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4ee5e5f
Disable VideoCaptureTest due to flakyness
by Björn Terelius
· 84 minutes ago
main
master
37fb647
Disable the roll of 'android_ndk'
by Prashanth Swaminathan
· 5 hours ago
lkgr
36c945b
Update WebRTC code version (2023-06-08T04:11:54).
by webrtc-version-updater
· 7 hours ago
9d9c3f4
[Analysis] Remove old threshold fields
by Beining Chen
· 12 hours ago
89f64b9
Make packet info optional and only set for primary packets in NetEq.
by Jakob Ivarsson
· 18 hours ago
9e639fa
Migrate Android NDK to CIPD [1/2]
by Prashanth Swaminathan
· 19 hours ago
fc260a18
Add method SetTimestamp in TransformableAudioFrameInterface
by Palak Agarwal
· 21 hours ago
3403acb
av1: 8 threads for >720p and tiles config
by Jerome Jiang
· 21 hours ago
d615704
Enable frame dropping in libaom AV1 encoder
by Sergey Silkin
· 23 hours ago
a458fe5
Update WebRTC code version (2023-06-07T04:12:21).
by webrtc-version-updater
· 31 hours ago
09e0086
Remove ImplForTesting function from MediaChannel
by Harald Alvestrand
· 2 days ago
bd66cfe
Roll chromium_revision a5cd053713..a8db252505 (1153688:1153825)
by chromium-webrtc-autoroll
· 2 days ago
847208e
Remove transitional shim classes
by Harald Alvestrand
· 2 days ago
ade07ca
Rename current flexfec implementation flexfec_03
by Yosef Twaik
· 2 days ago
43df03d
Fix spelling mistake ReplaceRemoteDescriptionAndCheckE*r*or
by Philipp Hancke
· 2 days ago
6d25e96
Roll chromium_revision 404afa6a86..a5cd053713 (1153573:1153688)
by chromium-webrtc-autoroll
· 2 days ago
d3b71c7
Update WebRTC code version (2023-06-06T04:12:09).
by webrtc-version-updater
· 2 days ago
e00a12f
Roll chromium_revision 96ad22527d..404afa6a86 (1153423:1153573)
by chromium-webrtc-autoroll
· 2 days ago
8c4b9ea
Remove references to AudioCodec and VideoCodec constructors
by Florent Castelli
· 3 days ago
fd096da
Roll chromium_revision 8f3397a259..96ad22527d (1153256:1153423)
by chromium-webrtc-autoroll
· 3 days ago
77c6230
Add create functions for voice media send and receive channels.
by Harald Alvestrand
· 3 days ago
be316da
Ensure that RTCErrorOr<T, E> doesn't require T to be default constructible
by Florent Castelli
· 3 days ago
0740048
Roll chromium_revision f28b824184..8f3397a259 (1152496:1153256)
by chromium-webrtc-autoroll
· 3 days ago
b0ef5e4
Declare factory functions for video sender and receiver
by Harald Alvestrand
· 3 days ago
2f0c078
Split WebRtcVoiceChannel into Send and Receive classes
by Harald Alvestrand
· 3 days ago
1e04d61
Update WebRTC code version (2023-06-05T04:02:35).
by webrtc-version-updater
· 3 days ago
816f5b1
Create VP9Encoder with a VP9 codec object
by Florent Castelli
· 4 days ago
968e3c0
rtp_sender: fix typo with spatial_bitmask
by Alfred E. Heggestad
· 4 days ago
079ce25
Update WebRTC code version (2023-06-04T04:02:33).
by webrtc-version-updater
· 4 days ago
e10f025
Update WebRTC code version (2023-06-03T04:02:02).
by webrtc-version-updater
· 5 days ago
5278b39
Add H264Encoder::Create()
by Florent Castelli
· 6 days ago
811e24a
Move functionality from AudioCodec and VideoCodec into cricket::Codec
by Florent Castelli
· 6 days ago
b8651de
Roll chromium_revision d48b2929db..f28b824184 (1152392:1152496)
by chromium-webrtc-autoroll
· 6 days ago
54e95bc
Propagate time of the last received packet with Timestamp type
by Danil Chapovalov
· 6 days ago
9a34d80
Apply the "shim" pattern for WebRtcVoiceEngine
by Harald Alvestrand
· 6 days ago
b15a9f0
Fix perf tests.
by Jeremy Leconte
· 6 days ago
3488726
sdp: reject spec simulcast answers without the rid extension
by Philipp Hancke
· 6 days ago
f785bd4
Split WebRtcVideoMediaChannel into Send and Receive
by Harald Alvestrand
· 6 days ago
4ad141e
Add callback for send codec in audio too
by Harald Alvestrand
· 6 days ago
371b7af
Roll chromium_revision 2478b63fb4..d48b2929db (1151892:1152392)
by chromium-webrtc-autoroll
· 6 days ago
b29ee5b
Run the same perf tests on all platforms.
by Jeremy Leconte
· 6 days ago
267040e
Make native VideoTrack pointer public
by Jonas Oreland
· 6 days ago
cfc1a3a
Update vpython3 requests
by Brian Sheedy
· 6 days ago
eeacddb
Disable flaky PictureIdTests.
by Jeremy Leconte
· 6 days ago
d454815
Use //third_party/cpu_features directly
by Prashanth Swaminathan
· 6 days ago
dab505b
Update WebRTC code version (2023-06-02T04:02:59).
by webrtc-version-updater
· 6 days ago
063b45b
Roll chromium_revision faf350b988..2478b63fb4 (1151758:1151892)
by chromium-webrtc-autoroll
· 7 days ago
dba22d3
Move transceiver iteration loop over to the signaling thread.
by Tommi
· 7 days ago
513ab0c
Add a -d option to apply-iwyu
by Harald Alvestrand
· 7 days ago
e24b34c
Roll chromium_revision e26eb46a54..faf350b988 (1150524:1151758)
by chromium-webrtc-autoroll
· 7 days ago
b93f69a
In VideoCaptureV4L2 create the capture thread last in StartCapture
by Andreas Pehrson
· 7 days ago
e44a155
Add third_party/cpu_features license path.
by Jeremy Leconte
· 7 days ago
2d59853
Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController.
by Ying Wang
· 7 days ago
3d6e88e
Remove low_bandwidth_audio_test.
by Jeremy Leconte
· 7 days ago
6110fd9
Update WebRTC code version (2023-06-01T04:12:34).
by webrtc-version-updater
· 7 days ago
cb85143
Fix duplicate 'unix' OS and latest-revision deps
by Prashanth Swaminathan
· 8 days ago
2197300
Update ReceiveStatistics to use Timestamp/TimeDelta to represent time
by Danil Chapovalov
· 8 days ago
a9bba04
Updating AsyncAudioProcessing API, part 1.
by Peter Hanspers
· 8 days ago
56d69e2
Add //third_party/cpu_features to DEPS
by Prashanth Swaminathan
· 8 days ago
c18f083
Split MediaChannel concrete functions to MediaChannelUtil
by Harald Alvestrand
· 8 days ago
94a9d55
Update WebRTC code version (2023-05-31T04:11:01).
by webrtc-version-updater
· 8 days ago
b84fae6
Use sinf instead of std::sinf to improve libstdc++ compatibility
by Li-Yu Yu
· 9 days ago
9fa5057
Roll chromium_revision da88253915..e26eb46a54 (1150417:1150524)
by chromium-webrtc-autoroll
· 9 days ago
6acfbb0
Replace std::optional with absl::optional in RtpPacketHistory
by Per K
· 9 days ago
d8098fb
Delete struct RTCPReportBlock as no longer used
by Danil Chapovalov
· 9 days ago
d8b88d8
Use the VideoMediaChannelShim for all cases
by Harald Alvestrand
· 9 days ago
428836d
tools: fix small typo in python script
by Alfred E. Heggestad
· 9 days ago
4bf5238
sdp: reject BUNDLE with RTP header extension id collisions
by Philipp Hancke
· 9 days ago
b184634
Run webrtc_perf_tests on Fuchsia os.
by Jeremy Leconte
· 9 days ago
c73ea4f
More systematic null checks before calling native methods
by Xavier Lepaul
· 9 days ago
a3e9c0a
Roll chromium_revision c90a8a46d7..da88253915 (1150306:1150417)
by chromium-webrtc-autoroll
· 9 days ago
97c9623
Make a shim object implementing the VideoMediaChannel interface
by Harald Alvestrand
· 9 days ago
4c1e959
Change flexfec header reader to parse according to updated RFC.
by Yosef Twaik
· 9 days ago
e4a9a6d
Update WebRTC code version (2023-05-30T04:02:06).
by webrtc-version-updater
· 9 days ago
c5e4bcc
Roll chromium_revision 599c746c73..c90a8a46d7 (1150194:1150306)
by chromium-webrtc-autoroll
· 10 days ago
4b14cb7
Roll chromium_revision fa2e063162..599c746c73 (1150086:1150194)
by chromium-webrtc-autoroll
· 10 days ago
4aaacb4
Update WebRTC code version (2023-05-29T04:03:50).
by webrtc-version-updater
· 10 days ago
e641a97
In RtcpReceiver remove redundand way to represent RTCP report blocks
by Danil Chapovalov
· 11 days ago
b9de471
Update WebRTC code version (2023-05-28T04:11:22).
by webrtc-version-updater
· 11 days ago
98185b9
Roll chromium_revision 99b12997bf..fa2e063162 (1150050:1150086)
by chromium-webrtc-autoroll
· 12 days ago
a294353
Use type raw for video_codec_perf_tests.
by Mirko Bonadei
· 12 days ago
01c2efc
Roll chromium_revision bddf6cbe18..99b12997bf (1149812:1150050)
by chromium-webrtc-autoroll
· 12 days ago
9bc8d05
Update WebRTC code version (2023-05-27T04:12:09).
by webrtc-version-updater
· 12 days ago
9ac543c
Roll chromium_revision 1fc947a5da..bddf6cbe18 (1149703:1149812)
by chromium-webrtc-autoroll
· 13 days ago
87e74f9
Remove unused combined_audio_video_bwe.
by Yury Yarashevich
· 13 days ago
2bb686d
Stop running low_bandwith_audio_tests.
by Jeremy Leconte
· 13 days ago
6490999
Roll chromium_revision aae661725b..1fc947a5da (1148994:1149703)
by chromium-webrtc-autoroll
· 13 days ago
f0820ff
Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay
by Rasmus Brandt
· 13 days ago
9caef2a
Use a constant for invalid PipeWire file descriptor
by Jan Grulich
· 13 days ago
0f1a2c5
Change StreamDataCounters to use Timestamp instead of int64_t
by Danil Chapovalov
· 13 days ago
5f32fa4
Delete MediaBaseChannel class
by Harald Alvestrand
· 13 days ago
4f1dcbb
doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c
by Li-Yu Yu
· 13 days ago
f53b343
Cleanup RtcpTransceiver dependency on webrtc::Transport
by Danil Chapovalov
· 13 days ago
5f38949
Allow single-mline offers without BUNDLE group when using max-bundle
by Philipp Hancke
· 13 days ago
dff6e25
Update WebRTC code version (2023-05-26T04:05:22).
by webrtc-version-updater
· 13 days ago
6e23fa5
Cleanup WebRTC-PayloadTypes-Lower-Dynamic-Range trial
by Philipp Hancke
· 14 days ago
56d1260
PipeWire video capture: split portal and PipeWire implementations
by Jan Grulich
· 14 days ago
2264e7a
Fixes distortion in WGC screen capture path
by henrika
· 14 days ago
40a0fa9
Add new padding mode to RtpPacketHistory
by Per K
· 14 days ago
4206d31
[Analysis] Add new thresholds config schema
by Beining Chen
· 14 days ago
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