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<title>[211391] trunk/Source/ThirdParty/libwebrtc</title>
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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/211391">211391</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2017-01-30 16:06:48 -0800 (Mon, 30 Jan 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>[WebRTC] Upload a diff of WebKit libwebrtc code and original libwebrtc code
https://bugs.webkit.org/show_bug.cgi?id=167573
Patch by Youenn Fablet <youennf@gmail.com> on 2017-01-30
Reviewed by Alex Christensen.
* WebKit/patch-libwebrtc: Added.</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceThirdPartylibwebrtcChangeLog">trunk/Source/ThirdParty/libwebrtc/ChangeLog</a></li>
</ul>
<h3>Added Paths</h3>
<ul>
<li><a href="#trunkSourceThirdPartylibwebrtcWebKitpatchlibwebrtc">trunk/Source/ThirdParty/libwebrtc/WebKit/patch-libwebrtc</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceThirdPartylibwebrtcChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (211390 => 211391)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-01-30 23:37:57 UTC (rev 211390)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-01-31 00:06:48 UTC (rev 211391)
</span><span class="lines">@@ -1,3 +1,12 @@
</span><ins>+2017-01-30 Youenn Fablet <youennf@gmail.com>
+
+ [WebRTC] Upload a diff of WebKit libwebrtc code and original libwebrtc code
+ https://bugs.webkit.org/show_bug.cgi?id=167573
+
+ Reviewed by Alex Christensen.
+
+ * WebKit/patch-libwebrtc: Added.
+
</ins><span class="cx"> 2017-01-27 Dan Bernstein <mitz@apple.com>
</span><span class="cx">
</span><span class="cx"> Ignore Xcode’s project.xcworkspace and userdata directories in this new project like we do
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcWebKitpatchlibwebrtc"></a>
<div class="addfile"><h4>Added: trunk/Source/ThirdParty/libwebrtc/WebKit/patch-libwebrtc (0 => 211391)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/WebKit/patch-libwebrtc (rev 0)
+++ trunk/Source/ThirdParty/libwebrtc/WebKit/patch-libwebrtc 2017-01-31 00:06:48 UTC (rev 211391)
</span><span class="lines">@@ -0,0 +1,979 @@
</span><ins>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
+index ddeb1e5..4b2e232 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
+@@ -75,7 +75,7 @@ class DataChannelObserver {
+ // A data buffer was successfully received.
+ virtual void OnMessage(const DataBuffer& buffer) = 0;
+ // The data channel's buffered_amount has changed.
+- virtual void OnBufferedAmountChange(uint64_t previous_amount){};
++ virtual void OnBufferedAmountChange(uint64_t){};
+
+ protected:
+ virtual ~DataChannelObserver() {}
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
+index e11a2ad..9d7bf9d 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
+@@ -98,7 +98,7 @@ class SessionDescriptionInterface {
+ // 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;
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
+index baebad7..91a8705 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
+@@ -134,9 +134,9 @@ class VideoTrackInterface
+ 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 @@ class AudioSourceInterface : public MediaSourceInterface {
+ // 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 @@ class AudioTrackInterface : public MediaStreamTrackInterface {
+ // 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.
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h
+index 7a30ab4..f02f9b9 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactory.h
+@@ -81,10 +81,10 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
+ 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.
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
+index ea056bf..76cce17 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
+@@ -119,9 +119,9 @@ class MetricsObserverInterface : public rtc::RefCountInterface {
+ // |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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
+ // |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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
+ // |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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
+ // 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
+
+ // 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
+ // 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
+ // 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
+
+ // 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 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
+ // 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 @@ class PeerConnectionObserver {
+ // 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 @@ class PeerConnectionObserver {
+ // 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.
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsender.cc
+index 2e01d96..0dc7916 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsender.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsender.cc
+@@ -204,7 +204,7 @@ void AudioRtpSender::SetAudioSend() {
+ return;
+ }
+ cricket::AudioOptions options;
+-#if !defined(WEBRTC_CHROMIUM_BUILD)
++#if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_WEBKIT_BUILD)
+ // TODO(tommi): Remove this hack when we move CreateAudioSource out of
+ // PeerConnection. This is a bit of a strange way to apply local audio
+ // options since it is also applied to all streams/channels, local or remote.
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/base/checks.h b/Source/ThirdParty/libwebrtc/Source/webrtc/base/checks.h
+index 653ed32..57a8b51 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/base/checks.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/base/checks.h
+@@ -221,7 +221,9 @@ class FatalMessageVoidify {
+ #define RTC_UNREACHABLE_CODE_HIT false
+ #define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
+
++#ifndef FATAL
+ #define FATAL() rtc::FatalMessage(__FILE__, __LINE__).stream()
++#endif
+ // TODO(ajm): Consider adding RTC_NOTIMPLEMENTED macro when
+ // base/logging.h and system_wrappers/logging.h are consolidated such that we
+ // can match the Chromium behavior.
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h
+index 6a3c2ef..795d04e 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h
+@@ -40,7 +40,7 @@ class FunctorMessageHandler : public MessageHandler {
+ 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 @@ class FunctorMessageHandler<class std::unique_ptr<ReturnT>, FunctorT>
+ : 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:
+@@ -70,7 +70,7 @@ class FunctorMessageHandler<void, FunctorT> : public MessageHandler {
+ public:
+ explicit FunctorMessageHandler(const FunctorT& functor)
+ : functor_(functor) {}
+- virtual void OnMessage(Message* msg) {
++ virtual void OnMessage(Message*) {
+ functor_();
+ }
+ void result() const {}
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/base/network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/base/network.h
+index 7d509c6..56a7a7d 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/base/network.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/base/network.h
+@@ -118,7 +118,7 @@ class NetworkManager : public DefaultLocalAddressProvider {
+ // useful as binding to such interfaces allow default routing behavior like
+ // http traffic.
+ // TODO(guoweis): remove this body when chromium implements this.
+- virtual void GetAnyAddressNetworks(NetworkList* networks) {}
++ virtual void GetAnyAddressNetworks(NetworkList*) {}
+
+ // Dumps the current list of networks in the network manager.
+ virtual void DumpNetworks() {}
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h
+index e27a692..004b449 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/base/sanitizer.h
+@@ -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 @@ static inline void rtc_AsanPoison(const volatile void* ptr,
+ 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 @@ static inline void rtc_AsanUnpoison(const volatile void* ptr,
+ 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 @@ static inline void rtc_MsanMarkUninitialized(const volatile void* ptr,
+ 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 @@ static inline void rtc_MsanCheckInitialized(const volatile void* ptr,
+ 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
+ }
+
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/base/sha1digest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/base/sha1digest.cc
+index c090a06..c29c859 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/base/sha1digest.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/base/sha1digest.cc
+@@ -17,15 +17,15 @@ size_t Sha1Digest::Size() const {
+ }
+
+ void Sha1Digest::Update(const void* buf, size_t len) {
+- SHA1Update(&ctx_, static_cast<const uint8_t*>(buf), len);
++ SHA1_UPDATE(&ctx_, static_cast<const uint8_t*>(buf), len);
+ }
+
+ size_t Sha1Digest::Finish(void* buf, size_t len) {
+ if (len < kSize) {
+ return 0;
+ }
+- SHA1Final(&ctx_, static_cast<uint8_t*>(buf));
+- SHA1Init(&ctx_); // Reset for next use.
++ SHA1_FINAL(&ctx_, static_cast<uint8_t*>(buf));
++ SHA1_INIT(&ctx_); // Reset for next use.
+ return kSize;
+ }
+
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/base/sha1digest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/base/sha1digest.h
+index d321cb8..a5fb658 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/base/sha1digest.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/base/sha1digest.h
+@@ -12,7 +12,21 @@
+ #define WEBRTC_BASE_SHA1DIGEST_H_
+
+ #include "webrtc/base/messagedigest.h"
++
++#ifdef USE_RTC_SHA1
+ #include "webrtc/base/sha1.h"
++#define SHA1_INIT SHA1Init
++#define SHA1_INIT SHA1Init
++#define SHA1_UPDATE SHA1Update
++#define SHA1_FINAL SHA1Final
++#else
++#include <openssl/sha.h>
++#define SHA1_CTX SHA_CTX
++#define SHA1_INIT SHA1_Init
++#define SHA1_UPDATE SHA1_Update
++#define SHA1_FINAL(x, y) SHA1_Final(y, x)
++#define SHA1_DIGEST_SIZE 20
++#endif
+
+ namespace rtc {
+
+@@ -21,7 +35,7 @@ class Sha1Digest : public MessageDigest {
+ public:
+ enum { kSize = SHA1_DIGEST_SIZE };
+ Sha1Digest() {
+- SHA1Init(&ctx_);
++ SHA1_INIT(&ctx_);
+ }
+ size_t Size() const override;
+ void Update(const void* buf, size_t len) override;
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/base/socketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/base/socketserver.h
+index 7071f22..a4f12d7 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/base/socketserver.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/base/socketserver.h
+@@ -32,7 +32,7 @@ class SocketServer : public SocketFactory {
+ // When the socket server is installed into a Thread, this function is
+ // called to allow the socket server to use the thread's message queue for
+ // any messaging that it might need to perform.
+- virtual void SetMessageQueue(MessageQueue* queue) {}
++ virtual void SetMessageQueue(MessageQueue*) {}
+
+ // Sleeps until:
+ // 1) cms milliseconds have elapsed (unless cms == kForever)
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h
+index bc94906..3acd830 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/base/stream.h
+@@ -132,7 +132,7 @@ class StreamInterface : public MessageHandler {
+ // 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 @@ class StreamInterface : public MessageHandler {
+ // 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
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
+index 80b9d4d..f3af2da 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
+@@ -1159,13 +1159,13 @@ class DataMediaChannel : public MediaChannel {
+ 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,
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h
+index 7f0921f..482274e 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h
+@@ -32,7 +32,7 @@ class WebRtcVideoDecoderFactory {
+ webrtc::VideoCodecType type) = 0;
+ virtual webrtc::VideoDecoder* CreateVideoDecoderWithParams(
+ webrtc::VideoCodecType type,
+- VideoDecoderParams params) {
++ VideoDecoderParams) {
+ return CreateVideoDecoder(type);
+ }
+ virtual ~WebRtcVideoDecoderFactory() {}
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h
+index 35e4092..c95c6f7 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h
+@@ -36,9 +36,9 @@ class WebRtcVideoEncoderFactory {
+
+ 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 @@ class WebRtcVideoEncoderFactory {
+ // 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;
+ }
+
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine2.cc
+index c43dff8..dd57ee5 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine2.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine2.cc
+@@ -115,10 +115,13 @@ class WebRtcSimulcastEncoderFactory
+ const cricket::VideoCodec& codec) override {
+ RTC_DCHECK(factory_ != NULL);
+ // If it's a codec type we can simulcast, create a wrapped encoder.
++#ifndef RTC_DISABLE_VP8
++ ASSERT(webrtc::VP8Encoder::IsSupported());
+ if (CodecNamesEq(codec.name.c_str(), kVp8CodecName)) {
+ return new webrtc::SimulcastEncoderAdapter(
+ new EncoderFactoryAdapter(factory_));
+ }
++#endif
+ webrtc::VideoEncoder* encoder = factory_->CreateVideoEncoder(codec);
+ if (encoder) {
+ non_simulcast_encoders_.push_back(encoder);
+@@ -430,9 +433,11 @@ void AddCodecAndMaybeRtxCodec(const VideoCodec& codec,
+
+ std::vector<VideoCodec> DefaultVideoCodecList() {
+ std::vector<VideoCodec> codecs;
+- AddCodecAndMaybeRtxCodec(
++ if (webrtc::VP8Encoder::IsSupported() && webrtc::VP8Decoder::IsSupported()) {
++ AddCodecAndMaybeRtxCodec(
+ MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType, kVp8CodecName),
+ &codecs);
++ }
+ if (webrtc::VP9Encoder::IsSupported() && webrtc::VP9Decoder::IsSupported()) {
+ AddCodecAndMaybeRtxCodec(MakeVideoCodecWithDefaultFeedbackParams(
+ kDefaultVp9PlType, kVp9CodecName),
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
+index 9468f0c..e4960f9 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
+@@ -231,12 +231,14 @@ std::unique_ptr<AudioEncoder> CreateEncoder(
+ if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderOpus(speech_inst));
+ #endif
++#ifdef WEBRTC_CODEC_G711
+ if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmU(speech_inst));
+ if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmA(speech_inst));
+ if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcm16B(speech_inst));
++#endif
+ #ifdef WEBRTC_CODEC_ILBC
+ if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbc(speech_inst));
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
+index 05b65c3..40ac08f 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
+@@ -16,7 +16,9 @@
+ #include "webrtc/base/optional.h"
+ #include "webrtc/common_types.h"
+ #include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
++#ifdef WEBRTC_CODEC_G711
+ #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
++#endif
+ #ifdef WEBRTC_CODEC_G722
+ #include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+ #endif
+@@ -50,6 +52,7 @@ std::unique_ptr<AudioDecoder> Unique(AudioDecoder* d) {
+ // TODO(kwiberg): These factory functions should probably be moved to each
+ // decoder.
+ NamedDecoderConstructor decoder_constructors[] = {
++#ifdef WEBRTC_CODEC_G711
+ {"pcmu",
+ [](const SdpAudioFormat& format) {
+ return format.clockrate_hz == 8000 && format.num_channels >= 1
+@@ -62,6 +65,7 @@ NamedDecoderConstructor decoder_constructors[] = {
+ ? Unique(new AudioDecoderPcmA(format.num_channels))
+ : nullptr;
+ }},
++#endif
+ #ifdef WEBRTC_CODEC_ILBC
+ {"ilbc",
+ [](const SdpAudioFormat& format) {
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+index baa5d38..7cc4b13 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+@@ -104,6 +104,8 @@ void AudioEncoderPcm::Reset() {
+ speech_buffer_.clear();
+ }
+
++#ifdef WEBRTC_CODEC_G711
++
+ AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst)
+ : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {}
+
+@@ -138,4 +140,5 @@ AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
+ return AudioEncoder::CodecType::kPcmU;
+ }
+
++#endif
+ } // namespace webrtc
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+index 7213445..b1679de 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+@@ -71,6 +71,8 @@ class AudioEncoderPcm : public AudioEncoder {
+
+ struct CodecInst;
+
++#ifdef WEBRTC_CODEC_G711
++
+ class AudioEncoderPcmA final : public AudioEncoderPcm {
+ public:
+ struct Config : public AudioEncoderPcm::Config {
+@@ -118,7 +120,7 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
+ static const int kSampleRateHz = 8000;
+ RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
+ };
+-
++#endif
+ } // namespace webrtc
+
+ #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac.mm
+index f11b36d..ba10db7 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac.mm
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac.mm
+@@ -1021,7 +1021,7 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
+ return nullptr;
+ }
+
+- return capturer;
++ return std::move(capturer);
+ }
+
+ } // namespace webrtc
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h
+index 432ed40..2141f97 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h
+@@ -53,7 +53,7 @@ class Module {
+ //
+ // 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() {}
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
+index dd35142..c9e45ee 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
+@@ -19,6 +19,7 @@ namespace webrtc {
+
+ class VP8Encoder : public VideoEncoder {
+ public:
++ static bool IsSupported();
+ static VP8Encoder* Create();
+
+ virtual ~VP8Encoder() {}
+@@ -26,6 +27,7 @@ class VP8Encoder : public VideoEncoder {
+
+ class VP8Decoder : public VideoDecoder {
+ public:
++ static bool IsSupported();
+ static VP8Decoder* Create();
+
+ virtual ~VP8Decoder() {}
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
+index b6b463c..85b6a56 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
+@@ -103,10 +103,18 @@ int NumStreamsDisabled(const std::vector<bool>& streams) {
+ }
+ } // namespace
+
++bool VP8Encoder::IsSupported() {
++ return true;
++}
++
+ VP8Encoder* VP8Encoder::Create() {
+ return new VP8EncoderImpl();
+ }
+
++bool VP8Decoder::IsSupported() {
++ return true;
++}
++
+ VP8Decoder* VP8Decoder::Create() {
+ return new VP8DecoderImpl();
+ }
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
+index 3c84a14..def9f5d 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
+@@ -241,9 +241,9 @@ class Port : public PortInterface, public rtc::MessageHandler,
+ // 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 @@ class Port : public PortInterface, public rtc::MessageHandler,
+ }
+
+ // Extra work to be done in subclasses when a connection is destroyed.
+- virtual void HandleConnectionDestroyed(Connection* conn) {}
++ virtual void HandleConnectionDestroyed(Connection*) {}
+
+ private:
+ void Construct();
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
+index 153f63d..5533b31 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
+@@ -217,7 +217,7 @@ class StunAttribute {
+ 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.
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
+index f6d216d..ed860a8 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
+@@ -110,11 +110,11 @@ class StunRequest : public rtc::MessageHandler {
+
+ // 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();
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h
+index adef653..cd156f9 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transport.h
+@@ -267,11 +267,11 @@ class Transport : public sigslot::has_slots<> {
+
+ // 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 @@ class Transport : public sigslot::has_slots<> {
+ 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;
+ }
+
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCNSGLVideoView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCNSGLVideoView.m
+index 415efe8..d87102a 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCNSGLVideoView.m
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCNSGLVideoView.m
+@@ -8,6 +8,8 @@
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
++#import <AVFoundation/AVFoundation.h>
++
+ #if !TARGET_OS_IPHONE
+
+ #import "WebRTC/RTCNSGLVideoView.h"
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCOpenGLDefines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCOpenGLDefines.h
+index 1ef642b..52bd234 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCOpenGLDefines.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCOpenGLDefines.h
+@@ -8,6 +8,8 @@
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
++#import <AVFoundation/AVFoundation.h>
++
+ #if TARGET_OS_IPHONE
+ #define RTC_PIXEL_FORMAT GL_LUMINANCE
+ #define SHADER_VERSION
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc
+index 4bf7a4d..8cee4df 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc
+@@ -12,10 +12,8 @@
+ #include "webrtc/base/logging.h"
+ #include "webrtc/common_video/h264/profile_level_id.h"
+ #include "webrtc/media/base/codec.h"
+-#if defined(WEBRTC_IOS)
+ #include "webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_encoder.h"
+ #include "webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_decoder.h"
+-#endif
+
+ // TODO(kthelgason): delete this when CreateVideoDecoder takes
+ // a cricket::VideoCodec instead of webrtc::VideoCodecType.
+@@ -38,7 +36,6 @@ namespace webrtc {
+
+ VideoToolboxVideoEncoderFactory::VideoToolboxVideoEncoderFactory() {
+ // Hardware H264 encoding only supported on iOS for now.
+-#if defined(WEBRTC_IOS)
+ // TODO(magjed): Push Constrained High profile as well when negotiation is
+ // ready, http://crbug/webrtc/6337.
+ cricket::VideoCodec constrained_baseline(cricket::kH264CodecName);
+@@ -52,29 +49,24 @@ VideoToolboxVideoEncoderFactory::VideoToolboxVideoEncoderFactory() {
+ constrained_baseline.SetParam(cricket::kH264FmtpLevelAsymmetryAllowed, "1");
+ constrained_baseline.SetParam(cricket::kH264FmtpPacketizationMode, "1");
+ supported_codecs_.push_back(constrained_baseline);
+-#endif
+ }
+
+ VideoToolboxVideoEncoderFactory::~VideoToolboxVideoEncoderFactory() {}
+
+ VideoEncoder* VideoToolboxVideoEncoderFactory::CreateVideoEncoder(
+ const cricket::VideoCodec& codec) {
+-#if defined(WEBRTC_IOS)
+ if (FindMatchingCodec(supported_codecs_, codec)) {
+ LOG(LS_INFO) << "Creating HW encoder for " << codec.name;
+ return new H264VideoToolboxEncoder(codec);
+ }
+-#endif
+ LOG(LS_INFO) << "No HW encoder found for codec " << codec.name;
+ return nullptr;
+ }
+
+ void VideoToolboxVideoEncoderFactory::DestroyVideoEncoder(
+ VideoEncoder* encoder) {
+-#if defined(WEBRTC_IOS)
+ delete encoder;
+ encoder = nullptr;
+-#endif
+ }
+
+ const std::vector<cricket::VideoCodec>&
+@@ -85,9 +77,7 @@ VideoToolboxVideoEncoderFactory::supported_codecs() const {
+ // VideoToolboxVideoDecoderFactory
+
+ VideoToolboxVideoDecoderFactory::VideoToolboxVideoDecoderFactory() {
+-#if defined(WEBRTC_IOS)
+ supported_codecs_.push_back(cricket::VideoCodec("H264"));
+-#endif
+ }
+
+ VideoToolboxVideoDecoderFactory::~VideoToolboxVideoDecoderFactory() {}
+@@ -95,22 +85,18 @@ VideoToolboxVideoDecoderFactory::~VideoToolboxVideoDecoderFactory() {}
+ VideoDecoder* VideoToolboxVideoDecoderFactory::CreateVideoDecoder(
+ VideoCodecType type) {
+ const auto codec = cricket::VideoCodec(NameFromCodecType(type));
+-#if defined(WEBRTC_IOS)
+ if (FindMatchingCodec(supported_codecs_, codec)) {
+ LOG(LS_INFO) << "Creating HW decoder for " << codec.name;
+ return new H264VideoToolboxDecoder();
+ }
+-#endif
+ LOG(LS_INFO) << "No HW decoder found for codec " << codec.name;
+ return nullptr;
+ }
+
+ void VideoToolboxVideoDecoderFactory::DestroyVideoDecoder(
+ VideoDecoder* decoder) {
+-#if defined(WEBRTC_IOS)
+ delete decoder;
+ decoder = nullptr;
+-#endif
+ }
+
+ } // namespace webrtc
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/stringize_macros.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/stringize_macros.h
+index 9c8e7e9..06926c4 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/stringize_macros.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/stringize_macros.h
+@@ -33,6 +33,8 @@
+ // Then:
+ // STRINGIZE(A) produces "FOO"
+ // STRINGIZE(B(y)) produces "myobj->FunctionCall(y)"
++#ifndef STRINGIZE
+ #define STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
++#endif
+
+ #endif // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_STRINGIZE_MACROS_H_
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/typedefs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/typedefs.h
+index 98b32f7..d5f4afa 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/typedefs.h
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/typedefs.h
+@@ -99,13 +99,17 @@
+ #endif
+ #endif
+
++#ifndef NO_RETURN
+ // Annotate a function that will not return control flow to the caller.
+ #if defined(_MSC_VER)
+ #define NO_RETURN __declspec(noreturn)
+ #elif defined(__GNUC__)
++#ifndef NO_RETURN
+ #define NO_RETURN __attribute__ ((__noreturn__))
++#endif
+ #else
+ #define NO_RETURN
+ #endif
++#endif // NO_RETURN
+
+ #endif // WEBRTC_TYPEDEFS_H_
+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_encoder.cc
+index f4e81565..5b5c6c2 100644
+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_encoder.cc
++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_encoder.cc
+@@ -39,7 +39,7 @@ bool VideoEncoder::IsSupportedSoftware(EncoderType codec_type) {
+ case kH264:
+ return H264Encoder::IsSupported();
+ case kVp8:
+- return true;
++ return VP8Encoder::IsSupported();
+ case kVp9:
+ return VP9Encoder::IsSupported();
+ case kUnsupportedCodec:
</ins></span></pre>
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