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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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refs/heads/int/n/fp2
b3cb8ab
Merge "Merge upstream SHA 04cb763" am: 9a337512d9
by Chih-hung Hsieh
· 8 years ago
int/n/fp2
9a33751
Merge "Merge upstream SHA 04cb763"
by Chih-hung Hsieh
· 8 years ago
daef292
Merge upstream SHA 04cb763
by Alex Luebs
· 8 years ago
04cb763
Add tests for verifying transport feedback for audio and video.
by Stefan Holmer
· 8 years ago
fcfc804
Eliminate defines in talk/
by kjellander
· 8 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
· 8 years ago
2734d77
Remove assert which was incorrectly added to TcpPort::OnSentPacket.
by Stefan Holmer
· 8 years ago
55674ff
Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
by Stefan Holmer
· 8 years ago
31c8d2e
Update with new default boringssl no-aes cipher suites. Re-enable tests.
by Torbjorn Granlund
· 8 years ago
e5e0e57
Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
by tommi
· 8 years ago
688e308
Re-land: "Use an explicit identifier in Config"
by aluebs
· 8 years ago
7307952
Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
by Stefan Holmer
· 8 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
· 8 years ago
35aae2e
Remove libfuzzer trybot from default trybot set.
by kjellander
· 8 years ago
ff2a635
Add ramp-up tests for transport sequence number with and w/o audio.
by Stefan Holmer
· 8 years ago
709513d
Delete remnants of non-square pixel support from cricket::VideoFrame.
by nisse
· 8 years ago
beed828
Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
by Sergey Ulanov
· 8 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
· 8 years ago
8432e1f
Re-enable tests that failed under Linux_Msan.
by marpan
· 8 years ago
fca54f4
Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
by tommi
· 8 years ago
09d944f
Roll chromium_revision 346fea9..099be58 (369082:369139)
by kjellander
· 8 years ago
306efad
Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
by kjellander
· 8 years ago
292e192
Add build_protobuf variable.
by kjellander
· 8 years ago
a276e73
Clean the code for external denoiser.
by jackychen
· 8 years ago
2f7dea1
[rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way
by danilchap
· 8 years ago
ea8c0f6
Fix capture ntp time issue introduced with r11187.
by Stefan Holmer
· 8 years ago
365543d
Roll chromium_revision 131167b..346fea9 (368784:369082)
by kjellander
· 8 years ago
25249d9
Use an explicit identifier in Config
by aluebs
· 8 years ago
e591f93
Storing raw audio sink for default audio track.
by deadbeef
· 8 years ago
6955870
Convert channel counts to size_t.
by Peter Kasting
· 8 years ago
92e677a
[rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function
by danilchap
· 8 years ago
5584bf4
Make :rtc_base_approved a public dep of :rtc_base.
by jbroman
· 8 years ago
e84e96e
NetEq: Fix a typo in a comment
by Henrik Lundin
· 8 years ago
36220ae
Slap deprecation notices on Pass methods
by kwiberg
· 8 years ago
d20e651
Fix test bug introduced in r11101.
by Stefan Holmer
· 8 years ago
3e1cfa7
Delete unused method webrtc::VideoRendererInterface::SetSize.
by nisse
· 8 years ago
3235a27
Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg.
by Henrik Boström
· 8 years ago
2845a02
Remove unused enum RTPDirections.
by terelius
· 8 years ago
3842c5c
Wire-up BWE feedback for audio receive streams.
by Stefan Holmer
· 8 years ago
6183de6
Remove tools/refactoring.
by Peter Boström
· 8 years ago
127782b
Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal.
by nisse
· 8 years ago
16979e3
Update .gitignore
by Henrik Kjellander
· 8 years ago
67e94fb
Add unit test for stand-alone denoiser and fixed some bugs.
by jackychen
· 8 years ago
b2328d1
Remove additional channel constraints when Beamforming is enabled in AudioProcessing
by aluebs
· 8 years ago
e93ad1b
Roll chromium_revision 8c958e0..131167b (368561:368784)
by kjellander
· 8 years ago
2a34688
Make Beamforming dynamically settable for Android platform builds
by aluebs
· 8 years ago
2bc63a1
clang-format audio_device/mac.
by andrew
· 8 years ago
a7446d2
Change DTLS default from 1.0 to 1.2 for webrtc.
by Guo-wei Shieh
· 8 years ago
f6c318e
Update API for Objective-C RTCMediaSource.
by Jon Hjelle
· 8 years ago
e799bad
Move Objective-C video renderers to webrtc/api/objc.
by Jon Hjelle
· 8 years ago
8102879
Update API for Objective-C RTCMediaStreamTrack.
by Jon Hjelle
· 8 years ago
a2c353f
Update API for Objective-C RTCStats.
by Jon Hjelle
· 8 years ago
7e8145f
[rtp_rtcp] rtcp::Tmmbr moved into own file
by danilchap
· 8 years ago
27ed3cc
SCTP: Stopped accepting SSRCs higher than max. Seems to fix asan-related crash.
by lally
· 8 years ago
a9a1d2a
H.264: Default flags and pulling in openh264 and ffmpeg.
by hbos
· 8 years ago
7823495
Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes.
by Jon Hjelle
· 8 years ago
fd99dea
Roll chromium_revision 42ab10e..8c958e0 (368534:368561)
by kjellander
· 8 years ago
ef3d805
[rtp_rtcp] rtcp::Tmmbn moved into own file explicetly unchanged.
by danilchap
· 8 years ago
d36efeb
Roll chromium_revision e738b54..42ab10e (368533:368534)
by kjellander
· 8 years ago
4de0037
Roll chromium_revision 7d97c94..e738b54 (368514:368533)
by kjellander
· 8 years ago
3c05e6c
Disable EndToEndTest.TransportSeqNumOnAudioAndVideo for Dr Memory.
by kjellander
· 8 years ago
daa8749
Revert of Roll chromium_revision 7d97c94..951c006 (368514:368525) (patchset #1 id:1 of https://codereview.webrtc.org/1577573002/ )
by guoweis
· 8 years ago
db21f63
fix GN build break on native_client
by Guo-wei Shieh
· 8 years ago
6109fc1
Roll chromium_revision 7d97c94..951c006 (368514:368525)
by kjellander
· 8 years ago
0697db6
Roll chromium_revision 8a15a7f..7d97c94 (368391:368514)
by kjellander
· 8 years ago
684e995
Disable 2 video tests which fail on DrMemoryFull
by Guo-wei Shieh
· 8 years ago
f475d36
Properly handle different transports having different SSL roles.
by Taylor Brandstetter
· 8 years ago
25702cb
Misc. small cleanups.
by pkasting
· 8 years ago
5de688e
Roll chromium_revision ede5d4f..8a15a7f (368310:368391)
by kjellander
· 8 years ago
49c454e
Cleaning neteq_unittest resource files.
by minyue
· 8 years ago
f1685c7
Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac
by kjellander
· 8 years ago
e74eef1
Add CreateSend/ReceiveTransport() methods to CallTest.
by stefan
· 8 years ago
37ebcf0
Reland "Add APK targets to build libjingle tests for Android."
by phoglund
· 8 years ago
b71b4f0
Update attributes to match gclibc's ansidecl.h
by kjellander
· 8 years ago
004851c
Roll chromium_revision 32569c6..ede5d4f (368258:368310)
by kjellander
· 8 years ago
e1ca167
Add tracing to NetEqImpl::GetAudio
by henrik.lundin
· 8 years ago
ec80f03
Check the mic volume only periodically on Mac.
by andrew
· 8 years ago
fbeb97e
Fix clang warning in peerconnection_jni.cc
by perkj
· 8 years ago
59bac1a
Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest.
by asapersson
· 8 years ago
95ab30c
Roll chromium_revision 6dd04c2..32569c6 (368115:368258)
by kjellander
· 8 years ago
a2b1e03
Disable AudioDeviceAPITest.MicrophoneVolumeTests on Linux.
by kjellander
· 8 years ago
893505d
Adding unit test to ensure TURN server priorities are unique.
by Taylor Brandstetter
· 8 years ago
e5ba13b
Adding a way for a Java RtpSender to set a track without taking ownership.
by Taylor Brandstetter
· 8 years ago
ced8ec9
Roll chromium_revision bd5949f..6dd04c2 (368055:368115)
by kjellander
· 8 years ago
bedc17b
Fixing integer underflow in FileAudioDevice (webrtc issue 4554)
by A.Brauckmann
· 8 years ago
6938793
vp9 tests: Adjust some parameters and re-enable the tests.
by Marco
· 8 years ago
6f5ca08
Update API for Objective-C RTCMediaConstraints.
by hjon
· 8 years ago
9fea80f
Add audio streams to CallTest and a first A/V call test.
by Stefan Holmer
· 8 years ago
ecd21b4
Add ImplementationName to SimulcastEncoderAdapter.
by pbos
· 8 years ago
01f364e
Remove always-on options in OveruseFrameDetector.
by Peter Boström
· 8 years ago
30166cb
iOS stability improvement for device switching, including BT devices
by henrika
· 8 years ago
7776e78
Remove unused methods in VideoCodingModule.
by Peter Boström
· 8 years ago
3886fc8
Use pointer to generated FEC packet.
by Peter Boström
· 8 years ago
46ea3ce
AudioDeviceTest.StartPlayoutOnTwoInstances now verifies two active playing streams
by henrika
· 8 years ago
a49ad98
Roll chromium_revision 4662d4f..bd5949f (368042:368055)
by kjellander
· 8 years ago
cea7c2f
Replace manual casting to rvalue reference with calls to std::move
by kwiberg
· 8 years ago
a46a4c9
Roll chromium_revision 2a70cb1..4662d4f (367468:368042)
by kjellander
· 8 years ago
1fe48a5
Add implementation in metrics.h that uses atomic pointer.
by asapersson
· 8 years ago
4331fcd
Remove duplicate code in SocketDispatcher
by jbauch
· 8 years ago
44cc795
Roll chromium_revision 4df108a..2a70cb1 (367307:367468)
by kjellander
· 8 years ago
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