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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
/
source
/
talk
/
5a1feaee7445019eac8c125a2cedcd377d90be74
5a1feae
Update talk to 61699344.
by mallinath@webrtc.org
· 11 years ago
54caffd
(Auto)update libjingle 61549749-> 61608469
by henrike@webrtc.org
· 11 years ago
765411c
AppRTCDemo(android): clarified README on how to launch app using adb.
by fischman@webrtc.org
· 11 years ago
f8eb85f
PeerConnectionTest(java): unbreak following 61460797-p10
by fischman@webrtc.org
· 11 years ago
f5e5b3a
Update talk to 61549749.
by mallinath@webrtc.org
· 11 years ago
8a77f5b
Update talk to 61538839.
by wu@webrtc.org
· 11 years ago
f5b8e7a
Revert 5545 "Update libjingle to 61514460"
by wu@webrtc.org
· 11 years ago
195a757
Update libjingle to 61514460
by xians@webrtc.org
· 11 years ago
20477f1
PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack.
by fischman@webrtc.org
· 11 years ago
b0ffe70
PeerConnection(java): use MediaCodec for HW-accelerated video encode where available.
by fischman@webrtc.org
· 11 years ago
ac98111
PeerConnectionClient needs to initialize SSL. BUG=2911 R=fischman@webrtc.org
by jiayl@webrtc.org
· 11 years ago
ee78a38
Revert 5516 "Thread annotation of talk_base::CriticalSection."
by wjia@webrtc.org
· 11 years ago
d1acbe9
Add ability to receive calls for iOS BUG=2701 R=fischman@webrtc.org
by fischman@webrtc.org
· 11 years ago
493b4d8
Thread annotation of talk_base::CriticalSection.
by pbos@webrtc.org
· 11 years ago
94a6ec3
Revert 5511 "Revert 5510 "Disable failing libjingle_p2p_unittest..."
by kjellander@webrtc.org
· 11 years ago
7b8389d
Roll chromium_revision 245382:249215
by kjellander@webrtc.org
· 11 years ago
c83b5b5
Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux"
by kjellander@webrtc.org
· 11 years ago
0a85679
Disable failing libjingle_p2p_unittest test on Linux
by kjellander@webrtc.org
· 11 years ago
12a4134
Disable AsyncInvokeTest.CancelInvoker test
by sergeyu@chromium.org
· 11 years ago
9d89162
Don't use LOG() in callback.h
by sergeyu@chromium.org
· 11 years ago
104540b
Switching to NSS random number generator and adding init method to unittests.
by mallinath@webrtc.org
· 11 years ago
70022fa
Update libjingle to 61168196
by sergeyu@chromium.org
· 11 years ago
f5a2b48
Fix gunit compilation on VS2012.
by pbos@webrtc.org
· 11 years ago
7dafa35
PeerConnectionTest(java): remove the obsolete magical names of streams & tracks.
by fischman@webrtc.org
· 11 years ago
b3def34
PeerConnectionTest(java): test SCTP DataChannels.
by fischman@webrtc.org
· 11 years ago
7f447f9
Updating libjingle.gyp after addition new files yuvframescapturer.cc.
by mallinath@webrtc.org
· 11 years ago
b881d27
Update talk to 60923971
by mallinath@webrtc.org
· 11 years ago
d51e05d
Disable a test assert which fails due to usrsctp not cleaned up in SctpDataEngine.cc
by jiayl@webrtc.org
· 11 years ago
13a42bc
Enable SCTP and use OPENSSL on Anroid and NSS on other platforms.
by jiayl@webrtc.org
· 11 years ago
1531384
Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."
by mallinath@webrtc.org
· 11 years ago
62373d6
Revert 5447 "Update talk to 60420316."
by mallinath@webrtc.org
· 11 years ago
ccbced3
Update talk to 60420316.
by mallinath@webrtc.org
· 11 years ago
574db84
Revert 5421 "Fix deadlock on register/unregister observer while ..."
by mallinath@webrtc.org
· 11 years ago
e182eb5
Remove the check for audio codec num in WebRtcVoiceEngineTest.HasCorrectCodecs.
by wu@webrtc.org
· 11 years ago
f665ac8
Re-enable webrtcvoice/videoengine unittests.
by wu@webrtc.org
· 11 years ago
91c1924
Fix deadlock on register/unregister observer while there is a an going callback.
by andresp@webrtc.org
· 11 years ago
1b57131
Update talk to 60094938.
by wu@webrtc.org
· 11 years ago
da97277
Libjingle source code has some spelling mistakes and one of them is "renegotation", which should be "renegotiation".
by mallinath@webrtc.org
· 11 years ago
7d00e6a
enabling disabled data channels tests on win32. The real culprit was that ice candidates not included in SDP when there were failure causing transport channels never becoming writable.
by mallinath@webrtc.org
· 11 years ago
8f2c8d5
Android example apps: fixes issue where useful failure information was suppressed.
by henrike@webrtc.org
· 11 years ago
19b058e
Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples.
by henrike@webrtc.org
· 11 years ago
f32dd31
Update libjingle to 59676287
by sergeyu@chromium.org
· 11 years ago
68463fd
Revert 5387 "Re-enable webrtcvoice/videoengine unittests."
by wu@webrtc.org
· 11 years ago
650f4d9
Re-enable webrtcvoice/videoengine unittests.
by wu@webrtc.org
· 11 years ago
c85af8d
Roll Chromium 238260 -> 243863
by wjia@webrtc.org
· 11 years ago
8485ec6
pRevert 5371 "Revert 5367 "Update talk to 59410372.""
by henrika@webrtc.org
· 11 years ago
26c6b8e
PeerConnection(java): Add OnRenegotiationNeeded support
by fischman@webrtc.org
· 11 years ago
8d131ba
Revert 5367 "Update talk to 59410372."
by henrika@webrtc.org
· 11 years ago
3e7165a
Update talk to 59410372.
by mallinath@webrtc.org
· 11 years ago
849bc5c
Fix NaCl compilation
by sergeyu@chromium.org
· 11 years ago
bfe87f8
PeerConnection(java): replace ScopedLocalRef with ScopedLocalRefFrame and fix a local reference leak in OnMessage.
by fischman@webrtc.org
· 11 years ago
5d93b75
AppRTCDemo(android): close() the throw-away DataChannel.
by fischman@webrtc.org
· 11 years ago
6c12326
Fix a compile error on Android on sctpdataengine.cc.
by wu@webrtc.org
· 11 years ago
2a81a38
Update talk to 59039880.
by wu@webrtc.org
· 11 years ago
84ab7ba
Android build: make it quiet on success and not overly noisy on failure.
by fischman@webrtc.org
· 11 years ago
5cb853d
The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation.
by fischman@webrtc.org
· 11 years ago
5b910e0
objc/README: Remove outdated advice about target_os.
by fischman@webrtc.org
· 11 years ago
40ce061
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
by wu@webrtc.org
· 11 years ago
1114b94
Update talk to 58157731.
by mallinath@webrtc.org
· 11 years ago
f89a403
Update talk to 58127566 together with
by wu@webrtc.org
· 11 years ago
a064d5d
Revert 5274 "Update talk to 58113193 together with https://webrt..."
by wu@webrtc.org
· 11 years ago
01d88c7
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
by wu@webrtc.org
· 11 years ago
59a8426
AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure.
by fischman@webrtc.org
· 11 years ago
174831c
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
by henrike@webrtc.org
· 11 years ago
97bc7cf
PeerConnection(java): rationalize pointer-to-jlong conversion.
by fischman@webrtc.org
· 11 years ago
cfd6247
Update talk to 58037405.
by wu@webrtc.org
· 11 years ago
1a8ac10
Disable a libjingle unittest which is failing after a chromium roll out.
by turaj@webrtc.org
· 11 years ago
176e638
Roll chromium_revision 232627:238260
by kjellander@webrtc.org
· 11 years ago
43f871c
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
by andrew@webrtc.org
· 11 years ago
9856caf
Fix compilation errors on Fedora 20.
by fischman@webrtc.org
· 11 years ago
eb221b8
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc.
by sergeyu@chromium.org
· 11 years ago
f3eeaef
revert r5230
by sergeyu@chromium.org
· 11 years ago
58b7686
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc.
by sergeyu@chromium.org
· 11 years ago
97fbd30
Update libjingle to 57692857
by sergeyu@chromium.org
· 11 years ago
e2d0284
RTCPeerConnection(objc): avoid leaking ICE candidate on addition.
by fischman@webrtc.org
· 11 years ago
658f439
Disable PeerConnectionEndToEndTest for tsanv2 build.
by wu@webrtc.org
· 11 years ago
f890d20
AppRTCDemo(ios): remove codesigning hack now that gyp signs by default.
by fischman@webrtc.org
· 11 years ago
3db0458
Update talk to 56698267.
by wu@webrtc.org
· 11 years ago
ade23ca
Disable datachannel_unittest.cc
by sergeyu@chromium.org
· 11 years ago
19da465
Update talk to 56619788
by sergeyu@chromium.org
· 11 years ago
a39fcda
Update talk to 56183333.
by wu@webrtc.org
· 11 years ago
100f2b1
PeerConnection iOS: update README instructions
by fischman@webrtc.org
· 11 years ago
4646ae6
Update talk to 56092586.
by wu@webrtc.org
· 11 years ago
80303b1
Update talk to 55906045.
by wu@webrtc.org
· 11 years ago
2a439b6
Update talk to 55863981.
by wu@webrtc.org
· 11 years ago
d1f631d
Update talk to 55821645.
by wu@webrtc.org
· 11 years ago
0f0ae8d
Explicitly @synthesize ObjC @properties
by fischman@webrtc.org
· 11 years ago
5d12f8d
Remove frame_callback.h include in webrtcvie.h.
by pbos@webrtc.org
· 11 years ago
5c9dd59
Update libjingle to 55618622. Update libyuv to r826.
by wu@webrtc.org
· 11 years ago
95cabf5
Fix tsan failures for libjingle_unittest.
by wu@webrtc.org
· 11 years ago
8af0f41
Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread.
by wu@webrtc.org
· 11 years ago
e940b2e
Fix tsan failures on filevideocapturer.cc.
by wu@webrtc.org
· 11 years ago
a0054d5
Upgrade scoped_ptr to Chromium's latest version.
by andrew@webrtc.org
· 11 years ago
7ee621c
Reenable DTLS renegotiation unittest in libjingle.
by mallinath@webrtc.org
· 11 years ago
e5b4910
Thread::Stop() must be called before any subclass's destructor completes.
by wu@webrtc.org
· 11 years ago
303a6ed
AppRTCDemo(android): remove vestigial mentions of PowerManager
by fischman@webrtc.org
· 11 years ago
cc71202
Update talk to 54898858.
by wu@webrtc.org
· 11 years ago
bf0b532
TSan v2 suppressions and exclusions for libjingle tests.
by kjellander@webrtc.org
· 11 years ago
1c20c72
Disabling the DTLS renegotiation test case for PeerConnection.
by mallinath@webrtc.org
· 11 years ago
8841d7b
Update talk to 54527154.
by mallinath@webrtc.org
· 11 years ago
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