Firefox 65 WebRTC/WebAudio Release Notes:

Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 65:

WebRTC and WebAudio bugs: Bugzilla search for WebRTC and WebAudio related bugs marked Fixed in Firefox 65

Noteworthy Changes:

• HTTP Proxy authentication for WebRTC TURN TCP is now supported bug 1203503

• Updated libwebrtc to version 64 bug 1376873

• Several stats got values got updated or deprecated to get closer to spec

Audio/Video: MediaStreamGraph:

bug 1423241 Remove MediaStreamListener

bug 1504020 [wpt-sync] Sync PR 13840 - Add support for resizeMode in MediaStreamTrack.getSettings()

bug 1504082 [wpt-sync] Sync PR 13845 - Add extra utilities to content::media_constraints

bug 1504087 [wpt-sync] Sync PR 13846 - Wire the resizeMode property to the constraints parsing mechanism.

bug 1504098 [wpt-sync] Sync PR 13848 - Add support for the resizeMode constraint in getUserMedia()

bug 1504385 [wpt-sync] Sync PR 13870 - Add support for resizeMode in [MediaStreamTrack|InputDeviceInfo].getCapabilities()

bug 1504769 [wpt-sync] Sync PR 13927 - MediaDevices is SecureContext

Audio/Video: Recording:

bug 1453078 Intermittent dom/media/test/test_mediarecorder_principals.html | assertion count 1 is more than expected 0 assertions

bug 1458538 [ MediaRecorder ] The Pause and resume events don't work in Firefox

bug 1496377 test_mediarecorder_state_transition.html is not being run in tests (and fails)

bug 1500210 [wpt-sync] Sync PR 13602 - WPT test for https://github.com/w3c/mediacapture-record/pull/152

bug 1503518 [wpt-sync] Sync PR 13805 - WebKit export of https://bugs.webkit.org/show_bug.cgi?id=190778

Audio/Video: cubeb:

bug 1500109 Crash in audiounit_create_unit

bug 1501148 Refactor AudioIPC to make way for multiple OS backends (Windows support)

bug 1501605 Update cubeb from upstream to 04d58b6

bug 1502165 Crash in audiounit_get_devices_of_type

bug 1503240 Print commits in update.sh script

bug 1504932 replace README_Mozilla with moz.yaml

Web Audio:

bug 1476514 Preparation for running AudioWorklet from MSG thread

bug 1501619 [wpt-sync] Sync PR 13698 - [run_web_tests] Check for extra baselines

bug 1502004 Move AudioWorkletGlobalScope from dom/worklet to dom/media/webaudio

bug 1503132 Run offline MSG thread even when not rendering

bug 1503236 Move WorkletImpl reference from WorkletGlobalScope to classes inheriting WorkletGlobalScope

bug 1503950 when more than two AudioParam events of the same type (e.g. setValueAtTime) are added at the same time, the latter events are ignored

bug 1504723 Fix typo in enum name for some BiquadFilter and WaveShaper nodes

bug 1504982 Add AutomationRate webidl definition

bug 1505726 [wpt-sync] Sync PR 13980 - [media] Treat cross-origin redirect as TAINTED only for no-cors requests

bug 1508671 Perma web platform /worklets/audio-worklet-csp.https.html when Gecko 65 merges to Beta on 2018-12-03

bug 1508905 Allow dom/media/webaudio/blink to be reformatted with clang-format

WebRTC:

bug 1227519 Establish deprecation date for DHE cipher suites in WebRTC

bug 1324788 Update RTCIceCandidateStats to spec

bug 1368816 VideoCaptureExternalTest Rotation gtest is disabled

bug 1376873 Update WebRTC code to webrtc.org stable branch 64

bug 1436993 [wpt-sync] PR 9434 - Rename RTCIceCandidate ufrag field to usernameFragment

bug 1450733 [wpt-sync] Sync PR 10271 - Bring RTCCertificate interface up to date with Candidate Recommendation

bug 1457129 Intermittent /webrtc/RTCPeerConnection-track-stats.https.html | RTCPeerConnection.getStats(receivingTrack) is the same as RTCRtpReceiver.getStats() - assert_true: expected true got false

bug 1470067 Remove PtrVector

bug 1487278 Intermittent PID 11358 | Assertion failure: transceiver->IsAssociated() (ICE candidate was gathered before the transceiver was associated! This should never happen.) at /builds/worker/workspace/build/src/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp

bug 1489040 WebRTC ICE candidate stats ipAddress needs to be renamed

bug 1503023 An invalid state error can arise when a PeerConnection is closed before its certificates are initialized

bug 1503363 Don't assume that all WINNT targets support sse2

bug 1503444 [wpt-sync] Sync PR 13798 - WebKit export of https://bugs.webkit.org/show_bug.cgi?id=191077

bug 1504383 [wpt-sync] Sync PR 13869 - Check for AudioContext to enable generating audio tracks with it

bug 1504496 [wpt-sync] Sync PR 13892 - Move peerIdentity test from webrtc-pc to webrtc-identity

bug 1504498 [wpt-sync] Sync PR 13893 - Rename generateOffer to generateDataChannelOffer and remove use of legacy optionsRename generateOffer to generateDataChannelOffer and remove use of legacy options

bug 1504515 [wpt-sync] Sync PR 13903 - Move RTCRtpTransceiver tests related to OfferToReceive legacy options to webrtc/legacy

bug 1504517 [wpt-sync] Sync PR 13904 - Remove unneeded setting of onaddstream

bug 1504520 [wpt-sync] Sync PR 13907 - Let sender sends some media data so that getStats produce some outbou…

bug 1504561 [wpt-sync] Sync PR 13910 - Fix typo in RTCRtpReceiver-getParameters.html

bug 1504604 [wpt-sync] Sync PR 13914 - Fix RTCRtpTransceiver direction tests

bug 1504616 [wpt-sync] Sync PR 13915 - webrtc: rename DTLSTransport.transport to .iceTransport

bug 1504629 [wpt-sync] Sync PR 13916 - Update replaceTrack after removeTrack tests

bug 1504855 [wpt-sync] Sync PR 13931 - Implement RTCQuicStream.write()

bug 1504933 [wpt-sync] Sync PR 13940 - Update RTCPeerConnection-setRemoteDescription-tracks.https.html as MediaStream has no constraint on the order of the track

bug 1505067 [wpt-sync] Sync PR 13950 - Fix and re-enable simplecall-no-ssrcs.https.html.

bug 1505731 [wpt-sync] Sync PR 13981 - Fix typo in RTCPeerConnection-setLocalDescription-offer.html

bug 1506644 -msse2 flag slips in again in non-x86 builds

bug 1507039 [wpt-sync] Sync PR 14043 - Remove display of stats by default in webrtc/get-stats.html

bug 1507064 [wpt-sync] Sync PR 14046 - Implement RTCQuicStream.waitForWriteBufferedAmountBelow()

bug 1507216 Crash in mozalloc_abort | abort | webrtc::internal::Call::~Call

bug 1507228 [wpt-sync] Sync PR 14054 - webrtc: add test for legacy default stream

bug 1507977 [wpt-sync] Sync PR 14098 - Implement RTCQuicStream.readInto()

bug 1508007 [wpt-sync] Sync PR 14103 - Implement RTCQuicStream.waitForReadable()

bug 1508224 [wpt-sync] Sync PR 14122 - Prevent timeout when remote stats are not implemented

bug 1512517 Update WebRTC stat deprecation warnings

bug 979649 RTCP timestamps transmitted from Windows XP have significant clock drift

WebRTC: Audio/Video:

bug 1274392 Make echoCancellation:false flip the other audio processing defaults to false

bug 1406941 Write unittest for configuring AudioConduit

bug 1411681 H.264 High 4.0 profile streams no longer display when using WebRTC (regression)

bug 1425277 Have a unified MediaDataEncoder API

bug 1475209 Get rid of EnumDevResolver in MediaDevices.cpp by using a promise/mozpromise/pledge solution directly.

bug 1482150 Expand CubebDeviceEnumerator to enumerate output devices

bug 1492479 Have MediaManager::GetUserMedia() return a MozPromise

bug 1497175 Replace all remaining uses of Pledge with MozPromise, and remove Pledge (cleanup)

bug 1497552 Remove support for 44100 Hz in dtmf_tone_generator

bug 1497577 Upstream code to detect zero size windows in desktop capture

bug 1497602 Enable DirectX screen capturer on Windows

bug 1497606 Remove disable_composition_ in screen_capturer_win_gdi

bug 1497951 Have VP8/VP9 decoder wrapper to detect change of stream content

bug 1497974 Upstream changes to jitter_buffer.cc

bug 1498205 Upstream or remove PlatformUIThread

bug 1502172 Crash in video capture on Win 10

bug 1502313 Adding video to Facebook audio call on Linux fails even if the permission is granted

bug 1502927 Remove MediaStream.currentTime

bug 1503536 Call ApplySettings in MediaEngineWebRTCMicrophoneSource::Start

bug 1505284 Use H264 MediaDataDecoder for webrtc calls

bug 1508677 Use VP8/VP9 MediaDataDecoder for webrtc call

bug 1509548 Reenable the start time assertion in StreamTracks.h

bug 1509842 Re-enable AGC by default

bug 1509994 Move code from video_engine from media/webrtc/trunk/webrtc to elsewhere in tree

WebRTC: Networking:

bug 1194010 ICE TCP might connect to any port

bug 1203503 ICE/TURN/TCP via an HTTP-Proxy does not support Authentication

bug 1494301 Write a single API surface for mtransport

bug 1494312 Make mtransport API entirely async

bug 1502766 Firefox 64 does not respect the RTCConfiguration iceTransportPolicy

bug 1505733 Gather DTLS versions used in WebRTC

bug 1507700 interop with chrome/mdns candidates is broken

WebRTC: Signaling:

bug 1456417 addIceCandidate without sdpMLineIndex incorrectly assumes sdpMLineIndex = 0

bug 1496245 RTCPeerConnection createOffer generates malformed sdp after rollback

bug 1498793 Accept a=msid without track id

bug 1504252 Remove dead code in signaling/src/media

bug 1507413 Only 32 transceivers active simultaneously