- 12a93e0 Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
- 31bd97d Fix issues with sequence number wrap-around in jitter statistics by henrik.lundin@webrtc.org · 11 years ago
- e1e050e Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
- 21dc10d Make interface destructor virtual by sprang@webrtc.org · 11 years ago
- 3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
- 3b7da1e Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. by asapersson@webrtc.org · 11 years ago
- 63e3810 Typo in vie_autotest_win.cc by braveyao@webrtc.org · 11 years ago
- 7950b98 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
- 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
- 4747585 Added ViE API for getting overuse measure. by asapersson@webrtc.org · 11 years ago
- 3051951 Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
- c4af4cf Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
- 3009c81 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
- 346dbe7 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
- 7f9f840 Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
- 964d78e Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
- 09b40ec Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
- 8b0791c Fix DesktopAndCursorComposer to restore frames to the original state. by sergeyu@chromium.org · 11 years ago
- eb45a20 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
- 4590177 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
- 09f84e5 Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
- e028410 Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
- fe5678c Fix breakage after introducing new test. by stefan@webrtc.org · 11 years ago
- 4985c7b Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
- 69b1aa4 Increment RTP timestamps for padding packets by henrik.lundin@webrtc.org · 11 years ago
- 6904054 Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
- 2f9e587 Disable all vie_auto_tests on Linux for now (take 2) by kjellander@webrtc.org · 11 years ago
- 8167387 Disable all automated vie_auto_tests on Linux for now by kjellander@webrtc.org · 11 years ago
- b748c9d Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
- b43ac9f Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
- 591be3b Reimplementing NetEq4's AudioVector by henrik.lundin@webrtc.org · 11 years ago
- f4fbef3 Parse next RTCP XR report block after an unsupported block type. by asapersson@webrtc.org · 11 years ago
- 26f5492 Reducing opus_test runtime to pass Android test by minyue@webrtc.org · 11 years ago
- ff4fc2b MIPS optimizations for AECM audio processing module by andrew@webrtc.org · 11 years ago
- 731a87b Move audio_processing dependencies to a variable. by andrew@webrtc.org · 11 years ago
- 9965e3a Remove ".." from include_dirs in build/common. by pbos@webrtc.org · 11 years ago
- 3051ff7 Remove unnecessary include_dirs from audio_processing. by andrew@webrtc.org · 11 years ago
- 4caa53b Remove unneeded includes from trace_posix.cc. by andrew@webrtc.org · 11 years ago
- 7e97e4c 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
- b89fa69 Fix log build error for Chromium builds. by henrikg@webrtc.org · 11 years ago
- b4db9c3 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
- 9705beb Replace disabled logging with a restricted logging mode. by andrew@webrtc.org · 11 years ago
- 78726d1 Updated WebRTC version to 3.46 by elham@webrtc.org · 11 years ago
- b4d7835 Fix for video_processor_intergration_tests to run in parallel. by marpan@webrtc.org · 11 years ago
- 626d764 Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
- f4def77 Sending status fix for module. by asapersson@webrtc.org · 11 years ago
- 93bf70f Add missing dependencies to .isolate files by kjellander@webrtc.org · 11 years ago
- 690a03c Fix broken build on x86 Android by fischman@webrtc.org · 11 years ago
- 1bd9a7b Removed unused code. by asapersson@webrtc.org · 11 years ago
- f8a1798 Make video quality analysis unittests print to log instead of stdout. by kjellander@webrtc.org · 11 years ago
- af92d3e Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
- a191cb0 Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
- 6baaf30 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
- 7773eec Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
- a9a788e Address Clag Analyzer issues. by turaj@webrtc.org · 11 years ago
- 72cc32a Propagate estimated RTT from receivers to rtt observer. by asapersson@webrtc.org · 11 years ago
- 6646abd Video bandwidth not reported correctly by sprang@webrtc.org · 11 years ago
- 1d76e9b Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc by sergeyu@chromium.org · 11 years ago
- 6d905e9 Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build. by wu@webrtc.org · 11 years ago
- bd5d9fa Add delay limit to ChokeFilter. by solenberg@webrtc.org · 11 years ago
- c2b6166 Logging for BWE test framework. by solenberg@webrtc.org · 11 years ago
- 0b1bf64 Make video/ only depend on video_engine_core. by pbos@webrtc.org · 11 years ago
- 86868fe Stop DirectTransports in VideoSendStreamTests. by pbos@webrtc.org · 11 years ago
- 367af84 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. by turaj@webrtc.org · 11 years ago
- 991d66a Adding tl0idx consideration for continuity by mikhal@webrtc.org · 11 years ago
- 800bc1a Fix build/isolate.gypi path in webrtc_tests.gypi. by pbos@webrtc.org · 11 years ago
- f00942a Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. by fischman@webrtc.org · 11 years ago
- 24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
- 4ce7590 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
- 2714c79 Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
- 77035fd Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h by xians@webrtc.org · 11 years ago
- 2ea3645 Added a "interleaved_" flag to webrtc::AudioFrame. by xians@webrtc.org · 11 years ago
- 52ec7b6 Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_. by andrew@webrtc.org · 11 years ago
- 191f4fe Change the low-bitrate handling in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
- ecfef19 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
- 44a8ce5 Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined. by andrew@webrtc.org · 11 years ago
- 84c1485 Add an extended filter option to audioproc. by andrew@webrtc.org · 11 years ago
- a2d942a Fix for incorrect RTT estimation. A too low RTT value could be estimated. by asapersson@webrtc.org · 11 years ago
- 6036f56 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
- 76dad96 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
- 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
- b45ab8b Roll chromium_revision 228675:229708 by kjellander@webrtc.org · 11 years ago
- d7e9041 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. by andrew@webrtc.org · 11 years ago
- 45b5167 Add CurrentLayerId() to temporal layers. by marpan@webrtc.org · 11 years ago
- 6796d68 Updated WebRTC version to 3.45 by elham@webrtc.org · 11 years ago
- 4b3ff2d Framework for testing bandwidth estimation. by solenberg@webrtc.org · 11 years ago
- 4633e15 Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
- 7c46e95 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
- 63301bd Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
- c5b5ad1 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
- 50edafc Move audio_e2e_harness into include_tests==1 condition. by kjellander@webrtc.org · 11 years ago
- b9586f0 Add audio_e2e_test target to tools.gyp by kjellander@webrtc.org · 11 years ago
- b27e670 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
- 5e74d96 Have padding decay to zero if no frames are being captured. by stefan@webrtc.org · 11 years ago
- ba368fc Disable the -Wno-unused-const-variable Clang warning on Mac by kjellander@webrtc.org · 11 years ago
- 127d8ad Minor comment fix after clang reformat. by andrew@webrtc.org · 11 years ago
- 2873c4c MouseCursorMonitor implementation for OSX and Windows. by sergeyu@chromium.org · 11 years ago
- f7651ef Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
- 3d553d4 Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
- 51e0101 Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago