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,