- 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
- 0e8ff34 Refactoring for typing detection by niklas.enbom@webrtc.org · 11 years ago
- 89f9266 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 11 years ago
- 5e0194b VCM/Receiver: Return null when can't extract frame. by mikhal@webrtc.org · 11 years ago
- 570c6be Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest by mikhal@webrtc.org · 11 years ago
- 47f874a Relanding r3952: VCM: Updating receiver logic BUG=r1734 R=stefan@webrtc.org by mikhal@webrtc.org · 11 years ago
- 3acb72d VCM/JB: Break and skip to key if possible by mikhal@webrtc.org · 11 years ago
- 34e0403 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 11 years ago
- 5762c22 Fix jitter buffer unittest. by stefan@webrtc.org · 11 years ago
- 6ef6170 Correctly add packets to nack list when sequence number wraps. by stefan@webrtc.org · 11 years ago
- 358846f Fix crash in pacer. by pwestin@webrtc.org · 11 years ago
- d51d500 Revert r3952 "VCM: Updating receiver logic" by stefan@webrtc.org · 11 years ago
- b0fca12 Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest." by stefan@webrtc.org · 11 years ago
- 79971c6 Landing 1399004, Minor clean up on the un-used _measureDelay code by xians@webrtc.org · 11 years ago
- caba49f Add an option to override the TestToStderr trace printout time. by andrew@webrtc.org · 11 years ago
- cff84ec Consolidate all third party licenses in LICENSE_THIRD_PARTY. by andrew@webrtc.org · 11 years ago
- e0aad3c Updated WebRTC version number to 3.30 by elham@webrtc.org · 11 years ago
- 25d4818 VCM/JB: Porting jitter_buffer_test to gtest. by mikhal@webrtc.org · 11 years ago
- 92b5ea0 Remove 44.1 kHz workaround from AudioDevice on PulseAudio. by andrew@webrtc.org · 11 years ago
- 6cd8727 Remove 44.1 kHz workaround from AudioDevice on WASAPI. by andrew@webrtc.org · 11 years ago
- 8e35807 Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). by sergeyu@chromium.org · 11 years ago
- cb0cb7d VCM: Updating receiver logic by mikhal@webrtc.org · 11 years ago
- 5398063 Correct and update dir name by leozwang@webrtc.org · 11 years ago
- c22830f Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 11 years ago
- a257915 Get rid of some unnecessary copying when sending REMBs. by solenberg@webrtc.org · 11 years ago
- 7ee60d0 Formatting ACM tests by tina.legrand@webrtc.org · 11 years ago
- 3794539 Fix when SetMinimumPlayoutDelay is configured to 0 by pwestin@webrtc.org · 11 years ago
- 76318c5 Removing bad code resulting in flaky test. by pwestin@webrtc.org · 11 years ago
- c06da8c Adding trace and changing pacing constants by pwestin@webrtc.org · 11 years ago
- 6c98c86 Update third party license file by niklas.enbom@webrtc.org · 11 years ago
- 8f5edba Bugfix custom call stop. by pwestin@webrtc.org · 11 years ago
- c77e4da Allow voe_cmd_test to select Opus mono (now the default). by andrew@webrtc.org · 11 years ago
- c875f20 Relax VoE's max packet length threshold. by andrew@webrtc.org · 11 years ago
- 86f267b Disabled flaky test. by phoglund@webrtc.org · 11 years ago
- 2d6a699 Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 11 years ago
- bbb54b3 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 11 years ago
- 74161fc WebRTCDemo Android app to route audio to headphone when it's plugged in. by braveyao@webrtc.org · 11 years ago
- b1c40c5 Replace Resampler with PushResampler in transmit_mixer. by andrew@webrtc.org · 11 years ago
- a23b051 Consolidate common_audio into a single target. by andrew@webrtc.org · 11 years ago
- 22a3795 Add AEC suppression level option to audioproc. by andrew@webrtc.org · 11 years ago
- 10600ab Move WEBRTC_THREAD_RR and WEBRTC_CLOCK_TYPE_REALTIME to system_wrappers.gypi . by sergeyu@chromium.org · 11 years ago
- 9e65a61 Fixes two bugs in receive statistics. by stefan@webrtc.org · 11 years ago
- c5fbd58 Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync. by pwestin@webrtc.org · 11 years ago