webrtc wpt: add missing pc.close during cleanup running the html.js/codemod-peerconnection-addcleanup from the tools directory showed some places where a pc.close was missing BUG=836871 Change-Id: I135db200a4269eccfafc43217b66584603101434 Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1610767 Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Reviewed-by: Henrik Boström <hbos@chromium.org> Cr-Commit-Position: refs/heads/master@{#661283} diff --git a/webrtc/RTCCertificate-postMessage.html b/webrtc/RTCCertificate-postMessage.html index 5885f9f..ee8347c 100644 --- a/webrtc/RTCCertificate-postMessage.html +++ b/webrtc/RTCCertificate-postMessage.html
@@ -42,9 +42,10 @@ iframe.contentWindow.postMessage(certificate, "*"); let certificate2 = await promise; - new RTCPeerConnection({certificates: [certificate]}); - - new RTCPeerConnection({certificates: [certificate2]}); + const pc1 = new RTCPeerConnection({certificates: [certificate]}); + t.add_cleanup(() => pc1.close()); + const pc2 = new RTCPeerConnection({certificates: [certificate2]}); + t.add_cleanup(() => pc2.close()); assert_equals(certificate.expires, certificate2.expires); for (let fingerprint of certificate2.getFingerprints()) diff --git a/webrtc/RTCDataChannel-send.html b/webrtc/RTCDataChannel-send.html index 76d3524..4565a83 100644 --- a/webrtc/RTCDataChannel-send.html +++ b/webrtc/RTCDataChannel-send.html
@@ -300,6 +300,7 @@ promise_test(async t => { let pc1 = new RTCPeerConnection(); + t.add_cleanup(() => pc1.close()); let [channel1, channel2] = await createDataChannelPair(pc1); let message = 'hello888'; // 8 bytes while (message.length <= pc1.sctp.maxMessageSize) { diff --git a/webrtc/RTCIceTransport-extension.https.html b/webrtc/RTCIceTransport-extension.https.html index 206a4bb..94beb99 100644 --- a/webrtc/RTCIceTransport-extension.https.html +++ b/webrtc/RTCIceTransport-extension.https.html
@@ -98,7 +98,7 @@ iceTransport.gather({}); let candidate; do { - ({ candidate } = await watcher.wait_for('icecandidate')); + (({ candidate } = await watcher.wait_for('icecandidate'))); } while (candidate !== null); assert_equals(iceTransport.gatheringState, 'gathering'); await watcher.wait_for('gatheringstatechange'); diff --git a/webrtc/RTCPeerConnection-setLocalDescription.html b/webrtc/RTCPeerConnection-setLocalDescription.html index 2becbd3..c4671c3 100644 --- a/webrtc/RTCPeerConnection-setLocalDescription.html +++ b/webrtc/RTCPeerConnection-setLocalDescription.html
@@ -125,6 +125,7 @@ promise_test(async t => { const pc = new RTCPeerConnection(); + t.add_cleanup(() => pc.close()); const offer = await pc.createOffer(); let eventSequence = ''; const signalingstatechangeResolver = new Resolver(); diff --git a/webrtc/RTCRtpParameters-transactionId.html b/webrtc/RTCRtpParameters-transactionId.html index 472b043..eb98a83 100644 --- a/webrtc/RTCRtpParameters-transactionId.html +++ b/webrtc/RTCRtpParameters-transactionId.html
@@ -63,6 +63,7 @@ */ promise_test(async t => { const pc = new RTCPeerConnection(); + t.add_cleanup(() => pc.close()); const { sender } = pc.addTransceiver('audio'); await doOfferAnswerExchange(t, pc); @@ -73,7 +74,6 @@ validateSenderRtpParameters(param2); assert_not_equals(param1.transactionId, param2.transactionId); - }, `sender.getParameters() should return different transaction IDs for each call`); /*