Enable -Winconsistent-missing-override flag. The problem with gmock is worked around by commenting out any other override declarations in classes using gmock. NOPRESUBMIT=True BUG=webrtc:3970 Review-Url: https://codereview.webrtc.org/1921653002 Cr-Commit-Position: refs/heads/master@{#12563}
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index 3ae09da..b31a07c 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn
@@ -117,6 +117,7 @@ cflags += [ "-Wimplicit-fallthrough", "-Wthread-safety", + "-Winconsistent-missing-override", ] } }
diff --git a/webrtc/api/dtmfsender.h b/webrtc/api/dtmfsender.h index e2c6735..c85557f 100644 --- a/webrtc/api/dtmfsender.h +++ b/webrtc/api/dtmfsender.h
@@ -83,7 +83,7 @@ DtmfSender(); // Implements MessageHandler. - virtual void OnMessage(rtc::Message* msg); + void OnMessage(rtc::Message* msg) override; // The DTMF sending task. void DoInsertDtmf();
diff --git a/webrtc/api/dtmfsender_unittest.cc b/webrtc/api/dtmfsender_unittest.cc index e6fa7fc..efe568d 100644 --- a/webrtc/api/dtmfsender_unittest.cc +++ b/webrtc/api/dtmfsender_unittest.cc
@@ -96,7 +96,7 @@ return true; } - virtual sigslot::signal0<>* GetOnDestroyedSignal() { + sigslot::signal0<>* GetOnDestroyedSignal() override { return &SignalDestroyed; }
diff --git a/webrtc/api/java/jni/peerconnection_jni.cc b/webrtc/api/java/jni/peerconnection_jni.cc index 34358ad..10b8134 100644 --- a/webrtc/api/java/jni/peerconnection_jni.cc +++ b/webrtc/api/java/jni/peerconnection_jni.cc
@@ -202,7 +202,7 @@ } void OnIceCandidatesRemoved( - const std::vector<cricket::Candidate>& candidates) { + const std::vector<cricket::Candidate>& candidates) override { ScopedLocalRefFrame local_ref_frame(jni()); jobjectArray candidates_array = ToJavaCandidateArray(jni(), candidates); jmethodID m =
diff --git a/webrtc/api/mediastream.h b/webrtc/api/mediastream.h index 2a77f0d..1f80f25 100644 --- a/webrtc/api/mediastream.h +++ b/webrtc/api/mediastream.h
@@ -31,10 +31,10 @@ bool AddTrack(VideoTrackInterface* track) override; bool RemoveTrack(AudioTrackInterface* track) override; bool RemoveTrack(VideoTrackInterface* track) override; - virtual rtc::scoped_refptr<AudioTrackInterface> - FindAudioTrack(const std::string& track_id); - virtual rtc::scoped_refptr<VideoTrackInterface> - FindVideoTrack(const std::string& track_id); + rtc::scoped_refptr<AudioTrackInterface> + FindAudioTrack(const std::string& track_id) override; + rtc::scoped_refptr<VideoTrackInterface> + FindVideoTrack(const std::string& track_id) override; AudioTrackVector GetAudioTracks() override { return audio_tracks_; } VideoTrackVector GetVideoTracks() override { return video_tracks_; }
diff --git a/webrtc/api/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h index 995c760..1992087 100644 --- a/webrtc/api/peerconnectionfactory.h +++ b/webrtc/api/peerconnectionfactory.h
@@ -35,7 +35,7 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface { public: - virtual void SetOptions(const Options& options) { + void SetOptions(const Options& options) override { options_ = options; }
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc index 738b736..1a8dd57 100644 --- a/webrtc/api/peerconnectioninterface_unittest.cc +++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -424,8 +424,8 @@ state_ = pc_->signaling_state(); } } - virtual void OnSignalingChange( - PeerConnectionInterface::SignalingState new_state) { + void OnSignalingChange( + PeerConnectionInterface::SignalingState new_state) override { EXPECT_EQ(pc_->signaling_state(), new_state); state_ = new_state; }
diff --git a/webrtc/api/rtpsender.h b/webrtc/api/rtpsender.h index fe61cbd..86de765 100644 --- a/webrtc/api/rtpsender.h +++ b/webrtc/api/rtpsender.h
@@ -98,8 +98,8 @@ void Stop() override; - RtpParameters GetParameters() const; - bool SetParameters(const RtpParameters& parameters); + RtpParameters GetParameters() const override; + bool SetParameters(const RtpParameters& parameters) override; private: // TODO(nisse): Since SSRC == 0 is technically valid, figure out @@ -164,8 +164,8 @@ void Stop() override; - RtpParameters GetParameters() const; - bool SetParameters(const RtpParameters& parameters); + RtpParameters GetParameters() const override; + bool SetParameters(const RtpParameters& parameters) override; private: bool can_send_track() const { return track_ && ssrc_; }
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc index 4cd1425..0ec3cfe 100644 --- a/webrtc/api/rtpsenderreceiver_unittest.cc +++ b/webrtc/api/rtpsenderreceiver_unittest.cc
@@ -43,7 +43,11 @@ // Helper class to test RtpSender/RtpReceiver. class MockAudioProvider : public AudioProviderInterface { public: - ~MockAudioProvider() override {} + // TODO(nisse): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. + ~MockAudioProvider() /* override */ {} MOCK_METHOD2(SetAudioPlayout, void(uint32_t ssrc, @@ -58,8 +62,8 @@ MOCK_METHOD2(SetAudioRtpParameters, bool(uint32_t ssrc, const RtpParameters&)); - void SetRawAudioSink(uint32_t, - std::unique_ptr<AudioSinkInterface> sink) override { + void SetRawAudioSink( + uint32_t, std::unique_ptr<AudioSinkInterface> sink) /* override */ { sink_ = std::move(sink); }
diff --git a/webrtc/api/statscollector_unittest.cc b/webrtc/api/statscollector_unittest.cc index 760db0f..2924c51 100644 --- a/webrtc/api/statscollector_unittest.cc +++ b/webrtc/api/statscollector_unittest.cc
@@ -67,6 +67,10 @@ class MockWebRtcSession : public webrtc::WebRtcSession { public: + // TODO(nisse): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. explicit MockWebRtcSession(webrtc::MediaControllerInterface* media_controller) : WebRtcSession(media_controller, rtc::Thread::Current(), @@ -85,7 +89,7 @@ // Workaround for gmock's inability to cope with move-only return values. std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate( - const std::string& transport_name) override { + const std::string& transport_name) /* override */ { return std::unique_ptr<rtc::SSLCertificate>( GetRemoteSSLCertificate_ReturnsRawPointer(transport_name)); }
diff --git a/webrtc/api/test/fakeaudiocapturemodule.h b/webrtc/api/test/fakeaudiocapturemodule.h index 30ad3f8..098243f 100644 --- a/webrtc/api/test/fakeaudiocapturemodule.h +++ b/webrtc/api/test/fakeaudiocapturemodule.h
@@ -174,12 +174,12 @@ int32_t ResetAudioDevice() override; int32_t SetLoudspeakerStatus(bool enable) override; int32_t GetLoudspeakerStatus(bool* enabled) const override; - virtual bool BuiltInAECIsAvailable() const { return false; } - virtual int32_t EnableBuiltInAEC(bool enable) { return -1; } - virtual bool BuiltInAGCIsAvailable() const { return false; } - virtual int32_t EnableBuiltInAGC(bool enable) { return -1; } - virtual bool BuiltInNSIsAvailable() const { return false; } - virtual int32_t EnableBuiltInNS(bool enable) { return -1; } + bool BuiltInAECIsAvailable() const override { return false; } + int32_t EnableBuiltInAEC(bool enable) override { return -1; } + bool BuiltInAGCIsAvailable() const override { return false; } + int32_t EnableBuiltInAGC(bool enable) override { return -1; } + bool BuiltInNSIsAvailable() const override { return false; } + int32_t EnableBuiltInNS(bool enable) override { return -1; } // End of functions inherited from webrtc::AudioDeviceModule. // The following function is inherited from rtc::MessageHandler.
diff --git a/webrtc/api/test/fakeaudiocapturemodule_unittest.cc b/webrtc/api/test/fakeaudiocapturemodule_unittest.cc index 8ac1acc..d0dcd85 100644 --- a/webrtc/api/test/fakeaudiocapturemodule_unittest.cc +++ b/webrtc/api/test/fakeaudiocapturemodule_unittest.cc
@@ -31,7 +31,7 @@ memset(rec_buffer_, 0, sizeof(rec_buffer_)); } - virtual void SetUp() { + void SetUp() override { fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); EXPECT_TRUE(fake_audio_capture_module_.get() != NULL); }
diff --git a/webrtc/api/test/fakedtlsidentitystore.h b/webrtc/api/test/fakedtlsidentitystore.h index 89c4084..c6f5a3c 100644 --- a/webrtc/api/test/fakedtlsidentitystore.h +++ b/webrtc/api/test/fakedtlsidentitystore.h
@@ -146,7 +146,7 @@ const char* get_cert() { return kKeysAndCerts[key_index_].cert_pem; } // rtc::MessageHandler implementation. - void OnMessage(rtc::Message* msg) { + void OnMessage(rtc::Message* msg) override { MessageData* message_data = static_cast<MessageData*>(msg->pdata); rtc::scoped_refptr<webrtc::DtlsIdentityRequestObserver> observer = message_data->data();
diff --git a/webrtc/api/videotrack.h b/webrtc/api/videotrack.h index 2f87532..60a0a64 100644 --- a/webrtc/api/videotrack.h +++ b/webrtc/api/videotrack.h
@@ -33,11 +33,11 @@ const rtc::VideoSinkWants& wants) override; void RemoveSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override; - virtual VideoTrackSourceInterface* GetSource() const { + VideoTrackSourceInterface* GetSource() const override { return video_source_.get(); } - virtual bool set_enabled(bool enable); - virtual std::string kind() const; + bool set_enabled(bool enable) override; + std::string kind() const override; protected: VideoTrack(const std::string& id, VideoTrackSourceInterface* video_source);
diff --git a/webrtc/api/videotracksource.h b/webrtc/api/videotracksource.h index 7100612..10e24ab 100644 --- a/webrtc/api/videotracksource.h +++ b/webrtc/api/videotracksource.h
@@ -36,8 +36,8 @@ void Stop() override{}; void Restart() override{}; - virtual bool is_screencast() const { return false; } - virtual rtc::Optional<bool> needs_denoising() const { + bool is_screencast() const override { return false; } + rtc::Optional<bool> needs_denoising() const override { return rtc::Optional<bool>(); } bool GetStats(Stats* stats) override { return false; }
diff --git a/webrtc/api/webrtcsession.h b/webrtc/api/webrtcsession.h index 89b77bb..970f967 100644 --- a/webrtc/api/webrtcsession.h +++ b/webrtc/api/webrtcsession.h
@@ -267,10 +267,10 @@ const RtpParameters& parameters) override; // Implements DtmfProviderInterface. - virtual bool CanInsertDtmf(const std::string& track_id); - virtual bool InsertDtmf(const std::string& track_id, - int code, int duration); - virtual sigslot::signal0<>* GetOnDestroyedSignal(); + bool CanInsertDtmf(const std::string& track_id) override; + bool InsertDtmf(const std::string& track_id, + int code, int duration) override; + sigslot::signal0<>* GetOnDestroyedSignal() override; // Implements DataChannelProviderInterface. bool SendData(const cricket::SendDataParams& params,
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc index 24e830e..5e9b039 100644 --- a/webrtc/api/webrtcsession_unittest.cc +++ b/webrtc/api/webrtcsession_unittest.cc
@@ -191,7 +191,7 @@ // Some local candidates are removed. void OnIceCandidatesRemoved( - const std::vector<cricket::Candidate>& candidates) { + const std::vector<cricket::Candidate>& candidates) override { num_candidates_removed_ += candidates.size(); }
diff --git a/webrtc/base/callback_unittest.cc b/webrtc/base/callback_unittest.cc index db294cd..aba1e0c 100644 --- a/webrtc/base/callback_unittest.cc +++ b/webrtc/base/callback_unittest.cc
@@ -34,7 +34,7 @@ int AddRef() const override { return ++count_; } - int Release() const { + int Release() const override { return --count_; } int RefCount() const { return count_; }
diff --git a/webrtc/base/fakesslidentity.h b/webrtc/base/fakesslidentity.h index 9f98c4e..3b0df29 100644 --- a/webrtc/base/fakesslidentity.h +++ b/webrtc/base/fakesslidentity.h
@@ -37,13 +37,13 @@ certs_.push_back(FakeSSLCertificate(*it)); } } - virtual FakeSSLCertificate* GetReference() const { + FakeSSLCertificate* GetReference() const override { return new FakeSSLCertificate(*this); } - virtual std::string ToPEMString() const { + std::string ToPEMString() const override { return data_; } - virtual void ToDER(Buffer* der_buffer) const { + void ToDER(Buffer* der_buffer) const override { std::string der_string; VERIFY(SSLIdentity::PemToDer(kPemTypeCertificate, data_, &der_string)); der_buffer->SetData(der_string.c_str(), der_string.size()); @@ -57,19 +57,19 @@ void set_digest_algorithm(const std::string& algorithm) { digest_algorithm_ = algorithm; } - virtual bool GetSignatureDigestAlgorithm(std::string* algorithm) const { + bool GetSignatureDigestAlgorithm(std::string* algorithm) const override { *algorithm = digest_algorithm_; return true; } - virtual bool ComputeDigest(const std::string& algorithm, - unsigned char* digest, - size_t size, - size_t* length) const { + bool ComputeDigest(const std::string& algorithm, + unsigned char* digest, + size_t size, + size_t* length) const override { *length = rtc::ComputeDigest(algorithm, data_.c_str(), data_.size(), digest, size); return (*length != 0); } - virtual std::unique_ptr<SSLCertChain> GetChain() const { + std::unique_ptr<SSLCertChain> GetChain() const override { if (certs_.empty()) return nullptr; std::vector<SSLCertificate*> new_certs(certs_.size());
diff --git a/webrtc/base/rtccertificategenerator_unittest.cc b/webrtc/base/rtccertificategenerator_unittest.cc index 750839c..a6e88a1 100644 --- a/webrtc/base/rtccertificategenerator_unittest.cc +++ b/webrtc/base/rtccertificategenerator_unittest.cc
@@ -36,13 +36,13 @@ RTCCertificateGenerator* generator() const { return generator_.get(); } RTCCertificate* certificate() const { return certificate_.get(); } - void OnSuccess(const scoped_refptr<RTCCertificate>& certificate) { + void OnSuccess(const scoped_refptr<RTCCertificate>& certificate) override { RTC_CHECK(signaling_thread_->IsCurrent()); RTC_CHECK(certificate); certificate_ = certificate; generate_async_completed_ = true; } - void OnFailure() { + void OnFailure() override { RTC_CHECK(signaling_thread_->IsCurrent()); certificate_ = nullptr; generate_async_completed_ = true;
diff --git a/webrtc/base/sslstreamadapter_unittest.cc b/webrtc/base/sslstreamadapter_unittest.cc index ac9fef9..dc62ac0 100644 --- a/webrtc/base/sslstreamadapter_unittest.cc +++ b/webrtc/base/sslstreamadapter_unittest.cc
@@ -562,7 +562,7 @@ } // Test data transfer for TLS - virtual void TestTransfer(int size) { + void TestTransfer(int size) override { LOG(LS_INFO) << "Starting transfer test with " << size << " bytes"; // Create some dummy data to send. size_t received; @@ -591,7 +591,7 @@ recv_stream_.GetBuffer(), size)); } - void WriteData() { + void WriteData() override { size_t position, tosend, size; rtc::StreamResult rv; size_t sent; @@ -627,7 +627,7 @@ } }; - virtual void ReadData(rtc::StreamInterface *stream) { + void ReadData(rtc::StreamInterface *stream) override { char buffer[1600]; size_t bread; int err2; @@ -691,7 +691,7 @@ new SSLDummyStreamDTLS(this, "s2c", &server_buffer_, &client_buffer_); } - virtual void WriteData() { + void WriteData() override { unsigned char *packet = new unsigned char[1600]; while (sent_ < count_) { @@ -720,7 +720,7 @@ delete [] packet; } - virtual void ReadData(rtc::StreamInterface *stream) { + void ReadData(rtc::StreamInterface *stream) override { unsigned char buffer[2000]; size_t bread; int err2; @@ -756,7 +756,7 @@ } } - virtual void TestTransfer(int count) { + void TestTransfer(int count) override { count_ = count; WriteData();
diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi index 948cc8a..1409e0b 100644 --- a/webrtc/build/common.gypi +++ b/webrtc/build/common.gypi
@@ -309,6 +309,7 @@ 'cflags': [ '-Wimplicit-fallthrough', '-Wthread-safety', + '-Winconsistent-missing-override', ], }], ],
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc index 8412564..329c1f2 100644 --- a/webrtc/call/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc
@@ -395,7 +395,7 @@ EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_); } - virtual Action OnSendRtp(const uint8_t* packet, size_t length) { + Action OnSendRtp(const uint8_t* packet, size_t length) override { rtc::CritScope lock(&crit_); RTPHeader header; EXPECT_TRUE(parser_->Parse(packet, length, &header));
diff --git a/webrtc/common_video/i420_buffer_pool.cc b/webrtc/common_video/i420_buffer_pool.cc index c382e93..8896260 100644 --- a/webrtc/common_video/i420_buffer_pool.cc +++ b/webrtc/common_video/i420_buffer_pool.cc
@@ -30,7 +30,7 @@ const uint8_t* DataU() const override { return buffer_->DataU(); } const uint8_t* DataV() const override { return buffer_->DataV(); } - bool IsMutable() { return HasOneRef(); } + bool IsMutable() override { return HasOneRef(); } // Make the IsMutable() check here instead of in |buffer_|, because the pool // also has a reference to |buffer_|. uint8_t* MutableDataY() override {
diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h index 24c4106..1cddda8 100644 --- a/webrtc/media/base/fakemediaengine.h +++ b/webrtc/media/base/fakemediaengine.h
@@ -482,23 +482,23 @@ return sinks_; } int max_bps() const { return max_bps_; } - virtual bool SetSendParameters(const VideoSendParameters& params) { + bool SetSendParameters(const VideoSendParameters& params) override { return (SetSendCodecs(params.codecs) && SetSendRtpHeaderExtensions(params.extensions) && SetMaxSendBandwidth(params.max_bandwidth_bps)); } - virtual bool SetRecvParameters(const VideoRecvParameters& params) { + bool SetRecvParameters(const VideoRecvParameters& params) override { return (SetRecvCodecs(params.codecs) && SetRecvRtpHeaderExtensions(params.extensions)); } - virtual bool AddSendStream(const StreamParams& sp) { + bool AddSendStream(const StreamParams& sp) override { return RtpHelper<VideoMediaChannel>::AddSendStream(sp); } - virtual bool RemoveSendStream(uint32_t ssrc) { + bool RemoveSendStream(uint32_t ssrc) override { return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc); } - virtual bool GetSendCodec(VideoCodec* send_codec) { + bool GetSendCodec(VideoCodec* send_codec) override { if (send_codecs_.empty()) { return false; } @@ -516,9 +516,9 @@ return true; } - virtual bool SetSend(bool send) { return set_sending(send); } - virtual bool SetVideoSend(uint32_t ssrc, bool enable, - const VideoOptions* options) { + bool SetSend(bool send) override { return set_sending(send); } + bool SetVideoSend(uint32_t ssrc, bool enable, + const VideoOptions* options) override { if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, !enable)) { return false; } @@ -536,20 +536,20 @@ bool HasSource(uint32_t ssrc) const { return sources_.find(ssrc) != sources_.end(); } - virtual bool AddRecvStream(const StreamParams& sp) { + bool AddRecvStream(const StreamParams& sp) override { if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp)) return false; sinks_[sp.first_ssrc()] = NULL; return true; } - virtual bool RemoveRecvStream(uint32_t ssrc) { + bool RemoveRecvStream(uint32_t ssrc) override { if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc)) return false; sinks_.erase(ssrc); return true; } - virtual bool GetStats(VideoMediaInfo* info) { return false; } + bool GetStats(VideoMediaInfo* info) override { return false; } private: bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
diff --git a/webrtc/media/engine/fakewebrtcvideoengine.h b/webrtc/media/engine/fakewebrtcvideoengine.h index b445d21..f8b8cbb 100644 --- a/webrtc/media/engine/fakewebrtcvideoengine.h +++ b/webrtc/media/engine/fakewebrtcvideoengine.h
@@ -182,8 +182,8 @@ num_created_encoders_(0), encoders_have_internal_sources_(false) {} - virtual webrtc::VideoEncoder* CreateVideoEncoder( - webrtc::VideoCodecType type) { + webrtc::VideoEncoder* CreateVideoEncoder( + webrtc::VideoCodecType type) override { rtc::CritScope lock(&crit_); if (supported_codec_types_.count(type) == 0) { return NULL; @@ -203,7 +203,7 @@ return false; } - virtual void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) { + void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override { rtc::CritScope lock(&crit_); encoders_.erase( std::remove(encoders_.begin(), encoders_.end(), encoder), @@ -211,12 +211,12 @@ delete encoder; } - virtual const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs() - const { + const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs() + const override { return codecs_; } - virtual bool EncoderTypeHasInternalSource( + bool EncoderTypeHasInternalSource( webrtc::VideoCodecType type) const override { return encoders_have_internal_sources_; }
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h index 4aa6ea3..5343800 100644 --- a/webrtc/media/engine/fakewebrtcvoiceengine.h +++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -298,7 +298,7 @@ channels_[channel]->associate_send_channel = accociate_send_channel; return 0; } - webrtc::RtcEventLog* GetEventLog() { return nullptr; } + webrtc::RtcEventLog* GetEventLog() override { return nullptr; } // webrtc::VoECodec WEBRTC_STUB(NumOfCodecs, ()); @@ -449,11 +449,11 @@ WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec)); WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)); WEBRTC_STUB(EnableBuiltInAEC, (bool enable)); - virtual bool BuiltInAECIsAvailable() const { return false; } + bool BuiltInAECIsAvailable() const override { return false; } WEBRTC_STUB(EnableBuiltInAGC, (bool enable)); - virtual bool BuiltInAGCIsAvailable() const { return false; } + bool BuiltInAGCIsAvailable() const override { return false; } WEBRTC_STUB(EnableBuiltInNS, (bool enable)); - virtual bool BuiltInNSIsAvailable() const { return false; } + bool BuiltInNSIsAvailable() const override { return false; } // webrtc::VoENetwork WEBRTC_FUNC(RegisterExternalTransport, (int channel, @@ -661,17 +661,17 @@ int reportingThreshold, int penaltyDecay, int typeEventDelay)); - int EnableHighPassFilter(bool enable) { + int EnableHighPassFilter(bool enable) override { highpass_filter_enabled_ = enable; return 0; } - bool IsHighPassFilterEnabled() { + bool IsHighPassFilterEnabled() override { return highpass_filter_enabled_; } - bool IsStereoChannelSwappingEnabled() { + bool IsStereoChannelSwappingEnabled() override { return stereo_swapping_enabled_; } - void EnableStereoChannelSwapping(bool enable) { + void EnableStereoChannelSwapping(bool enable) override { stereo_swapping_enabled_ = enable; } int GetNetEqCapacity() const {
diff --git a/webrtc/media/engine/webrtcvideocapturer.h b/webrtc/media/engine/webrtcvideocapturer.h index b6b3938..1efa4ad 100644 --- a/webrtc/media/engine/webrtcvideocapturer.h +++ b/webrtc/media/engine/webrtcvideocapturer.h
@@ -61,14 +61,14 @@ protected: void OnSinkWantsChanged(const rtc::VideoSinkWants& wants) override; // Override virtual methods of the parent class VideoCapturer. - virtual bool GetPreferredFourccs(std::vector<uint32_t>* fourccs); + bool GetPreferredFourccs(std::vector<uint32_t>* fourccs) override; private: // Callback when a frame is captured by camera. - virtual void OnIncomingCapturedFrame(const int32_t id, - const webrtc::VideoFrame& frame); - virtual void OnCaptureDelayChanged(const int32_t id, - const int32_t delay); + void OnIncomingCapturedFrame(const int32_t id, + const webrtc::VideoFrame& frame) override; + void OnCaptureDelayChanged(const int32_t id, + const int32_t delay) override; // Used to signal captured frames on the same thread as invoked Start(). // With WebRTC's current VideoCapturer implementations, this will mean a
diff --git a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h index cfee353..938e39e 100644 --- a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h +++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
@@ -47,7 +47,7 @@ // Returns the next encoded packet. Returns NULL if the test duration was // exceeded. Ownership of the packet is handed over to the caller. // Inherited from PacketSource. - Packet* NextPacket(); + Packet* NextPacket() override; // Inherited from AudioPacketizationCallback. int32_t SendData(FrameType frame_type,
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc index 503acdd..dc6bbf6 100644 --- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -608,7 +608,7 @@ ~AcmIsacMtTestOldApi() {} - void SetUp() { + void SetUp() override { AudioCodingModuleTestOldApi::SetUp(); RegisterCodec(); // Must be called before the threads start below. @@ -642,7 +642,7 @@ ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); } - void InsertPacket() { + void InsertPacket() override { int num_calls = packet_cb_.num_calls(); // Store locally for thread safety. if (num_calls > last_packet_number_) { // Get the new payload out from the callback handler. @@ -661,7 +661,7 @@ &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_)); } - void InsertAudio() { + void InsertAudio() override { // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS // this call confuses the number of samples with the number of bytes, and // ends up copying only half of what it should. @@ -677,7 +677,7 @@ // This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but // here it is using the constants defined in this class (i.e., shorter test // run). - virtual bool TestDone() { + bool TestDone() override { if (packet_cb_.num_calls() > kNumPackets) { rtc::CritScope lock(&crit_sect_); if (pull_audio_count_ > kNumPullCalls) { @@ -728,7 +728,7 @@ clock_ = fake_clock_.get(); } - void SetUp() { + void SetUp() override { AudioCodingModuleTestOldApi::SetUp(); // Set up input audio source to read from specified file, loop after 5 // seconds, and deliver blocks of 10 ms. @@ -757,7 +757,7 @@ codec_registration_thread_.SetPriority(rtc::kRealtimePriority); } - void TearDown() { + void TearDown() override { AudioCodingModuleTestOldApi::TearDown(); receive_thread_.Stop(); codec_registration_thread_.Stop(); @@ -1737,7 +1737,7 @@ } // Inherited from test::AudioSink. - bool WriteArray(const int16_t* audio, size_t num_samples) { + bool WriteArray(const int16_t* audio, size_t num_samples) override { // Skip checking the first output frame, since it has a number of zeros // due to how NetEq is initialized. if (first_output_) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc index 32f36c5..276eb60 100644 --- a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc +++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -25,10 +25,10 @@ IsacSpeedTest(); void SetUp() override; void TearDown() override; - virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream, - size_t max_bytes, size_t* encoded_bytes); - virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes, - int16_t* out_data); + float EncodeABlock(int16_t* in_data, uint8_t* bit_stream, + size_t max_bytes, size_t* encoded_bytes) override; + float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes, + int16_t* out_data) override; ISACFIX_MainStruct *ISACFIX_main_inst_; };
diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h index 6fafc25..2ffb30b 100644 --- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h +++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
@@ -22,7 +22,11 @@ class MockAudioEncoder : public AudioEncoder { public: - ~MockAudioEncoder() override { Die(); } + // TODO(nisse): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. + ~MockAudioEncoder() /* override */ { Die(); } MOCK_METHOD0(Die, void()); MOCK_METHOD1(Mark, void(std::string desc)); MOCK_CONST_METHOD0(SampleRateHz, int());
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc index 4d1aa42..7165d29 100644 --- a/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc +++ b/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -23,10 +23,10 @@ OpusSpeedTest(); void SetUp() override; void TearDown() override; - virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream, - size_t max_bytes, size_t* encoded_bytes); - virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes, - int16_t* out_data); + float EncodeABlock(int16_t* in_data, uint8_t* bit_stream, + size_t max_bytes, size_t* encoded_bytes) override; + float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes, + int16_t* out_data) override; WebRtcOpusEncInst* opus_encoder_; WebRtcOpusDecInst* opus_decoder_; };
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc index 42f2c1e..77622bc 100644 --- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -783,11 +783,15 @@ class MockAudioDecoder : public AudioDecoder { public: - void Reset() override {} + // TODO(nisse): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. + void Reset() /* override */ {} MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t)); MOCK_METHOD5(DecodeInternal, int(const uint8_t*, size_t, int, int16_t*, SpeechType*)); - size_t Channels() const override { return kChannels; } + size_t Channels() const /* override */ { return kChannels; } } decoder_; const uint8_t kFirstPayloadValue = 1;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc index 770ebd5..1a77abc 100644 --- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -24,31 +24,36 @@ class MockAudioDecoder final : public AudioDecoder { public: + // TODO(nisse): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. static const int kPacketDuration = 960; // 48 kHz * 20 ms explicit MockAudioDecoder(size_t num_channels) : num_channels_(num_channels), fec_enabled_(false) { } - ~MockAudioDecoder() override { Die(); } + ~MockAudioDecoder() /* override */ { Die(); } MOCK_METHOD0(Die, void()); MOCK_METHOD0(Reset, void()); int PacketDuration(const uint8_t* encoded, - size_t encoded_len) const override { + size_t encoded_len) const /* override */ { return kPacketDuration; } int PacketDurationRedundant(const uint8_t* encoded, - size_t encoded_len) const override { + size_t encoded_len) const /* override */ { return kPacketDuration; } - bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override { + bool PacketHasFec( + const uint8_t* encoded, size_t encoded_len) const /* override */ { return fec_enabled_; } - size_t Channels() const override { return num_channels_; } + size_t Channels() const /* override */ { return num_channels_; } void set_fec_enabled(bool enable_fec) { fec_enabled_ = enable_fec; } @@ -60,7 +65,7 @@ size_t encoded_len, int /*sample_rate_hz*/, int16_t* decoded, - SpeechType* speech_type) override { + SpeechType* speech_type) /* override */ { *speech_type = kSpeech; memset(decoded, 0, sizeof(int16_t) * kPacketDuration * Channels()); return kPacketDuration * Channels(); @@ -70,7 +75,7 @@ size_t encoded_len, int sample_rate_hz, int16_t* decoded, - SpeechType* speech_type) override { + SpeechType* speech_type) /* override */ { return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, speech_type); } @@ -294,7 +299,3 @@ } // namespace test } // namespace webrtc - - - -
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc index 2ebd192..62bfc1b 100644 --- a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc +++ b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -43,8 +43,8 @@ NetEqIsacQualityTest(); void SetUp() override; void TearDown() override; - virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples, - rtc::Buffer* payload, size_t max_bytes); + int EncodeBlock(int16_t* in_data, size_t block_size_samples, + rtc::Buffer* payload, size_t max_bytes) override; private: ISACFIX_MainStruct* isac_encoder_; int bit_rate_kbps_;
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc index baa0d67..a6117a4 100644 --- a/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc +++ b/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -103,8 +103,8 @@ NetEqOpusQualityTest(); void SetUp() override; void TearDown() override; - virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples, - rtc::Buffer* payload, size_t max_bytes); + int EncodeBlock(int16_t* in_data, size_t block_size_samples, + rtc::Buffer* payload, size_t max_bytes) override; private: WebRtcOpusEncInst* opus_encoder_; OpusRepacketizer* repacketizer_;
diff --git a/webrtc/modules/congestion_controller/congestion_controller.cc b/webrtc/modules/congestion_controller/congestion_controller.cc index 6985e67..14a73fe 100644 --- a/webrtc/modules/congestion_controller/congestion_controller.cc +++ b/webrtc/modules/congestion_controller/congestion_controller.cc
@@ -81,7 +81,7 @@ return rbe_->LatestEstimate(ssrcs, bitrate_bps); } - void SetMinBitrate(int min_bitrate_bps) { + void SetMinBitrate(int min_bitrate_bps) override { CriticalSectionScoped cs(crit_sect_.get()); rbe_->SetMinBitrate(min_bitrate_bps); min_bitrate_bps_ = min_bitrate_bps;
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc index 898ec02..f6cbe74 100644 --- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -157,8 +157,8 @@ : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {} virtual ~TestCallback() {} - virtual void StatisticsUpdated(const RtcpStatistics& statistics, - uint32_t ssrc) { + void StatisticsUpdated(const RtcpStatistics& statistics, + uint32_t ssrc) override { ssrc_ = ssrc; stats_ = statistics; ++num_calls_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h index 1511afb..0630adb 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
@@ -50,7 +50,7 @@ static const size_t kRrBaseLength = 4; static const size_t kMaxNumberOfReportBlocks = 0x1F; - size_t BlockLength() const { + size_t BlockLength() const override { return kHeaderLength + kRrBaseLength + report_blocks_.size() * ReportBlock::kLength; }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc index 091d271..283c284 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -406,7 +406,8 @@ return payload; } - int GetPayloadTypeFrequency(const RtpUtility::Payload& payload) const { + int GetPayloadTypeFrequency( + const RtpUtility::Payload& payload) const override { return payload.typeSpecific.Audio.frequency; } }; @@ -456,7 +457,8 @@ return payload; } - int GetPayloadTypeFrequency(const RtpUtility::Payload& payload) const { + int GetPayloadTypeFrequency( + const RtpUtility::Payload& payload) const override { return kVideoPayloadTypeFrequency; } };
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h index bec1578..d5d89ba 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -31,15 +31,15 @@ // The following three methods implement the TelephoneEventHandler interface. // Forward DTMFs to decoder for playout. - void SetTelephoneEventForwardToDecoder(bool forward_to_decoder); + void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) override; // Is forwarding of outband telephone events turned on/off? - bool TelephoneEventForwardToDecoder() const; + bool TelephoneEventForwardToDecoder() const override; // Is TelephoneEvent configured with payload type payload_type - bool TelephoneEventPayloadType(const int8_t payload_type) const; + bool TelephoneEventPayloadType(const int8_t payload_type) const override; - TelephoneEventHandler* GetTelephoneEventHandler() { return this; } + TelephoneEventHandler* GetTelephoneEventHandler() override { return this; } // Returns true if CNG is configured with payload type payload_type. If so, // the frequency and cng_payload_type_has_changed are filled in.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h index dc89b8f..486eced 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -34,7 +34,7 @@ int64_t timestamp, bool is_first_packet) override; - TelephoneEventHandler* GetTelephoneEventHandler() { return NULL; } + TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; } int GetPayloadTypeFrequency() const override;
diff --git a/webrtc/modules/utility/include/mock/mock_process_thread.h b/webrtc/modules/utility/include/mock/mock_process_thread.h index 3d39307..621fcee 100644 --- a/webrtc/modules/utility/include/mock/mock_process_thread.h +++ b/webrtc/modules/utility/include/mock/mock_process_thread.h
@@ -21,6 +21,10 @@ class MockProcessThread : public ProcessThread { public: + // TODO(nisse): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. MOCK_METHOD0(Start, void()); MOCK_METHOD0(Stop, void()); MOCK_METHOD1(WakeUp, void(Module* module)); @@ -31,7 +35,7 @@ // 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(std::unique_ptr<ProcessTask> task) override { + void PostTask(std::unique_ptr<ProcessTask> task) /* override */ { PostTask(task.get()); } };
diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h index 697d759..44169cc 100644 --- a/webrtc/modules/utility/source/file_recorder_impl.h +++ b/webrtc/modules/utility/source/file_recorder_impl.h
@@ -45,23 +45,23 @@ virtual ~FileRecorderImpl(); // FileRecorder functions. - virtual int32_t RegisterModuleFileCallback(FileCallback* callback); - virtual FileFormats RecordingFileFormat() const; - virtual int32_t StartRecordingAudioFile( + int32_t RegisterModuleFileCallback(FileCallback* callback) override; + FileFormats RecordingFileFormat() const override; + int32_t StartRecordingAudioFile( const char* fileName, const CodecInst& codecInst, uint32_t notificationTimeMs) override; - virtual int32_t StartRecordingAudioFile( + int32_t StartRecordingAudioFile( OutStream& destStream, const CodecInst& codecInst, uint32_t notificationTimeMs) override; - virtual int32_t StopRecording(); - virtual bool IsRecording() const; - virtual int32_t codec_info(CodecInst& codecInst) const; - virtual int32_t RecordAudioToFile( + int32_t StopRecording() override; + bool IsRecording() const override; + int32_t codec_info(CodecInst& codecInst) const override; + int32_t RecordAudioToFile( const AudioFrame& frame, - const TickTime* playoutTS = NULL); - virtual int32_t StartRecordingVideoFile( + const TickTime* playoutTS = NULL) override; + int32_t StartRecordingVideoFile( const char* fileName, const CodecInst& audioCodecInst, const VideoCodec& videoCodecInst, @@ -69,7 +69,7 @@ { return -1; } - virtual int32_t RecordVideoToFile(const VideoFrame& videoFrame) { + int32_t RecordVideoToFile(const VideoFrame& videoFrame) override { return -1; }
diff --git a/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc b/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc index d226013..b9721cd 100644 --- a/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc +++ b/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
@@ -101,10 +101,10 @@ virtual ~RealTimeTemporalLayers() {} - virtual bool ConfigureBitrates(int bitrate_kbit, - int max_bitrate_kbit, - int framerate, - vpx_codec_enc_cfg_t* cfg) { + bool ConfigureBitrates(int bitrate_kbit, + int max_bitrate_kbit, + int framerate, + vpx_codec_enc_cfg_t* cfg) override { temporal_layers_ = CalculateNumberOfTemporalLayers(temporal_layers_, framerate); temporal_layers_ = std::min(temporal_layers_, max_temporal_layers_); @@ -184,7 +184,7 @@ return true; } - virtual int EncodeFlags(uint32_t timestamp) { + int EncodeFlags(uint32_t timestamp) override { frame_counter_++; return CurrentEncodeFlags(); } @@ -196,16 +196,16 @@ return encode_flags_[index]; } - virtual int CurrentLayerId() const { + int CurrentLayerId() const override { assert(layer_ids_length_ > 0 && layer_ids_ != NULL); int index = frame_counter_ % layer_ids_length_; assert(index >= 0 && index < layer_ids_length_); return layer_ids_[index]; } - virtual void PopulateCodecSpecific(bool base_layer_sync, - CodecSpecificInfoVP8* vp8_info, - uint32_t timestamp) { + void PopulateCodecSpecific(bool base_layer_sync, + CodecSpecificInfoVP8* vp8_info, + uint32_t timestamp) override { assert(temporal_layers_ > 0); if (temporal_layers_ == 1) {
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc index 9a7e1b2..aafcd79 100644 --- a/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc +++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter_unittest.cc
@@ -107,34 +107,40 @@ class MockVideoEncoder : public VideoEncoder { public: + // TODO(nisse): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. int32_t InitEncode(const VideoCodec* codecSettings, int32_t numberOfCores, - size_t maxPayloadSize) override { + size_t maxPayloadSize) /* override */ { codec_ = *codecSettings; return 0; } int32_t Encode(const VideoFrame& inputImage, const CodecSpecificInfo* codecSpecificInfo, - const std::vector<FrameType>* frame_types) override { + const std::vector<FrameType>* frame_types) /* override */ { return 0; } int32_t RegisterEncodeCompleteCallback( - EncodedImageCallback* callback) override { + EncodedImageCallback* callback) /* override */ { callback_ = callback; return 0; } - int32_t Release() override { return 0; } + int32_t Release() /* override */ { return 0; } - int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) override { + int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) /* override */ { return 0; } MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt)); - bool SupportsNativeHandle() const override { return supports_native_handle_; } + bool SupportsNativeHandle() const /* override */ { + return supports_native_handle_; + } virtual ~MockVideoEncoder() {}
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h index 2b2aa5d..8d5b74f 100644 --- a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h +++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
@@ -168,7 +168,7 @@ virtual ~SpyingTemporalLayers() { delete layers_; } - virtual int EncodeFlags(uint32_t timestamp) { + int EncodeFlags(uint32_t timestamp) override { return layers_->EncodeFlags(timestamp); }
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h index 6906a32..f8af642 100644 --- a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h +++ b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
@@ -40,21 +40,21 @@ virtual ~VP8EncoderImpl(); - virtual int Release(); + int Release() override; - virtual int InitEncode(const VideoCodec* codec_settings, - int number_of_cores, - size_t max_payload_size); + int InitEncode(const VideoCodec* codec_settings, + int number_of_cores, + size_t max_payload_size) override; - virtual int Encode(const VideoFrame& input_image, - const CodecSpecificInfo* codec_specific_info, - const std::vector<FrameType>* frame_types); + int Encode(const VideoFrame& input_image, + const CodecSpecificInfo* codec_specific_info, + const std::vector<FrameType>* frame_types) override; - virtual int RegisterEncodeCompleteCallback(EncodedImageCallback* callback); + int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override; - virtual int SetChannelParameters(uint32_t packet_loss, int64_t rtt); + int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override; - virtual int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate); + int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate) override; void OnDroppedFrame() override {}
diff --git a/webrtc/modules/video_coding/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/jitter_buffer_unittest.cc index eb7d78b..af9c20a 100644 --- a/webrtc/modules/video_coding/jitter_buffer_unittest.cc +++ b/webrtc/modules/video_coding/jitter_buffer_unittest.cc
@@ -215,7 +215,7 @@ protected: TestBasicJitterBuffer() : scoped_field_trial_(GetParam()) {} - virtual void SetUp() { + void SetUp() override { clock_.reset(new SimulatedClock(0)); jitter_buffer_.reset(new VCMJitterBuffer( clock_.get(),
diff --git a/webrtc/p2p/base/faketransportcontroller.h b/webrtc/p2p/base/faketransportcontroller.h index c099c8c..e2bdc10 100644 --- a/webrtc/p2p/base/faketransportcontroller.h +++ b/webrtc/p2p/base/faketransportcontroller.h
@@ -231,7 +231,7 @@ } bool SetLocalCertificate( - const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) { + const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override { local_cert_ = certificate; return true; } @@ -257,7 +257,7 @@ bool GetSslCipherSuite(int* cipher_suite) override { return false; } - rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const { + rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const override { return local_cert_; }
diff --git a/webrtc/p2p/base/port_unittest.cc b/webrtc/p2p/base/port_unittest.cc index 7e787e0..7231d59 100644 --- a/webrtc/p2p/base/port_unittest.cc +++ b/webrtc/p2p/base/port_unittest.cc
@@ -965,7 +965,7 @@ void set_next_client_tcp_socket(AsyncPacketSocket* next_client_tcp_socket) { next_client_tcp_socket_ = next_client_tcp_socket; } - rtc::AsyncResolverInterface* CreateAsyncResolver() { + rtc::AsyncResolverInterface* CreateAsyncResolver() override { return NULL; }
diff --git a/webrtc/test/mock_voice_engine.h b/webrtc/test/mock_voice_engine.h index fac088b..b9eb05f 100644 --- a/webrtc/test/mock_voice_engine.h +++ b/webrtc/test/mock_voice_engine.h
@@ -24,6 +24,10 @@ // able to get the various interfaces as usual, via T::GetInterface(). class MockVoiceEngine : public VoiceEngineImpl { public: + // TODO(nisse): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. MockVoiceEngine() : VoiceEngineImpl(new Config(), true) { // Increase ref count so this object isn't automatically deleted whenever // interfaces are Release():d. @@ -36,7 +40,7 @@ return new testing::NiceMock<MockVoEChannelProxy>(); })); } - ~MockVoiceEngine() override { + ~MockVoiceEngine() /* override */ { // Decrease ref count before base class d-tor is called; otherwise it will // trigger an assertion. --_ref_count; @@ -45,7 +49,8 @@ MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy*(int channel_id)); // VoiceEngineImpl - std::unique_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id) override { + std::unique_ptr<voe::ChannelProxy> GetChannelProxy( + int channel_id) /* override */ { return std::unique_ptr<voe::ChannelProxy>(ChannelProxyFactory(channel_id)); }
diff --git a/webrtc/test/rtp_file_reader.cc b/webrtc/test/rtp_file_reader.cc index 3687ef7..476767a 100644 --- a/webrtc/test/rtp_file_reader.cc +++ b/webrtc/test/rtp_file_reader.cc
@@ -131,7 +131,7 @@ } bool Init(const std::string& filename, - const std::set<uint32_t>& ssrc_filter) { + const std::set<uint32_t>& ssrc_filter) override { file_ = fopen(filename.c_str(), "rb"); if (file_ == NULL) { printf("ERROR: Can't open file: %s\n", filename.c_str());
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc index ef1771a..0e02ff8 100644 --- a/webrtc/video/end_to_end_tests.cc +++ b/webrtc/video/end_to_end_tests.cc
@@ -1765,7 +1765,7 @@ ~BweObserver() {} - test::PacketTransport* CreateReceiveTransport() { + test::PacketTransport* CreateReceiveTransport() override { receive_transport_ = new test::PacketTransport( nullptr, this, test::PacketTransport::kReceiver, FakeNetworkPipe::Config()); @@ -2248,7 +2248,7 @@ return SEND_PACKET; } // Send stream should send SR packets (and DLRR packets if enabled). - virtual Action OnSendRtcp(const uint8_t* packet, size_t length) { + Action OnSendRtcp(const uint8_t* packet, size_t length) override { rtc::CritScope lock(&crit_); RTCPUtility::RTCPParserV2 parser(packet, length, true); EXPECT_TRUE(parser.IsValid());
diff --git a/webrtc/video/overuse_frame_detector_unittest.cc b/webrtc/video/overuse_frame_detector_unittest.cc index 06cff38..67d0532 100644 --- a/webrtc/video/overuse_frame_detector_unittest.cc +++ b/webrtc/video/overuse_frame_detector_unittest.cc
@@ -53,7 +53,7 @@ class OveruseFrameDetectorTest : public ::testing::Test, public CpuOveruseMetricsObserver { protected: - virtual void SetUp() { + void SetUp() override { clock_.reset(new SimulatedClock(1234)); observer_.reset(new MockCpuOveruseObserver()); options_.min_process_count = 0;
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc index 5df2425..53cb72d 100644 --- a/webrtc/video/video_send_stream_tests.cc +++ b/webrtc/video/video_send_stream_tests.cc
@@ -708,7 +708,7 @@ } } - virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) { + void EncodedFrameCallback(const EncodedFrame& encoded_frame) override { // Increase frame size for next encoded frame, in the context of the // encoder thread. if (!use_fec_ && @@ -999,8 +999,8 @@ size_t GetNumVideoStreams() const override { return 3; } - virtual void OnFrameGeneratorCapturerCreated( - test::FrameGeneratorCapturer* frame_generator_capturer) { + void OnFrameGeneratorCapturerCreated( + test::FrameGeneratorCapturer* frame_generator_capturer) override { rtc::CritScope lock(&crit_); capturer_ = frame_generator_capturer; } @@ -1040,7 +1040,7 @@ } private: - virtual Action OnSendRtp(const uint8_t* packet, size_t length) { + Action OnSendRtp(const uint8_t* packet, size_t length) override { if (RtpHeaderParser::IsRtcp(packet, length)) return DROP_PACKET;
diff --git a/webrtc/video/vie_channel.h b/webrtc/video/vie_channel.h index 0411857..92adc4e 100644 --- a/webrtc/video/vie_channel.h +++ b/webrtc/video/vie_channel.h
@@ -81,11 +81,10 @@ CallStatsObserver* GetStatsObserver(); // Implements VCMReceiveCallback. - virtual int32_t FrameToRender(VideoFrame& video_frame); // NOLINT + int32_t FrameToRender(VideoFrame& video_frame) override; // NOLINT // Implements VCMReceiveCallback. - virtual int32_t ReceivedDecodedReferenceFrame( - const uint64_t picture_id); + int32_t ReceivedDecodedReferenceFrame(const uint64_t picture_id) override; // Implements VCMReceiveCallback. void OnIncomingPayloadType(int payload_type) override; @@ -97,20 +96,20 @@ void OnFrameCountsUpdated(const FrameCounts& frame_counts) override; // Implements VCMDecoderTimingCallback. - virtual void OnDecoderTiming(int decode_ms, - int max_decode_ms, - int current_delay_ms, - int target_delay_ms, - int jitter_buffer_ms, - int min_playout_delay_ms, - int render_delay_ms); + void OnDecoderTiming(int decode_ms, + int max_decode_ms, + int current_delay_ms, + int target_delay_ms, + int jitter_buffer_ms, + int min_playout_delay_ms, + int render_delay_ms) override; // Implements FrameTypeCallback. - virtual int32_t RequestKeyFrame(); + int32_t RequestKeyFrame() override; // Implements FrameTypeCallback. - virtual int32_t SliceLossIndicationRequest( - const uint64_t picture_id); + int32_t SliceLossIndicationRequest( + const uint64_t picture_id) override; // Implements VideoPacketRequestCallback. int32_t ResendPackets(const uint16_t* sequence_numbers,