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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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38c121c484e12f677c2cb6afb882cd024bd469c1
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talk
1795c35
Add default implementation of Add/RemoveObserver.
by pbos@webrtc.org
· 10 years ago
8cad943
Revert 7327 "Update isolate.gypi files + link to isolate_driver.py"
by kjellander@webrtc.org
· 10 years ago
02cd306
Update isolate.gypi files + link to isolate_driver.py
by kjellander@webrtc.org
· 10 years ago
359d720
Allow Android apps to set video renderer scaling type.
by glaznev@webrtc.org
· 10 years ago
7dfb7fa
Reland disallowing blocking calls on the worker thread.
by jiayl@webrtc.org
· 10 years ago
6266240
Disable flaky tests:
by asapersson@webrtc.org
· 10 years ago
34f2a9e
Initialize SSL in unittest_main.cc.
by pbos@webrtc.org
· 10 years ago
bebc75e
Fix the duplicated candidate problem when using multiple STUN servers.
by jiayl@webrtc.org
· 10 years ago
a21d071
Reverting part of
by thorcarpenter@google.com
· 10 years ago
0530511
Explicitly initialize SSL for tests.
by pbos@webrtc.org
· 10 years ago
3987b6d
Fix a problem in Thread::Send.
by jiayl@webrtc.org
· 10 years ago
d60d79a
Thread annotation of rtc::CriticalSection.
by pbos@webrtc.org
· 10 years ago
38344ed
Move thread_annotations.h to webrtc/base/.
by pbos@webrtc.org
· 10 years ago
8166fae
Change Android video renderer to maintain video aspect
by glaznev@webrtc.org
· 10 years ago
90668b1
Switch HW video decoder to output byte buffers if video
by glaznev@webrtc.org
· 10 years ago
1b7dcc1
(Auto)update libjingle 76169599-> 76176062
by buildbot@webrtc.org
· 10 years ago
2c1bcea
Enable ipv6 by default for webrtc under a Finch experiment.
by guoweis@webrtc.org
· 10 years ago
3987f10
Revert "Remove DTMF status methods from Voice Engine" r7276
by henrik.lundin@webrtc.org
· 10 years ago
bf7b9e0
Remove DTMF status methods from Voice Engine
by henrik.lundin@webrtc.org
· 10 years ago
0a2087a
Skeleton for registering external encoders/decoders.
by pbos@webrtc.org
· 10 years ago
83f95ba
Remove engine-level SetOptions.
by pbos@webrtc.org
· 10 years ago
64a2f10
Remove Get/SetNetEQPlayoutMode APIs
by henrik.lundin@webrtc.org
· 10 years ago
97ed393
Reapply 23529005 after fixing the build break issue (Chromium:582133002)
by guoweis@webrtc.org
· 10 years ago
ed5ca1f
(Auto)update libjingle 75925673-> 75926712
by buildbot@webrtc.org
· 10 years ago
c98f217
(Auto)update libjingle 75924589-> 75925673
by buildbot@webrtc.org
· 10 years ago
0c9fe72
(Auto)update libjingle 75922684-> 75924589
by buildbot@webrtc.org
· 10 years ago
ebf2757
Fix HW video decoder crash on some Android KK devices.
by glaznev@webrtc.org
· 10 years ago
c1eebfa
Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc.
by thorcarpenter@google.com
· 10 years ago
e658124
Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD.
by glaznev@webrtc.org
· 10 years ago
bbe0a85
Config struct for VideoEncoder.
by pbos@webrtc.org
· 10 years ago
6e5c784
(Auto)update libjingle 75875619-> 75878731
by buildbot@webrtc.org
· 10 years ago
b5a5c44
(Auto)update libjingle 75865376-> 75875619
by buildbot@webrtc.org
· 10 years ago
d7acf11
(Auto)update libjingle 75854833-> 75865376
by buildbot@webrtc.org
· 10 years ago
ccb3e3f
(Auto)update libjingle 75854418-> 75854833
by buildbot@webrtc.org
· 10 years ago
dcc1f04
(Auto)update libjingle 75852725-> 75853560
by buildbot@webrtc.org
· 10 years ago
0b435ba
A few fixes to avoid crash in HW codec on device orientation change.
by glaznev@webrtc.org
· 10 years ago
83af77b
Revert maximum video codec resolution on Android back to 720p again.
by glaznev@webrtc.org
· 10 years ago
933d88a
(Auto)update libjingle 75818332-> 75837294
by buildbot@webrtc.org
· 10 years ago
42731bd
Avoid writing a double/float to a string to avoid a crash.
by jiayl@webrtc.org
· 10 years ago
6cd6ba8
Expose VP8/H264 defaults through video_encoder.h.
by pbos@webrtc.org
· 10 years ago
ab071da
Split video_render_module implementation into default and internal implementation.
by andresp@webrtc.org
· 10 years ago
369a637
Implemented Network::GetBestIP() selection logic as following.
by guoweis@webrtc.org
· 10 years ago
3b67f8e
Enable HW video decoding on Qualcomm devices.
by glaznev@webrtc.org
· 10 years ago
4a5061f
talk/p2p/base: removed unused variable "port_"
by henrike@webrtc.org
· 10 years ago
a74eda1
Split video_capture_module specific implementation (external vs internal capture)
by andresp@webrtc.org
· 10 years ago
85ef770
Split video engine android initialization into each internal module initialization.
by andresp@webrtc.org
· 10 years ago
ab990ae
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
by pbos@webrtc.org
· 10 years ago
6a9b155
(Auto)update libjingle 75683337-> 75695882
by buildbot@webrtc.org
· 10 years ago
a59c501
Java VideoRenderer class may be backed by two different native
by glaznev@webrtc.org
· 10 years ago
40c2aa3
Implemented Network::GetBestIP() selection logic as following.
by guoweis@webrtc.org
· 10 years ago
f8bff76
Implemented Network::GetBestIP() selection logic as following.
by guoweis@webrtc.org
· 10 years ago
cddd17c
Recreate VideoStreams when setting resolution.
by pbos@webrtc.org
· 10 years ago
88e85ad
Add pbos@webrtc.org (myself) to talk/media/webrtc/.
by pbos@webrtc.org
· 10 years ago
80132e4
(Auto)update libjingle 75610402-> 75610402
by buildbot@webrtc.org
· 10 years ago
595b23c
Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."
by kjellander@webrtc.org
· 10 years ago
6ae5a6d
Add a target for the approved subset of rtc_base.
by andrew@webrtc.org
· 10 years ago
9967845
HW video decoding optimization to better support HD resolution:
by glaznev@webrtc.org
· 10 years ago
cd309e3
Enable ipv6 by default for webrtc under a Finch experiment.
by guoweis@webrtc.org
· 10 years ago
000d867
Make BW checks > 0 in peerconnection_unittest.cc.
by pbos@webrtc.org
· 10 years ago
7f82635
Stop building talk/xmllite since it is no longer used.
by henrike@webrtc.org
· 10 years ago
a42a3ad
(Auto)update libjingle 75390072-> 75428737
by buildbot@webrtc.org
· 10 years ago
7e31197
Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..."
by fbarchard@google.com
· 10 years ago
192a54f
Temporary revert maximum video codec resolution back to 1080p.
by glaznev@webrtc.org
· 10 years ago
3decd9b
Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
by henrike@webrtc.org
· 10 years ago
ea77334
(Auto)update libjingle 75302540-> 75327856
by buildbot@webrtc.org
· 10 years ago
1d8f780
Stop building talk/sound since it is no longer used.
by henrike@webrtc.org
· 10 years ago
1d53f64
Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD.
by glaznev@webrtc.org
· 10 years ago
307d3db
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
by henrikg@webrtc.org
· 10 years ago
c665dcb
Revert 7145 "Stop building talk/sound since it is no longer used."
by sprang@webrtc.org
· 10 years ago
1972ff8
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
by henrik.lundin@webrtc.org
· 10 years ago
4c87645
Stop building talk/sound since it is no longer used.
by henrike@webrtc.org
· 10 years ago
3472dcd
Fix frame rate selection for Android camera.
by glaznev@webrtc.org
· 10 years ago
b2efb67
Put base tests in webrtc_tests.gyp
by henrike@webrtc.org
· 10 years ago
b6d6928
Enable shared socket for TurnPort.
by jiayl@webrtc.org
· 10 years ago
5d639b3
(Auto)update libjingle 75141932-> 75179475
by buildbot@webrtc.org
· 10 years ago
7d4891d
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
by jiayl@webrtc.org
· 10 years ago
54cf150
ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that.
by fbarchard@google.com
· 10 years ago
22406fc
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
by jiayl@webrtc.org
· 10 years ago
3d81b1b
Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got
by mallinath@webrtc.org
· 10 years ago
4d19e05
Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own.
by andresp@webrtc.org
· 10 years ago
b420191
Expose VideoEncoders with webrtc/video_encoder.h.
by pbos@webrtc.org
· 10 years ago
8b0b211
Revert 7093: "Implementing ICE Transports type handling in libjingle transport."
by henrike@webrtc.org
· 10 years ago
7118e61
Finish work queue in SctpDataMediaChannelTest.
by pbos@webrtc.org
· 10 years ago
0e52772
Fix a bot-breaking memory leak from early returning in ParseMediaDescription.
by jiayl@webrtc.org
· 10 years ago
c172320
Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.
by jiayl@webrtc.org
· 10 years ago
fd42f9d
(Auto)update libjingle 74955991-> 75042522
by buildbot@webrtc.org
· 10 years ago
7256d31
Implementing ICE Transports type handling in libjingle transport.
by mallinath@webrtc.org
· 10 years ago
cc06056
Remove unnecessary include from testutils.cc.
by thorcarpenter@google.com
· 10 years ago
992febb
(Auto)update libjingle 74873066-> 74873164
by buildbot@webrtc.org
· 10 years ago
a3344cf
Fix webrtcvideoframe tests.
by thorcarpenter@google.com
· 10 years ago
ddb85ab
Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07
by jiayl@webrtc.org
· 10 years ago
af5fa95
(Auto)update libjingle 74857067-> 74860820
by buildbot@webrtc.org
· 10 years ago
7e3bd3d
(Auto)update libjingle 74851128-> 74857067
by buildbot@webrtc.org
· 10 years ago
bc6fa18
(Auto)update libjingle 74825992-> 74851128
by buildbot@webrtc.org
· 10 years ago
818b7b3
(Auto)update libjingle 74825084-> 74825992
by buildbot@webrtc.org
· 10 years ago
dfbcf81
Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice.
by jiayl@webrtc.org
· 10 years ago
f1427c6
Revert 7070 "TurnPort should retry allocation with a new address on error
by henrike@webrtc.org
· 10 years ago
4b23404
Reduce maximum video resolution for Android.
by glaznev@webrtc.org
· 10 years ago
574f2f6
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
by jiayl@webrtc.org
· 10 years ago
52055a2
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
by jiayl@webrtc.org
· 10 years ago
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