Diff
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-01-20 22:30:24 UTC (rev 210987)
@@ -1,3 +1,81 @@
+2017-01-20 Youenn Fablet <[email protected]>
+
+ [WebRTC] libwebrtc headers are incompatible with WebKit compilation flags
+ https://bugs.webkit.org/show_bug.cgi?id=167242
+
+ Reviewed by Alex Christensen.
+
+ WebKit is enforcing -Wunused-parameter and -Wunused-variable which conflict with some included libwertc headers.
+ Removed unused parameter names for inlined functions.
+
+ * Source/webrtc/api/jsep.h:
+ (webrtc::SessionDescriptionInterface::RemoveCandidates):
+ * Source/webrtc/api/mediastreaminterface.h:
+ (webrtc::AudioSourceInterface::SetVolume):
+ (webrtc::AudioSourceInterface::RegisterAudioObserver):
+ (webrtc::AudioSourceInterface::UnregisterAudioObserver):
+ (webrtc::AudioSourceInterface::AddSink):
+ (webrtc::AudioSourceInterface::RemoveSink):
+ (webrtc::AudioTrackInterface::GetSignalLevel):
+ * Source/webrtc/api/peerconnectionfactory.h:
+ * Source/webrtc/api/peerconnectioninterface.h:
+ (webrtc::MetricsObserverInterface::IncrementEnumCounter):
+ (webrtc::PeerConnectionInterface::AddTrack):
+ (webrtc::PeerConnectionInterface::RemoveTrack):
+ (webrtc::PeerConnectionInterface::CreateSender):
+ (webrtc::PeerConnectionInterface::GetStats):
+ (webrtc::PeerConnectionInterface::CreateOffer):
+ (webrtc::PeerConnectionInterface::CreateAnswer):
+ (webrtc::PeerConnectionInterface::UpdateIce):
+ (webrtc::PeerConnectionInterface::SetConfiguration):
+ (webrtc::PeerConnectionInterface::RemoveIceCandidates):
+ (webrtc::PeerConnectionInterface::StartRtcEventLog):
+ (webrtc::PeerConnectionObserver::OnAddStream):
+ (webrtc::PeerConnectionObserver::OnRemoveStream):
+ (webrtc::PeerConnectionObserver::OnDataChannel):
+ (webrtc::PeerConnectionObserver::OnIceCandidatesRemoved):
+ (webrtc::PeerConnectionObserver::OnIceConnectionReceivingChange):
+ * Source/webrtc/api/rtpsender.cc:
+ * Source/webrtc/base/messagehandler.h:
+ (rtc::FunctorMessageHandler::OnMessage):
+ * Source/webrtc/base/sanitizer.h:
+ (rtc_AsanPoison):
+ (rtc_AsanUnpoison):
+ (rtc_MsanMarkUninitialized):
+ (rtc_MsanCheckInitialized):
+ * Source/webrtc/base/stream.h:
+ (rtc::StreamInterface::ConsumeReadData):
+ (rtc::StreamInterface::ConsumeWriteBuffer):
+ * Source/webrtc/media/base/mediachannel.h:
+ (cricket::DataMediaChannel::GetStats):
+ (cricket::DataMediaChannel::OnNetworkRouteChanged):
+ * Source/webrtc/media/engine/webrtcvideodecoderfactory.h:
+ (cricket::WebRtcVideoDecoderFactory::CreateVideoDecoderWithParams):
+ * Source/webrtc/media/engine/webrtcvideoencoderfactory.h:
+ (cricket::WebRtcVideoEncoderFactory::VideoCodec::VideoCodec):
+ (cricket::WebRtcVideoEncoderFactory::EncoderTypeHasInternalSource):
+ * Source/webrtc/media/engine/webrtcvideoengine2.cc:
+ * Source/webrtc/modules/include/module.h:
+ (webrtc::Module::ProcessThreadAttached):
+ * Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc:
+ * Source/webrtc/p2p/base/port.h:
+ (cricket::Port::HandleIncomingPacket):
+ (cricket::Port::HandleConnectionDestroyed):
+ (cricket::Connection::set_receiving_timeout):
+ * Source/webrtc/p2p/base/stun.h:
+ (cricket::StunAttribute::SetOwner):
+ * Source/webrtc/p2p/base/stunrequest.h:
+ (cricket::StunRequest::Prepare):
+ (cricket::StunRequest::OnResponse):
+ (cricket::StunRequest::OnErrorResponse):
+ * Source/webrtc/p2p/base/transport.h:
+ (cricket::Transport::SetLocalCertificate):
+ (cricket::Transport::GetLocalCertificate):
+ (cricket::Transport::GetSslRole):
+ (cricket::Transport::SetSslMaxProtocolVersion):
+ * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc:
+ * Source/webrtc/typedefs.h:
+
2017-01-20 Youenn Fablet <[email protected]>
[WebRTC] Update libwertc AudioRtpSender::SetAudioSend
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -98,7 +98,7 @@
// Removes the candidates from the description.
// Returns the number of candidates removed.
virtual size_t RemoveCandidates(
- const std::vector<cricket::Candidate>& candidates) { return 0; }
+ const std::vector<cricket::Candidate>&) { return 0; }
// Returns the number of m- lines in the session description.
virtual size_t number_of_mediasections() const = 0;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -134,9 +134,9 @@
public rtc::VideoSourceInterface<VideoFrame> {
public:
// Register a video sink for this track.
- void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
- const rtc::VideoSinkWants& wants) override{};
- void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override{};
+ void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>*,
+ const rtc::VideoSinkWants&) override {};
+ void RemoveSink(rtc::VideoSinkInterface<VideoFrame>*) override {};
virtual VideoTrackSourceInterface* GetSource() const = 0;
@@ -174,15 +174,15 @@
// Sets the volume to the source. |volume| is in the range of [0, 10].
// TODO(tommi): This method should be on the track and ideally volume should
// be applied in the track in a way that does not affect clones of the track.
- virtual void SetVolume(double volume) {}
+ virtual void SetVolume(double) {}
// Registers/unregisters observer to the audio source.
- virtual void RegisterAudioObserver(AudioObserver* observer) {}
- virtual void UnregisterAudioObserver(AudioObserver* observer) {}
+ virtual void RegisterAudioObserver(AudioObserver*) {}
+ virtual void UnregisterAudioObserver(AudioObserver*) {}
// TODO(tommi): Make pure virtual.
- virtual void AddSink(AudioTrackSinkInterface* sink) {}
- virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
+ virtual void AddSink(AudioTrackSinkInterface*) {}
+ virtual void RemoveSink(AudioTrackSinkInterface*) {}
};
// Interface of the audio processor used by the audio track to collect
@@ -230,7 +230,7 @@
// Return true on success, otherwise false.
// TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
// after Chrome has the correct implementation of the interface.
- virtual bool GetSignalLevel(int* level) { return false; }
+ virtual bool GetSignalLevel(int*) { return false; }
// Get the audio processor used by the audio track. Return NULL if the track
// does not have any processor.
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -81,10 +81,10 @@
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
void StopAecDump() override;
// TODO(ivoc) Remove after Chrome is updated.
- bool StartRtcEventLog(rtc::PlatformFile file) override { return false; }
+ bool StartRtcEventLog(rtc::PlatformFile) override { return false; }
// TODO(ivoc) Remove after Chrome is updated.
- bool StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) override {
+ bool StartRtcEventLog(rtc::PlatformFile,
+ int64_t) override {
return false;
}
// TODO(ivoc) Remove after Chrome is updated.
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -119,9 +119,9 @@
// |type| is the type of the enum counter to be incremented. |counter|
// is the particular counter in that type. |counter_max| is the next sequence
// number after the highest counter.
- virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
- int counter,
- int counter_max) {}
+ virtual void IncrementEnumCounter(PeerConnectionEnumCounterType,
+ int /* counter */,
+ int /* counter_max */) {}
// This is used to handle sparse counters like SSL cipher suites.
// TODO(guoweis): Remove the implementation once the dependency's interface
@@ -389,14 +389,14 @@
// |streams| indicates which stream labels the track should be associated
// with.
virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
- MediaStreamTrackInterface* track,
- std::vector<MediaStreamInterface*> streams) {
+ MediaStreamTrackInterface*,
+ std::vector<MediaStreamInterface*>) {
return nullptr;
}
// Remove an RtpSender from this PeerConnection.
// Returns true on success.
- virtual bool RemoveTrack(RtpSenderInterface* sender) {
+ virtual bool RemoveTrack(RtpSenderInterface*) {
return false;
}
@@ -410,8 +410,8 @@
// |stream_id| is used to populate the msid attribute; if empty, one will
// be generated automatically.
virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
- const std::string& kind,
- const std::string& stream_id) {
+ const std::string& /* kind */,
+ const std::string& /* stream_id */) {
return rtc::scoped_refptr<RtpSenderInterface>();
}
@@ -433,7 +433,7 @@
// TODO(hbos): Default implementation that does nothing only exists as to not
// break third party projects. As soon as they have been updated this should
// be changed to "= 0;".
- virtual void GetStats(RTCStatsCollectorCallback* callback) {}
+ virtual void GetStats(RTCStatsCollectorCallback*) {}
virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
@@ -444,23 +444,23 @@
// Create a new offer.
// The CreateSessionDescriptionObserver callback will be called when done.
- virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) {}
+ virtual void CreateOffer(CreateSessionDescriptionObserver*,
+ const MediaConstraintsInterface*) {}
// TODO(jiayl): remove the default impl and the old interface when chromium
// code is updated.
- virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
- const RTCOfferAnswerOptions& options) {}
+ virtual void CreateOffer(CreateSessionDescriptionObserver*,
+ const RTCOfferAnswerOptions&) {}
// Create an answer to an offer.
// The CreateSessionDescriptionObserver callback will be called when done.
- virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
- const RTCOfferAnswerOptions& options) {}
+ virtual void CreateAnswer(CreateSessionDescriptionObserver*,
+ const RTCOfferAnswerOptions&) {}
// Deprecated - use version above.
// TODO(hta): Remove and remove default implementations when all callers
// are updated.
- virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) {}
+ virtual void CreateAnswer(CreateSessionDescriptionObserver*,
+ const MediaConstraintsInterface*) {}
// Sets the local session description.
// JsepInterface takes the ownership of |desc| even if it fails.
@@ -475,11 +475,11 @@
// Restarts or updates the ICE Agent process of gathering local candidates
// and pinging remote candidates.
// TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
- virtual bool UpdateIce(const IceServers& configuration,
- const MediaConstraintsInterface* constraints) {
+ virtual bool UpdateIce(const IceServers&,
+ const MediaConstraintsInterface*) {
return false;
}
- virtual bool UpdateIce(const IceServers& configuration) { return false; }
+ virtual bool UpdateIce(const IceServers&) { return false; }
// Sets the PeerConnection's global configuration to |config|.
// Any changes to STUN/TURN servers or ICE candidate policy will affect the
// next gathering phase, and cause the next call to createOffer to generate
@@ -488,7 +488,7 @@
// TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
// PeerConnectionInterface implement it.
virtual bool SetConfiguration(
- const PeerConnectionInterface::RTCConfiguration& config) {
+ const PeerConnectionInterface::RTCConfiguration&) {
return false;
}
// Provides a remote candidate to the ICE Agent.
@@ -501,7 +501,7 @@
// Removes a group of remote candidates from the ICE agent.
virtual bool RemoveIceCandidates(
- const std::vector<cricket::Candidate>& candidates) {
+ const std::vector<cricket::Candidate>&) {
return false;
}
@@ -518,8 +518,8 @@
// automatically after 10 minutes have passed, or when the StopRtcEventLog
// function is called.
// TODO(ivoc): Make this pure virtual when Chrome is updated.
- virtual bool StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) {
+ virtual bool StartRtcEventLog(rtc::PlatformFile,
+ int64_t /* max_size_bytes */) {
return false;
}
@@ -553,21 +553,21 @@
// pointer version.
// Triggered when media is received on a new stream from remote peer.
- virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
+ virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface>) {}
// Deprecated; please use the version that uses a scoped_refptr.
- virtual void OnAddStream(MediaStreamInterface* stream) {}
+ virtual void OnAddStream(MediaStreamInterface*) {}
// Triggered when a remote peer close a stream.
- virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
+ virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface>) {
}
// Deprecated; please use the version that uses a scoped_refptr.
- virtual void OnRemoveStream(MediaStreamInterface* stream) {}
+ virtual void OnRemoveStream(MediaStreamInterface*) {}
// Triggered when a remote peer opens a data channel.
virtual void OnDataChannel(
- rtc::scoped_refptr<DataChannelInterface> data_channel){};
+ rtc::scoped_refptr<DataChannelInterface>){};
// Deprecated; please use the version that uses a scoped_refptr.
- virtual void OnDataChannel(DataChannelInterface* data_channel) {}
+ virtual void OnDataChannel(DataChannelInterface*) {}
// Triggered when renegotiation is needed. For example, an ICE restart
// has begun.
@@ -588,10 +588,10 @@
// TODO(honghaiz): Make this a pure virtual method when all its subclasses
// implement it.
virtual void OnIceCandidatesRemoved(
- const std::vector<cricket::Candidate>& candidates) {}
+ const std::vector<cricket::Candidate>&) {}
// Called when the ICE connection receiving status changes.
- virtual void OnIceConnectionReceivingChange(bool receiving) {}
+ virtual void OnIceConnectionReceivingChange(bool /* receiving */) {}
protected:
// Dtor protected as objects shouldn't be deleted via this interface.
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -40,7 +40,7 @@
public:
explicit FunctorMessageHandler(const FunctorT& functor)
: functor_(functor) {}
- virtual void OnMessage(Message* msg) {
+ virtual void OnMessage(Message*) {
result_ = functor_();
}
const ReturnT& result() const { return result_; }
@@ -56,7 +56,7 @@
: public MessageHandler {
public:
explicit FunctorMessageHandler(const FunctorT& functor) : functor_(functor) {}
- virtual void OnMessage(Message* msg) { result_ = std::move(functor_()); }
+ virtual void OnMessage(Message*) { result_ = std::move(functor_()); }
std::unique_ptr<ReturnT> result() { return std::move(result_); }
private:
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -42,6 +42,12 @@
#define RTC_NO_SANITIZE(what)
#endif
+#if !RTC_HAS_ASAN
+#define SANITIZER_UNUSED3(x, y, z) (void)&(x); \
+ (void)&(y); \
+ (void)&(z)
+#endif
+
// Ask ASan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being unaddressable, so that reads and writes are not allowed. ASan may
// narrow the range to the nearest alignment boundaries.
@@ -50,6 +56,8 @@
size_t num_elements) {
#if RTC_HAS_ASAN
ASAN_POISON_MEMORY_REGION(ptr, element_size * num_elements);
+#else
+ SANITIZER_UNUSED3(ptr, element_size, num_elements);
#endif
}
@@ -61,6 +69,8 @@
size_t num_elements) {
#if RTC_HAS_ASAN
ASAN_UNPOISON_MEMORY_REGION(ptr, element_size * num_elements);
+#else
+ SANITIZER_UNUSED3(ptr, element_size, num_elements);
#endif
}
@@ -71,6 +81,8 @@
size_t num_elements) {
#if RTC_HAS_MSAN
__msan_poison(ptr, element_size * num_elements);
+#else
+ SANITIZER_UNUSED3(ptr, element_size, num_elements);
#endif
}
@@ -82,6 +94,8 @@
size_t num_elements) {
#if RTC_HAS_MSAN
__msan_check_mem_is_initialized(ptr, element_size * num_elements);
+#else
+ SANITIZER_UNUSED3(ptr, element_size, num_elements);
#endif
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -132,7 +132,7 @@
// processed. Read and ConsumeReadData invalidate the buffer returned by
// GetReadData.
virtual const void* GetReadData(size_t* data_len);
- virtual void ConsumeReadData(size_t used) {}
+ virtual void ConsumeReadData(size_t) {}
// GetWriteBuffer returns a pointer to a buffer which is owned by the stream.
// The buffer has a capacity of buf_len bytes. NULL is returned if there is
@@ -146,7 +146,7 @@
// when it is available. If the requested amount is too large, return an
// error.
virtual void* GetWriteBuffer(size_t* buf_len);
- virtual void ConsumeWriteBuffer(size_t used) {}
+ virtual void ConsumeWriteBuffer(size_t) {}
// Write data_len bytes found in data, circumventing any throttling which
// would could cause SR_BLOCK to be returned. Returns true if all the data
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -1159,13 +1159,13 @@
virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
// TODO(pthatcher): Implement this.
- virtual bool GetStats(DataMediaInfo* info) { return true; }
+ virtual bool GetStats(DataMediaInfo*) { return true; }
virtual bool SetSend(bool send) = 0;
virtual bool SetReceive(bool receive) = 0;
- virtual void OnNetworkRouteChanged(const std::string& transport_name,
- const rtc::NetworkRoute& network_route) {}
+ virtual void OnNetworkRouteChanged(const std::string& /* transport_name */,
+ const rtc::NetworkRoute&) {}
virtual bool SendData(
const SendDataParams& params,
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -32,7 +32,7 @@
webrtc::VideoCodecType type) = 0;
virtual webrtc::VideoDecoder* CreateVideoDecoderWithParams(
webrtc::VideoCodecType type,
- VideoDecoderParams params) {
+ VideoDecoderParams) {
return CreateVideoDecoder(type);
}
virtual ~WebRtcVideoDecoderFactory() {}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -36,9 +36,9 @@
VideoCodec(webrtc::VideoCodecType t,
const std::string& nm,
- int w,
- int h,
- int fr)
+ int /* w */,
+ int /* h */,
+ int /* fr */)
: type(t), name(nm) {}
};
@@ -70,7 +70,7 @@
// frames to be delivered via webrtc::VideoEncoder::Encode. This flag is used
// as the internal_source parameter to
// webrtc::ViEExternalCodec::RegisterExternalSendCodec.
- virtual bool EncoderTypeHasInternalSource(webrtc::VideoCodecType type) const {
+ virtual bool EncoderTypeHasInternalSource(webrtc::VideoCodecType) const {
return false;
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -53,7 +53,7 @@
//
// NOTE: This method is not called from the worker thread itself, but from
// the thread that registers/deregisters the module or calls Start/Stop.
- virtual void ProcessThreadAttached(ProcessThread* process_thread) {}
+ virtual void ProcessThreadAttached(ProcessThread*) {}
protected:
virtual ~Module() {}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -241,9 +241,9 @@
// port implemented this method.
// TODO(mallinath) - Make it pure virtual.
virtual bool HandleIncomingPacket(
- rtc::AsyncPacketSocket* socket, const char* data, size_t size,
- const rtc::SocketAddress& remote_addr,
- const rtc::PacketTime& packet_time) {
+ rtc::AsyncPacketSocket*, const char*, size_t,
+ const rtc::SocketAddress&,
+ const rtc::PacketTime&) {
ASSERT(false);
return false;
}
@@ -360,7 +360,7 @@
}
// Extra work to be done in subclasses when a connection is destroyed.
- virtual void HandleConnectionDestroyed(Connection* conn) {}
+ virtual void HandleConnectionDestroyed(Connection*) {}
private:
void Construct();
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -217,7 +217,7 @@
virtual StunAttributeValueType value_type() const = 0;
// Only XorAddressAttribute needs this so far.
- virtual void SetOwner(StunMessage* owner) {}
+ virtual void SetOwner(StunMessage*) {}
// Reads the body (not the type or length) for this type of attribute from
// the given buffer. Return value is true if successful.
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -110,11 +110,11 @@
// Fills in a request object to be sent. Note that request's transaction ID
// will already be set and cannot be changed.
- virtual void Prepare(StunMessage* request) {}
+ virtual void Prepare(StunMessage*) {}
// Called when the message receives a response or times out.
- virtual void OnResponse(StunMessage* response) {}
- virtual void OnErrorResponse(StunMessage* response) {}
+ virtual void OnResponse(StunMessage*) {}
+ virtual void OnErrorResponse(StunMessage*) {}
virtual void OnTimeout() {}
// Called when the message is sent.
virtual void OnSent();
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h (210986 => 210987)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h 2017-01-20 22:19:40 UTC (rev 210986)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h 2017-01-20 22:30:24 UTC (rev 210987)
@@ -267,11 +267,11 @@
// Must be called before applying local session description.
virtual void SetLocalCertificate(
- const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {}
+ const rtc::scoped_refptr<rtc::RTCCertificate>&) {}
// Get a copy of the local certificate provided by SetLocalCertificate.
virtual bool GetLocalCertificate(
- rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
+ rtc::scoped_refptr<rtc::RTCCertificate>*) {
return false;
}
@@ -320,10 +320,10 @@
bool RemoveRemoteCandidates(const std::vector<Candidate>& candidates,
std::string* error);
- virtual bool GetSslRole(rtc::SSLRole* ssl_role) const { return false; }
+ virtual bool GetSslRole(rtc::SSLRole*) const { return false; }
// Must be called before channel is starting to connect.
- virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) {
+ virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion) {
return false;
}
