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51dbe82
setOfferedHeaderExtensions: stop any filtered extension
by Philipp Hancke
· 3 hours ago
lkgr
main
master
e9c3e51
Add a DEPS hook to download llvm-cov and llvm-profdata based on .gclient custom_vars.
by Björn Terelius
· 18 hours ago
c7a0620
Add an ICE switch reason for a switch requested by an application.
by Sameer Vijaykar
· 20 hours ago
95d12ad
Create unit test for the population of capture_start_ntp_time
by Harald Alvestrand
· 22 hours ago
4b0d6f9
Upgrade Linux MSan to Focal
by Tom Anderson
· 2 days ago
b0e1cb2
Adds WebRTC.DesktopCapture.Win.DirectXCapturerResult UMA
by henrika
· 3 days ago
fd29662
Fix typo in histogram name
by Johannes Kron
· 4 days ago
5e7301f
Remove rid and rrid from list of extensions that can be used for audio
by Philipp Hancke
· 4 days ago
c6ff4bc
Do not transfer ownership of codecs to tester
by Sergey Silkin
· 4 days ago
be9b576
Move video video receiver transformable frame to modules/rtc_rtcp/source
by Tony Herre
· 4 days ago
b311f6a
Add UMA histograms to track usage of fullscreen detection
by Johannes Kron
· 4 days ago
85abbdf
RtcEventLogImpl: Add test cases
by Xuanxi Leng
· 4 days ago
d1831cb
Treat non DTLS/SCTP Protocol Based Data Channels as Unsupported Media
by Dor Hen
· 4 days ago
f0be3be
Add pipewire/portal video capture support
by Michael Olbrich
· 5 days ago
fad9a6d
Delete deprecated Create method and config from AudioCodingModule
by Henrik Lundin
· 5 days ago
101c6aa
Remove leftover function signatures.
by Fredrik Solenberg
· 5 days ago
6c60f72
Refactor video codec testing stats
by Sergey Silkin
· 5 days ago
97d1c34
Enable rotation tests marked as expected failures
by Andreas Pehrson
· 5 days ago
65ab5fd
Cleanup RemoteEstimatorProxy::IncomingPacket
by Danil Chapovalov
· 5 days ago
ba846cc
Add a test that shows when channel_receive fires RR
by Harald Alvestrand
· 6 days ago
84f7569
Break apart AudioCodingModule and AcmReceiver
by Henrik Lundin
· 6 days ago
c5455e7
Allow RTX ssrc to be updated on receive streams
by Per K
· 6 days ago
be03c09
Only serialize non-stopped RTP header extensions
by Philipp Hancke
· 6 days ago
1f206b8
Use ArrayView in the IncomingRtcpPacket function.
by Harald Alvestrand
· 6 days ago
16a8792
Propagate received video csrcs to encodedframe metadata
by Tony Herre
· 6 days ago
0507fbd
[Desktop Capture] Remove disabled test.
by mark a. foltz
· 6 days ago
cb4b0a6
Check FMA3 support before use it in SincResampler
by Henrik Lundin
· 7 days ago
0b7184c
Add possibility to set MetricsSet metadata.
by Mirko Bonadei
· 7 days ago
217b384
Remove rtp header extension from config of Call audio and video receivers
by Per K
· 7 days ago
3541732
Add a Config struct to AcmReceiver, and a ctor using it
by Henrik Lundin
· 7 days ago
3274051
Add method to get FD for physical socket
by Artem Titov
· 7 days ago
d78f8e7
Fix doc path
by Artem Titov
· 7 days ago
e592283
Add 'metadata' field to MetricsSet proto.
by Mirko Bonadei
· 7 days ago
a617867
Reland "Migrate WebRTC documentation to new renderer"
by Artem Titov
· 7 days ago
2367103
Add android-tiramisuprivacysandbox to DEPS.
by Mirko Bonadei
· 7 days ago
3963a95
Enforce policy that SDP munging requires special approval
by Harald Alvestrand
· 8 days ago
4822031
shared_screencast_stream: Set WL capturer id
by Salman
· 8 days ago
9214718
Roll chromium_revision 2fd12db504..e182675fbb (1098462:1098562)
by chromium-webrtc-autoroll
· 8 days ago
dad91a6
Send periodic TransportFeedback based on extension version
by Per K
· 8 days ago
2ded55e
Cleanup Thread::BlockingCall
by Danil Chapovalov
· 8 days ago
8155227
sdp: add rtcp-fb:* lines for common feedback
by Philipp Hancke
· 8 days ago
68564bb
[infra] Clean up mb_config.pyl after reclient migration
by Junji Watanabe
· 8 days ago
942abaa
Roll chromium_revision 7faf2e057e..2fd12db504 (1098356:1098462)
by chromium-webrtc-autoroll
· 8 days ago
f6a1f7e
Roll chromium_revision ff5a43c15d..7faf2e057e (1098253:1098356)
by chromium-webrtc-autoroll
· 9 days ago
863a07f
Update WebRTC code version (2023-01-29T04:02:18).
by webrtc-version-updater
· 9 days ago
dc7a14b
Roll chromium_revision 5195bd9e41..ff5a43c15d (1098136:1098253)
by chromium-webrtc-autoroll
· 10 days ago
6ff7713
base_capturer_pipewire: Send frames via callback
by Salman
· 10 days ago
bf216a7
Roll chromium_revision d9876846b5..5195bd9e41 (1097987:1098136)
by chromium-webrtc-autoroll
· 11 days ago
26340b0
desktop_capturer: Support frame rate negotiation via pipewire
by Salman
· 11 days ago
4c4566a
Roll chromium_revision 6da6f5ebed..d9876846b5 (1097837:1097987)
by chromium-webrtc-autoroll
· 11 days ago
64ce699
Propagate Video CSRCs modified by an insertable streams frame transform
by Tony Herre
· 11 days ago
db20831
Update RTP timestamp based on capture timestamp when audio send stream is resumed.
by Jakob Ivarsson
· 11 days ago
9d21eb3
Roll chromium_revision 1f33300013..6da6f5ebed (1097480:1097837)
by chromium-webrtc-autoroll
· 11 days ago
a2653bc
Export some more symbols for use in chromium tests.
by Sameer Vijaykar
· 11 days ago
ca1cfd4
Add missing `src/third_party/jdk11` dependency and roll chromium into webrtc
by landrey
· 11 days ago
0f2ce5c
Revert "Migrate WebRTC documentation to new renderer"
by Artem Titov
· 12 days ago
b0d8a37
Ensure CallTest derived tests per default set min/max audio bitrate.
by Per K
· 12 days ago
6c032cb
in rtcp::TransportFeedback do not memorise all described packet
by Danil Chapovalov
· 12 days ago
dcb09ff
Reset encoder when audio send stream is stopped.
by Jakob Ivarsson
· 12 days ago
3eceaf4
Migrate WebRTC documentation to new renderer
by Artem Titov
· 12 days ago
94d5f6a
Add missing include
by Bjorn Terelius
· 12 days ago
66efab2
Measure RTCPMuxPolicy at time of connect
by Philipp Hancke
· 12 days ago
73e0cc8
Delete unused Audio Bwe integration test.
by Per K
· 12 days ago
cfbb247
Update WebRTC code version (2023-01-26T04:01:54).
by webrtc-version-updater
· 12 days ago
e15b9ff
Add a basic unittest for webrtc::voe::ChannelReceive
by Harald Alvestrand
· 13 days ago
664cf14
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
by Per K
· 13 days ago
7a67dce
prefer absl::optional for rtx-time
by Philipp Hancke
· 13 days ago
7c43d24
Roll chromium_revision a484be4b74..e5191e93ab (1096680:1096792)
by chromium-webrtc-autoroll
· 13 days ago
5adc2b6
Correct RTCAudioPlayoutStats type and add kind field.
by Fredrik Hernqvist
· 13 days ago
0c1c972
Fix gtest-output and resultdb for fuchsia
by Christoffer Jansson
· 13 days ago
5683a12
Increase expiration timeout for Perf bots
by landrey
· 13 days ago
5671c64
Stop overriding extensions in rampup tests
by Per K
· 13 days ago
8773349
Update win10 mixins.
by Jeremy Leconte
· 13 days ago
92dcc2d
Roll chromium_revision 98774e2693..a484be4b74 (1096567:1096680)
by chromium-webrtc-autoroll
· 13 days ago
10f1bf3
Remove unused enum `FrameCombiner::LimiterType`
by Alessio Bazzica
· 13 days ago
f2a083f
Revert "Delete PacketReceiver::DeliverPacket from all implementations"
by Andrey Logvin
· 13 days ago
07577b5
Update WebRTC code version (2023-01-25T04:11:56).
by webrtc-version-updater
· 13 days ago
508979b
Roll chromium_revision 87e5077aae..98774e2693 (1096404:1096567)
by chromium-webrtc-autoroll
· 13 days ago
cc1c932
Roll chromium_revision 78e2f876c5..87e5077aae (1096262:1096404)
by chromium-webrtc-autoroll
· 14 days ago
ea36cc2
Roll chromium_revision 632253d282..78e2f876c5 (1096118:1096262)
by chromium-webrtc-autoroll
· 14 days ago
897ea04
Delete PacketReceiver::DeliverPacket from all implementations
by Per K
· 14 days ago
0540627
SVC: Add test for SVC fallback
by Florent Castelli
· 14 days ago
1e43ce6
Roll chromium_revision f8b9751f30..632253d282 (1095656:1096118)
by chromium-webrtc-autoroll
· 14 days ago
e2c29c5
Use PacketReceiver::DeliverRtpPacket in RtpReplayer
by Per K
· 2 weeks ago
0793ee7
Remove FakePortAllocator's dependency on ScopedKeyValueConfig.
by Sameer Vijaykar
· 2 weeks ago
ace52a8
[infra] Remove CQ shadow builders with reclient
by Junji Watanabe
· 2 weeks ago
2810c14
[infra] Add todo for reclient migration cleanup
by Junji Watanabe
· 2 weeks ago
22821de
Make capture timestamp optional in ADM.
by Jakob Ivarsson
· 2 weeks ago
6e62729
Roll chromium_revision cdf104c2c9..f8b9751f30 (1095545:1095656)
by chromium-webrtc-autoroll
· 2 weeks ago
5b55b27
Version 3: Various changes on the pre-echo delay estimator:
by Jesús de Vicente Peña
· 2 weeks ago
438b5b4
WebRtcVideoChannel creates default stream with dummy SSRC on received RTX packet.
by Per K
· 2 weeks ago
05ce032
Remove the //rtc_base target
by Florent Castelli
· 2 weeks ago
9ad10bc
Only generate codec stats for the voice send and receive codec
by Philipp Hancke
· 2 weeks ago
41748c9
Roll chromium_revision a3aaddaf0c..cdf104c2c9 (1095442:1095545)
by chromium-webrtc-autoroll
· 2 weeks ago
d3b9e71
Migrate linux_libfuzzer_rel to use reclient
by Junji Watanabe
· 2 weeks ago
d506651
Update WebRTC code version (2023-01-23T04:11:41).
by webrtc-version-updater
· 2 weeks ago
5e7ae14
Roll chromium_revision 0c3ca8fbc7..a3aaddaf0c (1095332:1095442)
by chromium-webrtc-autoroll
· 2 weeks ago
c8270b1d
Roll chromium_revision e0c02c1406..0c3ca8fbc7 (1095203:1095332)
by chromium-webrtc-autoroll
· 2 weeks ago
d2c15aa
Roll chromium_revision 92aea4500e..e0c02c1406 (1095006:1095203)
by chromium-webrtc-autoroll
· 3 weeks ago
62ba379
Add some RTC_EXPORT needed by Chromium.
by Mirko Bonadei
· 3 weeks ago
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