1342389
|
Crash in CleanupPerAppKey | `anonymous namespace''::wasap...
|
Audio/Video: cubeb
|
1344653
|
Update cubeb from upstream to a3c012f
|
Audio/Video: cubeb
|
1355306
|
getUserMedia doesn't work in more than one tab at a time now
|
WebRTC: Audio/Video
|
1357683
|
Update cubeb from upstream to 6e52314
|
Audio/Video: cubeb
|
1359451
|
Update cubeb from upstream to 17503c4
|
Audio/Video: cubeb
|
1359477
|
Crash in libsystem_kernel.dylib@0x19d42 called from audio...
|
Audio/Video: cubeb
|
1362334
|
Update cubeb from upstream to 26a50b0
|
Audio/Video: cubeb
|
1366707
|
Update cubeb from upstream to 96cdb17
|
Audio/Video: cubeb
|
1367702
|
Crashes in __pthread_kill | abort | __assert_rtn (when pl...
|
Audio/Video: cubeb
|
1368910
|
Update cubeb from upstream to 3428c2b
|
Audio/Video: cubeb
|
1337418
|
Crash in nsACString_internal::Assign | nsACString_interna...
|
WebRTC: Audio/Video
|
1320994
|
Non deterministic recording-device-events notification co...
|
WebRTC: Audio/Video
|
1334421
|
Assertion failure: aInstanceSize == 0 || entry->GetClassS...
|
WebRTC: Audio/Video
|
1348381
|
VP8TrackEncoder crash: [@mozilla::VP8TrackEncoder::Prepar...
|
Audio/Video: Recordi
|
1363253
|
Intermittent browser/base/content/test/webrtc/browser_dev...
|
WebRTC
|
1358030
|
[Firefox for Android] add mochitest to test remote h/w en...
|
WebRTC: Audio/Video
|
1363259
|
getusermedia constraint frameRate doesn't take effect
|
WebRTC: Audio/Video
|
1349421
|
Integer overflow in dom/media/encoder/OpusTrackEncoder.cp...
|
Audio/Video: Recordi
|
1354457
|
MediaRecorder.pause() stopped working in 51
|
Audio/Video: Recordi
|
1358756
|
Crash in `anonymous namespace''::refill
|
Audio/Video: cubeb
|
1358373
|
Assertion in ChromiumCDMChild::Allocate due lack of appro...
|
Audio/Video: GMP
|
1366639
|
Add telemetry to track max number of shmems used by PChro...
|
Audio/Video: GMP
|
964133
|
Hook up webrtc.org unit tests so they can be built and ru...
|
WebRTC
|
1354350
|
libyuv builds with NVALGRIND under MOZ_VALGRIND builds
|
WebRTC
|
1354353
|
nICEr and nrappkit builds with NVALGRIND under MOZ_VALGR...
|
WebRTC
|
1325513
|
RTP header extensions potentially read out of bounds
|
WebRTC
|
1335966
|
Crash in nr_ice_component_consent_timer_cb
|
WebRTC: Networking
|
1355010
|
Fx 55 enforcing correct extmap direction causes call fail...
|
WebRTC
|
1355259
|
ICE TCP user pref filters out all candidates
|
WebRTC: Networking
|
1358889
|
Crash in mozilla::SingletonThreadHolder::ReleaseUse_i
|
WebRTC: Networking
|
1361206
|
RTP Header Extension IDs in Offer/Answer Exchange
|
WebRTC: Signaling
|
1363563
|
Firefox crashes when using WebRTC and renegotiating with ...
|
WebRTC: Networking
|
1365090
|
Crash when renegotiating SDP and simulcast is enabled
|
WebRTC: Networking
|
1367930
|
Handle RID Encondings mis-match for simulcast
|
WebRTC: Signaling
|
1348174
|
Test origin-unique deviceId persistence.
|
WebRTC: Audio/Video
|
1359662
|
Screen-sharing track.getSettings() returns crazy high val...
|
WebRTC: Audio/Video
|
1366415
|
Unprefix autoGainControl and noiseSuppression constraints
|
WebRTC: Audio/Video
|
1320561
|
Crash in `anonymous namespace''::handle_channel_layout
|
Audio/Video: cubeb
|
1358896
|
Crash in audiounit_setup_stream
|
Audio/Video: cubeb
|
1345511
|
Move nICEr stun local address discovery to an IPC call to...
|
WebRTC: Networking
|
1350055
|
--disable-webrtc build fails with "netwerk/ipc/PNecko.ipd...
|
WebRTC
|
1350568
|
Crash in Abort | corrupted actor state | mozalloc_abort |...
|
WebRTC: Networking
|
1241066
|
getStats API always returns 1 for mozRTT
|
WebRTC
|
1344970
|
Rename mozRtt stat to roundTripTime and change behavior t...
|
WebRTC
|
1355220
|
Add sender.getStats() and receiver.getStats()
|
WebRTC
|
1359775
|
Add RTCRtpContributingSourceStats to webrtc stats report
|
WebRTC
|
1357640
|
Permaorange in test_mediarecorder_record_changing_video_r...
|
Audio/Video: Recordi
|
1337641
|
Intermittent browser/base/content/test/webrtc/browser_dev...
|
Audio/Video: cubeb
|
1341666
|
Assertion failure: false (We should be reviving the graph...
|
Audio/Video: MediaSt
|
1348344
|
Add some code to debug the drift on OSX when aggregate de...
|
Audio/Video: cubeb
|
1358868
|
Crash in `anonymous namespace''::wasapi_stream_start
|
Audio/Video: cubeb
|
1338521
|
Video fails to appear (or freezes?) after refreshing page...
|
WebRTC: Audio/Video
|
1346005
|
Add logging for audio frames
|
WebRTC: Audio/Video
|
1349233
|
While screen sharing on Cisco Spark (web app) the receivi...
|
WebRTC
|
1350912
|
TIAS can be below the minimum bitrate for high-resolution...
|
WebRTC: Networking
|
1353476
|
Crash in mozilla::camera::CamerasParent::IsShuttingDown
|
WebRTC: Audio/Video
|
1367578
|
DataChannel logging broken since Network team updated it ...
|
WebRTC: Networking
|
1369724
|
Fake H264 codec used in mochitests randomly fails causing...
|
WebRTC: Audio/Video
|
1360060
|
Enable use of rust version of cubeb PulseAudio backend.
|
Audio/Video: cubeb
|
1232043
|
Handle resolution changes for VP8 in MediaRecorder
|
Audio/Video: Recordi
|
1368448
|
Intermittent dom/media/test/test_bug1113600.html | appli...
|
Audio/Video: MediaSt
|
1265755
|
[Firefox for Android] Utilize hardware encoders to save p...
|
WebRTC: Audio/Video
|
1359668
|
Screen-sharing track.getSettings() should reflect actual ...
|
WebRTC: Audio/Video
|
1364776
|
[MediaRecorder] requestData fails if called on a paused r...
|
Audio/Video: Recordi
|
1344812
|
Fix clang 5.0 warnings for ClearKeySessionManager.cpp: [-...
|
Audio/Video: GMP
|
862837
|
Signaling Unittests should build under Windows
|
WebRTC: Signaling
|
1342523
|
We need better ICE duration telemetry
|
WebRTC: Networking
|
1344556
|
Nor a=simulcast/a=rid if calling sender.setParameters() b...
|
WebRTC: Signaling
|
1176415
|
a=ssrc lines on m=application lines
|
WebRTC: Signaling
|
1323723
|
RTCPeerConnection: stop sending media when applying a sec...
|
WebRTC: Networking
|
1335262
|
Add max-message-size support for data-channels
|
WebRTC: Signaling
|
1342579
|
Test to verify we can connect to different certs and ICE ...
|
WebRTC
|
1343640
|
Add hex dumping of RTP and RTCP to mtransport
|
WebRTC: Networking
|
1347813
|
the incoming audio's packet loss is zero when get the p...
|
WebRTC: Networking
|
1354222
|
Fix log level casting warnings
|
WebRTC
|
1364325
|
Replace AddLocalRTPExtensions with SetLocalRTPExtensions
|
WebRTC: Signaling
|
1365081
|
Data channel fails to open if maxPacketLifeTime is set
|
WebRTC: Networking
|
1366581
|
single m-line offers does not use bundle
|
WebRTC: Signaling
|
1349480
|
getUserMedia({}) returns NotSupportedError
|
WebRTC
|
1368875
|
Telemetry WEBRTC_GET_USER_MEDIA_TYPE only counts callers ...
|
WebRTC: Audio/Video
|
1350893
|
Crash in arena_malloc | je_malloc | cubeb_stream::cubeb_s...
|
Audio/Video: cubeb
|
1352929
|
Update libcubeb to revision 04826edb
|
Audio/Video: cubeb
|
1303867
|
Missing DTLS alert on PeerConnection.close()
|
WebRTC: Networking
|
1325991
|
Firefox emits a=bundle-only with a non-zero port
|
WebRTC: Signaling
|
1351531
|
Intermittent dom/media/tests/mochitest/test_peerConnectio...
|
WebRTC
|
1351590
|
Intermittent dom/media/tests/mochitest/test_peerConnectio...
|
WebRTC
|
1353575
|
ice_unittest fails on external stun server tests - switch...
|
WebRTC: Networking
|
1358224
|
simulcast mochitests (offer and answer) need work to filt...
|
WebRTC
|
1361139
|
remove test-related ssrc filtering from MediaPipeline now...
|
WebRTC: Signaling
|
1365291
|
Make sure 'this' is captured on dispatch to STS thread in...
|
WebRTC
|
1353910
|
Permafailing on Aurora since e10s-multi Windows browser_d...
|
WebRTC
|
1325173
|
RtpStreamId is truncated when read from RTP extension header
|
WebRTC: Networking
|
1348657
|
Implement framesEncoded, pliCount, nackCount and firCount...
|
WebRTC: Networking
|
1264343
|
test_peerConnection_scaleResolution.html times out on And...
|
WebRTC: Audio/Video
|
1354993
|
[Firefox for Android] Add PictureID into VP8 CodecSpecifi...
|
WebRTC: Audio/Video
|
1351803
|
--disable-webrtc build broken in cubeb_wasapi.obj : error...
|
Audio/Video: cubeb
|
1371362
|
PeerConnectionImpl has unused counters mNumAudioStreams a...
|
WebRTC: Signaling
|
1343143
|
Enable VP8 temporal scalability for simulcast
|
WebRTC: Audio/Video
|
1353030
|
cleanup and document WrapRunnable uses
|
Audio/Video: GMP
|
1344909
|
mingw build broken with VideoDecoder.cpp error: 'std::thr...
|
Audio/Video: GMP
|
1346665
|
Implement PulseAudio backend for Cubeb in Rust
|
Audio/Video: cubeb
|
1360053
|
Cleanup unused sources files in cubeb-pulse-rs
|
Audio/Video: cubeb
|
1093835
|
test_peerConnection_addSecondAudioStream.html needs verif...
|
WebRTC
|
1332845
|
Intermittent dom/media/test/test_mediarecorder_principals...
|
Audio/Video: Recordi
|
1342950
|
MOZ_ASSERT(true) in SourceMediaStream::AddDirectTrackList...
|
Audio/Video: MediaSt
|
1315850
|
[EME] Implement ChromiumCDM as native GMP API
|
Audio/Video: GMP
|
1334111
|
EME: PersistentState should be disabled in private browsi...
|
Audio/Video: GMP
|
1344614
|
Improve GMP clock from second to millisecond precision
|
Audio/Video: GMP
|
1351132
|
Logging doesn't work properly in ChromiumCDMChild
|
Audio/Video: GMP
|
1351874
|
ChromiumCDMProxy::Shutdown() fails to dispatch to Chromiu...
|
Audio/Video: GMP
|
1351953
|
Use Shmems to send encrypted video frames and samples thr...
|
Audio/Video: GMP
|
1351954
|
gmp-clearkey's WMF decoder doesn't allocate its video fra...
|
Audio/Video: GMP
|
1352924
|
Fix GMP async shutdown
|
Audio/Video: GMP
|
1360959
|
Fix assert in ChromiumCDMChild::RecvDecrypt()
|
Audio/Video: GMP
|
1345791
|
Crash in mozilla::NrIceCtx::SetConnectionState
|
WebRTC: Networking
|
1348021
|
media/libcubeb/src/cubeb_log.cpp:76:17: undefined referen...
|
Audio/Video: cubeb
|
1346913
|
Rewrite waitForRtpFlow in mochitests to use new async/aw...
|
WebRTC
|
1341238
|
Patch to allow users to force a particular libcubeb audio...
|
Audio/Video: cubeb
|
1345313
|
Label runnables under dom/media/gtest
|
Audio/Video: GMP
|
1346678
|
Label runnables in dom/media/gmp/GMPParent.cpp and its fr...
|
Audio/Video: GMP
|
1346679
|
Label runnables in dom/media/gmp/GMPCDMCallbackProxy.cpp
|
Audio/Video: GMP
|
1346681
|
Label runnables in dom/media/gmp/GMPVideoEncoderParent.cpp
|
Audio/Video: GMP
|
971528
|
Support stereo capture in gUM
|
WebRTC: Audio/Video
|
1356212
|
Intermittent dom/media/test/test_mediarecorder_record_cha...
|
Audio/Video: Recordi
|
1337294
|
PeerConnectionMedia.cpp: Unnecessary call to std::string:...
|
WebRTC: Networking
|
1345761
|
Remove the unused do_GetMainThread() call from GMPService...
|
Audio/Video: GMP
|
1363261
|
MediaManager keeps accumulating callIds on successful gUM...
|
WebRTC: Audio/Video
|
1349008
|
SystemGroup used undefined in GMPVideoEncoderParent.cpp
|
Audio/Video: GMP
|
1352556
|
Possible integer overflow in usage of MFGetAttributeSize ...
|
Audio/Video: GMP
|
1359854
|
WebRTC in v53.0 is not recognizing TIAS bitrates (bitrate...
|
WebRTC
|
1354207
|
signaling disables valgrind
|
WebRTC: Signaling
|
1361445
|
convert MediaEngineSource to use NS_DECL_OWNINGTHREAD
|
WebRTC
|
1369108
|
Ask for WebRTC app permissions asynchronously on Android
|
WebRTC
|
1349595
|
Possible integer overflow in allocation size in GMPVideoi...
|
Audio/Video: GMP
|
1349604
|
Possible integer overflow in allocation size in WidevineV...
|
Audio/Video: GMP
|
1369422
|
Simplify MediaStreamGraph::IsNonRealtime and make it not...
|
Audio/Video: MediaSt
|
1369698
|
Fix deprecation warning for `URL.createObjectURL(MediaStr...
|
WebRTC: Audio/Video
|
1369967
|
Crash in mozilla::AudioProxyThread::InternalProcessAudioC...
|
WebRTC: Audio/Video
|
1358938
|
Remove gonk config.
|
Audio/Video: cubeb
|
1367646
|
Update cubeb from upstream to 087dc94
|
Audio/Video: cubeb
|
1371319
|
Startup crash in libpulsecommon-10.0.so@0x54b39 following...
|
Audio/Video: cubeb
|
1372057
|
Limit rust cubeb PulseAudio backend to nightly builds
|
Audio/Video: cubeb
|
1166955
|
nsISupportsUtils.h(49) : warning C4005: 'NS_IF_ADDREF' : ...
|
WebRTC
|
1343930
|
Headset plugging/unplugging results in tab crash when the...
|
Audio/Video: cubeb
|
1345049
|
Update cubeb from upstream to f07ee6d
|
Audio/Video: cubeb
|
1340163
|
Origin-unique deviceId persistence (used in enumerateDevi...
|
WebRTC: Audio/Video
|
1353028
|
Nightly users can't see/hear stream on Talky.io
|
WebRTC
|
1343691
|
RTCP stats missing in FF53 (fallout from webrtc.org 49 up...
|
WebRTC
|
1343972
|
Crash in `anonymous namespace''::com_ptr<T>::receive (unp...
|
Audio/Video: cubeb
|
1337810
|
Jitsi video freezes after a few minutes of conversation
|
WebRTC: Audio/Video
|
1339246
|
When switching from sendrecv to recvonly to sendrecv, FF ...
|
WebRTC: Networking
|
1346620
|
Netflix has never worked on Firefox: "The WidevineCdm plu...
|
Audio/Video: GMP
|