| 1297099
|
Audio/Video: MediaSt
|
P1
|
10
|
apehrson
|
RESO
|
2017-02-09
|
LoadManagerSingleton weak reference used on multiple threads
|
| 1299172
|
Audio/Video: MediaSt
|
P1
|
10
|
apehrson
|
RESO
|
2017-01-10
|
Intermittent dom/media/test/test_bug879717.html | application crashed [@ mozilla::MediaStreamGraph::DispatchToMainThreadAfterStreamStateUpdate(already_AddRefed<nsIRunnable>)]
|
| 1299451
|
Audio/Video: MediaSt
|
P1
|
17
|
apehrson
|
RESO
|
2016-09-02
|
HTMLMediaElement's mozCaptureStream may result in no output
|
| 1300529
|
Audio/Video: MediaSt
|
P1
|
15
|
apehrson
|
RESO
|
2016-09-07
|
HTMLMediaElement::CaptureStreamInternal called incorrectly
|
| 1293194
|
Audio/Video: GMP
|
P1
|
15
|
ayumiqmazaky
|
RESO
|
2016-08-19
|
Building with --disable-eme fails with GMPDecoderModule.cpp:15:30: fatal error: mozilla/EMEUtils.h: No such file or directory
|
| 1272877
|
Audio/Video: Recordi
|
P1
|
17
|
bonchiang
|
RESO
|
2016-08-30
|
Autophone - Intermittent Android 6.0 - PROCESS-CRASH | dom/media/test/test_mediarecorder_bitrate.html | application crashed [@ mozilla::gl::GLBlitHelper::InitTexQuadProgram]
|
| 1296684
|
WebRTC: Audio/Video
|
P1
|
15
|
bonchiang
|
RESO
|
2016-08-25
|
"MediaEngine.h:14:10: fatal error: 'DeviceChangeCallback.h' file not found" with --disable-webrtc
|
| 1201363
|
Audio/Video: MediaSt
|
P1
|
18
|
ctai
|
RESO
|
2019-02-27
|
Stop buffering video in the MediaStreamGraph
|
| 1299714
|
Audio/Video: Recordi
|
P1
|
13
|
ctai
|
RESO
|
2016-09-03
|
Add back the missing timeout protection back to |VideoTrackEncoder::Init|.
|
| 1300871
|
Audio/Video: MediaSt
|
P1
|
|
ctai
|
RESO
|
2016-09-19
|
MediaRecorder video and audio are very out of sync on win10 if setting media.navigator.audio.full_duplex=false
|
| 1255371
|
WebRTC: Networking
|
P1
|
15
|
dminor
|
RESO
|
2016-08-26
|
Invalid RTCP SR sent by Firefox when h264 is used
|
| 1293422
|
WebRTC: Audio/Video
|
P1
|
12
|
dminor
|
RESO
|
2016-08-14
|
[openh264] Mode 0 crashes FF >= 50 immediately
|
| 1297058
|
WebRTC: Audio/Video
|
P1
|
18
|
dminor
|
RESO
|
2016-09-16
|
Don't send FEC for H.264 with NACK enabled
|
| 1297416
|
WebRTC: Networking
|
P1
|
15
|
docfaraday
|
RESO
|
2017-01-27
|
Implement draft-ietf-rtcweb-ip-handling
|
| 1290365
|
WebRTC: Networking
|
P1
|
19
|
drno
|
RESO
|
2016-08-18
|
TURN/TCP with hostnames doesn't work on Linux
|
| 1284683
|
WebRTC: Audio/Video
|
P1
|
15
|
jib
|
RESO
|
2016-08-22
|
Unable to use device ids from the content process to call getUserMedia on the chrome process
|
| 1284909
|
WebRTC: Audio/Video
|
P1
|
15
|
jib
|
RESO
|
2016-08-31
|
Need to write code for observing deviceId constraints for screensharing
|
| 1290629
|
WebRTC: Audio/Video
|
P1
|
19
|
jib
|
RESO
|
2018-08-03
|
nightly 20160729 compile fails without webrtc, fatal error: webrtc/video_engine/include/vie_capture.h: No such file or directory
|
| 1302348
|
Audio/Video: cubeb
|
P1
|
15
|
kinetik
|
RESO
|
2016-09-16
|
Report cubeb context init failures via telemetry
|
| 1295690
|
WebRTC: Signaling
|
P1
|
19
|
mfroman
|
RESO
|
2016-08-19
|
Bug detecting red/ulpfec codecs in JsepTrack
|
| 1312764
|
Audio/Video: cubeb
|
P1
|
15
|
padenot
|
RESO
|
2016-10-27
|
Backout the rounding changes from cubeb_wasapi.cpp from aurora
|
| 1255737
|
Audio/Video: MediaSt
|
P1
|
10
|
rjesup
|
RESO
|
2020-07-21
|
Async shutdown crash with "MediaStreamGraph shutdown: blocking on msg thread"
|
| 1271585
|
WebRTC: Audio/Video
|
P1
|
18
|
rjesup
|
RESO
|
2017-09-12
|
Back out and rewrite the resampling bypass code and WebRTCEngine to MSG code
|
| 1290075
|
WebRTC: Audio/Video
|
P1
|
10
|
rjesup
|
RESO
|
2022-11-22
|
Can't open audio inputs via GetUserMedia() in Android (or desktop without full_duplex)
|
| 1290866
|
WebRTC: Audio/Video
|
P1
|
15
|
rjesup
|
RESO
|
2016-08-05
|
aOutHandle not passed out of MediaDevice Allocate() functions
|
| 1294407
|
WebRTC
|
P1
|
10
|
rjesup
|
RESO
|
2018-10-10
|
Firefox crash when packets with missing headers are received
|
| 1294753
|
Audio/Video: Recordi
|
P1
|
17
|
rjesup
|
RESO
|
2016-09-06
|
MediaRecorder sometimes records broken audio timestamps
|
| 1297101
|
WebRTC
|
P1
|
15
|
rjesup
|
RESO
|
2016-08-23
|
Improve green-ness of Android VMs
|
| 1298698
|
Audio/Video: MediaSt
|
P1
|
10
|
rjesup
|
RESO
|
2016-09-02
|
EnsureNextIteration() may leave MediaStreamGraph asleep
|
| 1302935
|
WebRTC
|
P1
|
10
|
rjesup
|
RESO
|
2016-09-16
|
WebRTC VP9 support doesn't set the gof field correctly due to use of old libvpx
|
| 1259788
|
Audio/Video: MediaSt
|
P2
|
25
|
apehrson
|
RESO
|
2019-04-15
|
video.mozCaptureStream doesn't work with a MediaStream src
|
| 1300600
|
WebRTC
|
P2
|
25
|
apehrson
|
RESO
|
2016-09-18
|
Intermittent dom/media/tests/mochitest/test_getUserMedia_basicAudio.html | Test timed out.
|
| 1290830
|
Audio/Video: GMP
|
P2
|
25
|
ayumiqmazaky
|
RESO
|
2016-08-31
|
Make Decryptor APIs reusable by not only GMP framework
|
| 1286429
|
WebRTC: Audio/Video
|
P2
|
27
|
bonchiang
|
RESO
|
2017-06-20
|
Implement mediaDevices.ondevicechange for Mac OSX
|
| 1292335
|
Audio/Video: MediaSt
|
P2
|
25
|
ctai
|
RESO
|
2016-08-28
|
Intermittent dom/media/test/test_streams_capture_origin.html | expected black - got "255,255,0,255", expected "0,0,0,255"
|
| 1293015
|
Audio/Video: MediaSt
|
P2
|
25
|
ctai
|
RESO
|
2016-08-21
|
Intermittent dom/media/test/test_bug879717.html | v3 (Stream of 320x240.ogv) should have gotten the 'loadeddata' event callback
|
| 1295687
|
WebRTC
|
P2
|
29
|
dholbert
|
RESO
|
2016-08-19
|
media/webrtc/trunk/testing/gtest/src/gtest-internal-inl.h:213:8: warning: private field 'pretty_' is not used [-Wunused-private-field]
|
| 1298991
|
WebRTC: Networking
|
P2
|
25
|
drno
|
RESO
|
2016-10-03
|
nICEr creates duplicated prflx candidate pairs
|
| 1302481
|
WebRTC: Networking
|
P2
|
|
drno
|
RESO
|
2016-09-14
|
New ICE unit test no NAT <-> sym NAT
|
| 1213441
|
WebRTC
|
P2
|
25
|
jib
|
RESO
|
2016-10-04
|
Remove RTCPeerConnection.removeStream for good.
|
| 1274479
|
Audio/Video: cubeb
|
P2
|
25
|
kinetik
|
RESO
|
2016-09-15
|
Crash in shutdownhang | stuck in winmm_stream_destroy and winmm_buffer_thread
|
| 1298189
|
WebRTC: Signaling
|
P2
|
21
|
mfroman
|
RESO
|
2016-08-30
|
Logging broken in media/webrtc/signaling/test
|
| 1278113
|
WebRTC
|
P2
|
21
|
nobody
|
RESO
|
2016-08-17
|
Intermittent test_peerConnection_verifyAudioAfterRenegotiation.html | Test timed out
|
| 1297808
|
WebRTC: Networking
|
P2
|
25
|
rjesup
|
RESO
|
2017-08-28
|
Latent (?) overflow in RTCPPacketInformation::AddApplicationData() could cause buffer overrun
|
| 1300810
|
WebRTC: Audio/Video
|
P2
|
22
|
rjesup
|
RESO
|
2016-09-15
|
Log enumeration results from cubeb in GetUserMedia
|
| 1284038
|
WebRTC
|
P3
|
35
|
andrew
|
RESO
|
2021-11-29
|
Intermittent browser/base/content/test/webrtc/browser_devices_get_user_media.js | expected notification recording-device-events - Got 2, expected 1
|
| 1298515
|
Audio/Video: MediaSt
|
P3
|
35
|
apehrson
|
RESO
|
2016-09-14
|
Intermittent dom/media/tests/mochitest/test_getUserMedia_audioCapture.html | Assertion count 50 is greater than expected range 0-0 assertions.
|
| 1275461
|
WebRTC: Networking
|
P3
|
|
docfaraday
|
RESO
|
2016-09-11
|
Intermittent test_peerConnection_simulcastOffer.html | Error in test execution: Error: Timed out waiting for frames timeout/<@http://mochi.test:8888/tests/dom/media/tests/mochitest/head.js:462:63 ... promise callback*timeout@http://mochi.test:8888/tests/d
|
| 1298586
|
WebRTC
|
P3
|
35
|
docfaraday
|
RESO
|
2017-04-08
|
Intermittent dom/media/tests/mochitest/test_peerConnection_simulcastOffer.html | sink is 1/2 width of source - got 50, expected 25
|
| 1293172
|
WebRTC: Networking
|
P3
|
|
drno
|
RESO
|
2016-08-09
|
Improve the DTLS handshake error log message
|
| 1293206
|
WebRTC
|
P3
|
35
|
drno
|
RESO
|
2016-08-17
|
Intermittent dom/media/tests/mochitest/test_peerConnection_basicAudioNATRelay.html | application crashed [@ libc-2.23.so + 0x35418]
|
| 1295415
|
WebRTC: Audio/Video
|
P3
|
35
|
jib
|
RESO
|
2016-08-17
|
Two coverity issues in MediaEngineWebRTC
|
| 1300446
|
Audio/Video: cubeb
|
P3
|
|
mozbugz
|
RESO
|
2016-09-05
|
Check cubeb_init so that a null context is not actually used
|
| 1197328
|
WebRTC
|
P5
|
55
|
palmieri.igor
|
RESO
|
2016-08-23
|
remove PR_snprintf calls in media/{webrtc,mtransport}/
|
| 1297911
|
WebRTC: Audio/Video
|
--
|
|
bonchiang
|
RESO
|
2016-08-26
|
Followup to bug 1286429 - if block needs to be braced
|
| 1303411
|
WebRTC: Audio/Video
|
--
|
|
dholbert
|
RESO
|
2016-09-17
|
VideoConduit.cpp:747:26: warning: '&&' within '||' [-Wlogical-op-parentheses]
|
| 1303419
|
WebRTC: Audio/Video
|
--
|
|
rjesup
|
RESO
|
2016-09-21
|
Crash in mozilla::MediaEngineWebRTCMicrophoneSource::FreeChannel
|
| 1279135
|
WebRTC: Audio/Video
|
P1
|
19
|
dminor
|
VERI
|
2016-08-31
|
RTCPeerConnection.removeTrack causes freezing video when removing an audio track
|
| 1286480
|
Audio/Video: GMP
|
P1
|
10
|
haftandilian
|
VERI
|
2017-01-30
|
[10.12] Widevine CDM always crashes on Amazon since upgrade to macOS Sierra
|
| 1278612
|
Audio/Video: cubeb
|
P1
|
10
|
kinetik
|
VERI
|
2016-09-09
|
OS X sound output device doesn't update when OS output device is changed
|
| 1295552
|
WebRTC: Networking
|
P1
|
15
|
mfroman
|
VERI
|
2016-08-31
|
no relay candidates after ice restarts
|
| 1293976
|
Audio/Video: MediaSt
|
P1
|
5
|
rjesup
|
VERI
|
2017-04-27
|
Crash in mozilla::MediaStreamGraph::NotifyOutputData since Firefox 49
|
| 1294095
|
WebRTC: Networking
|
P1
|
5
|
rjesup
|
VERI
|
2017-02-09
|
Crash in mozilla::DataChannelConnection::SctpDtlsOutput
|
| 1297083
|
Audio/Video: MediaSt
|
P1
|
10
|
rjesup
|
VERI
|
2016-09-01
|
Distorted/slowed-down audio with getUserMedia simultaneously in two tabs
|