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16cad083760474d71bdc3e990fd2f85c0245f000
16cad08
Wire up bandwidth stats to the new API and webrtcvideoengine2.
by stefan@webrtc.org
· 10 years ago
31e29d7
(Auto)update libjingle 79205306-> 79244016
by buildbot@webrtc.org
· 10 years ago
007bff3
(Auto)update libjingle 79200114-> 79205306
by buildbot@webrtc.org
· 10 years ago
e889fa5
Cleanup RTCVideoRenderer interface.
by tkchin@webrtc.org
· 10 years ago
1fc1bd0
(Auto)update libjingle 79169148-> 79192489
by buildbot@webrtc.org
· 10 years ago
562748d
AppRTCDemoActivity: use differnet Themes for different API levels
by mcasas@webrtc.org
· 10 years ago
19c8d5c
Remove protected files from talk/PRESUBMIT.py.
by pbos@webrtc.org
· 10 years ago
e540565
Falling back on single-stream on multiple SSRC.
by pbos@webrtc.org
· 10 years ago
4fb0d5a
ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
by perkj@webrtc.org
· 10 years ago
2efcfc4
Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send.
by tommi@webrtc.org
· 10 years ago
42536c6
Prepare for removal of PeerConnectionObserver::OnError.
by perkj@webrtc.org
· 10 years ago
b428c1d
(Auto)update libjingle 79104430-> 79104922
by buildbot@webrtc.org
· 10 years ago
561e751
Android AppRTCDemo improvements:
by glaznev@webrtc.org
· 10 years ago
377ebd5
Implement external decoder support in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
9d70622
Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan
by henrik.lundin@webrtc.org
· 10 years ago
11643cc
Update Android projects to API level 21.
by kjellander@webrtc.org
· 10 years ago
3434a47
Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64
by kjellander@webrtc.org
· 10 years ago
8a771f6
Update all .isolate files for the new format.
by kjellander@webrtc.org
· 10 years ago
74b9ec2
Update Android projects to API level 20.
by kjellander@webrtc.org
· 10 years ago
990afb6
Implement conference-mode temporal-layer screencast.
by pbos@webrtc.org
· 10 years ago
a1feeae
Configure A/V sync in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
36330f3
Adapting bitrate according to maxplaybackrate for Opus.
by minyue@webrtc.org
· 10 years ago
723f605
arm64 iOS build.
by tkchin@webrtc.org
· 10 years ago
b4c42ca
Improve the logging when a TCP connection is deleted.
by jiayl@webrtc.org
· 10 years ago
abe48e1
Cleaning up r7562-7567.
by minyue@webrtc.org
· 10 years ago
1975557
(Auto)update libjingle 78822708-> 78823675
by buildbot@webrtc.org
· 10 years ago
0042172
Revert 7563 "before rebase" due to wrong submission
by minyue@webrtc.org
· 10 years ago
2f5b8b6
Revert 7564 "to submit" due to wrong submission
by minyue@webrtc.org
· 10 years ago
d19bf75
to submit
by minyue@webrtc.org
· 10 years ago
0defac2
before rebase
by minyue@webrtc.org
· 10 years ago
395822f
adding default rates
by minyue@webrtc.org
· 10 years ago
5cc97a5
Use external VideoDecoders in VideoReceiveStream.
by pbos@webrtc.org
· 10 years ago
ecffe16
(Auto)update libjingle 78738075-> 78738103
by buildbot@webrtc.org
· 10 years ago
271ce95
ApprtDemo Android: Switch between front and back camera.
by perkj@webrtc.org
· 10 years ago
4bc2320
Renaming bandwidth to bitrate in webrtcvoiceengine.
by minyue@webrtc.org
· 10 years ago
5910fdf
move xmpp and p2p to webrtc
by henrike@webrtc.org
· 10 years ago
c103da3
(Auto)update libjingle 78642371-> 78680406
by buildbot@webrtc.org
· 10 years ago
a62c37c
(Auto)update libjingle 78616359-> 78642371
by buildbot@webrtc.org
· 10 years ago
f5dbd44
Check if a datachannel in the current local description is an sctp channel before assuming rtp.
by tommi@webrtc.org
· 10 years ago
b09dc2b
Adding setting screen to AppRTCDemo.
by glaznev@webrtc.org
· 10 years ago
10f2fab
(Auto)update libjingle 78583324-> 78583691
by buildbot@webrtc.org
· 10 years ago
1d3853c
Fix the SrtpFilter crash caused by two local offers.
by pthatcher@webrtc.org
· 10 years ago
6a9c800
Implement screencast settings for WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
bdfc44c
Use flags set by the port allocator.
by braveyao@webrtc.org
· 10 years ago
11a355d
(Auto)update libjingle 78430441-> 78445452
by buildbot@webrtc.org
· 10 years ago
c1577e9
(Auto)update libjingle 78427027-> 78430441
by buildbot@webrtc.org
· 10 years ago
6ae496e
Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected.
by perkj@webrtc.org
· 10 years ago
f9122b7
patch from issue 25469004
by pthatcher@webrtc.org
· 10 years ago
7b2755f
(Auto)update libjingle 78381351-> 78389679
by buildbot@webrtc.org
· 10 years ago
3140efa
(Auto)update libjingle 78344087-> 78381351
by buildbot@webrtc.org
· 10 years ago
190c56c
Add macros and APIs for webrtc histograms.
by asapersson@webrtc.org
· 10 years ago
98ff6c0
(Auto)update libjingle 78296920-> 78342456
by buildbot@webrtc.org
· 10 years ago
7fca7cd
(Auto)update libjingle 78273470-> 78296920
by buildbot@webrtc.org
· 10 years ago
a8314ba
Merging Henrik's and Peter's changes for AppRTCDemo
by glaznev@webrtc.org
· 10 years ago
bf7ac93
(Auto)update libjingle 78262388-> 78262615
by buildbot@webrtc.org
· 10 years ago
81275c6
Remove some disabled tests in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
ee11aad
(Auto)update libjingle 78193292-> 78199328
by buildbot@webrtc.org
· 10 years ago
d6fe0ff
Fix local address leakage when IceTransportsType is relay
by guoweis@webrtc.org
· 10 years ago
8135c2a
(Auto)update libjingle 78106439-> 78193292
by buildbot@webrtc.org
· 10 years ago
b51b408
Avoid using EGLContext class for Android 4.1 and below.
by glaznev@webrtc.org
· 10 years ago
010c874
Set up start bitrate in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
2ce4efe
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
by henrike@webrtc.org
· 10 years ago
36d12d8
(Auto)update libjingle 77953038-> 77970462
by buildbot@webrtc.org
· 10 years ago
1ad2ce0
Cleaning up Android AppRTCDemo.
by glaznev@webrtc.org
· 10 years ago
40aac8c
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
by henrike@webrtc.org
· 10 years ago
a1fed0a
(Auto)update libjingle 77701902-> 77709729
by buildbot@webrtc.org
· 10 years ago
8f580fc
(Auto)update libjingle 77689511-> 77696841
by buildbot@webrtc.org
· 10 years ago
7f06727
Remove unused (no-op) VideoOptions.
by pbos@webrtc.org
· 10 years ago
70d23c7
libjingle: use _stricmp instead of deprecated stricmp.
by henrike@webrtc.org
· 10 years ago
bfddf7a
Wire up external encoders.
by pbos@webrtc.org
· 10 years ago
6ca136a
(Auto)update libjingle 77554188-> 77629208
by buildbot@webrtc.org
· 10 years ago
aeaf150
Removes xmllite from talk/xmllite since webrtc/xmllite is used instead.
by henrike@webrtc.org
· 10 years ago
6b38543
(Auto)update libjingle 77414393-> 77554188
by buildbot@webrtc.org
· 10 years ago
9782bca
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
by xians@webrtc.org
· 10 years ago
54490d0
Change setting VP8 codec specific info values by HW VP8 encoder
by glaznev@webrtc.org
· 10 years ago
8d740c8
Remove bad waiting code from video decoder release function.
by glaznev@webrtc.org
· 10 years ago
36b9910
(Auto)update libjingle 77263371-> 77296420
by buildbot@webrtc.org
· 10 years ago
aa972cf
Protect send_/recv_streams_ in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
1f6d92b
Make the media content send only if offerToReceive is false while local streams exist.
by jiayl@webrtc.org
· 10 years ago
83d855e
Initialize sctp_paddrparams in OpenSctpSocket().
by pbos@webrtc.org
· 10 years ago
d7e2a0e
Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder.
by glaznev@webrtc.org
· 10 years ago
ee8ba51
Remove potential deadlock in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
d253970
Isolate: Remove use of --ignore_broken_items
by kjellander@webrtc.org
· 10 years ago
10093cd
Fixing build issue with L-sdk
by henrike@webrtc.org
· 10 years ago
16d9b3a
talk: removes empty directories base and sound.
by henrike@webrtc.org
· 10 years ago
69486f2
Wire up CPU adaptation in WebRtcVideoEngine2.
by pbos@webrtc.org
· 10 years ago
82879aa
Switch to SW video decoder on Android after getting 2 or more
by glaznev@webrtc.org
· 10 years ago
73bd001
Revert 7355 "Fix parallelization in libjingle_p2p_unittest."
by henrike@webrtc.org
· 10 years ago
3cd90fb
Fix parallelization in libjingle_p2p_unittest.
by pbos@webrtc.org
· 10 years ago
bee83ea
Reland "Remove DTMF status methods from Voice Engine" r7276
by henrik.lundin@webrtc.org
· 10 years ago
28fd205
Revert 7338 "Fixed the android build by making the interface pur..."
by xians@webrtc.org
· 10 years ago
25ebcf7
Fixed the android build by making the interface pure virtual.
by xians@webrtc.org
· 10 years ago
04c1c1e
Add default implementation of Add/RemoveObserver.
by pbos@webrtc.org
· 10 years ago
be51d7c
Revert 7327 "Update isolate.gypi files + link to isolate_driver.py"
by kjellander@webrtc.org
· 10 years ago
1e3db44
Update isolate.gypi files + link to isolate_driver.py
by kjellander@webrtc.org
· 10 years ago
c1c0fa6
Allow Android apps to set video renderer scaling type.
by glaznev@webrtc.org
· 10 years ago
002a4b0
Reland disallowing blocking calls on the worker thread.
by jiayl@webrtc.org
· 10 years ago
697892f
Disable flaky tests:
by asapersson@webrtc.org
· 10 years ago
bccb37b
Initialize SSL in unittest_main.cc.
by pbos@webrtc.org
· 10 years ago
4888d59
Fix the duplicated candidate problem when using multiple STUN servers.
by jiayl@webrtc.org
· 10 years ago
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