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vie_channel.cc
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Commit message (
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Author
2014-11-03
Add stats for video:
asapersson@webrtc.org
2014-09-08
Fix RTT calculations for send-only channels.
stefan@webrtc.org
2014-08-14
Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
stefan@webrtc.org
2014-08-07
Remove the old H264 code now that a new H.264 packetizer has been implemented.
stefan@webrtc.org
2014-07-11
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
stefan@webrtc.org
2014-07-11
Remove the send-side cname getter APIs from voice and video engine.
stefan@webrtc.org
2014-07-10
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module i...
andresp@webrtc.org
2014-07-08
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module...
andresp@webrtc.org
2014-07-07
Preserve RTP states for restarted VideoSendStreams.
pbos@webrtc.org
2014-07-04
Configure RTX send status on new modules.
pbos@webrtc.org
2014-06-30
Reserve RTP/RTCP modules in SetSSRC.
pbos@webrtc.org
2014-06-11
Add APIs to enable padding with redundant payloads.
stefan@webrtc.org
2014-06-05
Have RTX be enabled by setting an RTX payload type instead of by setting an R...
stefan@webrtc.org
2014-05-23
Fix deadlock in RegisterPreDecodeImageCallback.
pbos@webrtc.org
2014-05-02
Pointers were not dereferenced in GetRtpStatistics.
asapersson@webrtc.org
2014-05-02
Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a chan...
stefan@webrtc.org
2014-04-25
Replace scoped_array<T> with scoped_ptr<T[]>.
andrew@webrtc.org
2014-04-15
Propagate capture ntp timestamp from rtp to renderer.
wu@webrtc.org
2014-04-11
Cleaned up logging in video_coding.
stefan@webrtc.org
2014-04-08
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
andresp@webrtc.org
2014-04-07
Log Fixit for parts of video_engine folder.
mflodman@webrtc.org
2014-03-24
Implement ViE forwarding to RBE of packets for BWE coming in through the ViEN...
solenberg@webrtc.org
2014-02-24
Fix to get total number of sent and received rtcp packets.
asapersson@webrtc.org
2014-02-19
Add RTCP packet type counter (for getting statistics such as sent/received NA...
asapersson@webrtc.org
2014-02-11
Remove ViE external encryption API.
solenberg@webrtc.org
2014-02-10
Add stats of incoming frame delays for debugging bandwidth estimation.
jiayl@webrtc.org
2014-01-23
Add callbacks for receive channel RTP statistics
sprang@webrtc.org
2014-01-20
Add configuration and test for extended RTCP reference time reports to new vi...
asapersson@webrtc.org
2014-01-07
Wire up statistics in video send stream of new video engine api
sprang@webrtc.org
2013-12-19
Add callbacks for receive channel RTCP statistics.
sprang@webrtc.org
2013-12-16
Revert r5294 to re-roll r5293.
pbos@webrtc.org
2013-12-15
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
turaj@webrtc.org
2013-12-13
Auto instantiate RBE depending on whether AST or TOF is available in incoming...
solenberg@webrtc.org
2013-12-13
Callback for send bitrate estimates - new roll
sprang@webrtc.org
2013-12-13
Update talk to 58127566 together with
wu@webrtc.org
2013-12-12
Revert 5274 "Update talk to 58113193 together with https://webrt..."
wu@webrtc.org
2013-12-12
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5...
wu@webrtc.org
2013-12-11
Revert 5259 "Callback for send bitrate estimates"
sprang@webrtc.org
2013-12-11
Callback for send bitrate estimates
sprang@webrtc.org
2013-12-05
Add callbacks for send channel rtp statistics
sprang@webrtc.org
2013-12-05
Add API to query video engine for the send-side delay.
stefan@webrtc.org
2013-12-05
Fix bug where fraction_lost is always set to 0 when getting received RTCP sta...
stefan@webrtc.org
2013-12-05
Add callbacks for send channel rtcp statistics
sprang@webrtc.org
2013-12-04
Add send frame rate statistics callback
sprang@webrtc.org
2013-12-04
Adds support for sending redundant payloads over RTX.
stefan@webrtc.org
2013-11-26
Implement and test EncodedImageCallback in new ViE API.
sprang@webrtc.org
2013-11-21
Added API for enabling/disabling RTCP Receiver Reference Time extension.
asapersson@webrtc.org
2013-11-20
Add possibility to get the last processed RTT from the call stats class (to b...
asapersson@webrtc.org
2013-11-06
Sending status fix for module.
asapersson@webrtc.org
2013-11-04
Removed unused code.
asapersson@webrtc.org
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