summaryrefslogtreecommitdiff
path: root/video_engine
AgeCommit message (Expand)Author
2013-04-05Add GYP target for WebRTC Video demo for Android.kjellander@webrtc.org
2013-04-05Permit arbitrary payload names for kVideoCodecGeneric.pbos@webrtc.org
2013-04-04Remove WEBRTC_*_ENGINE_NETWORK_API usepwestin@webrtc.org
2013-04-04Adds event traces and counters for WebRTC receive side.edjee@google.com
2013-04-02Remove UDP transport API from ViEpwestin@webrtc.org
2013-04-01Updated Webrtc version to 3.28elham@webrtc.org
2013-03-28Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRT...solenberg@webrtc.org
2013-03-27Fix broken audio.leozwang@webrtc.org
2013-03-27Fix flakiness in network up/down event tests when running under memcheck.stefan@webrtc.org
2013-03-27WebRTCDemo: remove unnecessary stop & start during orientation change which i...fischman@webrtc.org
2013-03-27Add interface to signal a network down event.stefan@webrtc.org
2013-03-22Updated WebRTC version to 3.27 elham@webrtc.org
2013-03-22Bugfix for extended RTP/RTCP test pwestin@webrtc.org
2013-03-22Move the VIE tests to use external transport instead of the built in udp tran...pwestin@webrtc.org
2013-03-22Add min and target bitrate to VideoCodec.marpan@webrtc.org
2013-03-19Follow-up fix for r3681.stefan@webrtc.org
2013-03-18Updated WebRTC version number to 3.26elham@webrtc.org
2013-03-18Change VCM interface to take target bitrate in bits per second.stefan@webrtc.org
2013-03-18Generic video-codec support.pbos@webrtc.org
2013-03-15Adding RTX on sourcemikhal@webrtc.org
2013-03-13Revert r3667 and r3665pwestin@webrtc.org
2013-03-13 Added destructors for tests to control destruct orderpwestin@webrtc.org
2013-03-13Increasing size of nack list in buffered mode.mikhal@webrtc.org
2013-03-13Removed the engine API:s related to transport such as SetSendDestination, the...pwestin@webrtc.org
2013-03-08Lazy capture_device_info acquisition.pbos@webrtc.org
2013-03-06Enabling bufffering mode with no sync module or VoEmikhal@webrtc.org
2013-03-04Updated version number to 3.25elham@webrtc.org
2013-03-01Update integration tests for idempotent RTP header settings.bemasc@google.com
2013-03-01Destroy VCM and VPM instead of delete.mflodman@webrtc.org
2013-02-22Handle multiple calls to set initial delaymikhal@webrtc.org
2013-02-20Stop and restart fix.mflodman@webrtc.org
2013-02-20Fixed typo in vie_autotest_loopback.cc.pbos@webrtc.org
2013-02-19Rename webrtc::StatsObserver to webrtc::CallStatsObserverfischman@webrtc.org
2013-02-19fixing nack list size calculationmikhal@webrtc.org
2013-02-19Updated version number to 3.24elham@webrtc.org
2013-02-18Refactoring temporal layers implementation and adding VideoCodecMode for easi...stefan@webrtc.org
2013-02-16Add VoE interface to VieRTP testmikhal@webrtc.org
2013-02-15Adding a receive side API for buffering mode.mikhal@webrtc.org
2013-02-14Roll Chromium revision 176094:182149kjellander@webrtc.org
2013-02-14Remove MultiStreamMode from test.stefan@webrtc.org
2013-02-14Reset ssrc when calling SetSendCodec.mflodman@webrtc.org
2013-02-12Sync libvpx and its gyp wrapper from Chromium.andrew@webrtc.org
2013-02-12Increase maximum resolution to 4k x 3k. fbarchard@google.com
2013-02-11Android NDK build toolskjellander@webrtc.org
2013-02-11Set SingleStream BWE in unittests.stefan@webrtc.org
2013-02-10Updates to send side streaming mode:mikhal@webrtc.org
2013-02-08Update version number to 3.23tnakamura@webrtc.org
2013-02-07Made it possible to render custom call output to file.phoglund@webrtc.org
2013-02-06Don't report an error for GetEstimatedReceiveBandwidth if there is no validmflodman@webrtc.org
2013-02-06Enable indefinitely running vie_auto_test optionkjellander@webrtc.org
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-22 21:38:01 +0000