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<title>[210987] trunk/Source/ThirdParty/libwebrtc</title>
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<div id="msg">
<dl class="meta">
<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/210987">210987</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2017-01-20 14:30:24 -0800 (Fri, 20 Jan 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>[WebRTC] libwebrtc headers are incompatible with WebKit compilation flags
https://bugs.webkit.org/show_bug.cgi?id=167242
Patch by Youenn Fablet <youenn@apple.com> on 2017-01-20
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:</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceThirdPartylibwebrtcChangeLog">trunk/Source/ThirdParty/libwebrtc/ChangeLog</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcapijseph">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcapimediastreaminterfaceh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcapipeerconnectionfactoryh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcapipeerconnectioninterfaceh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcbasemessagehandlerh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcbasesanitizerh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcbasestreamh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmediabasemediachannelh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmediaenginewebrtcvideodecoderfactoryh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmediaenginewebrtcvideoencoderfactoryh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesincludemoduleh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcp2pbaseporth">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcp2pbasestunh">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcp2pbasestunrequesth">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcp2pbasetransporth">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceThirdPartylibwebrtcChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -1,3 +1,81 @@
</span><ins>+2017-01-20 Youenn Fablet <youenn@apple.com>
+
+ [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:
+
</ins><span class="cx"> 2017-01-20 Youenn Fablet <youennf@gmail.com>
</span><span class="cx">
</span><span class="cx"> [WebRTC] Update libwertc AudioRtpSender::SetAudioSend
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcapijseph"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -98,7 +98,7 @@
</span><span class="cx"> // Removes the candidates from the description.
</span><span class="cx"> // Returns the number of candidates removed.
</span><span class="cx"> virtual size_t RemoveCandidates(
</span><del>- const std::vector<cricket::Candidate>& candidates) { return 0; }
</del><ins>+ const std::vector<cricket::Candidate>&) { return 0; }
</ins><span class="cx">
</span><span class="cx"> // Returns the number of m- lines in the session description.
</span><span class="cx"> virtual size_t number_of_mediasections() const = 0;
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcapimediastreaminterfaceh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -134,9 +134,9 @@
</span><span class="cx"> public rtc::VideoSourceInterface<VideoFrame> {
</span><span class="cx"> public:
</span><span class="cx"> // Register a video sink for this track.
</span><del>- void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
- const rtc::VideoSinkWants& wants) override{};
- void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override{};
</del><ins>+ void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>*,
+ const rtc::VideoSinkWants&) override {};
+ void RemoveSink(rtc::VideoSinkInterface<VideoFrame>*) override {};
</ins><span class="cx">
</span><span class="cx"> virtual VideoTrackSourceInterface* GetSource() const = 0;
</span><span class="cx">
</span><span class="lines">@@ -174,15 +174,15 @@
</span><span class="cx"> // Sets the volume to the source. |volume| is in the range of [0, 10].
</span><span class="cx"> // TODO(tommi): This method should be on the track and ideally volume should
</span><span class="cx"> // be applied in the track in a way that does not affect clones of the track.
</span><del>- virtual void SetVolume(double volume) {}
</del><ins>+ virtual void SetVolume(double) {}
</ins><span class="cx">
</span><span class="cx"> // Registers/unregisters observer to the audio source.
</span><del>- virtual void RegisterAudioObserver(AudioObserver* observer) {}
- virtual void UnregisterAudioObserver(AudioObserver* observer) {}
</del><ins>+ virtual void RegisterAudioObserver(AudioObserver*) {}
+ virtual void UnregisterAudioObserver(AudioObserver*) {}
</ins><span class="cx">
</span><span class="cx"> // TODO(tommi): Make pure virtual.
</span><del>- virtual void AddSink(AudioTrackSinkInterface* sink) {}
- virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
</del><ins>+ virtual void AddSink(AudioTrackSinkInterface*) {}
+ virtual void RemoveSink(AudioTrackSinkInterface*) {}
</ins><span class="cx"> };
</span><span class="cx">
</span><span class="cx"> // Interface of the audio processor used by the audio track to collect
</span><span class="lines">@@ -230,7 +230,7 @@
</span><span class="cx"> // Return true on success, otherwise false.
</span><span class="cx"> // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
</span><span class="cx"> // after Chrome has the correct implementation of the interface.
</span><del>- virtual bool GetSignalLevel(int* level) { return false; }
</del><ins>+ virtual bool GetSignalLevel(int*) { return false; }
</ins><span class="cx">
</span><span class="cx"> // Get the audio processor used by the audio track. Return NULL if the track
</span><span class="cx"> // does not have any processor.
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcapipeerconnectionfactoryh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -81,10 +81,10 @@
</span><span class="cx"> bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
</span><span class="cx"> void StopAecDump() override;
</span><span class="cx"> // TODO(ivoc) Remove after Chrome is updated.
</span><del>- bool StartRtcEventLog(rtc::PlatformFile file) override { return false; }
</del><ins>+ bool StartRtcEventLog(rtc::PlatformFile) override { return false; }
</ins><span class="cx"> // TODO(ivoc) Remove after Chrome is updated.
</span><del>- bool StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) override {
</del><ins>+ bool StartRtcEventLog(rtc::PlatformFile,
+ int64_t) override {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="cx"> // TODO(ivoc) Remove after Chrome is updated.
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcapipeerconnectioninterfaceh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -119,9 +119,9 @@
</span><span class="cx"> // |type| is the type of the enum counter to be incremented. |counter|
</span><span class="cx"> // is the particular counter in that type. |counter_max| is the next sequence
</span><span class="cx"> // number after the highest counter.
</span><del>- virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
- int counter,
- int counter_max) {}
</del><ins>+ virtual void IncrementEnumCounter(PeerConnectionEnumCounterType,
+ int /* counter */,
+ int /* counter_max */) {}
</ins><span class="cx">
</span><span class="cx"> // This is used to handle sparse counters like SSL cipher suites.
</span><span class="cx"> // TODO(guoweis): Remove the implementation once the dependency's interface
</span><span class="lines">@@ -389,14 +389,14 @@
</span><span class="cx"> // |streams| indicates which stream labels the track should be associated
</span><span class="cx"> // with.
</span><span class="cx"> virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
</span><del>- MediaStreamTrackInterface* track,
- std::vector<MediaStreamInterface*> streams) {
</del><ins>+ MediaStreamTrackInterface*,
+ std::vector<MediaStreamInterface*>) {
</ins><span class="cx"> return nullptr;
</span><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> // Remove an RtpSender from this PeerConnection.
</span><span class="cx"> // Returns true on success.
</span><del>- virtual bool RemoveTrack(RtpSenderInterface* sender) {
</del><ins>+ virtual bool RemoveTrack(RtpSenderInterface*) {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -410,8 +410,8 @@
</span><span class="cx"> // |stream_id| is used to populate the msid attribute; if empty, one will
</span><span class="cx"> // be generated automatically.
</span><span class="cx"> virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
</span><del>- const std::string& kind,
- const std::string& stream_id) {
</del><ins>+ const std::string& /* kind */,
+ const std::string& /* stream_id */) {
</ins><span class="cx"> return rtc::scoped_refptr<RtpSenderInterface>();
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -433,7 +433,7 @@
</span><span class="cx"> // TODO(hbos): Default implementation that does nothing only exists as to not
</span><span class="cx"> // break third party projects. As soon as they have been updated this should
</span><span class="cx"> // be changed to "= 0;".
</span><del>- virtual void GetStats(RTCStatsCollectorCallback* callback) {}
</del><ins>+ virtual void GetStats(RTCStatsCollectorCallback*) {}
</ins><span class="cx">
</span><span class="cx"> virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
</span><span class="cx"> const std::string& label,
</span><span class="lines">@@ -444,23 +444,23 @@
</span><span class="cx">
</span><span class="cx"> // Create a new offer.
</span><span class="cx"> // The CreateSessionDescriptionObserver callback will be called when done.
</span><del>- virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) {}
</del><ins>+ virtual void CreateOffer(CreateSessionDescriptionObserver*,
+ const MediaConstraintsInterface*) {}
</ins><span class="cx">
</span><span class="cx"> // TODO(jiayl): remove the default impl and the old interface when chromium
</span><span class="cx"> // code is updated.
</span><del>- virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
- const RTCOfferAnswerOptions& options) {}
</del><ins>+ virtual void CreateOffer(CreateSessionDescriptionObserver*,
+ const RTCOfferAnswerOptions&) {}
</ins><span class="cx">
</span><span class="cx"> // Create an answer to an offer.
</span><span class="cx"> // The CreateSessionDescriptionObserver callback will be called when done.
</span><del>- virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
- const RTCOfferAnswerOptions& options) {}
</del><ins>+ virtual void CreateAnswer(CreateSessionDescriptionObserver*,
+ const RTCOfferAnswerOptions&) {}
</ins><span class="cx"> // Deprecated - use version above.
</span><span class="cx"> // TODO(hta): Remove and remove default implementations when all callers
</span><span class="cx"> // are updated.
</span><del>- virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) {}
</del><ins>+ virtual void CreateAnswer(CreateSessionDescriptionObserver*,
+ const MediaConstraintsInterface*) {}
</ins><span class="cx">
</span><span class="cx"> // Sets the local session description.
</span><span class="cx"> // JsepInterface takes the ownership of |desc| even if it fails.
</span><span class="lines">@@ -475,11 +475,11 @@
</span><span class="cx"> // Restarts or updates the ICE Agent process of gathering local candidates
</span><span class="cx"> // and pinging remote candidates.
</span><span class="cx"> // TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
</span><del>- virtual bool UpdateIce(const IceServers& configuration,
- const MediaConstraintsInterface* constraints) {
</del><ins>+ virtual bool UpdateIce(const IceServers&,
+ const MediaConstraintsInterface*) {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><del>- virtual bool UpdateIce(const IceServers& configuration) { return false; }
</del><ins>+ virtual bool UpdateIce(const IceServers&) { return false; }
</ins><span class="cx"> // Sets the PeerConnection's global configuration to |config|.
</span><span class="cx"> // Any changes to STUN/TURN servers or ICE candidate policy will affect the
</span><span class="cx"> // next gathering phase, and cause the next call to createOffer to generate
</span><span class="lines">@@ -488,7 +488,7 @@
</span><span class="cx"> // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
</span><span class="cx"> // PeerConnectionInterface implement it.
</span><span class="cx"> virtual bool SetConfiguration(
</span><del>- const PeerConnectionInterface::RTCConfiguration& config) {
</del><ins>+ const PeerConnectionInterface::RTCConfiguration&) {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="cx"> // Provides a remote candidate to the ICE Agent.
</span><span class="lines">@@ -501,7 +501,7 @@
</span><span class="cx">
</span><span class="cx"> // Removes a group of remote candidates from the ICE agent.
</span><span class="cx"> virtual bool RemoveIceCandidates(
</span><del>- const std::vector<cricket::Candidate>& candidates) {
</del><ins>+ const std::vector<cricket::Candidate>&) {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -518,8 +518,8 @@
</span><span class="cx"> // automatically after 10 minutes have passed, or when the StopRtcEventLog
</span><span class="cx"> // function is called.
</span><span class="cx"> // TODO(ivoc): Make this pure virtual when Chrome is updated.
</span><del>- virtual bool StartRtcEventLog(rtc::PlatformFile file,
- int64_t max_size_bytes) {
</del><ins>+ virtual bool StartRtcEventLog(rtc::PlatformFile,
+ int64_t /* max_size_bytes */) {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -553,21 +553,21 @@
</span><span class="cx"> // pointer version.
</span><span class="cx">
</span><span class="cx"> // Triggered when media is received on a new stream from remote peer.
</span><del>- virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
</del><ins>+ virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface>) {}
</ins><span class="cx"> // Deprecated; please use the version that uses a scoped_refptr.
</span><del>- virtual void OnAddStream(MediaStreamInterface* stream) {}
</del><ins>+ virtual void OnAddStream(MediaStreamInterface*) {}
</ins><span class="cx">
</span><span class="cx"> // Triggered when a remote peer close a stream.
</span><del>- virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
</del><ins>+ virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface>) {
</ins><span class="cx"> }
</span><span class="cx"> // Deprecated; please use the version that uses a scoped_refptr.
</span><del>- virtual void OnRemoveStream(MediaStreamInterface* stream) {}
</del><ins>+ virtual void OnRemoveStream(MediaStreamInterface*) {}
</ins><span class="cx">
</span><span class="cx"> // Triggered when a remote peer opens a data channel.
</span><span class="cx"> virtual void OnDataChannel(
</span><del>- rtc::scoped_refptr<DataChannelInterface> data_channel){};
</del><ins>+ rtc::scoped_refptr<DataChannelInterface>){};
</ins><span class="cx"> // Deprecated; please use the version that uses a scoped_refptr.
</span><del>- virtual void OnDataChannel(DataChannelInterface* data_channel) {}
</del><ins>+ virtual void OnDataChannel(DataChannelInterface*) {}
</ins><span class="cx">
</span><span class="cx"> // Triggered when renegotiation is needed. For example, an ICE restart
</span><span class="cx"> // has begun.
</span><span class="lines">@@ -588,10 +588,10 @@
</span><span class="cx"> // TODO(honghaiz): Make this a pure virtual method when all its subclasses
</span><span class="cx"> // implement it.
</span><span class="cx"> virtual void OnIceCandidatesRemoved(
</span><del>- const std::vector<cricket::Candidate>& candidates) {}
</del><ins>+ const std::vector<cricket::Candidate>&) {}
</ins><span class="cx">
</span><span class="cx"> // Called when the ICE connection receiving status changes.
</span><del>- virtual void OnIceConnectionReceivingChange(bool receiving) {}
</del><ins>+ virtual void OnIceConnectionReceivingChange(bool /* receiving */) {}
</ins><span class="cx">
</span><span class="cx"> protected:
</span><span class="cx"> // Dtor protected as objects shouldn't be deleted via this interface.
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcbasemessagehandlerh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -40,7 +40,7 @@
</span><span class="cx"> public:
</span><span class="cx"> explicit FunctorMessageHandler(const FunctorT& functor)
</span><span class="cx"> : functor_(functor) {}
</span><del>- virtual void OnMessage(Message* msg) {
</del><ins>+ virtual void OnMessage(Message*) {
</ins><span class="cx"> result_ = functor_();
</span><span class="cx"> }
</span><span class="cx"> const ReturnT& result() const { return result_; }
</span><span class="lines">@@ -56,7 +56,7 @@
</span><span class="cx"> : public MessageHandler {
</span><span class="cx"> public:
</span><span class="cx"> explicit FunctorMessageHandler(const FunctorT& functor) : functor_(functor) {}
</span><del>- virtual void OnMessage(Message* msg) { result_ = std::move(functor_()); }
</del><ins>+ virtual void OnMessage(Message*) { result_ = std::move(functor_()); }
</ins><span class="cx"> std::unique_ptr<ReturnT> result() { return std::move(result_); }
</span><span class="cx">
</span><span class="cx"> private:
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcbasesanitizerh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -42,6 +42,12 @@
</span><span class="cx"> #define RTC_NO_SANITIZE(what)
</span><span class="cx"> #endif
</span><span class="cx">
</span><ins>+#if !RTC_HAS_ASAN
+#define SANITIZER_UNUSED3(x, y, z) (void)&(x); \
+ (void)&(y); \
+ (void)&(z)
+#endif
+
</ins><span class="cx"> // Ask ASan to mark the memory range [ptr, ptr + element_size * num_elements)
</span><span class="cx"> // as being unaddressable, so that reads and writes are not allowed. ASan may
</span><span class="cx"> // narrow the range to the nearest alignment boundaries.
</span><span class="lines">@@ -50,6 +56,8 @@
</span><span class="cx"> size_t num_elements) {
</span><span class="cx"> #if RTC_HAS_ASAN
</span><span class="cx"> ASAN_POISON_MEMORY_REGION(ptr, element_size * num_elements);
</span><ins>+#else
+ SANITIZER_UNUSED3(ptr, element_size, num_elements);
</ins><span class="cx"> #endif
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -61,6 +69,8 @@
</span><span class="cx"> size_t num_elements) {
</span><span class="cx"> #if RTC_HAS_ASAN
</span><span class="cx"> ASAN_UNPOISON_MEMORY_REGION(ptr, element_size * num_elements);
</span><ins>+#else
+ SANITIZER_UNUSED3(ptr, element_size, num_elements);
</ins><span class="cx"> #endif
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -71,6 +81,8 @@
</span><span class="cx"> size_t num_elements) {
</span><span class="cx"> #if RTC_HAS_MSAN
</span><span class="cx"> __msan_poison(ptr, element_size * num_elements);
</span><ins>+#else
+ SANITIZER_UNUSED3(ptr, element_size, num_elements);
</ins><span class="cx"> #endif
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -82,6 +94,8 @@
</span><span class="cx"> size_t num_elements) {
</span><span class="cx"> #if RTC_HAS_MSAN
</span><span class="cx"> __msan_check_mem_is_initialized(ptr, element_size * num_elements);
</span><ins>+#else
+ SANITIZER_UNUSED3(ptr, element_size, num_elements);
</ins><span class="cx"> #endif
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcbasestreamh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -132,7 +132,7 @@
</span><span class="cx"> // processed. Read and ConsumeReadData invalidate the buffer returned by
</span><span class="cx"> // GetReadData.
</span><span class="cx"> virtual const void* GetReadData(size_t* data_len);
</span><del>- virtual void ConsumeReadData(size_t used) {}
</del><ins>+ virtual void ConsumeReadData(size_t) {}
</ins><span class="cx">
</span><span class="cx"> // GetWriteBuffer returns a pointer to a buffer which is owned by the stream.
</span><span class="cx"> // The buffer has a capacity of buf_len bytes. NULL is returned if there is
</span><span class="lines">@@ -146,7 +146,7 @@
</span><span class="cx"> // when it is available. If the requested amount is too large, return an
</span><span class="cx"> // error.
</span><span class="cx"> virtual void* GetWriteBuffer(size_t* buf_len);
</span><del>- virtual void ConsumeWriteBuffer(size_t used) {}
</del><ins>+ virtual void ConsumeWriteBuffer(size_t) {}
</ins><span class="cx">
</span><span class="cx"> // Write data_len bytes found in data, circumventing any throttling which
</span><span class="cx"> // would could cause SR_BLOCK to be returned. Returns true if all the data
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmediabasemediachannelh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -1159,13 +1159,13 @@
</span><span class="cx"> virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
</span><span class="cx">
</span><span class="cx"> // TODO(pthatcher): Implement this.
</span><del>- virtual bool GetStats(DataMediaInfo* info) { return true; }
</del><ins>+ virtual bool GetStats(DataMediaInfo*) { return true; }
</ins><span class="cx">
</span><span class="cx"> virtual bool SetSend(bool send) = 0;
</span><span class="cx"> virtual bool SetReceive(bool receive) = 0;
</span><span class="cx">
</span><del>- virtual void OnNetworkRouteChanged(const std::string& transport_name,
- const rtc::NetworkRoute& network_route) {}
</del><ins>+ virtual void OnNetworkRouteChanged(const std::string& /* transport_name */,
+ const rtc::NetworkRoute&) {}
</ins><span class="cx">
</span><span class="cx"> virtual bool SendData(
</span><span class="cx"> const SendDataParams& params,
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmediaenginewebrtcvideodecoderfactoryh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -32,7 +32,7 @@
</span><span class="cx"> webrtc::VideoCodecType type) = 0;
</span><span class="cx"> virtual webrtc::VideoDecoder* CreateVideoDecoderWithParams(
</span><span class="cx"> webrtc::VideoCodecType type,
</span><del>- VideoDecoderParams params) {
</del><ins>+ VideoDecoderParams) {
</ins><span class="cx"> return CreateVideoDecoder(type);
</span><span class="cx"> }
</span><span class="cx"> virtual ~WebRtcVideoDecoderFactory() {}
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmediaenginewebrtcvideoencoderfactoryh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -36,9 +36,9 @@
</span><span class="cx">
</span><span class="cx"> VideoCodec(webrtc::VideoCodecType t,
</span><span class="cx"> const std::string& nm,
</span><del>- int w,
- int h,
- int fr)
</del><ins>+ int /* w */,
+ int /* h */,
+ int /* fr */)
</ins><span class="cx"> : type(t), name(nm) {}
</span><span class="cx"> };
</span><span class="cx">
</span><span class="lines">@@ -70,7 +70,7 @@
</span><span class="cx"> // frames to be delivered via webrtc::VideoEncoder::Encode. This flag is used
</span><span class="cx"> // as the internal_source parameter to
</span><span class="cx"> // webrtc::ViEExternalCodec::RegisterExternalSendCodec.
</span><del>- virtual bool EncoderTypeHasInternalSource(webrtc::VideoCodecType type) const {
</del><ins>+ virtual bool EncoderTypeHasInternalSource(webrtc::VideoCodecType) const {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcmodulesincludemoduleh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -53,7 +53,7 @@
</span><span class="cx"> //
</span><span class="cx"> // NOTE: This method is not called from the worker thread itself, but from
</span><span class="cx"> // the thread that registers/deregisters the module or calls Start/Stop.
</span><del>- virtual void ProcessThreadAttached(ProcessThread* process_thread) {}
</del><ins>+ virtual void ProcessThreadAttached(ProcessThread*) {}
</ins><span class="cx">
</span><span class="cx"> protected:
</span><span class="cx"> virtual ~Module() {}
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcp2pbaseporth"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -241,9 +241,9 @@
</span><span class="cx"> // port implemented this method.
</span><span class="cx"> // TODO(mallinath) - Make it pure virtual.
</span><span class="cx"> virtual bool HandleIncomingPacket(
</span><del>- rtc::AsyncPacketSocket* socket, const char* data, size_t size,
- const rtc::SocketAddress& remote_addr,
- const rtc::PacketTime& packet_time) {
</del><ins>+ rtc::AsyncPacketSocket*, const char*, size_t,
+ const rtc::SocketAddress&,
+ const rtc::PacketTime&) {
</ins><span class="cx"> ASSERT(false);
</span><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="lines">@@ -360,7 +360,7 @@
</span><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> // Extra work to be done in subclasses when a connection is destroyed.
</span><del>- virtual void HandleConnectionDestroyed(Connection* conn) {}
</del><ins>+ virtual void HandleConnectionDestroyed(Connection*) {}
</ins><span class="cx">
</span><span class="cx"> private:
</span><span class="cx"> void Construct();
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcp2pbasestunh"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -217,7 +217,7 @@
</span><span class="cx"> virtual StunAttributeValueType value_type() const = 0;
</span><span class="cx">
</span><span class="cx"> // Only XorAddressAttribute needs this so far.
</span><del>- virtual void SetOwner(StunMessage* owner) {}
</del><ins>+ virtual void SetOwner(StunMessage*) {}
</ins><span class="cx">
</span><span class="cx"> // Reads the body (not the type or length) for this type of attribute from
</span><span class="cx"> // the given buffer. Return value is true if successful.
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcp2pbasestunrequesth"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -110,11 +110,11 @@
</span><span class="cx">
</span><span class="cx"> // Fills in a request object to be sent. Note that request's transaction ID
</span><span class="cx"> // will already be set and cannot be changed.
</span><del>- virtual void Prepare(StunMessage* request) {}
</del><ins>+ virtual void Prepare(StunMessage*) {}
</ins><span class="cx">
</span><span class="cx"> // Called when the message receives a response or times out.
</span><del>- virtual void OnResponse(StunMessage* response) {}
- virtual void OnErrorResponse(StunMessage* response) {}
</del><ins>+ virtual void OnResponse(StunMessage*) {}
+ virtual void OnErrorResponse(StunMessage*) {}
</ins><span class="cx"> virtual void OnTimeout() {}
</span><span class="cx"> // Called when the message is sent.
</span><span class="cx"> virtual void OnSent();
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcp2pbasetransporth"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h (210986 => 210987)</h4>
<pre class="diff"><span>
<span class="info">--- 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)
</span><span class="lines">@@ -267,11 +267,11 @@
</span><span class="cx">
</span><span class="cx"> // Must be called before applying local session description.
</span><span class="cx"> virtual void SetLocalCertificate(
</span><del>- const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {}
</del><ins>+ const rtc::scoped_refptr<rtc::RTCCertificate>&) {}
</ins><span class="cx">
</span><span class="cx"> // Get a copy of the local certificate provided by SetLocalCertificate.
</span><span class="cx"> virtual bool GetLocalCertificate(
</span><del>- rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
</del><ins>+ rtc::scoped_refptr<rtc::RTCCertificate>*) {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -320,10 +320,10 @@
</span><span class="cx"> bool RemoveRemoteCandidates(const std::vector<Candidate>& candidates,
</span><span class="cx"> std::string* error);
</span><span class="cx">
</span><del>- virtual bool GetSslRole(rtc::SSLRole* ssl_role) const { return false; }
</del><ins>+ virtual bool GetSslRole(rtc::SSLRole*) const { return false; }
</ins><span class="cx">
</span><span class="cx"> // Must be called before channel is starting to connect.
</span><del>- virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) {
</del><ins>+ virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion) {
</ins><span class="cx"> return false;
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre>
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