Attachment 349154 Details for Bug 189390 – Patch

[patch] Patch

bug-189390-20180907102704.patch (text/plain), 96.26 KB, created by youenn fablet on 2018-09-07 10:27:04 PDT

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Creator: youenn fablet

Created: 2018-09-07 10:27:04 PDT

Size: 96.26 KB

patch

obsolete

>Subversion Revision: 235784
>diff --git a/Source/ThirdParty/libwebrtc/ChangeLog b/Source/ThirdParty/libwebrtc/ChangeLog
>index ddec699540f9f5997ef48bc9655b54a53aa7990b..4912e6174ffc360ec8ea491369a5a8d683f14994 100644
>--- a/Source/ThirdParty/libwebrtc/ChangeLog
>+++ b/Source/ThirdParty/libwebrtc/ChangeLog
>@@ -1,3 +1,15 @@
>+2018-09-07  Youenn Fablet  <youenn@apple.com>
>+
>+        Add support for unified plan transceivers
>+        https://bugs.webkit.org/show_bug.cgi?id=189390
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Expose more symbols.
>+        * Configurations/libwebrtc.iOS.exp:
>+        * Configurations/libwebrtc.iOSsim.exp:
>+        * Configurations/libwebrtc.mac.exp:
>+
> 2018-09-05  Youenn Fablet  <youenn@apple.com>
> 
>         Expose RTCRtpSender.setParameters
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 68c6ab7b8e58805e4223fe6c4e009ddb9532807e..5d9204a2107921aacd57d77afc0dae23920ea997 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,83 @@
>+2018-09-07  Youenn Fablet  <youenn@apple.com>
>+
>+        Add support for unified plan transceivers
>+        https://bugs.webkit.org/show_bug.cgi?id=189390
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Keep previous transceiver behavior when unified plan flag is off.
>+        Otherwise, use the libwebrtc transceiver API to create and use unified plan transceivers.
>+        Fuel the implementation of transceivers through a dedicated backend.
>+
>+        Update transceiver IDL and make some smaller fixes at the same time:
>+        - Make sure remote sources have a proper name as per https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpreceiver.
>+        - Add support for transceiver.currentDirection.
>+
>+        Our mock peer connections are only supporting plan B APIs at the moment.
>+        We therefore mandate plan B when using such mocks until we can upgrade mocks to support unified plan APIs.
>+
>+        Covered by modified and rebased tests.
>+
>+        * Modules/mediastream/PeerConnectionBackend.cpp:
>+        (WebCore::PeerConnectionBackend::addTrack):
>+        * Modules/mediastream/PeerConnectionBackend.h:
>+        * Modules/mediastream/RTCPeerConnection.cpp:
>+        (WebCore::RTCPeerConnection::addTrack):
>+        (WebCore::RTCPeerConnection::addTransceiver):
>+        * Modules/mediastream/RTCRtpSender.cpp:
>+        (WebCore::RTCRtpSender::RTCRtpSender):
>+        * Modules/mediastream/RTCRtpTransceiver.cpp:
>+        (WebCore::RTCRtpTransceiver::mid const):
>+        (WebCore::RTCRtpTransceiver::currentDirection const):
>+        * Modules/mediastream/RTCRtpTransceiver.h:
>+        (WebCore::RTCRtpTransceiver::backend):
>+        * Modules/mediastream/RTCRtpTransceiver.idl:
>+        * Modules/mediastream/RTCRtpTransceiverBackend.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
>+        (WebCore::LibWebRTCMediaEndpoint::addTrack):
>+        (WebCore::LibWebRTCMediaEndpoint::newTransceiver):
>+        (WebCore::LibWebRTCMediaEndpoint::addTransceiver):
>+        (WebCore::LibWebRTCMediaEndpoint::transceiverBackendFromSender):
>+        (WebCore::LibWebRTCMediaEndpoint::OnAddTrack):
>+        (WebCore::LibWebRTCMediaEndpoint::OnTrack):
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
>+        (WebCore::createReceiverForSource):
>+        (WebCore::LibWebRTCPeerConnectionBackend::createReceiver):
>+        (WebCore::LibWebRTCPeerConnectionBackend::videoReceiver):
>+        (WebCore::LibWebRTCPeerConnectionBackend::audioReceiver):
>+        (WebCore::LibWebRTCPeerConnectionBackend::addTrack):
>+        (WebCore::LibWebRTCPeerConnectionBackend::addTransceiver):
>+        (WebCore::backendFromRTPTransceiver):
>+        (WebCore::LibWebRTCPeerConnectionBackend::existingTransceiver):
>+        (WebCore::LibWebRTCPeerConnectionBackend::newTransceiver):
>+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp:
>+        (WebCore::LibWebRTCRtpSenderBackend::replaceTrack):
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp: Added.
>+        (WebCore::LibWebRTCRtpTransceiverBackend::createReceiverBackend):
>+        (WebCore::LibWebRTCRtpTransceiverBackend::createSenderBackend):
>+        (WebCore::LibWebRTCRtpTransceiverBackend::direction const):
>+        (WebCore::LibWebRTCRtpTransceiverBackend::currentDirection const):
>+        (WebCore::LibWebRTCRtpTransceiverBackend::setDirection):
>+        (WebCore::LibWebRTCRtpTransceiverBackend::mid):
>+        (WebCore::LibWebRTCRtpTransceiverBackend::stop):
>+        * Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.h: Added.
>+        * Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp:
>+        (WebCore::toRTCRtpTransceiverDirection):
>+        (WebCore::fromRTCRtpTransceiverDirection):
>+        (WebCore::fromRtpTransceiverInit):
>+        * Modules/mediastream/libwebrtc/LibWebRTCUtils.h:
>+        * WebCore.xcodeproj/project.pbxproj:
>+        * platform/mediastream/RealtimeIncomingAudioSource.cpp:
>+        (WebCore::RealtimeIncomingAudioSource::RealtimeIncomingAudioSource):
>+        * platform/mediastream/RealtimeIncomingVideoSource.cpp:
>+        (WebCore::RealtimeIncomingVideoSource::RealtimeIncomingVideoSource):
>+        * platform/mediastream/RealtimeMediaSource.h:
>+        * testing/Internals.cpp:
>+        (WebCore::Internals::useMockRTCPeerConnectionFactory):
>+
> 2018-09-06  Ryosuke Niwa  <rniwa@webkit.org>
> 
>         ShadowRoot should have its own node flag
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>index d64cdf08b41b39a57afdb13fb8a78716a650a52d..76497e3c57c1733d166ecaf0750f9ee10024b4aa 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>@@ -212,3 +212,5 @@ __ZN6webrtc12RtpExtensionD1Ev
> __ZN6webrtc21RtpEncodingParametersC1ERKS0_
> __ZN6webrtc21RtpEncodingParametersC1Ev
> __ZN6webrtc21RtpEncodingParametersD1Ev
>+__ZN6webrtc18RtpTransceiverInitC1Ev
>+__ZN6webrtc18RtpTransceiverInitD1Ev
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>index 4fc1c4abd7a524b70ce8d50a333c180c63f6d4ff..1e713ce54b8a7d852a43292b8753cc745e9312c4 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>@@ -213,3 +213,6 @@ __ZN6webrtc12RtpExtensionD1Ev
> __ZN6webrtc21RtpEncodingParametersC1ERKS0_
> __ZN6webrtc21RtpEncodingParametersC1Ev
> __ZN6webrtc21RtpEncodingParametersD1Ev
>+
>+__ZN6webrtc18RtpTransceiverInitC1Ev
>+__ZN6webrtc18RtpTransceiverInitD1Ev
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>index 4fc1c4abd7a524b70ce8d50a333c180c63f6d4ff..c2e741b36a79a94e9199f13c16344ed853f03b0c 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>@@ -213,3 +213,5 @@ __ZN6webrtc12RtpExtensionD1Ev
> __ZN6webrtc21RtpEncodingParametersC1ERKS0_
> __ZN6webrtc21RtpEncodingParametersC1Ev
> __ZN6webrtc21RtpEncodingParametersD1Ev
>+__ZN6webrtc18RtpTransceiverInitC1Ev
>+__ZN6webrtc18RtpTransceiverInitD1Ev
>diff --git a/Source/WebCore/CMakeLists.txt b/Source/WebCore/CMakeLists.txt
>index ec77463ef1fe6ba471f46b28bddcb97f38c0972a..e2fa2d92d3fba7fec029764aee4a52852b082347 100644
>--- a/Source/WebCore/CMakeLists.txt
>+++ b/Source/WebCore/CMakeLists.txt
>@@ -1572,6 +1572,7 @@ if (USE_LIBWEBRTC)
>       Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>       Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverBackend.cpp
>       Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>+      Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp
>       Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
>       )
> endif ()
>diff --git a/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp b/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp
>index 333113f20e3c657b433b159a67e1af4284f1c707..2f174180cb9c5aea128f8c5cf1d9a7b3cee623c0 100644
>--- a/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp
>@@ -530,7 +530,7 @@ void PeerConnectionBackend::markAsNeedingNegotiation()
>         m_peerConnection.scheduleNegotiationNeededEvent();
> }
> 
>-ExceptionOr<Ref<RTCRtpSender>> PeerConnectionBackend::addTrack(RTCRtpSender*, MediaStreamTrack&, const Vector<String>&)
>+ExceptionOr<Ref<RTCRtpSender>> PeerConnectionBackend::addTrack(MediaStreamTrack&, Vector<String>&&)
> {
>     return Exception { NotSupportedError, "Not implemented"_s };
> }
>diff --git a/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h b/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h
>index bc6d2f6f3b95f44dff2b64ae2bb24c159668be1c..284a64a15a0c0fcef895237a73e8a9eb1cb0fba6 100644
>--- a/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h
>+++ b/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h
>@@ -100,7 +100,7 @@ public:
> 
>     virtual void getStats(MediaStreamTrack*, Ref<DeferredPromise>&&) = 0;
> 
>-    virtual ExceptionOr<Ref<RTCRtpSender>> addTrack(RTCRtpSender*, MediaStreamTrack&, const Vector<String>&);
>+    virtual ExceptionOr<Ref<RTCRtpSender>> addTrack(MediaStreamTrack&, Vector<String>&&);
>     virtual void removeTrack(RTCRtpSender&) { }
> 
>     virtual ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(const String&, const RTCRtpTransceiverInit&);
>diff --git a/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp b/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>index 4ad4f7cbf6a7749cebdf72e741a041152beec2ed..566fa3314fa676404fd907ff7bc3d802d2b6815f 100644
>--- a/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>+++ b/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>@@ -122,22 +122,7 @@ ExceptionOr<Ref<RTCRtpSender>> RTCPeerConnection::addTrack(Ref<MediaStreamTrack>
>     for (auto stream : streams)
>         mediaStreamIds.append(stream.get().id());
> 
>-    RTCRtpSender* sender = nullptr;
>-
>-    // Reuse an existing sender with the same track kind if it has never been used to send before.
>-    for (auto& transceiver : m_transceiverSet->list()) {
>-        auto& existingSender = transceiver->sender();
>-        if (existingSender.trackKind() == track->kind() && existingSender.trackId().isNull() && !transceiver->hasSendingDirection()) {
>-            existingSender.setTrack(track.copyRef());
>-            existingSender.setMediaStreamIds(WTFMove(mediaStreamIds));
>-            transceiver->enableSendingDirection();
>-            sender = &existingSender;
>-            
>-            break;
>-        }
>-    }
>-
>-    return m_backend->addTrack(sender, track.get(), mediaStreamIds);
>+    return m_backend->addTrack(track.get(), WTFMove(mediaStreamIds));
> }
> 
> ExceptionOr<void> RTCPeerConnection::removeTrack(RTCRtpSender& sender)
>@@ -171,9 +156,15 @@ ExceptionOr<Ref<RTCRtpTransceiver>> RTCPeerConnection::addTransceiver(AddTransce
>         if (kind != "audio" && kind != "video")
>             return Exception { TypeError };
> 
>+        if (isClosed())
>+            return Exception { InvalidStateError };
>+
>         return m_backend->addTransceiver(kind, init);
>     }
> 
>+    if (isClosed())
>+        return Exception { InvalidStateError };
>+
>     auto track = WTF::get<RefPtr<MediaStreamTrack>>(withTrack).releaseNonNull();
>     return m_backend->addTransceiver(WTFMove(track), init);
> }
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h b/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h
>index db6b7498b9e083919abb57af71acb4fb64de5f41..ac3da1058f98734da8537e2a80139bd289b34f2a 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h
>@@ -41,7 +41,7 @@ namespace WebCore {
> 
> class RTCRtpReceiver : public RTCRtpSenderReceiverBase {
> public:
>-    static Ref<RTCRtpReceiver> create(Ref<MediaStreamTrack>&& track, std::unique_ptr<RTCRtpReceiverBackend>&& backend = nullptr)
>+    static Ref<RTCRtpReceiver> create(Ref<MediaStreamTrack>&& track, std::unique_ptr<RTCRtpReceiverBackend>&& backend)
>     {
>         return adoptRef(*new RTCRtpReceiver(WTFMove(track), WTFMove(backend)));
>     }
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp b/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>index 75839e19e52a0550525e0cd0d586a761ceb22f36..3a321f6cb510fa5f325bd59d75f3cbd3efcd7e77 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp
>@@ -33,6 +33,8 @@
> 
> #if ENABLE(WEB_RTC)
> 
>+#include "RuntimeEnabledFeatures.h"
>+
> namespace WebCore {
> 
> Ref<RTCRtpSender> RTCRtpSender::create(Ref<MediaStreamTrack>&& track, Vector<String>&& mediaStreamIds, std::unique_ptr<RTCRtpSenderBackend>&& backend)
>@@ -53,6 +55,7 @@ RTCRtpSender::RTCRtpSender(String&& trackKind, Vector<String>&& mediaStreamIds,
>     , m_mediaStreamIds(WTFMove(mediaStreamIds))
>     , m_backend(WTFMove(backend))
> {
>+    ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled() || m_backend);
> }
> 
> void RTCRtpSender::setTrackToNull()
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp
>index 674b321cee1388e2e7f28eecff9f013593467ab1..601246e28a6ca35dbf7b19abfc2cdbc6db3b5a3e 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.cpp
>@@ -47,11 +47,9 @@ RTCRtpTransceiver::RTCRtpTransceiver(Ref<RTCRtpSender>&& sender, Ref<RTCRtpRecei
> {
> }
> 
>-const String& RTCRtpTransceiver::mid() const
>+String RTCRtpTransceiver::mid() const
> {
>-    if (!m_backend)
>-        return m_mid;
>-    return m_backend->mid();
>+    return m_backend ? m_backend->mid() : String { };
> }
> 
> bool RTCRtpTransceiver::hasSendingDirection() const
>@@ -66,6 +64,13 @@ RTCRtpTransceiverDirection RTCRtpTransceiver::direction() const
>     return m_backend->direction();
> }
> 
>+std::optional<RTCRtpTransceiverDirection> RTCRtpTransceiver::currentDirection() const
>+{
>+    if (!m_backend)
>+        return std::nullopt;
>+    return m_backend->currentDirection();
>+}
>+
> void RTCRtpTransceiver::setDirection(RTCRtpTransceiverDirection direction)
> {
>     m_direction = direction;
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h
>index 802672b50709bb3d3f3bb2e37a760f98fb28fc47..b991d72a11afb8a62aadb444686c8edc8e35ad6a 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.h
>@@ -55,8 +55,9 @@ public:
>     void disableSendingDirection();
> 
>     RTCRtpTransceiverDirection direction() const;
>+    std::optional<RTCRtpTransceiverDirection> currentDirection() const;
>     void setDirection(RTCRtpTransceiverDirection);
>-    const String& mid() const;
>+    String mid() const;
> 
>     RTCRtpSender& sender() { return m_sender.get(); }
>     RTCRtpReceiver& receiver() { return m_receiver.get(); }
>@@ -69,10 +70,11 @@ public:
>     // transport each.
>     RTCIceTransport& iceTransport() { return m_iceTransport.get(); }
> 
>+    RTCRtpTransceiverBackend* backend() { return m_backend.get(); }
>+
> private:
>     RTCRtpTransceiver(Ref<RTCRtpSender>&&, Ref<RTCRtpReceiver>&&, std::unique_ptr<RTCRtpTransceiverBackend>&&);
> 
>-    String m_mid;
>     RTCRtpTransceiverDirection m_direction;
> 
>     Ref<RTCRtpSender> m_sender;
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.idl b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.idl
>index 407871ebc760f53f2705830804bd9d021cb5f04f..aa166028f2cc724912736512fc779092aa02ced6 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.idl
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpTransceiver.idl
>@@ -36,13 +36,11 @@ typedef RTCRtpTransceiverDirection RtpTransceiverDirection;
>     EnabledAtRuntime=PeerConnection
> ] interface RTCRtpTransceiver {
>     readonly attribute DOMString? mid;
>-    // FIXME 169662: missing [SameObject]
>-    readonly attribute RTCRtpSender sender;
>-    // FIXME 169662: missing [SameObject]
>-    readonly attribute RTCRtpReceiver receiver;
>+    [SameObject] readonly attribute RTCRtpSender sender;
>+    [SameObject] readonly attribute RTCRtpReceiver receiver;
>     readonly attribute boolean stopped;
>     readonly attribute RtpTransceiverDirection direction;
>-    // FIXME 169662: missing currentDirection
>+    readonly attribute RTCRtpTransceiverDirection? currentDirection;
>     void setDirection(RtpTransceiverDirection direction);
>     void stop();
>     // FIXME 169662: missing setCodecPreferences
>diff --git a/Source/WebCore/Modules/mediastream/RTCRtpTransceiverBackend.h b/Source/WebCore/Modules/mediastream/RTCRtpTransceiverBackend.h
>index 8dfdf968aec278d9688b7711cd1aab1af6f1b0ae..f280ea40005adf31936f3bf3267e13cb1fddd8bb 100644
>--- a/Source/WebCore/Modules/mediastream/RTCRtpTransceiverBackend.h
>+++ b/Source/WebCore/Modules/mediastream/RTCRtpTransceiverBackend.h
>@@ -35,9 +35,10 @@ public:
>     virtual ~RTCRtpTransceiverBackend() = default;
> 
>     virtual RTCRtpTransceiverDirection direction() const = 0;
>+    virtual std::optional<RTCRtpTransceiverDirection> currentDirection() const = 0;
>     virtual void setDirection(RTCRtpTransceiverDirection) = 0;
> 
>-    virtual const String& mid() = 0;
>+    virtual String mid() = 0;
>     virtual void stop() = 0;
> };
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>index efa19adc72f4189ebb8940482601a5e6a3f82c1d..89efce152748c128683e9217d6610656e8f04a4c 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>@@ -34,6 +34,7 @@
> #include "LibWebRTCProvider.h"
> #include "LibWebRTCRtpReceiverBackend.h"
> #include "LibWebRTCRtpSenderBackend.h"
>+#include "LibWebRTCRtpTransceiverBackend.h"
> #include "LibWebRTCStatsCollector.h"
> #include "LibWebRTCUtils.h"
> #include "Logging.h"
>@@ -197,7 +198,6 @@ void LibWebRTCMediaEndpoint::doSetRemoteDescription(RTCSessionDescription& descr
> 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];
>@@ -208,35 +208,41 @@ bool LibWebRTCMediaEndpoint::addTrack(LibWebRTCRtpSenderBackend& sender, MediaSt
>         });
>     }
> 
>-    std::vector<std::string> ids;
>-    for (auto& id : mediaStreamIds)
>-        ids.push_back(id.utf8().data());
>-
>+    LibWebRTCRtpSenderBackend::Source source;
>+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> rtcTrack;
>     switch (track.privateTrack().type()) {
>     case RealtimeMediaSource::Type::Audio: {
>         auto audioSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
>-        auto audioTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), audioSource.ptr());
>-        sender.setSource(WTFMove(audioSource));
>-        auto rtpSender = m_backend->AddTrack(audioTrack.get(), WTFMove(ids));
>-        if (!rtpSender.ok())
>-            return false;
>-        sender.setRTCSender(rtpSender.MoveValue());
>-        return true;
>+        rtcTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), audioSource.ptr());
>+        source = WTFMove(audioSource);
>+        break;
>     }
>     case RealtimeMediaSource::Type::Video: {
>         auto videoSource = RealtimeOutgoingVideoSource::create(track.privateTrack());
>-        auto videoTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
>-        sender.setSource(WTFMove(videoSource));
>-        auto rtpSender = m_backend->AddTrack(videoTrack.get(), WTFMove(ids));
>-        if (!rtpSender.ok())
>-            return false;
>-        sender.setRTCSender(rtpSender.MoveValue());
>-        return true;
>+        rtcTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
>+        source = WTFMove(videoSource);
>+        break;
>     }
>     case RealtimeMediaSource::Type::None:
>         ASSERT_NOT_REACHED();
>+        return false;
>     }
>-    return false;
>+
>+    sender.setSource(WTFMove(source));
>+    if (auto rtpSender = sender.rtcSender()) {
>+        rtpSender->SetTrack(rtcTrack.get());
>+        return true;
>+    }
>+
>+    std::vector<std::string> ids;
>+    for (auto& id : mediaStreamIds)
>+        ids.push_back(id.utf8().data());
>+
>+    auto newRTPSender = m_backend->AddTrack(rtcTrack.get(), WTFMove(ids));
>+    if (!newRTPSender.ok())
>+        return false;
>+    sender.setRTCSender(newRTPSender.MoveValue());
>+    return true;
> }
> 
> void LibWebRTCMediaEndpoint::removeTrack(LibWebRTCRtpSenderBackend& sender)
>@@ -361,20 +367,131 @@ void LibWebRTCMediaEndpoint::addRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiv
>     }
> 
>     receiver->setBackend(std::make_unique<LibWebRTCRtpReceiverBackend>(WTFMove(rtcReceiver)));
>-    
>-    auto* track = receiver->track();
>-    ASSERT(track);
>+    auto& track = *receiver->track();
>+    fireTrackEvent(receiver.releaseNonNull(), track, rtcStreams, nullptr);
>+}
> 
>+void LibWebRTCMediaEndpoint::fireTrackEvent(Ref<RTCRtpReceiver>&& receiver, Ref<MediaStreamTrack>&& track, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& rtcStreams, RefPtr<RTCRtpTransceiver>&& transceiver)
>+{
>     Vector<RefPtr<MediaStream>> streams;
>     for (auto& rtcStream : rtcStreams) {
>         auto& mediaStream = mediaStreamFromRTCStream(*rtcStream.get());
>         streams.append(&mediaStream);
>-        mediaStream.addTrackFromPlatform(*track);
>+        mediaStream.addTrackFromPlatform(track.get());
>     }
>     m_peerConnectionBackend.connection().fireEvent(RTCTrackEvent::create(eventNames().trackEvent,
>-        Event::CanBubble::No, Event::IsCancelable::No, WTFMove(receiver), track, WTFMove(streams), nullptr));
>+        Event::CanBubble::No, Event::IsCancelable::No, WTFMove(receiver), WTFMove(track), WTFMove(streams), WTFMove(transceiver)));
>+}
>+
>+static inline void setExistingReceiverSourceTrack(RealtimeMediaSource& existingSource, webrtc::RtpReceiverInterface& rtcReceiver)
>+{
>+    switch (rtcReceiver.media_type()) {
>+    case cricket::MEDIA_TYPE_AUDIO: {
>+        ASSERT(existingSource.type() == RealtimeMediaSource::Type::Audio);
>+        rtc::scoped_refptr<webrtc::AudioTrackInterface> audioTrack = static_cast<webrtc::AudioTrackInterface*>(rtcReceiver.track().get());
>+        static_cast<RealtimeIncomingAudioSource&>(existingSource).setSourceTrack(WTFMove(audioTrack));
>+        return;
>+    }
>+    case cricket::MEDIA_TYPE_VIDEO: {
>+        ASSERT(existingSource.type() == RealtimeMediaSource::Type::Video);
>+        rtc::scoped_refptr<webrtc::VideoTrackInterface> videoTrack = static_cast<webrtc::VideoTrackInterface*>(rtcReceiver.track().get());
>+        static_cast<RealtimeIncomingVideoSource&>(existingSource).setSourceTrack(WTFMove(videoTrack));
>+        return;
>+    }
>+    case cricket::MEDIA_TYPE_DATA:
>+        ASSERT_NOT_REACHED();
>+        return;
>+    }
>+}
>+
>+static inline RefPtr<RealtimeMediaSource> sourceFromNewReceiver(webrtc::RtpReceiverInterface& rtcReceiver)
>+{
>+    auto rtcTrack = rtcReceiver.track();
>+    switch (rtcReceiver.media_type()) {
>+    case cricket::MEDIA_TYPE_DATA:
>+        return nullptr;
>+    case cricket::MEDIA_TYPE_AUDIO: {
>+        rtc::scoped_refptr<webrtc::AudioTrackInterface> audioTrack = static_cast<webrtc::AudioTrackInterface*>(rtcTrack.get());
>+        return RealtimeIncomingAudioSource::create(WTFMove(audioTrack), fromStdString(rtcTrack->id()));
>+    }
>+    case cricket::MEDIA_TYPE_VIDEO: {
>+        rtc::scoped_refptr<webrtc::VideoTrackInterface> videoTrack = static_cast<webrtc::VideoTrackInterface*>(rtcTrack.get());
>+        return RealtimeIncomingVideoSource::create(WTFMove(videoTrack), fromStdString(rtcTrack->id()));
>+    }
>+    }
>+}
>+
>+void LibWebRTCMediaEndpoint::newTransceiver(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>&& rtcTransceiver)
>+{
>+    auto* transceiver = m_peerConnectionBackend.existingTransceiver([&](auto& transceiverBackend) {
>+        return rtcTransceiver.get() == transceiverBackend.rtcTransceiver();
>+    });
>+    if (transceiver) {
>+        setExistingReceiverSourceTrack(transceiver->receiver().track()->source(), *rtcTransceiver->receiver());
>+        return;
>+    }
>+
>+    auto rtcReceiver = rtcTransceiver->receiver();
>+    auto source = sourceFromNewReceiver(*rtcReceiver);
>+    if (!source)
>+        return;
>+
>+    auto& newTransceiver = m_peerConnectionBackend.newRemoteTransceiver(std::make_unique<LibWebRTCRtpTransceiverBackend>(WTFMove(rtcTransceiver)), source.releaseNonNull());
>+
>+    fireTrackEvent(makeRef(newTransceiver.receiver()), *newTransceiver.receiver().track(), rtcReceiver->streams(), makeRef(newTransceiver));
>+}
>+
>+template<typename T>
>+std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::createTransceiverBackends(T&& trackOrKind, const RTCRtpTransceiverInit& init, LibWebRTCRtpSenderBackend::Source&& source)
>+{
>+    auto result = m_backend->AddTransceiver(WTFMove(trackOrKind), fromRtpTransceiverInit(init));
>+    if (!result.ok())
>+        return std::nullopt;
>+
>+    auto transceiver = std::make_unique<LibWebRTCRtpTransceiverBackend>(result.MoveValue());
>+    return LibWebRTCMediaEndpoint::Backends { transceiver->createSenderBackend(m_peerConnectionBackend, WTFMove(source)), transceiver->createReceiverBackend(), WTFMove(transceiver) };
>+}
>+
>+std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::addTransceiver(const String& trackKind, const RTCRtpTransceiverInit& init)
>+{
>+    auto type = trackKind == "audio" ? cricket::MediaType::MEDIA_TYPE_AUDIO : cricket::MediaType::MEDIA_TYPE_VIDEO;
>+    return createTransceiverBackends(type, init, nullptr);
>+}
>+
>+std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::addTransceiver(MediaStreamTrack& track, const RTCRtpTransceiverInit& init)
>+{
>+    LibWebRTCRtpSenderBackend::Source source;
>+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> rtcTrack;
>+    switch (track.privateTrack().type()) {
>+    case RealtimeMediaSource::Type::None:
>+        return std::nullopt;
>+    case RealtimeMediaSource::Type::Audio: {
>+        auto audioSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
>+        rtcTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), audioSource.ptr());
>+        source = WTFMove(audioSource);
>+        break;
>+    }
>+    case RealtimeMediaSource::Type::Video: {
>+        auto videoSource = RealtimeOutgoingVideoSource::create(track.privateTrack());
>+        rtcTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
>+        source = WTFMove(videoSource);
>+        break;
>+    }
>+    }
>+
>+    return createTransceiverBackends(WTFMove(rtcTrack), init, WTFMove(source));
> }
> 
>+std::unique_ptr<LibWebRTCRtpTransceiverBackend> LibWebRTCMediaEndpoint::transceiverBackendFromSender(LibWebRTCRtpSenderBackend& backend)
>+{
>+    for (auto& transceiver : m_backend->GetTransceivers()) {
>+        if (transceiver->sender().get() == backend.rtcSender())
>+            return std::make_unique<LibWebRTCRtpTransceiverBackend>(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>(transceiver));
>+    }
>+    return nullptr;
>+}
>+
>+
> void LibWebRTCMediaEndpoint::removeRemoteStream(webrtc::MediaStreamInterface& rtcStream)
> {
>     bool removed = m_streams.remove(&rtcStream);
>@@ -403,6 +520,9 @@ void LibWebRTCMediaEndpoint::OnRemoveStream(rtc::scoped_refptr<webrtc::MediaStre
> 
> void LibWebRTCMediaEndpoint::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& streams)
> {
>+    if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
>+        return;
>+
>     callOnMainThread([protectedThis = makeRef(*this), receiver = WTFMove(receiver), streams]() mutable {
>         if (protectedThis->isStopped())
>             return;
>@@ -410,6 +530,19 @@ void LibWebRTCMediaEndpoint::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverIn
>     });
> }
> 
>+void LibWebRTCMediaEndpoint::OnTrack(rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver)
>+{
>+    if (!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
>+        return;
>+
>+    callOnMainThread([protectedThis = makeRef(*this), transceiver = WTFMove(transceiver)]() mutable {
>+        if (protectedThis->isStopped())
>+            return;
>+        protectedThis->newTransceiver(WTFMove(transceiver));
>+    });
>+}
>+
>+
> std::unique_ptr<RTCDataChannelHandler> LibWebRTCMediaEndpoint::createDataChannel(const String& label, const RTCDataChannelInit& options)
> {
>     auto init = LibWebRTCDataChannelHandler::fromRTCDataChannelInit(options);
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>index f2ba6bc525a32ef42c448208f136c06743e0abc6..3fd50969092aeee02308026b6375659262db42db 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>@@ -28,10 +28,8 @@
> 
> #include "LibWebRTCObservers.h"
> #include "LibWebRTCProvider.h"
>-#include "PeerConnectionBackend.h"
>+#include "LibWebRTCRtpSenderBackend.h"
> #include "RTCRtpReceiver.h"
>-#include "RealtimeOutgoingAudioSource.h"
>-#include "RealtimeOutgoingVideoSource.h"
> #include <Timer.h>
> 
> #pragma GCC diagnostic push
>@@ -61,7 +59,8 @@ namespace WebCore {
> 
> class LibWebRTCProvider;
> class LibWebRTCPeerConnectionBackend;
>-class LibWebRTCRtpSenderBackend;
>+class LibWebRTCRtpReceiverBackend;
>+class LibWebRTCRtpTransceiverBackend;
> class MediaStreamTrack;
> class RTCSessionDescription;
> 
>@@ -101,6 +100,15 @@ public:
>     bool addTrack(LibWebRTCRtpSenderBackend&, MediaStreamTrack&, const Vector<String>&);
>     void removeTrack(LibWebRTCRtpSenderBackend&);
> 
>+    struct Backends {
>+        std::unique_ptr<LibWebRTCRtpSenderBackend> senderBackend;
>+        std::unique_ptr<LibWebRTCRtpReceiverBackend> receiverBackend;
>+        std::unique_ptr<LibWebRTCRtpTransceiverBackend> transceiverBackend;
>+    };
>+    std::optional<Backends> addTransceiver(const String& trackKind, const RTCRtpTransceiverInit&);
>+    std::optional<Backends> addTransceiver(MediaStreamTrack&, const RTCRtpTransceiverInit&);
>+    std::unique_ptr<LibWebRTCRtpTransceiverBackend> transceiverBackendFromSender(LibWebRTCRtpSenderBackend&);
>+
> private:
>     LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend&, LibWebRTCProvider&);
> 
>@@ -110,6 +118,8 @@ private:
>     void OnRemoveStream(rtc::scoped_refptr<webrtc::MediaStreamInterface>) final;
>     void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface>) final;
>     void OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&) final;
>+    void OnTrack(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>) final;
>+
>     void OnRenegotiationNeeded() final;
>     void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState) final;
>     void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState) final;
>@@ -125,6 +135,12 @@ private:
>     void addRemoteStream(webrtc::MediaStreamInterface&);
>     void addRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&);
>     void removeRemoteStream(webrtc::MediaStreamInterface&);
>+    void newTransceiver(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>&&);
>+
>+    void fireTrackEvent(Ref<RTCRtpReceiver>&&, Ref<MediaStreamTrack>&&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&, RefPtr<RTCRtpTransceiver>&&);
>+
>+    template<typename T>
>+    std::optional<Backends> createTransceiverBackends(T&&, const RTCRtpTransceiverInit&, LibWebRTCRtpSenderBackend::Source&&);
> 
>     void OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>&) final;
>     void gatherStatsForLogging();
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>index 50824599bd207174dee176d9f9831294a3ff01b6..1da68d4133e6c7286082f693b84f66a869857fdb 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>@@ -33,6 +33,7 @@
> #include "LibWebRTCMediaEndpoint.h"
> #include "LibWebRTCRtpReceiverBackend.h"
> #include "LibWebRTCRtpSenderBackend.h"
>+#include "LibWebRTCRtpTransceiverBackend.h"
> #include "MediaEndpointConfiguration.h"
> #include "Page.h"
> #include "RTCIceCandidate.h"
>@@ -200,12 +201,12 @@ void LibWebRTCPeerConnectionBackend::doAddIceCandidate(RTCIceCandidate& candidat
>     addIceCandidateSucceeded();
> }
> 
>-static inline Ref<RTCRtpReceiver> createReceiverForSource(ScriptExecutionContext& context, Ref<RealtimeMediaSource>&& source)
>+static inline Ref<RTCRtpReceiver> createReceiverForSource(ScriptExecutionContext& context, Ref<RealtimeMediaSource>&& source, std::unique_ptr<RTCRtpReceiverBackend>&& backend)
> {
>     auto remoteTrackPrivate = MediaStreamTrackPrivate::create(WTFMove(source), String { source->id() });
>     auto remoteTrack = MediaStreamTrack::create(context, WTFMove(remoteTrackPrivate));
> 
>-    return RTCRtpReceiver::create(WTFMove(remoteTrack));
>+    return RTCRtpReceiver::create(WTFMove(remoteTrack), WTFMove(backend));
> }
> 
> static inline Ref<RealtimeMediaSource> createEmptySource(const String& trackKind, String&& trackId)
>@@ -219,7 +220,7 @@ static inline Ref<RealtimeMediaSource> createEmptySource(const String& trackKind
> 
> Ref<RTCRtpReceiver> LibWebRTCPeerConnectionBackend::createReceiver(const String& trackKind, const String& trackId)
> {
>-    auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), createEmptySource(trackKind, String(trackId)));
>+    auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), createEmptySource(trackKind, String(trackId)), nullptr);
>     m_pendingReceivers.append(receiver.copyRef());
>     return receiver;
> }
>@@ -237,7 +238,7 @@ LibWebRTCPeerConnectionBackend::VideoReceiver LibWebRTCPeerConnectionBackend::vi
>         }
>     }
>     auto source = RealtimeIncomingVideoSource::create(nullptr, WTFMove(trackId));
>-    auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), source.copyRef());
>+    auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), source.copyRef(), nullptr);
> 
>     auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
>     auto transceiver = RTCRtpTransceiver::create(RTCRtpSender::create("video", { }, WTFMove(senderBackend)), receiver.copyRef(), nullptr);
>@@ -260,7 +261,7 @@ LibWebRTCPeerConnectionBackend::AudioReceiver LibWebRTCPeerConnectionBackend::au
>         }
>     }
>     auto source = RealtimeIncomingAudioSource::create(nullptr, WTFMove(trackId));
>-    auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), source.copyRef());
>+    auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), source.copyRef(), nullptr);
> 
>     auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
>     auto transceiver = RTCRtpTransceiver::create(RTCRtpSender::create("audio", { }, WTFMove(senderBackend)), receiver.copyRef(), nullptr);
>@@ -316,11 +317,57 @@ RefPtr<RTCSessionDescription> LibWebRTCPeerConnectionBackend::remoteDescription(
> 
> static inline LibWebRTCRtpSenderBackend& backendFromRTPSender(RTCRtpSender& sender)
> {
>+    ASSERT(!sender.isStopped());
>     return static_cast<LibWebRTCRtpSenderBackend&>(*sender.backend());
> }
> 
>-ExceptionOr<Ref<RTCRtpSender>> LibWebRTCPeerConnectionBackend::addTrack(RTCRtpSender* sender, MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
>+
>+static inline RefPtr<RTCRtpSender> findExistingSender(const Vector<std::reference_wrapper<RTCRtpSender>>& senders, LibWebRTCRtpSenderBackend& senderBackend)
>+{
>+    ASSERT(senderBackend.rtcSender());
>+    for (RTCRtpSender& sender : senders) {
>+        if (!sender.isStopped() && senderBackend.rtcSender() == backendFromRTPSender(sender).rtcSender())
>+            return makeRef(sender);
>+    }
>+    return nullptr;
>+}
>+
>+ExceptionOr<Ref<RTCRtpSender>> LibWebRTCPeerConnectionBackend::addTrack(MediaStreamTrack& track, Vector<String>&& mediaStreamIds)
> {
>+    if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled()) {
>+        auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
>+        if (!m_endpoint->addTrack(*senderBackend, track, mediaStreamIds))
>+            return Exception { TypeError, "Unable to add track"_s };
>+
>+        if (auto sender = findExistingSender(m_peerConnection.getSenders(), *senderBackend)) {
>+            sender->setTrack(makeRef(track));
>+            sender->setMediaStreamIds(WTFMove(mediaStreamIds));
>+            return sender.releaseNonNull();
>+        }
>+
>+        auto transceiverBackend = m_endpoint->transceiverBackendFromSender(*senderBackend);
>+
>+        auto sender = RTCRtpSender::create(makeRef(track), WTFMove(mediaStreamIds), WTFMove(senderBackend));
>+        auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), createEmptySource(track.kind(), createCanonicalUUIDString()), transceiverBackend->createReceiverBackend());
>+        auto transceiver = RTCRtpTransceiver::create(sender.copyRef(), WTFMove(receiver), WTFMove(transceiverBackend));
>+        m_peerConnection.addInternalTransceiver(WTFMove(transceiver));
>+        return WTFMove(sender);
>+    }
>+
>+    RTCRtpSender* sender = nullptr;
>+    // Reuse an existing sender with the same track kind if it has never been used to send before.
>+    for (auto& transceiver : m_peerConnection.getTransceivers()) {
>+        auto& existingSender = transceiver->sender();
>+        if (!existingSender.isStopped() && existingSender.trackKind() == track.kind() && existingSender.trackId().isNull() && !transceiver->hasSendingDirection()) {
>+            existingSender.setTrack(makeRef(track));
>+            existingSender.setMediaStreamIds(WTFMove(mediaStreamIds));
>+            transceiver->enableSendingDirection();
>+            sender = &existingSender;
>+
>+            break;
>+        }
>+    }
>+
>     if (!sender) {
>         const String& trackKind = track.kind();
>         String trackId = createCanonicalUUIDString();
>@@ -340,8 +387,25 @@ ExceptionOr<Ref<RTCRtpSender>> LibWebRTCPeerConnectionBackend::addTrack(RTCRtpSe
>     return makeRef(*sender);
> }
> 
>+template<typename T>
>+ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addUnifiedPlanTransceiver(T&& trackOrKind, const RTCRtpTransceiverInit& init)
>+{
>+    auto backends = m_endpoint->addTransceiver(trackOrKind, init);
>+    if (!backends)
>+        return Exception { InvalidAccessError, "Unable to add transceiver"_s };
>+
>+    auto sender = RTCRtpSender::create(WTFMove(trackOrKind), Vector<String> { }, WTFMove(backends->senderBackend));
>+    auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), createEmptySource(sender->trackKind(), createCanonicalUUIDString()), WTFMove(backends->receiverBackend));
>+    auto transceiver = RTCRtpTransceiver::create(WTFMove(sender), WTFMove(receiver), WTFMove(backends->transceiverBackend));
>+    m_peerConnection.addInternalTransceiver(transceiver.copyRef());
>+    return WTFMove(transceiver);
>+}
>+
> ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addTransceiver(const String& trackKind, const RTCRtpTransceiverInit& init)
> {
>+    if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
>+        return addUnifiedPlanTransceiver(String { trackKind }, init);
>+
>     auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
>     auto newSender = RTCRtpSender::create(String(trackKind), Vector<String>(), WTFMove(senderBackend));
>     return completeAddTransceiver(WTFMove(newSender), init, createCanonicalUUIDString(), trackKind);
>@@ -349,6 +413,9 @@ ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addTransceiv
> 
> ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addTransceiver(Ref<MediaStreamTrack>&& track, const RTCRtpTransceiverInit& init)
> {
>+    if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
>+        return addUnifiedPlanTransceiver(WTFMove(track), init);
>+
>     auto senderBackend = std::make_unique<LibWebRTCRtpSenderBackend>(*this, nullptr);
>     auto& backend = *senderBackend;
>     auto sender = RTCRtpSender::create(track.copyRef(), Vector<String>(), WTFMove(senderBackend));
>@@ -358,6 +425,29 @@ ExceptionOr<Ref<RTCRtpTransceiver>> LibWebRTCPeerConnectionBackend::addTransceiv
>     return completeAddTransceiver(WTFMove(sender), init, track->id(), track->kind());
> }
> 
>+static inline LibWebRTCRtpTransceiverBackend& backendFromRTPTransceiver(RTCRtpTransceiver& transceiver)
>+{
>+    return static_cast<LibWebRTCRtpTransceiverBackend&>(*transceiver.backend());
>+}
>+
>+RTCRtpTransceiver* LibWebRTCPeerConnectionBackend::existingTransceiver(WTF::Function<bool(LibWebRTCRtpTransceiverBackend&)>&& matchingFunction)
>+{
>+    for (auto& transceiver : m_peerConnection.getTransceivers()) {
>+        if (matchingFunction(backendFromRTPTransceiver(*transceiver)))
>+            return transceiver.get();
>+    }
>+    return nullptr;
>+}
>+
>+RTCRtpTransceiver& LibWebRTCPeerConnectionBackend::newRemoteTransceiver(std::unique_ptr<LibWebRTCRtpTransceiverBackend>&& transceiverBackend, Ref<RealtimeMediaSource>&& receiverSource)
>+{
>+    auto sender = RTCRtpSender::create(receiverSource->type() == RealtimeMediaSource::Type::Audio ? "audio"_s : "video"_s, Vector<String> { }, transceiverBackend->createSenderBackend(*this, nullptr));
>+    auto receiver = createReceiverForSource(*m_peerConnection.scriptExecutionContext(), WTFMove(receiverSource), transceiverBackend->createReceiverBackend());
>+    auto transceiver = RTCRtpTransceiver::create(WTFMove(sender), WTFMove(receiver), WTFMove(transceiverBackend));
>+    m_peerConnection.addInternalTransceiver(transceiver.copyRef());
>+    return transceiver.get();
>+}
>+
> Ref<RTCRtpTransceiver> LibWebRTCPeerConnectionBackend::completeAddTransceiver(Ref<RTCRtpSender>&& sender, const RTCRtpTransceiverInit& init, const String& trackId, const String& trackKind)
> {
>     auto transceiver = RTCRtpTransceiver::create(WTFMove(sender), createReceiver(trackKind, trackId), nullptr);
>@@ -373,21 +463,13 @@ void LibWebRTCPeerConnectionBackend::removeTrack(RTCRtpSender& sender)
>     m_endpoint->removeTrack(backendFromRTPSender(sender));
> }
> 
>-template<typename Source>
>-static inline bool updateTrackSource(Source& source, MediaStreamTrack* track)
>-{
>-    if (!track) {
>-        source.stop();
>-        return true;
>-    }
>-    return source.setSource(track->privateTrack());
>-}
>-
> void LibWebRTCPeerConnectionBackend::applyRotationForOutgoingVideoSources()
> {
>     for (auto& transceiver : m_peerConnection.getTransceivers()) {
>-        if (auto* videoSource = backendFromRTPSender(transceiver->sender()).videoSource())
>-            videoSource->setApplyRotation(true);
>+        if (!transceiver->sender().isStopped()) {
>+            if (auto* videoSource = backendFromRTPSender(transceiver->sender()).videoSource())
>+                videoSource->setApplyRotation(true);
>+        }
>     }
> }
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>index d376f0ec178eef99f804a5c24198b76ddad8dd34..1f6850d1752325a114b7944745ab15f247ae19c3 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>@@ -38,11 +38,13 @@ namespace WebCore {
> 
> class LibWebRTCMediaEndpoint;
> class LibWebRTCProvider;
>+class LibWebRTCRtpTransceiverBackend;
> class RTCRtpReceiver;
> class RTCSessionDescription;
> class RTCStatsReport;
> class RealtimeIncomingAudioSource;
> class RealtimeIncomingVideoSource;
>+class RealtimeMediaSource;
> class RealtimeOutgoingAudioSource;
> class RealtimeOutgoingVideoSource;
> 
>@@ -84,12 +86,15 @@ private:
> 
>     void getStatsSucceeded(const DeferredPromise&, Ref<RTCStatsReport>&&);
> 
>-    ExceptionOr<Ref<RTCRtpSender>> addTrack(RTCRtpSender*, MediaStreamTrack&, const Vector<String>&) final;
>+    ExceptionOr<Ref<RTCRtpSender>> addTrack(MediaStreamTrack&, Vector<String>&&) final;
>     void removeTrack(RTCRtpSender&) final;
> 
>     ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(const String&, const RTCRtpTransceiverInit&) final;
>     ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(Ref<MediaStreamTrack>&&, const RTCRtpTransceiverInit&) final;
> 
>+    RTCRtpTransceiver* existingTransceiver(WTF::Function<bool(LibWebRTCRtpTransceiverBackend&)>&&);
>+    RTCRtpTransceiver& newRemoteTransceiver(std::unique_ptr<LibWebRTCRtpTransceiverBackend>&&, Ref<RealtimeMediaSource>&&);
>+
>     struct VideoReceiver {
>         Ref<RTCRtpReceiver> receiver;
>         Ref<RealtimeIncomingVideoSource> source;
>@@ -108,6 +113,9 @@ private:
> 
>     Ref<RTCRtpReceiver> createReceiver(const String& trackKind, const String& trackId);
> 
>+    template<typename T>
>+    ExceptionOr<Ref<RTCRtpTransceiver>> addUnifiedPlanTransceiver(T&& trackOrKind, const RTCRtpTransceiverInit&);
>+
>     Ref<LibWebRTCMediaEndpoint> m_endpoint;
>     bool m_isLocalDescriptionSet { false };
>     bool m_isRemoteDescriptionSet { false };
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>index ad96b1fccef776375bc3c68f5f1e6b57108e73f4..a0e1488b1eecf887d5cc998d737ad6ec97787555 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.cpp
>@@ -90,7 +90,7 @@ void LibWebRTCRtpSenderBackend::replaceTrack(ScriptExecutionContext& context, RT
>         protectedSender->setTrack(track.releaseNonNull());
>         if (!hasTrack) {
>             // FIXME: In case of unified plan, we should use m_rtcSender->SetTrack and no longer need m_peerConnectionBackend.
>-            auto result = m_peerConnectionBackend->addTrack(protectedSender.ptr(), *protectedSender->track(), { });
>+            auto result = m_peerConnectionBackend->addTrack(*protectedSender->track(), { });
>             if (result.hasException()) {
>                 promise.reject(result.releaseException());
>                 return;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>index 9dff4aa9a167776fd6b060f1a31e0eef08b82ff9..c7344043a365b242b821015bea74b1c32b81f0a4 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>@@ -27,6 +27,7 @@
> #if ENABLE(WEB_RTC)
> 
> #include "LibWebRTCMacros.h"
>+#include "LibWebRTCPeerConnectionBackend.h"
> #include "RTCRtpSenderBackend.h"
> #include "RealtimeOutgoingAudioSource.h"
> #include "RealtimeOutgoingVideoSource.h"
>@@ -46,12 +47,20 @@ class LibWebRTCPeerConnectionBackend;
> 
> class LibWebRTCRtpSenderBackend final : public RTCRtpSenderBackend {
> public:
>-    explicit LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender)
>+    LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender)
>         : m_peerConnectionBackend(makeWeakPtr(&backend))
>         , m_rtcSender(WTFMove(rtcSender))
>     {
>     }
> 
>+    using Source = Variant<std::nullptr_t, Ref<RealtimeOutgoingAudioSource>, Ref<RealtimeOutgoingVideoSource>>;
>+    LibWebRTCRtpSenderBackend(LibWebRTCPeerConnectionBackend& backend, rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender, Source&& source)
>+        : m_peerConnectionBackend(makeWeakPtr(&backend))
>+        , m_rtcSender(WTFMove(rtcSender))
>+        , m_source(WTFMove(source))
>+    {
>+    }
>+
>     void setRTCSender(rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender) { m_rtcSender = WTFMove(rtcSender); }
>     webrtc::RtpSenderInterface* rtcSender() { return m_rtcSender.get(); }
> 
>@@ -71,24 +80,19 @@ public:
>         );
>     }
> 
>-    bool hasNoSource() const
>+    bool hasSource() const
>     {
>         return WTF::switchOn(m_source,
>-            [] (const std::nullptr_t&) { return true; },
>-            [] (const auto&) { return false; }
>+            [] (const std::nullptr_t&) { return false; },
>+            [] (const auto&) { return true; }
>         );
>     }
> 
>-    void setSource(Ref<RealtimeOutgoingAudioSource>&& source)
>-    {
>-        ASSERT(hasNoSource());
>-        m_source = WTFMove(source);
>-    }
>-
>-    void setSource(Ref<RealtimeOutgoingVideoSource>&& source)
>+    void setSource(Source&& source)
>     {
>-        ASSERT(hasNoSource());
>+        ASSERT(!hasSource());
>         m_source = WTFMove(source);
>+        ASSERT(hasSource());
>     }
> 
> private:
>@@ -98,7 +102,7 @@ private:
> 
>     WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
>     rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;
>-    Variant<std::nullptr_t, Ref<RealtimeOutgoingAudioSource>, Ref<RealtimeOutgoingVideoSource>> m_source;
>+    Source m_source;
> };
> 
> } // namespace WebCore
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..850cfe9e401aedff13125c7a9c46ec7b7b499733
>--- /dev/null
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp
>@@ -0,0 +1,78 @@
>+/*
>+ * Copyright (C) 2018 Apple Inc.
>+ *
>+ * Redistribution and use in source and binary forms, with or without
>+ * modification, are permitted provided that the following conditions
>+ * are met:
>+ * 1.  Redistributions of source code must retain the above copyright
>+ *     notice, this list of conditions and the following disclaimer.
>+ * 2.  Redistributions in binary form must reproduce the above copyright
>+ *     notice, this list of conditions and the following disclaimer in the
>+ *     documentation and/or other materials provided with the distribution.
>+ *
>+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
>+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
>+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
>+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
>+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
>+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
>+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
>+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
>+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
>+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
>+ */
>+
>+#include "config.h"
>+#include "LibWebRTCRtpTransceiverBackend.h"
>+
>+#if ENABLE(WEB_RTC) && USE(LIBWEBRTC)
>+
>+#include "LibWebRTCRtpReceiverBackend.h"
>+#include "LibWebRTCRtpSenderBackend.h"
>+#include "LibWebRTCUtils.h"
>+
>+namespace WebCore {
>+
>+std::unique_ptr<LibWebRTCRtpReceiverBackend> LibWebRTCRtpTransceiverBackend::createReceiverBackend()
>+{
>+    return std::make_unique<LibWebRTCRtpReceiverBackend>(m_rtcTransceiver->receiver());
>+}
>+
>+std::unique_ptr<LibWebRTCRtpSenderBackend> LibWebRTCRtpTransceiverBackend::createSenderBackend(LibWebRTCPeerConnectionBackend& backend, LibWebRTCRtpSenderBackend::Source&& source)
>+{
>+    return std::make_unique<LibWebRTCRtpSenderBackend>(backend, m_rtcTransceiver->sender(), WTFMove(source));
>+}
>+
>+RTCRtpTransceiverDirection LibWebRTCRtpTransceiverBackend::direction() const
>+{
>+    return toRTCRtpTransceiverDirection(m_rtcTransceiver->direction());
>+}
>+
>+std::optional<RTCRtpTransceiverDirection> LibWebRTCRtpTransceiverBackend::currentDirection() const
>+{
>+    auto value = m_rtcTransceiver->current_direction();
>+    if (!value)
>+        return std::nullopt;
>+    return toRTCRtpTransceiverDirection(*value);
>+}
>+
>+void LibWebRTCRtpTransceiverBackend::setDirection(RTCRtpTransceiverDirection direction)
>+{
>+    m_rtcTransceiver->SetDirection(fromRTCRtpTransceiverDirection(direction));
>+}
>+
>+String LibWebRTCRtpTransceiverBackend::mid()
>+{
>+    if (auto mid = m_rtcTransceiver->mid())
>+        return fromStdString(*mid);
>+    return String { };
>+}
>+
>+void LibWebRTCRtpTransceiverBackend::stop()
>+{
>+    m_rtcTransceiver->Stop();
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..c47ff2019cc2a62bff97835c11daefc01dcc53db
>--- /dev/null
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.h
>@@ -0,0 +1,69 @@
>+/*
>+ * Copyright (C) 2018 Apple Inc.
>+ *
>+ * Redistribution and use in source and binary forms, with or without
>+ * modification, are permitted provided that the following conditions
>+ * are met:
>+ * 1.  Redistributions of source code must retain the above copyright
>+ *     notice, this list of conditions and the following disclaimer.
>+ * 2.  Redistributions in binary form must reproduce the above copyright
>+ *     notice, this list of conditions and the following disclaimer in the
>+ *     documentation and/or other materials provided with the distribution.
>+ *
>+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
>+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
>+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
>+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
>+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
>+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
>+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
>+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
>+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
>+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
>+ */
>+
>+#pragma once
>+
>+#if ENABLE(WEB_RTC) && USE(LIBWEBRTC)
>+
>+#include "LibWebRTCMacros.h"
>+#include "LibWebRTCRtpSenderBackend.h"
>+#include "RTCRtpTransceiverBackend.h"
>+
>+#pragma GCC diagnostic push
>+#pragma GCC diagnostic ignored "-Wunused-parameter"
>+
>+#include <webrtc/api/rtptransceiverinterface.h>
>+#include <webrtc/rtc_base/scoped_ref_ptr.h>
>+
>+#pragma GCC diagnostic pop
>+
>+namespace WebCore {
>+
>+class LibWebRTCRtpReceiverBackend;
>+
>+class LibWebRTCRtpTransceiverBackend final : public RTCRtpTransceiverBackend {
>+public:
>+    explicit LibWebRTCRtpTransceiverBackend(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>&& rtcTransceiver)
>+        : m_rtcTransceiver(WTFMove(rtcTransceiver))
>+    {
>+    }
>+
>+    std::unique_ptr<LibWebRTCRtpReceiverBackend> createReceiverBackend();
>+    std::unique_ptr<LibWebRTCRtpSenderBackend> createSenderBackend(LibWebRTCPeerConnectionBackend&, LibWebRTCRtpSenderBackend::Source&&);
>+
>+    webrtc::RtpTransceiverInterface* rtcTransceiver() { return m_rtcTransceiver.get(); }
>+
>+private:
>+    RTCRtpTransceiverDirection direction() const final;
>+    std::optional<RTCRtpTransceiverDirection> currentDirection() const final;
>+    void setDirection(RTCRtpTransceiverDirection) final;
>+    String mid() final;
>+    void stop() final;
>+
>+    rtc::scoped_refptr<webrtc::RtpTransceiverInterface> m_rtcTransceiver;
>+};
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(WEB_RTC) && USE(LIBWEBRTC)
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
>index 2f3c10ce05985f71355f614556116552d6f29939..64509d0635b59929e89b872fdb2d887d2b4e9067 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
>@@ -28,10 +28,18 @@
> #if USE(LIBWEBRTC)
> 
> #include "LibWebRTCMacros.h"
>+#include "RTCPeerConnection.h"
> #include "RTCRtpParameters.h"
>-#include <webrtc/api/rtpparameters.h>
> #include <wtf/text/WTFString.h>
> 
>+#pragma GCC diagnostic push
>+#pragma GCC diagnostic ignored "-Wunused-parameter"
>+
>+#include <webrtc/api/rtpparameters.h>
>+#include <webrtc/api/rtptransceiverinterface.h>
>+
>+#pragma GCC diagnostic pop
>+
> namespace WebCore {
> 
> static inline RTCRtpParameters::EncodingParameters toRTCEncodingParameters(const webrtc::RtpEncodingParameters& rtcParameters)
>@@ -177,6 +185,41 @@ webrtc::RtpParameters fromRTCRtpParameters(const RTCRtpParameters& parameters)
>     return rtcParameters;
> }
> 
>+RTCRtpTransceiverDirection toRTCRtpTransceiverDirection(webrtc::RtpTransceiverDirection rtcDirection)
>+{
>+    switch (rtcDirection) {
>+    case webrtc::RtpTransceiverDirection::kSendRecv:
>+        return RTCRtpTransceiverDirection::Sendrecv;
>+    case webrtc::RtpTransceiverDirection::kSendOnly:
>+        return RTCRtpTransceiverDirection::Sendonly;
>+    case webrtc::RtpTransceiverDirection::kRecvOnly:
>+        return RTCRtpTransceiverDirection::Recvonly;
>+    case webrtc::RtpTransceiverDirection::kInactive:
>+        return RTCRtpTransceiverDirection::Inactive;
>+    };
>+}
>+
>+webrtc::RtpTransceiverDirection fromRTCRtpTransceiverDirection(RTCRtpTransceiverDirection direction)
>+{
>+    switch (direction) {
>+    case RTCRtpTransceiverDirection::Sendrecv:
>+        return webrtc::RtpTransceiverDirection::kSendRecv;
>+    case RTCRtpTransceiverDirection::Sendonly:
>+        return webrtc::RtpTransceiverDirection::kSendOnly;
>+    case RTCRtpTransceiverDirection::Recvonly:
>+        return webrtc::RtpTransceiverDirection::kRecvOnly;
>+    case RTCRtpTransceiverDirection::Inactive:
>+        return webrtc::RtpTransceiverDirection::kInactive;
>+    };
>+}
>+
>+webrtc::RtpTransceiverInit fromRtpTransceiverInit(const RTCRtpTransceiverInit& init)
>+{
>+    webrtc::RtpTransceiverInit rtcInit;
>+    rtcInit.direction = fromRTCRtpTransceiverDirection(init.direction);
>+    return rtcInit;
>+}
>+
> }; // namespace WebCore
> 
> #endif // USE(LIBWEBRTC)
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h
>index ba63751f1a24deeb1569c323cadcb9bc42e9dfae..c1b307c2228149aa4f51e7a593fd467c464633f2 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.h
>@@ -30,15 +30,25 @@
> 
> namespace webrtc {
> struct RtpParameters;
>+struct RtpTransceiverInit;
>+
>+enum class RtpTransceiverDirection;
> }
> 
> namespace WebCore {
> 
> struct RTCRtpParameters;
>+struct RTCRtpTransceiverInit;
>+
>+enum class RTCRtpTransceiverDirection;
> 
> RTCRtpParameters toRTCRtpParameters(const webrtc::RtpParameters&);
> webrtc::RtpParameters fromRTCRtpParameters(const RTCRtpParameters&);
> 
>+RTCRtpTransceiverDirection toRTCRtpTransceiverDirection(webrtc::RtpTransceiverDirection);
>+webrtc::RtpTransceiverDirection fromRTCRtpTransceiverDirection(RTCRtpTransceiverDirection);
>+webrtc::RtpTransceiverInit fromRtpTransceiverInit(const RTCRtpTransceiverInit&);
>+
> inline String fromStdString(const std::string& value)
> {
>     return String::fromUTF8(value.data(), value.length());
>diff --git a/Source/WebCore/WebCore.xcodeproj/project.pbxproj b/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>index 4a8ac36dffdb0f3b2e8b894f13fe42c26e47255c..7929ff638f3ae43725aecc0ce92c73409bf58a53 100644
>--- a/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>+++ b/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>@@ -1103,6 +1103,7 @@
> 		4186BD3E213EE3400001826F /* LibWebRTCUtils.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41D1A049213EDDFD0063FB6B /* LibWebRTCUtils.cpp */; };
> 		4186BD3F213EE3430001826F /* LibWebRTCRtpSenderBackend.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41D1A04B213EDDFE0063FB6B /* LibWebRTCRtpSenderBackend.cpp */; };
> 		4186BD40213EE3450001826F /* LibWebRTCRtpReceiverBackend.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41D1A04A213EDDFE0063FB6B /* LibWebRTCRtpReceiverBackend.cpp */; };
>+		4186BD4E2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 4186BD4D2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp */; };
> 		41885B9311B6FDA6003383BB /* FormSubmission.h in Headers */ = {isa = PBXBuildFile; fileRef = 41885B9111B6FDA6003383BB /* FormSubmission.h */; settings = {ATTRIBUTES = (Private, ); }; };
> 		418A06D0133C04D500CD379C /* EventDispatcher.h in Headers */ = {isa = PBXBuildFile; fileRef = 418A06CE133C04D500CD379C /* EventDispatcher.h */; };
> 		418F88050FF957AF0080F045 /* JSAbstractWorker.h in Headers */ = {isa = PBXBuildFile; fileRef = 418F88030FF957AE0080F045 /* JSAbstractWorker.h */; };
>@@ -7288,6 +7289,8 @@
> 		4186BD3B213EDE380001826F /* LibWebRTCRtpReceiverBackend.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = LibWebRTCRtpReceiverBackend.h; path = libwebrtc/LibWebRTCRtpReceiverBackend.h; sourceTree = "<group>"; };
> 		4186BD3D213EDE390001826F /* LibWebRTCRtpSenderBackend.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = LibWebRTCRtpSenderBackend.h; path = libwebrtc/LibWebRTCRtpSenderBackend.h; sourceTree = "<group>"; };
> 		4186BD46214072B60001826F /* RTCRtpTransceiverBackend.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCRtpTransceiverBackend.h; sourceTree = "<group>"; };
>+		4186BD4B2140A8050001826F /* LibWebRTCRtpTransceiverBackend.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = LibWebRTCRtpTransceiverBackend.h; path = libwebrtc/LibWebRTCRtpTransceiverBackend.h; sourceTree = "<group>"; };
>+		4186BD4D2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = LibWebRTCRtpTransceiverBackend.cpp; path = libwebrtc/LibWebRTCRtpTransceiverBackend.cpp; sourceTree = "<group>"; };
> 		41885B9111B6FDA6003383BB /* FormSubmission.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = FormSubmission.h; sourceTree = "<group>"; };
> 		41885B9211B6FDA6003383BB /* FormSubmission.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = FormSubmission.cpp; sourceTree = "<group>"; };
> 		418A06CE133C04D500CD379C /* EventDispatcher.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = EventDispatcher.h; sourceTree = "<group>"; };
>@@ -16894,6 +16897,8 @@
> 				4186BD3B213EDE380001826F /* LibWebRTCRtpReceiverBackend.h */,
> 				41D1A04B213EDDFE0063FB6B /* LibWebRTCRtpSenderBackend.cpp */,
> 				4186BD3D213EDE390001826F /* LibWebRTCRtpSenderBackend.h */,
>+				4186BD4D2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp */,
>+				4186BD4B2140A8050001826F /* LibWebRTCRtpTransceiverBackend.h */,
> 				41D28D0B2139E01D00F4206F /* LibWebRTCStatsCollector.cpp */,
> 				41D28D0C2139E01E00F4206F /* LibWebRTCStatsCollector.h */,
> 				41D1A049213EDDFD0063FB6B /* LibWebRTCUtils.cpp */,
>@@ -28781,7 +28786,6 @@
> 				D6489D26166FFCF1007C031B /* JSHTMLTemplateElement.h in Headers */,
> 				A80E7E9D0A1A83E3007FB8C5 /* JSHTMLTextAreaElement.h in Headers */,
> 				83E359A21BB1031D002CEB98 /* JSHTMLTimeElement.h in Headers */,
>-				E42050172141901B0066EF3B /* ProcessWarming.h in Headers */,
> 				A80E7B0C0A19D606007FB8C5 /* JSHTMLTitleElement.h in Headers */,
> 				070756D414239A4F00414161 /* JSHTMLTrackElement.h in Headers */,
> 				1A85B2110A1B258700D8C87C /* JSHTMLUListElement.h in Headers */,
>@@ -29789,6 +29793,7 @@
> 				B71FE6DF11091CB300DAEF77 /* PrintContext.h in Headers */,
> 				51F645D51FECDBCE00B54DED /* Process.h in Headers */,
> 				A8EA7EBC0A1945D000A8EF5F /* ProcessingInstruction.h in Headers */,
>+				E42050172141901B0066EF3B /* ProcessWarming.h in Headers */,
> 				E44613EC0CD681B500FADA75 /* ProgressEvent.h in Headers */,
> 				A715E653134BBBEC00D8E713 /* ProgressShadowElement.h in Headers */,
> 				1A2A68240B5BEDE70002A480 /* ProgressTracker.h in Headers */,
>@@ -31525,6 +31530,7 @@
> 				417612B11E3A994000C3D81D /* LibWebRTCPeerConnectionBackend.cpp in Sources */,
> 				4186BD40213EE3450001826F /* LibWebRTCRtpReceiverBackend.cpp in Sources */,
> 				4186BD3F213EE3430001826F /* LibWebRTCRtpSenderBackend.cpp in Sources */,
>+				4186BD4E2140B9E80001826F /* LibWebRTCRtpTransceiverBackend.cpp in Sources */,
> 				41D28D0D2139E05800F4206F /* LibWebRTCStatsCollector.cpp in Sources */,
> 				4186BD3E213EE3400001826F /* LibWebRTCUtils.cpp in Sources */,
> 				9759E93E14EF1CF80026A2DD /* LoadableTextTrack.cpp in Sources */,
>diff --git a/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp b/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp
>index 471ab3d7683b3b8ffc502bf34652479735c21abb..52331381fe5bbce3dc3cb47cef201eb03219d234 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp
>+++ b/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.cpp
>@@ -41,6 +41,7 @@ RealtimeIncomingAudioSource::RealtimeIncomingAudioSource(rtc::scoped_refptr<webr
>     : RealtimeMediaSource(WTFMove(audioTrackId), RealtimeMediaSource::Type::Audio, String())
>     , m_audioTrack(WTFMove(audioTrack))
> {
>+    setName("remote audio");
>     notifyMutedChange(!m_audioTrack);
> }
> 
>diff --git a/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.cpp b/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.cpp
>index 259b46bbf45d8e6c14684049acaadc7fb0bb7a4c..f760d40f75f60aadc8af21d2d0f57d67cc0c5508 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.cpp
>+++ b/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.cpp
>@@ -41,6 +41,7 @@ RealtimeIncomingVideoSource::RealtimeIncomingVideoSource(rtc::scoped_refptr<webr
>     : RealtimeMediaSource(WTFMove(videoTrackId), RealtimeMediaSource::Type::Video, String())
>     , m_videoTrack(WTFMove(videoTrack))
> {
>+    setName("remote video");
>     m_currentSettings.setWidth(640);
>     m_currentSettings.setHeight(480);
>     notifyMutedChange(!m_videoTrack);
>diff --git a/Source/WebCore/platform/mediastream/RealtimeMediaSource.h b/Source/WebCore/platform/mediastream/RealtimeMediaSource.h
>index 1e1bb82af525fe6da3aa30603c5294b8cb02ea56..057240ffc4404d0675fbf53eba6d4b97ba9fcf9a 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeMediaSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeMediaSource.h
>@@ -158,7 +158,7 @@ public:
>     virtual void setInterrupted(bool, bool);
> 
>     const String& name() const { return m_name; }
>-    void setName(const String& name) { m_name = name; }
>+    void setName(String&& name) { m_name = WTFMove(name); }
> 
>     unsigned fitnessScore() const { return m_fitnessScore; }
> 
>diff --git a/Source/WebCore/testing/Internals.cpp b/Source/WebCore/testing/Internals.cpp
>index ef1ea4eb731815ede59d8a70c3d4ae92f8840019..f75ab245ae7009a43813884dc4fc25741639bc0b 100644
>--- a/Source/WebCore/testing/Internals.cpp
>+++ b/Source/WebCore/testing/Internals.cpp
>@@ -1386,6 +1386,9 @@ void Internals::emulateRTCPeerConnectionPlatformEvent(RTCPeerConnection& connect
> 
> void Internals::useMockRTCPeerConnectionFactory(const String& testCase)
> {
>+    // FIXME: We should upgrade mocks to support unified plan APIs, until then use plan B in tests using mock.
>+
>+    ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled());
>     if (!LibWebRTCProvider::webRTCAvailable())
>         return;
> 
>diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
>index 08bc975fdb9c5f43e3d8622c200d3033302b3c80..84fa954c4baf14c7c509e44ee2978cdac87768cd 100644
>--- a/LayoutTests/ChangeLog
>+++ b/LayoutTests/ChangeLog
>@@ -1,3 +1,48 @@
>+2018-09-07  Youenn Fablet  <youenn@apple.com>
>+
>+        Add support for unified plan transceivers
>+        https://bugs.webkit.org/show_bug.cgi?id=189390
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Update tests using mock to enforce plan B for now until unified plan mocks are supported.
>+
>+        * fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt:
>+        * fast/mediastream/RTCPeerConnection-icecandidate-event.html:
>+        * fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html:
>+        * fast/mediastream/RTCPeerConnection-inspect-answer.html:
>+        * fast/mediastream/RTCPeerConnection-inspect-offer.html:
>+        * fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html:
>+        * fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html:
>+        * fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html:
>+        * fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html:
>+        * fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html:
>+        * webrtc/calling-peerconnection-once-closed.html:
>+        Once closed, addTransceiver is expected to throw.
>+        * webrtc/libwebrtc/release-while-creating-offer.html:
>+        * webrtc/libwebrtc/release-while-getting-stats.html:
>+        * webrtc/libwebrtc/release-while-setting-local-description.html:
>+        * webrtc/video-getParameters.html: sender and receiver parameters are no longer the same.
>+        Testing them individually.
>+        * webrtc/video-with-receiver.html:
>+        Test is working only with legacy mode.
>+
>+2018-09-06  Daniel Bates  <dabates@apple.com>
>+
>+        [iOS] Add a test to ensure that DOM keyup events have the correct details
>+        https://bugs.webkit.org/show_bug.cgi?id=189327
>+
>+        Reviewed by Wenson Hsieh.
>+
>+        Until we fix <rdar://problem/35282761> DOM keyup events will have incorrect details on iOS
>+        when dispatched in response to a hardware keyboard press. For now, land expected failure results.
>+
>+        * fast/events/ios/keyup-expected.txt: Added.
>+        * fast/events/ios/keyup.html: Added.
>+        * resources/ui-helper.js:
>+        (window.UIHelper.typeCharacter): Escape backslash characters in order to build up
>+        a well-formed UI script to evaluate.
>+
> 2018-09-07  Zalan Bujtas  <zalan@apple.com>
> 
>         [iOS] Unreviewed test gardening.
>diff --git a/LayoutTests/imported/w3c/ChangeLog b/LayoutTests/imported/w3c/ChangeLog
>index 2eccbd80bdcc792388aa0c50434cdb73b8e80a27..98fdf1f4af7f785f710480edffe2d4262ce004ef 100644
>--- a/LayoutTests/imported/w3c/ChangeLog
>+++ b/LayoutTests/imported/w3c/ChangeLog
>@@ -1,3 +1,19 @@
>+2018-09-07  Youenn Fablet  <youenn@apple.com>
>+
>+        Add support for unified plan transceivers
>+        https://bugs.webkit.org/show_bug.cgi?id=189390
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt:
>+        * web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt:
>+        Regression comes from the fact that the sender was never used but transceiver direction is sendrecv.
>+        This might need further clarification in the spec or implementation of libwebrtc.
>+        * web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt:
>+        * web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt:
>+        * web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt:
>+        * web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt:
>+
> 2018-09-05  Youenn Fablet  <youenn@apple.com>
> 
>         Expose RTCRtpSender.setParameters
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt b/LayoutTests/fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt
>index 7ddfc1c64d7ecb2f1f40ab92667babde83889856..80195ba0b6fdc88623f352017bc0aa1a2d26c64c 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt
>@@ -11,7 +11,7 @@ PASS Remote description set
> 
> *** Define sdpMid, badSdpMid, sdpMLineIndex and badSdpMLineIndex for testing
> PASS sdpMLineIndex is not badSdpMLineIndex
>-FAIL sdpMid should not be null.
>+PASS sdpMid is not null
> PASS sdpMid is not badSdpMid
> FAIL promise pc.addIceCandidate(new RTCIceCandidate({candidate: validCandidate, sdpMid: badSdpMid})) fulfilled unexpectedly.
> FAIL promise pc.addIceCandidate(new RTCIceCandidate({candidate: validCandidate, sdpMLineIndex: badSdpMLineIndex})) fulfilled unexpectedly.
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-icecandidate-event.html b/LayoutTests/fast/mediastream/RTCPeerConnection-icecandidate-event.html
>index 48488d551ad9f664862462e37b24aa8f4ba4a146..44e2d3cfe79f82d5edcd0e7b2f27033d8bba444d 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-icecandidate-event.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-icecandidate-event.html
>@@ -8,6 +8,9 @@
>         <script>
>             description("Test RTCPeerConnection 'icecandidate' event and gathering done");
> 
>+            if (window.testRunner)
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
>             if (window.internals)
>                 internals.useMockRTCPeerConnectionFactory("ICECandidates");
> 
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html b/LayoutTests/fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html
>index 4db056479a4b1f75b9e61d561e64074bceabe4a8..ccc2b9210dae437a4bb1d8ad7b935bb15a5f770a 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-iceconnectionstatechange-event.html
>@@ -7,6 +7,8 @@
>     <body>
>         <script>
>             description("Test RTCPeerConnection 'iceconnectionstatechange' event");
>+            if (window.testRunner)
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
> 
>             if (window.internals)
>                 internals.useMockRTCPeerConnectionFactory("ICEConnectionState");
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-answer.html b/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-answer.html
>index 04dd6e388c8f99478b8794d9a263d84d776d384a..ae134285805a075c31f3f987b0e35e5f63424e47 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-answer.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-answer.html
>@@ -6,6 +6,9 @@
>     </head>
>     <body>
>         <script>
>+            if (window.testRunner)
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
>             if (window.internals)
>                 internals.useMockRTCPeerConnectionFactory("");
> 
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-offer.html b/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-offer.html
>index 9c3c20c8b03c39336f985a589979e6f6e935951d..8e636382b0f543d36adb11e8a628d57e36448589 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-offer.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-inspect-offer.html
>@@ -6,6 +6,9 @@
>     </head>
>     <body>
>         <script>
>+            if (window.testRunner)
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
>             if (window.internals)
>                 internals.useMockRTCPeerConnectionFactory("");
> 
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html b/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html
>index 4821e468fb78ec44452f666a978d1a1a5e14957e..e829328bb736b5b5e1259367f58fc0827a7ccd72 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html
>@@ -6,6 +6,9 @@
>     </head>
>     <body>
>         <script>
>+            if (window.testRunner)
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
>             if (window.internals)
>                 internals.useMockRTCPeerConnectionFactory("");
> 
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html b/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html
>index 6887f3b410a582944cca1dd6d64fe4191ec2e914..79e6865193dd6c3f370fe2803e0b1591a5bd59da 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs.html
>@@ -11,9 +11,10 @@
> 
>             description("Test setting up media between two RTCPeerConnection instances with a single SDP dialog.");
> 
>-            if (window.testRunner)
>+            if (window.testRunner) {
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
>                 testRunner.setUserMediaPermission(true);
>-            else {
>+            } else {
>                 debug("This test can not be run without the testRunner");
>                 finishJSTest();
>             }
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html b/LayoutTests/fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html
>index 973e139958ee35295f8c430eff2227944c00bd18..0e3b5fd729dc060cd4dc7b9b5b2387799a79d6a4 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html
>@@ -6,6 +6,9 @@
>     </head>
>     <body>
>         <script>
>+            if (window.testRunner)
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
>             if (window.internals)
>                 internals.useMockRTCPeerConnectionFactory("");
> 
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html b/LayoutTests/fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html
>index 229e4b5c7ebe3f4b85d9d4e8b50e193609241f25..cbd516f3d73aed26c88b7b561c51055df6d9a870 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html
>@@ -6,6 +6,9 @@
>     </head>
>     <body>
>         <script>
>+            if (window.testRunner)
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
>             if (window.internals)
>                 internals.useMockRTCPeerConnectionFactory("");
> 
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html b/LayoutTests/fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html
>index cac37f078a704f8450e0f6b021046b5a53386eaa..81183019b15c9583ff6731a17d6883c45c793362 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html
>@@ -6,6 +6,9 @@
>     </head>
>     <body>
>         <script>
>+            if (window.testRunner)
>+                testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
>             if (window.internals)
>                 internals.useMockRTCPeerConnectionFactory("");
> 
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt
>index 03dcca2d51821d47b6b2f858a8944350c7c67522..35dc84de7b5fb1f702558b73395fb7df43dbd854 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https-expected.txt
>@@ -2,8 +2,8 @@
> FAIL insertDTMF() should succeed if tones contains valid DTMF characters promise_test: Unhandled rejection with value: object "ReferenceError: Can't find variable: RTCDTMFSender"
> FAIL insertDTMF() should throw InvalidCharacterError if tones contains invalid DTMF characters promise_test: Unhandled rejection with value: object "ReferenceError: Can't find variable: RTCDTMFSender"
> FAIL insertDTMF() should throw InvalidStateError if transceiver is stopped assert_throws: function "() => dtmfSender.insertDTMF('')" threw object "TypeError: undefined is not an object (evaluating 'dtmfSender.insertDTMF')" that is not a DOMException InvalidStateError: property "code" is equal to undefined, expected 11
>-FAIL insertDTMF() should throw InvalidStateError if transceiver.currentDirection is recvonly assert_equals: expected (string) "inactive" but got (undefined) undefined
>-FAIL insertDTMF() should throw InvalidStateError if transceiver.currentDirection is inactive assert_equals: expected (string) "inactive" but got (undefined) undefined
>+FAIL insertDTMF() should throw InvalidStateError if transceiver.currentDirection is recvonly assert_equals: expected (string) "inactive" but got (object) null
>+FAIL insertDTMF() should throw InvalidStateError if transceiver.currentDirection is inactive assert_equals: expected (string) "inactive" but got (object) null
> FAIL insertDTMF() should set toneBuffer to provided tones normalized, with old tones overridden promise_test: Unhandled rejection with value: object "ReferenceError: Can't find variable: RTCDTMFSender"
> FAIL insertDTMF() after remove and close should reject assert_throws: function "() =>
>                       dtmfSender.insertDTMF('123')" threw object "TypeError: undefined is not an object (evaluating 'dtmfSender.insertDTMF')" that is not a DOMException InvalidStateError: property "code" is equal to undefined, expected 11
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt
>index 78565279eb53ebda2af9176227842185b4760859..067d6437f65ff37e6377d0f112b0e652506508c7 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt
>@@ -5,6 +5,6 @@ PASS addTrack with single track argument and single mediaStream should succeed
> PASS addTrack with single track argument and multiple mediaStreams should succeed 
> PASS Adding the same track multiple times should throw InvalidAccessError 
> PASS addTrack with existing sender with null track, same kind, and recvonly direction should reuse sender 
>-PASS addTrack with existing sender with null track, same kind, and sendrecv direction should create new sender 
>+FAIL addTrack with existing sender with null track, same kind, and sendrecv direction should create new sender assert_not_equals: got disallowed value object "[object RTCRtpSender]"
> PASS addTrack with existing sender with null track, different kind, and recvonly direction should create new sender 
> 
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt
>index ff610595f475ecbabec0b25f78ead629657d07a5..81ff9f88b82575f514797ee3536546df2e60eda5 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addTransceiver-expected.txt
>@@ -1,11 +1,11 @@
> 
> PASS addTransceiver() with string argument as invalid kind should throw TypeError 
>-FAIL addTransceiver('audio') should return an audio transceiver assert_equals: expected (object) null but got (undefined) undefined
>-FAIL addTransceiver('video') should return a video transceiver assert_equals: expected "remote video" but got ""
>+PASS addTransceiver('audio') should return an audio transceiver 
>+PASS addTransceiver('video') should return a video transceiver 
> PASS addTransceiver() with direction sendonly should have result transceiver.direction be the same 
> PASS addTransceiver() with direction inactive should have result transceiver.direction be the same 
> PASS addTransceiver() with invalid direction should throw TypeError 
>-FAIL addTransceiver(track) should have result with sender.track be given track assert_equals: expected "remote audio" but got ""
>+PASS addTransceiver(track) should have result with sender.track be given track 
> PASS addTransceiver(track) multiple times should create multiple transceivers 
> FAIL addTransceiver() with rid containing invalid non-alphanumeric characters should throw TypeError assert_throws: function "() =>
>       pc.addTransceiver('audio', {
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt
>index f83abb2959d4be93f17809e85e2032566d1073d4..a812574e1293102e4678d4864b9f8582353bab36 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-onnegotiationneeded-expected.txt
>@@ -3,9 +3,9 @@ Harness Error (TIMEOUT), message = null
> 
> PASS Creating first data channel should fire negotiationneeded event 
> PASS calling createDataChannel twice should fire negotiationneeded event once 
>-TIMEOUT addTransceiver() should fire negotiationneeded event Test timed out
>-PASS Calling addTransceiver() twice should fire negotiationneeded event once 
>-PASS Calling both addTransceiver() and createDataChannel() should fire negotiationneeded event once 
>+PASS addTransceiver() should fire negotiationneeded event 
>+FAIL Calling addTransceiver() twice should fire negotiationneeded event once assert_unreached: Pending promise should never be resolved. Instead it is fulfilled with: [object Object] Reached unreachable code
>+FAIL Calling both addTransceiver() and createDataChannel() should fire negotiationneeded event once assert_unreached: Pending promise should never be resolved. Instead it is fulfilled with: [object Object] Reached unreachable code
> PASS negotiationneeded event should not fire if signaling state is not stable 
>-NOTRUN negotiationneeded event should fire only after signaling state go back to stable 
>+TIMEOUT negotiationneeded event should fire only after signaling state go back to stable Test timed out
> 
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt
>index aae4e59f0e1c16eb760bf292261764b61db87346..b0943ddfc683feee38c2fbe16730fc1b1d74981b 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https-expected.txt
>@@ -5,10 +5,10 @@ PASS addTransceiver - Calling removeTrack on different connection that is closed
> PASS addTrack - Calling removeTrack on different connection that is closed should throw InvalidStateError 
> FAIL addTransceiver - Calling removeTrack on different connection should throw InvalidAccessError assert_throws: function "() => pc2.removeTrack(sender)" did not throw
> FAIL addTrack - Calling removeTrack on different connection should throw InvalidAccessError assert_throws: function "() => pc2.removeTrack(sender)" did not throw
>-FAIL addTransceiver - Calling removeTrack with valid sender should set sender.track to null assert_equals: expected (object) null but got (undefined) undefined
>+FAIL addTransceiver - Calling removeTrack with valid sender should set sender.track to null assert_equals: direction should not be altered expected "sendrecv" but got "recvonly"
> PASS addTrack - Calling removeTrack with valid sender should set sender.track to null 
>-FAIL Calling removeTrack with currentDirection sendrecv should set direction to recvonly assert_equals: expected (object) null but got (undefined) undefined
>-FAIL Calling removeTrack with currentDirection sendonly should set direction to inactive assert_equals: expected (object) null but got (undefined) undefined
>-FAIL Calling removeTrack with currentDirection recvonly should not change direction assert_equals: expected (object) null but got (undefined) undefined
>-FAIL Calling removeTrack with currentDirection inactive should not change direction assert_equals: expected (object) null but got (undefined) undefined
>+FAIL Calling removeTrack with currentDirection sendrecv should set direction to recvonly assert_equals: expected "sendrecv" but got "sendonly"
>+PASS Calling removeTrack with currentDirection sendonly should set direction to inactive 
>+FAIL Calling removeTrack with currentDirection recvonly should not change direction assert_equals: expected "recvonly" but got "inactive"
>+PASS Calling removeTrack with currentDirection inactive should not change direction 
> 
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt
>index 897129a16eae3af755ca5bcb8c41688848b312c8..7b0d13d934de63e06d1684a17a2d7e5e4a918e25 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-setDescription-transceiver-expected.txt
>@@ -1,7 +1,7 @@
> 
>-FAIL setLocalDescription(offer) with m= section should assign mid to corresponding transceiver assert_equals: Expect transceiver.mid to set to valid string value expected "string" but got "object"
>+PASS setLocalDescription(offer) with m= section should assign mid to corresponding transceiver 
> FAIL setRemoteDescription(offer) with m= section and no existing transceiver should create corresponding transceiver promise_test: Unhandled rejection with value: object "TypeError: pc2.setRemoteDescrption is not a function. (In 'pc2.setRemoteDescrption(offer)', 'pc2.setRemoteDescrption' is undefined)"
>-FAIL setLocalDescription(rollback) should unset transceiver.mid assert_not_equals: got disallowed value null
>-FAIL setLocalDescription(rollback) should only unset transceiver mids associated with current round assert_not_equals: got disallowed value null
>-FAIL setRemoteDescription(rollback) should remove newly created transceiver from transceiver list assert_equals: expected 1 but got 0
>+FAIL setLocalDescription(rollback) should unset transceiver.mid promise_test: Unhandled rejection with value: object "InvalidStateError: Description type incompatible with current signaling state"
>+FAIL setLocalDescription(rollback) should only unset transceiver mids associated with current round promise_test: Unhandled rejection with value: object "InvalidStateError: Description type incompatible with current signaling state"
>+FAIL setRemoteDescription(rollback) should remove newly created transceiver from transceiver list promise_test: Unhandled rejection with value: object "InvalidStateError: Description type incompatible with current signaling state"
> 
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt
>index f95f5c6eaab31ca57fcc9b55599d1e4f6222b2db..ba722a0ba013ad3d99e5ae2a1fecad9b14b0a507 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setDirection-expected.txt
>@@ -1,5 +1,5 @@
> 
>-FAIL setDirection should change transceiver.direction assert_equals: expected (object) null but got (undefined) undefined
>+PASS setDirection should change transceiver.direction 
> PASS setDirection with same direction should have no effect 
>-FAIL setDirection should change transceiver.direction independent of transceiver.currentDirection assert_equals: expected (object) null but got (undefined) undefined
>+FAIL setDirection should change transceiver.direction independent of transceiver.currentDirection assert_equals: expected "recvonly" but got "inactive"
> 
>diff --git a/LayoutTests/webrtc/calling-peerconnection-once-closed.html b/LayoutTests/webrtc/calling-peerconnection-once-closed.html
>index c72d93864463afae666922f07e1c4c0df5655f4c..c6a41bd55cb2690290ca7b3cc61bd0753082736b 100644
>--- a/LayoutTests/webrtc/calling-peerconnection-once-closed.html
>+++ b/LayoutTests/webrtc/calling-peerconnection-once-closed.html
>@@ -42,7 +42,7 @@ promise_test(() => {
> 
> promise_test(() => {
>     return closedConnection().then((connection) => {
>-        connection.addTransceiver("video");
>+        assert_throws("InvalidStateError", () => { connection.addTransceiver("video"); });
>     });
> }, "Ensuring closed connection addTransceiver does not crash");
> 
>diff --git a/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html b/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html
>index cc7bed9f89d66965e2cf6d07249e2274202eaced..518aef79865ef2661235148654c70478044be508 100644
>--- a/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html
>+++ b/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html
>@@ -7,6 +7,9 @@
> <script>
> self.jsTestIsAsync = true;
> 
>+if (window.testRunner)
>+    testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
> if (window.internals)
>     internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileCreatingOffer");
> 
>diff --git a/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html b/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html
>index a94449ef60fcebeea2812e9c34a78e9189febb26..3f120da4004254fbd5e63d40c39d3717354aec6b 100644
>--- a/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html
>+++ b/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html
>@@ -7,6 +7,9 @@
> <script>
> self.jsTestIsAsync = true;
> 
>+if (window.testRunner)
>+    testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
> if (window.internals)
>     internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileGettingStats");
> 
>diff --git a/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html b/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html
>index 62e899e7f1fa755be8833a75720b9dc9cbbd6ef4..fa9c6534a6544d3bd943b4888680e082b554df3c 100644
>--- a/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html
>+++ b/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html
>@@ -10,6 +10,9 @@ self.jsTestIsAsync = true;
> // Silence unhandled rejection messages.
> window.onunhandledrejection = () => false;
> 
>+if (window.testRunner)
>+    testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
> if (window.internals)
>     internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileSettingDescription");
> 
>diff --git a/LayoutTests/webrtc/video-getParameters.html b/LayoutTests/webrtc/video-getParameters.html
>index e3a2931a296a68007f55f06e745a3c524856adb7..9e43930cecd5e2eb7e2911d0ef3436163fdb848b 100644
>--- a/LayoutTests/webrtc/video-getParameters.html
>+++ b/LayoutTests/webrtc/video-getParameters.html
>@@ -42,16 +42,13 @@ promise_test((test) => {
> 
>         senderParameters.encodings[0].fec.ssrc = 1;
>         senderParameters.encodings[0].rtx.ssrc = 1;
>-        receiverParameters.encodings[0].fec.ssrc = 1;
>-        receiverParameters.encodings[0].rtx.ssrc = 1;
> 
>         senderParameters.transactionId = "";
>         receiverParameters.transactionId = "";
> 
>-        assert_equals(JSON.stringify(senderParameters), JSON.stringify(receiverParameters), "testing sender vs. receiver parameters");
>-
>         senderParameters.encodings[0].ssrc = 1;
>-        assert_equals(JSON.stringify(senderParameters), '{"codecs":[],"degradationPreference":"balanced","encodings":[{"active":true,"fec":{"ssrc":1},"maxBitrate":0,"maxFramerate":0,"priority":"medium","rid":"","rtx":{"ssrc":1},"scaleResolutionDownBy":1,"ssrc":1}],"headerExtensions":[],"transactionId":""}', "Testing sanitized parameters");
>+        assert_equals(JSON.stringify(senderParameters), '{"codecs":[],"degradationPreference":"balanced","encodings":[{"active":true,"fec":{"ssrc":1},"maxBitrate":0,"maxFramerate":0,"priority":"medium","rid":"","rtx":{"ssrc":1},"scaleResolutionDownBy":1,"ssrc":1}],"headerExtensions":[],"transactionId":""}', "Testing sanitized sender parameters");
>+        assert_equals(JSON.stringify(receiverParameters), '{"codecs":[],"degradationPreference":"balanced","encodings":[],"headerExtensions":[],"transactionId":""}', "Testing sanitized receiver parameters");
>     });
> }, "Basic video stats");
>         </script>
>diff --git a/LayoutTests/webrtc/video-with-receiver.html b/LayoutTests/webrtc/video-with-receiver.html
>index c0cf821acef6cf452a0ff2bdcce0101f49e0c956..63bceef455fd744d7c4a37808e66edee3419ebfd 100644
>--- a/LayoutTests/webrtc/video-with-receiver.html
>+++ b/LayoutTests/webrtc/video-with-receiver.html
>@@ -14,6 +14,9 @@
> video = document.getElementById("video");
> canvas = document.getElementById("canvas");
> 
>+if (window.testRunner)
>+    testRunner.setWebRTCUnifiedPlanEnabled(false);
>+
> function testImage()
> {
>     canvas.width = video.videoWidth;

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Attachments on bug 189390: 349115 | 349120 | 349132 | 349149 | 349152 | 349154 | 349188