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AgeCommit message (Expand)Author
2014-06-16Implements start bitrate for new video API.mflodman@webrtc.org
2014-06-12Enable pacing by default and remove the option to disable it from the new API.stefan@webrtc.org
2014-06-11Add APIs to enable padding with redundant payloads.stefan@webrtc.org
2014-06-06Make VideoSendStream/VideoReceiveStream configs const.pbos@webrtc.org
2014-06-05Adding back platform specific renderer to video loopback test.mflodman@webrtc.org
2014-06-05Have RTX be enabled by setting an RTX payload type instead of by setting an R...stefan@webrtc.org
2014-05-29Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avo...wu@webrtc.org
2014-05-26First incoming packet was not accounted for in receive stats. Changed call or...asapersson@webrtc.org
2014-05-23Revert "Revert "Remove VideoSendStreamInput::PutFrame.""pbos@webrtc.org
2014-05-23Revert "Remove VideoSendStreamInput::PutFrame."pbos@webrtc.org
2014-05-23Remove VideoSendStreamInput::PutFrame.pbos@webrtc.org
2014-05-20Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky.stefan@webrtc.org
2014-05-15Fix Win VideoSendStream::...::ToString() compiles.pbos@webrtc.org
2014-05-15Add ToString() to VideoSendStream::Config.pbos@webrtc.org
2014-05-14Add DeliveryStatus enum to DeliverPacket().pbos@webrtc.org
2014-05-14Re-enable the BitrateEstimatorTest cases for the Call API.solenberg@webrtc.org
2014-05-13Move gflags usage to video_loopback.pbos@webrtc.org
2014-05-07Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can b...wu@webrtc.org
2014-05-05* Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter.wu@webrtc.org
2014-04-28Add thread annotations to Call API.pbos@webrtc.org
2014-04-28Disable flaky CaptureNtpTimeWithNetworkJitter.pbos@webrtc.org
2014-04-28Disabling flaky CanReceiveFec.pbos@webrtc.org
2014-04-24Calculate local/remote clock delta and capture ntp timestamp in receiver's ti...wu@webrtc.org
2014-04-24Remove TraceCallback use from Call.pbos@webrtc.org
2014-04-24Rename Start/Stop in Video{Send,Receive}Streams.pbos@webrtc.org
2014-04-22Let A/V sync test use default AudioCoding modulehenrik.lundin@webrtc.org
2014-04-14Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.fischman@webrtc.org
2014-04-10Disable UsesTraceCallbackpbos@webrtc.org
2014-04-08Implement FEC support in VideoReceiveStream.pbos@webrtc.org
2014-04-08Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.andresp@webrtc.org
2014-04-07Clean up traces and logs in RemoteBitrateEstimator.stefan@webrtc.org
2014-03-26Re-submit: rev5775andresp@webrtc.org
2014-03-25Revert 5775 "Modify bitrate controller to update bitrate based o..."andrew@webrtc.org
2014-03-25Change sprintf format string from %zu to %ihenrik.lundin@webrtc.org
2014-03-25Modify bitrate controller to update bitrate based on process call and notandresp@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
2014-03-20Extend perf tests to perform rampup on single stream.andresp@webrtc.org
2014-03-19Disabling SendsSetSimulcastSsrcs.pbos@webrtc.org
2014-03-19Disable flaky CanSwitchToUseAllSsrcs.pbos@webrtc.org
2014-03-19Simplify pacer interface.pbos@webrtc.org
2014-03-19Remove internal codecs from VideoSendStream.pbos@webrtc.org
2014-03-18Re-comitting r5711: "Fixing a flaky test in video_engine_tests"henrik.lundin@webrtc.org
2014-03-18Revert 5711 "Fixing a flaky test in video_engine_tests"turaj@webrtc.org
2014-03-17Fixing a flaky test in video_engine_testshenrik.lundin@webrtc.org
2014-03-17Refactor rampup tests:andresp@webrtc.org
2014-03-13Stopping network threads before tearing down testhenrik.lundin@webrtc.org
2014-03-13Re-landing "Routing SuspendChange to VideoSendStream::Stats"henrik.lundin@webrtc.org
2014-03-13Implement minimum transmit bitrate.pbos@webrtc.org
2014-03-13Enable all RampUpTest.UpDownUp* testshenrik.lundin@webrtc.org
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