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AgeCommit message (Expand)Author
2014-07-24Make sure padding is sent on the first sending RTP module.mflodman@webrtc.org
2014-07-23Fix flaky ramp-up test.stefan@webrtc.org
2014-07-20Check before send/receive rtp header extensions.pbos@webrtc.org
2014-07-15Make RTCP sender report send media bytes.pbos@webrtc.org
2014-07-11Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.stefan@webrtc.org
2014-07-11Remove the send-side cname getter APIs from voice and video engine.stefan@webrtc.org
2014-07-10Skip encoding in fake VP8 encoder.pbos@webrtc.org
2014-07-10Support VP8 encoder settings in VideoSendStream.pbos@webrtc.org
2014-07-09Add full stack test cases with a fake network pipe.stefan@webrtc.org
2014-07-08Refactor ramp-up tests to have separate help files for the test classes, to m...stefan@webrtc.org
2014-07-08Some refactoring inside rtp_rtcp/.pbos@webrtc.org
2014-07-08Extract RTP-header SSRC inline in Call.pbos@webrtc.org
2014-07-07Add test for VideoEncoder setup/teardown.pbos@webrtc.org
2014-07-07Preserve RTP states for restarted VideoSendStreams.pbos@webrtc.org
2014-07-07Fix data races related with traces in bitrate estimator test.andresp@webrtc.org
2014-07-07Remove GetDefaultConfigs() from Call.pbos@webrtc.org
2014-07-04Add boilerplate code for H.264.stefan@webrtc.org
2014-07-04Configure RTX send status on new modules.pbos@webrtc.org
2014-07-04Adding pbos as video/ owner and removing persons never working with this folder.mflodman@webrtc.org
2014-06-30Reserve RTP/RTCP modules in SetSSRC.pbos@webrtc.org
2014-06-27Refactor Call-based tests.pbos@webrtc.org
2014-06-23GN: Add BUILD.gn files + kjellander to OWNERSkjellander@webrtc.org
2014-06-20Add tests of texture frames in video_send_stream_test.wuchengli@chromium.org
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
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