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gerrit-public.fairphone.software
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platform
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external
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webrtc
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3c85cad1d4e54c6e4859e4787740d64f70880c08
3c85cad
Roll chromium_revision 7a4fb8d..f527e86 (370025:370073)
by kjellander
· 9 years ago
61046eb
Rename RWLockGeneric to RWLockWinXP to more accurately reflect when it's used.
by tommi
· 9 years ago
3860c7f
Fix parsing of CLANG_REVISON from tools/clang/scripts/update.py
by kjellander
· 9 years ago
c4c8485
Deleted renderer-related SetSize methods, and all uses.
by nisse
· 9 years ago
81354f5
Added mute logic to VideoTrackRenderers.
by nisse
· 9 years ago
8d6fab8
Remove two dead 'using' instances.
by Peter Boström
· 9 years ago
2067826
Remove dependency on ConditionVariableWrapper and CriticalSectionWrapper in UdpSocketPosix.
by Tommi
· 9 years ago
233bfd2
Move keyframe requests outside encoder mutex.
by Peter Boström
· 9 years ago
49c7402
Roll chromium_revision ad2f344..7a4fb8d (370010:370025)
by kjellander
· 9 years ago
aff4b70
Simplify the implementation of LoggingTest.
by tommi
· 9 years ago
f8c2bac
Add a gyp/gn variable for whether to use iLBC or not
by kwiberg
· 9 years ago
f5a3a93
Add 5-argument wrapper WebRtcVideoFrame::InitToBlack
by Niels Möller
· 9 years ago
d142067
Roll chromium_revision 1c9621e..ad2f344 (369979:370010)
by kjellander
· 9 years ago
34ed2b9
[rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function
by danilchap
· 9 years ago
8b1e431
Delete remnants of non-square pixel support from cricket::VideoFrame.
by nisse
· 9 years ago
33c1dca
Roll chromium_revision 89ca041..1c9621e (369966:369979)
by kjellander
· 9 years ago
9d2a3c5
Roll chromium_revision 4b805fe..89ca041 (369965:369966)
by kjellander
· 9 years ago
e110e5c
Roll chromium_revision 6058a7b..4b805fe (369961:369965)
by kjellander
· 9 years ago
d7db862
Roll chromium_revision 9e8fb7a..6058a7b (369957:369961)
by kjellander
· 9 years ago
c1cf0d3
Roll chromium_revision 0a79aa1..9e8fb7a (369950:369957)
by kjellander
· 9 years ago
011df0a
Roll chromium_revision 553c2cb..0a79aa1 (369932:369950)
by kjellander
· 9 years ago
f624a22
Roll chromium_revision 46fd746..553c2cb (369797:369932)
by kjellander
· 9 years ago
cec0a08
Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
by honghaiz
· 9 years ago
56271ed
fix bug 5430
by guoweis
· 9 years ago
f4decb5
Add QP statistics logging to Android HW encoder.
by glaznev
· 9 years ago
305ca25
Roll chromium_revision ff895e2..46fd746 (369726:369797)
by kjellander
· 9 years ago
884f585
Storing raw audio sink for default audio track.
by deadbeef
· 9 years ago
1567d0b
[rtp_rtcp] rtcp::Sdes moved into own file
by Danil Chapovalov
· 9 years ago
79a5a83
Adapt to boringssl's new defaults.
by torbjorng
· 9 years ago
2c13297
[rtp_rtcp] rtcp::Rpsi moved into own file
by Danil Chapovalov
· 9 years ago
256e5b2
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1557593002/
by Danil Chapovalov
· 9 years ago
a132197
Roll chromium_revision 6e188de..ff895e2 (369712:369726)
by kjellander
· 9 years ago
5679da1
[rtp_rtcp] rtcp::Fir moved into own file
by Danil Chapovalov
· 9 years ago
a5eba6c
[rtp_rtcp] rtcp::Remb moved into own file
by Danil Chapovalov
· 9 years ago
d66b44d
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
by ivoc
· 9 years ago
74e8df81
Roll chromium_revision 9946592..6e188de (369667:369712)
by kjellander
· 9 years ago
0f7d293
Revert changes to default option setting in https://codereview.webrtc.org/1500633002/
by solenberg
· 9 years ago
5602f65
setup_links.py fix so that FFmpeg compiles on windows.
by hbos
· 9 years ago
6a59ad3
Revert of Remove libfuzzer trybot from default trybot set. (patchset #1 id:1 of https://codereview.webrtc.org/1585963002/ )
by kjellander
· 9 years ago
301830f
Roll chromium_revision 099be58..9946592 (369139:369667)
by kjellander
· 9 years ago
dc305db
Add ApplyPacketOptions()
by Sergey Ulanov
· 9 years ago
20ac434
Fix a test bot failure.
by Honghai Zhang
· 9 years ago
e1f9d83
Adding AddTrack/RemoveTrack to native PeerConnection API.
by deadbeef
· 9 years ago
d9e62f5
Fixed sending Rtp packets with non zero csrcs and certain extensions.
by danilchap
· 9 years ago
67b1e1a
Put options as the argument of the java PeerConnectionFactory constructor.
by honghaiz
· 9 years ago
5d332ac
Fix expectation bug in the RTPSender unit test.
by terelius
· 9 years ago
04cb763
Add tests for verifying transport feedback for audio and video.
by Stefan Holmer
· 9 years ago
fcfc804
Eliminate defines in talk/
by kjellander
· 9 years ago
3542013
Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
by sprang
· 9 years ago
2734d77
Remove assert which was incorrectly added to TcpPort::OnSentPacket.
by Stefan Holmer
· 9 years ago
55674ff
Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
by Stefan Holmer
· 9 years ago
31c8d2e
Update with new default boringssl no-aes cipher suites. Re-enable tests.
by Torbjorn Granlund
· 9 years ago
e5e0e57
Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
by tommi
· 9 years ago
688e308
Re-land: "Use an explicit identifier in Config"
by aluebs
· 9 years ago
7307952
Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
by Stefan Holmer
· 9 years ago
268493a
Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
by nisse
· 9 years ago
35aae2e
Remove libfuzzer trybot from default trybot set.
by kjellander
· 9 years ago
ff2a635
Add ramp-up tests for transport sequence number with and w/o audio.
by Stefan Holmer
· 9 years ago
709513d
Delete remnants of non-square pixel support from cricket::VideoFrame.
by nisse
· 9 years ago
beed828
Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
by Sergey Ulanov
· 9 years ago
2d110be
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
by deadbeef
· 9 years ago
8432e1f
Re-enable tests that failed under Linux_Msan.
by marpan
· 9 years ago
fca54f4
Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
by tommi
· 9 years ago
09d944f
Roll chromium_revision 346fea9..099be58 (369082:369139)
by kjellander
· 9 years ago
306efad
Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
by kjellander
· 9 years ago
292e192
Add build_protobuf variable.
by kjellander
· 9 years ago
a276e73
Clean the code for external denoiser.
by jackychen
· 9 years ago
2f7dea1
[rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way
by danilchap
· 9 years ago
ea8c0f6
Fix capture ntp time issue introduced with r11187.
by Stefan Holmer
· 9 years ago
365543d
Roll chromium_revision 131167b..346fea9 (368784:369082)
by kjellander
· 9 years ago
25249d9
Use an explicit identifier in Config
by aluebs
· 9 years ago
e591f93
Storing raw audio sink for default audio track.
by deadbeef
· 9 years ago
6955870
Convert channel counts to size_t.
by Peter Kasting
· 9 years ago
92e677a
[rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function
by danilchap
· 9 years ago
5584bf4
Make :rtc_base_approved a public dep of :rtc_base.
by jbroman
· 9 years ago
e84e96e
NetEq: Fix a typo in a comment
by Henrik Lundin
· 9 years ago
36220ae
Slap deprecation notices on Pass methods
by kwiberg
· 9 years ago
d20e651
Fix test bug introduced in r11101.
by Stefan Holmer
· 9 years ago
3e1cfa7
Delete unused method webrtc::VideoRendererInterface::SetSize.
by nisse
· 9 years ago
3235a27
Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg.
by Henrik Boström
· 9 years ago
2845a02
Remove unused enum RTPDirections.
by terelius
· 9 years ago
3842c5c
Wire-up BWE feedback for audio receive streams.
by Stefan Holmer
· 9 years ago
6183de6
Remove tools/refactoring.
by Peter Boström
· 9 years ago
127782b
Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal.
by nisse
· 9 years ago
16979e3
Update .gitignore
by Henrik Kjellander
· 9 years ago
67e94fb
Add unit test for stand-alone denoiser and fixed some bugs.
by jackychen
· 9 years ago
b2328d1
Remove additional channel constraints when Beamforming is enabled in AudioProcessing
by aluebs
· 9 years ago
e93ad1b
Roll chromium_revision 8c958e0..131167b (368561:368784)
by kjellander
· 9 years ago
2a34688
Make Beamforming dynamically settable for Android platform builds
by aluebs
· 9 years ago
2bc63a1
clang-format audio_device/mac.
by andrew
· 9 years ago
a7446d2
Change DTLS default from 1.0 to 1.2 for webrtc.
by Guo-wei Shieh
· 9 years ago
f6c318e
Update API for Objective-C RTCMediaSource.
by Jon Hjelle
· 9 years ago
e799bad
Move Objective-C video renderers to webrtc/api/objc.
by Jon Hjelle
· 9 years ago
8102879
Update API for Objective-C RTCMediaStreamTrack.
by Jon Hjelle
· 9 years ago
a2c353f
Update API for Objective-C RTCStats.
by Jon Hjelle
· 9 years ago
7e8145f
[rtp_rtcp] rtcp::Tmmbr moved into own file
by danilchap
· 9 years ago
27ed3cc
SCTP: Stopped accepting SSRCs higher than max. Seems to fix asan-related crash.
by lally
· 9 years ago
a9a1d2a
H.264: Default flags and pulling in openh264 and ffmpeg.
by hbos
· 9 years ago
7823495
Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes.
by Jon Hjelle
· 9 years ago
fd99dea
Roll chromium_revision 42ab10e..8c958e0 (368534:368561)
by kjellander
· 9 years ago
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