Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
refs/tags/rel/10/fp2/22.10.0-rel.0
/
webrtc
715411c
Remove dependency on std::tr1
by Haibo Huang
· 6 years ago
09c7a57
Remove dead symlink.
by Steven Moreland
· 7 years ago
647707f
Add 'vendor: true' to headers to fix vndk builds
by Dan Willemsen
· 7 years ago
7358fda
Convert to Android.bp
by Dan Willemsen
· 7 years ago
aa415f4
Remove WEBRTC_BUILD_NEON_LIBS
by Dan Willemsen
· 7 years ago
c3f927f
Remove unused gnustl webrtc variants
by Dan Willemsen
· 7 years ago
60c71f6
Merge "Suppress non-critical warnings in webrtc." am: cd56b9ddae
by Chih-Hung Hsieh
· 7 years ago
107cb52
Suppress non-critical warnings in webrtc.
by Chih-Hung Hsieh
· 7 years ago
1bc6674
Merge "Move all libwebrtc* to vendor image. (2/2)" am: 3d524127ba
by Yifan Hong
· 7 years ago
e54a1ce
Move all libwebrtc* to vendor image. (2/2)
by Yifan Hong
· 7 years ago
e867596
Merge "libwebrtc_base depends on liblog" am: c07e6f5974 am: f08d623d4e
by Yifan Hong
· 7 years ago
bfb116c
Merge "Move all libwebrtc* to vendor image." am: 12aeeac6a5 am: 83cb4ee1d7
by Yifan Hong
· 7 years ago
e0d75fe
libwebrtc_base depends on liblog
by Yifan Hong
· 7 years ago
11d9a8b
Move all libwebrtc* to vendor image.
by Yifan Hong
· 7 years ago
76c221c
Merge "Use SSL_CTX_set_max_proto_version instead of SSL_CTX_set_max_version." am: 0d85cc3c3c am: 1335ab5cae
by davidben
· 7 years ago
0d85cc3
Merge "Use SSL_CTX_set_max_proto_version instead of SSL_CTX_set_max_version."
by Treehugger Robot
· 7 years ago
458ffd5
Use SSL_CTX_set_max_proto_version instead of SSL_CTX_set_max_version.
by davidben
· 8 years ago
c1c39a8
Merge "webrtc: Use the NDK cpufeatures directly" am: 6d99081fdd am: d70176e5a0
by Elliott Hughes
· 7 years ago
83d6729
webrtc: Use the NDK cpufeatures directly
by Jake Weinstein
· 7 years ago
9306ff4
Merge "Leave only an empty top level OWNERS file." am: 82eac3519a am: 4cc6c9fa1f
by Chih-hung Hsieh
· 7 years ago
2622ea7
Leave only an empty top level OWNERS file.
by Chih-Hung Hsieh
· 7 years ago
f81b461
Fix build warnings
by Glenn Kasten
· 7 years ago
3accb22
Merge "Use arch-specific variables" am: b33ba455c4 am: 3f1ab425f5 am: c55415d27b
by Victor Khimenko
· 8 years ago
33d86be
Use arch-specific variables
by Victor Khimenko
· 8 years ago
2c7ba82
Fix build warnings
by Glenn Kasten
· 8 years ago
4e188dd
Suppress unused-parameter warnings.
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
ff2a635
Add ramp-up tests for transport sequence number with and w/o audio.
by Stefan Holmer
· 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
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
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
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
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
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
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
ef3d805
[rtp_rtcp] rtcp::Tmmbn moved into own file explicetly unchanged.
by danilchap
· 8 years ago
db21f63
fix GN build break on native_client
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
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
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
59bac1a
Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest.
by asapersson
· 8 years ago
a2b1e03
Disable AudioDeviceAPITest.MicrophoneVolumeTests on Linux.
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
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
67e83d6
Update API for Objective-C RTCSessionDescription.
by Jon Hjelle
· 8 years ago
29d5e57
Update API for Objective-C RTCIceCandidate.
by Jon Hjelle
· 8 years ago
335ecf5
Disable VideoCaptureTest.Capabilities and CreateDelete fails on Mac
by kjellander
· 8 years ago
b680274
Fix a flaky turnport test failure
by Honghai Zhang
· 8 years ago
6b9ab92
Cease all future TURN requests when a TURN refresh request fails for a given TURN port.
by honghaiz
· 8 years ago
Next »