- 6c9726a Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
- 519d7cf Include files from webrtc/.. paths in video_processing/ by pbos@webrtc.org · 11 years ago
- d8f7f53 Include files from webrtc/.. paths in remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
- abf0cd8 Include files from webrtc/.. paths in common_audio/ by pbos@webrtc.org · 11 years ago
- 104218e Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
- 56041ab Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
- 9e0d3ec Refactor jitter buffer to use separate lists for decodable and incomplete frames. by stefan@webrtc.org · 11 years ago
- 3f2091a Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith(). by sergeyu@chromium.org · 11 years ago
- 087f8c6 Remove dead testRateControl.cc by pbos@webrtc.org · 11 years ago
- 2f30cee Removed dead testH263Parser.cc by pbos@webrtc.org · 11 years ago
- 2ea6127 Remove dead bitstreamTest.cc. by pbos@webrtc.org · 11 years ago
- 77fa22e Make sure GlxRenderer frees its resources. by pbos@webrtc.org · 11 years ago
- 1a37edd Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
- 06721fc Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
- c0dba24 CreateEmptyFrame casts from size_t to int. by pbos@webrtc.org · 11 years ago
- d9f9185 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
- 3d6a8bf Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
- 2660072 Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
- 53e452d Log the type of recycled frames. by stefan@webrtc.org · 11 years ago
- 6bd2847 Log a message when a key frame packet is received by hclam@chromium.org · 11 years ago
- 1286255 Fix failing tests on 32 bit Linux. by solenberg@webrtc.org · 11 years ago
- 6de406d 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
- d822fe4 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
- d921161 Disable WindowCapturer tests on OSX and Linux by sergeyu@chromium.org · 11 years ago
- e52fbdd Add direct_dependent_settings in common.gypi. by sergeyu@chromium.org · 11 years ago
- 2e87970 Refactor VCM/Timing. No changes in functionality. by mikhal@webrtc.org · 11 years ago
- 1da8866 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. by stefan@webrtc.org · 11 years ago
- be72fe4 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
- c9a4463 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
- 08dbe39 Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
- 9a15fc3 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
- 4602e44 Log timestamp of the frame when it's dropped from the render module by hclam@chromium.org · 11 years ago
- e37160f Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
- 5f600c8 Revert 4000 "Reverting r3978" by andrew@webrtc.org · 11 years ago
- 04958f7 Revert 4001 "Revert 3977" by andrew@webrtc.org · 11 years ago
- 3ec8ef6 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
- 02175b6 Recalibrate point sample expectation by fbarchard@google.com · 11 years ago
- 967320b Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
- 091158d Window capturer implementation for Windows. by sergeyu@chromium.org · 11 years ago
- 0313a3a Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
- ecd1591 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
- f2e6fb3 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
- 329c951 Improve wraparound handling in the render time extrapolator. by stefan@webrtc.org · 11 years ago
- 3f3dcd1 Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
- db9d0be Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
- db298d5 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
- 49ba1dc Add one unit test for NACKing a key frame by hclam@chromium.org · 11 years ago
- fc8382b Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
- 5ff68ae Avoid resetting encoder on identical settings. by pbos@webrtc.org · 11 years ago
- b421849 Bugfix: VCM would report wrong sentBitrate by marpan@webrtc.org · 11 years ago
- 4df4e2c Formatted FEC stuff. by phoglund@webrtc.org · 11 years ago
- 1d76489 Moved force_volume_max to its own gyp file to avoid a circular dependency. by phoglund@webrtc.org · 11 years ago
- f20975f Wrote a small portable tool for forcing the mic volume to 100%. by phoglund@webrtc.org · 11 years ago
- dc8c883 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago
- 3932563 Log too long non-decodable duration events. by stefan@webrtc.org · 11 years ago
- 864f9d7 Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
- 1e77b3b Add handling of the absolute send time header extension to the rtp_rtcp module. by solenberg@webrtc.org · 11 years ago
- ba4ccdd Updating NACK RTX test by mikhal@webrtc.org · 11 years ago
- dfdfaf5 VCM/JB: Bug fix in ExtractAndSetDecode BUG=1771 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
- 8197221 RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. by solenberg@webrtc.org · 11 years ago
- e07cbc5 CoreAudio Win: release resources safely under certain rare circumstances. by braveyao@webrtc.org · 11 years ago
- 5fa31f7 Linux support for typing detection by niklas.enbom@webrtc.org · 11 years ago
- 356329b Address sanitizer out of bounds read in iSAC by turaj@webrtc.org · 11 years ago
- 1a25618 Remove const for plain data types in common_video/ by pbos@webrtc.org · 11 years ago
- 2b2e78c Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
- 1aecacb Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. by stefan@webrtc.org · 11 years ago
- 28a9f65 Removed Mac capture crash and memory leak. by mflodman@webrtc.org · 11 years ago
- 6bbd8b1 Add script for comparing video quality by kjellander@webrtc.org · 11 years ago
- b2a298c Reformatted FEC tables. by phoglund@webrtc.org · 11 years ago
- 60003b2 Remove const for plain data types in common_audio/ by pbos@webrtc.org · 11 years ago
- 07a1c11 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
- 1e8424f Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 11 years ago
- 4981e61 Fix typo in log statement. witdh should be width. by fbarchard@google.com · 11 years ago
- 302f731 Add more tracing for key frames. by justinlin@chromium.org · 11 years ago
- f308b75 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. by vikasmarwaha@webrtc.org · 11 years ago
- 34d0fec Updated WebRTC version to 3.31 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
- aedb73b Revert 4008 "Avoid resetting video encoder for similar configs." by phoglund@webrtc.org · 11 years ago
- 196ed2e Disabled flaky codec test (RunsCodecTestWithoutErrors) by phoglund@webrtc.org · 11 years ago
- 16dfb75 Avoid resetting video encoder for similar configs. by pbos@webrtc.org · 11 years ago
- ad2b368 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
- af6696e Remove TEXT(x) for BUILDINFO macros. by pbos@webrtc.org · 11 years ago
- 93219bb Added a config class to ease passing a set of options across webrtc. by andresp@webrtc.org · 11 years ago
- ffeeec8 Add svn:eol-style back which is lost in r3993 mistakenly. by braveyao@webrtc.org · 11 years ago
- 601501f Revert 3977 BUG=webrtc:1749 by tnakamura@webrtc.org · 11 years ago
- e88f9d5 Reverting r3978 by elham@webrtc.org · 11 years ago
- 98a1ee2 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
- df08ae4 Use 2 threads for HD, or 1 for VGA or less. by fbarchard@google.com · 11 years ago
- b181cac 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
- 4ddb5bd WebRTCDemo Android doesn't hangle activity recreation correctly. by braveyao@webrtc.org · 11 years ago
- 6483be5 Drop Virtual webcam check script as moved into buildbot scripts. by kjellander@webrtc.org · 11 years ago
- f795df0 Add fischman into OWNERS of WebRTCDemo Android. by braveyao@webrtc.org · 11 years ago
- d3d364e Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 11 years ago
- ae2d248 Update SincResampler with the latest Chromium code. by andrew@webrtc.org · 11 years ago
- e155626 Clean creation of VideoEngine: by andresp@webrtc.org · 11 years ago
- 6027565 Formatted dtmf_queue. by phoglund@webrtc.org · 11 years ago
- 5187bfa Add script to ensure virtual webcam is running. by kjellander@webrtc.org · 11 years ago
- 3759823 Disable clang C++11 warnings to permit OVERRIDE keyword. by pbos@webrtc.org · 11 years ago
- bf8b98a Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem. by stefan@webrtc.org · 11 years ago
- db2e80b Enable protobuf use in Chromium. by andrew@webrtc.org · 11 years ago
- ee706f6 Update protoc.gypi to match Chromium's latest. by andrew@webrtc.org · 11 years ago