Thu Mar 28 2024 07:37:22 PDT
  • Target Milestone: mozilla53
  • Component: Audio/Video: cubeb, Audio/Video: GMP, Audio/Video: MediaStreamGraph, Audio/Video: Recording, WebRTC, WebRTC: Audio/Video, WebRTC: Networking, WebRTC: Signaling
  • Product: Core
  • Status: RESOLVED, VERIFIED, CLOSED
  • Resolution: FIXED

115 bugs found.
ID Comp Updated Pri Rank Assignee Status Summary
1221574 Audio/Video: cubeb 2017-01-24 P1 17 achronop RESO Write a full-duplex Android OpenSL ES cubeb backend
1317234 Audio/Video: cubeb 2017-02-23 P1 15 achronop RESO audiounit_stream_init() sometimes gets stuck forever on O...
1326176 Audio/Video: cubeb 2017-04-27 P1 15 achronop RESO Crash in jemalloc_crash | arena_dalloc_small | je_free | ...
1328284 Audio/Video: cubeb 2017-04-05 P1 19 achronop RESO Crash after removing HDMI cable [@ libsystem_kernel.dylib...
1310355 WebRTC 2016-12-29 P1 19 andrew.swan RESO Improve resiliency for using webrtc permission hooks
1317501 Audio/Video: MediaSt 2019-03-31 P1 15 apehrson RESO MediaStreamGraph processes runnables at unsafe time
1321235 Audio/Video: MediaSt 2019-03-31 P1 15 apehrson RESO Can not remove a stopped media track using removeTrack on...
1329075 Audio/Video: MediaSt 2017-01-12 P1 17 apehrson RESO Null-deref in [@ HTMLMediaElement::StreamCaptureTrackSour...
1273372 Audio/Video: GMP 2017-01-13 P1 19 bobowencode RESO [EME] Crash in mozilla::gmp::GMPChild::ProcessingError
1317670 WebRTC 2017-05-19 P1 10 bonchiang RESO ref_fuzz5 crashes Nightly in mozilla::MediaManager::OnNav...
1322305 Audio/Video: GMP 2017-08-28 P1 chris RESO Use-after-free in GMP process if GMPDecryptorChild actor ...
1271681 WebRTC: Signaling 2016-12-29 P1 17 dminor RESO Move SDP-related test cases from signaling_unittests to s...
1271682 WebRTC: Signaling 2017-01-04 P1 17 dminor RESO Move JSEP-related tests from signaling_unittests to jsep_...
1316888 WebRTC: Signaling 2016-11-16 P1 15 dminor RESO Port sdp_unittest to xul gtest
1317009 WebRTC: Signaling 2017-01-25 P1 17 dminor RESO Port jsep_session_unittest and jsep_track_unittest to xul...
1317714 WebRTC: Audio/Video 2016-12-10 P1 17 dminor RESO port mediaconduit_unittests to xul gtest
1317726 WebRTC: Signaling 2016-11-25 P1 17 dminor RESO sdp_file_parser still depends upon xpcom glue
1322707 WebRTC: Signaling 2017-01-24 P1 17 dminor RESO Stop building signaling_unittest and mediapipeline_unittests
1329922 WebRTC: Audio/Video 2017-01-18 P1 19 dminor RESO [DTMF] Tones are not heard when duration is set to lowest...
1266667 WebRTC: Networking 2017-01-10 P1 19 drno RESO [e10s] active ICE TCP fails because multiple connections ...
1319268 WebRTC 2017-01-19 P1 15 drno RESO Extend WebRTC ICE Telemetry probes
1320150 WebRTC: Networking 2016-11-30 P1 drno RESO ICE consent signals connected too earlier for non bundled...
1324995 WebRTC: Networking 2017-01-13 P1 15 drno RESO Crash in jemalloc_crash | je_free | r_free | stun_get_win...
1329976 WebRTC: Audio/Video 2017-01-20 P1 13 florian RESO getUserMedia(audio, video) when already capturing audio f...
1277037 WebRTC: Audio/Video 2019-12-18 P1 15 giles RESO MOZ_CRASH: Could not start cubeb stream for MSG.
1317764 WebRTC: Signaling 2016-12-29 P1 15 jbeich RESO --disable-tests fails to build: media/webrtc/signaling/fu...
1313966 WebRTC 2024-03-06 P1 15 jib RESO RTCSessionDescription interface doesn't match spec
1319542 WebRTC 2019-01-07 P1 15 jib RESO Update pc.createDataChannel's RTCDataChannelInit dict to ...
1314514 Audio/Video: cubeb 2018-01-24 P1 15 kinetik RESO gtestify the cubeb unit tests
1319623 Audio/Video: cubeb 2019-04-21 P1 15 kinetik RESO Valgrind reports uninitialized memory use in pulse_stream...
1322546 WebRTC: Networking 2016-12-21 P1 20 m_kato RESO Cannot compile nrappkit with WINVER=0x0600 or later
1322503 WebRTC 2017-01-17 P1 15 na-g RESO Firefox's RTCStatsType is not spec-compatible (missing hy...
1328440 WebRTC 2017-01-10 P1 20 na-g RESO Legacy PeerConnection.getStats should return a legacy sta...
1329762 WebRTC 2017-01-12 P1 15 na-g RESO Strengthen deprecation warning of legacy PeerConnection.g...
1318619 Audio/Video: cubeb 2017-04-10 P1 15 nobody RESO Update cubeb from upstream to 7f74039f92
1302231 Audio/Video: MediaSt 2017-08-28 P1 10 padenot RESO Crash in mozilla::MediaStreamGraph::NotifyOutputData sinc...
1310224 Audio/Video: cubeb 2017-09-05 P1 15 padenot RESO OOM crash in output-only scenario on Windows/WASAPI
1331869 Audio/Video: cubeb 2018-02-20 P1 15 padenot RESO Update cubeb from upstream to d96e35f02d
1328142 WebRTC: Signaling 2017-03-13 P1 12 paulrkerr RESO Webrtc.org 49 update broke simulcast
1326311 Audio/Video: Recordi 2017-03-05 P1 15 pierre.brochard.1982 RESO The Media Recorder API crash when we do a lot of stop/sta...
1250356 WebRTC 2021-10-21 P1 10 rjesup RESO Update WebRTC code to webrtc.org stable branch 49
1308481 WebRTC 2020-07-21 P1 15 rjesup RESO TIAS bitrate limitation does not work when resolution cha...
1319566 WebRTC: Audio/Video 2020-07-21 P1 15 rjesup RESO Crash in nsTArray_Impl<T>::DestructRange | nsTArray_Impl<...
1330091 WebRTC 2017-01-16 P1 13 rjesup RESO Renegotiation doesn't actually change the codec configura...
1330212 Audio/Video: MediaSt 2018-04-07 P1 15 rjesup RESO Intermittent dom/media/tests/mochitest/test_getUserMedia_...
1330318 WebRTC: Audio/Video 2017-01-18 P1 17 rjesup RESO Setting b=TIAS caps us at 2kbps
1331158 WebRTC 2017-01-18 P1 13 rjesup RESO Renegotiation doesn't actually change the receive codec c...
1332139 WebRTC: Audio/Video 2017-01-23 P1 17 rjesup RESO Drop ifdefs in webrtc vp9 interface code for handling old...
1329594 Audio/Video: cubeb 2017-04-05 P2 23 achronop RESO Crash in cubeb_resampler_destroy
1231848 Audio/Video: Recordi 2017-05-04 P2 22 apehrson RESO CanvasStream + MediaRecorder does not create variable fra...
1305949 Audio/Video: MediaSt 2016-12-30 P2 25 apehrson RESO Do some cleaning around direct listeners and video sinks
1314886 Audio/Video: MediaSt 2019-03-31 P2 25 apehrson RESO Intermittent dom/media/test/test_streams_element_capture_...
1319445 Audio/Video: MediaSt 2017-10-11 P2 25 apehrson RESO Disable direct audio listeners for RTCPeerConnection with...
1323079 WebRTC 2017-12-03 P2 27 apehrson RESO Intermittent dom/media/tests/mochitest/test_peerConnectio...
1326386 WebRTC: Audio/Video 2016-12-30 P2 25 apehrson RESO webrtc.org 49 update mismerged away a mochitest
1330676 Audio/Video: Recordi 2017-01-16 P2 25 apehrson RESO MediaRecorder's CBR setting causes really bad perceived v...
1330918 Audio/Video: Recordi 2017-01-24 P2 25 apehrson RESO Make MediaRecorder use timestamps for video
1330919 Audio/Video: MediaSt 2017-01-19 P2 25 apehrson RESO Set proper timestamps on video frames from canvas.capture...
1332584 Audio/Video: Recordi 2017-01-27 P2 25 apehrson RESO MediaRecorder doesn't record the last frame of a video track
1313758 WebRTC: Audio/Video 2016-12-02 P2 21 bonchiang RESO WebRTC getUserMedia mediaSource 'browser' broken: Cause: ...
1329562 WebRTC: Audio/Video 2017-03-14 P2 25 cpeterson RESO Remove WebRTC checks for Windows Vista
1307461 WebRTC: Signaling 2017-02-14 P2 25 dminor RESO Intermittent mediapipeline_unittest | test failed with re...
1316886 WebRTC: Signaling 2017-01-25 P2 dminor RESO Port sdp_file_parser unit test to standalone binary
1317044 WebRTC: Signaling 2017-02-02 P2 dminor RESO Intermittent mediapipeline_unittest | test failed with re...
1318180 WebRTC: Networking 2017-01-14 P2 25 drno RESO Cannot createOffer after network change
1322338 WebRTC 2016-12-10 P2 25 drno RESO Point out lack of STUN/TURN server in ICE failure message
1322438 WebRTC: Networking 2016-12-10 P2 25 drno RESO Change ICE failed message depending on presence of relay ...
1322659 WebRTC 2017-05-05 P2 25 drno RESO Too many STUN/TURN servers don't help with conectivity
1263312 WebRTC 2017-03-10 P2 23 jib RESO Have addIceCandidate, setLocalDescription et al take dict...
1329193 WebRTC 2017-01-11 P2 25 jib RESO More overhaul PeerConnection.js with modern JavaScript
1223692 WebRTC: Audio/Video 2017-01-16 P2 25 johannkoenig RESO Update libvpx to 1.6.0
1328330 WebRTC: Audio/Video 2017-01-19 P2 25 johannkoenig RESO vp8 error concealment should be removed
1331498 WebRTC: Audio/Video 2019-02-05 P2 johannkoenig RESO Update libvpx to 1.6.1
1321628 WebRTC: Networking 2017-10-06 P2 25 mfroman RESO add ice restart and rollback counts to about:webrtc
1328429 WebRTC: Signaling 2017-01-07 P2 21 mfroman RESO When no redundant encodings are specified for RED in offe...
1316261 WebRTC: Networking 2016-12-08 P2 mt RESO System CA's cause big and fragmented DTLS messages
1324608 WebRTC: Networking 2016-12-21 P2 25 nobody RESO RtpStreamId RTP header extension indicates incorrect head...
1326011 WebRTC 2016-12-29 P2 25 nobody RESO webrtc/trunk/webrtc/base/platform_thread.cc:44:47: error:...
1332354 Audio/Video: cubeb 2017-01-20 P2 21 padenot RESO Allow enabling cubeb log by flipping a pref
1320101 WebRTC: Audio/Video 2017-05-11 P2 25 rjesup RESO Setting b=TIAS caps us at 2kbps
1326442 WebRTC: Audio/Video 2017-01-26 P2 rjesup RESO VideoConduit code should simply reconfigure the VideoSend...
1330696 Audio/Video: MediaSt 2017-01-13 P3 35 apehrson RESO Add profiler labels to Canvas frame capturing
1332598 Audio/Video: Recordi 2017-01-21 P3 35 apehrson RESO Improve logging of VP8TrackEncoder
1317660 WebRTC: Audio/Video 2016-11-19 P3 33 ayumiqmazaky RESO Fix CID 1394336: Resource leaks in TestAudioPacketizer.cpp
1316215 Audio/Video: GMP 2017-12-03 P3 chris RESO Convert GMPService to MozPromise
1317473 Audio/Video: GMP 2016-11-16 P3 chris RESO GMPServiceParent::AddOnGMPThread(path) can't handle a mix...
1317822 Audio/Video: GMP 2016-11-16 P3 chris RESO Move GMPCrashHelper into its own file
1325185 Audio/Video: GMP 2016-12-22 P3 chris RESO Fix operator precedence in GMPUtils' ToHexString()
1331829 Audio/Video: GMP 2017-01-19 P3 chris RESO Remove GMP async shutdown
1332149 Audio/Video: GMP 2017-01-19 P3 chris RESO Don't expose plugin-container or sandbox vouchers to GMPs.
1193731 WebRTC: Signaling 2016-11-22 P3 35 drno RESO onicegatheringstatechange doesn't work
1318803 WebRTC: Networking 2016-11-22 P3 drno RESO Provide IPC reason for STUN filter blocking
1318965 Audio/Video: GMP 2021-07-20 P3 jay.harris RESO Convert gmp-clearkey to use Chromium ContentDecryptionMod...
1319197 Audio/Video: GMP 2016-11-24 P3 jay.harris RESO Remove audio decoding from gmp-clearkey
1197021 WebRTC 2016-11-22 P3 jib RESO Remove last remnants of already retired backwards compati...
1230381 WebRTC: Signaling 2017-10-25 P3 25 jib RESO PeerConnection.js: "TypeError: e is null" or "TypeError: ...
1294769 Audio/Video: GMP 2020-02-16 P3 25 mozbugz RESO gmp-clearkey AnnexB::ConvertFrameInPlace potential out of...
1313258 Audio/Video: GMP 2018-07-15 P3 35 padenot RESO Intermittent FATAL ERROR: AsyncShutdown timeout in xpcom-...
1326463 WebRTC: Audio/Video 2017-01-09 P3 35 rjesup RESO Build failure in webrtc with sndio after bug 1250356
1321609 WebRTC: Audio/Video 2016-12-29 P4 45 apehrson RESO PeerConnection tests sometimes expect media flow on recei...
1323998 WebRTC: Networking 2016-12-22 P4 45 cykesiopka.bmo+mozbz RESO Stop using Scoped.h NSS types in dtlsidentity.(cpp|h) and...
1322745 Audio/Video: Recordi 2016-12-12 P5 55 apehrson RESO VP8TrackEncoder::GetSourceSurface can be improved
1306359 WebRTC: Audio/Video 2016-12-17 P5 55 cykesiopka.bmo+mozbz RESO Stop using Scoped.h NSS types in RTCCertificate.(cpp|h)
1332585 Audio/Video: Recordi 2017-02-15 -- apehrson RESO Add some VideoTrackEncoder unit tests
1318132 WebRTC: Audio/Video 2016-11-19 -- jib RESO Coverity issue in CamerasChild
1318163 WebRTC 2016-11-19 -- jib RESO Remove unimplemented and non-spec getStreamById from RTCP...
1322274 WebRTC 2017-10-25 -- jib RESO Overhaul PeerConnection.js with modern JavaScript
1323095 WebRTC 2017-03-11 -- jib RESO Add deprecation warnings to callback-based pc.getStats()
1329932 WebRTC: Networking 2017-01-10 -- nobody RESO Remove unneeded nsXPCOMGlue includes
1332905 Audio/Video: cubeb 2017-01-30 -- padenot RESO Crash in abort | `anonymous namespace''::wasapi_stream_init
1320891 WebRTC 2016-11-29 -- sledru RESO Make some webrtc tests build with gcc 7.0 and --enable-wa...
1056934 WebRTC: Networking 2017-08-30 P1 15 docfaraday VERI Support TURN TLS in WebRTC
1307754 WebRTC: Audio/Video 2016-12-29 P1 12 rjesup VERI Webrtc. FF Beta 50.0b4. No signal from microphone.
1326288 WebRTC: Audio/Video 2017-03-24 P1 12 rjesup VERI VP9 decoding broken by webrtc.org 49 update - YCbCr point...
1270572 WebRTC: Audio/Video 2021-05-27 P2 22 bonchiang VERI While page already has a live track, getUserMedia should ...
115 bugs found.

File a new bug in the "Core" product