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 ...
|