diff options
| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
| commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
| tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /testing/web-platform/meta/webrtc | |
| parent | Initial commit. (diff) | |
| download | firefox-upstream.tar.xz firefox-upstream.zip | |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/meta/webrtc')
165 files changed, 1771 insertions, 0 deletions
diff --git a/testing/web-platform/meta/webrtc/RTCCertificate-postMessage.html.ini b/testing/web-platform/meta/webrtc/RTCCertificate-postMessage.html.ini new file mode 100644 index 0000000000..0d2beefdb3 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCCertificate-postMessage.html.ini @@ -0,0 +1,13 @@ +[RTCCertificate-postMessage.html] + [Check cross-origin created RTCCertificate] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531875 + + [Check cross-origin RTCCertificate serialization] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1525241 + + [Check same-origin RTCCertificate serialization] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1525241 + diff --git a/testing/web-platform/meta/webrtc/RTCCertificate.html.ini b/testing/web-platform/meta/webrtc/RTCCertificate.html.ini new file mode 100644 index 0000000000..e4a56f48cf --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCCertificate.html.ini @@ -0,0 +1,12 @@ +[RTCCertificate.html] + [RTCCertificate should have at least one fingerprint] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1525241 + expected: FAIL + + [RTCPeerConnection({ certificates }) should generate offer SDP with fingerprint of provided certificate] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1525241 + expected: FAIL + + [RTCPeerConnection({ certificates }) should generate offer SDP with fingerprint of all provided certificates] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531880 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCConfiguration-bundlePolicy.html.ini b/testing/web-platform/meta/webrtc/RTCConfiguration-bundlePolicy.html.ini new file mode 100644 index 0000000000..8a7347a085 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCConfiguration-bundlePolicy.html.ini @@ -0,0 +1,3 @@ +[RTCConfiguration-bundlePolicy.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini b/testing/web-platform/meta/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini new file mode 100644 index 0000000000..c73263bfc8 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini @@ -0,0 +1,3 @@ +[RTCConfiguration-iceCandidatePoolSize.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1529398 + diff --git a/testing/web-platform/meta/webrtc/RTCConfiguration-iceServers.html.ini b/testing/web-platform/meta/webrtc/RTCConfiguration-iceServers.html.ini new file mode 100644 index 0000000000..2098e6171c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCConfiguration-iceServers.html.ini @@ -0,0 +1,30 @@ +[RTCConfiguration-iceServers.html] + [setConfiguration(config) - with url field should throw TypeError] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1529588 + expected: FAIL + + [new RTCPeerConnection(config) - with url field should throw TypeError] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1529588 + expected: FAIL + + [setConfiguration(config) - with invalid stun url should throw SyntaxError] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1529588 + expected: FAIL + + [new RTCPeerConnection(config) - with invalid stun url should throw SyntaxError] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1529588 + expected: FAIL + + [setConfiguration(config) - with invalid turn url should throw SyntaxError] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1529588 + expected: FAIL + + [new RTCPeerConnection(config) - with invalid turn url should throw SyntaxError] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1529588 + expected: FAIL + + [new RTCPeerConnection(config) - with turns server, and object credential should throw InvalidAccessError] + expected: FAIL + + [setConfiguration(config) - with turns server, and object credential should throw InvalidAccessError] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCConfiguration-iceTransportPolicy.html.ini b/testing/web-platform/meta/webrtc/RTCConfiguration-iceTransportPolicy.html.ini new file mode 100644 index 0000000000..836f49a014 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCConfiguration-iceTransportPolicy.html.ini @@ -0,0 +1,3 @@ +[RTCConfiguration-iceTransportPolicy.html] + expected: + if (os == "android") and fission: [TIMEOUT, OK] diff --git a/testing/web-platform/meta/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini b/testing/web-platform/meta/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini new file mode 100644 index 0000000000..44c813e62f --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini @@ -0,0 +1,3 @@ +[RTCConfiguration-rtcpMuxPolicy.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1339203 + diff --git a/testing/web-platform/meta/webrtc/RTCDTMFSender-insertDTMF.https.html.ini b/testing/web-platform/meta/webrtc/RTCDTMFSender-insertDTMF.https.html.ini new file mode 100644 index 0000000000..fcf7927101 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDTMFSender-insertDTMF.https.html.ini @@ -0,0 +1,5 @@ +[RTCDTMFSender-insertDTMF.https.html] + expected: + if (os == "linux") and not debug and fission and (processor == "x86"): [OK, CRASH] + if (os == "linux") and not debug and fission and (processor == "x86_64"): [OK, CRASH] + if (os == "linux") and not debug and not fission: [OK, CRASH] diff --git a/testing/web-platform/meta/webrtc/RTCDTMFSender-ontonechange-long.https.html.ini b/testing/web-platform/meta/webrtc/RTCDTMFSender-ontonechange-long.https.html.ini new file mode 100644 index 0000000000..5e38bec776 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDTMFSender-ontonechange-long.https.html.ini @@ -0,0 +1,7 @@ +[RTCDTMFSender-ontonechange-long.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [insertDTMF with duration greater than 6000 should be clamped to 6000] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1402717 + expected: + if (os == "win") and not debug and (processor == "x86_64"): [PASS, FAIL] diff --git a/testing/web-platform/meta/webrtc/RTCDTMFSender-ontonechange.https.html.ini b/testing/web-platform/meta/webrtc/RTCDTMFSender-ontonechange.https.html.ini new file mode 100644 index 0000000000..8795a16236 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDTMFSender-ontonechange.https.html.ini @@ -0,0 +1,8 @@ +[RTCDTMFSender-ontonechange.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Calling insertDTMF() multiple times in the middle of tonechange events should cause future tonechanges to be updated the last provided tones] + expected: + if (processor == "x86") and (os == "win") and not debug: [PASS, FAIL] diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-binaryType.window.js.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-binaryType.window.js.ini new file mode 100644 index 0000000000..e1acfa6e3c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-binaryType.window.js.ini @@ -0,0 +1,21 @@ +[RTCDataChannel-binaryType.window.html] + [Setting invalid binaryType 'arraybuffer ' should throw SyntaxError] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728325 + + [Setting invalid binaryType 'undefined' should throw SyntaxError] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728325 + + [Setting invalid binaryType 'null' should throw SyntaxError] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728325 + + [Setting invalid binaryType '' should throw SyntaxError] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728325 + + [Setting invalid binaryType 'jellyfish' should throw SyntaxError] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728325 + diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-close.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-close.html.ini new file mode 100644 index 0000000000..b7758b5e86 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-close.html.ini @@ -0,0 +1,34 @@ +[RTCDataChannel-close.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Close datachannel causes onclosing and onclose to be called] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1611953 + expected: FAIL + + [Close datachannel causes closing and close event to be called] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1641026 + expected: FAIL + + [Close peerconnection causes close event and error to be called on datachannel] + bug: Probably https://bugzilla.mozilla.org/show_bug.cgi?id=1611953 + expected: FAIL + + [Close negotiated datachannel causes closing and close event to be called] + bug: Probably https://bugzilla.mozilla.org/show_bug.cgi?id=1641026 + expected: FAIL + + [Close negotiated datachannel causes onclosing and onclose to be called] + bug: Probably https://bugzilla.mozilla.org/show_bug.cgi?id=1611953 + expected: FAIL + + [Close peerconnection causes close event and error to be called on negotiated datachannel] + bug: Probably https://bugzilla.mozilla.org/show_bug.cgi?id=1611953 + expected: FAIL + + [Close peerconnection causes close event and error on many channels, negotiated datachannel] + bug: Probably https://bugzilla.mozilla.org/show_bug.cgi?id=1611953 + expected: FAIL + + [Close peerconnection causes close event and error on many channels, datachannel] + bug: Probably https://bugzilla.mozilla.org/show_bug.cgi?id=1611953 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-iceRestart.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-iceRestart.html.ini new file mode 100644 index 0000000000..a0d14b4c91 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-iceRestart.html.ini @@ -0,0 +1,5 @@ +[RTCDataChannel-iceRestart.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728342 + expected: + if (os == "android") and fission: [ERROR, TIMEOUT] + ERROR diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-id.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-id.html.ini new file mode 100644 index 0000000000..3176e3a2dd --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-id.html.ini @@ -0,0 +1,3 @@ +[RTCDataChannel-id.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-send-blob-order.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-send-blob-order.html.ini new file mode 100644 index 0000000000..719963a084 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-send-blob-order.html.ini @@ -0,0 +1,2 @@ +[RTCDataChannel-send-blob-order.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1577830 diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-send.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-send.html.ini new file mode 100644 index 0000000000..da2dbd3928 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-send.html.ini @@ -0,0 +1,8 @@ +[RTCDataChannel-send.html] + [Datachannel send() up to max size should succeed, above max size should fail] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [Negotiated datachannel send() up to max size should succeed, above max size should fail] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCDataChannelEvent-constructor.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannelEvent-constructor.html.ini new file mode 100644 index 0000000000..f10331515f --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannelEvent-constructor.html.ini @@ -0,0 +1,3 @@ +[RTCDataChannelEvent-constructor.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCDtlsTransport-getRemoteCertificates.html.ini b/testing/web-platform/meta/webrtc/RTCDtlsTransport-getRemoteCertificates.html.ini new file mode 100644 index 0000000000..c11e369673 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDtlsTransport-getRemoteCertificates.html.ini @@ -0,0 +1,3 @@ +[RTCDtlsTransport-getRemoteCertificates.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + diff --git a/testing/web-platform/meta/webrtc/RTCDtlsTransport-state.html.ini b/testing/web-platform/meta/webrtc/RTCDtlsTransport-state.html.ini new file mode 100644 index 0000000000..8cc396c9f9 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDtlsTransport-state.html.ini @@ -0,0 +1,2 @@ +[RTCDtlsTransport-state.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 diff --git a/testing/web-platform/meta/webrtc/RTCError.html.ini b/testing/web-platform/meta/webrtc/RTCError.html.ini new file mode 100644 index 0000000000..c18125686c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCError.html.ini @@ -0,0 +1,3 @@ +[RTCError.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1527916 + diff --git a/testing/web-platform/meta/webrtc/RTCIceCandidate-constructor.html.ini b/testing/web-platform/meta/webrtc/RTCIceCandidate-constructor.html.ini new file mode 100644 index 0000000000..09e167e607 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCIceCandidate-constructor.html.ini @@ -0,0 +1,10 @@ +[RTCIceCandidate-constructor.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [new RTCIceCandidate({ ... }) with nondefault values for all fields, tcp candidate] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1322186 + expected: FAIL + + [new RTCIceCandidate({ ... }) with nondefault values for all fields] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1322186 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCIceConnectionState-candidate-pair.https.html.ini b/testing/web-platform/meta/webrtc/RTCIceConnectionState-candidate-pair.https.html.ini new file mode 100644 index 0000000000..c4716c7a10 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCIceConnectionState-candidate-pair.https.html.ini @@ -0,0 +1,3 @@ +[RTCIceConnectionState-candidate-pair.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCIceTransport.html.ini b/testing/web-platform/meta/webrtc/RTCIceTransport.html.ini new file mode 100644 index 0000000000..8c69d2d02b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCIceTransport.html.ini @@ -0,0 +1,3 @@ +[RTCIceTransport.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1307994 + diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-SLD-SRD-timing.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-SLD-SRD-timing.https.html.ini new file mode 100644 index 0000000000..c77caf17eb --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-SLD-SRD-timing.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-SLD-SRD-timing.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini new file mode 100644 index 0000000000..f63e7601fe --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-add-track-no-deadlock.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate-connectionSetup.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate-connectionSetup.html.ini new file mode 100644 index 0000000000..0bb6046112 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate-connectionSetup.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-addIceCandidate-connectionSetup.html] + expected: + if (os == "android") and fission: [TIMEOUT, OK] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate-timing.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate-timing.https.html.ini new file mode 100644 index 0000000000..c5c95119bd --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate-timing.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-addIceCandidate-timing.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate.html.ini new file mode 100644 index 0000000000..1342de4090 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-addIceCandidate.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-addTrack.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-addTrack.https.html.ini new file mode 100644 index 0000000000..0fed304db5 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-addTrack.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-addTrack.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-addTransceiver.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-addTransceiver.https.html.ini new file mode 100644 index 0000000000..021fb12c16 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-addTransceiver.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-addTransceiver.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-canTrickleIceCandidates.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-canTrickleIceCandidates.html.ini new file mode 100644 index 0000000000..a0670048b0 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-canTrickleIceCandidates.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-canTrickleIceCandidates.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-candidate-in-sdp.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-candidate-in-sdp.https.html.ini new file mode 100644 index 0000000000..8ec019df36 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-candidate-in-sdp.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-candidate-in-sdp.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-capture-video.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-capture-video.https.html.ini new file mode 100644 index 0000000000..51cce359d7 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-capture-video.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-capture-video.https.html] + disabled: true + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1541471 diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-connectionState.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-connectionState.https.html.ini new file mode 100644 index 0000000000..9b3b7b5d8b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-connectionState.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-connectionState.https.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1265827 + diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-constructor.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-constructor.html.ini new file mode 100644 index 0000000000..c0e4f5b87d --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-constructor.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-constructor.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [new RTCPeerConnection({ iceCandidatePoolSize: toNumberThrows })] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1529398 + expected: FAIL + + [connectionState initial value] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1265827 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-createAnswer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-createAnswer.html.ini new file mode 100644 index 0000000000..35c57c1175 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-createAnswer.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-createAnswer.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-createDataChannel.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-createDataChannel.html.ini new file mode 100644 index 0000000000..4dc340406a --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-createDataChannel.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-createDataChannel.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-createOffer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-createOffer.html.ini new file mode 100644 index 0000000000..644e655a6b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-createOffer.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-createOffer.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-description-attributes-timing.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-description-attributes-timing.https.html.ini new file mode 100644 index 0000000000..e9fe32efdc --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-description-attributes-timing.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-description-attributes-timing.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-explicit-rollback-iceGatheringState.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-explicit-rollback-iceGatheringState.html.ini new file mode 100644 index 0000000000..f0b918067b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-explicit-rollback-iceGatheringState.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-explicit-rollback-iceGatheringState.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-generateCertificate.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-generateCertificate.html.ini new file mode 100644 index 0000000000..3b39eaf57b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-generateCertificate.html.ini @@ -0,0 +1,7 @@ +[RTCPeerConnection-generateCertificate.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [generateCertificate() with 0 expires parameter should generate expired cert] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1402717 + expected: + if os == "win": [PASS, FAIL] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-getStats.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-getStats.https.html.ini new file mode 100644 index 0000000000..3a0ca2faaf --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-getStats.https.html.ini @@ -0,0 +1,50 @@ +[RTCPeerConnection-getStats.https.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: TIMEOUT + [getStats() with no argument should return stats report containing peer-connection stats on an empty PC] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531087 + expected: FAIL + + [getStats() with connected peer connections having tracks and data channel should return all mandatory to implement stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225720 + expected: FAIL + + [getStats() track with stream returns peer-connection and outbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: TIMEOUT + + [getStats() track without stream returns peer-connection and outbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN + + [getStats() audio outbound-rtp contains all mandatory stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN + + [getStats() video outbound-rtp contains all mandatory stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN + + [getStats() audio and video validate all mandatory stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN + + [getStats() on track associated with RTCRtpSender should return stats report containing outbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN + + [getStats() on track associated with RTCRtpReceiver should return stats report containing inbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN + + [getStats() inbound-rtp contains all mandatory stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN + + [getStats(track) should not work if multiple senders have the same track] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN + + [RTCStats.timestamp increases with time passing] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: NOTRUN diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-getTransceivers.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-getTransceivers.html.ini new file mode 100644 index 0000000000..3f0356a39e --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-getTransceivers.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-getTransceivers.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-helper-test.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-helper-test.html.ini new file mode 100644 index 0000000000..28c870e1ee --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-helper-test.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-helper-test.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini new file mode 100644 index 0000000000..061c4df4ae --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-iceConnectionState-disconnected.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini new file mode 100644 index 0000000000..d51203fe9b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini @@ -0,0 +1,18 @@ +[RTCPeerConnection-iceConnectionState.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [connection with one data channel should eventually have connected connection state] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [iceConnectionState changes at the right time, with bundle policy max-bundle] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL + + [iceConnectionState changes at the right time, with bundle policy max-compat] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL + + [iceConnectionState changes at the right time, with bundle policy balanced] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-iceGatheringState.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceGatheringState.html.ini new file mode 100644 index 0000000000..c0b4aa8954 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceGatheringState.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-iceGatheringState.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [connection with one data channel should eventually have connected connection state] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [renegotiation that closes all transports should result in ICE gathering state "new"] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728353 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-mandatory-getStats.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-mandatory-getStats.https.html.ini new file mode 100644 index 0000000000..4d0890a047 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-mandatory-getStats.https.html.ini @@ -0,0 +1,93 @@ +[RTCPeerConnection-mandatory-getStats.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [RTCRtpStreamStats's transportId] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [RTCPeerConnectionStats's dataChannelsOpened] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531087 + expected: FAIL + + [RTCPeerConnectionStats's dataChannelsClosed] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531087 + expected: FAIL + + [RTCTransportStats's bytesSent] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [RTCTransportStats's bytesReceived] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [RTCTransportStats's selectedCandidatePairId] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [RTCTransportStats's localCertificateId] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [RTCTransportStats's remoteCertificateId] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [RTCIceCandidatePairStats's totalRoundTripTime] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1542938 + expected: FAIL + + [RTCIceCandidatePairStats's currentRoundTripTime] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1542938 + expected: FAIL + + [RTCIceCandidateStats's url] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1508543 + expected: FAIL + + [RTCCertificateStats's fingerprint] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225724 + expected: FAIL + + [RTCCertificateStats's fingerprintAlgorithm] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225724 + expected: FAIL + + [RTCCertificateStats's base64Certificate] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225724 + expected: FAIL + + [RTCReceivedRtpStreamStats's framesDropped] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728357 + expected: FAIL + + [RTCMediaSourceStats's trackIdentifier] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728364 + expected: FAIL + + [RTCMediaSourceStats's kind] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728364 + expected: FAIL + + [RTCAudioSourceStats's totalAudioEnergy] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728364 + expected: FAIL + + [RTCAudioSourceStats's totalSamplesDuration] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728364 + expected: FAIL + + [RTCVideoSourceStats's width] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728364 + expected: FAIL + + [RTCVideoSourceStats's height] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728364 + expected: FAIL + + [RTCVideoSourceStats's framesPerSecond] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728364 + expected: FAIL + + [RTCInboundRtpStreamStats's trackIdentifier] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-ondatachannel.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-ondatachannel.html.ini new file mode 100644 index 0000000000..c602e68241 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-ondatachannel.html.ini @@ -0,0 +1,17 @@ +[RTCPeerConnection-ondatachannel.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1613433 + expected: [OK, TIMEOUT] + [In-band negotiated channel created on remote peer should match the same configuration as local peer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1613433 + expected: [PASS, TIMEOUT] + + [In-band negotiated channel created on remote peer should match the same (default) configuration as local peer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1613433 + expected: [PASS, NOTRUN] + + [Open event should not be raised when sending and immediately closing the channel in the datachannel event] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1613433 + + [Negotiated channel should not fire datachannel event on remote peer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1613433 + expected: [PASS, NOTRUN] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-onicecandidateerror.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-onicecandidateerror.https.html.ini new file mode 100644 index 0000000000..81878a328c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-onicecandidateerror.https.html.ini @@ -0,0 +1,2 @@ +[RTCPeerConnection-onicecandidateerror.https.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1561441 diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini new file mode 100644 index 0000000000..5464a8db11 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini @@ -0,0 +1,6 @@ +[RTCPeerConnection-onnegotiationneeded.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Calling setStreams should cause negotiationneeded to fire] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1510802 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini new file mode 100644 index 0000000000..037ef0f292 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-onsignalingstatechanged.https.html] + expected: + if (os == "android") and fission: [TIMEOUT, OK] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-ontrack.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-ontrack.https.html.ini new file mode 100644 index 0000000000..8d045d50aa --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-ontrack.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-ontrack.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-operations.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-operations.https.html.ini new file mode 100644 index 0000000000..aad205ea8c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-operations.https.html.ini @@ -0,0 +1,6 @@ +[RTCPeerConnection-operations.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [sender.getStats does NOT use the operations chain] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1620689 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation-stress-glare-linear.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation-stress-glare-linear.https.html.ini new file mode 100644 index 0000000000..b6edfd4e41 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation-stress-glare-linear.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-perfect-negotiation-stress-glare-linear.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation-stress-glare.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation-stress-glare.https.html.ini new file mode 100644 index 0000000000..bad47d2298 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation-stress-glare.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-perfect-negotiation-stress-glare.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation.https.html.ini new file mode 100644 index 0000000000..e89f5f2feb --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-perfect-negotiation.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-perfect-negotiation.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-plan-b-is-not-supported.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-plan-b-is-not-supported.html.ini new file mode 100644 index 0000000000..9445133330 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-plan-b-is-not-supported.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-plan-b-is-not-supported.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-relay-canvas.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-relay-canvas.https.html.ini new file mode 100644 index 0000000000..65b6db550f --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-relay-canvas.https.html.ini @@ -0,0 +1,9 @@ +[RTCPeerConnection-relay-canvas.https.html] + disabled: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1728435 + [Two PeerConnections relaying a canvas source] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1730024 + expected: + if (os == "linux") and (processor == "x86_64") and swgl and not fission: [PASS, FAIL] + if (os == "mac") and not debug: [PASS, FAIL] + if (os == "linux") and (processor == "x86"): FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini new file mode 100644 index 0000000000..a25479f377 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini @@ -0,0 +1,12 @@ +[RTCPeerConnection-remote-track-mute.https.html] + prefs: [media.peerconnection.mute_on_bye_or_timeout:true] + expected: + if (os == "linux") and debug and not fission and swgl: [OK, TIMEOUT] + if (os == "linux") and debug and fission: [OK, TIMEOUT] + if (os == "android") and fission: [OK, TIMEOUT] + if (os == "linux") and not debug: [OK, TIMEOUT] + [pc.close() on one side causes mute events on the other] + expected: + if (os == "linux") and debug and not fission and swgl: [PASS, TIMEOUT] + if (os == "linux") and debug and fission: [PASS, TIMEOUT] + if (os == "linux") and not debug: [PASS, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-removeTrack.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-removeTrack.https.html.ini new file mode 100644 index 0000000000..80222e09e7 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-removeTrack.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-removeTrack.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-restartIce-onnegotiationneeded.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-restartIce-onnegotiationneeded.https.html.ini new file mode 100644 index 0000000000..a036e5216c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-restartIce-onnegotiationneeded.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-restartIce-onnegotiationneeded.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-restartIce.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-restartIce.https.html.ini new file mode 100644 index 0000000000..3783fb1f0c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-restartIce.https.html.ini @@ -0,0 +1,12 @@ +[RTCPeerConnection-restartIce.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [restartIce() survives remote offer containing partial restart] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1468993 + expected: FAIL + + [restartIce() survives remote offer containing partial restart (perfect negotiation)] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1468993 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setDescription-transceiver.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setDescription-transceiver.html.ini new file mode 100644 index 0000000000..36d3a7e2c9 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setDescription-transceiver.html.ini @@ -0,0 +1,6 @@ +[RTCPeerConnection-setDescription-transceiver.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [setRemoteDescription should set transceiver inactive if its corresponding m section is rejected] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728367 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini new file mode 100644 index 0000000000..6dce2e9b4e --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setLocalDescription-answer.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini new file mode 100644 index 0000000000..713305bf10 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setLocalDescription-offer.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-parameterless.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-parameterless.https.html.ini new file mode 100644 index 0000000000..c9bc3f8623 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-parameterless.https.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-setLocalDescription-parameterless.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Parameterless SLD() uses [[LastCreatedAnswer\]\] if it is still valid] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1055080 + expected: FAIL + + [Parameterless SLD() uses [[LastCreatedOffer\]\] if it is still valid] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1055080 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini new file mode 100644 index 0000000000..f7157156c1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setLocalDescription-pranswer.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1004510 + diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini new file mode 100644 index 0000000000..210ebb896a --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setLocalDescription-rollback.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription.html.ini new file mode 100644 index 0000000000..c0f61a2b26 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setLocalDescription.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-answer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-answer.html.ini new file mode 100644 index 0000000000..28f695a84f --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-answer.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setRemoteDescription-answer.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini new file mode 100644 index 0000000000..b37c93a298 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setRemoteDescription-nomsid.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini new file mode 100644 index 0000000000..967242ee5a --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini @@ -0,0 +1,11 @@ +[RTCPeerConnection-setRemoteDescription-offer.html] + expected: + if (os == "linux") and not debug and fission and (processor == "x86_64"): [OK, CRASH] + if (os == "android") and fission: [OK, TIMEOUT] + [setRemoteDescription(offer) with invalid SDP should reject with RTCError] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1527916 + expected: FAIL + + [setRemoteDescription(invalidOffer) from have-local-offer does not undo rollback] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1527916 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini new file mode 100644 index 0000000000..3a414305a2 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setRemoteDescription-pranswer.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1004510 + diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html.ini new file mode 100644 index 0000000000..a670939244 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setRemoteDescription-replaceTrack.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini new file mode 100644 index 0000000000..e483127d38 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini @@ -0,0 +1,14 @@ +[RTCPeerConnection-setRemoteDescription-rollback.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [explicit rollback of local offer should remove transceivers and transport] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL + + [rollback of a local offer to negotiated stable state should enable applying of a remote offer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL + + [rollback of a remote offer with stream changes] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1510802 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-simulcast.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-simulcast.https.html.ini new file mode 100644 index 0000000000..3f5ada9415 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-simulcast.https.html.ini @@ -0,0 +1,6 @@ +[RTCPeerConnection-setRemoteDescription-simulcast.https.html] + restart-after: + if (os == "win") and debug and (bits == 32): bug 1641974 + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini new file mode 100644 index 0000000000..31b2d788e1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini @@ -0,0 +1,5 @@ +[RTCPeerConnection-setRemoteDescription-tracks.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription.html.ini new file mode 100644 index 0000000000..cd3fbcb695 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setRemoteDescription.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-transceivers.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-transceivers.https.html.ini new file mode 100644 index 0000000000..a8b71b261f --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-transceivers.https.html.ini @@ -0,0 +1,7 @@ +[RTCPeerConnection-transceivers.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Closing the PC stops the transceivers] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-transport-stats.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-transport-stats.https.html.ini new file mode 100644 index 0000000000..80281f56ae --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-transport-stats.https.html.ini @@ -0,0 +1,6 @@ +[RTCPeerConnection-transport-stats.https.html] + [DTLS statistics on transport-stats after setLocalDescription] + expected: FAIL + + [ICE statistics on transport-stats after setLocalDescription] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnection-videoDetectorTest.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-videoDetectorTest.html.ini new file mode 100644 index 0000000000..0b9b2e2dc6 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-videoDetectorTest.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-videoDetectorTest.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1526207 + expected: + if (os == "android") and release_or_beta: OK + if os == "android": TIMEOUT + [Signal detector detects track change within reasonable time] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1526207 + expected: + if (os == "android") and release_or_beta: PASS + if os == "android": TIMEOUT diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnectionIceErrorEvent.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnectionIceErrorEvent.html.ini new file mode 100644 index 0000000000..1c02072b31 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnectionIceErrorEvent.html.ini @@ -0,0 +1,5 @@ +[RTCPeerConnectionIceErrorEvent.html] + [RTCPeerConnectionIceErrorEvent constructed from init parameters] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728335 + expected: FAIL + diff --git a/testing/web-platform/meta/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini new file mode 100644 index 0000000000..56ee8f056e --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini @@ -0,0 +1,9 @@ +[RTCPeerConnectionIceEvent-constructor.html] + [RTCPeerConnectionIceEvent with no eventInitDict (default)] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531911 + + [RTCPeerConnectionIceEvent with empty object as eventInitDict (default)] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531911 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpParameters-codecs.html.ini b/testing/web-platform/meta/webrtc/RTCRtpParameters-codecs.html.ini new file mode 100644 index 0000000000..d9906f9583 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpParameters-codecs.html.ini @@ -0,0 +1,3 @@ +[RTCRtpParameters-codecs.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1531458 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpParameters-encodings.html.ini b/testing/web-platform/meta/webrtc/RTCRtpParameters-encodings.html.ini new file mode 100644 index 0000000000..bbe6fec2dd --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpParameters-encodings.html.ini @@ -0,0 +1,3 @@ +[RTCRtpParameters-encodings.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCRtpParameters-headerExtensions.html.ini b/testing/web-platform/meta/webrtc/RTCRtpParameters-headerExtensions.html.ini new file mode 100644 index 0000000000..11de88b591 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpParameters-headerExtensions.html.ini @@ -0,0 +1,3 @@ +[RTCRtpParameters-headerExtensions.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1531458 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpParameters-rtcp.html.ini b/testing/web-platform/meta/webrtc/RTCRtpParameters-rtcp.html.ini new file mode 100644 index 0000000000..dc458b4c83 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpParameters-rtcp.html.ini @@ -0,0 +1,3 @@ +[RTCRtpParameters-rtcp.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1531458 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-getCapabilities.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getCapabilities.html.ini new file mode 100644 index 0000000000..27d1343b44 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getCapabilities.html.ini @@ -0,0 +1,3 @@ +[RTCRtpReceiver-getCapabilities.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1531460 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini new file mode 100644 index 0000000000..fd7c25ec70 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini @@ -0,0 +1,4 @@ +[RTCRtpReceiver-getContributingSources.https.html] + expected: + if (os == "linux") and not debug and fission and (processor == "x86_64"): [OK, CRASH] + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-getParameters.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getParameters.html.ini new file mode 100644 index 0000000000..398ae39f2a --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getParameters.html.ini @@ -0,0 +1,3 @@ +[RTCRtpReceiver-getParameters.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1531464 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-getStats.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getStats.https.html.ini new file mode 100644 index 0000000000..962c23bb2c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getStats.https.html.ini @@ -0,0 +1,22 @@ +[RTCRtpReceiver-getStats.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [receiver.getStats() via addTransceiver should return stats report containing inbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225720 + expected: FAIL + + [receiver.getStats() via addTrack should return stats report containing inbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225720 + expected: FAIL + + [receiver.getStats() should work on a stopped transceiver] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1056433 + expected: + if (os == "linux") and debug and not swgl and fission: [PASS, FAIL] + [FAIL, PASS] + + [receiver.getStats() should work with a closed PeerConnection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1056433 + expected: + if (os == "linux") and debug and not swgl and fission: [PASS, FAIL] + [FAIL, PASS] diff --git a/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini new file mode 100644 index 0000000000..24607ca70c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini @@ -0,0 +1,7 @@ +[RTCRtpReceiver-getSynchronizationSources.https.html] + expected: + if (os == "linux") and not debug and fission and (processor == "x86_64"): [OK, CRASH] + if (os == "android") and fission: [OK, TIMEOUT] + [[audio-only\] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1525394 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender-encode-same-track-twice.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender-encode-same-track-twice.https.html.ini new file mode 100644 index 0000000000..d9798e1b48 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-encode-same-track-twice.https.html.ini @@ -0,0 +1,3 @@ +[RTCRtpSender-encode-same-track-twice.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT, CRASH] diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender-getCapabilities.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender-getCapabilities.html.ini new file mode 100644 index 0000000000..9043863c72 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-getCapabilities.html.ini @@ -0,0 +1,3 @@ +[RTCRtpSender-getCapabilities.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1531460 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender-getStats.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender-getStats.https.html.ini new file mode 100644 index 0000000000..07f3589d6e --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-getStats.https.html.ini @@ -0,0 +1,10 @@ +[RTCRtpSender-getStats.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [sender.getStats() via addTransceiver should return stats report containing outbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225720 + expected: FAIL + + [sender.getStats() via addTrack should return stats report containing outbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225720 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender-replaceTrack.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender-replaceTrack.https.html.ini new file mode 100644 index 0000000000..0d352e212f --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-replaceTrack.https.html.ini @@ -0,0 +1,15 @@ +[RTCRtpSender-replaceTrack.https.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1526207 + expected: + if (os == "android") and release_or_beta: OK + if os == "android": TIMEOUT + [ReplaceTrack transmits the new track not the old track] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1526207 + expected: + if (os == "android") and release_or_beta: PASS + if os == "android": TIMEOUT + [ReplaceTrack null -> new track transmits the new track] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1526207 + expected: + if (os == "android") and release_or_beta: PASS + if os == "android": NOTRUN diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender-setParameters.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender-setParameters.html.ini new file mode 100644 index 0000000000..8d2e0b157c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-setParameters.html.ini @@ -0,0 +1,5 @@ +[RTCRtpSender-setParameters.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [setParameters() with already used parameters should reject with InvalidStateError] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender-setStreams.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender-setStreams.https.html.ini new file mode 100644 index 0000000000..775a42ef3f --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-setStreams.https.html.ini @@ -0,0 +1,2 @@ +[RTCRtpSender-setStreams.https.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1510802 diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender-transport.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender-transport.https.html.ini new file mode 100644 index 0000000000..c345c4a20e --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-transport.https.html.ini @@ -0,0 +1,3 @@ +[RTCRtpSender-transport.https.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender.https.html.ini new file mode 100644 index 0000000000..f10874049a --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender.https.html.ini @@ -0,0 +1,3 @@ +[RTCRtpSender.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCRtpTransceiver-direction.html.ini b/testing/web-platform/meta/webrtc/RTCRtpTransceiver-direction.html.ini new file mode 100644 index 0000000000..a33d9a362b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpTransceiver-direction.html.ini @@ -0,0 +1,3 @@ +[RTCRtpTransceiver-direction.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini b/testing/web-platform/meta/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini new file mode 100644 index 0000000000..d78f524c4b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini @@ -0,0 +1,3 @@ +[RTCRtpTransceiver-setCodecPreferences.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1396922 + diff --git a/testing/web-platform/meta/webrtc/RTCRtpTransceiver-stop.html.ini b/testing/web-platform/meta/webrtc/RTCRtpTransceiver-stop.html.ini new file mode 100644 index 0000000000..1a998de790 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpTransceiver-stop.html.ini @@ -0,0 +1,10 @@ +[RTCRtpTransceiver-stop.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [If a transceiver is stopped, transceivers should end up in state stopped] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [If a transceiver is stopped, transceivers, senders and receivers should disappear after offer/answer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCRtpTransceiver-stopping.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpTransceiver-stopping.https.html.ini new file mode 100644 index 0000000000..16ae84fda4 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpTransceiver-stopping.https.html.ini @@ -0,0 +1,32 @@ +[RTCRtpTransceiver-stopping.https.html] + expected: + if (os == "android") and fission: [TIMEOUT, OK] + [[audio\] Locally stopped transceiver goes from stopping to stopped] + expected: FAIL + + [[audio\] Remotely stopping a transceiver ends the track] + expected: FAIL + + [[audio\] Remotely stopped transceiver goes directly to stopped] + expected: FAIL + + [[audio\] Rollback when removing transceiver does end the track] + expected: FAIL + + [[audio\] Rollback when removing transceiver makes it stopped] + expected: FAIL + + [[video\] Locally stopped transceiver goes from stopping to stopped] + expected: FAIL + + [[video\] Remotely stopping a transceiver ends the track] + expected: FAIL + + [[video\] Remotely stopped transceiver goes directly to stopped] + expected: FAIL + + [[video\] Rollback when removing transceiver does end the track] + expected: FAIL + + [[video\] Rollback when removing transceiver makes it stopped] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCRtpTransceiver.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpTransceiver.https.html.ini new file mode 100644 index 0000000000..4fd4d434fb --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpTransceiver.https.html.ini @@ -0,0 +1,45 @@ +[RTCRtpTransceiver.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 + expected: + if (os == "android") and fission: [OK, CRASH, TIMEOUT] + [checkStop] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [checkStopAfterCreateOffer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [checkStopAfterSetLocalOffer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [checkStopAfterSetRemoteOffer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [checkStopAfterCreateAnswer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [checkStopAfterSetLocalAnswer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [checkLocalRollback] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [checkRemoteRollback] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296 + expected: FAIL + + [checkCurrentDirection] + expected: FAIL + + [checkAddTransceiverThenAddTrackPairs] + expected: FAIL + + [checkMsectionReuse] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/RTCSctpTransport-constructor.html.ini b/testing/web-platform/meta/webrtc/RTCSctpTransport-constructor.html.ini new file mode 100644 index 0000000000..207959b3ae --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCSctpTransport-constructor.html.ini @@ -0,0 +1,3 @@ +[RTCSctpTransport-constructor.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + diff --git a/testing/web-platform/meta/webrtc/RTCSctpTransport-events.html.ini b/testing/web-platform/meta/webrtc/RTCSctpTransport-events.html.ini new file mode 100644 index 0000000000..6b15559a47 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCSctpTransport-events.html.ini @@ -0,0 +1,3 @@ +[RTCSctpTransport-events.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + diff --git a/testing/web-platform/meta/webrtc/RTCSctpTransport-maxChannels.html.ini b/testing/web-platform/meta/webrtc/RTCSctpTransport-maxChannels.html.ini new file mode 100644 index 0000000000..a62a5ad259 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCSctpTransport-maxChannels.html.ini @@ -0,0 +1,3 @@ +[RTCSctpTransport-maxChannels.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + diff --git a/testing/web-platform/meta/webrtc/RTCSctpTransport-maxMessageSize.html.ini b/testing/web-platform/meta/webrtc/RTCSctpTransport-maxMessageSize.html.ini new file mode 100644 index 0000000000..a3c32e1e3c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCSctpTransport-maxMessageSize.html.ini @@ -0,0 +1,3 @@ +[RTCSctpTransport-maxMessageSize.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + diff --git a/testing/web-platform/meta/webrtc/RTCTrackEvent-constructor.html.ini b/testing/web-platform/meta/webrtc/RTCTrackEvent-constructor.html.ini new file mode 100644 index 0000000000..8554879345 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCTrackEvent-constructor.html.ini @@ -0,0 +1,3 @@ +[RTCTrackEvent-constructor.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RTCTrackEvent-fire.html.ini b/testing/web-platform/meta/webrtc/RTCTrackEvent-fire.html.ini new file mode 100644 index 0000000000..0af431ddcf --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCTrackEvent-fire.html.ini @@ -0,0 +1,4 @@ +[RTCTrackEvent-fire.html] + prefs: [media.peerconnection.sdp.alternate_parse_mode:never, media.peerconnection.sdp.parser:sipcc] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/RollbackEvents.https.html.ini b/testing/web-platform/meta/webrtc/RollbackEvents.https.html.ini new file mode 100644 index 0000000000..82af8e6d4f --- /dev/null +++ b/testing/web-platform/meta/webrtc/RollbackEvents.https.html.ini @@ -0,0 +1,26 @@ +[RollbackEvents.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [[audio\] Track with stream: removal due to disassociation in rollback and then add it back again] + expected: FAIL + + [[audio\] Track without stream: removal due to disassociation in rollback and then add it back] + expected: FAIL + + [[audio\] Track with stream: removal due to direction changing and then add back using rollback] + expected: FAIL + + [[audio\] Track without stream: removal due to direction changing and then add back using rollback] + expected: FAIL + + [[video\] Track with stream: removal due to disassociation in rollback and then add it back again] + expected: FAIL + + [[video\] Track without stream: removal due to disassociation in rollback and then add it back] + expected: FAIL + + [[video\] Track with stream: removal due to direction changing and then add back using rollback] + expected: FAIL + + [[video\] Track without stream: removal due to direction changing and then add back using rollback] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/__dir__.ini b/testing/web-platform/meta/webrtc/__dir__.ini new file mode 100644 index 0000000000..a2c84b5e8c --- /dev/null +++ b/testing/web-platform/meta/webrtc/__dir__.ini @@ -0,0 +1,3 @@ +prefs: [media.navigator.permission.disabled:true, media.navigator.streams.fake:true, privacy.resistFingerprinting.reduceTimerPrecision.jitter:false, privacy.reduceTimerPrecision:false, media.peerconnection.ice.trickle_grace_period:10000, media.peerconnection.ice.obfuscate_host_addresses:false, media.peerconnection.allow_old_setParameters:false] +lsan-allowed: [Alloc, MakeAndAddRef, Malloc, NS_NewDOMDataChannel, NS_NewRunnableFunction, PR_Realloc, ParentContentActorCreateFunc, allocate, mozilla::DataChannelConnection::Create, mozilla::DataChannelConnection::HandleOpenRequestMessage, mozilla::DataChannelConnection::Open, mozilla::MediaPacket::Copy, mozilla::MediaPipeline::MediaPipeline, mozilla::WeakPtr, mozilla::dom::DocGroup::Create, mozilla::dom::DocGroup::DocGroup, mozilla::runnable_args_func, nsRefPtrDeque, sctp_add_vtag_to_timewait, sctp_hashinit_flags] +leak-threshold: [default:3020800, rdd:51200, tab:51200] diff --git a/testing/web-platform/meta/webrtc/getstats.html.ini b/testing/web-platform/meta/webrtc/getstats.html.ini new file mode 100644 index 0000000000..b490c5f1d8 --- /dev/null +++ b/testing/web-platform/meta/webrtc/getstats.html.ini @@ -0,0 +1,7 @@ +[getstats.html] + expected: + if (os == "linux") and not debug and fission and (processor == "x86_64"): [OK, CRASH] + if (os == "android") and fission: [OK, TIMEOUT] + [Can get stats from a basic WebRTC call.] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531087 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/historical.html.ini b/testing/web-platform/meta/webrtc/historical.html.ini new file mode 100644 index 0000000000..8cefe2c852 --- /dev/null +++ b/testing/web-platform/meta/webrtc/historical.html.ini @@ -0,0 +1,34 @@ +[historical.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [RTCDataChannel member reliable should not exist] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1285683 + expected: FAIL + + [RTCPeerConnection member addStream should not exist] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531808 + expected: FAIL + + [RTCPeerConnection member getLocalStreams should not exist] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531810 + expected: FAIL + + [RTCPeerConnection member getRemoteStreams should not exist] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531810 + expected: FAIL + + [RTCPeerConnection member onaddstream should not exist] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1241291 + expected: FAIL + + [mozRTCIceCandidate interface should not exist] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531812 + expected: FAIL + + [mozRTCPeerConnection interface should not exist] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531812 + expected: FAIL + + [mozRTCSessionDescription interface should not exist] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531812 + 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 new file mode 100644 index 0000000000..7859065251 --- /dev/null +++ b/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini @@ -0,0 +1,485 @@ +[idlharness.https.window.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "protocol" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "foundation" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relatedAddress" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute tcpType] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "type" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute candidate] + expected: FAIL + + [RTCIceCandidate interface: attribute priority] + expected: FAIL + + [RTCIceCandidate interface: attribute foundation] + expected: FAIL + + [RTCIceCandidate interface: attribute port] + expected: FAIL + + [RTCPeerConnection interface: attribute onicecandidateerror] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relatedPort" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "tcpType" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute usernameFragment] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "component" with the proper type] + expected: FAIL + + [RTCSessionDescription interface: attribute type] + expected: FAIL + + [RTCIceCandidate interface: attribute sdpMLineIndex] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onconnectionstatechange" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute protocol] + expected: FAIL + + [RTCIceCandidate interface: attribute component] + expected: FAIL + + [Test driver for asyncInitTransports] + expected: FAIL + + [RTCIceCandidate interface: attribute relatedPort] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "sctp" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onicecandidateerror" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "port" with the proper type] + expected: FAIL + + [RTCSessionDescription interface: attribute sdp] + expected: FAIL + + [RTCPeerConnection interface: attribute sctp] + expected: FAIL + + [RTCIceCandidate interface: attribute sdpMid] + expected: FAIL + + [RTCPeerConnection interface: attribute onconnectionstatechange] + expected: FAIL + + [RTCIceCandidate interface: attribute relatedAddress] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "priority" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "connectionState" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: attribute connectionState] + expected: FAIL + + [RTCSctpTransport interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCIceTransport interface: operation getSelectedCandidatePair()] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCDtlsTransport must be primary interface of idlTestObjects.dtlsTransport] + expected: FAIL + + [RTCErrorEvent must be primary interface of new RTCErrorEvent('error')] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "onstatechange" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "ongatheringstatechange" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCErrorEvent interface: new RTCErrorEvent('error') must inherit property "error" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute errorText] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "onstatechange" with the proper type] + expected: FAIL + + [RTCDTMFSender interface: attribute canInsertDTMF] + expected: FAIL + + [RTCRtpReceiver interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').receiver with too few arguments must throw TypeError] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "component" with the proper type] + expected: FAIL + + [RTCIceTransport interface object length] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getRemoteParameters()" with the proper type] + expected: FAIL + + [RTCIceTransport interface: attribute gatheringState] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCRtpReceiver interface: operation getCapabilities(DOMString)] + expected: FAIL + + [RTCSctpTransport interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "onselectedcandidatepairchange" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface object name] + expected: FAIL + + [RTCIceTransport must be primary interface of idlTestObjects.iceTransport] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: new RTCPeerConnectionIceEvent('ice') must inherit property "url" with the proper type] + expected: FAIL + + [RTCErrorEvent interface object length] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface object length] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "maxChannels" with the proper type] + expected: FAIL + + [RTCSctpTransport interface: attribute transport] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "getRemoteCertificates()" with the proper type] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "state" with the proper type] + expected: FAIL + + [RTCRtpSender interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError] + expected: FAIL + + [RTCRtpSender interface: operation getCapabilities(DOMString)] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getLocalParameters()" with the proper type] + expected: FAIL + + [RTCSctpTransport interface: attribute onstatechange] + expected: FAIL + + [RTCIceTransport interface: attribute state] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "onstatechange" with the proper type] + expected: FAIL + + [RTCSctpTransport interface object name] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCSctpTransport interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [Stringification of idlTestObjects.dtlsTransport] + expected: FAIL + + [RTCCertificate interface: idlTestObjects.certificate must inherit property "getFingerprints()" with the proper type] + expected: FAIL + + [RTCSctpTransport interface: existence and properties of interface object] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getSelectedCandidatePair()" with the proper type] + expected: FAIL + + [RTCErrorEvent interface: attribute error] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [Stringification of new RTCErrorEvent('error')] + expected: FAIL + + [RTCCertificate interface: operation getFingerprints()] + expected: FAIL + + [RTCSctpTransport interface: attribute state] + expected: FAIL + + [RTCSctpTransport interface: attribute maxChannels] + expected: FAIL + + [RTCIceTransport interface: operation getLocalParameters()] + expected: FAIL + + [RTCIceTransport interface: attribute ongatheringstatechange] + expected: FAIL + + [RTCSctpTransport interface: attribute maxMessageSize] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getParameters()" with the proper type] + expected: FAIL + + [RTCRtpReceiver interface: operation getParameters()] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCIceTransport interface: attribute onselectedcandidatepairchange] + expected: FAIL + + [RTCIceTransport interface: attribute onstatechange] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "role" with the proper type] + expected: FAIL + + [RTCIceTransport interface: operation getRemoteParameters()] + expected: FAIL + + [Stringification of idlTestObjects.iceTransport] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getLocalCandidates()" with the proper type] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getRemoteCandidates()" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute errorCode] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "gatheringState" with the proper type] + expected: FAIL + + [RTCErrorEvent interface object name] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "onerror" with the proper type] + expected: FAIL + + [RTCIceTransport interface object name] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: attribute url] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface object] + expected: FAIL + + [RTCIceTransport interface: attribute role] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCIceTransport interface: attribute component] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "state" with the proper type] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "state" with the proper type] + expected: FAIL + + [RTCDtlsTransport interface: operation getRemoteCertificates()] + expected: FAIL + + [RTCIceTransport interface: operation getRemoteCandidates()] + expected: FAIL + + [RTCSctpTransport must be primary interface of idlTestObjects.sctpTransport] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "maxMessageSize" with the proper type] + expected: FAIL + + [Stringification of idlTestObjects.sctpTransport] + expected: FAIL + + [RTCIceTransport interface: operation getLocalCandidates()] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute url] + expected: FAIL + + [RTCSctpTransport interface object length] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "transport" with the proper type] + expected: FAIL + + [RTCDtlsTransport interface: attribute onerror] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "address" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute address] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "iceTransport" with the proper type] + expected: FAIL + + [RTCDtlsTransport interface: attribute iceTransport] + expected: FAIL + + [RTCError interface: attribute sentAlert] + expected: FAIL + + [RTCError interface object name] + expected: FAIL + + [RTCError interface object length] + expected: FAIL + + [RTCError interface: attribute errorDetail] + expected: FAIL + + [RTCError interface: attribute sctpCauseCode] + expected: FAIL + + [RTCError interface: attribute sdpLineNumber] + expected: FAIL + + [RTCError interface: attribute receivedAlert] + expected: FAIL + + [RTCError interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCError interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCError interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCError interface: existence and properties of interface object] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "onclosing" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute onclosing] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute address] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute port] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "errorText" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "port" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "url" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent must be primary interface of new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 });] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "errorCode" with the proper type] + expected: FAIL + + [Stringification of new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 });] + expected: FAIL + + [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>)] + expected: FAIL + + [RTCRtpSender interface: operation setStreams(MediaStream...)] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "setStreams(MediaStream...)" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "setCodecPreferences(sequence<RTCRtpCodecCapability>)" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface: calling setCodecPreferences(sequence<RTCRtpCodecCapability>) on new RTCPeerConnection().addTransceiver('audio') with too few arguments must throw TypeError] + expected: FAIL + + [RTCRtpSender interface: calling setStreams(MediaStream...) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError] + expected: FAIL + + [RTCPeerConnection interface: operation setRemoteDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)] + expected: FAIL + + [RTCPeerConnection interface: operation setRemoteDescription(RTCSessionDescriptionInit)] + expected: FAIL + + [RTCSessionDescription interface object length] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "binaryType" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute relayProtocol] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relayProtocol" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute url] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "url" with the proper type] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini b/testing/web-platform/meta/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini new file mode 100644 index 0000000000..96c0421c9f --- /dev/null +++ b/testing/web-platform/meta/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-createOffer-offerToReceive.html] + expected: + if (os == "linux") and not fission and not debug: [OK, CRASH] + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini b/testing/web-platform/meta/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini new file mode 100644 index 0000000000..fd4f576bad --- /dev/null +++ b/testing/web-platform/meta/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini @@ -0,0 +1,3 @@ +[RTCRtpTransceiver-with-OfferToReceive-options.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/legacy/__dir__.ini b/testing/web-platform/meta/webrtc/legacy/__dir__.ini new file mode 100644 index 0000000000..70e26bcb8f --- /dev/null +++ b/testing/web-platform/meta/webrtc/legacy/__dir__.ini @@ -0,0 +1 @@ +lsan-allowed: [NewSegment, mozilla::layers::BufferTextureData::CreateInternal] diff --git a/testing/web-platform/meta/webrtc/legacy/onaddstream.https.html.ini b/testing/web-platform/meta/webrtc/legacy/onaddstream.https.html.ini new file mode 100644 index 0000000000..a71eb3422a --- /dev/null +++ b/testing/web-platform/meta/webrtc/legacy/onaddstream.https.html.ini @@ -0,0 +1,3 @@ +[onaddstream.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/no-media-call.html.ini b/testing/web-platform/meta/webrtc/no-media-call.html.ini new file mode 100644 index 0000000000..d6a4db1ba4 --- /dev/null +++ b/testing/web-platform/meta/webrtc/no-media-call.html.ini @@ -0,0 +1,5 @@ +[no-media-call.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [RTCPeerConnection No-Media Connection Test] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/promises-call.html.ini b/testing/web-platform/meta/webrtc/promises-call.html.ini new file mode 100644 index 0000000000..6afa3a954f --- /dev/null +++ b/testing/web-platform/meta/webrtc/promises-call.html.ini @@ -0,0 +1,3 @@ +[promises-call.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/RTCPeerConnection-payloadTypes.html.ini b/testing/web-platform/meta/webrtc/protocol/RTCPeerConnection-payloadTypes.html.ini new file mode 100644 index 0000000000..f63530850c --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/RTCPeerConnection-payloadTypes.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-payloadTypes.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/__dir__.ini b/testing/web-platform/meta/webrtc/protocol/__dir__.ini new file mode 100644 index 0000000000..c6a51b9705 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/__dir__.ini @@ -0,0 +1,2 @@ +lsan-allowed: [Create, CreateNullDecoderModule, IPC::Channel::Channel, MakeRefPtr, MakeUnique, NewPage, NewSegment, PLDHashTable::Add, PLDHashTable::ChangeTable, PLDHashTable::MakeEntryHandle, Realloc, RevocableStore::RevocableStore, allocate, already_AddRefed, maybe_pod_malloc, mozilla::FFmpegDecoderModule, mozilla::KnowsCompositorVideo::TryCreateForIdentifier, mozilla::detail::UniqueSelector, mozilla::ipc::IProtocol::ActorConnected, mozilla::ipc::MessageChannel::Open, mozilla::layers::BufferTextureData::CreateInternal, mozilla::layers::ImageContainer::CreatePlanarYCbCrImage, mozilla::layers::ImageContainer::EnsureRecycleAllocatorForRDD, mozilla::layers::TextureClient::CreateIPDLActor, mozilla::layers::TextureClientRecycleAllocator::CreateOrRecycle, mozilla::layers::VideoBridgeChild::Open, sctp_add_vtag_to_timewait, sctp_hashinit_flags] +leak-threshold: [default:3020800, rdd:51200] diff --git a/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini b/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini new file mode 100644 index 0000000000..9c79e38d51 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini @@ -0,0 +1,13 @@ +[bundle.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [not negotiating BUNDLE creates two separate ice and dtls transports] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL + + [bundles on the first transport and closes the second] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1307996 + expected: FAIL + + [max-bundle with an offer without bundle only negotiates the first m-line] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini b/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini new file mode 100644 index 0000000000..c29b1c9c1f --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini @@ -0,0 +1,19 @@ +[candidate-exchange.https.html] + expected: + if (os == "linux") and not debug and fission: [OK, CRASH] + if (os == "android") and fission: [OK, TIMEOUT] + [Adding only caller -> callee candidates gives a connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [Adding only callee -> caller candidates gives a connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [Explicit offer/answer exchange gives a connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [Adding callee -> caller candidates from end-of-candidates gives a connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini b/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini new file mode 100644 index 0000000000..51d47dc6ce --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini @@ -0,0 +1,34 @@ +[crypto-suite.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [srtpCipher is acceptable on video-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [srtpCipher is acceptable on data-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [tlsGroup is acceptable on video-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [tlsGroup is acceptable on data-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [dtlsCipher is acceptable on video-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [dtlsCipher is acceptable on data-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [tlsVersion is acceptable on video-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [tlsVersion is acceptable on data-only] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini b/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini new file mode 100644 index 0000000000..e5a1ec8db9 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini @@ -0,0 +1,6 @@ +[dtls-fingerprint-validation.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: TIMEOUT + [Connection fails if one side provides a wrong DTLS fingerprint] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1635922 + expected: TIMEOUT diff --git a/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini b/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini new file mode 100644 index 0000000000..b399895d1b --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini @@ -0,0 +1,14 @@ +[dtls-setup.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [PC with setup=actpass should have a dtlsRole of client] + expected: FAIL + + [PC with setup=active should have a dtlsRole of server] + expected: FAIL + + [PC with setup=passive should have a dtlsRole of client] + expected: FAIL + + [dtlsRole is `unknown` before negotiation of the DTLS handshake] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini b/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini new file mode 100644 index 0000000000..1dbd03b13c --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini @@ -0,0 +1,57 @@ +[h264-profile-levels.https.html] + [Level 1 H264 video is appropriately constrained] + expected: FAIL + + [Level 2 H264 video is appropriately constrained] + expected: FAIL + + [Level 3 H264 video is appropriately constrained] + expected: FAIL + + [Level 4 H264 video is appropriately constrained] + expected: FAIL + + [Level 5 H264 video is appropriately constrained] + expected: FAIL + + [Level 6 H264 video is appropriately constrained] + expected: FAIL + + [Level 1.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 1.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 1.3 H264 video is appropriately constrained] + expected: FAIL + + [Level 2.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 2.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 3.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 3.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 4.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 4.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 5.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 5.2 H264 video is appropriately constrained] + expected: FAIL + + [Level 6.1 H264 video is appropriately constrained] + expected: FAIL + + [Level 6.2 H264 video is appropriately constrained] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini b/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini new file mode 100644 index 0000000000..840f90f7e7 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini @@ -0,0 +1,3 @@ +[handover-datachannel.html] + [Handover with datachannel reinitiated from new callee completes] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/handover.html.ini b/testing/web-platform/meta/webrtc/protocol/handover.html.ini new file mode 100644 index 0000000000..3f168c5b49 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/handover.html.ini @@ -0,0 +1,8 @@ +[handover.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Negotiation of handover initiated at callee works] + expected: FAIL + + [Negotiation of handover initiated at caller works] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini b/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini new file mode 100644 index 0000000000..8801b5c5f1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini @@ -0,0 +1,6 @@ +[ice-state.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [PC should enter disconnected state when a failing candidate is sent] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1557053 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini b/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini new file mode 100644 index 0000000000..09560fcfec --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini @@ -0,0 +1,10 @@ +[ice-ufragpwd.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [setRemoteDescription with a ice-ufrag containing a non-ice-char fails] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1617686 + expected: FAIL + + [setRemoteDescription with a ice-pwd containing a non-ice-char fails] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1617686 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/jsep-initial-offer.https.html.ini b/testing/web-platform/meta/webrtc/protocol/jsep-initial-offer.https.html.ini new file mode 100644 index 0000000000..19cdc7b1e2 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/jsep-initial-offer.https.html.ini @@ -0,0 +1,3 @@ +[jsep-initial-offer.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/missing-fields.html.ini b/testing/web-platform/meta/webrtc/protocol/missing-fields.html.ini new file mode 100644 index 0000000000..4a96a2feea --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/missing-fields.html.ini @@ -0,0 +1,3 @@ +[missing-fields.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/msid-generate.html.ini b/testing/web-platform/meta/webrtc/protocol/msid-generate.html.ini new file mode 100644 index 0000000000..099a2d0723 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/msid-generate.html.ini @@ -0,0 +1,29 @@ +[msid-generate.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [AddTrack with a stream produces MSID with a stream ID] + expected: FAIL + + [AddTrack with two streams produces two MSID lines] + expected: FAIL + + [AddTrack with the stream twice produces single MSID with a stream ID] + expected: FAIL + + [AddTransceiver with a stream produces MSID with a stream ID] + expected: FAIL + + [AddTransceiver with two streams produces two MSID lines] + expected: FAIL + + [AddTransceiver with the stream twice produces single MSID with a stream ID] + expected: FAIL + + [SetStreams with a stream produces MSID with a stream ID] + expected: FAIL + + [SetStreams with two streams produces two MSID lines] + expected: FAIL + + [SetStreams with the stream twice produces single MSID with a stream ID] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/msid-parse.html.ini b/testing/web-platform/meta/webrtc/protocol/msid-parse.html.ini new file mode 100644 index 0000000000..a5976d5758 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/msid-parse.html.ini @@ -0,0 +1,3 @@ +[msid-parse.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini new file mode 100644 index 0000000000..4b058cbe13 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini @@ -0,0 +1,4 @@ +[rtp-clockrate.html] + expected: TIMEOUT + [video rtp timestamps increase by approximately 90000 per second] + expected: TIMEOUT diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini new file mode 100644 index 0000000000..b42afbaaa6 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini @@ -0,0 +1,11 @@ +[rtp-demuxing.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + if (os == "mac") and not debug: [OK, TIMEOUT] + [Can demux two video tracks with different payload types on a bundled connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531460 + expected: FAIL + + [Can demux two video tracks with the same payload type on an unbundled connection] + expected: + if (os == "mac") and not debug: [PASS, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini new file mode 100644 index 0000000000..5c77f6e741 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini @@ -0,0 +1,11 @@ +[rtp-extension-support.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [RTP header extension urn:3gpp:video-orientation is present in offer] + expected: FAIL + + [RTP header extension urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id is present in offer] + expected: FAIL + + [RTP header extension urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id is present in offer] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini new file mode 100644 index 0000000000..277eb7a7f6 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini @@ -0,0 +1,5 @@ +[rtp-payloadtypes.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [setRemoteDescription with a codec in the range 64-95 throws an InvalidAccessError] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/rtx-codecs.https.html.ini b/testing/web-platform/meta/webrtc/protocol/rtx-codecs.https.html.ini new file mode 100644 index 0000000000..634ed07ca7 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtx-codecs.https.html.ini @@ -0,0 +1,3 @@ +[rtx-codecs.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/sctp-format.html.ini b/testing/web-platform/meta/webrtc/protocol/sctp-format.html.ini new file mode 100644 index 0000000000..d6d1b12461 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/sctp-format.html.ini @@ -0,0 +1,4 @@ +[sctp-format.html] + max-asserts: 3 + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/sdes-dont-dont-dont.html.ini b/testing/web-platform/meta/webrtc/protocol/sdes-dont-dont-dont.html.ini new file mode 100644 index 0000000000..87ec9989b9 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/sdes-dont-dont-dont.html.ini @@ -0,0 +1,9 @@ +[sdes-dont-dont-dont.html] + expected: + if (os == "win") and debug and (processor == "x86_64") and not swgl: OK + if (os == "android") and debug and not fission: OK + if (os == "android") and debug and fission: [OK, TIMEOUT] + if (os == "win") and not debug: OK + if os == "mac": OK + [OK, ERROR] + max-asserts: 3 diff --git a/testing/web-platform/meta/webrtc/protocol/simulcast-answer.html.ini b/testing/web-platform/meta/webrtc/protocol/simulcast-answer.html.ini new file mode 100644 index 0000000000..a29b91f67c --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/simulcast-answer.html.ini @@ -0,0 +1,6 @@ +[simulcast-answer.html] + max-asserts: 3 + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Using the ~rid SDP syntax in a remote offer does not control the local encodings active flag] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini b/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini new file mode 100644 index 0000000000..1422fe0bc8 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/simulcast-offer.html.ini @@ -0,0 +1,3 @@ +[simulcast-offer.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/protocol/split.https.html.ini b/testing/web-platform/meta/webrtc/protocol/split.https.html.ini new file mode 100644 index 0000000000..4c5f3695ca --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/split.https.html.ini @@ -0,0 +1,3 @@ +[split.https.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1643050 + diff --git a/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini b/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini new file mode 100644 index 0000000000..2e54d190b9 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini @@ -0,0 +1,5 @@ +[unknown-mediatypes.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [Unknown media types are rejected with the port set to 0] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini b/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini new file mode 100644 index 0000000000..c14dffd01b --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini @@ -0,0 +1,11 @@ +[video-codecs.https.html] + max-asserts: 3 + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [H.264 and VP8 should be supported in initial offer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1534688 + expected: FAIL + + [H.264 and VP8 should be negotiated after handshake] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1534687 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini b/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini new file mode 100644 index 0000000000..e59f7d68cf --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini @@ -0,0 +1,6 @@ +[vp8-fmtp.html] + expected: + if (os == "win") and debug: [OK, TIMEOUT] + if (os == "android") and fission: [OK, TIMEOUT] + [setRemoteDescription parses max-fr and max-fs fmtp parameters] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/receiver-track-live.https.html.ini b/testing/web-platform/meta/webrtc/receiver-track-live.https.html.ini new file mode 100644 index 0000000000..37eeb63057 --- /dev/null +++ b/testing/web-platform/meta/webrtc/receiver-track-live.https.html.ini @@ -0,0 +1,3 @@ +[receiver-track-live.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/recvonly-transceiver-can-become-sendrecv.https.html.ini b/testing/web-platform/meta/webrtc/recvonly-transceiver-can-become-sendrecv.https.html.ini new file mode 100644 index 0000000000..de8a6527d3 --- /dev/null +++ b/testing/web-platform/meta/webrtc/recvonly-transceiver-can-become-sendrecv.https.html.ini @@ -0,0 +1,3 @@ +[recvonly-transceiver-can-become-sendrecv.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/simplecall-no-ssrcs.https.html.ini b/testing/web-platform/meta/webrtc/simplecall-no-ssrcs.https.html.ini new file mode 100644 index 0000000000..6ca786ec3c --- /dev/null +++ b/testing/web-platform/meta/webrtc/simplecall-no-ssrcs.https.html.ini @@ -0,0 +1,3 @@ +[simplecall-no-ssrcs.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/simplecall.https.html.ini b/testing/web-platform/meta/webrtc/simplecall.https.html.ini new file mode 100644 index 0000000000..74f49de931 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simplecall.https.html.ini @@ -0,0 +1,3 @@ +[simplecall.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/simulcast/basic.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/basic.https.html.ini new file mode 100644 index 0000000000..6acb6d79ca --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/basic.https.html.ini @@ -0,0 +1,3 @@ +[basic.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/simulcast/getStats.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/getStats.https.html.ini new file mode 100644 index 0000000000..8a85d3ff87 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/getStats.https.html.ini @@ -0,0 +1,2 @@ +[getStats.https.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1643001, https://bugzilla.mozilla.org/show_bug.cgi?id=1787474 diff --git a/testing/web-platform/meta/webrtc/simulcast/h264.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/h264.https.html.ini new file mode 100644 index 0000000000..e5d2f82b1a --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/h264.https.html.ini @@ -0,0 +1,3 @@ +[h264.https.html] + [H264 simulcast setup with two spatial layers] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/simulcast/negotiation-encodings.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/negotiation-encodings.https.html.ini new file mode 100644 index 0000000000..33b1f3c1bf --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/negotiation-encodings.https.html.ini @@ -0,0 +1,3 @@ +[negotiation-encodings.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/simulcast/rid-manipulation.html.ini b/testing/web-platform/meta/webrtc/simulcast/rid-manipulation.html.ini new file mode 100644 index 0000000000..b84864478f --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/rid-manipulation.html.ini @@ -0,0 +1,5 @@ +[rid-manipulation.html] + expected: + if (os == "android") and fission: [TIMEOUT, OK] + [Remote reanswer altering rids does not throw an exception.] + expected: FAIL 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..cd80f27535 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini @@ -0,0 +1,18 @@ +[setParameters-active.https.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1787474 + expected: + if (os == "android") and swgl: [TIMEOUT, OK] + [OK, TIMEOUT] + [Simulcast setParameters active=false on first encoding stops sending frames for that encoding] + expected: [PASS, TIMEOUT] + + [Simulcast setParameters active=false on second encoding stops sending frames for that encoding] + expected: + if swgl and (os == "android"): [TIMEOUT, PASS, NOTRUN] + [PASS, TIMEOUT, NOTRUN] + + [Simulcast setParameters active=false stops sending frames] + expected: + if (os == "android") and swgl: [NOTRUN, PASS, TIMEOUT] + if (os == "mac") and not debug: [PASS, FAIL, TIMEOUT, NOTRUN] + [PASS, TIMEOUT, NOTRUN] diff --git a/testing/web-platform/meta/webrtc/simulcast/setParameters-encodings.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/setParameters-encodings.https.html.ini new file mode 100644 index 0000000000..9457c3f67e --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/setParameters-encodings.https.html.ini @@ -0,0 +1,3 @@ +[setParameters-encodings.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] diff --git a/testing/web-platform/meta/webrtc/simulcast/vp8.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/vp8.https.html.ini new file mode 100644 index 0000000000..b4e521ada4 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/vp8.https.html.ini @@ -0,0 +1,5 @@ +[vp8.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [VP8 simulcast setup with two spatial layers] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/toJSON.html.ini b/testing/web-platform/meta/webrtc/toJSON.html.ini new file mode 100644 index 0000000000..4f70284923 --- /dev/null +++ b/testing/web-platform/meta/webrtc/toJSON.html.ini @@ -0,0 +1,3 @@ +[toJSON.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] |