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AgeCommit message (Expand)Author
2014-05-06Update makefiles after merge of Chromium at 268379Android Chromium Automerger
2014-04-30Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-29Upping start bitrate to min, if set to a lower value i SetSendCodec.mflodman@webrtc.org
2014-04-28Update makefiles after merge of Chromium at 266543Android Chromium Automerger
2014-04-28Disable flaky RunsRtpRtcpTestWIthoutErrors.pbos@webrtc.org
2014-04-25Replace scoped_array<T> with scoped_ptr<T[]>.andrew@webrtc.org
2014-04-25Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-25Casting char to int in logs.asapersson@webrtc.org
2014-04-24Calculate local/remote clock delta and capture ntp timestamp in receiver's ti...wu@webrtc.org
2014-04-23Update makefiles after merge of Chromium at 265680Android Chromium Automerger
2014-04-23Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-15Propagate capture ntp timestamp from rtp to renderer.wu@webrtc.org
2014-04-14Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.fischman@webrtc.org
2014-04-11Cleaned up logging in video_coding.stefan@webrtc.org
2014-04-10Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-09Update makefiles after merge of Chromium at 262754Android Chromium Automerger
2014-04-08Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.andresp@webrtc.org
2014-04-07Updated WebRTC version to 3.52elham@webrtc.org
2014-04-07Update makefiles after merge of Chromium at 262110Android Chromium Automerger
2014-04-07Log Fixit for parts of video_engine folder.mflodman@webrtc.org
2014-04-04Update makefiles after merge of Chromium at 261622Android Chromium Automerger
2014-04-01Update makefiles after merge of Chromium at 260927Android Chromium Automerger
2014-03-26Re-submit: rev5775andresp@webrtc.org
2014-03-26Add API to allow deducting bitrate from incoming estimates before the capacit...solenberg@webrtc.org
2014-03-25Revert 5775 "Modify bitrate controller to update bitrate based o..."andrew@webrtc.org
2014-03-25Removing VideoCodecDerived and moving methods inside VideoCodec.mallinath@webrtc.org
2014-03-25Updated WebRTC version to 3.51elham@webrtc.org
2014-03-25Modify bitrate controller to update bitrate based on process call and notandresp@webrtc.org
2014-03-25Adding API for setting bandwidth estimation configurations.stefan@webrtc.org
2014-03-24Add configuration for ability to use the encode usage measure for triggering ...asapersson@webrtc.org
2014-03-24Implement ViE forwarding to RBE of packets for BWE coming in through the ViEN...solenberg@webrtc.org
2014-03-24Have changes to REMB trigger RTCP to be sent immediately.stefan@webrtc.org
2014-03-24VoE changes to allow forwarding of packets from VoE to ViE BWE.solenberg@webrtc.org
2014-03-24Add AIMD option to BWE API.stefan@webrtc.org
2014-03-21Refactor in BitrateController module.andresp@webrtc.org
2014-03-21Adding operator== and != methods for CodecInst and VideoCodec structures.mallinath@webrtc.org
2014-03-20Add ability to configure cpu overuse options via an API.asapersson@webrtc.org
2014-03-19Fixes RTX related bugs.stefan@webrtc.org
2014-03-19Simplify pacer interface.pbos@webrtc.org
2014-03-19Fix a deadlock in ViEEncoder::DeliverFrame.wuchengli@chromium.org
2014-03-13Implement minimum transmit bitrate.pbos@webrtc.org
2014-03-07Avoid crash in ViEEncoder::DeRegisterExternalEncoder().fischman@webrtc.org
2014-03-06Adding a new ramp-up-down-up testhenrik.lundin@webrtc.org
2014-02-27Adds APIs for reporting pacer queuing delay.jiayl@webrtc.org
2014-02-24Fix to get total number of sent and received rtcp packets.asapersson@webrtc.org
2014-02-20Updated WebRTC version to 3.50elham@webrtc.org
2014-02-20Modified overuse detection thresholds.asapersson@webrtc.org
2014-02-19Add RTCP packet type counter (for getting statistics such as sent/received NA...asapersson@webrtc.org
2014-02-18Remove external encryption API for VoE.solenberg@webrtc.org
2014-02-17Reset estimate if no frame has been seen for a certain time (to avoid large j...asapersson@webrtc.org
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