diff options
Diffstat (limited to 'testing/web-platform/meta/webrtc')
7 files changed, 30 insertions, 10 deletions
diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-send-close.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-send-close.html.ini index e9f2be5f4f..63a12be1c6 100644 --- a/testing/web-platform/meta/webrtc/RTCDataChannel-send-close.html.ini +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-send-close.html.ini @@ -1,7 +1,6 @@ [RTCDataChannel-send-close.html] expected: - if (os == "win") and (processor == "x86"): CRASH - if os == "android": CRASH + if os == "android": [CRASH, TIMEOUT] [TIMEOUT, OK] [Datachannel should be able to send and receive all string messages on close] expected: [FAIL, TIMEOUT, NOTRUN] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini index 7fc41ec7d8..ff8279db28 100644 --- a/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini @@ -1,4 +1,6 @@ [RTCPeerConnection-iceConnectionState.https.html] + expected: + if tsan: CRASH [iceConnectionState changes at the right time, with bundle policy max-bundle] bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html.ini new file mode 100644 index 0000000000..fa8ca81b19 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html.ini @@ -0,0 +1,3 @@ +[RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html] + [measure raising and lowering audio jitterBufferTarget] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html.ini new file mode 100644 index 0000000000..511e0d93b3 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html.ini @@ -0,0 +1,3 @@ +[RTCRtpReceiver-video-jitterBufferTarget-stats.html] + [measure raising and lowering video jitterBufferTarget] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini b/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini index 6a26d4498c..37d9821fde 100644 --- a/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini +++ b/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini @@ -215,26 +215,29 @@ [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "address" with the proper type] expected: FAIL - [RTCRtpTransceiver interface: operation setCodecPreferences(sequence<RTCRtpCodecCapability>)] + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "binaryType" with the proper type] expected: FAIL - [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "setCodecPreferences(sequence<RTCRtpCodecCapability>)" with the proper type] + [RTCIceCandidate interface: attribute relayProtocol] expected: FAIL - [RTCRtpTransceiver interface: calling setCodecPreferences(sequence<RTCRtpCodecCapability>) on new RTCPeerConnection().addTransceiver('audio') with too few arguments must throw TypeError] + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relayProtocol" with the proper type] expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "binaryType" with the proper type] + [RTCIceCandidate interface: attribute url] expected: FAIL - [RTCIceCandidate interface: attribute relayProtocol] + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "url" with the proper type] expected: FAIL - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relayProtocol" with the proper type] + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "jitterBufferTarget" with the proper type] expected: FAIL - [RTCIceCandidate interface: attribute url] + [RTCRtpTransceiver interface: operation setCodecPreferences(sequence<RTCRtpCodec>)] expected: FAIL - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "url" with the proper type] + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "setCodecPreferences(sequence<RTCRtpCodec>)" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface: calling setCodecPreferences(sequence<RTCRtpCodec>) on new RTCPeerConnection().addTransceiver('audio') with too few arguments must throw TypeError] expected: FAIL diff --git a/testing/web-platform/meta/webrtc/legacy/simplecall_callbacks.https.html.ini b/testing/web-platform/meta/webrtc/legacy/simplecall_callbacks.https.html.ini new file mode 100644 index 0000000000..f640335196 --- /dev/null +++ b/testing/web-platform/meta/webrtc/legacy/simplecall_callbacks.https.html.ini @@ -0,0 +1,6 @@ +[simplecall_callbacks.https.html] + expected: + if (os == "mac") and not debug: [OK, TIMEOUT] + [Can set up a basic WebRTC call.] + expected: + if (os == "mac") and not debug: [PASS, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini new file mode 100644 index 0000000000..9a1ece0772 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini @@ -0,0 +1,4 @@ +[setParameters-active.https.html] + [Simulcast setParameters active=false stops sending frames] + expected: + if (os == "mac") and not debug: [PASS, FAIL] |