Media/WebRTC/ReleaseNotes/68
Contents
- 1 Firefox 68 WebRTC/WebAudio Release Notes:
- 1.1 Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 68:
- 1.2 Audio/Video: GMP:
- 1.3 Audio/Video: MediaStreamGraph:
- 1.4 Audio/Video: Recording:
- 1.5 Audio/Video: cubeb:
- 1.6 Web Audio:
- 1.7 WebRTC:
- 1.8 WebRTC: Audio/Video:
- 1.9 WebRTC: Networking:
- 1.10 WebRTC: Signaling:
- 1.11 Intermittent Test failures:
- 1.12 Web Platform Tests:
Firefox 68 WebRTC/WebAudio Release Notes:
Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 68:
WebRTC and WebAudio bugs: Bugzilla search for WebRTC and WebAudio related bugs marked Fixed in Firefox 68
Audio/Video: GMP:
bug 1535010 GMPDiskStorage.cpp and GMPServiceParent.cpp do unnecessary I/O syscalls to create directories during startup
bug 1544602 Assertion failure: IsAtomic<bool>::value || NS_IsMainThread() (Non-atomic static pref 'media.gmp.insecure.allow' being accessed on background thread)
Audio/Video: MediaStreamGraph:
bug 1423253 Kill NotifyPull for video tracks
bug 1538630 Check a predicate when waiting on condition variables in GraphRunner
bug 1538640 wait for GraphRunner thread shutdown
bug 1539045 use AppendMessage() for ForceShutDown()
bug 1541290 Crash on Web Speech API, (Speech Recognition portion) when feeding audio from the microphone
bug 1551855 Add dedicated pref to enable GraphRunner also when audio worklets is disabled
Audio/Video: Recording:
bug 1532391 Extend lifetime of MEDIA_RECORDER_RECORDING_DURATION, MEDIA_RECORDER_TRACK_ENCODER_INIT_TIMEOUT_TYPE, and SCALARS_MEDIARECORDER.RECORDING_COUNT telemetry probes
bug 1538113 Fix warnings from bug 1423253 landing
bug 1538727 Assertion failure: false (Not connected to this video track), at /builds/worker/workspace/build/src/dom/media/encoder/MediaEncoder.cpp:612
bug 1542685 Crash in [@ mozilla::DriftCompensator::GetVideoTime]
Audio/Video: cubeb:
bug 1531833 Miscellaneous audio improvements on Windows and Android
bug 1533539 Crash in [@ mozilla::CubebUtils::GetCubebContextUnlocked]
bug 1536605 New warnings in audioipc with rust 1.34 due to ATOMIC_USIZE_INIT usage
bug 1541101 crash in [@ audiounit_stream_start ]
bug 1541805 Crash in [@ wasapi_init]
bug 1545279 Crash in [@ monitor_device_notifications::notify]
bug 1546872 Audio devices appear to only be enumerated on content process creation
bug 1552342 Update libcubeb to pick up PR 507.
Web Audio:
bug 1324548 Implement MediaStreamTrackAudioSourceNode
bug 1375562 Random actions cause a suspended AudioContext to resume.
bug 1445923 WebAudio: Remove b2g dead code
bug 1456269 Construct OscillatorNode with PeriodicWave fails
bug 1456962 Update default channel attributes for DynamicsCompressorNode
bug 1477205 Stop throwing error when creating AudioNodes on a closed context
bug 1528319 Reloading after creating AudioContext causes InvalidStateError
bug 1530178 [Web Audio API] copyFromChannel/copyToChannel error occurs
bug 1538470 Crash in [@ mozilla::dom::AudioNode::AudioNode]
bug 1539522 windows/aarch64 - dom/media/webaudio/test/test_audioContextSuspendResumeClose.html | Test timed out.
bug 1541311 add support for AudioWorkletNode.numberOfInputs/Outputs
bug 1541467 AddressSanitizer: SEGV /builds/worker/workspace/build/src/obj-firefox/dist/include/nsPIDOMWindow.h:517:38 in WindowID
bug 1549041 audible is no longer fired with tabs.onUpdated after a tab is reloaded or a navigation happened
WebRTC:
bug 1072388 Cannot call createOffer/setLocalDescription in "have-local-offer" state, nor createAnswer/setRemoteDescription in "have-remote-offer" state
bug 1496359 [Wayland] We need to implement PipeWire support
bug 1512281 Add a pref to turn off RTCP in WebRTC to prevent regressions in which the local stats are used for the remote stats (again)
bug 1515716 Refactor WebRTC RTP stats types
bug 1525323 Assertion failure: false (A non-finished SourceMediaStream wasn't fed enough data by NotifyPull), at /builds/worker/workspace/build/src/dom/media/MediaStreamGraph.cpp:1254
bug 1531494 Remove all non-implemented RTC stats dictionaries and fields from the WebIDL and the IPC code
bug 1532898 Move WebRTC Video Telemetry recording to the VideoConduit
bug 1534466 implement getSynchronizationSources for Video
bug 1535766 Crash in [@ mozilla::WebrtcGmpVideoEncoder::Encoded]
bug 1537567 windows/aarch64 - dom/media/tests/mochitest/test_peerConnection_setParameters_scaleResolutionDownBy.html | Error in test execution: Error: Timeout checkScaleDownBy@http://mochi.test:8888/tests/dom/media/tests/mochitest/test_peerConnection_setParameter
bug 1538359 windows/aarch64 - Intermittent dom/media/tests/mochitest/test_getUserMedia_audioCapture.html | Error executing test: Error: Audio analysis timed out waitForAnalysisSuccess@https://example.com/tests/dom/media/tests/mochitest/head.js:196:63 ... @https://
bug 1538508 FF 66.0 breaks H.264 basline constrained for WebRTC
bug 1539220 Browser crashes when User tries to set avatar image with Camera option.
bug 1539809 Assertion failure: !oldTransceiver.HasLevel() || !HasLevel() || oldTransceiver.GetLevel() == GetLevel(), at /builds/worker/workspace/build/src/media/webrtc/signaling/src/jsep/JsepTransceiver.h:61
bug 1541553 Once a non-zero RTT is reported we are allowed to report RTTs of zero, we should do so
bug 1543938 Perma LeakSanitizer | leak at alloc, __rdl_alloc, alloc::alloc::alloc, _$LT$alloc..alloc..Global$u20$as$u20$core..alloc..Alloc$GT$::alloc
bug 1545090 Assertion failure: !aTransportId.empty(), at /builds/worker/workspace/build/src/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.cpp:393
bug 1547278 Assertion failure: false, at /builds/worker/workspace/build/src/media/webrtc/signaling/src/peerconnection/MediaTransportHandler.cpp:493
bug 1548097 getContributingSources and getSynchronizationSources should return results sorted by playout time in descending order
WebRTC: Audio/Video:
bug 1335740 Disable getUserMedia on non-secure origins
bug 1407415 CamerasParent::StopVideoCapture() should try and avoid blocking the IPDL Background ("PBackground") thread
bug 1494675 Remove the AllocationHandle API
bug 1497559 Remove support for application capturing from our local copy of webrtc.org
bug 1506884 Audit and document member access from threads in AudioConduit
bug 1528078 Add telemetry for getUserMedia/getDisplayMedia/enumerateDevices secure vs insecure vs legacy
bug 1532576 Fallback openh264 gmp source file is out of date
bug 1533071 Enable openh264 plugin for win64-aarch64
bug 1534313 Make the CubebDeviceEnumerator the only path to enumerate audio devices
bug 1540251 Workaround unset OpenH264 NAL size in WebrtcGmpVideoEncoder::Encoded
bug 1540434 Crash in [@ mozilla::GetUserMediaWindowListener::Remove]
bug 1546865 getUserMedia({audio}) right after audioTrack.stop() fails with AbortError
bug 1549383 Bustage on src/dom/media/systemservices/CamerasParent.cpp when Gecko 68 merges to Beta on 2019-05-07
bug 1549699 Restore previous behaviour for audio devices on Windows and Android
bug 1551361 Add more logging to the basic RTP extensions test
WebRTC: Networking:
bug 1318167 Add support for ICE end of candidate
bug 1518609 Add Telemetry to determine when maxRetransmitTime in DataChannel init can be deprecated
bug 1535868 Negotiating DTLS without SRTP extension results in random crashes
bug 1545827 Get webrtc https proxy for TURN/TCP working with the socket process
bug 1546562 ICE Restart fails when re-negotiating after ICE failure.
bug 1546691 PeerConnectionObserver can spontaneously go away when network is lost
bug 1548272 DataChannel::GetOrdered makes a racy access to DataChannel::mFlags
bug 1550540 Crash with failed "@mozilla.org/peerconnection;1" instance
bug 1551702 Hide DataChannelConnection ctor, set local port on construction
bug 1551740 Assertion failure: !stream->obsolete, at media/mtransport/nricectx.cpp:420
WebRTC: Signaling:
bug 1225877 Parse latest a=simulcast and a=rid
bug 1240897 Firefox incorrectly generates "a=setup" line in answer when negotiated DTLS role is "passive".
bug 1288105 Opus payload type mis-match results in broken audio
bug 1518672 signalingstatechange event fires too soon.
bug 1529595 Remove "token" from RTCIceCredentialType
bug 1529612 RTCDataChannel.bufferedAmount is updated too soon after sending data
bug 1529635 RTCIceCandidate constructor validation for sdpMid/sdpMLineIndex is not implemented
bug 1529695 Implement RTCDataChannel.negotiated
bug 1529708 RTCIceConnectionState-candidate-pair.https.html.ini needs to be removed
bug 1531078 Fuzzy Date.now precision could cause tests to fail
bug 1531110 Handle setLocalDescription (either offer or answer) with empty sdp string
bug 1531122 JsepSessionImpl can erroneously compare a locally-created offer with a locally set answer
bug 1531803 RTCTrackEvent-fire.html wpt wants ontrack events when a=msid is altered in remote description
bug 1531828 RTCDTMFSender ontonechange events should stop if the transceiver stops sending
bug 1531894 createDataChannel throws InvalidParameterError instead of TypeError if both maxRetransmits and maxPacketLifeTime are set
bug 1531904 RTCPeerConnection.createDataChannel doesn't do a very good job of validating stream ids
bug 1531908 RTCPeerConnection.createDataChannel does not check the length of the label
bug 1531910 RTCPeerConnection.createDataChannel does not check the length of the protocol
bug 1531914 RTCRtpTransceiver-stop.html wpt has a flawed test: "A stopped sendonly transceiver should generate an inactive m-section in the offer"
bug 1534673 Stop paying attention to msid-semantic when parsing a=msid
bug 1534683 webrtc/protocol/msid-parse.html uses malformed sdp
bug 1534692 mock-idp.js does not seem to be working in the webrtc wpt
bug 1535410 RTCPeerConnection.addIceCandidate validation for sdpMid/sdpMLineIndex is not implemented
bug 1535442 Pay attention to ufrag when incorporating ICE candidates into SDP
bug 1536631 Invalid modifications to SDP should result in an InvalidModificationError
bug 1540752 Clean up cruft left in meta/webrtc/idlharness.https.window.js.ini
bug 1542021 NS_ERROR_UNEXPECTED from PeerConnection.jsm in signaling rollback demo
bug 1542343 RTCDataChannel-send.html disabled on aarch64
bug 1542345 RTCRtpTransceiver.https.html disabled on aarch64
bug 1542907 Should ignore multiple identical msids
bug 1543425 Calling createOffer then transceiver.stop() on a just-added transceiver can cause nullptr crashes
bug 1543427 Setting local offer, then transceiver.stop(), then local rollback, then a remote offer, then addIceCandidate can cause nullptr crashes
bug 1543429 Rejecting the bundle tag can lead to JSEP errors
bug 1546396 meta/webrtc/RTCPeerConnection-connectionState.https.html.ini needs to be updated
bug 1546402 A bunch of new failing tests in webrtc/RTCPeerConnection-createDataChannel.html
bug 1546404 A bunch of new failing tests in webrtc/RTCPeerConnection-ondatachannel.html
bug 1546406 Need to update meta/webrtc/RTCSctpTransport-events.html.ini
bug 1546408 simulcast-answer.html is flawed
bug 1546981 webrtc/RTCPeerConnection-setLocalDescription-answer.html has a duplicate test-case
Intermittent Test failures:
bug 1373123 Intermittent dom/media/tests/mochitest/test_peerConnection_stats.html | Error in test execution: Error: Waiting for synced RTCP timed out after at least 15000ms waitForSyncedRtcp@http://mochi.test:8888/tests/dom/media/tests/mochitest/test_peerConnection_
bug 1407650 Intermittent dom/media/test/test_mediarecorder_record_changing_video_resolution.html | Expected number of resize events - got 2, expected 3
bug 1504336 Intermittent dom/media/tests/mochitest/test_peerConnection_simulcastOddResolution.html | Width 640 should be within 10% of 1280 for rid 'foo'
bug 1511542 Intermittent GECKO(1060) | Assertion failure: NS_IsMainThread(), at /builds/worker/workspace/build/src/media/webrtc/signaling/src/media-conduit/VideoConduit.cpp:451
bug 1538232 Intermittent GECKO(1801) | Assertion failure: !iter->IsNull(), at /builds/worker/workspace/build/src/dom/media/encoder/TrackEncoder.cpp:618
bug 1541030 Intermittent Assertion failure: mStream (How come we don't have a stream here?), at /builds/worker/workspace/build/src/dom/media/webaudio/AudioNode.cpp:280
Web Platform Tests:
bug 1504514 [wpt-sync] Sync PR 13902 - An incoming offer can generate simulcast
bug 1531387 [wpt-sync] Sync PR 15535 - Updates RTCIceTransport to standard state API.
bug 1532123 [wpt-sync] Sync PR 15531 - Exposing RID attribute in RTCRtpCodingParameters.
bug 1532151 [wpt-sync] Sync PR 15520 - Add support for AudioContextOptions sampleRate
bug 1532522 [wpt-sync] Sync PR 15621 - Mark MediaDevices-related interfaces as SecureContext
bug 1534108 [wpt-sync] Sync PR 15651 - RTCError: Make "message" optional and be the last argument.
bug 1534144 [wpt-sync] Sync PR 15710 - ABSN with null buffer should output silence
bug 1535709 [wpt-sync] Sync PR 15778 - Add SctpTransport API
bug 1535797 [wpt-sync] Sync PR 15438 - Update RTCDataChannel bufferedamountlow implementation.
bug 1535850 [wpt-sync] Sync PR 15790 - Use matching sample rate for the context as for the reference file
bug 1536651 [wpt-sync] Sync PR 15911 - webrtc wpt: fix use of helper function
bug 1537584 [wpt-sync] Sync PR 15945 - Create RTCIceTransport using a webrtc::IceTransportInterface object.
bug 1538312 [wpt-sync] Sync PR 15957 - Revert "Create RTCIceTransport using a webrtc::IceTransportInterface object."
bug 1538392 [wpt-sync] Sync PR 16017 - Reland "Create RTCIceTransport using a webrtc::IceTransportInterface object."
bug 1539655 [wpt-sync] Sync PR 16046 - Include html WebIDL in idlharness for WebRTC
bug 1539679 [wpt-sync] Sync PR 15925 - webrtc wpt: add connectionState tests
bug 1539996 [wpt-sync] Sync PR 16092 - Adding WPT for accepting an offer to receive simulcast.
bug 1541338 [wpt-sync] Sync PR 16131 - s/transciever/transceiver
bug 1541501 [wpt-sync] Sync PR 16038 - Data channel tests updated by Lennart Grahl <lennart.grahl@gmail.com>
bug 1541505 [wpt-sync] Sync PR 16037 - Add RTCSctpTransport basic state tests
bug 1541509 [wpt-sync] Sync PR 16042 - Replace generateOffer by generateAudioReceiveOnlyOffer
bug 1541549 [wpt-sync] Sync PR 16053 - Revert "Reland "Create RTCIceTransport using a webrtc::IceTransportInterface object.""