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gerrit-public.fairphone.software
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fp2-dev
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platform
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external
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chromium_org
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third_party
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libjingle
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source
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talk
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a77a80167b9acf91dfcd6c13fb76c0be2718030a
a77a801
Merge from Chromium at DEPS revision 278856
by Torne (Richard Coles)
· 10 years ago
99bc8f3
Merge from Chromium at DEPS revision 278205
by Torne (Richard Coles)
· 10 years ago
0a0ddb3
Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at e25766c6c8584153a564a6af63bc92d2adcf1e94
by Android Chromium Automerger
· 10 years ago
e25766c
(Auto)update libjingle 69543894-> 69555283
by buildbot@webrtc.org
· 10 years ago
02632fd
(Auto)update libjingle 69506154-> 69515138
by buildbot@webrtc.org
· 10 years ago
121ec58
Fix a memory leak in SctpDataMediaChannelTest.
by jiayl@webrtc.org
· 10 years ago
d19ccc1
Properly shut down the SCTP stack.
by jiayl@webrtc.org
· 10 years ago
81f0809
Makes the sid of a closed DataChannel available to reuse per the spec.
by jiayl@webrtc.org
· 10 years ago
96ca749
Increasing tolerances quite a bit to fight flakes.
by phoglund@webrtc.org
· 10 years ago
07617d7
(Auto)update libjingle 69359922-> 69365993
by buildbot@webrtc.org
· 10 years ago
d25cd98
(Auto)update libjingle 69337301-> 69359922
by buildbot@webrtc.org
· 10 years ago
d607b29
(Auto)update libjingle 69306183-> 69323802
by buildbot@webrtc.org
· 10 years ago
6de09a1
Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 2e507b2e465fe7f63c74e04d136ec2e23810c61f
by Android Chromium Automerger
· 10 years ago
f0150cc
Implement RTP extension support in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
0ae1298
Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 75ecb2a27cbb9da53ff201c0216c91065527675e
by Android Chromium Automerger
· 10 years ago
0b3c33a
(Auto)update libjingle 69292418-> 69293749
by buildbot@webrtc.org
· 10 years ago
3d341e2
(Auto)update libjingle 69291002-> 69292418
by buildbot@webrtc.org
· 10 years ago
2e507b2
(Auto)update libjingle 69278008-> 69291002
by buildbot@webrtc.org
· 10 years ago
a4717a4
(Auto)update libjingle 69276003-> 69278008
by buildbot@webrtc.org
· 10 years ago
95c71fa
(Auto)update libjingle 69260070-> 69276003
by buildbot@webrtc.org
· 10 years ago
27edee5
(Auto)update libjingle 69188577-> 69260070
by buildbot@webrtc.org
· 10 years ago
aad3e19
Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc.
by glaznev@webrtc.org
· 10 years ago
5a09f52
Add extra logging and latency restriction to VP8 HW encoder.
by glaznev@webrtc.org
· 10 years ago
b449ff6
(Auto)update libjingle 69144530-> 69164179
by buildbot@webrtc.org
· 10 years ago
6d33fab
(Auto)update libjingle 69143161-> 69144530
by buildbot@webrtc.org
· 10 years ago
aa77587
Add NACK feedback parameter to WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
2352cb3
Implement RTX tests+fixes in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
6bb6290
Remove engine_codecs_ cache from unittests.
by pbos@webrtc.org
· 10 years ago
4d8e980
Fix GYP DEPTH for libjingle isolate files
by kjellander@webrtc.org
· 10 years ago
a90dbe7
Pass GYP DEPTH variable to isolate.
by kjellander@webrtc.org
· 10 years ago
5c2c63e
(Auto)update libjingle 69131548-> 69132244
by buildbot@webrtc.org
· 10 years ago
a43c422
Initial owners file for talk/media/webrtc/.
by pbos@webrtc.org
· 10 years ago
ec0b1b8
(Auto)update libjingle 69102234-> 69116997
by buildbot@webrtc.org
· 10 years ago
2b70e77
Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck'
by jiayl@webrtc.org
· 10 years ago
20576e1
Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck
by jiayl@webrtc.org
· 10 years ago
7ac6cc6
(Auto)update libjingle 69097619-> 69099564
by buildbot@webrtc.org
· 10 years ago
ab9ed35
Fix the flaky RTP DataChannel test.
by jiayl@webrtc.org
· 10 years ago
1884535
(Auto)update libjingle 69069003-> 69082899
by buildbot@webrtc.org
· 10 years ago
890e587
Fixed GetStats when local and remote track are using the same ssrc.
by xians@webrtc.org
· 10 years ago
75ecb2a
(Auto)update libjingle 69049090-> 69054765
by buildbot@webrtc.org
· 10 years ago
d2ca8ce
(Auto)update libjingle 69005149-> 69049090
by buildbot@webrtc.org
· 10 years ago
fb2b140
(Auto)update libjingle 68985065-> 69005149
by buildbot@webrtc.org
· 10 years ago
7249cf2
Re-land webrtcmediaengine.cc part of r6397.
by pbos@webrtc.org
· 10 years ago
81fabb8
(Auto)update libjingle 68982444-> 68983526
by buildbot@webrtc.org
· 10 years ago
09c019f
Revert 6397 "(Auto)update libjingle 68949184-> 68982444"
by minyue@webrtc.org
· 10 years ago
9dab816
(Auto)update libjingle 68949184-> 68982444
by buildbot@webrtc.org
· 10 years ago
c52cd46
Adds end to end DataChannel tests.
by jiayl@webrtc.org
· 10 years ago
1c1377b
Add support for NVidia VP8 HW encoder.
by glaznev@webrtc.org
· 10 years ago
bd0d8c2
Revert 6380 "Replace libjingle_root with talk_root variable."
by kjellander@webrtc.org
· 10 years ago
91e3af0
(Auto)update libjingle 68891947-> 68893961
by buildbot@webrtc.org
· 10 years ago
1ef658d
Move WebRtcVideoEngine2 fakes to unittest header.
by pbos@webrtc.org
· 10 years ago
b723f16
Replace libjingle_root with talk_root variable.
by kjellander@webrtc.org
· 10 years ago
04dadd5
Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at cef4b39b7ec2035a69d46360d8101a785fd154d3
by Android Chromium Automerger
· 10 years ago
ae73e64
Remove unused test_env.py from isolate files + fix nss path.
by kjellander@webrtc.org
· 10 years ago
aa7cbe2
Adds support for the "apt" format parameter and turns on the RTX feature.
by stefan@webrtc.org
· 10 years ago
92d181b
Merge from Chromium at DEPS revision 275586
by Torne (Richard Coles)
· 10 years ago
19f7dc9
Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio."
by jiayl@webrtc.org
· 10 years ago
596929a
APPRTCDemo(objc): Remove regex parsing in favor of JSON struct.
by tkchin@webrtc.org
· 10 years ago
50af306
Add OpenGL Android video renderer which can display multiple
by glaznev@webrtc.org
· 10 years ago
444d3db
Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio.
by jiayl@webrtc.org
· 10 years ago
eb70c0e
AppRTCDemo(android): support app (UI) & capture rotation.
by fischman@webrtc.org
· 10 years ago
cef4b39
(Auto)update libjingle 68701339-> 68703656
by buildbot@webrtc.org
· 10 years ago
daeb064
Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at b5f0570d3fef0be89c236b1a063e0da5bdfb383f
by Android Chromium Automerger
· 10 years ago
ffd44a8
Fix C++11 -Wnarrowing in channel_unittest.cc.
by pbos@webrtc.org
· 10 years ago
214cc46
(Auto)update libjingle 68689052-> 68689059
by buildbot@webrtc.org
· 10 years ago
7254487
Make VideoSendStream/VideoReceiveStream configs const.
by pbos@webrtc.org
· 10 years ago
256edd7
Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 5ffd4a6be1c7f26558ee14fcf52badce9e6d2a99
by Android Chromium Automerger
· 10 years ago
57a0606
(Auto)update libjingle 68646004-> 68648993
by buildbot@webrtc.org
· 10 years ago
b5f0570
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
by wu@webrtc.org
· 10 years ago
91621f8
AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct.
by fischman@webrtc.org
· 10 years ago
b931d7e
Remove static initializer from WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
5ffd4a6
(Auto)update libjingle 68562943-> 68571194
by buildbot@webrtc.org
· 10 years ago
5eb79e1
Fix retain cycle in RTCEAGLVideoView.
by tkchin@webrtc.org
· 10 years ago
8c080a4
(Auto)update libjingle 68507189-> 68543735
by buildbot@webrtc.org
· 10 years ago
f44966d
(Auto)update libjingle 68506654-> 68507189
by buildbot@webrtc.org
· 10 years ago
3904bc8
(Auto)update libjingle 68501302-> 68506654
by buildbot@webrtc.org
· 10 years ago
8549b80
(Auto)update libjingle 68499439-> 68501302
by buildbot@webrtc.org
· 10 years ago
7c69197
(Auto)update libjingle 68495561-> 68499439
by buildbot@webrtc.org
· 10 years ago
27abd57
talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291).
by henrike@webrtc.org
· 10 years ago
8c1588b
(Auto)update libjingle 68465410-> 68487517
by buildbot@webrtc.org
· 10 years ago
24caaad
PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED.
by fischman@webrtc.org
· 10 years ago
6caba77
Add empty webrtcmediaengine.cc.
by pbos@webrtc.org
· 10 years ago
695e7a4
Merge from Chromium at DEPS revision 273901
by Torne (Richard Coles)
· 10 years ago
842c369
(Auto)update libjingle 68379861-> 68445177
by buildbot@webrtc.org
· 10 years ago
fc78ef6
Remove kMaxWaitForStatsMs from tsanv2 compilation.
by pbos@webrtc.org
· 10 years ago
ac85501
(Auto)update libjingle 68275107-> 68379861
by buildbot@webrtc.org
· 10 years ago
69adcdf
Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang.
by pbos@webrtc.org
· 10 years ago
1ef7bf9
Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback
by jiayl@webrtc.org
· 10 years ago
ae50b2d
Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates.
by jiayl@webrtc.org
· 10 years ago
0baf0a7
Fix AppRTC target configuration in libjingle_examples.gyp.
by tkchin@webrtc.org
· 10 years ago
52348f6
Implement mac version of AppRTCDemo.
by tkchin@webrtc.org
· 10 years ago
b278723
Fix two bugs in DataChannel state transition.
by jiayl@webrtc.org
· 10 years ago
a5702ad
(Auto)update libjingle 68230113-> 68244456
by buildbot@webrtc.org
· 10 years ago
50f1b41
(Auto)update libjingle 68230011-> 68230113
by buildbot@webrtc.org
· 10 years ago
5b17e9b
Implement new-API test RecvStreamWithoutRtx.
by pbos@webrtc.org
· 10 years ago
a812f30
Log default receive stream creation.
by pbos@webrtc.org
· 10 years ago
84f5e34
Implement and fix new-API NackIsEnabled test.
by pbos@webrtc.org
· 10 years ago
f365c83
(Auto)update libjingle 68203780-> 68206793
by buildbot@webrtc.org
· 10 years ago
742af79
Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
by jiayl@webrtc.org
· 10 years ago
4cb7ef9
AppRTCDemo(android): run in full-screen & immersive mode.
by fischman@webrtc.org
· 10 years ago
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