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d29c689
Expose adaptive_ptime from Android SDK.
by Yura Yaroshevich
· 3 years ago
d71b38e
Update WebRTC code version (2021-04-19T04:03:03).
by webrtc-version-updater
· 3 years ago
d46a174
Expose adaptive_ptime from iOS SDK.
by Yura Yaroshevich
· 3 years ago
7fa8d46
Slight code clarification in RemoveStoppedTransceivers.
by Tomas Gunnarsson
· 3 years ago
0ee5bcf
Update WebRTC code version (2021-04-18T04:03:49).
by webrtc-version-updater
· 3 years ago
e632402
Remove rtp data channel related code from media_channel.*
by Tomas Gunnarsson
· 3 years ago
18ac30c
Update WebRTC code version (2021-04-17T04:04:03).
by webrtc-version-updater
· 3 years ago
983b620
Remove third_party/xstream from DEPS
by Bjorn Terelius
· 3 years ago
78aa5cd
dcsctp: Ensure packet size doesn't exceed MTU
by Victor Boivie
· 3 years ago
7af57c6
Remove RTP data implementation
by Harald Alvestrand
· 3 years ago
f981cb3
Add video/g3doc/stats.md to the doc site menu
by Artem Titov
· 3 years ago
15e078c
Fix unsignalled ssrc race in WebRtcVideoChannel.
by Henrik Boström
· 3 years ago
882d007
Add documentation for video/stats.
by Åsa Persson
· 3 years ago
0131a4d
Delete StreamAdapterInterface
by Niels Möller
· 3 years ago
b291da8
Add conceptual docs for modules/video_coding
by Rasmus Brandt
· 3 years ago
dd36198
Revert "Expose AV1 encoder&decoder from Android SDK."
by Björn Terelius
· 3 years ago
220a252
Delete unused class MessageBufferReader
by Niels Möller
· 3 years ago
6c127a1
Add Stable Writable Connection Ping Interval parameter to RTCConfiguration.
by Derek Bailey
· 3 years ago
74b1bbe
Remove unused a gn variable related to gtk
by Byoungchan Lee
· 3 years ago
a43528c
Update WebRTC code version (2021-04-16T04:04:52).
by webrtc-version-updater
· 3 years ago
3ceb16e
[Android] Set use_raw_android_executable explicitly for test() template.
by Peter Kotwicz
· 3 years ago
0f57e0b
Make libjingle_peerconnection_metrics_default_jni available in Linux builds.
by Mirko Bonadei
· 3 years ago
9fea310
Fix crash in WindowCapturerWinGdi::CaptureFrame.
by Austin Orion
· 3 years ago
a80c3e5
sctp: Reorganize build targets
by Florent Castelli
· 3 years ago
6c7c495
doc: fix ice metadata + spelling
by Philipp Hancke
· 3 years ago
fedd502
Expose AV1 encoder&decoder from Android SDK.
by Yura Yaroshevich
· 3 years ago
572f50f
Delete left-over references to AsyncInvoker
by Niels Möller
· 3 years ago
affd219
Delete AsyncInvoker usage from SimulatedPacketTransport
by Niels Möller
· 3 years ago
bc959b6
Remove enable_rtp_data_channel
by Harald Alvestrand
· 3 years ago
fa8a946
Remove obsolete DCHECK in remote_audio_source.cc.
by Henrik Boström
· 3 years ago
17490b5
Fix regression in UsrSctpReliabilityTest
by Niels Möller
· 3 years ago
403e328
Fix build with rtc_libvpx_build_vp9=false
by Byoungchan Lee
· 3 years ago
980c460
AGC2: retuning and large refactoring
by Alessio Bazzica
· 3 years ago
d28434b
Configure GN to use python3 to exec_script.
by Mirko Bonadei
· 3 years ago
dad500a
Remove PacketBuffers internal mutex.
by philipel
· 3 years ago
61982a7
AGC2 lightweight noise floor estimator
by Alessio Bazzica
· 3 years ago
3ab7a55
Reformat pacer doc and add it into sitemap
by Artem Titov
· 3 years ago
9aec8c2
Use default rtp parameters to init wrappers in iOS
by Yura Yaroshevich
· 3 years ago
89f3dd5
Make RTC_LOG_THREAD_BLOCK_COUNT less spammy for known call counts
by Tomas Gunnarsson
· 3 years ago
5744b7f
Fix formatting in sitemap.md
by Artem Titov
· 3 years ago
08d30a2
Add documentation for video/adaptation
by Evan Shrubsole
· 3 years ago
24bc419
Revert "Fix RTP header extension encryption"
by Björn Terelius
· 3 years ago
dea5721
Adding g3doc for AudioProcessingModule (APM)
by Per Åhgren
· 3 years ago
9861f96
dcsctp: Add operators on TimeMs and DurationMs
by Victor Boivie
· 3 years ago
8181b4f
Add conceptual documentation for NetEq.
by Jakob Ivarsson
· 3 years ago
a743303
Fix RTP header extension encryption
by Lennart Grahl
· 3 years ago
84ba164
Change from sakal@webrtc.org to xalep@webrtc.org in OWNERS files.
by Mirko Bonadei
· 3 years ago
c54f672
dcsctp: Fix post-review comments for DataTracker
by Victor Boivie
· 3 years ago
0498519
Add g3doc for audio coding module.
by Minyue Li
· 3 years ago
1fad94f
Remove ErleUncertainty
by Gustaf Ullberg
· 3 years ago
77d73a6
Document SctpTransport
by Harald Alvestrand
· 3 years ago
1d2d169
Update WebRTC code version (2021-04-14T04:04:15).
by webrtc-version-updater
· 3 years ago
e871e02
Add telemetry to measure usage, perf, and errors in Desktop Capturers.
by Austin Orion
· 3 years ago
efcfa4b
Roll chromium_revision 0bde1c5411..1a13f11499 (871876:872016)
by chromium-webrtc-autoroll
· 3 years ago
250fbb3
dcsctp: Make Sequence Number API more consistent
by Victor Boivie
· 3 years ago
ce423ce
Track last packet receive times in RtpVideoStreamReceiver instead of the PacketBuffer.
by philipel
· 3 years ago
cd83ae2
Speed up FrameCombiner::Combine by 3x
by Steve Anton
· 3 years ago
32347b5
Add readme for pacing module
by Erik Språng
· 3 years ago
09c7f1e
Add architecture section about PeerConnection test framework
by Artem Titov
· 3 years ago
79cbe69
Removes incorrect test expectation.
by Erik Språng
· 3 years ago
3db3a06
Adding g3doc for AudioDeviceModule (ADM) - part of the AudioEngine
by henrika
· 3 years ago
df1edc9
API description: PeerConnection description
by Harald Alvestrand
· 3 years ago
11b3089
Roll chromium_revision 74f869d04b..0bde1c5411 (871745:871876)
by chromium-webrtc-autoroll
· 3 years ago
1fded2f
dcsctp: Fix build dependencies
by Florent Castelli
· 3 years ago
e082984
Add death test for WrappingAsyncResolver
by Harald Alvestrand
· 3 years ago
a168bb9
Add index.md documentation page for PC level test framework
by Artem Titov
· 3 years ago
696cea0
Refactor some RtpSender-level tests into RtpRtcp-level tests
by Erik Språng
· 3 years ago
5fe0b37
Roll chromium_revision 7e70585ca5..74f869d04b (871605:871745)
by chromium-webrtc-autoroll
· 3 years ago
c8cf0a6
Remove MDNS message implementation
by Harald Alvestrand
· 3 years ago
eff79cf
Roll chromium_revision 2dffe06711..7e70585ca5 (871492:871605)
by chromium-webrtc-autoroll
· 3 years ago
067dce7
Fix processing of dropped frame for runtime added participant
by Andrey Logvin
· 3 years ago
dc53ce6
Revert "Add addr in error msg if stun sock sent with error"
by Mirko Bonadei
· 3 years ago
8093935
Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492)
by Mirko Bonadei
· 3 years ago
5051693
[Battery]: TaskQueuePacedSender not started by default.
by Etienne Pierre-doray
· 3 years ago
9ff75a6
Add addr in error msg if stun sock sent with error
by Yura Yaroshevich
· 3 years ago
3928e8f
dcsctp: Disable packet fuzzers
by Victor Boivie
· 3 years ago
0aa1a19
Add module overview of ICE
by Jonas Oreland
· 3 years ago
9de39f6
Add titovartem@webrtc.org as owner for /g3doc
by Artem Titov
· 3 years ago
4af6f2b
Move threading documentation for API into g3doc structure
by Harald Alvestrand
· 3 years ago
a3575cb
Remove tautological 'unsigned expr < 0' comparisons
by Anton Bikineev
· 3 years ago
22379fc
sctp: Rename SctpTransport to UsrSctpTransport
by Florent Castelli
· 3 years ago
606bd6d
dcsctp: Use correct field width for PPID
by Victor Boivie
· 3 years ago
9d60936
dcsctp: Fix relative dependency paths in timer/
by Victor Boivie
· 3 years ago
1003219
srtp: compare key length to srtp policy key length
by Philipp Hancke
· 3 years ago
5691053
IceStatesReachCompletionWithRemoteHostname: disable on Linux.
by Markus Handell
· 3 years ago
9071957
Remove unused members in tests.
by Åsa Persson
· 3 years ago
55de292
Use relative paths for //net/dcsctp/public:socket.
by Mirko Bonadei
· 3 years ago
1cdeb0a
addIceCandidate with callback into Android's SDK.
by Yura Yaroshevich
· 3 years ago
f075917
Ensure TaskQueuePacedSender dont depend on PacketRouter
by Per Kjellander
· 3 years ago
1c73e03
Update WebRTC code version (2021-04-12T04:03:54).
by webrtc-version-updater
· 3 years ago
cb70aa7
dcsctp: Add Reassembly Queue
by Victor Boivie
· 3 years ago
8a13d2c
dcsctp: Add Traditional Reassembly Streams
by Victor Boivie
· 3 years ago
b2d539b
dcsctp: Add Data Tracker
by Victor Boivie
· 3 years ago
50fc1df
dcsctp: Add SCTP packet corpus
by Victor Boivie
· 3 years ago
a4da76a
Update WebRTC code version (2021-04-10T04:03:36).
by webrtc-version-updater
· 3 years ago
061d898
Update WgcScreenSource* to use device indices instead of HMONITORs.
by Austin Orion
· 3 years ago
f2f9bb6
Fixing a buffer copy issue in DesktopFrame
by Joe Downing
· 3 years ago
fc5d276
Fix dropped frames not counted issue
by Johannes Kron
· 3 years ago
9410217
dcsctp: Add SCTP packet fuzzer
by Victor Boivie
· 3 years ago
3c31ee0
Reduce logging for PC supported codecs in PC level tests
by Artem Titov
· 3 years ago
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