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webrtc
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HEAD
625e1d9
VP8 ref finder unittest cleanup
by philipel
· 3 hours ago
master
86f04ad
Populate “fractionLost” stats for remote inbound rtp streams
by Di Wu
· 4 hours ago
6512af0
Add root doc site definition for WebRTC documentation
by Artem Titov
· 4 hours ago
f57f2cd
doc: add M88/M89 release notes
by Philipp Hancke
· 4 hours ago
64f7da0
Use input_state to get pixels for single active stream.
by Åsa Persson
· 5 hours ago
31e06cb
addIceCandidate: prefer ice candidate sdpMid over sdpMLineIndex
by Philipp Hancke
· 7 hours ago
df6d4ca
Prevent TURN from connecting to ports < 1024 (except 443)
by Harald Alvestrand
· 7 hours ago
bfc3ef0
Mention ninja all in webrtc development documantation.
by Andrey Logvin
· 8 hours ago
4785402
Replace RecursiveCriticalSection with Mutex in ProcessThreadImpl
by Niels Möller
· 8 hours ago
1e2da37
Update WebRTC code version (2021-02-28T04:02:30).
by webrtc-version-updater
· 2 days ago
ac615b6
Update WebRTC code version (2021-02-27T04:03:14).
by webrtc-version-updater
· 3 days ago
c96aa30
Minor refactoring of RtpVideoSender
by Erik Språng
· 3 days ago
484acf2
Add ability to configure sampling rate for input/output video dumps in PC level framework
by Artem Titov
· 3 days ago
9d9b8de
Reland "Rename SIGNALING and WORKER to PRIMARY and SECONDARY"
by Mirko Bonadei
· 3 days ago
376cf38
Replace RecursiveCriticalSection with Mutex in EmulatedEndpointImpl
by Niels Möller
· 3 days ago
lkgr
07a01d0
Revert "Rename SIGNALING and WORKER to PRIMARY and SECONDARY"
by Mirko Bonadei
· 4 days ago
a37f2bd
Rename SIGNALING and WORKER to PRIMARY and SECONDARY
by Harald Alvestrand
· 4 days ago
2aeab5e
Make the PC proxy invoke LookupDtlsTransportByMid on the network thread
by Tomas Gunnarsson
· 4 days ago
5cfcf22
modules/desktop_capture: replace memcpy with libyuv::CopyPlane
by Zhaoliang Ma
· 4 days ago
258e989
Use default ResolutionBitrateLimits for simulcast with one active stream if not configured
by Åsa Persson
· 4 days ago
1124ed1
Communicate encoder resolutions via rtc::VideoSinkWants.
by Henrik Boström
· 4 days ago
bb52bdf
Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium"
by Johannes Kron
· 4 days ago
cd5127b
Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium"
by Mirko Bonadei
· 4 days ago
49b2792
Update WebRTC code version (2021-02-25T04:03:13).
by webrtc-version-updater
· 5 days ago
c500977
Change the safe SCTP MTU size to 1191
by Tomas Gunnarsson
· 5 days ago
dfe1971
Enable use of rtc::SystemTimeNanos() provided by Chromium
by Johannes Kron
· 5 days ago
dac39c5
Reland "Add test for odd sizes with spatial layers"
by Sergio Garcia Murillo
· 5 days ago
61d1773
Remove deactivated RTP modules from PacketRouter map.
by Erik Språng
· 5 days ago
451a8af
Feed the clock skew to AbsoluteCaptureTimeReceiver in audio receiver.
by Minyue Li
· 5 days ago
cd0373f
peerconnection: add was-ever-connected boolean flag
by Philipp Hancke
· 5 days ago
2ee9415
AndroidVideoDecoder: Ignore format updates with zero dimensions
by Raman Budny
· 5 days ago
eaedde7
Remove old workaround in PacingController
by Erik Språng
· 5 days ago
0093a38
Fix low-latency renderer with unset playout delays
by Johannes Kron
· 5 days ago
198299c
Roll src/third_party/libjpeg_turbo/ fa0de0767..7b4981b65 (2 commits)
by Keiichi Enomoto
· 5 days ago
77475ec
Delete unused sigslot variables.
by Lahiru Ginnaliya Gamathige
· 6 days ago
c32f00e
Remove usage of sigslot and replace with a function pointer.
by Lahiru Ginnaliya Gamathige
· 6 days ago
5fec23c
Update WebRTC code version (2021-02-24T04:02:50).
by webrtc-version-updater
· 6 days ago
8af6b49
Populate jitter stats for video RTP streams
by Di Wu (RP Room Eng)
· 6 days ago
373bb7b
Don't use SystemTimeNanos() for current wallclock time on WINUWP
by Johannes Kron
· 6 days ago
d0844a8
Revert "Add test for odd sizes with spatial layers"
by Florent Castelli
· 6 days ago
09226fc
Disable high-pass filtering of the AEC reference
by Gustaf Ullberg
· 7 days ago
e5caa9e
Update WebRTC code version (2021-02-23T04:03:09).
by webrtc-version-updater
· 7 days ago
90ea0a6
AV1: Change multithreading, speed, qp settings
by Jerome Jiang
· 7 days ago
6fe3fa1
Add test for odd sizes with spatial layers
by Sergio Garcia Murillo
· 7 days ago
28547e9
Fix typos in network emulation default routing
by Artem Titov
· 7 days ago
d44532a
Delete unused sigslot SignalAddressReady and MSG_ID_ADDRESS_BOUND
by Niels Möller
· 7 days ago
77ee854
Extract sequencing from RtpSender
by Erik Språng
· 7 days ago
7013b3b
Add deprecation section to webrtc style guide
by Danil Chapovalov
· 7 days ago
e904161
Replace RTC_DEPRECATED with ABSL_DEPRECATED
by Danil Chapovalov
· 7 days ago
8ef1d7b
Add a missing lock in VideoBroadcaster::OnDiscardedFrame().
by Mirta Dvornicic
· 7 days ago
2bfddf7
Add thread annotations and docs in ProcessThreadImpl.
by Niels Möller
· 7 days ago
bc9dc5a
Upload all values instead of only mean and err into histograms
by Artem Titov
· 7 days ago
3d37e06
Introduce default routes for network emulation
by Artem Titov
· 7 days ago
1dd94a0
Use pixels from single active stream if set in CanDecreaseResolutionTo
by Åsa Persson
· 7 days ago
42dd9bc
Add documentation about DefaultVideoQualityAnalyzer
by Artem Titov
· 7 days ago
a21ea29
Update WebRTC code version (2021-02-20T04:03:37).
by webrtc-version-updater
· 10 days ago
04a6529
AV1: set superblock to 64x64 for 720p 4 threads.
by Jerome Jiang
· 10 days ago
1fbff10
In RtpVideoStreamReceiver change way to track time for the last received packet.
by Danil Chapovalov
· 10 days ago
f3dc47e
Ending a statement with a semicolon
by Zhaoliang Ma
· 10 days ago
da20c73
Add build argument rtc_exclude_system_time
by Johannes Kron
· 10 days ago
072c008
Reland "Replace RecursiveCriticalSection with Mutex in RTCAudioSession."
by Niels Möller
· 10 days ago
ae096ef
Remove log message if balanced/cpu speed field trial is not set.
by Åsa Persson
· 10 days ago
16359f6
Delay creation of decoders until they are needed
by Johannes Kron
· 10 days ago
c9b9930
Add L2T3 K-SVC structure
by Emil Lundmark
· 10 days ago
753c76a
Update WebRTC code version (2021-02-19T04:02:42).
by webrtc-version-updater
· 11 days ago
735e33f
Add S3T3 video scalability structure
by Danil Chapovalov
· 11 days ago
0f71871
Reland "Batch assign RTP seq# for all packets of a frame."
by Erik Språng
· 11 days ago
9915db3
Move Call's histogram reporting code into destructor.
by Tomas Gunnarsson
· 11 days ago
17f914c
Revert "Batch assign RTP seq# for all packets of a frame."
by Jeremy Leconte
· 11 days ago
e11b4ae
doc: show how to build the fuzzers
by Philipp Hancke
· 12 days ago
86d3725
Update WebRTC code version (2021-02-18T04:03:24).
by webrtc-version-updater
· 12 days ago
5cc9957
Batch assign RTP seq# for all packets of a frame.
by Erik Språng
· 12 days ago
e927c0f
QualityScalingTests: Move encoder factory creation to ScalingObserver.
by Åsa Persson
· 12 days ago
3999384
Reland "Reland "Split peer_connection_integrationtest.cc into pieces""
by Harald Alvestrand
· 12 days ago
62b6c92
Refactor LossBasedBandwidthEstimation
by Per Kjellander
· 12 days ago
ebc563e
Update the call to RuntimeEnvironment.application
by Artem Titov
· 12 days ago
89c40e2
Revert "Reland "Split peer_connection_integrationtest.cc into pieces""
by Harald Alvestrand
· 12 days ago
60c0b44
Use CallbackList for DtlsState in dtls_transport.
by Lahiru Ginnaliya Gamathige
· 13 days ago
46e5a2f
Update WebRTC code version (2021-02-17T04:02:09).
by webrtc-version-updater
· 13 days ago
772066b
Reland "Split peer_connection_integrationtest.cc into pieces"
by Harald Alvestrand
· 13 days ago
3562318
Delete unused functions in RtpSender, RtcpSender and RtcpReceiver
by Danil Chapovalov
· 13 days ago
f4e3e2b
Delete rtc::Callback0 and friends.
by Niels Möller
· 13 days ago
d6c81db
Replace VideoLayerFrameId with int64_t.
by philipel
· 13 days ago
bdf78cb
Bug fixes to EglBase10Impl.getNativeEglContext.
by Sami Kalliomäki
· 13 days ago
067b050
Delete deprecated unused functions from RtpRtcp interface
by Danil Chapovalov
· 13 days ago
2c8d929
QualityScalingTests: Add tests for VP9.
by Åsa Persson
· 13 days ago
c79bd43
Delete friendship between VirtualSocket and VirtualSocketServer
by Niels Möller
· 13 days ago
686ad4f
Resolve relative paths in sdk build scripts.
by Yura Yaroshevich
· 14 days ago
d1bbec3
Update WebRTC code version (2021-02-16T04:03:42).
by webrtc-version-updater
· 14 days ago
b73c9f0
Extract SystemTimeNanos to its own file
by Johannes Kron
· 14 days ago
8408c99
Remove 'secondary sink' concept from webrtc::VideoReceiveStream.
by Tomas Gunnarsson
· 2 weeks ago
a33f41b
Support getNativeEglContext in EglBase10Impl.
by Sami Kalliomäki
· 2 weeks ago
8623c75
Remove ctor for BuiltInNetworkBehaviorConfig
by Per Kjellander
· 2 weeks ago
453a125
Remove no longer needed FrameDroppingOn setting in QualityScalingTests.
by Åsa Persson
· 2 weeks ago
51746ce
Revert "Replace RecursiveCriticalSection with Mutex in RTCAudioSession."
by Niels Moller
· 2 weeks ago
17ec2fc
Remove log line that states that FlexFEC is disabled.
by Rasmus Brandt
· 2 weeks ago
6e35ece
Destroy PC properly to stop input video before closing video writer
by Artem Titov
· 2 weeks ago
9aa9b8d
Prepare to replace VideoLayerFrameId with int64_t.
by philipel
· 2 weeks ago
563fbc1
Replace RecursiveCriticalSection with Mutex in DxgiDuplicatorController
by Niels Möller
· 2 weeks ago
410c998
Const correct NetworkEmulationManager::GetStats
by Per Kjellander
· 2 weeks ago
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