| 1213414
|
|
Implement channelCount audio constraint
|
Core
|
WebRTC: Audio/Video
|
achronop
|
RESO
|
FIXE
|
2017-08-24
|
| 1374275
|
|
Update cubeb from upstream to 18153b9
|
Core
|
Audio/Video: cubeb
|
achronop
|
RESO
|
FIXE
|
2017-06-27
|
| 1380233
|
|
Update cubeb from upstream to bb18984
|
Core
|
Audio/Video: cubeb
|
achronop
|
RESO
|
FIXE
|
2017-07-14
|
| 1381015
|
|
Update cubeb from upstream to a329c6a
|
Core
|
Audio/Video: cubeb
|
achronop
|
RESO
|
FIXE
|
2019-07-04
|
| 1384053
|
|
Update cubeb from upstream to 09aeb59
|
Core
|
Audio/Video: cubeb
|
achronop
|
RESO
|
FIXE
|
2017-07-28
|
| 1385481
|
|
Update cubeb from upstream to 42da8d2
|
Core
|
Audio/Video: cubeb
|
achronop
|
RESO
|
FIXE
|
2017-08-01
|
| 1372687
|
|
[Mdm2] Intermittent test_peerConnection_basicVideoRemoteHwEncoder.html | java-exception org.mozilla.gecko.mozglue.GeckoLoader$AbortException: abort() called from :0xdd6216d9 () at org.mozilla.gecko.mozglue.GeckoLoader.abort(GeckoLoader.java:532)
|
Core
|
WebRTC
|
bonchiang
|
RESO
|
FIXE
|
2017-07-10
|
| 1374640
|
|
Multiple calls to getUsermedia return the same window / screen
|
Core
|
WebRTC: Audio/Video
|
bonchiang
|
RESO
|
FIXE
|
2017-07-13
|
| 1374938
|
|
mediaDevices.ondevicechange and cam contraints are broken on Mac OSX
|
Core
|
WebRTC: Audio/Video
|
bonchiang
|
RESO
|
FIXE
|
2017-07-20
|
| 1376957
|
|
Widevine CDM 970 fails to load on Windows 7
|
Core
|
Audio/Video: GMP
|
chris
|
RESO
|
FIXE
|
2017-08-14
|
| 1383580
|
|
EME video playback fails due to hitting CDM IPC shmem limit
|
Core
|
Audio/Video: GMP
|
chris
|
RESO
|
FIXE
|
2017-07-26
|
| 1349539
|
|
Protect against integer overflow in webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
|
Core
|
WebRTC: Audio/Video
|
dminor
|
RESO
|
FIXE
|
2018-02-01
|
| 1355947
|
|
Use TestNrSocket to build a fake ICE implementation for testing
|
Core
|
WebRTC: Signaling
|
dminor
|
RESO
|
FIXE
|
2017-06-28
|
| 1368030
|
|
Intermittent dom/media/tests/mochitest/test_getUserMedia_basicScreenshare.html | application terminated with exit code 5
|
Core
|
WebRTC: Audio/Video
|
dminor
|
RESO
|
FIXE
|
2018-02-01
|
| 1378412
|
|
Build error on Linux 32 bit due to a warning in task_queue_libevent.cc with clang 4.0
|
Core
|
WebRTC
|
dminor
|
RESO
|
FIXE
|
2017-07-12
|
| 1379743
|
|
Screensharing with resize constraints is broken (graphics garbage) (regression)
|
Core
|
WebRTC: Audio/Video
|
dminor
|
RESO
|
FIXE
|
2017-08-30
|
| 1379836
|
|
Fix AEC Logging
|
Core
|
WebRTC
|
dminor
|
RESO
|
FIXE
|
2017-07-19
|
| 1382681
|
|
Crash in java.lang.RuntimeException: Camera is already stopped! at org.webrtc.videoengine.VideoCaptureAndroid.stopCaptureOnCameraThread(VideoCaptureAndroid.java)
|
Core
|
WebRTC
|
dminor
|
RESO
|
FIXE
|
2017-08-14
|
| 1384874
|
|
Build fail on OSX 10.11.6 after Bug 1368030 landed
|
Core
|
WebRTC
|
dminor
|
RESO
|
FIXE
|
2017-07-28
|
| 1264479
|
|
Implement RTCPeerConnection attributes: currentLocalDescription pendingLocalDescription currentRemoteDescription pendingRemoteDescription
|
Core
|
WebRTC: Signaling
|
drno
|
RESO
|
FIXE
|
2018-05-14
|
| 1372383
|
|
[Libfuzzer] Heap-buffer-overflow in sdp_parse_attr_fmtp when parsing annex p attribute
|
Core
|
WebRTC: Signaling
|
drno
|
RESO
|
FIXE
|
2020-05-19
|
| 1372467
|
|
[Libfuzzer] Heap-buffer-overflow in sdp_parse_attr_fmtp
|
Core
|
WebRTC: Signaling
|
drno
|
RESO
|
FIXE
|
2020-05-19
|
| 1373144
|
|
Ignore multiple MSID's as preparation for Transceivers
|
Core
|
WebRTC: Signaling
|
drno
|
RESO
|
FIXE
|
2018-09-22
|
| 1377959
|
|
jvm_android.cc passes va_list to varags methods
|
Core
|
WebRTC
|
froydnj+bz
|
RESO
|
FIXE
|
2018-02-01
|
| 1385115
|
|
dom/media/gmp/GMPChild.cpp:438:33: error: 'XUL_LIB_FILE' was not declared in this scope
|
Core
|
Audio/Video: GMP
|
jbeich
|
RESO
|
FIXE
|
2017-08-01
|
| 1379392
|
|
Firefox crashes (UAF) whenever camera is not readable (NotReadableError) in OSX. regression.
|
Core
|
WebRTC: Audio/Video
|
jib
|
RESO
|
FIXE
|
2017-08-28
|
| 1382095
|
|
Mainthread-refcounts on MediaEngineSource in setLastPrefs runnable can sidestep proper cleanup on shutdown.
|
Core
|
WebRTC: Audio/Video
|
jib
|
RESO
|
FIXE
|
2017-07-21
|
| 1379400
|
|
Intermittent dom/media/tests/mochitest/test_peerConnection_stats.html | candidate-pair.bytesReceived is a sane number (20,000<>800,000) for a short test. value=801806
|
Core
|
WebRTC: Signaling
|
mfroman
|
RESO
|
FIXE
|
2017-07-12
|
| 1380430
|
|
Backport current webrtc.org support for RtpStreamId (and the much better handling of variable size RtpHeaderExtensions)
|
Core
|
WebRTC
|
mfroman
|
RESO
|
FIXE
|
2017-07-19
|
| 1383272
|
|
Fix RTP parsing error for RtpStreamId
|
Core
|
WebRTC: Networking
|
mfroman
|
RESO
|
FIXE
|
2017-07-22
|
| 1377803
|
|
Remove unnecessary plarena.h #includes
|
Core
|
WebRTC: Signaling
|
n.nethercote
|
RESO
|
FIXE
|
2017-07-05
|
| 1343515
|
|
RTPStats inbound-rtp.jitter is wrong when isRemote=true
|
Core
|
WebRTC
|
na-g
|
RESO
|
FIXE
|
2020-07-21
|
| 1374465
|
|
Webrtc video inbound-rtp does not include framesDecoded
|
Core
|
WebRTC
|
na-g
|
RESO
|
FIXE
|
2017-07-07
|
| 1341555
|
|
Label runnables in dom/media/MediaStreamGraph.cpp
|
Core
|
Audio/Video: MediaSt
|
padenot
|
RESO
|
FIXE
|
2017-06-30
|
| 1373213
|
|
Repair cubeb logging
|
Core
|
Audio/Video: cubeb
|
padenot
|
RESO
|
FIXE
|
2017-07-18
|
| 1378067
|
|
Label more runnables in dom/media (GraphDriver.cpp, CubebUtils.cpp, DOMMediaStream.cpp)
|
Core
|
Audio/Video: MediaSt
|
padenot
|
RESO
|
FIXE
|
2018-07-08
|
| 1341285
|
|
Update WebRTC code to webrtc.org stable branch 57
|
Core
|
WebRTC
|
rjesup
|
RESO
|
FIXE
|
2022-03-17
|
| 1373475
|
|
cubeb-pulse-rs: Panic in pulse_format_to_cubeb_format
|
Core
|
Audio/Video: cubeb
|
u480271
|
RESO
|
FIXE
|
2017-06-18
|
| 1375873
|
|
all tabs crash in nightly (thread '<unnamed>' panicked at 'SinkInfo contains invalid flags')
|
Core
|
Audio/Video: cubeb
|
u480271
|
RESO
|
FIXE
|
2018-06-02
|
| 1376000
|
|
All Tabs Crash in libpulse
|
Core
|
Audio/Video: cubeb
|
u480271
|
RESO
|
FIXE
|
2018-09-22
|
| 1364038
|
|
getUserMedia returns an unusable stream after removing and reconnecting camera
|
Core
|
WebRTC: Audio/Video
|
bonchiang
|
RESO
|
FIXE
|
2021-11-21
|
| 1374204
|
|
[Firefox for Android] set preference for hardware encoder to true in nightly
|
Core
|
WebRTC: Audio/Video
|
bonchiang
|
RESO
|
FIXE
|
2017-06-23
|
| 1197045
|
|
Add some basic audio hardware/driver/format information to about:support
|
Core
|
Audio/Video: cubeb
|
cchang
|
RESO
|
FIXE
|
2022-01-28
|
| 1328169
|
|
mediaconduit_unittests are no longer being built (or run)
|
Core
|
WebRTC: Audio/Video
|
dminor
|
RESO
|
FIXE
|
2017-07-14
|
| 1370601
|
|
Make it possible for offerer and answerer to switch roles in jsep_session_unittest.cpp
|
Core
|
WebRTC: Signaling
|
dminor
|
RESO
|
FIXE
|
2017-06-12
|
| 1371161
|
|
Port SDP file parser to LibFuzzer
|
Core
|
WebRTC: Signaling
|
drno
|
RESO
|
FIXE
|
2017-06-14
|
| 1373103
|
|
nicer log filled with read callback messages if TCP is used
|
Core
|
WebRTC: Networking
|
drno
|
RESO
|
FIXE
|
2017-06-15
|
| 1373450
|
|
MaxMessageSize zero is ambiguous
|
Core
|
WebRTC: Signaling
|
drno
|
RESO
|
FIXE
|
2017-06-16
|
| 1374440
|
|
max-message-size should only be emitted with EOR support
|
Core
|
WebRTC: Signaling
|
drno
|
RESO
|
FIXE
|
2017-06-22
|
| 1383575
|
|
Remove TURNS log warning
|
Core
|
WebRTC: Networking
|
drno
|
RESO
|
FIXE
|
2017-07-25
|
| 1355048
|
|
Add a WebrtcVideoDecoder MediaDataDecoder class.
|
Core
|
WebRTC: Audio/Video
|
jya-moz
|
RESO
|
FIXE
|
2017-07-17
|
| 1343755
|
|
Label runnables in netwerk/sctp/datachannel/
|
Core
|
WebRTC: Networking
|
kershaw
|
RESO
|
FIXE
|
2017-06-28
|
| 1339906
|
|
Update IceCandidatePairStats
|
Core
|
WebRTC: Networking
|
mfroman
|
RESO
|
FIXE
|
2019-04-22
|
| 1368159
|
|
Tweak error log line to warn or info at ice_ctx.c:710
|
Core
|
WebRTC: Networking
|
mfroman
|
RESO
|
FIXE
|
2017-06-15
|
| 1373015
|
|
Permafailing dom/media/tests/mochitest/test_peerConnection_stats.html | candidate-pair.state is succeeded. value=cancelled
|
Core
|
WebRTC
|
mfroman
|
RESO
|
FIXE
|
2017-06-25
|
| 1382972
|
|
Intermittent /webrtc/RTCPeerConnection-addIceCandidate.html | Add valid candidate should never resolve when pc is closed - Peer connection is closed
|
Core
|
WebRTC
|
mfroman
|
RESO
|
FIXE
|
2017-07-30
|
| 1383069
|
|
Intermittent TEST-UNEXPECTED-PASS | /webrtc/RTCPeerConnection-setLocalDescription.html | setLocalDescription(offer) should never resolve if connection is closed in parallel - expected FAIL
|
Core
|
WebRTC
|
mfroman
|
RESO
|
FIXE
|
2017-07-30
|
| 1305813
|
|
Ensure RID isn't empty when constructing RTP header
|
Core
|
WebRTC: Signaling
|
nobody
|
RESO
|
FIXE
|
2017-06-26
|
| 1330360
|
|
Use multiple MediaStreamGraph in the same process
|
Core
|
Audio/Video: MediaSt
|
padenot
|
RESO
|
FIXE
|
2022-03-25
|
| 1371719
|
|
Don't create an MSG if it's not going to be used immediately
|
Core
|
Audio/Video: MediaSt
|
padenot
|
RESO
|
FIXE
|
2017-07-31
|
| 1372247
|
|
Make the AudioOutputObserver a member of the MediaEngineWebRTCMicrophoneSource so it's not a global singleton anymore
|
Core
|
Audio/Video: MediaSt
|
padenot
|
RESO
|
FIXE
|
2017-09-18
|
| 1371841
|
|
PeerConnectionImpl::RecordEndOfCallTelemetry sends telemetry when no connection information was exchanged
|
Core
|
WebRTC: Signaling
|
pe5
|
RESO
|
FIXE
|
2017-06-13
|
| 1376357
|
|
Enable ESLint for dom/media/*.js
|
Core
|
WebRTC
|
standard8
|
RESO
|
FIXE
|
2017-06-30
|
| 1374494
|
|
Update cubeb-pulse-rs to to 6451581
|
Core
|
Audio/Video: cubeb
|
u480271
|
RESO
|
FIXE
|
2017-08-04
|
| 1365852
|
|
Intermittent browser/base/content/test/webrtc/browser_devices_get_user_media_unprompted_access_queue_request.js | leaked 1 window(s) until shutdown [url = chrome://browser/content/webrtcIndicator.xul]
|
Core
|
WebRTC: Audio/Video
|
bonchiang
|
RESO
|
FIXE
|
2017-06-15
|
| 1372080
|
|
[EME] Reorder frames coming out of Widevine CDM
|
Core
|
Audio/Video: GMP
|
chris
|
RESO
|
FIXE
|
2017-12-05
|
| 1381720
|
|
[EME] Update content_decryption_module.h
|
Core
|
Audio/Video: GMP
|
chris
|
RESO
|
FIXE
|
2017-11-08
|
| 1379982
|
|
mach webrtc-gtest does not work on mac
|
Core
|
WebRTC
|
dminor
|
RESO
|
FIXE
|
2017-07-21
|
| 1373858
|
|
Intermittent dom/media/tests/mochitest/test_peerConnection_stats.html | candidate-pair.bytesSent was tested.
|
Core
|
WebRTC
|
mfroman
|
RESO
|
FIXE
|
2017-07-02
|
| 1383137
|
|
Intermittent /webrtc/RTCPeerConnection-generateCertificate.html | generateCertificate() with 0 expires parameter should generate expired cert - assert_less_than_equal: expected a number less than or equal to 1500650276906 but got 1500650276907
|
Core
|
WebRTC
|
mfroman
|
RESO
|
FIXE
|
2019-04-24
|
| 1312623
|
|
Intermittent dom/media/tests/mochitest/test_peerConnection_setLocalAnswerInStable.html | application crashed [@ mozilla::CycleCollectedJSContext::ProcessMetastableStateQueue]
|
Core
|
Audio/Video: MediaSt
|
padenot
|
RESO
|
FIXE
|
2018-04-07
|
| 1384826
|
|
Media: WebRTC: Fix build config for MIPS
|
Core
|
WebRTC
|
r
|
RESO
|
FIXE
|
2017-07-27
|
| 1376885
|
|
Implement a way to deny gUM requests in continuous integration
|
Core
|
WebRTC: Audio/Video
|
apehrson
|
RESO
|
FIXE
|
2017-08-15
|
| 1374499
|
|
Increase number of shmems pre-allocated by PChromiumCDM.
|
Core
|
Audio/Video: GMP
|
chris
|
RESO
|
FIXE
|
2017-06-20
|
| 1196974
|
|
Remove mozDontOfferDataChannel/mozBundleOnly from RTCOfferOptions
|
Core
|
WebRTC: Signaling
|
drno
|
RESO
|
FIXE
|
2017-08-02
|
| 1377093
|
|
Build mediapipeline tests on BSDs
|
Core
|
WebRTC: Audio/Video
|
jbeich
|
RESO
|
FIXE
|
2017-06-30
|
| 1383020
|
|
mediaconduit_unittests.cpp: suggest explicit braces to avoid ambiguous 'else' [-Werror=dangling-else]
|
Core
|
WebRTC: Signaling
|
sledru
|
RESO
|
FIXE
|
2017-08-02
|
| 1377078
|
|
ooura_fft.cc/ooura_fft_sse2.cc has '//' in #include patch which causes build failure
|
Core
|
WebRTC: Audio/Video
|
stransky
|
RESO
|
FIXE
|
2018-09-22
|
| 1378826
|
|
Removing the last track from a recorder results in crash
|
Core
|
Audio/Video: Recordi
|
brycebugemail
|
VERI
|
FIXE
|
2018-02-01
|
| 1382718
|
|
MediaRecorder throws RecordErrorEvent on valid window mediaStream with no further info
|
Core
|
Audio/Video: Recordi
|
brycebugemail
|
VERI
|
FIXE
|
2017-08-02
|
| 1375708
|
|
Netflix broken on Linux in Firefox 54
|
Core
|
Audio/Video: GMP
|
chris
|
VERI
|
FIXE
|
2017-07-06
|
| 1372509
|
|
Self-XSS XUL Injection in about:webrtc
|
Core
|
WebRTC
|
jib
|
VERI
|
FIXE
|
2021-10-21
|
| 1371000
|
|
Firefox not seeing all windows available to share
|
Core
|
WebRTC
|
na-g
|
VERI
|
FIXE
|
2022-01-10
|
| 1375238
|
|
WebRTC 57 fails to build with Clang i386 due to -Wc++11-narrowing
|
Core
|
WebRTC
|
rjesup
|
VERI
|
FIXE
|
2017-07-05
|
| 1384655
|
|
Build with --enable-alsa --enable-tests is broken by webrtc::AudioDeviceLinuxALSA
|
Core
|
WebRTC
|
padenot
|
VERI
|
FIXE
|
2017-08-10
|