summaryrefslogtreecommitdiff
path: root/video_engine/vie_channel_manager.cc
AgeCommit message (Collapse)Author
2014-05-12Fixes log spam introduced with r6041.stefan@webrtc.org
We shouldn't return an error if we don't yet have a valid estimate. BUG=crbug/371714 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15469006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6111 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-02Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a ↵stefan@webrtc.org
channel group instead of splitting it up among channels. This fixes an issue where the user doesn't know which channels are "active" and therefore can't properly sum the estimates for all channels. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6041 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-07Log Fixit for parts of video_engine folder.mflodman@webrtc.org
BUG=3153 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5853 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-26Add API to allow deducting bitrate from incoming estimates before the ↵solenberg@webrtc.org
capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5791 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25Adding API for setting bandwidth estimation configurations.stefan@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5773 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29Connect webrtc::Config to WrappingBitrateEstimatorhenrik.lundin@webrtc.org
This is the second CL for this change. Connection to the ViE API remains to be done. BUG=2698 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5455 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16Revert r5294 to re-roll r5293.pbos@webrtc.org
To fix races in test each stream now owns its own encoder/decoder. R=mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/5919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-15Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."turaj@webrtc.org
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. > > BUG= > R=mflodman@webrtc.org, stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/5409004 TBR=solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Auto instantiate RBE depending on whether AST or TOF is available in ↵solenberg@webrtc.org
incoming packet stream. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Add possibility to get the last processed RTT from the call stats class (to ↵asapersson@webrtc.org
be used by RTP/RTCP module). R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2383004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12Replace MapWrapper with std::map<>.pbos@webrtc.org
MapWrapper was needed on some platforms where STL wasn't supported, we now use std::map<> directly. BUG=2164 TEST=trybots R=henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2001004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4530 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ↵solenberg@webrtc.org
ChannelGroup. - Changed implementation of SetReceiveAbsoluteSendTimeStatus API so the RBE instance is changed when at least one channel in a group has the extension enabled. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1553005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4113 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17Include files from webrtc/.. paths in video_engine/pbos@webrtc.org
BUG=1662 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1492004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14Adding a factory to remote bitrate estimator and allow it to be set via config.andresp@webrtc.org
Additionally: - clean api to set remote bitrate estimator mode. - clean api to set over use detector options. R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1448006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13Wiring down config from video engine until video coding and remote bitrate ↵andresp@webrtc.org
estimator modules instantiation. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1450008 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4007 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-22Removed unused variable.mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1320013 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3881 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-22Fixing Coverity issues.mflodman@webrtc.org
BUG=C14457, C10611 Review URL: https://webrtc-codereview.appspot.com/1320012 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3880 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-14Reset ssrc when calling SetSendCodec.mflodman@webrtc.org
BUG=1398 TEST=Tested locally. Review URL: https://webrtc-codereview.appspot.com/1107004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3511 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26Wire up CallStats to provide modules with correct RTT.mflodman@webrtc.org
BUG=769 TEST=Manual test since there is no ViE APi to get RTT for receive channels. Review URL: https://webrtc-codereview.appspot.com/937027 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3163 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-13Enable paced sender. pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/965016 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-07Removed ViEBaseObserver.mflodman@webrtc.org
BUG=1037 TEST=Still compiles and ViE autotest passes. Review URL: https://webrtc-codereview.appspot.com/929012 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3052 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-25Only remove encoder state feedback for send channels.mflodman@webrtc.org
BUG=1000 TEST=See bug Review URL: https://webrtc-codereview.appspot.com/938004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2994 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-25Revert the revert in r2988 since that wasn't the issue.mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/931005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2992 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-24Reverse Merged r2884 & r2888 from trunk.vikasmarwaha@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2988 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22Move src/ -> webrtc/andrew@webrtc.org
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-26 01:28:38 +0000