Diff
Modified: trunk/LayoutTests/ChangeLog (218794 => 218795)
--- trunk/LayoutTests/ChangeLog 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/ChangeLog 2017-06-25 18:06:06 UTC (rev 218795)
@@ -1,3 +1,25 @@
+2017-06-25 Youenn Fablet <[email protected]>
+
+ Remove use of mock webrtc backend factory at injected bundle reset time
+ https://bugs.webkit.org/show_bug.cgi?id=173817
+
+ Reviewed by Darin Adler.
+
+ Making some tests less flaky.
+
+ * TestExpectations:
+ * fast/mediastream/RTCPeerConnection-media-setup-two-dialogs-expected.txt:
+ * platform/mac/webrtc/captureCanvas-webrtc-software-encoder.html:
+ * webrtc/audio-peer-connection-webaudio.html:
+ * webrtc/captureCanvas-webrtc.html:
+ * webrtc/datachannel/bufferedAmountLowThreshold-default-expected.txt: Added.
+ * webrtc/datachannel/bufferedAmountLowThreshold-default.html: Added.
+ * webrtc/datachannel/bufferedAmountLowThreshold-expected.txt:
+ * webrtc/datachannel/bufferedAmountLowThreshold.html:
+ * webrtc/routines.js:
+ * webrtc/video-replace-muted-track.html:
+ * webrtc/video-unmute.html:
+
2017-06-24 Joseph Pecoraro <[email protected]>
Remove Reflect.enumerate
Modified: trunk/LayoutTests/TestExpectations (218794 => 218795)
--- trunk/LayoutTests/TestExpectations 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/TestExpectations 2017-06-25 18:06:06 UTC (rev 218795)
@@ -770,6 +770,7 @@
# Media Sessions is not yet enabled by default: ENABLE(MEDIA_SESSION)
media/session [ Skip ]
+webrtc/datachannel/multiple-connections.html [ Slow ]
webrtc/negotiatedneeded-event-addStream.html [ Pass Crash ]
webrtc/video-replace-track.html [ Pass Failure ]
webrtc/video-getParameters.html [ Failure ]
Modified: trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs-expected.txt (218794 => 218795)
--- trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs-expected.txt 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/fast/mediastream/RTCPeerConnection-media-setup-two-dialogs-expected.txt 2017-06-25 18:06:06 UTC (rev 218795)
@@ -21,14 +21,85 @@
A: create offer
A: got offer, set it as local description
A: local offer set
-FAIL pcA.signalingState should be have-local-offer. Was stable.
-FAIL pcA.pendingLocalDescription is not an instance of RTCSessionDescription
+PASS pcA.signalingState is 'have-local-offer'
+PASS pcA.pendingLocalDescription is an instance of RTCSessionDescription
PASS pcA.currentLocalDescription is null
A: send offer to B
A --- offer --> B
B: got offer from A, set it as remote description
-FAIL Error in promise chain: TypeError: Argument 1 ('description') to RTCPeerConnection.setRemoteDescription must be an instance of RTCSessionDescription
+----------
+PASS B: got remote track event
+----------
+B: remote offer set
+PASS pcB.signalingState is 'have-remote-offer'
+PASS pcB.pendingRemoteDescription is an instance of RTCSessionDescription
+PASS pcB.currentRemoteDescription is null
+
+B: create answer (without local media)
+B: got answer, set it as local description
+B: local answer set
+PASS pcB.signalingState is 'stable'
+PASS pcB.pendingLocalDescription is null
+PASS pcB.currentLocalDescription is an instance of RTCSessionDescription
+PASS pcB.pendingRemoteDescription is null
+PASS pcB.currentRemoteDescription is an instance of RTCSessionDescription
+
+B: send answer to A
+A <-- answer -- B
+A: got answer from B, set it as remote description
+A: remote answer set
+PASS pcA.signalingState is 'stable'
+PASS pcA.pendingLocalDescription is null
+PASS pcA.currentLocalDescription is an instance of RTCSessionDescription
+PASS pcA.pendingRemoteDescription is null
+PASS pcA.currentRemoteDescription is an instance of RTCSessionDescription
+
+PASS First offer/answer dialog completed
+========================================
+
+PASS pcB.getSenders().length is 1
+B: add media (should reuse sender)
+PASS pcB.getSenders().length is 1
+B: create offer
+B: got offer, set it as local description
+B: local offer set (session update so we have both pending and current local descriptions)
+PASS pcB.signalingState is 'have-local-offer'
+PASS pcB.pendingLocalDescription is an instance of RTCSessionDescription
+PASS pcB.currentLocalDescription is an instance of RTCSessionDescription
+
+B: send offer to A
+A <-- offer --- B
+A: got offer from B, set it as remote description
+----------
+PASS A: got remote track event
+----------
+A: remote offer set (session update so we have both pending and current remote descriptions)
+PASS pcA.signalingState is 'have-remote-offer'
+PASS pcA.pendingRemoteDescription is an instance of RTCSessionDescription
+PASS pcA.currentRemoteDescription is an instance of RTCSessionDescription
+
+A: create answer
+A: got answer, set it as local description
+A: local answer set
+PASS pcA.signalingState is 'stable'
+PASS pcA.pendingLocalDescription is null
+PASS pcA.currentLocalDescription is an instance of RTCSessionDescription
+PASS pcA.pendingRemoteDescription is null
+PASS pcA.currentRemoteDescription is an instance of RTCSessionDescription
+
+A: send answer to B
+A --- answer -> B
+B: got answer from A, set it as remote description
+B: remote answer set
+PASS pcB.signalingState is 'stable'
+PASS pcB.pendingLocalDescription is null
+PASS pcB.currentLocalDescription is an instance of RTCSessionDescription
+PASS pcB.pendingRemoteDescription is null
+PASS pcB.currentRemoteDescription is an instance of RTCSessionDescription
+
+PASS Second offer/answer dialog completed
+
PASS successfullyParsed is true
TEST COMPLETE
Modified: trunk/LayoutTests/platform/mac/webrtc/captureCanvas-webrtc-software-encoder.html (218794 => 218795)
--- trunk/LayoutTests/platform/mac/webrtc/captureCanvas-webrtc-software-encoder.html 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/platform/mac/webrtc/captureCanvas-webrtc-software-encoder.html 2017-06-25 18:06:06 UTC (rev 218795)
@@ -39,10 +39,10 @@
if (isEqual === isSame)
return;
- if (count === 20)
+ if (count === 40)
return Promise.reject(testName + " failed");
- return waitFor(50).then(() => {
+ return waitFor(100).then(() => {
return testCanvas(testName, array1, isSame, ++count);
});
}
Modified: trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html (218794 => 218795)
--- trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html 2017-06-25 18:06:06 UTC (rev 218795)
@@ -8,6 +8,30 @@
<script src =""
<script>
var context = new webkitAudioContext();
+ var remoteStream;
+
+ async function checkForHumBipBop(stream, previousResults, counter)
+ {
+ if (!previousResults)
+ previousResults = {
+ heardHum : false,
+ heardBip : false,
+ heardBop : false
+ };
+ if (!counter)
+ counter = 1;
+ else if (++counter > 4)
+ return Promise.resolve(false);
+ results = await analyseAudio(stream, 1000, context);
+ previousResults.heardHum |= results.heardHum;
+ previousResults.heardBip |= results.heardBip;
+ previousResults.heardBop |= results.heardBop;
+ if (previousResults.heardHum && previousResults.heardBip && previousResults.heardBop)
+ return Promise.resolve(true);
+ var results = await checkForHumBipBop(stream, previousResults, counter);
+ return results;
+ }
+
promise_test((test) => {
if (window.testRunner)
testRunner.setUserMediaPermission(true);
@@ -21,12 +45,10 @@
});
setTimeout(() => reject("Test timed out"), 5000);
});
- }).then((remoteStream) => {
- return analyseAudio(remoteStream, 1000, context);
+ }).then((stream) => {
+ return checkForHumBipBop(stream);
}).then((results) => {
- assert_true(results.heardHum, "heard hum");
- assert_true(results.heardBip, "heard bip");
- assert_true(results.heardBop, "heard bop");
+ assert_true(results, "heard hum bip bop");
}).then(() => {
return context.close();
});
Modified: trunk/LayoutTests/webrtc/captureCanvas-webrtc.html (218794 => 218795)
--- trunk/LayoutTests/webrtc/captureCanvas-webrtc.html 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/webrtc/captureCanvas-webrtc.html 2017-06-25 18:06:06 UTC (rev 218795)
@@ -40,7 +40,7 @@
if (count === 20)
return Promise.reject(testName + " failed, expected " + JSON.stringify(array1) + " but got " + JSON.stringify(array2));
- return waitFor(50).then(() => {
+ return waitFor(100).then(() => {
return testCanvas(testName, array1, isSame, ++count);
});
}
@@ -89,9 +89,9 @@
}, "Checking canvas is green again");
promise_test((test) => {
- canvas1.width = 640;
- canvas1.height = 480;
- return waitForVideoSize(video, 640, 480);
+ canvas1.width = 640;
+ canvas1.height = 480;
+ return waitForVideoSize(video, 640, 480);
}, "Checking canvas size change");
</script>
</head>
Added: trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-default-expected.txt (0 => 218795)
--- trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-default-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-default-expected.txt 2017-06-25 18:06:06 UTC (rev 218795)
@@ -0,0 +1,3 @@
+
+PASS Default buffer threshold
+
Added: trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-default.html (0 => 218795)
--- trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-default.html (rev 0)
+++ trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-default.html 2017-06-25 18:06:06 UTC (rev 218795)
@@ -0,0 +1,66 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>Testing bufferedAmountLowThreashold default RTCDataChannel attribute and event</title>
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <script src =""
+ <script>
+window.addEventListener("unhandledrejection", event => {
+ event.preventDefault();
+});
+
+var localChannel;
+var remoteChannel;
+
+function closeDataChannels() {
+ localChannel.close();
+ remoteChannel.close();
+ closeConnections();
+}
+
+var longString = "abcdefgh";
+for (var cptr = 0; cptr < 14; ++cptr)
+ longString += longString;
+
+function sendMessages(channel)
+{
+ channel.send(longString);
+ channel.send(longString);
+ channel.send(longString);
+}
+
+function receiveMessages(event) {
+ if (++counter === 1)
+ return;
+}
+
+promise_test((test) => {
+ counter = 0;
+ var gotEvent = false;
+ return new Promise((resolve, reject) => {
+ createConnections((localConnection) => {
+ localChannel = localConnection.createDataChannel('sendDataChannel');
+ localChannel._onopen_ = () => { sendMessages(localChannel); };
+ localChannel._onbufferedamountlow_ = () => {
+ resolve();
+ }
+ assert_equals(localChannel.bufferedAmountLowThreshold, 0, "default bufferedAmountLowThreshold value");
+ }, (remoteConnection) => {
+ remoteConnection._ondatachannel_ = (event) => {
+ remoteChannel = event.channel;
+ remoteChannel._onmessage_ = receiveMessages;
+ };
+ });
+ setTimeout(() => reject('timed out'), 5000);
+ }).then (() => {
+ closeDataChannels();
+ });
+}, "Default buffer threshold");
+
+ </script>
+ </body>
+</html>
Modified: trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-expected.txt (218794 => 218795)
--- trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-expected.txt 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold-expected.txt 2017-06-25 18:06:06 UTC (rev 218795)
@@ -1,5 +1,4 @@
-PASS Default buffer threshold
PASS Large buffer threshold reached
PASS Medium buffer threshold not reached
Modified: trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold.html (218794 => 218795)
--- trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold.html 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/webrtc/datachannel/bufferedAmountLowThreshold.html 2017-06-25 18:06:06 UTC (rev 218795)
@@ -26,41 +26,11 @@
for (var cptr = 0; cptr < 14; ++cptr)
longString += longString;
-function sendMessages(channel)
-{
- channel.send(longString);
- channel.send(longString);
- channel.send(longString);
-}
-
function receiveMessages(event) {
if (++counter === 1)
return;
}
-promise_test((test) => {
- counter = 0;
- var gotEvent = false;
- return new Promise((resolve, reject) => {
- createConnections((localConnection) => {
- localChannel = localConnection.createDataChannel('sendDataChannel');
- localChannel._onopen_ = () => { sendMessages(localChannel); };
- localChannel._onbufferedamountlow_ = () => {
- resolve();
- }
- assert_equals(localChannel.bufferedAmountLowThreshold, 0, "default bufferedAmountLowThreshold value");
- }, (remoteConnection) => {
- remoteConnection._ondatachannel_ = (event) => {
- remoteChannel = event.channel;
- remoteChannel._onmessage_ = receiveMessages;
- };
- });
- setTimeout(() => reject('timed out'), 5000);
- }).then (() => {
- closeDataChannels();
- });
-}, "Default buffer threshold");
-
function sendContinuouslyMessages(channel)
{
try {
Modified: trunk/LayoutTests/webrtc/routines.js (218794 => 218795)
--- trunk/LayoutTests/webrtc/routines.js 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/webrtc/routines.js 2017-06-25 18:06:06 UTC (rev 218795)
@@ -144,7 +144,7 @@
if (++count > 20)
return Promise.reject("waitForVideoSize timed out, expected " + width + "x"+ height + " but got " + video.videoWidth + "x" + video.videoHeight);
- return waitFor(50).then(() => {
+ return waitFor(100).then(() => {
return waitForVideoSize(video, width, height, count);
});
}
@@ -152,10 +152,11 @@
async function doHumAnalysis(stream, expected)
{
var context = new webkitAudioContext();
- for (var cptr = 0; cptr < 10; cptr++) {
+ for (var cptr = 0; cptr < 20; cptr++) {
var results = await analyseAudio(stream, 200, context);
if (results.heardHum === expected)
return true;
+ await waitFor(50);
}
await context.close();
return false;
Modified: trunk/LayoutTests/webrtc/video-replace-muted-track.html (218794 => 218795)
--- trunk/LayoutTests/webrtc/video-replace-muted-track.html 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/webrtc/video-replace-muted-track.html 2017-06-25 18:06:06 UTC (rev 218795)
@@ -32,7 +32,7 @@
function pollVideoBlackCheck(expected, resolve, delay)
{
if (isVideoBlack() === expected || stopPolling) {
- resolve();
+ resolve(true);
return;
}
@@ -39,13 +39,13 @@
setTimeout(() => pollVideoBlackCheck(expected, resolve), delay);
}
-function checkVideoBlack(expected, duration, id, delay)
+async function checkVideoBlack(expected, duration, id, delay)
{
stopPolling = false;
return new Promise((resolve, reject) => {
pollVideoBlackCheck(expected, resolve, delay);
setTimeout(() => {
- reject("checkVideoBlack timed out for " + id + ", expected " + expected);
+ resolve(false);
stopPolling = true;
}, duration);
});
@@ -61,6 +61,15 @@
return imageData.data;
}
+async function checkForNoBlackFrames()
+{
+ for (var cptr = 0; cptr < 10; cptr++) {
+ if (!(await checkVideoBlack(true, 1500, "test3", 50)))
+ return true;
+ }
+ return false;
+}
+
promise_test((test) => {
if (window.testRunner)
testRunner.setUserMediaPermission(true);
@@ -93,11 +102,14 @@
}).then(() => {
return waitFor(200);
}).then(() => {
- // Let's ensure we have some non black frames sent.
return checkVideoBlack(false, 5000, "test2", 200);
+ }).then((result) => {
+ assert_true(result, "expect to receive some non black frames");
}).then(() => {
- // Let's ensure there are no black frames sent.
- return checkVideoBlack(true, 2000, "test3", 50).then(assert_unreached, () => { });
+ // Let's ensure there are no black frames being received.
+ return checkForNoBlackFrames();
+ }).then((result) => {
+ assert_true(result, "check no black frame");
});
}, "Switching from disabled to enabled track");
</script>
Modified: trunk/LayoutTests/webrtc/video-unmute.html (218794 => 218795)
--- trunk/LayoutTests/webrtc/video-unmute.html 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/LayoutTests/webrtc/video-unmute.html 2017-06-25 18:06:06 UTC (rev 218795)
@@ -40,7 +40,7 @@
if (!counter)
counter = 0;
- else if (++counter > 20) {
+ else if (++counter > 40) {
reject("test for " + id + " timed out");
return;
}
@@ -50,7 +50,7 @@
function checkVideoBlack(expected, id)
{
return new Promise((resolve, reject) => {
- pollVideoBlackCheck(expected, id, resolve);
+ pollVideoBlackCheck(expected, id, resolve, reject);
});
}
Modified: trunk/Source/WebCore/ChangeLog (218794 => 218795)
--- trunk/Source/WebCore/ChangeLog 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/Source/WebCore/ChangeLog 2017-06-25 18:06:06 UTC (rev 218795)
@@ -1,3 +1,20 @@
+2017-06-25 Youenn Fablet <[email protected]>
+
+ Remove use of mock webrtc backend factory at injected bundle reset time
+ https://bugs.webkit.org/show_bug.cgi?id=173817
+
+ Reviewed by Darin Adler.
+
+ * testing/Internals.cpp:
+ (WebCore::Internals::resetToConsistentState): Resetting the peer connection backend.
+ * testing/MockLibWebRTCPeerConnection.cpp: Doing some clean-up
+ (WebCore::useRealRTCPeerConnectionFactory):
+ (WebCore::MockLibWebRTCPeerConnectionFactory::MockLibWebRTCPeerConnectionFactory):
+ (WebCore::MockLibWebRTCPeerConnectionFactory::CreatePeerConnection):
+ (WebCore::MockLibWebRTCPeerConnectionFactory::CreateVideoTrack):
+ (WebCore::MockLibWebRTCPeerConnectionFactory::CreateAudioTrack):
+ * testing/MockLibWebRTCPeerConnection.h:
+
2017-06-25 Konstantin Tokarev <[email protected]>
Remove excessive headers from _javascript_Core
Modified: trunk/Source/WebCore/testing/Internals.cpp (218794 => 218795)
--- trunk/Source/WebCore/testing/Internals.cpp 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/Source/WebCore/testing/Internals.cpp 2017-06-25 18:06:06 UTC (rev 218795)
@@ -465,6 +465,10 @@
#endif
printContextForTesting() = nullptr;
+
+#if USE(LIBWEBRTC)
+ WebCore::useRealRTCPeerConnectionFactory();
+#endif
}
Internals::Internals(Document& document)
Modified: trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp (218794 => 218795)
--- trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp 2017-06-25 18:06:06 UTC (rev 218795)
@@ -47,6 +47,15 @@
return getRealPeerConnectionFactory().get();
}
+void useRealRTCPeerConnectionFactory()
+{
+ auto& factory = getRealPeerConnectionFactory();
+ if (!factory)
+ return;
+ LibWebRTCProvider::setPeerConnectionFactory(factory.get());
+ factory = nullptr;
+}
+
void useMockRTCPeerConnectionFactory(LibWebRTCProvider* provider, const String& testCase)
{
if (provider && !realPeerConnectionFactory()) {
@@ -152,25 +161,10 @@
MockLibWebRTCPeerConnectionFactory::MockLibWebRTCPeerConnectionFactory(String&& testCase)
: m_testCase(WTFMove(testCase))
{
- if (m_testCase == "TwoRealPeerConnections") {
- m_numberOfRealPeerConnections = 2;
- return;
- }
- if (m_testCase == "OneRealPeerConnection")
- m_numberOfRealPeerConnections = 1;
}
-rtc::scoped_refptr<webrtc::PeerConnectionInterface> MockLibWebRTCPeerConnectionFactory::CreatePeerConnection(const webrtc::PeerConnectionInterface::RTCConfiguration& configuration, std::unique_ptr<cricket::PortAllocator> portAllocator, std::unique_ptr<rtc::RTCCertificateGeneratorInterface> generator, webrtc::PeerConnectionObserver* observer)
+rtc::scoped_refptr<webrtc::PeerConnectionInterface> MockLibWebRTCPeerConnectionFactory::CreatePeerConnection(const webrtc::PeerConnectionInterface::RTCConfiguration&, std::unique_ptr<cricket::PortAllocator>, std::unique_ptr<rtc::RTCCertificateGeneratorInterface>, webrtc::PeerConnectionObserver* observer)
{
- if (!realPeerConnectionFactory())
- return nullptr;
-
- if (m_numberOfRealPeerConnections) {
- auto connection = realPeerConnectionFactory()->CreatePeerConnection(configuration, WTFMove(portAllocator), WTFMove(generator), observer);
- --m_numberOfRealPeerConnections;
- return connection;
- }
-
if (m_testCase == "ICECandidates")
return new rtc::RefCountedObject<MockLibWebRTCPeerConnectionForIceCandidates>(*observer);
@@ -191,15 +185,11 @@
rtc::scoped_refptr<webrtc::VideoTrackInterface> MockLibWebRTCPeerConnectionFactory::CreateVideoTrack(const std::string& id, webrtc::VideoTrackSourceInterface* source)
{
- if (m_testCase == "TwoRealPeerConnections")
- return realPeerConnectionFactory()->CreateVideoTrack(id, source);
return new rtc::RefCountedObject<MockLibWebRTCVideoTrack>(id, source);
}
rtc::scoped_refptr<webrtc::AudioTrackInterface> MockLibWebRTCPeerConnectionFactory::CreateAudioTrack(const std::string& id, webrtc::AudioSourceInterface* source)
{
- if (m_testCase == "TwoRealPeerConnections")
- return realPeerConnectionFactory()->CreateAudioTrack(id, source);
return new rtc::RefCountedObject<MockLibWebRTCAudioTrack>(id, source);
}
Modified: trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h (218794 => 218795)
--- trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h 2017-06-25 17:40:30 UTC (rev 218794)
+++ trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h 2017-06-25 18:06:06 UTC (rev 218795)
@@ -37,6 +37,7 @@
class MockRtpSender;
void useMockRTCPeerConnectionFactory(LibWebRTCProvider*, const String&);
+void useRealRTCPeerConnectionFactory();
class MockLibWebRTCPeerConnection : public webrtc::PeerConnectionInterface {
public:
@@ -256,7 +257,6 @@
private:
String m_testCase;
- unsigned m_numberOfRealPeerConnections { 0 };
};
} // namespace WebCore
