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0e231c1
Update WebRTC code version (2025-03-16T04:06:29).
by webrtc-version-updater
· 12 hours ago
lkgr
main
master
9634696
Update WebRTC code version (2025-03-15T04:02:31).
by webrtc-version-updater
· 36 hours ago
99d1807
Move CorruptionDetectionFilterSettings into webrtc namespace.
by Joachim Reiersen
· 2 days ago
b401d3d
Add a callback to use StreamStats::rtp_stats as the source of truth.
by Jeremy Leconte
· 2 days ago
c839686
Introduce SctpOptions struct as argument to SctpTransport::Start
by Philipp Hancke
· 2 days ago
a22e244
Enable test that there are no duplicates in a CodecList
by Harald Alvestrand
· 2 days ago
c2f1bc7
Split EmulatedNetworkInterface into two
by Danil Chapovalov
· 2 days ago
c3cb9b7
Add .rustfmt.toml file
by Mirko Bonadei
· 2 days ago
64e4714
Add field trial that change g2g metric to use abs. capture time.
by Olov Brändström
· 2 days ago
b922597
dtls-in-stun: Fix late SDP answer
by Jonas Oreland
· 2 days ago
024ceea
Run DataChannelIntegrationTests w/o media enginge (for most tests)
by Jonas Oreland
· 2 days ago
d5d9526
Update freshness for the h-cc-pairs style document
by Danil Chapovalov
· 2 days ago
42a1cde
Update WebRTC code version (2025-03-14T04:09:46).
by webrtc-version-updater
· 3 days ago
41bc570
Enable 0Hz capture for test::FrameGeneratorCapturer.
by Jeremy Leconte
· 3 days ago
c2c052c
Refresh abseil-in-webrtc rules and documentation
by Danil Chapovalov
· 3 days ago
c13956e
Move all files in p2p/test to webrtc namespace
by Evan Shrubsole
· 3 days ago
0d3b384
Avoids log spam in AudioDeviceGeneric
by henrika
· 3 days ago
b6909ee
dtls-in-stun: Remove dtls restart in dtls-in-stun handling
by Jonas Oreland
· 3 days ago
2cf4492
Log VideoEncoderConfig::simulcast_layers[]
by Sergey Silkin
· 3 days ago
8990f2a
Move rtc_certificate and rtc_certificate_generator to webrtc namespace
by Evan Shrubsole
· 3 days ago
c713071
dtls_transport: Extend logging a tiny bit
by Jonas Oreland
· 3 days ago
dd99114
Add RtcEventProcessor TieBreaker for LoggedRtcpPacketSenderReport.
by Rasmus Brandt
· 3 days ago
d200661
Move sanitizer.h to webrtc namespace
by Evan Shrubsole
· 3 days ago
64b076f4
Move socket_address.h to webrtc namespace
by Evan Shrubsole
· 3 days ago
e1912c5
Fix issue with protobuf that is blocking perf tests.
by Mirko Bonadei
· 3 days ago
63f9cc0
Apply include-cleaner to call/
by Danil Chapovalov
· 3 days ago
1bd545a
Revert "fix: h26x packet buffer video artifacts"
by Philip Eliasson
· 3 days ago
e0cbfe4
Update WebRTC code version (2025-03-13T04:04:53).
by webrtc-version-updater
· 3 days ago
01f2ec6
Prevent VideoQualityMetricsReporter from crashing because of negative DataRate.
by Jeremy Leconte
· 4 days ago
7e54bb9
Remove unused constant
by Rasmus Brandt
· 4 days ago
762d77e
Revert the deletion of WebRTC-VideoH26xPacketBuffer flag.
by Henrik Boström
· 4 days ago
b2e88f9
Improve test outcomes for WebRTC-PayloadTypesInTransport
by Harald Alvestrand
· 4 days ago
cc5db7a
Remove deprecated non-Optional datachannel parameters
by Harald Alvestrand
· 4 days ago
31d20a6
Move moving_max_counter.h to webrtc namespace
by Evan Shrubsole
· 4 days ago
9278968
Delete deprecated functions in test EmulatedNetworkManagerInterface
by Danil Chapovalov
· 4 days ago
9ca15b6
Increase precision of high pass filter
by Gustaf Ullberg
· 4 days ago
eb835d0
Move ssl_stream_adapter.h to webrtc namespace
by Evan Shrubsole
· 4 days ago
97feff2
Move function_view.h to webrtc namespace
by Evan Shrubsole
· 4 days ago
b1d704b
Move port_interface.h to webrtc namespace
by Evan Shrubsole
· 4 days ago
9f114ece
Delete sigslot_tester
by Evan Shrubsole
· 4 days ago
c9f82e6
Update WebRTC code version (2025-03-12T04:09:25).
by webrtc-version-updater
· 5 days ago
851bcba
dtls-in-stun: Redo handling of when ice complete before dtls
by Jonas Oreland
· 5 days ago
6202bf1
Move ip_address.h to webrtc namespace
by Evan Shrubsole
· 5 days ago
a0a83aa
Allow test::EmulatedEndpoint to rebind to a different network
by Danil Chapovalov
· 5 days ago
0a76875
Skip assembling frames if the stream has been decoded past that point.
by Philip Eliasson
· 5 days ago
a3866aa
Refactoring: Look up codec vendor via RtpTransceiver
by Harald Alvestrand
· 5 days ago
8a9e08b
Add QP threshold experiment for H265.
by Henrik Boström
· 5 days ago
e1dc6d5
Deprecate sigslot_tester
by Evan Shrubsole
· 5 days ago
b000e8e
Move test_client.h to webrtc namespace
by Evan Shrubsole
· 5 days ago
f13201d
Temporarily disable two failing PCFullStackTests that hit DCHECKs.
by Henrik Boström
· 5 days ago
576ec5a
PipeWire capture: Clean latest_available_frame on capture stop
by Jan Grulich
· 5 days ago
9e1b0ec
Move stream.h to webrtc namespace
by Evan Shrubsole
· 6 days ago
377b69a
Move test_certificate_verifier.h to webrtc namespace
by Evan Shrubsole
· 6 days ago
6738e57
Move base64.h to webrtc namespace
by Evan Shrubsole
· 6 days ago
f36d45f
Disable 'rtc_unittests' on iOS simulator.
by Jeremy Leconte
· 6 days ago
d9232b3
Refactor VideoQualityMetricsReporter with no impact.
by Jeremy Leconte
· 6 days ago
47ddc6e
Move test_echo_server.h to webrtc namespace
by Evan Shrubsole
· 6 days ago
4887bc3
Move socket_address_pair.h to webrtc namespace
by Evan Shrubsole
· 6 days ago
0a97878
Fix build with Pipewire 1.4
by K900
· 6 days ago
4203617
dtls-in-stun: Add new testcase
by Jonas Oreland
· 6 days ago
58c273b
Move bit_buffer.h to webrtc namespace
by Evan Shrubsole
· 7 days ago
c3fba45
Update WebRTC code version (2025-03-10T04:05:53).
by webrtc-version-updater
· 7 days ago
49ba3e2
Move byte_order.h to webrtc namespace
by Evan Shrubsole
· 7 days ago
5015b4d
Move crc32.h to webrtc namespace
by Evan Shrubsole
· 7 days ago
ccf5114
Update WebRTC code version (2025-03-09T04:08:42).
by webrtc-version-updater
· 8 days ago
cec4b95
Update WebRTC code version (2025-03-08T04:06:00).
by webrtc-version-updater
· 8 days ago
ee5e7e5
sdp munging: detect audio nack and changes to opus FEC and DTX
by Philipp Hancke
· 9 days ago
b101a7e
doc: SRTP_AES128_CM_HMAC_SHA1_32 is disabled by default
by Philipp Hancke
· 9 days ago
f45f111
Sequence checker shouldn't guard itself
by Artem Titov
· 9 days ago
4df984d
dtls-in-stun: Add callback for disabling of piggybacking
by Jonas Oreland
· 9 days ago
1816152
stats: do not expose ufrag before the first setRemoteDescription
by Philipp Hancke
· 10 days ago
2167733
Format fuzzer documentation
by Björn Terelius
· 10 days ago
3af24f3
Ensure one can build audioproc_f with apm_debug_dump=true
by Lionel Koenig Gélas
· 10 days ago
a5da882
Update fuzzer documentation
by Björn Terelius
· 10 days ago
e733ebc
dtls1.3 - patch 6
by Jonas Oreland
· 10 days ago
9789a44
Catch also IllegalStateException when creating hw encoder
by Liad Rubin
· 10 days ago
4a3dffe
Add SurfaceEglRenderer ctor with VideoFrameDrawer and propagate to super class
by Liad Rubin
· 10 days ago
924b73c
Checkout base/ only if fuzzing is enabled
by Byoungchan Lee
· 10 days ago
fb8032e
Move non-PT-assigning codec collection out of VoiceMediaEngine
by Harald Alvestrand
· 10 days ago
5568c98
Update WebRTC code version (2025-03-06T04:08:40).
by webrtc-version-updater
· 10 days ago
34bfdcd
Roll chromium_revision d4f6168ad3..a1723b7af1 (1428571:1428676)
by chromium-webrtc-autoroll
· 11 days ago
0e8160a
Roll chromium_revision 0b0f98e256..d4f6168ad3 (1428408:1428571)
by chromium-webrtc-autoroll
· 11 days ago
31f317e
Roll chromium_revision a82e0ea13a..0b0f98e256 (1428241:1428408)
by chromium-webrtc-autoroll
· 11 days ago
f1a3142
IWYU pc/test/
by Philipp Hancke
· 11 days ago
72a1d7d
Deprecate the cricket::Codec access functions in MediaEngine
by Harald Alvestrand
· 11 days ago
d9e96ef
Roll chromium_revision 547cac853e..a82e0ea13a (1428124:1428241)
by chromium-webrtc-autoroll
· 11 days ago
cba7209
Fix -Wthread-safety-analysis warning on sequence_checker.h
by Zequan Wu
· 11 days ago
3a7525e
dtls-in-stun: Improve test coverage
by Jonas Oreland
· 11 days ago
683a980
Implement FrameGeneratorCapturer::RequestRefreshFrame.
by Jeremy Leconte
· 11 days ago
55cb716
Use a vector of scalability modes in VideoQualityMetricsReporter.
by Jeremy Leconte
· 11 days ago
eec79a0
dtls-in-stun: Only read IceConfig.dtls_handshake_in_stun in 1 place.
by Jonas Oreland
· 11 days ago
b3373d8
Update WebRTC code version (2025-03-05T04:05:48).
by webrtc-version-updater
· 12 days ago
f4687eaf4
Roll chromium_revision 887a261e75..547cac853e (1427905:1428124)
by chromium-webrtc-autoroll
· 12 days ago
74c5a21
Support negative SenderCaptureTimeOffset in TransformableVideoFrameInterface
by Guido Urdaneta
· 12 days ago
a6b939a
Roll chromium_revision 3678ad0b60..887a261e75 (1427739:1427905)
by chromium-webrtc-autoroll
· 12 days ago
243485a
IWYU p2p/client and p2p/stun_prober
by Philipp Hancke
· 12 days ago
7d11c9f
Roll chromium_revision 767b207241..3678ad0b60 (1426198:1427739)
by chromium-webrtc-autoroll
· 12 days ago
fa3b0a5
Add DEPS needed for `ninja <...> all`
by Björn Terelius
· 12 days ago
f0964b5
Roll chromium_revision 36f59af467..767b207241 (1424618:1426198)
by Björn Terelius
· 12 days ago
a49e78a
Make WebRTC-Video-H26xPacketBuffer enabled-by-default (delete flag).
by Henrik Boström
· 12 days ago
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