrtc/video/payload_router_unittest.cc @@ -8,9 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include + #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_rtcp.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/video/payload_router.h" @@ -27,7 +28,7 @@ class PayloadRouterTest : public ::testing::Test { virtual void SetUp() { payload_router_.reset(new PayloadRouter()); } - rtc::scoped_ptr payload_router_; + std::unique_ptr payload_router_; }; TEST_F(PayloadRouterTest, SendOnOneModule) { diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc index 484924872b..52b6ff6075 100644 --- a/webrtc/video/replay.cc +++ b/webrtc/video/replay.cc @@ -11,13 +11,13 @@ #include #include +#include #include #include "gflags/gflags.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/checks.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" @@ -209,12 +209,12 @@ class DecoderBitstreamFileWriter : public EncodedFrameObserver { }; void RtpReplay() { - rtc::scoped_ptr playback_video( + std::unique_ptr playback_video( test::VideoRenderer::Create("Playback Video", 640, 480)); FileRenderPassthrough file_passthrough(flags::OutBase(), playback_video.get()); - rtc::scoped_ptr call(Call::Create(Call::Config())); + std::unique_ptr call(Call::Create(Call::Config())); test::NullTransport transport; VideoReceiveStream::Config receive_config(&transport); @@ -237,7 +237,7 @@ void RtpReplay() { encoder_settings.payload_name = flags::Codec(); encoder_settings.payload_type = flags::PayloadType(); VideoReceiveStream::Decoder decoder; - rtc::scoped_ptr bitstream_writer; + std::unique_ptr bitstream_writer; if (!flags::DecoderBitstreamFilename().empty()) { bitstream_writer.reset(new DecoderBitstreamFileWriter( flags::DecoderBitstreamFilename().c_str())); @@ -255,7 +255,7 @@ void RtpReplay() { VideoReceiveStream* receive_stream = call->CreateVideoReceiveStream(receive_config); - rtc::scoped_ptr rtp_reader(test::RtpFileReader::Create( + std::unique_ptr rtp_reader(test::RtpFileReader::Create( test::RtpFileReader::kRtpDump, flags::InputFile())); if (rtp_reader.get() == nullptr) { rtp_reader.reset(test::RtpFileReader::Create(test::RtpFileReader::kPcap, @@ -290,7 +290,7 @@ void RtpReplay() { break; case PacketReceiver::DELIVERY_UNKNOWN_SSRC: { RTPHeader header; - rtc::scoped_ptr parser(RtpHeaderParser::Create()); + std::unique_ptr parser(RtpHeaderParser::Create()); parser->Parse(packet.data, packet.length, &header); if (unknown_packets[header.ssrc] == 0) fprintf(stderr, "Unknown SSRC: %u!\n", header.ssrc); diff --git a/webrtc/video/send_statistics_proxy.h b/webrtc/video/send_statistics_proxy.h index 24a09b0006..66f03367b2 100644 --- a/webrtc/video/send_statistics_proxy.h +++ b/webrtc/video/send_statistics_proxy.h @@ -12,12 +12,12 @@ #define WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_ #include +#include #include #include "webrtc/base/criticalsection.h" #include "webrtc/base/exp_filter.h" #include "webrtc/base/ratetracker.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/modules/video_coding/include/video_codec_interface.h" @@ -174,7 +174,7 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver, const VideoSendStream::Stats start_stats_; }; - rtc::scoped_ptr uma_container_ GUARDED_BY(crit_); + std::unique_ptr uma_container_ GUARDED_BY(crit_); }; } // namespace webrtc diff --git a/webrtc/video/send_statistics_proxy_unittest.cc b/webrtc/video/send_statistics_proxy_unittest.cc index b3da5e978a..a98505f84d 100644 --- a/webrtc/video/send_statistics_proxy_unittest.cc +++ b/webrtc/video/send_statistics_proxy_unittest.cc @@ -12,6 +12,7 @@ #include "webrtc/video/send_statistics_proxy.h" #include +#include #include #include @@ -94,7 +95,7 @@ class SendStatisticsProxyTest : public ::testing::Test { } SimulatedClock fake_clock_; - rtc::scoped_ptr statistics_proxy_; + std::unique_ptr statistics_proxy_; VideoSendStream::Config config_; int avg_delay_ms_; int max_delay_ms_; diff --git a/webrtc/video/video_capture_input_unittest.cc b/webrtc/video/video_capture_input_unittest.cc index d20b999c2b..357914d997 100644 --- a/webrtc/video/video_capture_input_unittest.cc +++ b/webrtc/video/video_capture_input_unittest.cc @@ -9,12 +9,12 @@ */ #include "webrtc/video/video_capture_input.h" +#include #include #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/event.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/system_wrappers/include/ref_count.h" #include "webrtc/system_wrappers/include/scoped_vector.h" #include "webrtc/test/fake_texture_frame.h" @@ -82,12 +82,12 @@ class VideoCaptureInputTest : public ::testing::Test { SendStatisticsProxy stats_proxy_; - rtc::scoped_ptr mock_frame_callback_; + std::unique_ptr mock_frame_callback_; - rtc::scoped_ptr overuse_detector_; + std::unique_ptr overuse_detector_; // Used to send input capture frames to VideoCaptureInput. - rtc::scoped_ptr input_; + std::unique_ptr input_; // Input capture frames of VideoCaptureInput. ScopedVector input_frames_; diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc index 0fc125c759..84dbb5104e 100644 --- a/webrtc/video/video_quality_test.cc +++ b/webrtc/video/video_quality_test.cc @@ -21,7 +21,6 @@ #include "webrtc/base/checks.h" #include "webrtc/base/event.h" #include "webrtc/base/format_macros.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/timeutils.h" #include "webrtc/call.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" @@ -1039,7 +1038,7 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) { params_ = params; CheckParams(); - rtc::scoped_ptr local_preview( + std::unique_ptr local_preview( test::VideoRenderer::Create("Local Preview", params_.common.width, params_.common.height)); size_t stream_id = params_.ss.selected_stream; @@ -1050,7 +1049,7 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) { title += " - Stream #" + s.str(); } - rtc::scoped_ptr loopback_video( + std::unique_ptr loopback_video( test::VideoRenderer::Create(title.c_str(), params_.ss.streams[stream_id].width, params_.ss.streams[stream_id].height)); @@ -1059,7 +1058,7 @@ void VideoQualityTest::RunWithVideoRenderer(const Params& params) { // match the full stack tests. Call::Config call_config; call_config.bitrate_config = params_.common.call_bitrate_config; - rtc::scoped_ptr call(Call::Create(call_config)); + std::unique_ptr call(Call::Create(call_config)); test::LayerFilteringTransport transport( params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9, diff --git a/webrtc/video/video_quality_test.h b/webrtc/video/video_quality_test.h index dd2b011cc3..b476004aae 100644 --- a/webrtc/video/video_quality_test.h +++ b/webrtc/video/video_quality_test.h @@ -10,6 +10,7 @@ #ifndef WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_ #define WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_ +#include #include #include @@ -103,10 +104,10 @@ class VideoQualityTest : public test::CallTest { void SetupScreenshare(); // We need a more general capturer than the FrameGeneratorCapturer. - rtc::scoped_ptr capturer_; - rtc::scoped_ptr trace_to_stderr_; - rtc::scoped_ptr frame_generator_; - rtc::scoped_ptr encoder_; + std::unique_ptr capturer_; + std::unique_ptr trace_to_stderr_; + std::unique_ptr frame_generator_; + std::unique_ptr encoder_; VideoCodecUnion codec_settings_; Clock* const clock_; diff --git a/webrtc/video/video_receive_stream.h b/webrtc/video/video_receive_stream.h index 5510945818..f1061dc130 100644 --- a/webrtc/video/video_receive_stream.h +++ b/webrtc/video/video_receive_stream.h @@ -11,9 +11,9 @@ #ifndef WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ #define WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_ +#include #include -#include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" #include "webrtc/call/transport_adapter.h" #include "webrtc/common_video/include/incoming_video_stream.h" @@ -95,7 +95,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream, CallStats* const call_stats_; VieRemb* const remb_; - rtc::scoped_ptr vcm_; + std::unique_ptr vcm_; IncomingVideoStream incoming_video_stream_; ReceiveStatisticsProxy stats_proxy_; ViEChannel vie_channel_; diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc index e8f1101606..4243ee69de 100644 --- a/webrtc/video/video_send_stream_tests.cc +++ b/webrtc/video/video_send_stream_tests.cc @@ -8,6 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ #include // max +#include #include #include "testing/gtest/include/gtest/gtest.h" @@ -18,7 +19,6 @@ #include "webrtc/base/event.h" #include "webrtc/base/logging.h" #include "webrtc/base/platform_thread.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" #include "webrtc/call/transport_adapter.h" #include "webrtc/frame_callback.h" @@ -304,7 +304,7 @@ class FakeReceiveStatistics : public NullReceiveStatistics { RtcpStatistics stats_; }; - rtc::scoped_ptr lossy_stats_; + std::unique_ptr lossy_stats_; StatisticianMap stats_map_; }; @@ -442,8 +442,8 @@ class FecObserver : public test::EndToEndTest { EXPECT_TRUE(Wait()) << "Timed out waiting for FEC and media packets."; } - rtc::scoped_ptr transport_adapter_; - rtc::scoped_ptr encoder_; + std::unique_ptr transport_adapter_; + std::unique_ptr encoder_; const std::string payload_name_; const bool use_nack_; const bool expect_red_; @@ -562,7 +562,7 @@ void VideoSendStreamTest::TestNackRetransmission( EXPECT_TRUE(Wait()) << "Timed out while waiting for NACK retransmission."; } - rtc::scoped_ptr transport_adapter_; + std::unique_ptr transport_adapter_; int send_count_; uint32_t retransmit_ssrc_; uint8_t retransmit_payload_type_; @@ -758,7 +758,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format, EXPECT_TRUE(Wait()) << "Timed out while observing incoming RTP packets."; } - rtc::scoped_ptr transport_adapter_; + std::unique_ptr transport_adapter_; test::ConfigurableFrameSizeEncoder encoder_; const size_t max_packet_size_; @@ -937,7 +937,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) { EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr)); } - rtc::scoped_ptr transport_adapter_; + std::unique_ptr transport_adapter_; Clock* const clock_; VideoSendStream* stream_; @@ -1015,7 +1015,7 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) { } Clock* const clock_; - rtc::scoped_ptr transport_adapter_; + std::unique_ptr transport_adapter_; rtc::CriticalSection crit_; int64_t last_packet_time_ms_ GUARDED_BY(crit_); test::FrameGeneratorCapturer* capturer_ GUARDED_BY(crit_); @@ -1103,8 +1103,8 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) { << "Timeout while waiting for low bitrate stats after REMB."; } - rtc::scoped_ptr rtp_rtcp_; - rtc::scoped_ptr feedback_transport_; + std::unique_ptr rtp_rtcp_; + std::unique_ptr feedback_transport_; VideoSendStream* stream_; bool bitrate_capped_; } test; @@ -1292,7 +1292,7 @@ void ExpectEqualFramesVector(const std::vector& frames1, VideoFrame CreateVideoFrame(int width, int height, uint8_t data) { const int kSizeY = width * height * 2; - rtc::scoped_ptr buffer(new uint8_t[kSizeY]); + std::unique_ptr buffer(new uint8_t[kSizeY]); memset(buffer.get(), data, kSizeY); VideoFrame frame; frame.CreateFrame(buffer.get(), buffer.get(), buffer.get(), width, height, @@ -2168,7 +2168,7 @@ class Vp9HeaderObserver : public test::SendTest { VerifyTl0Idx(vp9); } - rtc::scoped_ptr vp9_encoder_; + std::unique_ptr vp9_encoder_; VideoCodecVP9 vp9_settings_; webrtc::VideoEncoderConfig encoder_config_; RTPHeader last_header_; diff --git a/webrtc/video/vie_channel.h b/webrtc/video/vie_channel.h index afb8d890a8..b89b800bab 100644 --- a/webrtc/video/vie_channel.h +++ b/webrtc/video/vie_channel.h @@ -13,11 +13,11 @@ #include #include +#include #include #include "webrtc/base/criticalsection.h" #include "webrtc/base/platform_thread.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" @@ -284,13 +284,13 @@ class ViEChannel : public VCMFrameTypeCallback, rtc::CriticalSection crit_; // Owned modules/classes. - rtc::scoped_ptr vcm_protection_callback_; + std::unique_ptr vcm_protection_callback_; VideoCodingModule* const vcm_; ViEReceiver vie_receiver_; // Helper to report call statistics. - rtc::scoped_ptr stats_observer_; + std::unique_ptr stats_observer_; // Not owned. ReceiveStatisticsProxy* receive_stats_callback_ GUARDED_BY(crit_); @@ -301,7 +301,7 @@ class ViEChannel : public VCMFrameTypeCallback, PacedSender* const paced_sender_; PacketRouter* const packet_router_; - const rtc::scoped_ptr bandwidth_observer_; + const std::unique_ptr bandwidth_observer_; TransportFeedbackObserver* const transport_feedback_observer_; int max_nack_reordering_threshold_; diff --git a/webrtc/video/vie_encoder.h b/webrtc/video/vie_encoder.h index 3bb6d3f3bf..77046cb9dc 100644 --- a/webrtc/video/vie_encoder.h +++ b/webrtc/video/vie_encoder.h @@ -11,10 +11,10 @@ #ifndef WEBRTC_VIDEO_VIE_ENCODER_H_ #define WEBRTC_VIDEO_VIE_ENCODER_H_ +#include #include #include "webrtc/base/criticalsection.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/call/bitrate_allocator.h" @@ -139,12 +139,12 @@ class ViEEncoder : public VideoEncoderRateObserver, const uint32_t number_of_cores_; const std::vector ssrcs_; - const rtc::scoped_ptr vp_; - const rtc::scoped_ptr qm_callback_; - const rtc::scoped_ptr vcm_; + const std::unique_ptr vp_; + const std::unique_ptr qm_callback_; + const std::unique_ptr vcm_; rtc::CriticalSection data_cs_; - rtc::scoped_ptr bitrate_observer_; + std::unique_ptr bitrate_observer_; SendStatisticsProxy* const stats_proxy_; I420FrameCallback* const pre_encode_callback_; diff --git a/webrtc/video/vie_receiver.h b/webrtc/video/vie_receiver.h index ccfbd459b6..b6e19cb26e 100644 --- a/webrtc/video/vie_receiver.h +++ b/webrtc/video/vie_receiver.h @@ -12,10 +12,10 @@ #define WEBRTC_VIDEO_VIE_RECEIVER_H_ #include +#include #include #include -#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/criticalsection.h" #include "webrtc/engine_configurations.h" #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" @@ -100,17 +100,17 @@ class ViEReceiver : public RtpData { rtc::CriticalSection receive_cs_; Clock* clock_; - rtc::scoped_ptr rtp_header_parser_; - rtc::scoped_ptr rtp_payload_registry_; - rtc::scoped_ptr rtp_receiver_; - const rtc::scoped_ptr rtp_receive_statistics_; - rtc::scoped_ptr fec_receiver_; + std::unique_ptr rtp_header_parser_; + std::unique_ptr rtp_payload_registry_; + std::unique_ptr rtp_receiver_; + const std::unique_ptr rtp_receive_statistics_; + std::unique_ptr fec_receiver_; RtpRtcp* rtp_rtcp_; std::vector rtp_rtcp_simulcast_; VideoCodingModule* vcm_; RemoteBitrateEstimator* remote_bitrate_estimator_; - rtc::scoped_ptr ntp_estimator_; + std::unique_ptr ntp_estimator_; bool receiving_; uint8_t restored_packet_[IP_PACKET_SIZE]; diff --git a/webrtc/video/vie_remb.h b/webrtc/video/vie_remb.h index 39dbc85311..d2c60dbeb1 100644 --- a/webrtc/video/vie_remb.h +++ b/webrtc/video/vie_remb.h @@ -16,7 +16,6 @@ #include #include "webrtc/base/criticalsection.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/include/module.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" diff --git a/webrtc/video/vie_remb_unittest.cc b/webrtc/video/vie_remb_unittest.cc index 3a69cdb07c..5f72b967cb 100644 --- a/webrtc/video/vie_remb_unittest.cc +++ b/webrtc/video/vie_remb_unittest.cc @@ -11,11 +11,11 @@ // This file includes unit tests for ViERemb. +#include #include #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_rtcp.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/modules/utility/include/mock/mock_process_thread.h" @@ -39,8 +39,8 @@ class ViERembTest : public ::testing::Test { vie_remb_.reset(new VieRemb(&fake_clock_)); } SimulatedClock fake_clock_; - rtc::scoped_ptr process_thread_; - rtc::scoped_ptr vie_remb_; + std::unique_ptr process_thread_; + std::unique_ptr vie_remb_; }; TEST_F(ViERembTest, OneModuleTestForSendingRemb) { diff --git a/webrtc/video/vie_sync_module.h b/webrtc/video/vie_sync_module.h index 5724ce799a..a5dff437f0 100644 --- a/webrtc/video/vie_sync_module.h +++ b/webrtc/video/vie_sync_module.h @@ -14,8 +14,9 @@ #ifndef WEBRTC_VIDEO_VIE_SYNC_MODULE_H_ #define WEBRTC_VIDEO_VIE_SYNC_MODULE_H_ +#include + #include "webrtc/base/criticalsection.h" -#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/include/module.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/video/stream_synchronization.h" @@ -50,7 +51,7 @@ class ViESyncModule : public Module { int voe_channel_id_; VoEVideoSync* voe_sync_interface_; TickTime last_sync_time_; - rtc::scoped_ptr sync_; + std::unique_ptr sync_; StreamSynchronization::Measurements audio_measurement_; StreamSynchronization::Measurements video_measurement_; };