summaryrefslogtreecommitdiff
path: root/video_engine
AgeCommit message (Expand)Author
2014-03-17Update makefiles after merge of Chromium at 257433Android Chromium Automerger
2014-03-17Update makefiles after merge of Chromium at 257420Android Chromium Automerger
2014-03-14Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-03-13Implement minimum transmit bitrate.pbos@webrtc.org
2014-03-12Update makefiles after merge of Chromium at 256489Android Chromium Automerger
2014-03-12Update makefiles after merge of Chromium at 256368Android Chromium Automerger
2014-03-11Update makefiles after merge of Chromium at 256314Android Chromium Automerger
2014-03-11Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-03-07Avoid crash in ViEEncoder::DeRegisterExternalEncoder().fischman@webrtc.org
2014-03-07Update makefiles after merge of Chromium at 255581Android Chromium Automerger
2014-03-06Adding a new ramp-up-down-up testhenrik.lundin@webrtc.org
2014-03-05Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-02-28Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-02-27Adds APIs for reporting pacer queuing delay.jiayl@webrtc.org
2014-02-26Update makefiles after merge of Chromium at 253634Android Chromium Automerger
2014-02-25Update makefiles after merge of Chromium at 253112Android Chromium Automerger
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
2014-02-14Adding a critical section missing in r5543.stefan@webrtc.org
2014-02-13Increase overuse and normal use thresholds for Mac.asapersson@webrtc.org
2014-02-13Fixes a race when writing to send_padding_.stefan@webrtc.org
2014-02-12Set pacing bitrates in SetEncoder.pbos@webrtc.org
2014-02-11Remove ViE external encryption API.solenberg@webrtc.org
2014-02-10Add stats of incoming frame delays for debugging bandwidth estimation.jiayl@webrtc.org
2014-01-29Connect webrtc::Config to WrappingBitrateEstimatorhenrik.lundin@webrtc.org
2014-01-29Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."mallinath@webrtc.org
2014-01-27Revert 5421 "Fix deadlock on register/unregister observer while ..."mallinath@webrtc.org
2014-01-23Fix deadlock on register/unregister observer while there is a an going callback.andresp@webrtc.org
2014-01-23Add callbacks for receive channel RTP statisticssprang@webrtc.org
2014-01-20Add configuration and test for extended RTCP reference time reports to new vi...asapersson@webrtc.org
2014-01-14Roll Chromium 238260 -> 243863wjia@webrtc.org
2014-01-13Updated Webrtc version to 3.49elham@webrtc.org
2014-01-13Removes usage of ListWrapper from several files.henrike@webrtc.org
2014-01-07Wire up statistics in video send stream of new video engine apisprang@webrtc.org
2013-12-20Add thread_annotations for clang targets.andresp@webrtc.org
2013-12-20If the configured start bitrate is higher than the configures maxmflodman@webrtc.org
2013-12-20Race condition in ViECapturer::RegisterObserversprang@webrtc.org
2013-12-19Update WebRTC to version 3.48tnakamura@webrtc.org
2013-12-19Add callbacks for receive channel RTCP statistics.sprang@webrtc.org
2013-12-18Integrate fake_network_pipe into direct_transport.stefan@webrtc.org
2013-12-18Remove media_file from VideoEngine dependencies.pbos@webrtc.org
2013-12-16Revert r5294 to re-roll r5293.pbos@webrtc.org
2013-12-15Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."turaj@webrtc.org
2013-12-13Auto instantiate RBE depending on whether AST or TOF is available in incoming...solenberg@webrtc.org
2013-12-13Callback for send bitrate estimates - new rollsprang@webrtc.org
2013-12-13Make sure channels in the same call are in the same channel group.mflodman@webrtc.org
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-23 05:23:39 +0000