Attachment 348870 Details for Bug 189270 – Patch for landing

[patch] Patch for landing

bug-189270-20180904163151.patch (text/plain), 21.32 KB, created by youenn fablet on 2018-09-04 16:31:51 PDT

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Creator: youenn fablet

Created: 2018-09-04 16:31:51 PDT

Size: 21.32 KB

patch

obsolete

>Subversion Revision: 235645
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 89382338d8c61f6cc9a690473bb70e81f99bbe41..2e2dd3441ee2c589329d6e598828bee605853dc5 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,51 @@
>+2018-09-04  Youenn Fablet  <youenn@apple.com>
>+
>+        Make LibWebRTCRtpSenderBackend own its libwebrtc RTP sender backend
>+        https://bugs.webkit.org/show_bug.cgi?id=189270
>+
>+        Reviewed by Eric Carlson.
>+
>+        Previously, the libwebrtc senders were owned in LibWebRTCMediaEndpoint.
>+        This patch makes them be owned by LibWebRTCRtpSenderBackend.
>+        This simplifies the implementation of RTCRtpSender::getParameters.
>+        This will help simplify the implementation of RTCRtpSender::replaceTrack.
>+
>+        removeTrack is also refactored so that LibWebRTCMediaEndpoint directly uses libwebrtc constructs and not WebCore libwebrtc-agnostic ones.
>+
>+        No change of behavior.
>+
>+        * Modules/mediastream/PeerConnectionBackend.h:
>+        (WebCore::PeerConnectionBackend::removeTrack):
>+        (WebCore::PeerConnectionBackend::notifyRemovedTrack): Deleted.
>+        * Modules/mediastream/RTCPeerConnection.cpp:
>+        (WebCore::RTCPeerConnection::removeTrack):
>+        * Modules/mediastream/RTCRtpSender.cpp:
>+        (WebCore::RTCRtpSender::getParameters):
>+        * Modules/mediastream/RTCRtpSender.h:
>+        (WebCore::RTCRtpSender::backend):
>+        * Modules/mediastream/RTCRtpSenderBackend.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
>+        (WebCore::LibWebRTCMediaEndpoint::addTrack):
>+        (WebCore::LibWebRTCMediaEndpoint::removeTrack):
>+        (WebCore::LibWebRTCMediaEndpoint::doCreateOffer):
>+        (WebCore::LibWebRTCMediaEndpoint::stop):
>+        (WebCore::LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveAudio const): Deleted.
>+        (WebCore::LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveVideo const): Deleted.
>+        (WebCore::LibWebRTCMediaEndpoint::getRTCRtpSenderParameters): Deleted.
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
>+        (WebCore::backendFromRTPSender):
>+        (WebCore::LibWebRTCPeerConnectionBackend::addTrack):
>+        (WebCore::LibWebRTCPeerConnectionBackend::addTransceiver):
>+        (WebCore::LibWebRTCPeerConnectionBackend::removeTrack):
>+        (WebCore::LibWebRTCPeerConnectionBackend::shouldOfferAllowToReceive const):
>+        (WebCore::LibWebRTCPeerConnectionBackend::notifyRemovedTrack): Deleted.
>+        (WebCore::LibWebRTCPeerConnectionBackend::getParameters const): Deleted.
>+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp:
>+        (WebCore::LibWebRTCRtpSenderBackend::getParameters const):
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h:
>+
> 2018-09-04  Simon Fraser  <simon.fraser@apple.com>
> 
>         CSS reference filter that references a tiled feTurbulence is blank
>diff --git a/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h b/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h
>index 95e0f9932cae0e5af4f7b294e38dd4195064fc82..6292de8613ace3c05b58b5abf9b65102717857ad 100644
>--- a/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h
>+++ b/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h
>@@ -102,7 +102,7 @@ public:
> 
>     virtual Ref<RTCRtpReceiver> createReceiver(const String& transceiverMid, const String& trackKind, const String& trackId) = 0;
>     virtual ExceptionOr<Ref<RTCRtpSender>> addTrack(RTCRtpSender*, MediaStreamTrack&, const Vector<String>&);
>-    virtual void notifyRemovedTrack(RTCRtpSender&) { }
>+    virtual void removeTrack(RTCRtpSender&) { }
> 
>     virtual ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(const String&, const RTCRtpTransceiverInit&);
>     virtual ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(Ref<MediaStreamTrack>&&, const RTCRtpTransceiverInit&);
>diff --git a/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp b/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>index 31e69e216b49e44443d22b2b4c41ca18aeceebe8..2a5a13e0a088688bdb2ecd3a4d802b6971800e51 100644
>--- a/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>+++ b/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>@@ -157,9 +157,8 @@ ExceptionOr<void> RTCPeerConnection::removeTrack(RTCRtpSender& sender)
>     if (shouldAbort)
>         return { };
> 
>+    m_backend->removeTrack(sender);
>     sender.stop();
>-
>-    m_backend->notifyRemovedTrack(sender);
>     return { };
> }
> 
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp b/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>index 6a3d1bcd90886e033d5124dc9d4eb4b6dfdfb74c..5acdd466f4934049c3b497f03fccd7966647bfe1 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>@@ -96,7 +96,7 @@ RTCRtpParameters RTCRtpSender::getParameters()
> {
>     if (isStopped())
>         return { };
>-    return m_backend->getParameters(*this);
>+    return m_backend->getParameters();
> }
> 
> } // namespace WebCore
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpSender.h b/Source/WebCore/Modules/mediastream/RTCRtpSender.h
>index 4e1e4b5e3a7fa5f7d17c0df73496f6ead20ef1ef..a5d30924e3a03ba741eef8985e0e7503bbbd9a91 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpSender.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpSender.h
>@@ -57,6 +57,7 @@ public:
>     void replaceTrack(RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&);
> 
>     RTCRtpParameters getParameters();
>+    RTCRtpSenderBackend* backend() { return m_backend.get(); }
> 
> private:
>     RTCRtpSender(String&& trackKind, Vector<String>&& mediaStreamIds, std::unique_ptr<RTCRtpSenderBackend>&&);
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpSenderBackend.h b/Source/WebCore/Modules/mediastream/RTCRtpSenderBackend.h
>index 5d258a4c3e9cf491538264868166172ffc40a4d5..476543c98c6fa6ca465015c1bc58fcee7a97a431 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpSenderBackend.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpSenderBackend.h
>@@ -37,7 +37,7 @@ class RTCRtpSender;
> class RTCRtpSenderBackend {
> public:
>     virtual void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) = 0;
>-    virtual RTCRtpParameters getParameters(RTCRtpSender&) const = 0;
>+    virtual RTCRtpParameters getParameters() const = 0;
>     virtual ~RTCRtpSenderBackend() = default;
> };
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>index b5ebb7d356c82945ee824191c157a4c96ec10dee..98cbee5c5a3450f260d6260eb7e4199e4848af63 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>@@ -194,9 +194,10 @@ void LibWebRTCMediaEndpoint::doSetRemoteDescription(RTCSessionDescription& descr
>     startLoggingStats();
> }
> 
>-bool LibWebRTCMediaEndpoint::addTrack(RTCRtpSender& sender, MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
>+bool LibWebRTCMediaEndpoint::addTrack(LibWebRTCRtpSenderBackend& sender, MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
> {
>     ASSERT(m_backend);
>+    ASSERT(!sender.rtcSender());
> 
>     if (!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled()) {
>         String mediaStreamId = mediaStreamIds.isEmpty() ? createCanonicalUUIDString() : mediaStreamIds[0];
>@@ -217,18 +218,20 @@ bool LibWebRTCMediaEndpoint::addTrack(RTCRtpSender& sender, MediaStreamTrack& tr
>         auto audioTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), trackSource.ptr());
>         m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
>         auto rtpSender = m_backend->AddTrack(audioTrack.get(), WTFMove(ids));
>-        if (rtpSender.ok())
>-            m_senders.add(&sender, rtpSender.MoveValue());
>-        return rtpSender.ok();
>+        if (!rtpSender.ok())
>+            return false;
>+        sender.setRTCSender(rtpSender.MoveValue());
>+        return true;
>     }
>     case RealtimeMediaSource::Type::Video: {
>         auto videoSource = RealtimeOutgoingVideoSource::create(track.privateTrack());
>         auto videoTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
>         m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
>         auto rtpSender = m_backend->AddTrack(videoTrack.get(), WTFMove(ids));
>-        if (rtpSender.ok())
>-            m_senders.add(&sender, rtpSender.MoveValue());
>-        return rtpSender.ok();
>+        if (!rtpSender.ok())
>+            return false;
>+        sender.setRTCSender(rtpSender.MoveValue());
>+        return true;
>     }
>     case RealtimeMediaSource::Type::None:
>         ASSERT_NOT_REACHED();
>@@ -236,46 +239,10 @@ bool LibWebRTCMediaEndpoint::addTrack(RTCRtpSender& sender, MediaStreamTrack& tr
>     return false;
> }
> 
>-void LibWebRTCMediaEndpoint::removeTrack(RTCRtpSender& sender)
>+void LibWebRTCMediaEndpoint::removeTrack(LibWebRTCRtpSenderBackend& sender)
> {
>     ASSERT(m_backend);
>-
>-    auto rtcSender = m_senders.get(&sender);
>-    if (!rtcSender)
>-        return;
>-    m_backend->RemoveTrack(rtcSender.get());
>-}
>-
>-bool LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveAudio() const
>-{
>-    ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled());
>-    for (const auto& transceiver : m_peerConnectionBackend.connection().getTransceivers()) {
>-        if (transceiver->sender().trackKind() != "audio")
>-            continue;
>-
>-        if (transceiver->direction() == RTCRtpTransceiverDirection::Recvonly)
>-            return true;
>-
>-        if (transceiver->direction() == RTCRtpTransceiverDirection::Sendrecv && !m_senders.contains(&transceiver->sender()))
>-            return true;
>-    }
>-    return false;
>-}
>-
>-bool LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveVideo() const
>-{
>-    ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled());
>-    for (const auto& transceiver : m_peerConnectionBackend.connection().getTransceivers()) {
>-        if (transceiver->sender().trackKind() != "video")
>-            continue;
>-
>-        if (transceiver->direction() == RTCRtpTransceiverDirection::Recvonly)
>-            return true;
>-
>-        if (transceiver->direction() == RTCRtpTransceiverDirection::Sendrecv && !m_senders.contains(&transceiver->sender()))
>-            return true;
>-    }
>-    return false;
>+    m_backend->RemoveTrack(sender.rtcSender());
> }
> 
> void LibWebRTCMediaEndpoint::doCreateOffer(const RTCOfferOptions& options)
>@@ -288,9 +255,9 @@ void LibWebRTCMediaEndpoint::doCreateOffer(const RTCOfferOptions& options)
>     rtcOptions.voice_activity_detection = options.voiceActivityDetection;
> 
>     if (!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled()) {
>-        if (shouldOfferAllowToReceiveAudio())
>+        if (m_peerConnectionBackend.shouldOfferAllowToReceive("audio"_s))
>             rtcOptions.offer_to_receive_audio = webrtc::PeerConnectionInterface::RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>-        if (shouldOfferAllowToReceiveVideo())
>+        if (m_peerConnectionBackend.shouldOfferAllowToReceive("video"_s))
>             rtcOptions.offer_to_receive_video = webrtc::PeerConnectionInterface::RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>     }
>     m_backend->CreateOffer(&m_createSessionDescriptionObserver, rtcOptions);
>@@ -470,7 +437,6 @@ void LibWebRTCMediaEndpoint::stop()
>     m_backend->Close();
>     m_backend = nullptr;
>     m_streams.clear();
>-    m_senders.clear();
> }
> 
> void LibWebRTCMediaEndpoint::OnRenegotiationNeeded()
>@@ -614,14 +580,6 @@ void LibWebRTCMediaEndpoint::setRemoteSessionDescriptionFailed(const std::string
>     });
> }
> 
>-RTCRtpParameters LibWebRTCMediaEndpoint::getRTCRtpSenderParameters(RTCRtpSender& sender)
>-{
>-    auto rtcSender = m_senders.get(&sender);
>-    if (!rtcSender)
>-        return { };
>-    return fillRtpParameters(rtcSender->GetParameters());
>-}
>-
> void LibWebRTCMediaEndpoint::gatherStatsForLogging()
> {
>     LibWebRTCProvider::callOnWebRTCSignalingThread([protectedThis = makeRef(*this)] {
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>index b717495f18c5cb3c78c22bfab9eadeee4eaf73d8..f2ba6bc525a32ef42c448208f136c06743e0abc6 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>@@ -61,6 +61,7 @@ namespace WebCore {
> 
> class LibWebRTCProvider;
> class LibWebRTCPeerConnectionBackend;
>+class LibWebRTCRtpSenderBackend;
> class MediaStreamTrack;
> class RTCSessionDescription;
> 
>@@ -97,9 +98,8 @@ public:
>     RefPtr<RTCSessionDescription> pendingLocalDescription() const;
>     RefPtr<RTCSessionDescription> pendingRemoteDescription() const;
> 
>-    bool addTrack(RTCRtpSender&, MediaStreamTrack&, const Vector<String>&);
>-    void removeTrack(RTCRtpSender&);
>-    RTCRtpParameters getRTCRtpSenderParameters(RTCRtpSender&);
>+    bool addTrack(LibWebRTCRtpSenderBackend&, MediaStreamTrack&, const Vector<String>&);
>+    void removeTrack(LibWebRTCRtpSenderBackend&);
> 
> private:
>     LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend&, LibWebRTCProvider&);
>@@ -142,9 +142,6 @@ private:
>         : rtc::RefCountReleaseStatus::kDroppedLastRef;
>     }
> 
>-    bool shouldOfferAllowToReceiveAudio() const;
>-    bool shouldOfferAllowToReceiveVideo() const;
>-
> #if !RELEASE_LOG_DISABLED
>     const Logger& logger() const final { return m_logger.get(); }
>     const void* logIdentifier() const final { return m_logIdentifier; }
>@@ -167,7 +164,6 @@ private:
>     SetRemoteSessionDescriptionObserver<LibWebRTCMediaEndpoint> m_setRemoteSessionDescriptionObserver;
> 
>     HashMap<webrtc::MediaStreamInterface*, RefPtr<MediaStream>> m_streams;
>-    HashMap<RTCRtpSender*, rtc::scoped_refptr<webrtc::RtpSenderInterface>> m_senders;
> 
>     bool m_isInitiator { false };
>     Timer m_statsLogTimer;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>index c7c0644b747c657d44ca30cd80752ec58c7a0dd6..461eb5ad2b46f387c38c81d18e2404989a535f78 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>@@ -43,6 +43,7 @@
> #include "RealtimeIncomingVideoSource.h"
> #include "RealtimeOutgoingAudioSource.h"
> #include "RealtimeOutgoingVideoSource.h"
>+#include "RuntimeEnabledFeatures.h"
> 
> namespace WebCore {
> 
>@@ -331,6 +332,11 @@ RefPtr<RTCSessionDescription> LibWebRTCPeerConnectionBackend::remoteDescription(
>     return m_endpoint->remoteDescription();
> }
> 
>+static inline LibWebRTCRtpSenderBackend& backendFromRTPSender(RTCRtpSender& sender)
>+{
>+    return static_cast<LibWebRTCRtpSenderBackend&>(*sender.backend());
>+}
>+
> ExceptionOr<Ref<RTCRtpSender>> LibWebRTCPeerConnectionBackend::addTrack(RTCRtpSender* sender, MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
> {
>     if (!sender) {
>@@ -351,7 +357,7 @@ ExceptionOr<Ref<RTCRtpSender>> LibWebRTCPeerConnectionBackend::addTrack(RTCRtpSe
>         m_peerConnection.addInternalTransceiver(WTFMove(transceiver));
>     }
> 
>-    if (!m_endpoint->addTrack(*sender, track, mediaStreamIds))
>+    if (!m_endpoint->addTrack(backendFromRTPSender(*sender), track, mediaStreamIds))
>         return Exception { TypeError, "Unable to add track"_s };
> 
>     return makeRef(*sender);
>@@ -367,8 +373,9 @@ ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addTransceiv
> ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addTransceiver(Ref<MediaStreamTrack>&& track, const RTCRtpTransceiverInit& init)
> {
>     auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
>+    auto& backend = *senderBackend;
>     auto sender = RTCRtpSender::create(track.copyRef(), Vector<String>(), WTFMove(senderBackend));
>-    if (!m_endpoint->addTrack(sender.get(), track, Vector<String> { }))
>+    if (!m_endpoint->addTrack(backend, track, Vector<String> { }))
>         return Exception { InvalidAccessError, "Unable to add track"_s };
> 
>     return completeAddTransceiver(WTFMove(sender), init, track->id(), track->kind());
>@@ -386,9 +393,9 @@ Ref<RTCRtpTransceiver> LibWebRTCPeerConnectionBackend::completeAddTransceiver(Re
>     return transceiver;
> }
> 
>-void LibWebRTCPeerConnectionBackend::notifyRemovedTrack(RTCRtpSender& sender)
>+void LibWebRTCPeerConnectionBackend::removeTrack(RTCRtpSender& sender)
> {
>-    m_endpoint->removeTrack(sender);
>+    m_endpoint->removeTrack(backendFromRTPSender(sender));
> }
> 
> template<typename Source>
>@@ -444,17 +451,32 @@ void LibWebRTCPeerConnectionBackend::replaceTrack(RTCRtpSender& sender, RefPtr<M
>     }
> }
> 
>-RTCRtpParameters LibWebRTCPeerConnectionBackend::getParameters(RTCRtpSender& sender) const
>-{
>-    return m_endpoint->getRTCRtpSenderParameters(sender);
>-}
>-
> void LibWebRTCPeerConnectionBackend::applyRotationForOutgoingVideoSources()
> {
>     for (auto& source : m_videoSources)
>         source->setApplyRotation(true);
> }
> 
>+bool LibWebRTCPeerConnectionBackend::shouldOfferAllowToReceive(const char* kind) const
>+{
>+    ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled());
>+    for (const auto& transceiver : m_peerConnection.getTransceivers()) {
>+        if (transceiver->sender().trackKind() != kind)
>+            continue;
>+
>+        if (transceiver->direction() == RTCRtpTransceiverDirection::Recvonly)
>+            return true;
>+
>+        if (transceiver->direction() != RTCRtpTransceiverDirection::Sendrecv)
>+            continue;
>+
>+        auto& backend = static_cast<LibWebRTCRtpSenderBackend&>(*transceiver->sender().backend());
>+        if (!backend.rtcSender())
>+            return true;
>+    }
>+    return false;
>+}
>+
> } // namespace WebCore
> 
> #endif // USE(LIBWEBRTC)
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>index d10bb1a6545093330be9b6f724eb9af618dc62c1..b55f0cf2d49b855811e87f6893d1c59cfd1505f7 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>@@ -56,7 +56,8 @@ public:
>     bool hasVideoSources() const { return m_videoSources.size(); }
> 
>     void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&);
>-    RTCRtpParameters getParameters(RTCRtpSender&) const;
>+
>+    bool shouldOfferAllowToReceive(const char*) const;
> 
> private:
>     void doCreateOffer(RTCOfferOptions&&) final;
>@@ -90,7 +91,7 @@ private:
>     void getStatsSucceeded(const DeferredPromise&, Ref<RTCStatsReport>&&);
> 
>     ExceptionOr<Ref<RTCRtpSender>> addTrack(RTCRtpSender*, MediaStreamTrack&, const Vector<String>&) final;
>-    void notifyRemovedTrack(RTCRtpSender&) final;
>+    void removeTrack(RTCRtpSender&) final;
> 
>     ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(const String&, const RTCRtpTransceiverInit&) final;
>     ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(Ref<MediaStreamTrack>&&, const RTCRtpTransceiverInit&) final;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>index 36863a942c8ed33699b081246cd67d6886cfec1f..2b2735c274b652b4b882959fe7d1a2da074ba0d1 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>@@ -49,14 +49,11 @@ void LibWebRTCRtpSenderBackend::replaceTrack(RTCRtpSender& sender, RefPtr<MediaS
>     m_peerConnectionBackend->replaceTrack(sender, WTFMove(track), WTFMove(promise));
> }
> 
>-RTCRtpParameters LibWebRTCRtpSenderBackend::getParameters(RTCRtpSender& sender) const
>+RTCRtpParameters LibWebRTCRtpSenderBackend::getParameters() const
> {
>-    if (!m_rtcSender) {
>-        // FIXME: We should not need to go through the backend and should directly have m_rtcSender here.
>-        if (!m_peerConnectionBackend)
>-            return { };
>-        return m_peerConnectionBackend->getParameters(sender);
>-    }
>+    if (!m_rtcSender)
>+        return { };
>+
>     return fillRtpParameters(m_rtcSender->GetParameters());
> }
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>index 3fd331a89abed2e135f540910d1a97fd60afd582..24b1b1bc7d466481d81f74665d9c77625201718f 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>@@ -50,10 +50,12 @@ public:
>     {
>     }
> 
>-private:
>     void setRTCSender(rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender) { m_rtcSender = WTFMove(rtcSender); }
>+    webrtc::RtpSenderInterface* rtcSender() { return m_rtcSender.get(); }
>+
>+private:
>     void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) final;
>-    RTCRtpParameters getParameters(RTCRtpSender&) const final;
>+    RTCRtpParameters getParameters() const final;
> 
>     WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
>     rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;

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Attachments on bug 189270: 348857 | 348870