Firefox 59 WebRTC/WebAudio Release Notes:

Full listing of all WebRTC & WebAudio bugs marked as Fixed in Firefox 59:

WebRTC and WebAudio bugs: Bugzilla search for WebRTC and WebAudio related bugs marked Fixed in Firefox 59

Noteworthy Changes:

• Transceivers are now available in Firefox bug 1290948

• getContributingSources is now available bug 1363667

• Implemented muted/onmute/onunmute bug 1208378

• Firefox on Windows is no longer limited to 16 audio stream bug 1397793

Bug tickets fixed in Firefox 59 that affect WebRTC or Web Audio (full list):

Audio/Video: GMP:

bug 1404230 [EME] Support HDCP Policy check on MediaKeys

bug 1411205 [EME] Test case for Bug 1404230 HDCP Policy check on MediaKeys

bug 1415466 [EME] Update content_decryption_module.h and content_decryption_module_ext.h for CDM version 1.4.9.xxxx

bug 1416667 Use MOZ_CRASH_UNSAFE_PRINTF in GMPChild::ProcessingError to take aReason into account.

bug 1416686 Reduce the uses of IPC_FAIL_NO_REASON in GMPChild.cpp

bug 1417297 Convert gmp-clearkey to use Chromium ContentDecryptionModule_9 interface

bug 1417332 Convert CDM Error to CDM Exception when we got OnLegacySessionError from CDM

bug 1422669 Restore librlz from Bug 1332530 for calculating the machine ID.

bug 1422856 Stop using GetNativePath in GMPServiceParent

bug 1430517 Remove VIDEO_CHROMIUM_CDM_MAX_SHMEMS telemetry

Audio/Video: MediaStreamGraph:

bug 1406350 Create a loopback devices sink to source for automated testing on Linux/Pulse

bug 1415556 Clarify MediaStreamGraph code thread usage

bug 1419378 Return failure in AudioCallbackDriver::Init when out channels is 0

bug 1425623 Avoid heap memory allocation on MSG::UpdateGraph

Audio/Video: Recording:

bug 1403307 Intermittent dom/media/test/test_mediarecorder_pause_resume_video.html | Last frame should be green

bug 1424191 MediaRecorder(videoStream) unexpectedly stops itself

bug 1429765 Extend lifetime of MEDIA_RECORDER_RECORDING_DURATION, MEDIA_RECORDER_TRACK_ENCODER_INIT_TIMEOUT_TYPE, and SCALARS_MEDIARECORDER.RECORDING_COUNT telemetry probes

Audio/Video: cubeb:

bug 1370598 Don't cap latency at 512 frames on Macs that are not Macbooks or Macbook Air.

bug 1405877 Cubeb audioipc requires a named Unix-domain socket

bug 1420930 Update cubeb from upstream to 8a0a300

bug 1421267 Update cubeb from upstream to e17ba01

bug 1423901 Update cubeb from upstream to a88bf02

bug 1426174 Crash in `anonymous namespace::wasapi_stream_init

bug 1426791 All OSX tests are going to permacrash when Gecko 59 merges to Beta on 2018-01-11

bug 1426867 Crash in mozilla::CubebUtils::GetCubebContextUnlocked

bug 1427150 When building against pulseaudio >= 2.0, the resulting build does a buffer overflow read with pulseaudio < 2.0

bug 1427702 Update cubeb from upstream to bda37c2

bug 1429666 cubeb_resampler_speex calls data callback while draining

bug 1430870 Avoid static ctors in AudioIPC startup code

bug 1430996 Remove NIGHTLY_BUILD restriction to running cubeb-pulse-rs

bug 1431333 Cubeb logging does not work with cubeb-sandox on

Web Audio:

bug 1339889 Intermittent dom/media/webaudio/test/test_mediaElementAudioSourceNodeFidelity.html | Found unexpected noise during analysis.

bug 1424906 PeriodicWave disableNormalization false is incorrect

WebRTC:

bug 1363667 Add getContributingSources and getSynchronizationSources to RTCRtpReceiver

bug 1406135 Can't subscribe to large amount of webRTC streams (regression)

bug 1414167 Annotate SDPs on about:webrtc with offer and answer

bug 1414169 Show received ICE candidates on about:webrtc

bug 1416932 Add tests to detect inclusion of negotiated RTP header extensions in RTP packets

bug 1418522 Show unmatched candidates on about:webrtc

bug 1419093 Track RTC RTP source objects interface to dictionary spec change

bug 1421819 Only create webrtc::call() object on video calls

bug 1421830 test_peerConnection_scaleResolution.html sometimes generates log chatter until shutdown

bug 1421958 OfferToReceiveVideo and OfferToReceiveAudio should be of type Boolean not Long

bug 1424318 Crash in webrtc::FloatS16ToFloat

bug 1424342 WebRTC crashes in random places on Win

bug 1426130 trickle_caption_msg in about:webrtc is not properly localizable

bug 1426323 Media: WebRTC: Fix build config for MIPS64

bug 1426678 Assertion failure: mRawPtr != nullptr (You can't dereference a NULL RefPtr with operator*().), at /builds/worker/workspace/build/src/obj-firefox/dist/include/mozilla/RefPtr.h:370

bug 1427009 Crash in mozalloc_abort | abort | webrtc::StreamId::Set

bug 1429085 Assertion failure: false, at /builds/worker/workspace/build/src/media/webrtc/signaling/src/peerconnection/PeerConnectionMedia.cpp:439

bug 1429536 Assertion failure: !(aWidth&1), at /home/worker/workspace/build/src/dom/media/webrtc/MediaEngineDefault.cpp:145

bug 1430213 Pref toggle for RTCRtpReceiver getContributingSources and getSynchronizationSources APIs

WebRTC: Audio/Video:

bug 1208378 Implement MediaStreamTrack.muted/onmute/onunmute

bug 1372073 Neutralize the threat of fingerprinting of media devices API when 'privacy.resistFingerprinting' is true

bug 1376276 firefox53 cannot limit framerate in getusermedia with screen

bug 1388219 Support down-scaling per track in getUserMedia

bug 1388667 NormalizedConstraints doesn't work properly in multiple content processes case

bug 1397793 Remove webrtc.org "External" audio interface and switch gUM audio input to APM

bug 1399413 Multiple content process getUserMedia is rejected (all platforms except OSX)

bug 1404997 Move Opus decoding/NetEQ out of the MSG thread, it’s too expensive

bug 1406935 Write unittest for configuring VideoConduit

bug 1406936 Write unittest for re-configuring VideoConduit

bug 1406937 Write unittest for the video-encode path through VideoConduit

bug 1411739 MediaManager.cpp uses HostIsHttps instead of window.isSecureContext

bug 1411742 Remove unused prefs and member variables in media.getusermedia

bug 1412394 near perma fail dom/media/tests/crashtests/1367930_2.html | crash

bug 1418165 Protect sVideoCaptureThread with sThreadMonitor

bug 1418331 Crash in mozilla::camera::VideoEngine::CreateVideoCapture

bug 1418367 Webrtc web-platform-tests are disabled because number of concurrent AudioSessionConduits is very limited on windows and linux32

bug 1418694 WebRTC microphone input not working (regression)

bug 1418871 VideoCapture thread hangs due to deadlock in shutdown

bug 1420162 Remove USE_GRAPH_RATE because it's the default now, and we don't support anything else

bug 1420585 getUserMedia hangs when constraints can't be met

bug 1421706 recent regression - quick build up of latency on webrtc calls using Firefox Nightly

bug 1422875 fake:true constraint should not affect screen sharing (needed for testing screenshare+audio)

bug 1423228 Prevent using non-fake devices when testing screen-sharing

bug 1423515 Failed to send a devicechange event for permanent permission case (no live stream)

bug 1423819 Assertion failure: pthread_mutex_destroy(&mMutex) == 0 (pthread_mutex_destroy failed), at /builds/worker/workspace/build/src/xpcom/threads/RecursiveMutex.cpp:63

bug 1423893 Crash in webrtc::AudioBuffer::CopyFrom

bug 1423920 Crash in arena_t::DallocSmall | arena_dalloc | `anonymous namespace::wasapi_device_collection_destroy

bug 1423923 Properly feed reverse stream to the AudioProcessingModule

bug 1423929 Crash in webrtc::SincResampler::Resample

bug 1423930 Crash in webrtc::WebRtcAec_BufferFarend

bug 1423953 Crash in arena_t::MallocSmall | moz_xmalloc | nsTArray_base<T>::EnsureCapacity<T> | nsTArray_Impl<T>::AppendElements<T> | mozilla::MediaSegmentBase<T>::AppendChunk

bug 1423954 Crash in mozilla::SourceMediaStream::AppendToTrack

bug 1424660 Crash in mozilla::MediaEngineWebRTCMicrophoneSource::PacketizeAndProcess

bug 1425596 Stop busy looping in mFakeAudioDevice

bug 1425631 Use common SharedThreadPool across WebRTC

bug 1425904 Crash in RtlRaiseStatus | RtlpUnWaitCriticalSection | RtlLeaveCriticalSection | <T>::operator() | mozilla::MozPromise<T>::InvokeCallbackMethod<T>

bug 1426123 Avoid spurious divide-by-zero warning from coverity for SelectSendResolution()

bug 1426171 Potential crash if GraphRate is greater than 48kHz in WebrtcAudioConduit::GetAudioFrame

bug 1426486 Make GetInputStream()->AsSourceStream() invariant

bug 1428098 Don't reconfig the video encoder stack manually

bug 1428390 Risk for shutdown deadlock in CamerasChild

bug 1428392 Remove AudioOutputObserver

bug 1429219 Enforce providing simulcast encodings with enough bits to avoid encoder Init failure

bug 1430931 appear.in hits MOZ_CRASH(ArrayBufferInputStream not thread-safe)

WebRTC: Networking:

bug 1230759 Update libsrtp to version 2.2.0-pre

bug 1297418 Update sctp library from upstream

bug 1426059 Remove unused code in mtransport

WebRTC: Signaling:

bug 1290948 Implement RTCRtpTransceiver and pc.addTransceiver

bug 1400363 Update muted state on tracks when negotiation happens

bug 1404686 Crash - WebRtc - Null Pointer dereference in nsWrapperCache::HasWrapperFlag

bug 1421965 Crash in mozilla::MediaPipeline::MediaPipeline

bug 1422215 WebRTC - Use After Free in in JsepSessionImpl::CheckNegotiationNeeded()

bug 1423842 onaddstream changed behaviour with transceivers

bug 1425621 Lost ability to detect remote track removal (Remove remote tracks from their streams when negotiated away)

bug 1425697 Data Channel remote maximum message size slightly incorrect

bug 1425873 addTransceiver(<string>, {streams: [stream]) should fire ontrack with stream in streams argument.

bug 1425901 Use nsITimerCallback for DTMF timers

bug 1425956 Removing a track and later re-adding it to a peer connection causes InvalidSessionDescriptionError

bug 1425996 Various builds will be busted when Gecko 59 merges to Beta on 2018-01-11

bug 1427745 Enable ESLint rule mozilla/use-services for dom/media

bug 1430707 Hit MOZ_CRASH() at PeerConnectionMedia.cpp:490