Firefox 67 WebRTC/WebAudio Release Notes:

Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 67:

WebRTC and WebAudio bugs: Bugzilla search for WebRTC and WebAudio related bugs marked Fixed in Firefox 67

Audio/Video: GMP:

bug 1515210 no open h.264 cisco codec for webrtc support for aarch64

bug 1532354 Remove dead code from GMPServiceParent: ProcessPossiblePlugin and DeleteGMPServiceParent

bug 1532578 Automatic installation of OpenH264 plugin is broken on Android on Firefox 67

bug 1532756 Unable to load OpenH264 plugin on Android

Audio/Video: MediaStreamGraph:

bug 1473469 Run MediaStreamGraph from a single thread

bug 1521577 1.5 seconds audio delay in appear.in

bug 1528436 Perma TEST-UNEXPECTED-FAIL : xperf: File 'c:＼windows＼system32＼dxgi.dll' (normalized from 'C:＼Windows＼System32＼dxgi.dll') was accessed and we were not expecting it. DiskReadCount: 2, DiskWriteCount: 0, DiskReadBytes: 32768, DiskWriteBytes: 0

bug 1529399 Remove unnecessary wrapper runnable from CreateDirectTaskDrainer() for stable state runnables

bug 1534238 GraphRunner::Run can run before its constructor is finished

Audio/Video: cubeb:

bug 1481244 Crash in <name omitted> | mozilla::AudioStream::SetVolume

bug 1512445 Import Windows AudioIPC changes and enable in build (but leave disabled via pref)

bug 1524818 audioipc fails to build with nightly Rust: "error: use of deprecated item 'std::error::Error::cause': replaced by Error::source, which can support downcasting"

bug 1527659 Update cubeb from upstream to 3afc335

bug 1532645 Update cubeb-backend to workaround https://github.com/rust-lang/rust/issues/58881

Web Audio:

bug 1517324 start blocked AudioContext when it's MediaStreamAudioSourceNode starts

bug 1519562 AudioWorkletGlobalScope::RegisterProcessor: save descriptors in a map

bug 1524026 Web audio didn't produce sound even if AudioContext is resumed from blocked

bug 1524087 raptor-webaudio-firefox regression from clang 8

bug 1528876 enable AudioWorklet in Nightly Web Audio wpt

bug 1533911 windows/aarch64 - /webaudio/the-audio-api/the-audiobuffersourcenode-interface/sub-sample-buffer-stitching.html | X SNR (58.621820307137014 dB) is not greater than or equal to 85.586. Got 58.621820307137014. - assert_true: expected true got false

bug 1533912 windows/aarch64 - /webaudio/the-audio-api/the-audiobuffersourcenode-interface/sub-sample-scheduling.html | X With playbackRate 0.25: output0[18] is not close to 1.0499999999999998 within a relative error of 4.542e-8 (RelErr=0.07462636629740381).

bug 1535214 run AudioWorklet for realtime AudioContext on MSG thread

WebRTC:

bug 1231414 RTCPeerConnection.addTrack should not require a stream to be passed

bug 1515699 Remove webrtc ARM64 build workarounds after we switch to clang-cl

bug 1523795 Perma Assertion failure: get() (dereferencing a UniquePtr containing nullptr), at /builds/worker/workspace/build/src/obj-firefox/dist/include/mozilla/UniquePtr.h:302

bug 1525341 Jitter stat for WebRTC audio is always zero

bug 1526512 WebRTC RTCP stat roundTripTime is expressed in milliseconds not in (double) seconds

bug 1527526 AudioConduit is reporting RTP send statistics when it should be reporting RTP receive statistics

bug 1527633 Rename GetRTPStats to something more descriptive

bug 1530025 Video only stats functions should be moved off of MediaConduitInterface and onto VideoSessionConduit

WebRTC: Audio/Video:

bug 1213453 Implement MediaDeviceInfo.groupId

bug 1440601 Tabsharing exposed on Fennec Nightly. Disable it.

bug 1519535 Crash in CFPasteboardPromiseDataUsingBlock

bug 1520200 web.ciscospark.com sometimes displays a zoomed-in upper-left corner of the video feed

bug 1522238 Set frame timestamps in MediaPipeline

bug 1522488 WebRTC: cannot switch microphone to another one without page refresh

bug 1522773 Permafailing tier 2 Assertion failure: Request::mDisconnected, at /builds/worker/workspace/build/src/obj-firefox/dist/include/mozilla/MozPromise.h:429

bug 1523412 GetStreamCaps can fail with S_FALSE

bug 1523611 Firefox 65+ isn't enforcing >=libvpx-1.7.0 requirement

bug 1523817 Over 3s A/V sync delay when Firefox is the sender.

bug 1524145 No longer able to send stereo audio when setting fmtp stereo=1 in the sdp

bug 1524648 MediaEngineWebRTCMicrophoneSource::Deallocat asserts if another browser already has gUM on audio and video

bug 1525230 Video cropped when resolution changes

bug 1530488 Disable camera for aarch64 windows builds

bug 1535044 Missing mozilla namespace in TestGroupId.cpp

WebRTC: Networking:

bug 1490658 Support RTCIceCandidate.usernameFragment

bug 1494311 Make mtransport API more IPC friendly

bug 1510898 Disable SRTP Sha1_32 cipher in Nightly builds

bug 1521879 Use mtransport in a separate process, guarded by a pref

bug 1526477 Firefox failing to nominate ICE pairs when interoping with an ice-lite endpoint

bug 1528352 ICE failed with ICMP error (Windows only seems?)

bug 1530107 Crash in [@ mozilla::PeerConnectionImpl::PeerConnectionImpl]

bug 1530815 Crash in [@ r_assoc_fetch_bucket]

bug 979966 Need better r_log diagnostics when no candidate pairs exist for a stream

WebRTC: Signaling:

bug 1370562 PeerConnection stops functioning properly after receiving disabled m= line in SDP without a=inactive attribute

bug 1402912 Put multiple a=msid in SDP when a track is attached to multiple streams

bug 1508685 Local reoffer generated without mid when remote answer did not have mid

bug 1524642 RTCRtpTransceiver wpt has been quietly disabled

bug 1525397 Remove lsan suppressions from webrtc wpt

bug 1526733 Replace TCP/TLS/RTP/SAVP with correct DTLS value

bug 1528323 Make replaceTrack async even when transceiver is not associated

bug 1529403 webrtc wpt meta files need to have links to bugs when tests are expected to fail, timeout, or are disabled

bug 1529693 Figure out why RTCPeerConnection-createDataChannel.html is disabled on test-verify linux debug

bug 1529787 Renew SDP parser telemetry

bug 1530435 Preserve bug history of webrtc.sdp.parser_diff telemetry

bug 1531075 createOffer with no transceivers fails

bug 1531084 RTCPeerConnection.getStats needs to throw InvalidAccessError if a unique Receiver/Sender cannot be found

bug 1531094 RTCPeerConnection-getStats.html asserts the presence of "inbound-rtp" and "outbound-rtp" stats when no RTP streams exist

bug 1531103 RTCPeerConnection-onnegotiationneeded.html wpt has been quietly disabled

bug 1531143 RTCPeerConnection-setLocalDescription.html wpt has a bad renegotiation test-case

bug 1531144 Need to re-enable RTCPeerConnection-setRemoteDescription-answer.html wpt

bug 1531146 Need to re-enable RTCPeerConnection-setRemoteDescription-offer.html wpt

bug 1531148 RTCPeerConnection-setRemoteDescription-tracks.https.html wpt expects remote and local track ids to match

bug 1531156 RTCPeerConnection.currentRemote/LocalDescription can have the wrong type

bug 1531439 RTCPeerConnection-transceivers.https.html wpt expects remote and local track ids to match

bug 1531448 RTCPeerConnection.close() should stop the transceivers

bug 1531472 RTCRtpSender-replaceTrack.https.html wpt has been quietly disabled

bug 1531505 Firefox is parsing and using source-level msid

bug 1531811 Remove unused onremovestream from RTCPeerConnection.webidl

bug 1534607 webrtc web-platform-test meta subdirectories need bug links in them for disabled tests

bug 1534734 jsep-initial-offer.https.html is too strict

bug 1535100 Rework RTCDTMFSender-ontonechange tests to use cumulative time when checking whether tonechange events are happening too early

Intermittent Test failures:

bug 1412231 Intermittent Assertion failure: mAudioContextOperation == AudioContextOperation::Close (We should be reviving the graph?), at /builds/worker/workspace/build/src/dom/media/MediaStreamGraph.cpp:4051

bug 1518378 Intermittent Test Verify dom/media/webaudio/test/test_pannerNodeHRTFSymmetry.html | maxDifference: a, first bad index: b with test-data offset 0 and expected-data offset 0; corresponding values c and 0 --- differences - got d, expected +0 | Test timed out

bug 1518946 Intermittent browser/base/content/test/webrtc/browser_devices_get_user_media_queue_request.js | microphone selector hidden -

bug 1522535 Intermittent /webrtc/RTCPeerConnection-removeTrack.https.html | application crashed [@ libxul.so + 0xd1a437] (leaking the world)

bug 1533261 Intermittent PID 9884 | Assertion failure: gInstance, at z:/build/build/src/media/webrtc/signaling/src/peerconnection/PeerConnectionCtx.cpp:159

Web Platform Tests:

bug 1492316 [wpt-sync] Sync PR 13061 - Sub-sample accurate start for ABSN

bug 1498465 [wpt-sync] Sync PR 13477 - Compute azimuth correctly according to the spec

bug 1511573 [wpt-sync] Sync PR 14317 - MSID information change should trigger related track events

bug 1513504 [wpt-sync] Sync PR 14476 - Switch to new ICE state implementation

bug 1514670 [wpt-sync] Sync PR 14554 - Added tests for missing MID field in sdp

bug 1515979 [wpt-sync] Sync PR 14635 - Fix WebRTC test use of Resolver after PR 14417

bug 1517444 [wpt-sync] Sync PR 14699 - Wiring for webrtc DtlsTransport events and state

bug 1517740 [wpt-sync] Sync PR 14716 - Update tests that broke due to upstream changes of Resolver.

bug 1517942 [wpt-sync] Sync PR 14730 - Fixing web platform tests for PeerConnection.setRemoteDescription().

bug 1518754 [wpt-sync] Sync PR 14757 - RTCRtpReceiver.getSynchronizationSources() added.

bug 1526291 [wpt-sync] Sync PR 15139 - Handle role conflict in the standalone RTCIceTransport

bug 1526301 [wpt-sync] Sync PR 15005 - Update cached azimuth/elevation/cone gain

bug 1526316 [wpt-sync] Sync PR 15009 - MediaCapabilities: Add "transmission" type.

bug 1526399 [wpt-sync] Sync PR 15157 - Implement and ship RTCRtpEncodingParameters.scaleResolutionDownBy

bug 1526499 [wpt-sync] Sync PR 15132 - Add support for all RTCIceCandidate fields.

bug 1526532 [wpt-sync] Sync PR 14937 - Add RTCError, RTCErrorInit, RTCErrorDetailType and WPT coverage.

bug 1526599 [wpt-sync] Sync PR 15215 - Use oversampling to compute frame number

bug 1526647 [wpt-sync] Sync PR 15270 - Relax required SNR a bit

bug 1526801 [wpt-sync] Sync PR 15292 - [PeerConnection] Fire signalingstatechange event at the right time

bug 1526860 [wpt-sync] Sync PR 15214 - Round up to the next render quantum for suspend

bug 1527044 [wpt-sync] Sync PR 15133 - Deflake RTCPeerConnection-track-stats.https.html.

bug 1529821 [wpt-sync] Sync PR 15506 - Surface dtlsTransport via state-surfacer