Log - 0e03360591bfdc088a6dcceb09bcb8123c6fb77f - platform/external/webrtc

0e03360 Add OWNERS for resources/ by kjellander@webrtc.org · 11 years ago
7a36cb4 Add missing dependencies to .isolate files by kjellander@webrtc.org · 11 years ago
1e8b671 Roll chromium_revision 231713:232627 by kjellander@webrtc.org · 11 years ago
da7f658 Add svn:ignore to avoid re-download of resources by kjellander@webrtc.org · 11 years ago
b8cb85b Fix broken build on x86 Android by fischman@webrtc.org · 11 years ago
7b273a5 PeerConnection iOS: update README instructions by fischman@webrtc.org · 11 years ago
07a6fbe Update talk to 56092586. by wu@webrtc.org · 11 years ago
3779c1c Fix invalid .sha1 files for audio_coding by kjellander@webrtc.org · 11 years ago
8017458 Replace old resources download script with depot_tools by kjellander@webrtc.org · 11 years ago
a452fc2 Remove resources/ svn:ignore to prepare for updated resource handling by kjellander@webrtc.org · 11 years ago
58bcdee Roll chromium_revision 229708:231713 by kjellander@webrtc.org · 11 years ago
766154a Removed unused code. by asapersson@webrtc.org · 11 years ago
e2df8b7 Make video quality analysis unittests print to log instead of stdout. by kjellander@webrtc.org · 11 years ago
5dd2ecb Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
74e6e84 Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
d705649 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
1a4ed0d Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
de30501 Update talk to 55906045. by wu@webrtc.org · 11 years ago
58cd316 Address Clag Analyzer issues. by turaj@webrtc.org · 11 years ago
7d6bd22 Propagate estimated RTT from receivers to rtt observer. by asapersson@webrtc.org · 11 years ago
da2c37b Video bandwidth not reported correctly by sprang@webrtc.org · 11 years ago
773e727 Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc by sergeyu@chromium.org · 11 years ago
de748c8 Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build. by wu@webrtc.org · 11 years ago
dce70cc Add delay limit to ChokeFilter. by solenberg@webrtc.org · 11 years ago
f424cb8 Update talk to 55863981. by wu@webrtc.org · 11 years ago
d6e4663 Logging for BWE test framework. by solenberg@webrtc.org · 11 years ago
cecfd18 Update talk to 55821645. by wu@webrtc.org · 11 years ago
ec4cccc Update libyuv to 832. by wu@webrtc.org · 11 years ago
47ebbad Make video/ only depend on video_engine_core. by pbos@webrtc.org · 11 years ago
def22b4 Stop DirectTransports in VideoSendStreamTests. by pbos@webrtc.org · 11 years ago
55e1723 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. by turaj@webrtc.org · 11 years ago
9ca93a8 Explicitly @synthesize ObjC @properties by fischman@webrtc.org · 11 years ago
0aeb22e Adding tl0idx consideration for continuity by mikhal@webrtc.org · 11 years ago
0803c03 Fix build/isolate.gypi path in webrtc_tests.gypi. by pbos@webrtc.org · 11 years ago
b7a1718 Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. by fischman@webrtc.org · 11 years ago
16e03b7 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
850bcbe Remove frame_callback.h include in webrtcvie.h. by pbos@webrtc.org · 11 years ago
1a3a6e5 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
fe5d36b Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
97077a3 Update libjingle to 55618622. Update libyuv to r826. by wu@webrtc.org · 11 years ago
728bc0f Add qiang.lu@intel.com to WATCHLISTS. by fischman@webrtc.org · 11 years ago
c94abd3 Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h by xians@webrtc.org · 11 years ago
e4e5683 Clean up tsan suppression file: by wu@webrtc.org · 11 years ago
0729460 Added a "interleaved_" flag to webrtc::AudioFrame. by xians@webrtc.org · 11 years ago
442c5e4 Update adapter.js to use TURN transport parameters for FF version 27 & above. by vikasmarwaha@webrtc.org · 11 years ago
d674a56 Update dc1 demo as it was using invalid data Constraint (Reliable:true) for SCTP. The constraint Reliable is not supported by Standard and ignored in our implementation. See issue 2511. by vikasmarwaha@webrtc.org · 11 years ago
b3731da Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_. by andrew@webrtc.org · 11 years ago
b56d0e3 Change the low-bitrate handling in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
37bb497 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
d371a29 Fix tsan failures for libjingle_unittest. by wu@webrtc.org · 11 years ago
d1bcf11 Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined. by andrew@webrtc.org · 11 years ago
22858d4 Add an extended filter option to audioproc. by andrew@webrtc.org · 11 years ago
042e91c Fix for incorrect RTT estimation. A too low RTT value could be estimated. by asapersson@webrtc.org · 11 years ago
ba975e2 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
886aef0 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
8804a29 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
4d7116b Fix tsan failures on filevideocapturer.cc. by wu@webrtc.org · 11 years ago
90d8719 Radix should be specified when calling ParseInt function in adapter.js. Refer to issue 2490. by vikasmarwaha@webrtc.org · 11 years ago
8575980 Add default trybots for WebRTC try server. by kjellander@webrtc.org · 11 years ago
31628aa Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
06b60c0 Roll chromium_revision 228675:229708 by kjellander@webrtc.org · 11 years ago
621df67 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. by andrew@webrtc.org · 11 years ago
943e3b9 Add CurrentLayerId() to temporal layers. by marpan@webrtc.org · 11 years ago
50bc553 Reenable DTLS renegotiation unittest in libjingle. by mallinath@webrtc.org · 11 years ago
9c735c4 Updated WebRTC version to 3.45 by elham@webrtc.org · 11 years ago
8215106 Framework for testing bandwidth estimation. by solenberg@webrtc.org · 11 years ago
29dd0de Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
0d19ed9 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
fe1ef93 Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
e053629 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
eb61a85 Move audio_e2e_harness into include_tests==1 condition. by kjellander@webrtc.org · 11 years ago
88a3108 Add audio_e2e_test target to tools.gyp by kjellander@webrtc.org · 11 years ago
fb648da Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug. by wu@webrtc.org · 11 years ago
3c5d2b4 Thread::Stop() must be called before any subclass's destructor completes. by wu@webrtc.org · 11 years ago
3e00505 Have padding decay to zero if no frames are being captured. by stefan@webrtc.org · 11 years ago
893c07f Disable the -Wno-unused-const-variable Clang warning on Mac by kjellander@webrtc.org · 11 years ago
89b1e68 Minor comment fix after clang reformat. by andrew@webrtc.org · 11 years ago
1c82037 AppRTCDemo(android): remove vestigial mentions of PowerManager by fischman@webrtc.org · 11 years ago
2df89c0 MouseCursorMonitor implementation for OSX and Windows. by sergeyu@chromium.org · 11 years ago
6b426ba Final round of LSan suppressions (take 2) by kjellander@webrtc.org · 11 years ago
6342066 Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
b22049b Final round of LSan suppressions. by kjellander@webrtc.org · 11 years ago
8a7b89f More libjingle LSan suppressions. by kjellander@webrtc.org · 11 years ago
675e260 Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
e61da8c Suppressions and PRESUBMIT.py for LSan by kjellander@webrtc.org · 11 years ago
c11148b Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
603ed98 Suppress race condition warn in CallTest_ReceivesAndRetransmitsNack_Test by sprang@webrtc.org · 11 years ago
54e729b Remove tsan suppression for the failure that's already fixed. by wu@webrtc.org · 11 years ago
853dd07 Add issue links to the tsanv2 suppressions. by wu@webrtc.org · 11 years ago
e7771e2 Add /webrtc/modules/audio_device/android/test/{bin,gen,libs} to .gitignore by fischman@webrtc.org · 11 years ago
d030972 Remove unused kPowTableFrac which causes anroid clang build failure. by wu@webrtc.org · 11 years ago
1d1ffc9 Update talk to 54898858. by wu@webrtc.org · 11 years ago
83e9c89 Exclude more tests for TSan on Windows. by kjellander@webrtc.org · 11 years ago
d1cfa71 TSan v2 suppressions and exclusions for libjingle tests. by kjellander@webrtc.org · 11 years ago
25fce9a Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
b400aa7 Don't pad if only one stream is sent, except if auto muted. by stefan@webrtc.org · 11 years ago
e7009f3 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
5d957e2 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
9401524 Run FullStack tests without render windows. by pbos@webrtc.org · 11 years ago
5ed4f46 Remove TSan v2 disabled test in condition_variable_unittest.cc by kjellander@webrtc.org · 11 years ago