Media/WebRTC/ReleaseNotes/64

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Firefox 64 WebRTC/WebAudio Release Notes:

Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 64:

WebRTC and WebAudio bugs: Bugzilla search for WebRTC and WebAudio related bugs marked Fixed in Firefox 64

Noteworthy Changes:

  • scaleResolutionDownBy and maxBitrate can now be updated live on a connected PeerConnection bug 1253499
  • Enable automatic gain control (AGC) by default now in bug 1496714
  • SRTP is now using AES CPU instructions (through using NSS instead of build in libsrtp ciphers) and offers AEAD_AES_128_GCM and AEAD_AES_256_GCM through bug 1479665 and some follow up bugs
  • Datachannels should get more throughput in most scenarios through bug 1051685

Audio/Video: MediaStreamGraph:

bug 1258143 Remove LocalMediaStream (and its Stop()) from js

bug 1492627 [wpt-sync] Sync PR 13084 - WebKit export of https://bugs.webkit.org/show_bug.cgi?id=189516

bug 1493565 [wpt-sync] Sync PR 13181 - Make use of navigator.mediaDevices.getUserMedia instead of navigator.getUserMedia

bug 1495935 [wpt-sync] Sync PR 13321 - Include ended tracks when cloning MediaStreams.

bug 1498058 [wpt-sync] Sync PR 13361 - Fix media element field behavior when playing MediaStreams

Audio/Video: Recording:

bug 1482346 [wpt-sync] Sync PR 12401 - Upstream image_capture tests to WPT

bug 1487948 [wpt-sync] Sync PR 12790 - [Image Capture] Add focusDistance constraint.

bug 1489417 [wpt-sync] Sync PR 12893 - Add tests for mediacapture-image

bug 1490689 [wpt-sync] Sync PR 12967 - [Image Capture] Add exposureTime constraint.

bug 1496383 MediaRecorder state error cases does not match its W3C spec

Audio/Video: cubeb:

bug 1489052 Can't get audio to work over BT with Bluetooth headset

bug 1491152 2018 MacBook Pro sound fails after waking up from sleep

bug 1498519 Update cubeb from upstream to 4559815

bug 1500377 Update cubeb to a68892d and cubeb-pulse-rs to 100b858

bug 1500468 Enumerate device is broken in Linux

Web Audio:

bug 1473467 implement AudioWorkletGlobalScope::RegisterProcessor()

bug 1485198 [wpt-sync] Sync PR 12606 - Update audit.js and fix error catching logic in should().throw()

bug 1487963 PannerNode should throw when parameters are out of range

bug 1488242 Throw the correct error type in {ConstantSourceNode,AudioBufferSourceNode}.{Start,Stop}

bug 1488586 [wpt-sync] Sync PR 12834 - Throw errors for invalid rolloffFactor and coneOuterGain

bug 1488630 [wpt-sync] Sync PR 12837 - Fix cases where setValueCurveAtTime was not throwing errors

bug 1488949 [wpt-sync] Sync PR 12857 - Change detune min/max limits

bug 1489338 [wpt-sync] Sync PR 12887 - Slightly improve AudioBuffer resampling

bug 1490103 [wpt-sync] Sync PR 12938 - Honor given outputChannelCount for AudioWorkletNodeOptions

bug 1493779 AddressSanitizer: ILL /builds/worker/workspace/build/src/xpcom/base/nsDebugImpl.cpp:628:3 in NS_ABORT_OOM(unsigned long)

bug 1495582 [wpt-sync] Sync PR 13296 - ConvolverNode buffer can be set multiple times

bug 1496496 Sound indicator incorrectly shows when playing a silent web audio

bug 1497112 detect and optimize when AudioParam stream inputs are null

bug 1497757 [wpt-sync] Sync PR 13445 - Allow posting a SharedArrayBuffer to AudioWorklet

bug 1500238 StereoPanner does not handle a pan value of zero for mono signals

bug 1500303 apply input gain correctly for stereo-to-stereo StereoPanner

WebRTC:

bug 1253499 scaleResolutionDownBy and maxBitrate don't update live.

bug 1468451 Crash near null [@ mozilla::PeerConnectionMedia::AddTransceiver]

bug 1479632 Intermittent dom/media/tests/mochitest/test_peerConnection_stats.html | inbound-rtp.pliCount is a sane number for a short test. value=100

bug 1486693 [wpt-sync] Sync PR 12715 - WebKit export of https://bugs.webkit.org/show_bug.cgi?id=189040

bug 1487256 [wpt-sync] Sync PR 12750 - Add RTCQuicStream IDL + binding skeleton

bug 1487585 [wpt-sync] Sync PR 12771 - RTCQuicTransport: start() implementation

bug 1487848 [wpt-sync] Sync PR 12782 - webrtc: throw SyntaxError on {iceServers: []}

bug 1488832 Assertion failure: mInitDone, at /builds/worker/workspace/build/src/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp:190

bug 1489033 WebRTC local-candidate and remote-candidate stats need coverage

bug 1489487 [wpt-sync] Sync PR 12895 - Implement DTMF ToneBuffer in the blink layer

bug 1489623 Spec change: throw SyntaxError on RTCIceServer with no urls

bug 1490700 Divide-by-zero in [@webrtc::I420Buffer::CropAndScaleFrom]

bug 1491128 Add comment block to dom/webidl/RTCDTMFToneChangeEvent.webidl

bug 1493012 [wpt-sync] Sync PR 13123 - Add allowance for race in DTMF ontonechange events

bug 1494648 [wpt-sync] Sync PR 13241 - [Unified Plan] Remote MediaStreamTracks should be muted by default.

bug 1495477 [wpt-sync] Sync PR 13287 - Implement RTCQuicTransport.onquicstream and stream reset/finish

bug 1495626 [wpt-sync] Sync PR 13298 - Remove invalid RTCPeerConnection.addTransceiver() tests

bug 1495968 [wpt-sync] Sync PR 13322 - Remove and fix non-spec compliant WebRTC tests

bug 1495976 [wpt-sync] Sync PR 13323 - Reland: Implement RTCIceTransport.onselectedcandidatepairchange

bug 1498237 Intermittent dom/media/tests/mochitest/identity/test_fingerprints.html | Test timed out.

bug 1498679 Perma tier2 dom/media/tests/mochitest/test_setSinkId.html | Never enter here, this must fail| wpt/setSinkId.html | setSinkId fails with NotFoundError on made up deviceid - assert_unreached: Should have rejected: undefined Reached unreachable code

WebRTC: Audio/Video:

bug 1377146 Remove AudioStreamTrack and VideoStreamTrack

bug 1404992 Rework VideoConduit

bug 1479051 [macOS 10.14] WebRTC sites silently fail if user previously clicked "Don't Allow" for Firefox camera/mic access

bug 1479840 Test that enumerateDevices() neither resolves nor rejects when navigated away.

bug 1479841 Start replacing uses of Pledge with MozPromise in MediaManager (cleanup)

bug 1481152 Restrict number of concurrent uses of an audio input device in a child process

bug 1487057 Finish cleaning up audio input devices code

bug 1487419 Share screen doesn't work on Mac with getUserMedia Test Page with multiple monitors

bug 1489757 Changeset 549f0b8075d5 causes video streams to take a very long time to recover from packet loss

bug 1494498 Fix constraints logging.

bug 1494806 Exact constraints containing string arrays, e.g. {deviceId: {exact:['id']}} are treated as ideal.

bug 1495478 Intermittent /builds/worker/workspace/build/src/dom/media/webrtc/MediaTrackConstraints.cpp:484:3: error: use of undeclared identifier 'LogConstraints'

bug 1496529 [webrtc]H264 video decoding cause a crash/failure when using playback decoder.

bug 1496714 Consider enabling AGC by default

bug 1497254 Remove the concept of an Allocation from MediaEngineWebRTCAudio

bug 1497351 Do without new null defaults for dictionary-typed members in MediaStreamTrack.webidl

bug 1497390 Remove support for legacy mozAutoGainControl and mozNoiseSuppression constraints.

WebRTC: Networking:

bug 1051685 WebRTC data channels always use the default SCTP window size of 128K

bug 1435789 Deprecate RTCIceCandidateStats.mozLocalTransport and add protocol and relayProtocol

bug 1479665 Update libsrtp to 2.2.0-pre

bug 1480869 Stop using SRTP cipher suites from NSS

bug 1485883 SRTP extension using NSS extension handlers

bug 1486012 re-implement ICE restart as the creation of new ICE streams on the pre-existing context

bug 1491511 Add Telemetry for SRTP cipher usage in WebRTC

bug 1492834 Remove "Attempting to protect RTP" and related log messages

bug 1493146 Lengthen ice-pwd

bug 1498068 SRTP_AEAD_AES_128_GCM failed to connect

WebRTC: Signaling:

bug 1492248 UBSan: undefined-behavior media/webrtc/signaling/src/sdp/sipcc/sdp_token.c:1803:13

bug 1493765 Stop using NrIce* stuff in PeerConnectionImpl

bug 1495160 WEBRTC_DATACHANNEL_NEGOTIATED Telemetry broken since 59

bug 1495569 SDP offers without a=mid get rejected after creating an answer (Local descriptions must have a=mid attributes)

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