- 074eb20 Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
- b59962f Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
- 2e37985 Log the type of recycled frames. by stefan@webrtc.org · 11 years ago
- 9de67da Log a message when a key frame packet is received by hclam@chromium.org · 11 years ago
- de0b5fa Fix failing tests on 32 bit Linux. by solenberg@webrtc.org · 11 years ago
- d557734 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
- b7716d8 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
- 0204219 Disable WindowCapturer tests on OSX and Linux by sergeyu@chromium.org · 11 years ago
- ced13a5 Add direct_dependent_settings in common.gypi. by sergeyu@chromium.org · 11 years ago
- 389bb40 Refactor VCM/Timing. No changes in functionality. by mikhal@webrtc.org · 11 years ago
- 3740808 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. by stefan@webrtc.org · 11 years ago
- 471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
- 8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
- 141a00c Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
- 6169712 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
- 15bdfdf Log timestamp of the frame when it's dropped from the render module by hclam@chromium.org · 11 years ago
- cca5086 Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
- 366d158 Revert 4000 "Reverting r3978" by andrew@webrtc.org · 11 years ago
- 9038990 Revert 4001 "Revert 3977" by andrew@webrtc.org · 11 years ago
- 0e15695 Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension. by solenberg@webrtc.org · 11 years ago
- 6595271 Recalibrate point sample expectation by fbarchard@google.com · 11 years ago
- 453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
- e032f9f Window capturer implementation for Windows. by sergeyu@chromium.org · 11 years ago
- b9e5732 Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
- 9ea8c99 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
- 281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
- c7979e0 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 11 years ago
- de93f2c Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
- e8dc588 Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
- 8787048 Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. by turaj@webrtc.org · 11 years ago
- 673f50d Add one unit test for NACKing a key frame by hclam@chromium.org · 11 years ago
- 5a22c40 Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
- f6cb4b6 Avoid resetting encoder on identical settings. by pbos@webrtc.org · 11 years ago
- d434891 Bugfix: VCM would report wrong sentBitrate by marpan@webrtc.org · 11 years ago
- 4efbd60 Formatted FEC stuff. by phoglund@webrtc.org · 11 years ago
- fefc490 Moved force_volume_max to its own gyp file to avoid a circular dependency. by phoglund@webrtc.org · 11 years ago
- a4f0d20 Wrote a small portable tool for forcing the mic volume to 100%. by phoglund@webrtc.org · 11 years ago
- 2a9108f New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago
- 069f626 Log too long non-decodable duration events. by stefan@webrtc.org · 11 years ago
- 7645e4d Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
- a0b0025 Add handling of the absolute send time header extension to the rtp_rtcp module. by solenberg@webrtc.org · 11 years ago
- b960975 Updating NACK RTX test by mikhal@webrtc.org · 11 years ago
- 9c94651 VCM/JB: Bug fix in ExtractAndSetDecode BUG=1771 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
- 8129578 RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. by solenberg@webrtc.org · 11 years ago
- 88ba0a5 CoreAudio Win: release resources safely under certain rare circumstances. by braveyao@webrtc.org · 11 years ago
- 32b4e2f Linux support for typing detection by niklas.enbom@webrtc.org · 11 years ago
- f6e0404 Address sanitizer out of bounds read in iSAC by turaj@webrtc.org · 11 years ago
- 2cc529f Remove const for plain data types in common_video/ by pbos@webrtc.org · 11 years ago
- 0425392 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
- 5ba433c Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. by stefan@webrtc.org · 11 years ago
- 1fc08d3 Removed Mac capture crash and memory leak. by mflodman@webrtc.org · 11 years ago
- dc2b152 Add script for comparing video quality by kjellander@webrtc.org · 11 years ago
- efbf737 Reformatted FEC tables. by phoglund@webrtc.org · 11 years ago
- d10d6e1 Remove const for plain data types in common_audio/ by pbos@webrtc.org · 11 years ago
- ca7a9a2 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
- ee6f8a2 Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 11 years ago
- fb20e53 Fix typo in log statement. witdh should be width. by fbarchard@google.com · 11 years ago
- d474c13 Add more tracing for key frames. by justinlin@chromium.org · 11 years ago
- dcfeff7 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. by vikasmarwaha@webrtc.org · 11 years ago
- 08fe40f Updated WebRTC version to 3.31 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
- 99616fa Revert 4008 "Avoid resetting video encoder for similar configs." by phoglund@webrtc.org · 11 years ago
- 9062a9a Disabled flaky codec test (RunsCodecTestWithoutErrors) by phoglund@webrtc.org · 11 years ago
- 0dd675d Avoid resetting video encoder for similar configs. by pbos@webrtc.org · 11 years ago
- ac6d919 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
- 7d6e2a0 Remove TEXT(x) for BUILDINFO macros. by pbos@webrtc.org · 11 years ago
- 1dba621 Added a config class to ease passing a set of options across webrtc. by andresp@webrtc.org · 11 years ago
- e56cf2c Add svn:eol-style back which is lost in r3993 mistakenly. by braveyao@webrtc.org · 11 years ago
- 22aedca Revert 3977 BUG=webrtc:1749 by tnakamura@webrtc.org · 11 years ago
- 4de065d Reverting r3978 by elham@webrtc.org · 11 years ago
- 15c0af4 This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build. by fischman@webrtc.org · 11 years ago
- 238aa38 Use 2 threads for HD, or 1 for VGA or less. by fbarchard@google.com · 11 years ago
- f22bfed Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation. by fischman@webrtc.org · 11 years ago
- af89cc3 WebRTCDemo Android doesn't hangle activity recreation correctly. by braveyao@webrtc.org · 11 years ago
- 41b55fa Drop Virtual webcam check script as moved into buildbot scripts. by kjellander@webrtc.org · 11 years ago
- 4afe86b Add fischman into OWNERS of WebRTCDemo Android. by braveyao@webrtc.org · 11 years ago
- 7ab7268 Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 11 years ago
- 5dea86a Update SincResampler with the latest Chromium code. by andrew@webrtc.org · 11 years ago
- 90f05ed Clean creation of VideoEngine: by andresp@webrtc.org · 11 years ago
- a149ea3 Formatted dtmf_queue. by phoglund@webrtc.org · 11 years ago
- 4be3afb Add script to ensure virtual webcam is running. by kjellander@webrtc.org · 11 years ago
- d7ebd68 Disable clang C++11 warnings to permit OVERRIDE keyword. by pbos@webrtc.org · 11 years ago
- 9ccfe46 Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem. by stefan@webrtc.org · 11 years ago
- acdfffb Enable protobuf use in Chromium. by andrew@webrtc.org · 11 years ago
- bfa5ee2 Update protoc.gypi to match Chromium's latest. by andrew@webrtc.org · 11 years ago
- 28832e1 Refactoring for typing detection by niklas.enbom@webrtc.org · 11 years ago
- 06ad384 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 11 years ago
- 38fb7b0 VCM/Receiver: Return null when can't extract frame. by mikhal@webrtc.org · 11 years ago
- 957f938 Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest by mikhal@webrtc.org · 11 years ago
- 933f885 Relanding r3952: VCM: Updating receiver logic BUG=r1734 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
- 2fc86ee VCM/JB: Break and skip to key if possible by mikhal@webrtc.org · 11 years ago
- 98b2011 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 11 years ago
- 7bdebfd Fix jitter buffer unittest. by stefan@webrtc.org · 11 years ago
- 73631c9 Correctly add packets to nack list when sequence number wraps. by stefan@webrtc.org · 11 years ago
- 49e9c6c Fix crash in pacer. by pwestin@webrtc.org · 11 years ago
- 20eb558 Revert r3952 "VCM: Updating receiver logic" by stefan@webrtc.org · 11 years ago
- a9fc587 Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest." by stefan@webrtc.org · 11 years ago
- 000fca3 Landing 1399004, Minor clean up on the un-used _measureDelay code by xians@webrtc.org · 11 years ago
- 02ae32e Add an option to override the TestToStderr trace printout time. by andrew@webrtc.org · 11 years ago
- a0975ed Consolidate all third party licenses in LICENSE_THIRD_PARTY. by andrew@webrtc.org · 11 years ago
- b6c7447 Updated WebRTC version number to 3.30 by elham@webrtc.org · 11 years ago