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Firefox 66 WebRTC/WebAudio Release Notes:

Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 66:

WebRTC and WebAudio bugs: Bugzilla search for WebRTC and WebAudio related bugs marked Fixed in Firefox 66

Noteworthy Changes:

  • Fixed regression: Datachannel are allowed again without SRTP DTLS extension bug 1510487

Audio/Video: GMP:

bug 1516669 Convert gmp-clearkey to use Chromium ContentDecryptionModule_10 interface

Audio/Video: MediaStreamGraph:

bug 1513638 DOMMediaStream::CountUnderlyingStreams, resolve a Promise while in stable state.

bug 1513973 Audio input latency, possibly in MediaStreamGraph

bug 1518834 Muting locally on freezes video

Audio/Video: Recording:

bug 1514016 [ MediaRecorder ] New Pause/resume events in 65 fire synchronously, which is web incompatible.

bug 1515032 Add test case for playing blob consisting of multiple file blobs

Audio/Video: cubeb:

bug 1521791 Update cubeb from upstream to 67d37c1

Web Audio:

bug 1501709 AudioWorkletGlobalScope::RegisterProcessor: check descriptors and convert them to an internal representation

bug 1511120 Turn on the pref "media.autoplay.block-webaudio" on Nightly

bug 1512737 Missing tests for HRTF

bug 1513722 Run AudioWorklet from offline MSG thread

bug 1513733 start blocked AudioContext when it's source media element starts

bug 1518352 remove unnecessary WeakPtr support from PannerNode

bug 1518994 Enable NEON in AudioNodeEngine on aarch64

bug 1519430 Don't resume the context on user interaction or AudioScheduledSourceNode.start if the context was explicitely suspended

bug 1520021 clang-cl should not use AudioNodeEngine's NEON workarounds

bug 1520457 Adjust the message written in the console when the auto-play policy block an AudioContext


bug 1328194 Remove legacy PeerConnection.getStats and associated legacy stats type

bug 1347070 Add qpSum to local outbound RTCRTPStreamStats

bug 1421724 browser_devices_get_user_media_screen.js consistently timing out in ccov builds

bug 1486038 fix webrtc compilation errors on aarch64 windows

bug 1495446 In getStats(), RTCP timestamps have the wrong epoch

bug 1515205 Peer sees choppy motion/low frameRate in 1-1 call with Google Meet (regression)

bug 1515379 mFramesDeliveredToEncoder stat is should be initialized in VideoConduit

bug 1515548 Crash [@ webrtc::DesktopCaptureImpl::Run ] provoked by {frameRate: {max: 0}} (divide by zero)

bug 1517731 Enable mochitests for maxRetransmits and maxPacketLifeTime

bug 1518735 Make WebRTC PeerConnection stats mochitest easier to edit

bug 1519415 Perma-failing tier2 dom/media/tests/mochitest/test_getUserMedia_permission.html | Test timed out.

bug 813063 Missing LICENSE files etc.

WebRTC: Audio/Video:

bug 1321221 Implement getDisplayMedia for screen capture to comply with spec changes

bug 1371741 Disallow getUserMedia on nullprincipals (sandboxed iframes, top-level data urls).

bug 1439997 Switch OS X video capture to new version of code

bug 1474376 Make mediaSource legacy constraint values "screen" and "window" mean the same

bug 1497573 Remove DesktopCapture::Stop

bug 1497610 Upstream IsNewerOrSameTimestamp

bug 1497619 Restore thread check in

bug 1497992 Upstream or remove VideoReceiver::Reset

bug 1498253 Remove _current_sync_offset from channel.h

bug 1512280 Minor cleanups in MediaManager: shorten MediaManager::GetUserMedia(), better LOG macros

bug 1512459 Remove webrtc sndio audio device

bug 1514241 Chromium-specific code in HTMLMediaElement-captureStream WPT

bug 1515068 Assertion failure: mSrcStreamPausedGraphTime == GRAPH_TIME_MAX, at /builds/worker/workspace/build/src/dom/html/HTMLMediaElement.cpp:4665

bug 1515527 Log why GMPVideoEncoderParent::Encode fails

bug 1515873 Crash in mozilla::MediaManager::GetBackend

bug 1517681 Fix wpt MediaStream-default-feature-policy.https.html to comply with getUserMedia spec

bug 1518106 WebRTC microphone input not working (regression)

WebRTC: Networking:

bug 1510487 DTLS without SRTP extension (for datachannel only) closes connection

WebRTC: Signaling:

bug 1502899 Assertion failure: false, at /builds/worker/workspace/build/src/media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp:813

bug 1520289 Replace TCP/TLS/RTP/SAVPF with correct DTLS value

Intermittent Test failures:

bug 1517710 Intermittent PROCESS-CRASH | Main app process exited normally | application crashed [@ mozilla::MediaStreamGraphImpl::AppendMessage(mozilla::UniquePtr<mozilla::ControlMessage, mozilla::DefaultDelete<mozilla::ControlMessage> >)]

bug 1517711 Intermittent /feature-policy/reporting/microphone-report-only.https.html | application crashed [@ mozilla::detail::MutexImpl::lock()]

Web Platform Tests:

bug 1503351 [wpt-sync] Sync PR 13788 - Add RTCPeerConnection.connectionState and onconnectionstate.

bug 1507775 [wpt-sync] Sync PR 14089 - Remove timeout in async_test for mediacapture-streams tests

bug 1507797 [wpt-sync] Sync PR 14091 - Test getUserMedia non-applicable constraints are ignored

bug 1509173 [wpt-sync] Sync PR 14170 - webrtc-wpt: use addTrack(track, stream) to increase firefox compat

bug 1509239 [wpt-sync] Sync PR 14175 - Fix bad merge 'Merge branch 'master' into gecko/1498793'.

bug 1509310 [wpt-sync] Sync PR 14181 - Media Capabilities: switch MediaCapabilitiesInfo to a dictionary.

bug 1509602 [wpt-sync] Sync PR 14216 - Media Capabilities: implement Blink shell of encrypted media support.

bug 1509772 [wpt-sync] Sync PR 14227 - Remove the timeout in async_test for webrtc and xhr tests

bug 1511578 [wpt-sync] Sync PR 14319 - web platform tests for new networkPriority encoding parameter.

bug 1511855 [wpt-sync] Sync PR 14341 - Create RTCDtlsTransport objects in the blink layer

bug 1512176 [wpt-sync] Sync PR 14378 - Implement RTCRtpReceiver.getParameters()

bug 1512414 [wpt-sync] Sync PR 14393 - Add WPT tests for correct parsing of msid

bug 1513230 [wpt-sync] Sync PR 14462 - Reland "Create RTCDtlsTransport objects in the blink layer"

bug 1514125 [wpt-sync] Sync PR 14516 - WebKit export of

bug 1514432 [wpt-sync] Sync PR 14530 - WebKit export of