diff options
| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
| commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
| tree | f435a8308119effd964b339f76abb83a57c29483 /testing/web-platform/meta/webrtc | |
| parent | Initial commit. (diff) | |
| download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip | |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/meta/webrtc')
110 files changed, 1184 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-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..04bb84e712 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCConfiguration-iceServers.html.ini @@ -0,0 +1,24 @@ +[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 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-ontonechange.https.html.ini b/testing/web-platform/meta/webrtc/RTCDTMFSender-ontonechange.https.html.ini new file mode 100644 index 0000000000..aba16c4ab2 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDTMFSender-ontonechange.https.html.ini @@ -0,0 +1,3 @@ +[RTCDTMFSender-ontonechange.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 diff --git a/testing/web-platform/meta/webrtc/RTCDataChannel-GC.html.ini b/testing/web-platform/meta/webrtc/RTCDataChannel-GC.html.ini new file mode 100644 index 0000000000..db31aacbc1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-GC.html.ini @@ -0,0 +1,4 @@ +[RTCDataChannel-GC.html] + [While remote PC remains open, its datachannel should not be collected] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1858557 + expected: 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..ca36745a79 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-binaryType.window.js.ini @@ -0,0 +1,3 @@ +[RTCDataChannel-binaryType.window.html] + [Default binaryType value] + expected: FAIL 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..a2eabb9539 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-close.html.ini @@ -0,0 +1,62 @@ +[RTCDataChannel-close.html] + expected: + if (processor == "x86_64") and (os == "linux") and not fission and not debug and not asan: [OK, TIMEOUT] + if (processor == "x86_64") and (os == "win") and not debug: [OK, TIMEOUT] + if (processor == "x86") and not debug: [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: + if (processor == "x86_64") and (os == "linux") and not fission and not debug and not asan: [FAIL, NOTRUN] + if (processor == "x86_64") and (os == "win") and not debug: [FAIL, NOTRUN] + if (processor == "x86") and not debug: [FAIL, NOTRUN] + FAIL + + [Close negotiated datachannel causes onclosing and onclose to be called] + bug: Probably https://bugzilla.mozilla.org/show_bug.cgi?id=1611953 + expected: + if (processor == "x86_64") and (os == "linux") and not fission and not debug and not asan: [FAIL, NOTRUN] + if (processor == "x86_64") and (os == "win") and not debug: [FAIL, NOTRUN] + if (processor == "x86") and not debug: [FAIL, NOTRUN] + 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: + if (processor == "x86_64") and (os == "linux") and not fission and not debug and not asan: [FAIL, NOTRUN] + if (processor == "x86_64") and (os == "win") and not debug: [FAIL, NOTRUN] + if (processor == "x86") and not debug: [FAIL, NOTRUN] + 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: + if (processor == "x86_64") and (os == "linux") and not fission and not debug and not asan: [FAIL, NOTRUN] + if (processor == "x86_64") and (os == "win") and not debug: [FAIL, NOTRUN] + if (processor == "x86") and not debug: [FAIL, NOTRUN] + FAIL + + [Close peerconnection causes close event and error on many channels, datachannel] + bug: Probably https://bugzilla.mozilla.org/show_bug.cgi?id=1611953 + expected: + if (processor == "x86_64") and (os == "linux") and not fission and not debug and not asan: [FAIL, TIMEOUT] + if (processor == "x86_64") and (os == "win") and not debug: [FAIL, TIMEOUT] + if (processor == "x86") and not debug: [FAIL, TIMEOUT] + FAIL + + [Close peerconnection after negotiated datachannel close causes no events] + expected: + if (processor == "x86_64") and (os == "linux") and not fission and not debug and not asan: [PASS, NOTRUN] + if (processor == "x86_64") and (os == "win") and not debug: [PASS, NOTRUN] + if (processor == "x86") and not debug: [PASS, NOTRUN] 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..0ba52fcf7d --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-iceRestart.html.ini @@ -0,0 +1,10 @@ +[RTCDataChannel-iceRestart.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1728342 + expected: + if (os == "linux") and not swgl and not debug and not tsan and not fission and not asan: [ERROR, OK] + if (os == "linux") and not swgl and not debug and not tsan and fission: [ERROR, OK] + if (os == "linux") and not swgl and debug and fission: [ERROR, OK] + if (os == "linux") and not swgl and debug and not fission: [ERROR, OK] + if (os == "win") and not swgl and debug and (processor == "x86_64"): [ERROR, OK] + if (os == "win") and swgl: [ERROR, OK] + 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..15e9c598da --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-id.html.ini @@ -0,0 +1,3 @@ +[RTCDataChannel-id.html] + expected: + if (os == "win") and debug and (processor == "x86_64") and not swgl: [OK, CRASH] 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..7297ea8a74 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCDataChannel-send.html.ini @@ -0,0 +1,6 @@ +[RTCDataChannel-send.html] + [Datachannel binaryType should receive message as ArrayBuffer by default] + expected: FAIL + + [Negotiated datachannel binaryType should receive message as ArrayBuffer by default] + expected: FAIL 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..9bec62a2a7 --- /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=1805446 + 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..0c68ed7221 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCIceCandidate-constructor.html.ini @@ -0,0 +1,8 @@ +[RTCIceCandidate-constructor.html] + [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/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-GC.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-GC.https.html.ini new file mode 100644 index 0000000000..0cf20647af --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-GC.https.html.ini @@ -0,0 +1,8 @@ +[RTCPeerConnection-GC.https.html] + prefs: + # hw codecs disabled due to bug 1526207 + if os == "android": [media.navigator.mediadatadecoder_vpx_enabled:false, media.webrtc.hw.h264.enabled:false] + expected: + if (os == "win") and (processor == "x86_64") and debug and not swgl: [OK, CRASH] + if (os == "win") and (processor == "x86_64") and not debug: [OK, CRASH] + if (os == "win") and (processor == "x86"): [OK, CRASH] 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..6671543fff --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-addIceCandidate.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-addIceCandidate.html] + expected: + if (processor == "x86") and not debug: [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-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..bd68a49846 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-connectionState.https.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-connectionState.https.html] + [connection with one data channel should eventually have transports in connected state] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL 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..e30aeb8953 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-constructor.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-constructor.html] + [new RTCPeerConnection({ iceCandidatePoolSize: toNumberThrows })] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1529398 + expected: FAIL 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..4a87108c37 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-createDataChannel.html.ini @@ -0,0 +1,6 @@ +[RTCPeerConnection-createDataChannel.html] + [createDataChannel attribute default values] + expected: FAIL + + [createDataChannel with provided parameters should initialize attributes to provided values] + expected: FAIL 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..b4949aca01 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-generateCertificate.html.ini @@ -0,0 +1,5 @@ +[RTCPeerConnection-generateCertificate.html] + [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..462e7b8aad --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-getStats.https.html.ini @@ -0,0 +1,23 @@ +[RTCPeerConnection-getStats.https.html] + expected: + if (os == "win") and (processor == "x86_64") and not swgl: [OK, CRASH] + if (os == "android") and debug and not swgl: [OK, TIMEOUT] + [getStats() track without stream returns peer-connection and outbound-rtp stats] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1813847 + expected: [PASS, FAIL] + + [getStats() on track associated with RTCRtpSender should return stats report containing outbound-rtp stats] + expected: + if (os == "android") and debug and not swgl: [PASS, NOTRUN] + + [getStats() on track associated with RTCRtpReceiver should return stats report containing inbound-rtp stats] + expected: + if (os == "android") and debug and not swgl: [PASS, NOTRUN] + + [getStats(track) should not work if multiple senders have the same track] + expected: + if (os == "android") and debug and not swgl: [PASS, NOTRUN] + + [RTCStats.timestamp increases with time passing] + expected: + if (os == "android") and debug and not swgl: [PASS, 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-iceConnectionState.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini new file mode 100644 index 0000000000..e9900a5215 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini @@ -0,0 +1,16 @@ +[RTCPeerConnection-iceConnectionState.https.html] + [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=1278299 + expected: FAIL + + [iceConnectionState changes at the right time, with bundle policy max-compat] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [iceConnectionState changes at the right time, with bundle policy balanced] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + 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..c16c77891d --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-iceGatheringState.html.ini @@ -0,0 +1,8 @@ +[RTCPeerConnection-iceGatheringState.html] + [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..ac9b627ff2 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-mandatory-getStats.https.html.ini @@ -0,0 +1,59 @@ +[RTCPeerConnection-mandatory-getStats.https.html] + [RTCRtpStreamStats's transportId] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + 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 + + [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 + + [RTCIceCandidatePairStats's responsesReceived] + 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-operations.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-operations.https.html.ini new file mode 100644 index 0000000000..cfa53cbe53 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-operations.https.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-operations.https.html] + [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-relay-canvas.https.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-relay-canvas.https.html.ini new file mode 100644 index 0000000000..99fcd9b189 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-relay-canvas.https.html.ini @@ -0,0 +1,7 @@ +[RTCPeerConnection-relay-canvas.https.html] + disabled: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1728435 + if (os == "linux") and (processor == "x86"): https://bugzilla.mozilla.org/show_bug.cgi?id=1813323 + [Two PeerConnections relaying a canvas source] + expected: + if (os == "linux") and not debug: [PASS, 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..72acf393c4 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini @@ -0,0 +1,2 @@ +[RTCPeerConnection-remote-track-mute.https.html] + prefs: [media.peerconnection.mute_on_bye_or_timeout:true] 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..370dbcee23 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-restartIce.https.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-restartIce.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 + [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-setLocalDescription-answer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini new file mode 100644 index 0000000000..034e700dd1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini @@ -0,0 +1,6 @@ +[RTCPeerConnection-setLocalDescription-answer.html] + [Calling setLocalDescription(answer) from stable state should reject with InvalidStateError] + expected: FAIL + + [Calling setLocalDescription(answer) from have-local-offer state should reject with InvalidStateError] + expected: FAIL 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..8e2eb5fcf8 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setLocalDescription-parameterless.https.html.ini @@ -0,0 +1,8 @@ +[RTCPeerConnection-setLocalDescription-parameterless.https.html] + [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-setRemoteDescription-offer.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini new file mode 100644 index 0000000000..19a74d60e5 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-setRemoteDescription-offer.html] + expected: + if (os == "linux") and not debug and fission and (processor == "x86_64"): [OK, CRASH] + [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-rollback.html.ini b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini new file mode 100644 index 0000000000..3e84ce0b22 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini @@ -0,0 +1,8 @@ +[RTCPeerConnection-setRemoteDescription-rollback.html] + [explicit rollback of local offer should remove transceivers and transport] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1805474 + 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=1805474 + 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..62ae0afaec --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-simulcast.https.html.ini @@ -0,0 +1,4 @@ +[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 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..bf488e2f0c --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini @@ -0,0 +1,3 @@ +[RTCPeerConnection-setRemoteDescription-tracks.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 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..e77b55bfab --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCPeerConnection-transceivers.https.html.ini @@ -0,0 +1,5 @@ +[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] 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..2744e3e051 --- /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, OK] + [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, PASS] 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-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..dd1c0538e4 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getStats.https.html.ini @@ -0,0 +1,4 @@ +[RTCRtpReceiver-getStats.https.html] + [receiver.getStats() should work on a stopped transceiver but not have inbound-rtp objects] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1879605 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..6b8799454b --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini @@ -0,0 +1,4 @@ +[RTCRtpReceiver-getSynchronizationSources.https.html] + [[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..fd3804482a --- /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 not debug: [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..48aa66cee3 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-getCapabilities.html.ini @@ -0,0 +1,3 @@ +[RTCRtpSender-getCapabilities.html] + expected: + if (os == "android") and not debug: [OK, ERROR] 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..b7ae7fbe54 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-getStats.https.html.ini @@ -0,0 +1,4 @@ +[RTCRtpSender-getStats.https.html] + [sender.getStats() should work on a stopped transceiver but not have outbound-rtp stats] + expected: FAIL + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1879605 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..0b649149b7 --- /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, OK] + [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, PASS] + [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, PASS] diff --git a/testing/web-platform/meta/webrtc/RTCRtpSender-setParameters-keyFrame.html.ini b/testing/web-platform/meta/webrtc/RTCRtpSender-setParameters-keyFrame.html.ini new file mode 100644 index 0000000000..4fdd451713 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-setParameters-keyFrame.html.ini @@ -0,0 +1,6 @@ +[RTCRtpSender-setParameters-keyFrame.html] + [setParameters() second argument can be used to trigger keyFrame generation] + expected: FAIL + + [setParameters() second argument can be used to trigger keyFrame generation (simulcast)] + expected: FAIL 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..46d128f985 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpSender-transport.https.html.ini @@ -0,0 +1,12 @@ +[RTCRtpSender-transport.https.html] + [RTCRtpSender/receiver/SCTP transport at the right time, with bundle policy balanced] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [RTCRtpSender/receiver/SCTP transport at the right time, with bundle policy max-bundle] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL + + [RTCRtpSender/receiver/SCTP transport at the right time, with bundle policy max-compat] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1278299 + expected: FAIL 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.https.html.ini b/testing/web-platform/meta/webrtc/RTCRtpTransceiver.https.html.ini new file mode 100644 index 0000000000..db4f5f2282 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCRtpTransceiver.https.html.ini @@ -0,0 +1,3 @@ +[RTCRtpTransceiver.https.html] + restart-after: + if os == "android": https://bugzilla.mozilla.org/show_bug.cgi?id=1641237 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-fire.html.ini b/testing/web-platform/meta/webrtc/RTCTrackEvent-fire.html.ini new file mode 100644 index 0000000000..03bf543781 --- /dev/null +++ b/testing/web-platform/meta/webrtc/RTCTrackEvent-fire.html.ini @@ -0,0 +1,2 @@ +[RTCTrackEvent-fire.html] + prefs: [media.peerconnection.sdp.alternate_parse_mode:never, media.peerconnection.sdp.parser:sipcc] diff --git a/testing/web-platform/meta/webrtc/__dir__.ini b/testing/web-platform/meta/webrtc/__dir__.ini new file mode 100644 index 0000000000..736938ab3d --- /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, media.aboutwebrtc.hist.poll_interval_ms:2000] +lsan-allowed: [Alloc, MakeAndAddRef, MakeUnique, Malloc, NS_NewDOMDataChannel, NS_NewRunnableFunction, NewPage, PR_NewMonitor, PR_Realloc, ParentContentActorCreateFunc, WrapRelease, allocate, mozilla::DataChannelConnection::Create, mozilla::DataChannelConnection::Destroy, mozilla::DataChannelConnection::HandleOpenRequestMessage, mozilla::DataChannelConnection::Open, mozilla::MediaPacket::Copy, mozilla::MediaPipeline::MediaPipeline, mozilla::NrSocketBase::CreateSocket, mozilla::WeakPtr, mozilla::dom::DocGroup::Create, mozilla::dom::DocGroup::DocGroup, mozilla::runnable_args_func, nsRefPtrDeque, nsThread::nsThread, nsThreadManager::NewNamedThread, sctp_add_vtag_to_timewait, sctp_alloc_chunklist, sctp_alloc_hmaclist, sctp_alloc_sharedkey, sctp_hashinit_flags, sctp_inpcb_alloc] +leak-threshold: [default:3020800, rdd:51200, tab:51200] diff --git a/testing/web-platform/meta/webrtc/back-forward-cache-with-closed-webrtc-connection-ccns.https.tentative.window.js.ini b/testing/web-platform/meta/webrtc/back-forward-cache-with-closed-webrtc-connection-ccns.https.tentative.window.js.ini new file mode 100644 index 0000000000..0158f77366 --- /dev/null +++ b/testing/web-platform/meta/webrtc/back-forward-cache-with-closed-webrtc-connection-ccns.https.tentative.window.js.ini @@ -0,0 +1,3 @@ +[back-forward-cache-with-closed-webrtc-connection-ccns.https.tentative.window.html] + [Testing BFCache support for page with closed WebRTC connection and "Cache-Control: no-store" header.] + expected: PRECONDITION_FAILED diff --git a/testing/web-platform/meta/webrtc/back-forward-cache-with-open-webrtc-connection-ccns.https.tentative.window.js.ini b/testing/web-platform/meta/webrtc/back-forward-cache-with-open-webrtc-connection-ccns.https.tentative.window.js.ini new file mode 100644 index 0000000000..3801657b57 --- /dev/null +++ b/testing/web-platform/meta/webrtc/back-forward-cache-with-open-webrtc-connection-ccns.https.tentative.window.js.ini @@ -0,0 +1,3 @@ +[back-forward-cache-with-open-webrtc-connection-ccns.https.tentative.window.html] + [Testing BFCache support for page with open WebRTC connection and "Cache-Control: no-store" header.] + expected: PRECONDITION_FAILED 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..b9c8b6f268 --- /dev/null +++ b/testing/web-platform/meta/webrtc/getstats.html.ini @@ -0,0 +1,3 @@ +[getstats.html] + expected: + if (os == "linux") and not debug and fission and (processor == "x86_64"): [OK, CRASH] 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..20015d542b --- /dev/null +++ b/testing/web-platform/meta/webrtc/historical.html.ini @@ -0,0 +1,20 @@ +[historical.html] + [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 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..ed8b630bd3 --- /dev/null +++ b/testing/web-platform/meta/webrtc/idlharness.https.window.js.ini @@ -0,0 +1,378 @@ +[idlharness.https.window.html] + [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 + + [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 "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 + + [RTCIceCandidate interface: attribute sdpMid] + expected: FAIL + + [RTCIceCandidate interface: attribute relatedAddress] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "priority" with the proper type] + expected: FAIL + + [RTCIceTransport interface: operation getSelectedCandidatePair()] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCErrorEvent must be primary interface of new RTCErrorEvent('error')] + 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 + + [RTCDTMFSender interface: attribute canInsertDTMF] + 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 + + [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 + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "getRemoteCertificates()" with the proper type] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getLocalParameters()" with the proper type] + expected: FAIL + + [RTCIceTransport interface: attribute state] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "onstatechange" with the proper type] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCCertificate interface: idlTestObjects.certificate must inherit property "getFingerprints()" with the proper type] + 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 + + [RTCIceTransport interface: operation getLocalParameters()] + expected: FAIL + + [RTCIceTransport interface: attribute ongatheringstatechange] + 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 + + [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 + + [RTCIceTransport interface: operation getLocalCandidates()] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute url] + 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 + + [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 + + [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/__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/munge-dont.html.ini b/testing/web-platform/meta/webrtc/legacy/munge-dont.html.ini new file mode 100644 index 0000000000..91d01d5f4c --- /dev/null +++ b/testing/web-platform/meta/webrtc/legacy/munge-dont.html.ini @@ -0,0 +1,15 @@ +[munge-dont.html] + [RTCSessionDescription.type is read-only] + expected: FAIL + + [RTCSessionDescription.sdp is read-only] + expected: FAIL + + [RTCIceCandidate.candidate is read-only] + expected: FAIL + + [Rejects SDP munging between createOffer and setLocalDescription] + expected: FAIL + + [Rejects SDP munging between createAnswer and setLocalDescription] + expected: FAIL 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/additional-codecs.html.ini b/testing/web-platform/meta/webrtc/protocol/additional-codecs.html.ini new file mode 100644 index 0000000000..a0f2f64f43 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/additional-codecs.html.ini @@ -0,0 +1,3 @@ +[additional-codecs.html] + [Listing an additional codec in the answer causes it to be sent.] + expected: FAIL 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..8c84464872 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/bundle.https.html.ini @@ -0,0 +1,23 @@ +[bundle.https.html] + expected: + if (os == "android") and debug and not swgl: [OK, TIMEOUT] + if (os == "win") and not debug and (processor == "x86"): [OK, CRASH] + [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=1805480 + expected: + if (os == "android") and debug and not swgl: [FAIL, TIMEOUT] + FAIL + + [max-bundle with an offer without bundle only negotiates the first m-line] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1805484 + expected: + if (os == "android") and debug and not swgl: [FAIL, NOTRUN] + FAIL + + [sRD(offer) works with no transport attributes in a bundle-only m-section] + expected: + if (os == "android") and debug and not swgl: [PASS, NOTRUN] 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..297b54b1f8 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/candidate-exchange.https.html.ini @@ -0,0 +1,18 @@ +[candidate-exchange.https.html] + expected: + if (os == "linux") and not debug and fission: [OK, CRASH] + [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..116d12f4e4 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/crypto-suite.https.html.ini @@ -0,0 +1,24 @@ +[crypto-suite.https.html] + [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 + + [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-certificates.html.ini b/testing/web-platform/meta/webrtc/protocol/dtls-certificates.html.ini new file mode 100644 index 0000000000..3e512c9e93 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/dtls-certificates.html.ini @@ -0,0 +1,6 @@ +[dtls-certificates.html] + expected: + if (os == "android") and not debug: [OK, TIMEOUT] + [RTCPeerConnection establishes using rsa and rsa certificates] + expected: + if (os == "android") and not debug: [PASS, TIMEOUT] 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..34cafce8e1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/dtls-fingerprint-validation.html.ini @@ -0,0 +1,3 @@ +[dtls-fingerprint-validation.html] + expected: + if tsan: CRASH 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..3ad3443d9b --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/dtls-setup.https.html.ini @@ -0,0 +1,16 @@ +[dtls-setup.https.html] + [PC with setup=actpass should have a dtlsRole of client] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [PC with setup=active should have a dtlsRole of server] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [PC with setup=passive should have a dtlsRole of client] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + expected: FAIL + + [dtlsRole is `unknown` before negotiation of the DTLS handshake] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723 + 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..2f72b22a32 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/h264-profile-levels.https.html.ini @@ -0,0 +1,2 @@ +[h264-profile-levels.https.html] + disabled: https://bugzilla.mozilla.org/show_bug.cgi?id=1396922 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..9702bc1803 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/handover-datachannel.html.ini @@ -0,0 +1,2 @@ +[handover-datachannel.html] + disabled: https://github.com/web-platform-tests/wpt/issues/37561 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..b8feb75485 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/handover.html.ini @@ -0,0 +1,2 @@ +[handover.html] + disabled: https://github.com/web-platform-tests/wpt/issues/37561 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..9b13a9e695 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/ice-state.https.html.ini @@ -0,0 +1,4 @@ +[ice-state.https.html] + [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..1dcc567dbc --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/ice-ufragpwd.html.ini @@ -0,0 +1,8 @@ +[ice-ufragpwd.html] + [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/rtp-clockrate.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini new file mode 100644 index 0000000000..f8b32255b3 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-clockrate.html.ini @@ -0,0 +1,5 @@ +[rtp-clockrate.html] + [video rtp timestamps increase by approximately 90000 per second] + expected: + if (os == "win") and not debug and (processor == "x86"): [PASS, FAIL] + if (os == "mac") and not debug: [PASS, FAIL] 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..186f43bc00 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-demuxing.html.ini @@ -0,0 +1,10 @@ +[rtp-demuxing.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1818283 + expected: [OK, TIMEOUT] + [Can demux two video tracks with different payload types on a bundled connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1396922 + expected: FAIL + + [Can demux two video tracks with the same payload type on an unbundled connection] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1818283 + expected: [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..44bb7ecbae --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-extension-support.html.ini @@ -0,0 +1,4 @@ +[rtp-extension-support.html] + [RTP header extension urn:3gpp:video-orientation is present in offer] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1340372 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/protocol/rtp-headerextensions.html.ini b/testing/web-platform/meta/webrtc/protocol/rtp-headerextensions.html.ini new file mode 100644 index 0000000000..af9dc6374b --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-headerextensions.html.ini @@ -0,0 +1,12 @@ +[rtp-headerextensions.html] + [Video orientation header extension is supported.] + expected: FAIL + + [Negotiates the subset of supported extensions offered] + expected: FAIL + + [Supports header extensions with id=15] + expected: FAIL + + [Supports two-byte header extensions] + 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..59d1862d17 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/rtp-payloadtypes.html.ini @@ -0,0 +1,4 @@ +[rtp-payloadtypes.html] + [setRemoteDescription with a codec in the range 64-95 throws an InvalidAccessError] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1806181 + 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/transceiver-mline-recycling.html.ini b/testing/web-platform/meta/webrtc/protocol/transceiver-mline-recycling.html.ini new file mode 100644 index 0000000000..4ac0c264e1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/transceiver-mline-recycling.html.ini @@ -0,0 +1,6 @@ +[transceiver-mline-recycling.html] + [Reuses m-lines in local negotiation] + expected: FAIL + + [Reuses m-lines in remote negotiation] + expected: FAIL 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..9a216b2119 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/unknown-mediatypes.html.ini @@ -0,0 +1,4 @@ +[unknown-mediatypes.html] + [Unknown media types are rejected with the port set to 0] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1806185 + 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..659a322d55 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/video-codecs.https.html.ini @@ -0,0 +1,9 @@ +[video-codecs.https.html] + max-asserts: 3 + [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..812d1ea704 --- /dev/null +++ b/testing/web-platform/meta/webrtc/protocol/vp8-fmtp.html.ini @@ -0,0 +1,4 @@ +[vp8-fmtp.html] + [setRemoteDescription parses max-fr and max-fs fmtp parameters] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531464 + expected: FAIL 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..f91573adf1 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/basic.https.html.ini @@ -0,0 +1,12 @@ +[basic.https.html] + expected: + if (os == "win") and (processor == "x86_64") and debug and swgl: [OK, TIMEOUT] + if (os == "win") and (processor == "x86_64") and debug and not swgl: TIMEOUT + if (os == "win") and (processor == "x86"): [OK, TIMEOUT] + if (os == "linux") and not debug: [OK, TIMEOUT] + [Basic simulcast setup with two spatial layers] + expected: + if (os == "win") and (processor == "x86_64") and debug and swgl: [PASS, TIMEOUT] + if (os == "win") and (processor == "x86_64") and debug and not swgl: TIMEOUT + if (os == "win") and (processor == "x86"): [PASS, TIMEOUT] + if (os == "linux") and not debug: [PASS, 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..b1c8f8de45 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/h264.https.html.ini @@ -0,0 +1,4 @@ +[h264.https.html] + [H264 simulcast setup with two streams] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1396922 + 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..8145be5443 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/negotiation-encodings.https.html.ini @@ -0,0 +1,20 @@ +[negotiation-encodings.https.html] + expected: + if (os == "android") and fission: [OK, TIMEOUT] + [addTrack, then sRD(simulcast recv offer) results in simulcast] + expected: FAIL + + [sRD(simulcast offer) can narrow the simulcast envelope from a previous negotiation] + expected: FAIL + + [Duplicate rids in sRD(offer) are ignored] + expected: FAIL + + [Choices in rids in sRD(offer) are ignored] + expected: FAIL + + [Reordering of rids in sRD(reoffer) is ignored] + expected: FAIL + + [sRD(simulcast offer) can narrow the simulcast envelope from a previous negotiation by removing the first encoding] + expected: FAIL 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..73313b2a80 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/rid-manipulation.html.ini @@ -0,0 +1,2 @@ +[rid-manipulation.html] + disabled: https://github.com/web-platform-tests/wpt/issues/37564 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..5c93a4adea --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini @@ -0,0 +1,13 @@ +[setParameters-active.https.html] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1787474 + expected: [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: [PASS, TIMEOUT, NOTRUN] + + [Simulcast setParameters active=false stops sending frames] + expected: + 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..8b2d9e33dd --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/vp8.https.html.ini @@ -0,0 +1,4 @@ +[vp8.https.html] + [VP8 simulcast setup with two streams] + bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1396922 + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/simulcast/vp9-scalability-mode.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/vp9-scalability-mode.https.html.ini new file mode 100644 index 0000000000..28e7afe277 --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/vp9-scalability-mode.https.html.ini @@ -0,0 +1,3 @@ +[vp9-scalability-mode.https.html] + [VP9 simulcast setup with two streams and L1T2 set] + expected: FAIL diff --git a/testing/web-platform/meta/webrtc/simulcast/vp9.https.html.ini b/testing/web-platform/meta/webrtc/simulcast/vp9.https.html.ini new file mode 100644 index 0000000000..348df638cf --- /dev/null +++ b/testing/web-platform/meta/webrtc/simulcast/vp9.https.html.ini @@ -0,0 +1,3 @@ +[vp9.https.html] + [VP9 simulcast setup with two streams] + expected: FAIL |