Media/WebRTC/ReleaseNotes/65
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