Attachment 349915 Details for Bug 176281 – Patch

[patch] Patch

bug-176281-20180917113022.patch (text/plain), 9.71 KB, created by youenn fablet on 2018-09-17 11:30:22 PDT

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Creator: youenn fablet

Created: 2018-09-17 11:30:22 PDT

Size: 9.71 KB

patch

obsolete

>Subversion Revision: 236064
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 7c460113fcc6ec4210989614ed31fb8bc0757b6c..100bc5d51a9afbd19c37b87bf4b5d79b74bbac7d 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,25 @@
>+2018-09-17  Youenn Fablet  <youenn@apple.com>
>+
>+        track.onmute isn't called for a remote MediaStreamTrack when its counter part track is removed from the peer connection
>+        https://bugs.webkit.org/show_bug.cgi?id=176281
>+        <rdar://problem/44525674>
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Listen to libwebrtc remove track callbacks.
>+        Implement handling as per https://w3c.github.io/webrtc-pc/#process-remote-track-removal.
>+        This triggers a mute event on the track.
>+
>+        Test: webrtc/remove-track.html
>+
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
>+        (WebCore::LibWebRTCMediaEndpoint::removeRemoteTrack):
>+        (WebCore::LibWebRTCMediaEndpoint::OnRemoveTrack):
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
>+        (WebCore::LibWebRTCPeerConnectionBackend::shouldOfferAllowToReceive const):
>+        Drive by fix: Plan B code path does not mandate having an rtc backend for each sender.
>+
> 2018-09-17  Youenn Fablet  <youenn@apple.com>
> 
>         Enable VCP for iOS and reenable it for MacOS
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>index 8535d2b72bc16e67fe1d0f95252cacf738deb202..b25b564f2be71ae2ea5db0e9d39fbdfdb8e07e28 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>@@ -445,6 +445,22 @@ void LibWebRTCMediaEndpoint::newTransceiver(rtc::scoped_refptr<webrtc::RtpTransc
>     fireTrackEvent(makeRef(newTransceiver.receiver()), newTransceiver.receiver().track(), rtcReceiver->streams(), makeRef(newTransceiver));
> }
> 
>+void LibWebRTCMediaEndpoint::removeRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&& receiver)
>+{
>+    // FIXME: Support plan B code path.
>+    if (!RuntimeEnabledFeatures::sharedFeatures().webRTCUnifiedPlanEnabled())
>+        return;
>+
>+    auto* transceiver = m_peerConnectionBackend.existingTransceiver([&receiver](auto& transceiverBackend) {
>+        auto* rtcTransceiver = transceiverBackend.rtcTransceiver();
>+        return rtcTransceiver && receiver.get() == rtcTransceiver->receiver().get();
>+    });
>+    if (!transceiver)
>+        return;
>+
>+    transceiver->receiver().track().source().setMuted(true);
>+}
>+
> template<typename T>
> std::optional<LibWebRTCMediaEndpoint::Backends> LibWebRTCMediaEndpoint::createTransceiverBackends(T&& trackOrKind, const RTCRtpTransceiverInit& init, LibWebRTCRtpSenderBackend::Source&& source)
> {
>@@ -546,6 +562,14 @@ void LibWebRTCMediaEndpoint::OnTrack(rtc::scoped_refptr<webrtc::RtpTransceiverIn
>     });
> }
> 
>+void LibWebRTCMediaEndpoint::OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver)
>+{
>+    callOnMainThread([protectedThis = makeRef(*this), receiver = WTFMove(receiver)]() mutable {
>+        if (protectedThis->isStopped())
>+            return;
>+        protectedThis->removeRemoteTrack(WTFMove(receiver));
>+    });
>+}
> 
> std::unique_ptr<RTCDataChannelHandler> LibWebRTCMediaEndpoint::createDataChannel(const String& label, const RTCDataChannelInit& options)
> {
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>index 5806a830582d50da7bc61a6441d80e5c65422ef4..177e7aa143f67d9c26975170e785c43640dc39cd 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>@@ -118,6 +118,7 @@ private:
>     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 OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>) final;
> 
>     void OnRenegotiationNeeded() final;
>     void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState) final;
>@@ -135,6 +136,7 @@ private:
>     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 removeRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&&);
> 
>     void fireTrackEvent(Ref<RTCRtpReceiver>&&, Ref<MediaStreamTrack>&&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&, RefPtr<RTCRtpTransceiver>&&);
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>index c21d26bb0ca82ce605da48c11498802182b60121..3c4f99a5b32f6fafd8af3542c4eabe2be0134978 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>@@ -486,8 +486,8 @@ bool LibWebRTCPeerConnectionBackend::shouldOfferAllowToReceive(const char* kind)
>         if (transceiver->direction() != RTCRtpTransceiverDirection::Sendrecv)
>             continue;
> 
>-        auto& backend = static_cast<LibWebRTCRtpSenderBackend&>(*transceiver->sender().backend());
>-        if (!backend.rtcSender())
>+        auto* backend = static_cast<LibWebRTCRtpSenderBackend*>(transceiver->sender().backend());
>+        if (backend && !backend->rtcSender())
>             return true;
>     }
>     return false;
>diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
>index b3d9e0e4a77d9051201318ef18a39af0f82253c8..3edb1a8f4f955f313f8565ea570824fc2a7c97a1 100644
>--- a/LayoutTests/ChangeLog
>+++ b/LayoutTests/ChangeLog
>@@ -1,3 +1,14 @@
>+2018-09-17  Youenn Fablet  <youenn@apple.com>
>+
>+        track.onmute isn't called for a remote MediaStreamTrack when its counter part track is removed from the peer connection
>+        https://bugs.webkit.org/show_bug.cgi?id=176281
>+        <rdar://problem/44525674>
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * webrtc/remove-track-expected.txt: Added.
>+        * webrtc/remove-track.html: Added.
>+
> 2018-09-17  Youenn Fablet  <youenn@apple.com>
> 
>         Enable VCP for iOS and reenable it for MacOS
>diff --git a/LayoutTests/webrtc/remove-track-expected.txt b/LayoutTests/webrtc/remove-track-expected.txt
>new file mode 100644
>index 0000000000000000000000000000000000000000..de6c3c260587732c7619df5db0b7b7ccf014a86a
>--- /dev/null
>+++ b/LayoutTests/webrtc/remove-track-expected.txt
>@@ -0,0 +1,5 @@
>+
>+PASS Setup audio video exchange 
>+PASS Remove video track 
>+PASS Remove audio track 
>+
>diff --git a/LayoutTests/webrtc/remove-track.html b/LayoutTests/webrtc/remove-track.html
>new file mode 100644
>index 0000000000000000000000000000000000000000..827f522e42242801692bb94408766dd173ed6fd8
>--- /dev/null
>+++ b/LayoutTests/webrtc/remove-track.html
>@@ -0,0 +1,72 @@
>+<!doctype html>
>+<html>
>+    <head>
>+        <meta charset="utf-8">
>+        <title>Testing basic video exchange from offerer to receiver</title>
>+        <script src="../resources/testharness.js"></script>
>+        <script src="../resources/testharnessreport.js"></script>
>+    </head>
>+    <body>
>+        <script src ="routines.js"></script>
>+        <script>
>+let firstConnection, secondConnection;
>+let stream;
>+let remoteAudioTrack, remoteVideoTrack;
>+promise_test(async (test) => {
>+    if (window.testRunner)
>+        testRunner.setUserMediaPermission(true);
>+
>+    stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
>+    await new Promise((resolve, reject) => {
>+        createConnections((connection) => {
>+            firstConnection = connection;
>+            firstConnection.addTrack(stream.getVideoTracks()[0], stream);
>+            firstConnection.addTrack(stream.getAudioTracks()[0], stream);
>+        }, (connection) => {
>+            secondConnection = connection;
>+            secondConnection.ontrack = (trackEvent) => {
>+                if (!remoteVideoTrack) {
>+                    remoteVideoTrack = trackEvent.track;
>+                    return;
>+                }
>+                remoteAudioTrack = trackEvent.track;
>+                resolve();
>+            };
>+        });
>+        setTimeout(() => reject("Test timed out"), 5000);
>+    });
>+}, "Setup audio video exchange");
>+
>+async function renegotiate()
>+{
>+    let d = await firstConnection.createOffer();
>+    await firstConnection.setLocalDescription(d);
>+    await secondConnection.setRemoteDescription(firstConnection.localDescription);
>+    d = await secondConnection.createAnswer();
>+    await secondConnection.setLocalDescription(d);
>+}
>+
>+promise_test((test) => {
>+    const promise = new Promise((resolve, reject) => {
>+        remoteVideoTrack.onmute = resolve;
>+        setTimeout(() => reject("Test timed out"), 5000);
>+    });
>+     
>+    firstConnection.removeTrack(firstConnection.getSenders()[0]);
>+    renegotiate();
>+    return promise;
>+}, "Remove video track");
>+
>+promise_test((test) => {
>+    const promise = new Promise((resolve, reject) => {
>+        remoteAudioTrack.onmute = resolve;
>+        setTimeout(() => reject("Test timed out"), 5000);
>+    });
>+     
>+    firstConnection.removeTrack(firstConnection.getSenders()[1]);
>+    renegotiate();
>+    return promise;
>+}, "Remove audio track");
>+        </script>
>+    </body>
>+</html>

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Attachments on bug 176281: 319733 | 349915