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.""