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45d8674
Update WebRTC code version (2023-10-04T04:04:01).
by webrtc-version-updater
· 5 hours ago
main
master
4408575
Reland "Enable SRTP GCM ciphers by default"
by Saúl Ibarra Corretgé
· 12 hours ago
8db8824
Fix use-of-uninitialized-value and integer-overflow issues reported by chromium fuzz testing
by qwu16
· 20 hours ago
lkgr
40ed3ff
Improved event tracing in FrameCadenceAdapter
by henrika
· 22 hours ago
9209b50
Update WebRTC code version (2023-10-03T04:04:07).
by webrtc-version-updater
· 29 hours ago
3725398
Roll chromium_revision 6a1112fe10..1a86984556 (1204286:1204416)
by chromium-webrtc-autoroll
· 32 hours ago
4e5ae32
Roll chromium_revision 45a884046b..6a1112fe10 (1203912:1204286)
by chromium-webrtc-autoroll
· 36 hours ago
a557468
dcsctp: Only process meaningful FORWARD-TSN
by Victor Boivie
· 2 days ago
8e007ba
Remove field trial WebRTC-Turn-AllowSystemPorts
by Harald Alvestrand
· 2 days ago
bce7ce7
Adds support for tracking OnFrame PostTask delta times
by henrika
· 2 days ago
4d2d215
Roll chromium_revision cd02ca7e29..45a884046b (1203806:1203912)
by chromium-webrtc-autoroll
· 2 days ago
3272a99
Update WebRTC code version (2023-10-02T04:11:24).
by webrtc-version-updater
· 2 days ago
316c853
Roll chromium_revision b294093253..cd02ca7e29 (1203697:1203806)
by chromium-webrtc-autoroll
· 3 days ago
9cefc24
Update WebRTC code version (2023-10-01T04:06:57).
by webrtc-version-updater
· 3 days ago
cad0bfa
Roll chromium_revision 7be4bb3958..b294093253 (1203597:1203697)
by chromium-webrtc-autoroll
· 4 days ago
8212061
Update WebRTC code version (2023-09-30T04:13:02).
by webrtc-version-updater
· 4 days ago
7d8ef10
Roll chromium_revision a53f87ab6b..7be4bb3958 (1203480:1203597)
by chromium-webrtc-autoroll
· 4 days ago
11f087d
Roll chromium_revision 9b3bfd6cf6..a53f87ab6b (1203365:1203480)
by chromium-webrtc-autoroll
· 4 days ago
02d5ba1
Roll chromium_revision 3882779f44..9b3bfd6cf6 (1203188:1203365)
by chromium-webrtc-autoroll
· 5 days ago
dd15070
Adopt EglThread in EglRenderer once again.
by Linus Nilsson
· 5 days ago
7d3eb10
Roll chromium_revision 2cd0974cfa..3882779f44 (1202869:1203188)
by chromium-webrtc-autoroll
· 5 days ago
c27034b
Revert "Expose getCapabilities/setCodecPreferences for objc"
by Manashi Sarkar
· 5 days ago
a2f30e1
Expose getCapabilities/setCodecPreferences for objc
by David Liu
· 5 days ago
012c5a3
Remove more Codec-related templating in MediaSession
by Philipp Hancke
· 5 days ago
34ec5c3
Clear PacketBuffer on large negative jumps at the start of the video stream
by Danil Chapovalov
· 5 days ago
e9f9c28
Set permissions on experiments/field_trials.py
by Harald Alvestrand
· 5 days ago
a61b334
Relax the field trial policy to not require an open bug
by Emil Lundmark
· 5 days ago
5e252b7
Register WebRTC-Bwe-SubtractAdditionalBackoffTerm
by Emil Lundmark
· 5 days ago
191dd88
Update WebRTC code version (2023-09-29T04:02:17).
by webrtc-version-updater
· 5 days ago
ea32ab1
Roll chromium_revision 2506ef1631..2cd0974cfa (1202675:1202869)
by chromium-webrtc-autoroll
· 5 days ago
5ee308b
Roll chromium_revision 116647225e..2506ef1631 (1202523:1202675)
by chromium-webrtc-autoroll
· 6 days ago
c2bbe4b
Revert "Enable SRTP GCM ciphers by default"
by Manashi Sarkar
· 6 days ago
ebe207f
Add field trial for enabling SSL client hello extension permutation
by Philipp Hancke
· 6 days ago
a1475c2
Roll chromium_revision f7fb707ebb..116647225e (1201480:1202523)
by chromium-webrtc-autoroll
· 6 days ago
a93f581
dcsctp: Don't generate FORWARD-TSN across stream resets
by Victor Boivie
· 6 days ago
d863386
Enable SRTP GCM ciphers by default
by Saúl Ibarra Corretgé
· 6 days ago
98e71f5
Subtract an additional 5kbps of the bitrate when backing off.
by Björn Terelius
· 6 days ago
332c56f
MediaSession: ensure transport description factory exists
by Philipp Hancke
· 6 days ago
bbc7711
Reduce log verbosity in codec selection implementation
by Florent Castelli
· 6 days ago
2d508f1
Deprecate old names for EncodedImage::RtpTimestamp accessor and setter
by Danil Chapovalov
· 6 days ago
f97058e
Move static functions in media_session into anonymous namespace
by Philipp Hancke
· 6 days ago
21b5ec1
Add AV1 singlecast bitrate limits
by Sergey Silkin
· 7 days ago
d12e759
Add instructions for adding and removing field trials
by Emil Lundmark
· 7 days ago
83894d3
Fire MaybeSignalReadyToSend in a PostTask when recursive
by Harald Alvestrand
· 7 days ago
4001cc7
Populate field trial registry
by Emil Lundmark
· 7 days ago
5543a96
Add sub-command for validating that field trials conforms to the policy
by Emil Lundmark
· 7 days ago
9686a4b
Add sub-command for listing expired field trials
by Emil Lundmark
· 7 days ago
b7fca15
Refactor global variables to be immutable
by Emil Lundmark
· 7 days ago
7d81e18
Reformat field_trials.py to follow PEP-8
by Emil Lundmark
· 7 days ago
0505115
Pass the correct abs_capture_timestamp while cloning audio frame
by Palak Agarwal
· 8 days ago
3218d74
Roll chromium_revision ae93f006ea..f7fb707ebb (1201360:1201480)
by chromium-webrtc-autoroll
· 8 days ago
2bf1b99
Make CreateOffer/CreateAnswer return RTCErrorOr<SessionDescription>
by Philipp Hancke
· 8 days ago
06fbe63
dcsctp: Exit deferred stream reset on FORWARD-TSN
by Victor Boivie
· 8 days ago
259d95f
Roll chromium_revision 3a03dc546a..ae93f006ea (1201238:1201360)
by chromium-webrtc-autoroll
· 8 days ago
bfc2a35
Remove more codec-related templating
by Philipp Hancke
· 8 days ago
7829daf
Update WebRTC code version (2023-09-26T04:02:19).
by webrtc-version-updater
· 8 days ago
78c119c
Remove check on last_packet_received_time_ as it's no longer used.
by Ying Wang
· 8 days ago
9b5d795
Roll chromium_revision eb86ccf4cd..3a03dc546a (1201083:1201238)
by chromium-webrtc-autoroll
· 8 days ago
0ca4d62
Roll chromium_revision ae69785833..eb86ccf4cd (1200919:1201083)
by chromium-webrtc-autoroll
· 9 days ago
77df7ff
dcsctp: Cleanup use of matchers
by Victor Boivie
· 9 days ago
7892f05
Configure Pylint to follow PEP-8
by Emil Lundmark
· 9 days ago
7d1aff6
Unify RTP payload type validity checking
by Philipp Hancke
· 9 days ago
6bf2d31
Change PortInterface::Type to string_view and make type_ member const
by Tommi
· 9 days ago
070d386
Roll chromium_revision 286dbc6af0..ae69785833 (1200141:1200919)
by chromium-webrtc-autoroll
· 9 days ago
29d4a01
Reland: use loss based bwe v2 in the start phase.
by Diep Bui
· 9 days ago
ba97eec
Add string_view overload for Wrap method
by Artem Titov
· 9 days ago
b4d4bbc
Revert "Clean up last_packet_received_time_ as it's no longer used."
by Björn Terelius
· 9 days ago
9c58483
Rename EncodedImage property Timetamp to RtpTimestamp
by Danil Chapovalov
· 10 days ago
bbf27e0
Remove NSApplicationActivateIgnoringOtherApps
by Johannes Kron
· 10 days ago
850296b
Reapply "dcsctp: Negotiate zero checksum"
by Victor Boivie
· 11 days ago
63c50f5
Update WebRTC code version (2023-09-23T04:12:34).
by webrtc-version-updater
· 11 days ago
2f4bc64
Clean up last_packet_received_time_ as it's no longer used.
by Ying Wang
· 11 days ago
d2f4cf9
Roll chromium_revision c066d24408..286dbc6af0 (1200027:1200141)
by chromium-webrtc-autoroll
· 11 days ago
4aa2b40
Revert "Use loss based bwe v2 in the start phase."
by Diep Bui
· 11 days ago
b6c7ddd
Use loss based bwe v2 in the start phase.
by Diep Bui
· 12 days ago
b6ea0b2
Direcly call configure_reclient_cfgs.py instead of indirectly via fetch_reclient_cfgs.py
by Björn Terelius
· 12 days ago
1db3980
Remove upper_link_capacity from loss_based_bwe_v2.
by Diep Bui
· 12 days ago
70eec6d
Configure YAPF to follow PEP-8 altogether
by Emil Lundmark
· 12 days ago
4b39e86
Update WebRTC code version (2023-09-22T04:11:01).
by webrtc-version-updater
· 12 days ago
8500974
Roll chromium_revision 4b9a788892..c066d24408 (1199866:1200027)
by chromium-webrtc-autoroll
· 12 days ago
047eeb4
Roll chromium_revision f473cfebae..4b9a788892 (1199635:1199866)
by chromium-webrtc-autoroll
· 12 days ago
e3e030e
Roll chromium_revision 54d127d7c9..f473cfebae (1199499:1199635)
by chromium-webrtc-autoroll
· 13 days ago
e887cbe
Roll chromium_revision b3921f4990..54d127d7c9 (1198700:1199499)
by chromium-webrtc-autoroll
· 13 days ago
7ee64bd
Remove the upper link capacity usage in the loss based bwe.
by Diep Bui
· 13 days ago
c951d1b
audio: fix some typos
by Alfred E. Heggestad
· 13 days ago
6fc4d97
Make WEBRTC_UNSAFE_FUZZER_MODE dependent only on use_fuzzing_engine
by Greg Thompson
· 14 days ago
46da472
Revert "mac: Work around an inccorect availability annotation in the 13.3 SDK"
by Avi Drissman
· 14 days ago
5551776
Reject attempts to change the media kind for a m-line with a previously used mid
by Philipp Hancke
· 14 days ago
ec82627
Look through all candidates before falling back to default packetization
by Emil Lundmark
· 14 days ago
f14dfed
Move codecs() to MediaContentDescription
by Philipp Hancke
· 2 weeks ago
ae82df7
Add codec name H265 to support H265 in WebRTC
by qwu16
· 2 weeks ago
9596002
Update WebRTC code version (2023-09-20T04:02:40).
by webrtc-version-updater
· 2 weeks ago
dacd1fa
Roll chromium_revision b63463aa70..b3921f4990 (1198567:1198700)
by chromium-webrtc-autoroll
· 2 weeks ago
7917525
Roll chromium_revision dfc3d16403..b63463aa70 (1198356:1198567)
by chromium-webrtc-autoroll
· 2 weeks ago
4b583c7
Roll chromium_revision eef62e8a0c..dfc3d16403 (1197906:1198356)
by chromium-webrtc-autoroll
· 2 weeks ago
f8c70c9
fix: Handle out-of-range device index after GetDevicesInfo
by Youfa
· 2 weeks ago
2e7ed0d
Roll chromium_revision 6ac7929166..eef62e8a0c (1190797:1197906)
by Jeremy Leconte
· 2 weeks ago
e14d122
Remove deprecated SendRtp and SendRtcp functions
by Harald Alvestrand
· 2 weeks ago
090699a
Delete deprecated RtpSource timestamp_ms constructor and accessor
by Danil Chapovalov
· 2 weeks ago
7cdf66f
Add local capture clock offset to video RtpPacketInfos
by Olov Brändström
· 2 weeks ago
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