- 561990f - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
- 6ec2507 Disable WindowCapturer tests on OSX and Linux by sergeyu@chromium.org · 11 years ago
- 6ebfd34 Add direct_dependent_settings in common.gypi. by sergeyu@chromium.org · 11 years ago
- 5f8f112 Not to request to TURN server for local tests. Follow-up work to issue1197. by braveyao@webrtc.org · 11 years ago
- 106afff Roll libvpx to 196669. -pick up libvpx roll to 9981006d by marpan@webrtc.org · 11 years ago
- 2eaf98b Refactor VCM/Timing. No changes in functionality. by mikhal@webrtc.org · 11 years ago
- 3417eb4 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. by stefan@webrtc.org · 11 years ago
- 956aa7e Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
- 8a025e2 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
- d2541e8 Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
- 375deb4 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
- 0d540c3 Log timestamp of the frame when it's dropped from the render module by hclam@chromium.org · 11 years ago
- 69bb348 Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
- ac0ef48 Revert 4067 "libyuv roll to r698 for Core Media fourccs for OSX ..." by andrew@webrtc.org · 11 years ago
- f9825e5 Revert 4000 "Reverting r3978" by andrew@webrtc.org · 11 years ago
- 225f2b8 Revert 4001 "Revert 3977" by andrew@webrtc.org · 11 years ago
- c0352d5 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
- e5794cb Recalibrate point sample expectation by fbarchard@google.com · 11 years ago
- a58d729 libyuv roll to r698 for Core Media fourccs for OSX camtwist support and performance improvements in ARGB scaler. by fbarchard@google.com · 11 years ago
- cb9cff0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
- b10ccbe Window capturer implementation for Windows. by sergeyu@chromium.org · 11 years ago
- 5e2a1bb AppRTC: make requestTurn() failure non-fatal to call establishment. by fischman@webrtc.org · 11 years ago
- 8d6eb56 Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
- 5a602d7 Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
- 2163212 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
- f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
- 9f557c1 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 11 years ago
- 14d7700 Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
- e874a8f Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
- 8630cfe 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
- fe307e1 Add one unit test for NACKing a key frame by hclam@chromium.org · 11 years ago
- b3e5acf Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
- b9bb3d1 Avoid resetting encoder on identical settings. by pbos@webrtc.org · 11 years ago
- 890f609 Bugfix: VCM would report wrong sentBitrate by marpan@webrtc.org · 11 years ago
- 9919ad5 Formatted FEC stuff. by phoglund@webrtc.org · 11 years ago
- 5c1948d Moved force_volume_max to its own gyp file to avoid a circular dependency. by phoglund@webrtc.org · 11 years ago
- 61d3c55 Wrote a small portable tool for forcing the mic volume to 100%. by phoglund@webrtc.org · 11 years ago
- 29d5839 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago
- 2038214 Log too long non-decodable duration events. by stefan@webrtc.org · 11 years ago
- 4dee309 Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
- 7ebbea1 Add handling of the absolute send time header extension to the rtp_rtcp module. by solenberg@webrtc.org · 11 years ago
- 59a0667 Updated apprtc demo to interop with firefox. by vikasmarwaha@webrtc.org · 11 years ago
- 40298d4 Added webaudio-and-webtrc.html to the demos index.html. by vikasmarwaha@webrtc.org · 11 years ago
- 8c2e78b Roll chromium_revision 193311:199267 by fischman@webrtc.org · 11 years ago
- 6cfa390 Updating NACK RTX test by mikhal@webrtc.org · 11 years ago
- cb20a5b VCM/JB: Bug fix in ExtractAndSetDecode BUG=1771 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
- 5add4ad RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. by solenberg@webrtc.org · 11 years ago
- c93b1d0 CoreAudio Win: release resources safely under certain rare circumstance in GTalkplugin by braveyao@webrtc.org · 11 years ago
- e2a8006 Linux support for typing detection by niklas.enbom@webrtc.org · 11 years ago
- 4ce8389 Address sanitizer out of bounds read in iSAC by turaj@webrtc.org · 11 years ago
- 6bee05a Remove const for plain data types in common_video/ by pbos@webrtc.org · 11 years ago
- 29b2219 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
- 1673481 Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. by stefan@webrtc.org · 11 years ago
- 736c6f7 Fixed more perf expectations. by phoglund@webrtc.org · 11 years ago
- 80c7e3b Adjusted perf expectations for mac large tests. by phoglund@webrtc.org · 11 years ago
- bb984f5 Removed Mac capture crash and memory leak. by mflodman@webrtc.org · 11 years ago
- a6ff845 Add script for comparing video quality by kjellander@webrtc.org · 11 years ago
- 6d07ad9 Added protoc_wrapper to blacklist, fixed tools/PRESUBMIT.py which was passing in the wrong args to CheckLongLines. by phoglund@webrtc.org · 11 years ago
- 527f6c6 Reformatted FEC tables. by phoglund@webrtc.org · 11 years ago
- 8e3b594 Remove const for plain data types in common_audio/ by pbos@webrtc.org · 11 years ago
- 9213521 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
- 185bae4 Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 11 years ago
- c9cb4ff Fix typo in log statement. witdh should be width. by fbarchard@google.com · 11 years ago
- 7bfb3a3 Add more tracing for key frames. by justinlin@chromium.org · 11 years ago
- 941fcc5 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. by vikasmarwaha@webrtc.org · 11 years ago
- 1993a55 Added Stereo url paramter to apprtc demo. by vikasmarwaha@webrtc.org · 11 years ago
- 52b3905 Updated WebRTC version to 3.31 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
- 43bf6ce Revert 4008 "Avoid resetting video encoder for similar configs." by phoglund@webrtc.org · 11 years ago
- c53480f Disabled flaky codec test (RunsCodecTestWithoutErrors) by phoglund@webrtc.org · 11 years ago
- aa4efd1 Avoid resetting video encoder for similar configs. by pbos@webrtc.org · 11 years ago
- 7707d06 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
- 7a5615b New WebAudio-WebRTC demo. by henrika@webrtc.org · 11 years ago
- 7ee8228 Remove TEXT(x) for BUILDINFO macros. by pbos@webrtc.org · 11 years ago
- 6b68c28 Added a config class to ease passing a set of options across webrtc. by andresp@webrtc.org · 11 years ago
- 9ecd686 Add svn:eol-style back which is lost in r3993 mistakenly. by braveyao@webrtc.org · 11 years ago
- a404d1d Change watchlist. by leozwang@webrtc.org · 11 years ago
- 7311083 Revert 3977 BUG=webrtc:1749 by tnakamura@webrtc.org · 11 years ago
- 05ea12f Reverting r3978 by elham@webrtc.org · 11 years ago
- d6ed000 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
- 571b336 Updating perf by mikhal@webrtc.org · 11 years ago
- 1e3c794 Use 2 threads for HD, or 1 for VGA or less. by fbarchard@google.com · 11 years ago
- 0680670 Updating perf by mikhal@webrtc.org · 11 years ago
- 6a36f0e 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
- e525309 WebRTCDemo Android doesn't hangle activity recreation correctly. by braveyao@webrtc.org · 11 years ago
- 219762a Drop Virtual webcam check script as moved into buildbot scripts. by kjellander@webrtc.org · 11 years ago
- ebdfa8d Add fischman into OWNERS of WebRTCDemo Android. by braveyao@webrtc.org · 11 years ago
- d72262d Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 11 years ago
- c6a3755 Update SincResampler with the latest Chromium code. by andrew@webrtc.org · 11 years ago
- 4427273 Clean creation of VideoEngine: by andresp@webrtc.org · 11 years ago
- 6155be2 Add /tools/protoc_wrappers to .gitignore. by andrew@webrtc.org · 11 years ago
- aeb7d87 Tweaked webrtc_reformat. by phoglund@webrtc.org · 11 years ago
- 315d398 Formatted dtmf_queue. by phoglund@webrtc.org · 11 years ago
- 73a4d5a Add script to ensure virtual webcam is running. by kjellander@webrtc.org · 11 years ago
- f6d67ae Disable clang C++11 warnings to permit OVERRIDE keyword. by pbos@webrtc.org · 11 years ago
- d98e784 Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem. by stefan@webrtc.org · 11 years ago
- b55a12a Enable protobuf use in Chromium. by andrew@webrtc.org · 11 years ago
- e53084f Update protoc.gypi to match Chromium's latest. by andrew@webrtc.org · 11 years ago
- 3be565b Refactoring for typing detection by niklas.enbom@webrtc.org · 11 years ago
- ef14488 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 11 years ago
- 8f86cc8 VCM/Receiver: Return null when can't extract frame. by mikhal@webrtc.org · 11 years ago