Log - 26a736f764385826c477d308c08a1a69a5f4289d - fp2-dev/platform/external/chromium_org/third_party/webrtc

26a736f Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
5eca0c7 Fix breakage after introducing new test. by stefan@webrtc.org · 11 years ago
f8c47a1 Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
90e2fdd Increment RTP timestamps for padding packets by henrik.lundin@webrtc.org · 11 years ago
8f2997c Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
764b28e Disable all vie_auto_tests on Linux for now (take 2) by kjellander@webrtc.org · 11 years ago
d8dc0f5 Disable all automated vie_auto_tests on Linux for now by kjellander@webrtc.org · 11 years ago
04281a4 Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
8dda8d2 Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
7cba612 Reimplementing NetEq4's AudioVector by henrik.lundin@webrtc.org · 11 years ago
0db738b Parse next RTCP XR report block after an unsupported block type. by asapersson@webrtc.org · 11 years ago
c824f2c Reducing opus_test runtime to pass Android test by minyue@webrtc.org · 11 years ago
e5efa32 MIPS optimizations for AECM audio processing module by andrew@webrtc.org · 11 years ago
7821bd1 Move audio_processing dependencies to a variable. by andrew@webrtc.org · 11 years ago
590c60f Remove ".." from include_dirs in build/common. by pbos@webrtc.org · 11 years ago
01966bb Remove unnecessary include_dirs from audio_processing. by andrew@webrtc.org · 11 years ago
6dc6e03 Remove unneeded includes from trace_posix.cc. by andrew@webrtc.org · 11 years ago
e9274ae Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
6106bbc Fix log build error for Chromium builds. by henrikg@webrtc.org · 11 years ago
e0df4d7 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
0aa16d7 Replace disabled logging with a restricted logging mode. by andrew@webrtc.org · 11 years ago
c4a7861 Updated WebRTC version to 3.46 by elham@webrtc.org · 11 years ago
6196a56 Fix for video_processor_intergration_tests to run in parallel. by marpan@webrtc.org · 11 years ago
685e91a Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
1dc0158 Sending status fix for module. by asapersson@webrtc.org · 11 years ago
c359e28 Add missing dependencies to .isolate files by kjellander@webrtc.org · 11 years ago
4e0ea6a Fix broken build on x86 Android by fischman@webrtc.org · 11 years ago
65e4415 Removed unused code. by asapersson@webrtc.org · 11 years ago
e3709a8 Make video quality analysis unittests print to log instead of stdout. by kjellander@webrtc.org · 11 years ago
06977ab Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
f5fdd0c Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
a4a5bf2 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
987587e Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
565f991 Address Clag Analyzer issues. by turaj@webrtc.org · 11 years ago
f1262f3 Propagate estimated RTT from receivers to rtt observer. by asapersson@webrtc.org · 11 years ago
3c97268 Video bandwidth not reported correctly by sprang@webrtc.org · 11 years ago
0f78f7b Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc by sergeyu@chromium.org · 11 years ago
893c229 Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build. by wu@webrtc.org · 11 years ago
be03ea6 Add delay limit to ChokeFilter. by solenberg@webrtc.org · 11 years ago
77c834d Logging for BWE test framework. by solenberg@webrtc.org · 11 years ago
9a1635a Make video/ only depend on video_engine_core. by pbos@webrtc.org · 11 years ago
6671434 Stop DirectTransports in VideoSendStreamTests. by pbos@webrtc.org · 11 years ago
267f694 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. by turaj@webrtc.org · 11 years ago
9ce61d4 Adding tl0idx consideration for continuity by mikhal@webrtc.org · 11 years ago
56290ed Fix build/isolate.gypi path in webrtc_tests.gypi. by pbos@webrtc.org · 11 years ago
8e3e298 Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. by fischman@webrtc.org · 11 years ago
b581c90 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
d4ec1f5 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
4043e7e Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
b397091 Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h by xians@webrtc.org · 11 years ago
d080e35 Added a "interleaved_" flag to webrtc::AudioFrame. by xians@webrtc.org · 11 years ago
e2f3ebc Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_. by andrew@webrtc.org · 11 years ago
ae2b602 Change the low-bitrate handling in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
7af2f81 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
d6b231e Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined. by andrew@webrtc.org · 11 years ago
cddf2b1 Add an extended filter option to audioproc. by andrew@webrtc.org · 11 years ago
a881576 Fix for incorrect RTT estimation. A too low RTT value could be estimated. by asapersson@webrtc.org · 11 years ago
ce21c82 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
7606f43 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
2ba95be Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
e5fd264 Roll chromium_revision 228675:229708 by kjellander@webrtc.org · 11 years ago
a597fc2 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. by andrew@webrtc.org · 11 years ago
4535225 Add CurrentLayerId() to temporal layers. by marpan@webrtc.org · 11 years ago
1e1938a Updated WebRTC version to 3.45 by elham@webrtc.org · 11 years ago
61e533c Framework for testing bandwidth estimation. by solenberg@webrtc.org · 11 years ago
97d0fc6 Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
0c6fa57 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
3ba57eb Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
dea5a74 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
05ee189 Move audio_e2e_harness into include_tests==1 condition. by kjellander@webrtc.org · 11 years ago
8ee76b9 Add audio_e2e_test target to tools.gyp by kjellander@webrtc.org · 11 years ago
6c0739e 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
9b307a7 Have padding decay to zero if no frames are being captured. by stefan@webrtc.org · 11 years ago
6be4250 Disable the -Wno-unused-const-variable Clang warning on Mac by kjellander@webrtc.org · 11 years ago
c4579f3 Minor comment fix after clang reformat. by andrew@webrtc.org · 11 years ago
b746a33 MouseCursorMonitor implementation for OSX and Windows. by sergeyu@chromium.org · 11 years ago
224c0f5 Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
1e6493d Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
b1ef0d7 Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
de16548 Remove unused kPowTableFrac which causes anroid clang build failure. by wu@webrtc.org · 11 years ago
6133dd5 Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
e44b42d Don't pad if only one stream is sent, except if auto muted. by stefan@webrtc.org · 11 years ago
a19dab9 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
4fe8543 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
55ca27e Run FullStack tests without render windows. by pbos@webrtc.org · 11 years ago
1c83344 Remove TSan v2 disabled test in condition_variable_unittest.cc by kjellander@webrtc.org · 11 years ago
976adc0 Open file in binary in CreateFromYuvFile(). by pbos@webrtc.org · 11 years ago
28f6166 Add MouseCursorRenderer. by sergeyu@chromium.org · 11 years ago
92da5d7 Add MouseCursorCapturer interface with implementation for X11. by sergeyu@chromium.org · 11 years ago
0f281aa Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
7865560 Make RtpData and RtpFeedback destructors public. by stefan@webrtc.org · 11 years ago
2390091 Move ChromaGenerator to common_video/. by pbos@webrtc.org · 11 years ago
8dc840d Compile out unused kMinTrustedDelayMs. by andrew@webrtc.org · 11 years ago
c1b7718 Android: Fixes WebRTCDemo build (missing Java code). by henrike@webrtc.org · 11 years ago
53fa5da NetEq4: Removing templatization for AudioVector by henrik.lundin@webrtc.org · 11 years ago
f24a93f Remove empty line in SharedXDisplay::RemoveEventHandler. by sergeyu@chromium.org · 11 years ago
a171eae Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. by henrike@webrtc.org · 11 years ago
5e8b020 Add event handling in SharedXDisplay. by sergeyu@chromium.org · 11 years ago
a6295d3 Add DesktopCaptureOptions class. by sergeyu@chromium.org · 11 years ago
4b795a1 WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties. by henrike@webrtc.org · 11 years ago