Attachment 348389 Details for Bug 189068 – Patch

[patch] Patch

bug-189068-20180828212133.patch (text/plain), 22.40 KB, created by youenn fablet on 2018-08-28 21:21:35 PDT

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Creator: youenn fablet

Created: 2018-08-28 21:21:35 PDT

Size: 22.40 KB

patch

obsolete

>Subversion Revision: 235451
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index c74c169a3474bbc2a1565b1ef38774e8e1e55f93..eb12b696c1fa47713f48b6701be27b50e9cc4916 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,27 @@
>+2018-08-28  Youenn Fablet  <youenn@apple.com>
>+
>+        Add a runtime flag for WebRTC unified plan
>+        https://bugs.webkit.org/show_bug.cgi?id=189068
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Covered by existing updated tests.
>+        Main change is to call addTrack with a stream parameter so that on the other side, the track will be tied to a stream.
>+        Receive-only case in unified plan is not yet supported.
>+        This will be supported in follow-up patches.
>+
>+        * Modules/mediastream/RTCPeerConnection.cpp:
>+        (WebCore::RTCPeerConnection::setConfiguration): Set the correct exception type. Drive by fix.
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
>+        (WebCore::LibWebRTCMediaEndpoint::setConfiguration): Activate unified plan based on runtime flag.
>+        (WebCore::LibWebRTCMediaEndpoint::addTrack): Do not use AddStream in case of unified plan.
>+        (WebCore::LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveAudio const):
>+        (WebCore::LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveVideo const):
>+        (WebCore::LibWebRTCMediaEndpoint::doCreateOffer): Use legacy webrtc option for receive only cases only in plan B case.
>+        * page/RuntimeEnabledFeatures.h:
>+        (WebCore::RuntimeEnabledFeatures::webRTCUnifiedPlanEnabled const):
>+        (WebCore::RuntimeEnabledFeatures::setWebRTCUnifiedPlanEnabled):
>+
> 2018-08-28  Youenn Fablet  <youenn@apple.com>
> 
>         IDBDatabase should not return true to hasPendingActivity after being stopped
>diff --git a/Source/WebKit/ChangeLog b/Source/WebKit/ChangeLog
>index 960a1b818c0930f3f865cf845e50ad25c8f0f7c3..e521ddb1936b58d00dfbdb9e32c74321bf20e864 100644
>--- a/Source/WebKit/ChangeLog
>+++ b/Source/WebKit/ChangeLog
>@@ -1,3 +1,14 @@
>+2018-08-28  Youenn Fablet  <youenn@apple.com>
>+
>+        Add a runtime flag for WebRTC unified plan
>+        https://bugs.webkit.org/show_bug.cgi?id=189068
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * Shared/WebPreferences.yaml:
>+        * WebProcess/InjectedBundle/InjectedBundle.cpp:
>+        (WebKit::InjectedBundle::overrideBoolPreferenceForTestRunner):
>+
> 2018-08-27  Youenn Fablet  <youenn@apple.com>
> 
>         Remove WebRTC legacy API implementation
>diff --git a/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp b/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>index a04f81e1fb12727cffbd508918b23c760a22baef..786a0e1bc5a9604779e5cf039bc8d2d3ef52779d 100644
>--- a/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>+++ b/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp
>@@ -370,7 +370,7 @@ ExceptionOr<void> RTCPeerConnection::setConfiguration(RTCConfiguration&& configu
>         return servers.releaseException();
> 
>     if (!m_backend->setConfiguration({ servers.releaseReturnValue(), configuration.iceTransportPolicy, configuration.bundlePolicy, configuration.iceCandidatePoolSize }))
>-        return Exception { InvalidAccessError, "Bad Configuration Parameters" };
>+        return Exception { InvalidModificationError, "Bad Configuration Parameters" };
> 
>     m_configuration = WTFMove(configuration);
>     return { };
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>index 0ae30342f30a118f592e1dd5ec1cb1b82cfe8d46..0ba49473714dd9e2747f31d26bcfdca5804fa94f 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>@@ -79,6 +79,9 @@ LibWebRTCMediaEndpoint::LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend& p
> 
> bool LibWebRTCMediaEndpoint::setConfiguration(LibWebRTCProvider& client, webrtc::PeerConnectionInterface::RTCConfiguration&& configuration)
> {
>+    if (RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
>+        configuration.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
>+
>     if (!m_backend) {
>         m_backend = client.createPeerConnection(*this, WTFMove(configuration));
>         return !!m_backend;
>@@ -196,12 +199,14 @@ bool LibWebRTCMediaEndpoint::addTrack(RTCRtpSender& sender, MediaStreamTrack& tr
> {
>     ASSERT(m_backend);
> 
>-    String mediaStreamId = mediaStreamIds.isEmpty() ? createCanonicalUUIDString() : mediaStreamIds[0];
>-    rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream = m_localStreams.get(mediaStreamId);
>-    if (!mediaStream) {
>-        mediaStream = m_peerConnectionFactory.CreateLocalMediaStream(mediaStreamId.utf8().data());
>-        m_backend->AddStream(mediaStream);
>-        m_localStreams.add(mediaStreamId, mediaStream);
>+    if (!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled()) {
>+        String mediaStreamId = mediaStreamIds.isEmpty() ? createCanonicalUUIDString() : mediaStreamIds[0];
>+        rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream = m_localStreams.get(mediaStreamId);
>+        if (!mediaStream) {
>+            mediaStream = m_peerConnectionFactory.CreateLocalMediaStream(mediaStreamId.utf8().data());
>+            m_backend->AddStream(mediaStream);
>+            m_localStreams.add(mediaStreamId, mediaStream);
>+        }
>     }
> 
>     std::vector<std::string> ids;
>@@ -245,6 +250,7 @@ void LibWebRTCMediaEndpoint::removeTrack(RTCRtpSender& sender)
> 
> bool LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveAudio() const
> {
>+    ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled());
>     for (const auto& transceiver : m_peerConnectionBackend.connection().getTransceivers()) {
>         if (transceiver->sender().trackKind() != "audio")
>             continue;
>@@ -260,6 +266,7 @@ bool LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveAudio() const
> 
> bool LibWebRTCMediaEndpoint::shouldOfferAllowToReceiveVideo() const
> {
>+    ASSERT(!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled());
>     for (const auto& transceiver : m_peerConnectionBackend.connection().getTransceivers()) {
>         if (transceiver->sender().trackKind() != "video")
>             continue;
>@@ -281,11 +288,13 @@ void LibWebRTCMediaEndpoint::doCreateOffer(const RTCOfferOptions& options)
>     webrtc::PeerConnectionInterface::RTCOfferAnswerOptions rtcOptions;
>     rtcOptions.ice_restart = options.iceRestart;
>     rtcOptions.voice_activity_detection = options.voiceActivityDetection;
>-    // FIXME: offer_to_receive_audio and offer_to_receive_video are used as libwebrtc does not support transceivers yet.
>-    if (shouldOfferAllowToReceiveAudio())
>-        rtcOptions.offer_to_receive_audio = webrtc::PeerConnectionInterface::RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>-    if (shouldOfferAllowToReceiveVideo())
>-        rtcOptions.offer_to_receive_video = webrtc::PeerConnectionInterface::RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>+
>+    if (!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled()) {
>+        if (shouldOfferAllowToReceiveAudio())
>+            rtcOptions.offer_to_receive_audio = webrtc::PeerConnectionInterface::RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>+        if (shouldOfferAllowToReceiveVideo())
>+            rtcOptions.offer_to_receive_video = webrtc::PeerConnectionInterface::RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>+    }
>     m_backend->CreateOffer(&m_createSessionDescriptionObserver, rtcOptions);
> }
> 
>diff --git a/Source/WebCore/page/RuntimeEnabledFeatures.h b/Source/WebCore/page/RuntimeEnabledFeatures.h
>index cd9193af6a41dc4d2ed94d1bc6e14c8040394750..a60d8a38ef08f63726483181b7d2b752c6a82467 100644
>--- a/Source/WebCore/page/RuntimeEnabledFeatures.h
>+++ b/Source/WebCore/page/RuntimeEnabledFeatures.h
>@@ -116,6 +116,8 @@ public:
> #endif
> 
> #if ENABLE(WEB_RTC)
>+    bool webRTCUnifiedPlanEnabled() const { return m_isWebRTCUnifiedPlanEnabled; }
>+    void setWebRTCUnifiedPlanEnabled(bool isEnabled) { m_isWebRTCUnifiedPlanEnabled = isEnabled; }
>     bool peerConnectionEnabled() const { return m_isPeerConnectionEnabled; }
>     void setPeerConnectionEnabled(bool isEnabled) { m_isPeerConnectionEnabled = isEnabled; }
>     bool mdnsICECandidatesEnabled() const { return m_mdnsICECandidatesEnabled; }
>@@ -319,6 +321,7 @@ private:
> #endif
> 
> #if ENABLE(WEB_RTC)
>+    bool m_isWebRTCUnifiedPlanEnabled { true };
>     bool m_isPeerConnectionEnabled { true };
>     bool m_mdnsICECandidatesEnabled { false };
> #endif
>diff --git a/Source/WebKit/Shared/WebPreferences.yaml b/Source/WebKit/Shared/WebPreferences.yaml
>index 97bb9bd0d443dacdd11df080501474e869e7ead1..1f41929008bbecd0ca6d0b6e327f118da9d690ba 100644
>--- a/Source/WebKit/Shared/WebPreferences.yaml
>+++ b/Source/WebKit/Shared/WebPreferences.yaml
>@@ -494,6 +494,15 @@ PeerConnectionEnabled:
>   webcoreBinding: RuntimeEnabledFeatures
>   condition: ENABLE(WEB_RTC)
> 
>+WebRTCUnifiedPlanEnabled:
>+  type: bool
>+  defaultValue: true
>+  webcoreBinding: RuntimeEnabledFeatures
>+  condition: ENABLE(WEB_RTC)
>+  humanReadableName: "WebRTC Unified Plan"
>+  humanReadableDescription: "Use WebRTC Unified Plan"
>+  category: experimental
>+
> UseLegacyTextAlignPositionedElementBehavior:
>   type: bool
>   defaultValue: false
>diff --git a/Source/WebKit/WebProcess/InjectedBundle/InjectedBundle.cpp b/Source/WebKit/WebProcess/InjectedBundle/InjectedBundle.cpp
>index 38e2bcf6fcfbc158980814455cf3027b9187961d..ff0e185d08aa30e4c12789b7e88a33dc1965cf47 100644
>--- a/Source/WebKit/WebProcess/InjectedBundle/InjectedBundle.cpp
>+++ b/Source/WebKit/WebProcess/InjectedBundle/InjectedBundle.cpp
>@@ -241,6 +241,8 @@ void InjectedBundle::overrideBoolPreferenceForTestRunner(WebPageGroupProxy* page
> #if ENABLE(WEB_RTC)
>     if (preference == "WebKitMDNSICECandidatesEnabled")
>         RuntimeEnabledFeatures::sharedFeatures().setMDNSICECandidatesEnabled(enabled);
>+    if (preference == "WebKitWebRTCUnifiedPlanEnabled")
>+        RuntimeEnabledFeatures::sharedFeatures().setWebRTCUnifiedPlanEnabled(enabled);
> #endif
> 
>     if (preference == "WebKitIsSecureContextAttributeEnabled") {
>diff --git a/Tools/ChangeLog b/Tools/ChangeLog
>index d5340c5aaf3935276ed42858c2f2a15a5ac5648e..48869f3954557107b11a4693d1179be3bd5430d1 100644
>--- a/Tools/ChangeLog
>+++ b/Tools/ChangeLog
>@@ -1,3 +1,15 @@
>+2018-08-28  Youenn Fablet  <youenn@apple.com>
>+
>+        Add a runtime flag for WebRTC unified plan
>+        https://bugs.webkit.org/show_bug.cgi?id=189068
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * WebKitTestRunner/InjectedBundle/Bindings/TestRunner.idl:
>+        * WebKitTestRunner/InjectedBundle/TestRunner.cpp:
>+        (WTR::TestRunner::setWebRTCUnifiedPlanEnabled):
>+        * WebKitTestRunner/InjectedBundle/TestRunner.h:
>+
> 2018-08-27  Youenn Fablet  <youenn@apple.com>
> 
>         Remove WebRTC legacy API implementation
>diff --git a/Tools/WebKitTestRunner/InjectedBundle/Bindings/TestRunner.idl b/Tools/WebKitTestRunner/InjectedBundle/Bindings/TestRunner.idl
>index 9a55d9213262eaae1fb8c7f1ecca92051f0556a2..a46c46e5b0f94533ff4511f298ea93fab32b4d07 100644
>--- a/Tools/WebKitTestRunner/InjectedBundle/Bindings/TestRunner.idl
>+++ b/Tools/WebKitTestRunner/InjectedBundle/Bindings/TestRunner.idl
>@@ -321,6 +321,7 @@ interface TestRunner {
>     void setOpenPanelFiles(object filesArray);
> 
>     void setMDNSICECandidatesEnabled(boolean value);
>+    void setWebRTCUnifiedPlanEnabled(boolean value);
>     void setCustomUserAgent(DOMString userAgent);
> 
>     void terminateNetworkProcess();
>diff --git a/Tools/WebKitTestRunner/InjectedBundle/TestRunner.cpp b/Tools/WebKitTestRunner/InjectedBundle/TestRunner.cpp
>index ef98e634c162d50f02c6d6d7adb3582db1e0d254..d4bcaaef023ac95122c1ef720c712d19c6ae20ba 100644
>--- a/Tools/WebKitTestRunner/InjectedBundle/TestRunner.cpp
>+++ b/Tools/WebKitTestRunner/InjectedBundle/TestRunner.cpp
>@@ -442,6 +442,13 @@ void TestRunner::setMDNSICECandidatesEnabled(bool enabled)
>     WKBundleOverrideBoolPreferenceForTestRunner(injectedBundle.bundle(), injectedBundle.pageGroup(), key.get(), enabled);
> }
> 
>+void TestRunner::setWebRTCUnifiedPlanEnabled(bool enabled)
>+{
>+    WKRetainPtr<WKStringRef> key(AdoptWK, WKStringCreateWithUTF8CString("WebKitWebRTCUnifiedPlanEnabled"));
>+    auto& injectedBundle = InjectedBundle::singleton();
>+    WKBundleOverrideBoolPreferenceForTestRunner(injectedBundle.bundle(), injectedBundle.pageGroup(), key.get(), enabled);
>+}
>+
> void TestRunner::setCustomUserAgent(JSStringRef userAgent)
> {
>     WKRetainPtr<WKStringRef> messageName(AdoptWK, WKStringCreateWithUTF8CString("SetCustomUserAgent"));
>diff --git a/Tools/WebKitTestRunner/InjectedBundle/TestRunner.h b/Tools/WebKitTestRunner/InjectedBundle/TestRunner.h
>index 92964889a9a6ec606672a3c4a23862130dc3eeab..bb7a0cfd4e95873cecd2cee69159ad0cd79c1c8c 100644
>--- a/Tools/WebKitTestRunner/InjectedBundle/TestRunner.h
>+++ b/Tools/WebKitTestRunner/InjectedBundle/TestRunner.h
>@@ -131,6 +131,7 @@ public:
>     void setEncryptedMediaAPIEnabled(bool);
>     void setMediaDevicesEnabled(bool);
>     void setMDNSICECandidatesEnabled(bool);
>+    void setWebRTCUnifiedPlanEnabled(bool);
>     void setCustomUserAgent(JSStringRef);
> 
>     // Special DOM functions.
>diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
>index aebc9939977504dda94aea41fe1f2a51f266a9bc..c8d8eef590b533efad12c12e8008476871c811ac 100644
>--- a/LayoutTests/ChangeLog
>+++ b/LayoutTests/ChangeLog
>@@ -1,3 +1,17 @@
>+2018-08-28  Youenn Fablet  <youenn@apple.com>
>+
>+        Add a runtime flag for WebRTC unified plan
>+        https://bugs.webkit.org/show_bug.cgi?id=189068
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * fast/mediastream/RTCPeerConnection-add-removeTrack-expected.txt:
>+        * webrtc/peer-connection-audio-mute2.html:
>+        * webrtc/peer-connection-remote-audio-mute2.html:
>+        * webrtc/video-addTrack-expected.txt:
>+        * webrtc/video-addTrack.html:
>+        * webrtc/video-addTransceiver.html:
>+
> 2018-08-27  Youenn Fablet  <youenn@apple.com>
> 
>         Remove WebRTC legacy API implementation
>diff --git a/LayoutTests/imported/w3c/ChangeLog b/LayoutTests/imported/w3c/ChangeLog
>index 75ac6a409b0aa8bb0839be0dca254f24129abb3e..abb9d1ff458ee439bec630a4ed19529e4c454170 100644
>--- a/LayoutTests/imported/w3c/ChangeLog
>+++ b/LayoutTests/imported/w3c/ChangeLog
>@@ -1,3 +1,13 @@
>+2018-08-28  Youenn Fablet  <youenn@apple.com>
>+
>+        Add a runtime flag for WebRTC unified plan
>+        https://bugs.webkit.org/show_bug.cgi?id=189068
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * web-platform-tests/webrtc/RTCConfiguration-iceServers-expected.txt:
>+        * web-platform-tests/webrtc/RTCPeerConnection-addTrack.https-expected.txt:
>+
> 2018-08-27  Youenn Fablet  <youenn@apple.com>
> 
>         Remove WebRTC legacy API implementation
>diff --git a/LayoutTests/fast/mediastream/RTCPeerConnection-add-removeTrack-expected.txt b/LayoutTests/fast/mediastream/RTCPeerConnection-add-removeTrack-expected.txt
>index 89bcf5203409ee1ecc6ce961918332765fc4d761..d81f83330a233d7b7360efc1b9e2f9e6cc3367c4 100644
>--- a/LayoutTests/fast/mediastream/RTCPeerConnection-add-removeTrack-expected.txt
>+++ b/LayoutTests/fast/mediastream/RTCPeerConnection-add-removeTrack-expected.txt
>@@ -33,10 +33,10 @@ PASS transceiver.receiver is receiver
> PASS transceiver.stopped is false
> Try to add same track again
> PASS sender = pc.addTrack(track, stream) threw exception InvalidAccessError: The object does not support the operation or argument..
>-FAIL sender2 = pc.addTrack(track2, stream, stream2) should not throw exception. Threw exception InvalidAccessError: Unable to add track.
>+PASS sender2 = pc.addTrack(track2, stream, stream2) did not throw exception.
> PASS pc.getSenders().length is 2
> PASS pc.getSenders()[0] is sender
>-FAIL pc.getSenders()[1] should be undefined (of type undefined). Was [object RTCRtpSender] (of type object).
>+PASS pc.getSenders()[1] is sender2
> PASS pc.removeTrack(sender) did not throw exception.
> Sender is still in getSenders() list
> PASS pc.getSenders().length is 2
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCConfiguration-iceServers-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCConfiguration-iceServers-expected.txt
>index cd680aa4a7e714a85baf50e1b4d687bffad6ffa8..65c3e261965dc9cffd10e52f1e90492a6d854532 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCConfiguration-iceServers-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCConfiguration-iceServers-expected.txt
>@@ -73,7 +73,7 @@ FAIL new RTCPeerConnection(config) - with invalid stun url should throw SyntaxEr
> FAIL setConfiguration(config) - with invalid stun url should throw SyntaxError assert_throws: function "() =>
>       makePc({ iceServers: [{
>         urls: 'stun://example.org/foo?x=y'
>-      }] })" threw object "InvalidAccessError: Bad Configuration Parameters" that is not a DOMException SyntaxError: property "code" is equal to 15, expected 12
>+      }] })" threw object "InvalidModificationError: Bad Configuration Parameters" that is not a DOMException SyntaxError: property "code" is equal to 13, expected 12
> FAIL new RTCPeerConnection(config) - with empty urls and credentialType password should succeed assert_equals: expected (string) "password" but got (undefined) undefined
> FAIL setConfiguration(config) - with empty urls and credentialType password should succeed assert_equals: expected (string) "password" but got (undefined) undefined
> FAIL new RTCPeerConnection(config) - with empty urls and credentialType oauth should succeed assert_equals: expected (string) "oauth" but got (undefined) undefined
>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 78a29d490eb3a616f0e8aec2c00b4aaa7b8aa635..78565279eb53ebda2af9176227842185b4760859 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
>@@ -2,7 +2,7 @@
> PASS addTrack when pc is closed should throw InvalidStateError 
> PASS addTrack with single track argument and no mediaStream should succeed 
> PASS addTrack with single track argument and single mediaStream should succeed 
>-FAIL addTrack with single track argument and multiple mediaStreams should succeed promise_test: Unhandled rejection with value: object "InvalidAccessError: Unable to add track"
>+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 
>diff --git a/LayoutTests/webrtc/peer-connection-audio-mute2.html b/LayoutTests/webrtc/peer-connection-audio-mute2.html
>index 9dda885fd39ffdb72e6f7f8a44f94697a920dcf0..caa757f85614f4112ef7e0eee76fc6cde28e235d 100644
>--- a/LayoutTests/webrtc/peer-connection-audio-mute2.html
>+++ b/LayoutTests/webrtc/peer-connection-audio-mute2.html
>@@ -17,7 +17,7 @@
>             var remoteStream;
>             return new Promise((resolve, reject) => {
>                 createConnections((firstConnection) => {
>-                    firstConnection.addTrack(localTrack);
>+                    firstConnection.addTrack(localTrack, localStream);
>                 }, (secondConnection) => {
>                     secondConnection.ontrack = (trackEvent) => {
>                         remoteStream = trackEvent.streams[0];
>diff --git a/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html b/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html
>index 1ba5d33772ae9b9191d916a1ea3a6011e86a1629..ccd2eb29d4f54c9ab604586b9da32199316901f0 100644
>--- a/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html
>+++ b/LayoutTests/webrtc/peer-connection-remote-audio-mute2.html
>@@ -16,7 +16,7 @@
>             var remoteStream;
>             return new Promise((resolve, reject) => {
>                 createConnections((firstConnection) => {
>-                    firstConnection.addTrack(localStream.getAudioTracks()[0]);
>+                    firstConnection.addTrack(localStream.getAudioTracks()[0], localStream);
>                 }, (secondConnection) => {
>                     secondConnection.ontrack = (trackEvent) => {
>                         remoteStream = trackEvent.streams[0];
>diff --git a/LayoutTests/webrtc/video-addTrack-expected.txt b/LayoutTests/webrtc/video-addTrack-expected.txt
>index 609e289dae91a26d564ff2c518804a70e3ba3044..5af3fbeefde291304dd614d20d3bb4f3bad4924b 100644
>--- a/LayoutTests/webrtc/video-addTrack-expected.txt
>+++ b/LayoutTests/webrtc/video-addTrack-expected.txt
>@@ -1,4 +1,4 @@
>- 
>+
> 
> PASS Basic video exchange with addTrack - waiting for second track before playing 
> PASS Basic video exchange with addTrack - not waiting for second track to play 
>@@ -7,5 +7,5 @@ PASS  track 1, wait = true
> PASS Testing image result, wait = true 
> PASS  track 0, wait = false 
> PASS  track 1, wait = false 
>-FAIL Testing image result, wait = false The index is not in the allowed range.
>+PASS Testing image result, wait = false 
> 
>diff --git a/LayoutTests/webrtc/video-addTrack.html b/LayoutTests/webrtc/video-addTrack.html
>index 137d4e9a7242daea040e9b9ce5f0cd36524cb0db..45a1baa00870259c3b62ccb3d80c6e6a9e4326ef 100644
>--- a/LayoutTests/webrtc/video-addTrack.html
>+++ b/LayoutTests/webrtc/video-addTrack.html
>@@ -51,7 +51,7 @@ function testBasicVideoExchangeWithAddTrack(waitForSecondTrack)
>         return new Promise((resolve, reject) => {
>             createConnections((firstConnection) => {
>                 assert_equals(stream.getTracks().length, 2);
>-                stream.getTracks().forEach(track => firstConnection.addTrack(track));
>+                stream.getTracks().forEach(track => firstConnection.addTrack(track, stream));
>             }, (secondConnection) => {
>                 var count = 0;
>                 secondConnection.ontrack = (trackEvent) => {
>diff --git a/LayoutTests/webrtc/video-addTransceiver.html b/LayoutTests/webrtc/video-addTransceiver.html
>index 58f6773c0adbed467cbe5bf77e8b6a475b9a80d6..3ca754c6e692f8468909bd888ec8836a3ec4a388 100644
>--- a/LayoutTests/webrtc/video-addTransceiver.html
>+++ b/LayoutTests/webrtc/video-addTransceiver.html
>@@ -11,6 +11,8 @@
>         <canvas id="canvas" width="640" height="480"></canvas>
>         <script src ="routines.js"></script>
>         <script>
>+if (window.testRunner)
>+    testRunner.setWebRTCUnifiedPlanEnabled(false);
> 
> promise_test((test) => {
>     var pc = new RTCPeerConnection();

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Attachments on bug 189068: 348362 | 348375 | 348389 | 348412