Log - 53c85735256dc7d540deb0a5e2bbb2f2821c4bd4 - platform/external/webrtc

53c8573 Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
5a63655 Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
0b72f58 Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
5d85819 Fix DesktopAndCursorComposer to restore frames to the original state. by sergeyu@chromium.org · 11 years ago
7a05ae5 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. by turaj@webrtc.org · 11 years ago
9c5fb76 Exclude AV-sync test from Valgrind platforms. by pbos@webrtc.org · 11 years ago
ce8e093 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
28bf50f Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
b082ade Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
4cfa605 Fix breakage after introducing new test. by stefan@webrtc.org · 11 years ago
69969e2 Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
6e95d7a Increment RTP timestamps for padding packets by henrik.lundin@webrtc.org · 11 years ago
6488761 Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
183c727 Disable datachannel_unittest.cc by sergeyu@chromium.org · 11 years ago
a23f0ca Update talk to 56619788 by sergeyu@chromium.org · 11 years ago
e872285 Disable all vie_auto_tests on Linux for now (take 2) by kjellander@webrtc.org · 11 years ago
c848985 Disable all automated vie_auto_tests on Linux for now by kjellander@webrtc.org · 11 years ago
9b82f5a Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
03f3370 Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
e8433eb Reimplementing NetEq4's AudioVector by henrik.lundin@webrtc.org · 11 years ago
3859951 Parse next RTCP XR report block after an unsupported block type. by asapersson@webrtc.org · 11 years ago
3e42726 Reducing opus_test runtime to pass Android test by minyue@webrtc.org · 11 years ago
e03cafa MIPS optimizations for AECM audio processing module by andrew@webrtc.org · 11 years ago
b073010 Move audio_processing dependencies to a variable. by andrew@webrtc.org · 11 years ago
57eb858 Remove ".." from include_dirs in build/common. by pbos@webrtc.org · 11 years ago
6e908b3 Remove unnecessary include_dirs from audio_processing. by andrew@webrtc.org · 11 years ago
00ed170 Roll libvpx 225010:232686. by marpan@webrtc.org · 11 years ago
5973f3a Remove unneeded includes from trace_posix.cc. by andrew@webrtc.org · 11 years ago
48df381 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
bff9620 Fix log build error for Chromium builds. by henrikg@webrtc.org · 11 years ago
4c828e1 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
f1a4817 Replace disabled logging with a restricted logging mode. by andrew@webrtc.org · 11 years ago
5adc897 Updated WebRTC version to 3.46 by elham@webrtc.org · 11 years ago
a7855a8 Fix for xgetbv on Visual Studio 2010. by fbarchard@google.com · 11 years ago
bde3056 Fix for video_processor_intergration_tests to run in parallel. by marpan@webrtc.org · 11 years ago
c4225b6 Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
8bad50e Sending status fix for module. by asapersson@webrtc.org · 11 years ago
16d6254 Update talk to 56183333. by wu@webrtc.org · 11 years ago
d16d307 Fix bad Google Storage uploads of resource files. by kjellander@webrtc.org · 11 years ago
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