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<dl class="meta">
<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/211837">211837</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2017-02-07 13:56:42 -0800 (Tue, 07 Feb 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>[WebRTC] LibWebRTCEndpoint should not own objects that should be destroyed on the main thread
https://bugs.webkit.org/show_bug.cgi?id=167816
Patch by Youenn Fablet <youennf@gmail.com> on 2017-02-07
Reviewed by Alex Christensen.
Source/WebCore:
Tests: webrtc/libwebrtc/release-while-creating-offer.html
webrtc/libwebrtc/release-while-getting-stats.html
webrtc/libwebrtc/release-while-setting-local-description.html
Moving AV sources, stats promises, ICE candidates from LibWebRTCEndpoint to LibWebRTCPeerConnectionBackend.
This allows ensuring these are destroyed in the main thread.
* Modules/mediastream/MediaEndpointPeerConnection.cpp:
(WebCore::MediaEndpointPeerConnection::getStats):
* Modules/mediastream/MediaEndpointPeerConnection.h:
* Modules/mediastream/PeerConnectionBackend.h:
* Modules/mediastream/RTCPeerConnection.cpp:
(WebCore::RTCPeerConnection::getStats):
* Modules/mediastream/RTCPeerConnection.h:
* Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
(WebCore::LibWebRTCMediaEndpoint::doCreateOffer):
(WebCore::LibWebRTCMediaEndpoint::doCreateAnswer):
(WebCore::LibWebRTCMediaEndpoint::getStats):
(WebCore::LibWebRTCMediaEndpoint::StatsCollector::StatsCollector):
(WebCore::LibWebRTCMediaEndpoint::StatsCollector::OnStatsDelivered):
* Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
(WebCore::LibWebRTCMediaEndpoint::addIceCandidate):
(WebCore::LibWebRTCMediaEndpoint::isStopped):
* Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
(WebCore::LibWebRTCPeerConnectionBackend::~LibWebRTCPeerConnectionBackend):
(WebCore::LibWebRTCPeerConnectionBackend::getStats):
(WebCore::LibWebRTCPeerConnectionBackend::iceCandidateSucceeded):
(WebCore::LibWebRTCPeerConnectionBackend::iceCandidateFailed):
(WebCore::LibWebRTCPeerConnectionBackend::doSetLocalDescription):
(WebCore::LibWebRTCPeerConnectionBackend::doSetRemoteDescription):
(WebCore::LibWebRTCPeerConnectionBackend::doAddIceCandidate):
(WebCore::LibWebRTCPeerConnectionBackend::addAudioSource):
(WebCore::LibWebRTCPeerConnectionBackend::addVideoSource):
* Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
* testing/MockLibWebRTCPeerConnection.cpp:
(WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer):
(WebCore::releaseInNetworkThread):
(WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer::CreateOffer):
(WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats):
(WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats::GetStats):
(WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription):
(WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription::SetLocalDescription):
(WebCore::MockLibWebRTCPeerConnectionFactory::CreatePeerConnection):
* testing/MockLibWebRTCPeerConnection.h:
LayoutTests:
* webrtc/libwebrtc/release-while-creating-offer.html: Added.
* webrtc/libwebrtc/release-while-getting-stats.html: Added.
* webrtc/libwebrtc/release-while-setting-local-description.html: Added.</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkLayoutTestsChangeLog">trunk/LayoutTests/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamMediaEndpointPeerConnectioncpp">trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamMediaEndpointPeerConnectionh">trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamPeerConnectionBackendh">trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamRTCPeerConnectioncpp">trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamRTCPeerConnectionh">trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCMediaEndpointcpp">trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCMediaEndpointh">trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCPeerConnectionBackendcpp">trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCPeerConnectionBackendh">trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h</a></li>
<li><a href="#trunkSourceWebCoretestingMockLibWebRTCPeerConnectioncpp">trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp</a></li>
<li><a href="#trunkSourceWebCoretestingMockLibWebRTCPeerConnectionh">trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h</a></li>
</ul>
<h3>Added Paths</h3>
<ul>
<li>trunk/LayoutTests/webrtc/libwebrtc/</li>
<li><a href="#trunkLayoutTestswebrtclibwebrtcreleasewhilecreatingofferhtml">trunk/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html</a></li>
<li><a href="#trunkLayoutTestswebrtclibwebrtcreleasewhilegettingstatshtml">trunk/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html</a></li>
<li><a href="#trunkLayoutTestswebrtclibwebrtcreleasewhilesettinglocaldescriptionhtml">trunk/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkLayoutTestsChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/ChangeLog (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/ChangeLog 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/LayoutTests/ChangeLog 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -1,5 +1,16 @@
</span><span class="cx"> 2017-02-07 Youenn Fablet <youennf@gmail.com>
</span><span class="cx">
</span><ins>+ [WebRTC] LibWebRTCEndpoint should not own objects that should be destroyed on the main thread
+ https://bugs.webkit.org/show_bug.cgi?id=167816
+
+ Reviewed by Alex Christensen.
+
+ * webrtc/libwebrtc/release-while-creating-offer.html: Added.
+ * webrtc/libwebrtc/release-while-getting-stats.html: Added.
+ * webrtc/libwebrtc/release-while-setting-local-description.html: Added.
+
+2017-02-07 Youenn Fablet <youennf@gmail.com>
+
</ins><span class="cx"> [WebRTC] LibWebRTC WK2 network stack is not providing correct ports for ICE candidates
</span><span class="cx"> https://bugs.webkit.org/show_bug.cgi?id=167939
</span><span class="cx">
</span></span></pre></div>
<a id="trunkLayoutTestswebrtclibwebrtcreleasewhilecreatingofferhtml"></a>
<div class="addfile"><h4>Added: trunk/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html (0 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html (rev 0)
+++ trunk/LayoutTests/webrtc/libwebrtc/release-while-creating-offer.html 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -0,0 +1,28 @@
</span><ins>+<!DOCTYPE html>
+<html>
+<head>
+</head>
+<body>
+<script src="../../resources/js-test-pre.js"></script>
+<script>
+self.jsTestIsAsync = true;
+
+if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileCreatingOffer");
+
+(function() {
+ var pc = new RTCPeerConnection();
+ pc.addIceCandidate({ candidate : "2013266431 1 udp 2013266432 192.168.0.100 38838 typ host generation 0" });
+ pc.createOffer();
+ pc.close();
+})();
+
+if (window.GCController)
+ GCController.collect();
+
+setTimeout(finishJSTest, 100);
+</script>
+<div style="font-family: WebFont;">This test makes sure that RTCPeerConnection will free itself correctly even if released from the network thread.</div>
+<script src="../../resources/js-test-post.js"></script>
+</body>
+</html>
</ins></span></pre></div>
<a id="trunkLayoutTestswebrtclibwebrtcreleasewhilegettingstatshtml"></a>
<div class="addfile"><h4>Added: trunk/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html (0 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html (rev 0)
+++ trunk/LayoutTests/webrtc/libwebrtc/release-while-getting-stats.html 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -0,0 +1,27 @@
</span><ins>+<!DOCTYPE html>
+<html>
+<head>
+</head>
+<body>
+<script src="../../resources/js-test-pre.js"></script>
+<script>
+self.jsTestIsAsync = true;
+
+if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileGettingStats");
+
+(function() {
+ var pc = new RTCPeerConnection();
+ pc.getStats();
+ pc.close();
+})();
+
+if (window.GCController)
+ GCController.collect();
+
+setTimeout(finishJSTest, 100);
+</script>
+<div style="font-family: WebFont;">This test makes sure that RTCPeerConnection will free itself correctly even if released from the network thread.</div>
+<script src="../../resources/js-test-post.js"></script>
+</body>
+</html>
</ins></span></pre></div>
<a id="trunkLayoutTestswebrtclibwebrtcreleasewhilesettinglocaldescriptionhtml"></a>
<div class="addfile"><h4>Added: trunk/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html (0 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html (rev 0)
+++ trunk/LayoutTests/webrtc/libwebrtc/release-while-setting-local-description.html 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -0,0 +1,31 @@
</span><ins>+<!DOCTYPE html>
+<html>
+<head>
+</head>
+<body>
+<script src="../../resources/js-test-pre.js"></script>
+<script>
+self.jsTestIsAsync = true;
+
+if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("LibWebRTCReleasingWhileSettingDescription");
+
+(function() {
+ var pc = new RTCPeerConnection();
+ pc.addIceCandidate({ candidate : "2013266431 1 udp 2013266432 192.168.0.100 38838 typ host generation 0" });
+ pc.createOffer().then((offer) => {
+ setTimeout(function() {
+ if (window.GCController)
+ GCController.collect();
+ finishJSTest();
+ }, 0);
+ pc.setLocalDescription(offer);
+ pc.close();
+ });
+})();
+
+</script>
+<div style="font-family: WebFont;">This test makes sure that RTCPeerConnection backend will free itself correctly even if released from the network thread.</div>
+<script src="../../resources/js-test-post.js"></script>
+</body>
+</html>
</ins></span></pre></div>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/ChangeLog 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -1,3 +1,55 @@
</span><ins>+2017-02-07 Youenn Fablet <youennf@gmail.com>
+
+ [WebRTC] LibWebRTCEndpoint should not own objects that should be destroyed on the main thread
+ https://bugs.webkit.org/show_bug.cgi?id=167816
+
+ Reviewed by Alex Christensen.
+
+ Tests: webrtc/libwebrtc/release-while-creating-offer.html
+ webrtc/libwebrtc/release-while-getting-stats.html
+ webrtc/libwebrtc/release-while-setting-local-description.html
+
+ Moving AV sources, stats promises, ICE candidates from LibWebRTCEndpoint to LibWebRTCPeerConnectionBackend.
+ This allows ensuring these are destroyed in the main thread.
+
+ * Modules/mediastream/MediaEndpointPeerConnection.cpp:
+ (WebCore::MediaEndpointPeerConnection::getStats):
+ * Modules/mediastream/MediaEndpointPeerConnection.h:
+ * Modules/mediastream/PeerConnectionBackend.h:
+ * Modules/mediastream/RTCPeerConnection.cpp:
+ (WebCore::RTCPeerConnection::getStats):
+ * Modules/mediastream/RTCPeerConnection.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::LibWebRTCMediaEndpoint::doCreateOffer):
+ (WebCore::LibWebRTCMediaEndpoint::doCreateAnswer):
+ (WebCore::LibWebRTCMediaEndpoint::getStats):
+ (WebCore::LibWebRTCMediaEndpoint::StatsCollector::StatsCollector):
+ (WebCore::LibWebRTCMediaEndpoint::StatsCollector::OnStatsDelivered):
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
+ (WebCore::LibWebRTCMediaEndpoint::addIceCandidate):
+ (WebCore::LibWebRTCMediaEndpoint::isStopped):
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
+ (WebCore::LibWebRTCPeerConnectionBackend::~LibWebRTCPeerConnectionBackend):
+ (WebCore::LibWebRTCPeerConnectionBackend::getStats):
+ (WebCore::LibWebRTCPeerConnectionBackend::iceCandidateSucceeded):
+ (WebCore::LibWebRTCPeerConnectionBackend::iceCandidateFailed):
+ (WebCore::LibWebRTCPeerConnectionBackend::doSetLocalDescription):
+ (WebCore::LibWebRTCPeerConnectionBackend::doSetRemoteDescription):
+ (WebCore::LibWebRTCPeerConnectionBackend::doAddIceCandidate):
+ (WebCore::LibWebRTCPeerConnectionBackend::addAudioSource):
+ (WebCore::LibWebRTCPeerConnectionBackend::addVideoSource):
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
+ * testing/MockLibWebRTCPeerConnection.cpp:
+ (WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer):
+ (WebCore::releaseInNetworkThread):
+ (WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer::CreateOffer):
+ (WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats):
+ (WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats::GetStats):
+ (WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription):
+ (WebCore::MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription::SetLocalDescription):
+ (WebCore::MockLibWebRTCPeerConnectionFactory::CreatePeerConnection):
+ * testing/MockLibWebRTCPeerConnection.h:
+
</ins><span class="cx"> 2017-02-07 Myles C. Maxfield <mmaxfield@apple.com>
</span><span class="cx">
</span><span class="cx"> [Win] [GTK] [EFL] Compile (but don't use, yet) the platform-independent piece of ComplexTextController
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamMediaEndpointPeerConnectioncpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -666,11 +666,11 @@
</span><span class="cx"> addIceCandidateSucceeded();
</span><span class="cx"> }
</span><span class="cx">
</span><del>-void MediaEndpointPeerConnection::getStats(MediaStreamTrack*, PeerConnection::StatsPromise&& promise)
</del><ins>+void MediaEndpointPeerConnection::getStats(MediaStreamTrack*, Ref<DeferredPromise>&& promise)
</ins><span class="cx"> {
</span><span class="cx"> notImplemented();
</span><span class="cx">
</span><del>- promise.reject(NOT_SUPPORTED_ERR);
</del><ins>+ promise->reject(NOT_SUPPORTED_ERR);
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> Vector<RefPtr<MediaStream>> MediaEndpointPeerConnection::getRemoteStreams() const
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamMediaEndpointPeerConnectionh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.h (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.h 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.h 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -57,7 +57,7 @@
</span><span class="cx">
</span><span class="cx"> void setConfiguration(MediaEndpointConfiguration&&) final;
</span><span class="cx">
</span><del>- void getStats(MediaStreamTrack*, PeerConnection::StatsPromise&&) final;
</del><ins>+ void getStats(MediaStreamTrack*, Ref<DeferredPromise>&&) final;
</ins><span class="cx">
</span><span class="cx"> Vector<RefPtr<MediaStream>> getRemoteStreams() const final;
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamPeerConnectionBackendh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -88,7 +88,7 @@
</span><span class="cx">
</span><span class="cx"> virtual void setConfiguration(MediaEndpointConfiguration&&) = 0;
</span><span class="cx">
</span><del>- virtual void getStats(MediaStreamTrack*, PeerConnection::StatsPromise&&) = 0;
</del><ins>+ virtual void getStats(MediaStreamTrack*, Ref<DeferredPromise>&&) = 0;
</ins><span class="cx">
</span><span class="cx"> virtual Vector<RefPtr<MediaStream>> getRemoteStreams() const = 0;
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamRTCPeerConnectioncpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -382,7 +382,7 @@
</span><span class="cx"> return { };
</span><span class="cx"> }
</span><span class="cx">
</span><del>-void RTCPeerConnection::getStats(MediaStreamTrack* selector, PeerConnection::StatsPromise&& promise)
</del><ins>+void RTCPeerConnection::getStats(MediaStreamTrack* selector, Ref<DeferredPromise>&& promise)
</ins><span class="cx"> {
</span><span class="cx"> m_backend->getStats(selector, WTFMove(promise));
</span><span class="cx"> }
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamRTCPeerConnectionh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -106,7 +106,7 @@
</span><span class="cx"> const RTCConfiguration& getConfiguration() const { return m_configuration; }
</span><span class="cx"> ExceptionOr<void> setConfiguration(RTCConfiguration&&);
</span><span class="cx">
</span><del>- void getStats(MediaStreamTrack*, PeerConnection::StatsPromise&&);
</del><ins>+ void getStats(MediaStreamTrack*, Ref<DeferredPromise>&&);
</ins><span class="cx">
</span><span class="cx"> ExceptionOr<Ref<RTCDataChannel>> createDataChannel(ScriptExecutionContext&, String&&, RTCDataChannelInit&&);
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCMediaEndpointcpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -102,12 +102,6 @@
</span><span class="cx"> m_backend->SetRemoteDescription(&m_setRemoteSessionDescriptionObserver, sessionDescription.release());
</span><span class="cx"> }
</span><span class="cx">
</span><del>-void LibWebRTCMediaEndpoint::addPendingIceCandidates()
-{
- while (m_pendingCandidates.size())
- m_backend->AddIceCandidate(m_pendingCandidates.takeLast().release());
-}
-
</del><span class="cx"> static inline std::string streamId(RTCPeerConnection& connection)
</span><span class="cx"> {
</span><span class="cx"> auto& senders = connection.getSenders();
</span><span class="lines">@@ -139,11 +133,13 @@
</span><span class="cx"> auto trackSource = RealtimeOutgoingAudioSource::create(source);
</span><span class="cx"> auto rtcTrack = peerConnectionFactory().CreateAudioTrack(track->id().utf8().data(), trackSource.ptr());
</span><span class="cx"> trackSource->setTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>(rtcTrack));
</span><del>- m_audioSources.append(WTFMove(trackSource));
</del><ins>+ m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
</ins><span class="cx"> stream->AddTrack(WTFMove(rtcTrack));
</span><span class="cx"> } else {
</span><del>- m_videoSources.append(RealtimeOutgoingVideoSource::create(source));
- stream->AddTrack(peerConnectionFactory().CreateVideoTrack(track->id().utf8().data(), m_videoSources.last().ptr()));
</del><ins>+ auto videoSource = RealtimeOutgoingVideoSource::create(source);
+ auto videoTrack = peerConnectionFactory().CreateVideoTrack(track->id().utf8().data(), videoSource.ptr());
+ m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
+ stream->AddTrack(WTFMove(videoTrack));
</ins><span class="cx"> }
</span><span class="cx"> }
</span><span class="cx"> }
</span><span class="lines">@@ -169,11 +165,13 @@
</span><span class="cx"> auto trackSource = RealtimeOutgoingAudioSource::create(source);
</span><span class="cx"> auto rtcTrack = peerConnectionFactory().CreateAudioTrack(track->id().utf8().data(), trackSource.ptr());
</span><span class="cx"> trackSource->setTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>(rtcTrack));
</span><del>- m_audioSources.append(WTFMove(trackSource));
</del><ins>+ m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
</ins><span class="cx"> stream->AddTrack(WTFMove(rtcTrack));
</span><span class="cx"> } else {
</span><del>- m_videoSources.append(RealtimeOutgoingVideoSource::create(source));
- stream->AddTrack(peerConnectionFactory().CreateVideoTrack(track->id().utf8().data(), m_videoSources.last().ptr()));
</del><ins>+ auto videoSource = RealtimeOutgoingVideoSource::create(source);
+ auto videoTrack = peerConnectionFactory().CreateVideoTrack(track->id().utf8().data(), videoSource.ptr());
+ m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
+ stream->AddTrack(WTFMove(videoTrack));
</ins><span class="cx"> }
</span><span class="cx"> }
</span><span class="cx"> }
</span><span class="lines">@@ -182,16 +180,14 @@
</span><span class="cx"> m_backend->CreateAnswer(&m_createSessionDescriptionObserver, nullptr);
</span><span class="cx"> }
</span><span class="cx">
</span><del>-void LibWebRTCMediaEndpoint::getStats(MediaStreamTrack* track, PeerConnection::StatsPromise&& promise)
</del><ins>+void LibWebRTCMediaEndpoint::getStats(MediaStreamTrack* track, const DeferredPromise& promise)
</ins><span class="cx"> {
</span><del>- auto collector = StatsCollector::create(*this, WTFMove(promise), track);
- m_backend->GetStats(collector.ptr());
- m_statsCollectors.append(WTFMove(collector));
</del><ins>+ m_backend->GetStats(StatsCollector::create(*this, promise, track).get());
</ins><span class="cx"> }
</span><span class="cx">
</span><del>-LibWebRTCMediaEndpoint::StatsCollector::StatsCollector(LibWebRTCMediaEndpoint& endpoint, PeerConnection::StatsPromise&& promise, MediaStreamTrack* track)
</del><ins>+LibWebRTCMediaEndpoint::StatsCollector::StatsCollector(LibWebRTCMediaEndpoint& endpoint, const DeferredPromise& promise, MediaStreamTrack* track)
</ins><span class="cx"> : m_endpoint(endpoint)
</span><del>- , m_promise(WTFMove(promise))
</del><ins>+ , m_promise(promise)
</ins><span class="cx"> {
</span><span class="cx"> if (track)
</span><span class="cx"> m_id = track->id();
</span><span class="lines">@@ -199,14 +195,14 @@
</span><span class="cx">
</span><span class="cx"> void LibWebRTCMediaEndpoint::StatsCollector::OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>& report)
</span><span class="cx"> {
</span><del>- callOnMainThread([this, report, protector = makeRef(m_endpoint)] {
</del><ins>+ callOnMainThread([protectedThis = rtc::scoped_refptr<LibWebRTCMediaEndpoint::StatsCollector>(this), report] {
+ if (protectedThis->m_endpoint.isStopped())
+ return;
+
</ins><span class="cx"> // FIXME: Fulfill promise with the report
</span><span class="cx"> UNUSED_PARAM(report);
</span><del>- m_promise.reject(TypeError, ASCIILiteral("Stats API is not yet implemented"));
</del><span class="cx">
</span><del>- m_endpoint.m_statsCollectors.removeFirstMatching([this](const Ref<StatsCollector>& collector) {
- return this == collector.ptr();
- });
</del><ins>+ protectedThis->m_endpoint.m_peerConnectionBackend.iceCandidateFailed(protectedThis->m_promise, Exception { TypeError, ASCIILiteral("Stats API is not yet implemented") });
</ins><span class="cx"> });
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCMediaEndpointh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -65,13 +65,12 @@
</span><span class="cx"> void doSetRemoteDescription(RTCSessionDescription&);
</span><span class="cx"> void doCreateOffer();
</span><span class="cx"> void doCreateAnswer();
</span><del>- void getStats(MediaStreamTrack*, PeerConnection::StatsPromise&&);
</del><ins>+ void getStats(MediaStreamTrack*, const DeferredPromise&);
</ins><span class="cx"> std::unique_ptr<RTCDataChannelHandler> createDataChannel(const String&, const RTCDataChannelInit&);
</span><ins>+ bool addIceCandidate(webrtc::IceCandidateInterface& candidate) { return m_backend->AddIceCandidate(&candidate); }
</ins><span class="cx">
</span><del>- void storeIceCandidate(std::unique_ptr<webrtc::IceCandidateInterface>&& candidate) { m_pendingCandidates.append(WTFMove(candidate)); }
- void addPendingIceCandidates();
-
</del><span class="cx"> void stop();
</span><ins>+ bool isStopped() const { return !m_backend; }
</ins><span class="cx">
</span><span class="cx"> private:
</span><span class="cx"> LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend&, LibWebRTCProvider&);
</span><span class="lines">@@ -96,8 +95,6 @@
</span><span class="cx"> void addStream(webrtc::MediaStreamInterface&);
</span><span class="cx"> void addDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface>&&);
</span><span class="cx">
</span><del>- bool isStopped() const { return !m_backend; }
-
</del><span class="cx"> int AddRef() const { ref(); return static_cast<int>(refCount()); }
</span><span class="cx"> int Release() const { deref(); return static_cast<int>(refCount()); }
</span><span class="cx">
</span><span class="lines">@@ -143,19 +140,20 @@
</span><span class="cx"> LibWebRTCMediaEndpoint& m_endpoint;
</span><span class="cx"> };
</span><span class="cx">
</span><del>- class StatsCollector final : public RefCounted<StatsCollector>, public webrtc::RTCStatsCollectorCallback {
</del><ins>+ class StatsCollector final : public webrtc::RTCStatsCollectorCallback {
</ins><span class="cx"> public:
</span><del>- static Ref<StatsCollector> create(LibWebRTCMediaEndpoint& endpoint, PeerConnection::StatsPromise&& promise, MediaStreamTrack* track) { return adoptRef(* new StatsCollector(endpoint, WTFMove(promise), track)); }
</del><ins>+ static rtc::scoped_refptr<StatsCollector> create(LibWebRTCMediaEndpoint& endpoint, const DeferredPromise& promise, MediaStreamTrack* track) { return new StatsCollector(endpoint, promise, track); }
+
+ int AddRef() const { return m_endpoint.AddRef(); }
+ int Release() const { return m_endpoint.Release(); }
+
</ins><span class="cx"> private:
</span><del>- StatsCollector(LibWebRTCMediaEndpoint&, PeerConnection::StatsPromise&&, MediaStreamTrack*);
</del><ins>+ StatsCollector(LibWebRTCMediaEndpoint&, const DeferredPromise&, MediaStreamTrack*);
</ins><span class="cx">
</span><span class="cx"> void OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>&) final;
</span><span class="cx">
</span><del>- int AddRef() const final { ref(); return static_cast<int>(refCount()); }
- int Release() const final { deref(); return static_cast<int>(refCount()); }
-
</del><span class="cx"> LibWebRTCMediaEndpoint& m_endpoint;
</span><del>- PeerConnection::StatsPromise m_promise;
</del><ins>+ const DeferredPromise& m_promise;
</ins><span class="cx"> String m_id;
</span><span class="cx"> };
</span><span class="cx">
</span><span class="lines">@@ -167,11 +165,6 @@
</span><span class="cx"> SetRemoteSessionDescriptionObserver m_setRemoteSessionDescriptionObserver;
</span><span class="cx">
</span><span class="cx"> bool m_isInitiator { false };
</span><del>-
- Vector<std::unique_ptr<webrtc::IceCandidateInterface>> m_pendingCandidates;
- Vector<Ref<RealtimeOutgoingAudioSource>> m_audioSources;
- Vector<Ref<RealtimeOutgoingVideoSource>> m_videoSources;
- Vector<Ref<StatsCollector>> m_statsCollectors;
</del><span class="cx"> };
</span><span class="cx">
</span><span class="cx"> } // namespace WebCore
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCPeerConnectionBackendcpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -29,6 +29,7 @@
</span><span class="cx">
</span><span class="cx"> #include "Document.h"
</span><span class="cx"> #include "IceCandidate.h"
</span><ins>+#include "JSRTCStatsResponse.h"
</ins><span class="cx"> #include "LibWebRTCDataChannelHandler.h"
</span><span class="cx"> #include "LibWebRTCMediaEndpoint.h"
</span><span class="cx"> #include "MediaEndpointConfiguration.h"
</span><span class="lines">@@ -62,6 +63,10 @@
</span><span class="cx"> {
</span><span class="cx"> }
</span><span class="cx">
</span><ins>+LibWebRTCPeerConnectionBackend::~LibWebRTCPeerConnectionBackend()
+{
+}
+
</ins><span class="cx"> static webrtc::PeerConnectionInterface::RTCConfiguration configurationFromMediaEndpointConfiguration(MediaEndpointConfiguration&& configuration)
</span><span class="cx"> {
</span><span class="cx"> webrtc::PeerConnectionInterface::RTCConfiguration rtcConfiguration(webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive);
</span><span class="lines">@@ -91,17 +96,38 @@
</span><span class="cx"> m_endpoint->backend().SetConfiguration(configurationFromMediaEndpointConfiguration(WTFMove(configuration)));
</span><span class="cx"> }
</span><span class="cx">
</span><del>-void LibWebRTCPeerConnectionBackend::getStats(MediaStreamTrack* track, PeerConnection::StatsPromise&& promise)
</del><ins>+void LibWebRTCPeerConnectionBackend::getStats(MediaStreamTrack* track, Ref<DeferredPromise>&& promise)
</ins><span class="cx"> {
</span><del>- m_endpoint->getStats(track, WTFMove(promise));
</del><ins>+ if (m_endpoint->isStopped())
+ return;
+
+ auto& statsPromise = promise.get();
+ m_statsPromises.add(&statsPromise, WTFMove(promise));
+ m_endpoint->getStats(track, statsPromise);
</ins><span class="cx"> }
</span><span class="cx">
</span><ins>+void LibWebRTCPeerConnectionBackend::iceCandidateSucceeded(const DeferredPromise& promise, Ref<RTCStatsResponse>&& response)
+{
+ auto statsPromise = m_statsPromises.take(&promise);
+ ASSERT(statsPromise);
+ statsPromise.value()->resolve<IDLInterface<RTCStatsResponse>>(WTFMove(response));
+}
+
+void LibWebRTCPeerConnectionBackend::iceCandidateFailed(const DeferredPromise& promise, Exception&& exception)
+{
+ auto statsPromise = m_statsPromises.take(&promise);
+ ASSERT(statsPromise);
+ statsPromise.value()->reject(WTFMove(exception));
+}
+
</ins><span class="cx"> void LibWebRTCPeerConnectionBackend::doSetLocalDescription(RTCSessionDescription& description)
</span><span class="cx"> {
</span><span class="cx"> m_endpoint->doSetLocalDescription(description);
</span><span class="cx"> if (!m_isLocalDescriptionSet) {
</span><del>- if (m_isRemoteDescriptionSet)
- m_endpoint->addPendingIceCandidates();
</del><ins>+ if (m_isRemoteDescriptionSet) {
+ while (m_pendingCandidates.size())
+ m_endpoint->addIceCandidate(*m_pendingCandidates.takeLast().release());
+ }
</ins><span class="cx"> m_isLocalDescriptionSet = true;
</span><span class="cx"> }
</span><span class="cx"> }
</span><span class="lines">@@ -110,8 +136,10 @@
</span><span class="cx"> {
</span><span class="cx"> m_endpoint->doSetRemoteDescription(description);
</span><span class="cx"> if (!m_isRemoteDescriptionSet) {
</span><del>- if (m_isLocalDescriptionSet)
- m_endpoint->addPendingIceCandidates();
</del><ins>+ if (m_isLocalDescriptionSet) {
+ while (m_pendingCandidates.size())
+ m_endpoint->addIceCandidate(*m_pendingCandidates.takeLast().release());
+ }
</ins><span class="cx"> m_isRemoteDescriptionSet = true;
</span><span class="cx"> }
</span><span class="cx"> }
</span><span class="lines">@@ -154,8 +182,8 @@
</span><span class="cx">
</span><span class="cx"> // libwebrtc does not like that ice candidates are set before the description.
</span><span class="cx"> if (!m_isLocalDescriptionSet || !m_isRemoteDescriptionSet)
</span><del>- m_endpoint->storeIceCandidate(WTFMove(rtcCandidate));
- else if (!m_endpoint->backend().AddIceCandidate(rtcCandidate.get())) {
</del><ins>+ m_pendingCandidates.append(WTFMove(rtcCandidate));
+ else if (!m_endpoint->addIceCandidate(*rtcCandidate.get())) {
</ins><span class="cx"> ASSERT_NOT_REACHED();
</span><span class="cx"> addIceCandidateFailed(Exception { OperationError, ASCIILiteral("Failed to apply the received candidate") });
</span><span class="cx"> return;
</span><span class="lines">@@ -163,6 +191,16 @@
</span><span class="cx"> addIceCandidateSucceeded();
</span><span class="cx"> }
</span><span class="cx">
</span><ins>+void LibWebRTCPeerConnectionBackend::addAudioSource(Ref<RealtimeOutgoingAudioSource>&& source)
+{
+ m_audioSources.append(WTFMove(source));
+}
+
+void LibWebRTCPeerConnectionBackend::addVideoSource(Ref<RealtimeOutgoingVideoSource>&& source)
+{
+ m_videoSources.append(WTFMove(source));
+}
+
</ins><span class="cx"> void LibWebRTCPeerConnectionBackend::markAsNeedingNegotiation()
</span><span class="cx"> {
</span><span class="cx"> // FIXME: Implement this
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCPeerConnectionBackendh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -27,16 +27,24 @@
</span><span class="cx"> #if USE(LIBWEBRTC)
</span><span class="cx">
</span><span class="cx"> #include "PeerConnectionBackend.h"
</span><ins>+#include <wtf/HashMap.h>
</ins><span class="cx">
</span><ins>+namespace webrtc {
+class IceCandidateInterface;
+}
+
</ins><span class="cx"> namespace WebCore {
</span><span class="cx">
</span><span class="cx"> class LibWebRTCMediaEndpoint;
</span><span class="cx"> class RTCRtpReceiver;
</span><span class="cx"> class RTCSessionDescription;
</span><ins>+class RealtimeOutgoingAudioSource;
+class RealtimeOutgoingVideoSource;
</ins><span class="cx">
</span><span class="cx"> class LibWebRTCPeerConnectionBackend final : public PeerConnectionBackend {
</span><span class="cx"> public:
</span><span class="cx"> explicit LibWebRTCPeerConnectionBackend(RTCPeerConnection&);
</span><ins>+ ~LibWebRTCPeerConnectionBackend();
</ins><span class="cx">
</span><span class="cx"> private:
</span><span class="cx"> void doCreateOffer(RTCOfferOptions&&) final;
</span><span class="lines">@@ -47,7 +55,7 @@
</span><span class="cx"> void doStop() final;
</span><span class="cx"> std::unique_ptr<RTCDataChannelHandler> createDataChannelHandler(const String&, const RTCDataChannelInit&) final;
</span><span class="cx"> void setConfiguration(MediaEndpointConfiguration&&) final;
</span><del>- void getStats(MediaStreamTrack*, PeerConnection::StatsPromise&&) final;
</del><ins>+ void getStats(MediaStreamTrack*, Ref<DeferredPromise>&&) final;
</ins><span class="cx"> Ref<RTCRtpReceiver> createReceiver(const String& transceiverMid, const String& trackKind, const String& trackId) final;
</span><span class="cx">
</span><span class="cx"> // FIXME: API to implement for real
</span><span class="lines">@@ -59,7 +67,6 @@
</span><span class="cx"> RefPtr<RTCSessionDescription> currentRemoteDescription() const final { return nullptr; }
</span><span class="cx"> RefPtr<RTCSessionDescription> pendingRemoteDescription() const final { return nullptr; }
</span><span class="cx">
</span><del>-
</del><span class="cx"> Vector<RefPtr<MediaStream>> getRemoteStreams() const final { return { }; }
</span><span class="cx">
</span><span class="cx"> void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) final { }
</span><span class="lines">@@ -72,11 +79,21 @@
</span><span class="cx">
</span><span class="cx"> friend LibWebRTCMediaEndpoint;
</span><span class="cx"> RTCPeerConnection& connection() { return m_peerConnection; }
</span><ins>+ void addAudioSource(Ref<RealtimeOutgoingAudioSource>&&);
+ void addVideoSource(Ref<RealtimeOutgoingVideoSource>&&);
</ins><span class="cx">
</span><ins>+ void iceCandidateSucceeded(const DeferredPromise&, Ref<RTCStatsResponse>&&);
+ void iceCandidateFailed(const DeferredPromise&, Exception&&);
+
</ins><span class="cx"> private:
</span><span class="cx"> Ref<LibWebRTCMediaEndpoint> m_endpoint;
</span><span class="cx"> bool m_isLocalDescriptionSet { false };
</span><span class="cx"> bool m_isRemoteDescriptionSet { false };
</span><ins>+
+ Vector<std::unique_ptr<webrtc::IceCandidateInterface>> m_pendingCandidates;
+ Vector<Ref<RealtimeOutgoingAudioSource>> m_audioSources;
+ Vector<Ref<RealtimeOutgoingVideoSource>> m_videoSources;
+ HashMap<const DeferredPromise*, Ref<DeferredPromise>> m_statsPromises;
</ins><span class="cx"> };
</span><span class="cx">
</span><span class="cx"> } // namespace WebCore
</span></span></pre></div>
<a id="trunkSourceWebCoretestingMockLibWebRTCPeerConnectioncpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -32,6 +32,7 @@
</span><span class="cx"> #include <sstream>
</span><span class="cx"> #include <webrtc/api/mediastream.h>
</span><span class="cx"> #include <wtf/Function.h>
</span><ins>+#include <wtf/MainThread.h>
</ins><span class="cx">
</span><span class="cx"> namespace WebCore {
</span><span class="cx">
</span><span class="lines">@@ -68,7 +69,6 @@
</span><span class="cx"> });
</span><span class="cx"> }
</span><span class="cx">
</span><del>-
</del><span class="cx"> class MockLibWebRTCPeerConnectionForIceConnectionState : public MockLibWebRTCPeerConnection {
</span><span class="cx"> public:
</span><span class="cx"> explicit MockLibWebRTCPeerConnectionForIceConnectionState(webrtc::PeerConnectionObserver& observer) : MockLibWebRTCPeerConnection(observer) { }
</span><span class="lines">@@ -88,6 +88,52 @@
</span><span class="cx"> m_observer.OnIceConnectionChange(kIceConnectionNew);
</span><span class="cx"> }
</span><span class="cx">
</span><ins>+template<typename U> static inline void releaseInNetworkThread(MockLibWebRTCPeerConnection& mock, U& observer)
+{
+ mock.AddRef();
+ observer.AddRef();
+ callOnMainThread([&mock, &observer] {
+ callOnWebRTCNetworkThread([&mock, &observer]() {
+ observer.Release();
+ mock.Release();
+ });
+ });
+}
+
+class MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer : public MockLibWebRTCPeerConnection {
+public:
+ explicit MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer(webrtc::PeerConnectionObserver& observer) : MockLibWebRTCPeerConnection(observer) { }
+ virtual ~MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer() = default;
+
+private:
+ void CreateOffer(webrtc::CreateSessionDescriptionObserver* observer, const webrtc::MediaConstraintsInterface*) final { releaseInNetworkThread(*this, *observer); }
+};
+
+class MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats : public MockLibWebRTCPeerConnection {
+public:
+ explicit MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats(webrtc::PeerConnectionObserver& observer) : MockLibWebRTCPeerConnection(observer) { }
+ virtual ~MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats() = default;
+
+private:
+ bool GetStats(webrtc::StatsObserver*, webrtc::MediaStreamTrackInterface*, StatsOutputLevel) final;
+};
+
+bool MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats::GetStats(webrtc::StatsObserver* observer, webrtc::MediaStreamTrackInterface*, StatsOutputLevel)
+{
+ releaseInNetworkThread(*this, *observer);
+ return true;
+}
+
+class MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription : public MockLibWebRTCPeerConnection {
+public:
+ explicit MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription(webrtc::PeerConnectionObserver& observer) : MockLibWebRTCPeerConnection(observer) { }
+ virtual ~MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription() = default;
+
+private:
+ void SetLocalDescription(webrtc::SetSessionDescriptionObserver* observer, webrtc::SessionDescriptionInterface*) final { releaseInNetworkThread(*this, *observer); }
+};
+
+
</ins><span class="cx"> MockLibWebRTCPeerConnectionFactory::MockLibWebRTCPeerConnectionFactory(LibWebRTCProvider* provider, String&& testCase)
</span><span class="cx"> : m_provider(provider)
</span><span class="cx"> , m_testCase(WTFMove(testCase))
</span><span class="lines">@@ -116,6 +162,15 @@
</span><span class="cx"> if (m_testCase == "ICEConnectionState")
</span><span class="cx"> return new rtc::RefCountedObject<MockLibWebRTCPeerConnectionForIceConnectionState>(*observer);
</span><span class="cx">
</span><ins>+ if (m_testCase == "LibWebRTCReleasingWhileCreatingOffer")
+ return new rtc::RefCountedObject<MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileCreatingOffer>(*observer);
+
+ if (m_testCase == "LibWebRTCReleasingWhileGettingStats")
+ return new rtc::RefCountedObject<MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileGettingStats>(*observer);
+
+ if (m_testCase == "LibWebRTCReleasingWhileSettingDescription")
+ return new rtc::RefCountedObject<MockLibWebRTCPeerConnectionReleasedInNetworkThreadWhileSettingDescription>(*observer);
+
</ins><span class="cx"> return new rtc::RefCountedObject<MockLibWebRTCPeerConnection>(*observer);
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoretestingMockLibWebRTCPeerConnectionh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h (211836 => 211837)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h 2017-02-07 21:37:44 UTC (rev 211836)
+++ trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h 2017-02-07 21:56:42 UTC (rev 211837)
</span><span class="lines">@@ -45,29 +45,30 @@
</span><span class="cx"> explicit MockLibWebRTCPeerConnection(webrtc::PeerConnectionObserver& observer) : m_observer(observer) { }
</span><span class="cx">
</span><span class="cx"> private:
</span><del>- rtc::scoped_refptr<webrtc::StreamCollectionInterface> local_streams() { return nullptr; }
- rtc::scoped_refptr<webrtc::StreamCollectionInterface> remote_streams() { return nullptr; }
- rtc::scoped_refptr<webrtc::DtmfSenderInterface> CreateDtmfSender(webrtc::AudioTrackInterface*) { return nullptr; }
- bool GetStats(webrtc::StatsObserver*, webrtc::MediaStreamTrackInterface*, StatsOutputLevel) { return false; }
- const webrtc::SessionDescriptionInterface* local_description() const { return nullptr; }
- const webrtc::SessionDescriptionInterface* remote_description() const { return nullptr; }
- bool AddIceCandidate(const webrtc::IceCandidateInterface*) { return true; }
- void RegisterUMAObserver(webrtc::UMAObserver*) { }
- SignalingState signaling_state() { return kStable; }
- IceConnectionState ice_connection_state() { return kIceConnectionNew; }
- IceGatheringState ice_gathering_state() { return kIceGatheringNew; }
- void StopRtcEventLog() { }
- void Close() { }
</del><ins>+ rtc::scoped_refptr<webrtc::StreamCollectionInterface> local_streams() override { return nullptr; }
+ rtc::scoped_refptr<webrtc::StreamCollectionInterface> remote_streams() override { return nullptr; }
+ rtc::scoped_refptr<webrtc::DtmfSenderInterface> CreateDtmfSender(webrtc::AudioTrackInterface*) override { return nullptr; }
+ const webrtc::SessionDescriptionInterface* local_description() const override { return nullptr; }
+ const webrtc::SessionDescriptionInterface* remote_description() const override { return nullptr; }
+ bool AddIceCandidate(const webrtc::IceCandidateInterface*) override { return true; }
+ void RegisterUMAObserver(webrtc::UMAObserver*) override { }
+ SignalingState signaling_state() override { return kStable; }
+ IceConnectionState ice_connection_state() override { return kIceConnectionNew; }
+ IceGatheringState ice_gathering_state() override { return kIceGatheringNew; }
+ void StopRtcEventLog() override { }
+ void Close() override { }
</ins><span class="cx">
</span><span class="cx"> protected:
</span><del>- void SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) final;
</del><span class="cx"> void SetRemoteDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) final;
</span><del>- void CreateOffer(webrtc::CreateSessionDescriptionObserver*, const webrtc::MediaConstraintsInterface*) final;
</del><span class="cx"> void CreateAnswer(webrtc::CreateSessionDescriptionObserver*, const webrtc::MediaConstraintsInterface*) final;
</span><span class="cx"> rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(const std::string&, const webrtc::DataChannelInit*) final;
</span><span class="cx"> bool AddStream(webrtc::MediaStreamInterface*) final;
</span><span class="cx"> void RemoveStream(webrtc::MediaStreamInterface*) final;
</span><span class="cx">
</span><ins>+ void SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) override;
+ bool GetStats(webrtc::StatsObserver*, webrtc::MediaStreamTrackInterface*, StatsOutputLevel) override { return false; }
+ void CreateOffer(webrtc::CreateSessionDescriptionObserver*, const webrtc::MediaConstraintsInterface*) override;
+
</ins><span class="cx"> virtual void gotLocalDescription() { }
</span><span class="cx">
</span><span class="cx"> webrtc::PeerConnectionObserver& m_observer;
</span></span></pre>
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