diff --git a/media/BUILD.gn b/media/BUILD.gn index 026715a5ff..3042ac2a86 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -421,6 +421,7 @@ if (rtc_include_tests) { "base/fakertp.h", "base/fakevideocapturer.cc", "base/fakevideocapturer.h", + "base/fakevideorenderer.cc", "base/fakevideorenderer.h", "base/testutils.cc", "base/testutils.h", diff --git a/media/base/fakevideorenderer.cc b/media/base/fakevideorenderer.cc new file mode 100644 index 0000000000..fdd789a4b7 --- /dev/null +++ b/media/base/fakevideorenderer.cc @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "media/base/fakevideorenderer.h" + +namespace cricket { + +FakeVideoRenderer::FakeVideoRenderer() = default; + +void FakeVideoRenderer::OnFrame(const webrtc::VideoFrame& frame) { + rtc::CritScope cs(&crit_); + // TODO(zhurunz) Check with VP8 team to see if we can remove this + // tolerance on Y values. Some unit tests produce Y values close + // to 16 rather than close to zero, for supposedly black frames. + // Largest value observed is 34, e.g., running + // PeerConnectionIntegrationTest.SendAndReceive16To9AspectRatio. + black_frame_ = CheckFrameColorYuv(0, 48, 128, 128, 128, 128, &frame); + // Treat unexpected frame size as error. + ++num_rendered_frames_; + width_ = frame.width(); + height_ = frame.height(); + rotation_ = frame.rotation(); + timestamp_us_ = frame.timestamp_us(); +} + +} // namespace cricket diff --git a/media/base/fakevideorenderer.h b/media/base/fakevideorenderer.h index a7c532a7d2..f072c729e6 100644 --- a/media/base/fakevideorenderer.h +++ b/media/base/fakevideorenderer.h @@ -21,30 +21,9 @@ namespace cricket { // Faked video renderer that has a callback for actions on rendering. class FakeVideoRenderer : public rtc::VideoSinkInterface { public: - FakeVideoRenderer() - : errors_(0), - width_(0), - height_(0), - rotation_(webrtc::kVideoRotation_0), - timestamp_us_(0), - num_rendered_frames_(0), - black_frame_(false) {} + FakeVideoRenderer(); - virtual void OnFrame(const webrtc::VideoFrame& frame) { - rtc::CritScope cs(&crit_); - // TODO(zhurunz) Check with VP8 team to see if we can remove this - // tolerance on Y values. Some unit tests produce Y values close - // to 16 rather than close to zero, for supposedly black frames. - // Largest value observed is 34, e.g., running - // PeerConnectionIntegrationTest.SendAndReceive16To9AspectRatio. - black_frame_ = CheckFrameColorYuv(0, 48, 128, 128, 128, 128, &frame); - // Treat unexpected frame size as error. - ++num_rendered_frames_; - width_ = frame.width(); - height_ = frame.height(); - rotation_ = frame.rotation(); - timestamp_us_ = frame.timestamp_us(); - } + void OnFrame(const webrtc::VideoFrame& frame) override; int errors() const { return errors_; } int width() const { @@ -127,13 +106,13 @@ class FakeVideoRenderer : public rtc::VideoSinkInterface { return true; } - int errors_; - int width_; - int height_; - webrtc::VideoRotation rotation_; - int64_t timestamp_us_; - int num_rendered_frames_; - bool black_frame_; + int errors_ = 0; + int width_ = 0; + int height_ = 0; + webrtc::VideoRotation rotation_ = webrtc::kVideoRotation_0; + int64_t timestamp_us_ = 0; + int num_rendered_frames_ = 0; + bool black_frame_ = false; rtc::CriticalSection crit_; }; diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn index 34f103bb4e..4501391492 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn @@ -199,12 +199,6 @@ if (is_ios || is_mac) { "../rtc_base:rtc_base", "../rtc_base:rtc_base_approved", ] - - if (is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } } } @@ -234,11 +228,6 @@ if (is_ios || is_mac) { "..:common_objc", ":used_from_extension", ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } } rtc_static_library("videoframebuffer_objc") { @@ -316,11 +305,6 @@ if (is_ios || is_mac) { "..:common_objc", ":used_from_extension", ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } } rtc_static_library("ui_objc") { @@ -442,12 +426,6 @@ if (is_ios || is_mac) { public_configs = [ ":common_config_objc" ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - deps = [ ":common_objc", ":video_objc", @@ -519,12 +497,6 @@ if (is_ios || is_mac) { "objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h", ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - deps = [ ":wrapped_native_codec_objc", "../modules/video_coding:webrtc_vp8", @@ -541,12 +513,6 @@ if (is_ios || is_mac) { "objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h", ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - deps = [ ":wrapped_native_codec_objc", "../modules/video_coding:webrtc_vp9", @@ -575,12 +541,6 @@ if (is_ios || is_mac) { public_configs = [ ":common_config_objc" ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - deps = [ ":native_api", ":native_video", @@ -596,11 +556,6 @@ if (is_ios || is_mac) { "objc/Framework/Classes/PeerConnection/RTCMediaConstraints.mm", "objc/Framework/Headers/WebRTC/RTCMediaConstraints.h", ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } public_configs = [ ":common_config_objc" ] deps = [ @@ -771,12 +726,6 @@ if (is_ios || is_mac) { ] public_configs = [ ":common_config_objc" ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - deps = [ ":common_objc", ":mediaconstraints_objc", @@ -879,12 +828,6 @@ if (is_ios || is_mac) { ] include_dirs += [ "$root_out_dir/WebRTC.framework/Headers/" ] - - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } } bundle_data("sdk_unittests_bundle_data") { @@ -915,12 +858,6 @@ if (is_ios || is_mac) { ":sdk_unittests_sources", ] ldflags = [ "-all_load" ] - - if (is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } } } @@ -986,12 +923,6 @@ if (is_ios || is_mac) { sources += [ "objc/Framework/UnitTests/RTCAudioSessionTest.mm" ] deps += [ ":audio_objc" ] } - - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } } } @@ -1111,12 +1042,6 @@ if (is_ios || is_mac) { ] public_configs = [ ":common_config_objc" ] - - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - configs -= [ "//build/config/clang:find_bad_constructs" ] - } } } @@ -1131,12 +1056,6 @@ if (is_ios || is_mac) { configs += [ "..:common_objc" ] public_configs = [ ":common_config_objc" ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - deps = [ ":common_objc", ":videocodec_objc", @@ -1171,12 +1090,6 @@ if (is_ios || is_mac) { public_configs = [ ":common_config_objc" ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - deps = [ ":native_video", ":videocapturebase_objc", @@ -1254,11 +1167,6 @@ if (is_ios || is_mac) { "../rtc_base:checks", "../rtc_base:rtc_base_approved", ] - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } } rtc_static_library("videotoolbox_objc") { @@ -1309,12 +1217,6 @@ if (is_ios || is_mac) { "CoreVideo.framework", "VideoToolbox.framework", ] - - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin - # (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } } } } diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h index 75cd79ba58..1c8eb28895 100644 --- a/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h +++ b/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h @@ -18,12 +18,12 @@ namespace webrtc { class MediaConstraints : public MediaConstraintsInterface { public: - virtual ~MediaConstraints(); + ~MediaConstraints() override; MediaConstraints(); MediaConstraints(const MediaConstraintsInterface::Constraints& mandatory, const MediaConstraintsInterface::Constraints& optional); - virtual const Constraints& GetMandatory() const; - virtual const Constraints& GetOptional() const; + const Constraints& GetMandatory() const override; + const Constraints& GetOptional() const override; private: MediaConstraintsInterface::Constraints mandatory_; diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h index 15b05de3ac..1d02b63691 100644 --- a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h +++ b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h @@ -23,7 +23,7 @@ namespace webrtc { class PeerConnectionDelegateAdapter : public PeerConnectionObserver { public: PeerConnectionDelegateAdapter(RTCPeerConnection *peerConnection); - virtual ~PeerConnectionDelegateAdapter(); + ~PeerConnectionDelegateAdapter() override; void OnSignalingChange(PeerConnectionInterface::SignalingState new_state) override; diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm index fb16a31b3b..e39972261b 100644 --- a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm +++ b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm @@ -25,9 +25,7 @@ class StatsObserverAdapter : public StatsObserver { completion_handler_ = completionHandler; } - ~StatsObserverAdapter() { - completion_handler_ = nil; - } + ~StatsObserverAdapter() override { completion_handler_ = nil; } void OnComplete(const StatsReports& reports) override { RTC_DCHECK(completion_handler_); diff --git a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm index 947bba6e1c..103b3c095d 100644 --- a/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm +++ b/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm @@ -46,9 +46,7 @@ class CreateSessionDescriptionObserverAdapter completion_handler_ = completionHandler; } - ~CreateSessionDescriptionObserverAdapter() { - completion_handler_ = nil; - } + ~CreateSessionDescriptionObserverAdapter() override { completion_handler_ = nil; } void OnSuccess(SessionDescriptionInterface *desc) override { RTC_DCHECK(completion_handler_); @@ -86,9 +84,7 @@ class SetSessionDescriptionObserverAdapter : completion_handler_ = completionHandler; } - ~SetSessionDescriptionObserverAdapter() { - completion_handler_ = nil; - } + ~SetSessionDescriptionObserverAdapter() override { completion_handler_ = nil; } void OnSuccess() override { RTC_DCHECK(completion_handler_); diff --git a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc index a373050803..479ef3c6bb 100644 --- a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc +++ b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc @@ -286,6 +286,8 @@ AnnexBBufferReader::AnnexBBufferReader(const uint8_t* annexb_buffer, offset_ = offsets_.begin(); } +AnnexBBufferReader::~AnnexBBufferReader() = default; + bool AnnexBBufferReader::ReadNalu(const uint8_t** out_nalu, size_t* out_length) { RTC_DCHECK(out_nalu); diff --git a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h index 0155882648..cd5a51079a 100644 --- a/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h +++ b/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h @@ -57,7 +57,7 @@ CMVideoFormatDescriptionRef CreateVideoFormatDescription( class AnnexBBufferReader final { public: AnnexBBufferReader(const uint8_t* annexb_buffer, size_t length); - ~AnnexBBufferReader() {} + ~AnnexBBufferReader(); AnnexBBufferReader(const AnnexBBufferReader& other) = delete; void operator=(const AnnexBBufferReader& other) = delete; diff --git a/sdk/objc/Framework/Native/src/audio/audio_device_ios.h b/sdk/objc/Framework/Native/src/audio/audio_device_ios.h index 56326c12b7..23e54b1409 100644 --- a/sdk/objc/Framework/Native/src/audio/audio_device_ios.h +++ b/sdk/objc/Framework/Native/src/audio/audio_device_ios.h @@ -49,7 +49,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric, public rtc::MessageHandler { public: AudioDeviceIOS(); - ~AudioDeviceIOS(); + ~AudioDeviceIOS() override; void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override; @@ -65,11 +65,11 @@ class AudioDeviceIOS : public AudioDeviceGeneric, int32_t StartPlayout() override; int32_t StopPlayout() override; - bool Playing() const override { return playing_; } + bool Playing() const override; int32_t StartRecording() override; int32_t StopRecording() override; - bool Recording() const override { return recording_; } + bool Recording() const override; // These methods returns hard-coded delay values and not dynamic delay // estimates. The reason is that iOS supports a built-in AEC and the WebRTC diff --git a/sdk/objc/Framework/Native/src/audio/audio_device_ios.mm b/sdk/objc/Framework/Native/src/audio/audio_device_ios.mm index 3779c2552e..dfa4deae8d 100644 --- a/sdk/objc/Framework/Native/src/audio/audio_device_ios.mm +++ b/sdk/objc/Framework/Native/src/audio/audio_device_ios.mm @@ -271,6 +271,10 @@ int32_t AudioDeviceIOS::StopPlayout() { return 0; } +bool AudioDeviceIOS::Playing() const { + return playing_; +} + int32_t AudioDeviceIOS::StartRecording() { LOGI() << "StartRecording"; RTC_DCHECK_RUN_ON(&thread_checker_); @@ -305,6 +309,10 @@ int32_t AudioDeviceIOS::StopRecording() { return 0; } +bool AudioDeviceIOS::Recording() const { + return recording_; +} + int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const { delayMS = kFixedPlayoutDelayEstimate; return 0; diff --git a/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h b/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h index 2967b0adf7..3e2980896e 100644 --- a/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h +++ b/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h @@ -21,7 +21,7 @@ namespace webrtc { class ObjCVideoDecoderFactory : public VideoDecoderFactory { public: explicit ObjCVideoDecoderFactory(id); - ~ObjCVideoDecoderFactory(); + ~ObjCVideoDecoderFactory() override; id wrapped_decoder_factory() const; diff --git a/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h b/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h index 8c3a5a82c9..43a3bf431e 100644 --- a/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h +++ b/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h @@ -22,7 +22,7 @@ namespace webrtc { class ObjCVideoEncoderFactory : public VideoEncoderFactory { public: explicit ObjCVideoEncoderFactory(id); - ~ObjCVideoEncoderFactory(); + ~ObjCVideoEncoderFactory() override; id wrapped_encoder_factory() const; diff --git a/sdk/objc/Framework/Native/src/objc_video_track_source.h b/sdk/objc/Framework/Native/src/objc_video_track_source.h index d237980d40..e26b281571 100644 --- a/sdk/objc/Framework/Native/src/objc_video_track_source.h +++ b/sdk/objc/Framework/Native/src/objc_video_track_source.h @@ -31,16 +31,16 @@ class ObjCVideoTrackSource : public rtc::AdaptedVideoTrackSource { // This class can not be used for implementing screen casting. Hopefully, this // function will be removed before we add that to iOS/Mac. - bool is_screencast() const override { return false; } + bool is_screencast() const override; // Indicates that the encoder should denoise video before encoding it. // If it is not set, the default configuration is used which is different // depending on video codec. - absl::optional needs_denoising() const override { return false; } + absl::optional needs_denoising() const override; - SourceState state() const override { return SourceState::kLive; } + SourceState state() const override; - bool remote() const override { return false; } + bool remote() const override; void OnCapturedFrame(RTCVideoFrame* frame); diff --git a/sdk/objc/Framework/Native/src/objc_video_track_source.mm b/sdk/objc/Framework/Native/src/objc_video_track_source.mm index b430efa763..abb03781bf 100644 --- a/sdk/objc/Framework/Native/src/objc_video_track_source.mm +++ b/sdk/objc/Framework/Native/src/objc_video_track_source.mm @@ -39,6 +39,22 @@ ObjCVideoTrackSource::ObjCVideoTrackSource(RTCObjCVideoSourceAdapter *adapter) : adapter_.objCVideoTrackSource = this; } +bool ObjCVideoTrackSource::is_screencast() const { + return false; +} + +absl::optional ObjCVideoTrackSource::needs_denoising() const { + return false; +} + +MediaSourceInterface::SourceState ObjCVideoTrackSource::state() const { + return SourceState::kLive; +} + +bool ObjCVideoTrackSource::remote() const { + return false; +} + void ObjCVideoTrackSource::OnOutputFormatRequest(int width, int height, int fps) { cricket::VideoFormat format(width, height, cricket::VideoFormat::FpsToInterval(fps), 0); video_adapter()->OnOutputFormatRequest(format); diff --git a/sdk/objc/Framework/UnitTests/ObjCVideoTrackSource_xctest.mm b/sdk/objc/Framework/UnitTests/ObjCVideoTrackSource_xctest.mm index 7d111c8b12..a32ccce314 100644 --- a/sdk/objc/Framework/UnitTests/ObjCVideoTrackSource_xctest.mm +++ b/sdk/objc/Framework/UnitTests/ObjCVideoTrackSource_xctest.mm @@ -31,7 +31,7 @@ class ObjCCallbackVideoSink : public rtc::VideoSinkInterface public: ObjCCallbackVideoSink(VideoSinkCallback callback) : callback_(callback) {} - virtual void OnFrame(const webrtc::VideoFrame &frame) { + void OnFrame(const webrtc::VideoFrame &frame) override { callback_(NativeToObjCVideoFrame(frame)); } diff --git a/sdk/objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm b/sdk/objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm index 9644b2beac..739ec6af8c 100644 --- a/sdk/objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm +++ b/sdk/objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm @@ -43,7 +43,7 @@ typedef int32_t(^RecordedDataIsAvailableBlock)(const void* audioSamples, class MockAudioTransport : public webrtc::AudioTransport { public: MockAudioTransport() {} - ~MockAudioTransport() {} + ~MockAudioTransport() override {} void expectNeedMorePlayData(NeedMorePlayDataBlock block) { needMorePlayDataBlock = block; @@ -60,7 +60,7 @@ public: void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, - int64_t* ntp_time_ms) { + int64_t* ntp_time_ms) override { return needMorePlayDataBlock(nSamples, nBytesPerSample, nChannels, @@ -80,7 +80,7 @@ public: const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, - uint32_t& newMicLevel) { + uint32_t& newMicLevel) override { return recordedDataIsAvailableBlock(audioSamples, nSamples, nBytesPerSample, @@ -99,11 +99,9 @@ public: size_t number_of_frames, void* audio_data, int64_t* elapsed_time_ms, - int64_t* ntp_time_ms) { + int64_t* ntp_time_ms) override {} - } - -private: + private: NeedMorePlayDataBlock needMorePlayDataBlock; RecordedDataIsAvailableBlock recordedDataIsAvailableBlock; }; diff --git a/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm b/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm index 951b5ea260..ee52242ecf 100644 --- a/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm +++ b/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm @@ -299,7 +299,7 @@ namespace webrtc { class AudioSessionTest : public ::testing::Test { protected: - void TearDown() { + void TearDown() override { RTCAudioSession *session = [RTCAudioSession sharedInstance]; for (id delegate : session.delegates) { [session removeDelegate:delegate]; diff --git a/sdk/objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm b/sdk/objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm index 713ed9fc99..a8ea43cdf0 100644 --- a/sdk/objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm +++ b/sdk/objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm @@ -56,14 +56,14 @@ std::unique_ptr GetObjCDecoder(id TEST(ObjCVideoDecoderFactoryTest, InitDecodeReturnsOKOnSuccess) { std::unique_ptr decoder = GetObjCDecoder(CreateOKDecoderFactory()); - auto settings = new webrtc::VideoCodec(); + auto* settings = new webrtc::VideoCodec(); EXPECT_EQ(decoder->InitDecode(settings, 1), WEBRTC_VIDEO_CODEC_OK); } TEST(ObjCVideoDecoderFactoryTest, InitDecodeReturnsErrorOnFail) { std::unique_ptr decoder = GetObjCDecoder(CreateErrorDecoderFactory()); - auto settings = new webrtc::VideoCodec(); + auto* settings = new webrtc::VideoCodec(); EXPECT_EQ(decoder->InitDecode(settings, 1), WEBRTC_VIDEO_CODEC_ERROR); } diff --git a/sdk/objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm b/sdk/objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm index 2ae1ebbeb1..9377751d05 100644 --- a/sdk/objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm +++ b/sdk/objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm @@ -58,14 +58,14 @@ std::unique_ptr GetObjCEncoder(id TEST(ObjCVideoEncoderFactoryTest, InitEncodeReturnsOKOnSuccess) { std::unique_ptr encoder = GetObjCEncoder(CreateOKEncoderFactory()); - auto settings = new webrtc::VideoCodec(); + auto* settings = new webrtc::VideoCodec(); EXPECT_EQ(encoder->InitEncode(settings, 1, 0), WEBRTC_VIDEO_CODEC_OK); } TEST(ObjCVideoEncoderFactoryTest, InitEncodeReturnsErrorOnFail) { std::unique_ptr encoder = GetObjCEncoder(CreateErrorEncoderFactory()); - auto settings = new webrtc::VideoCodec(); + auto* settings = new webrtc::VideoCodec(); EXPECT_EQ(encoder->InitEncode(settings, 1, 0), WEBRTC_VIDEO_CODEC_ERROR); }