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video_engine
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vie_rtp_rtcp_impl.cc
Age
Commit message (
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Author
2014-07-11
Cast payload types to int for logging.
pbos@webrtc.org
2014-07-11
Remove the send-side cname getter APIs from voice and video engine.
stefan@webrtc.org
2014-07-07
Preserve RTP states for restarted VideoSendStreams.
pbos@webrtc.org
2014-06-11
Add APIs to enable padding with redundant payloads.
stefan@webrtc.org
2014-05-14
Remove WEBRTC_TRACE uses in video_engine/
pbos@webrtc.org
2014-05-02
Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a chan...
stefan@webrtc.org
2014-03-26
Add API to allow deducting bitrate from incoming estimates before the capacit...
solenberg@webrtc.org
2014-03-24
Implement ViE forwarding to RBE of packets for BWE coming in through the ViEN...
solenberg@webrtc.org
2014-03-13
Implement minimum transmit bitrate.
pbos@webrtc.org
2014-02-27
Adds APIs for reporting pacer queuing delay.
jiayl@webrtc.org
2014-02-19
Add RTCP packet type counter (for getting statistics such as sent/received NA...
asapersson@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
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
2013-12-11
Revert 5259 "Callback for send bitrate estimates"
sprang@webrtc.org
2013-12-11
Callback for send bitrate estimates
sprang@webrtc.org
2013-12-06
Fraction lost statistics not being reported
sprang@webrtc.org
2013-12-05
Add callbacks for send channel rtp statistics
sprang@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
2013-11-21
Added API for enabling/disabling RTCP Receiver Reference Time extension.
asapersson@webrtc.org
2013-11-20
Interface changes to old api, for use by new api transition.
sprang@webrtc.org
2013-08-15
Update talk to 50918584.
wu@webrtc.org
2013-07-16
Revert r4301
tnakamura@webrtc.org
2013-07-05
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
stefan@webrtc.org
2013-05-27
- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelG...
solenberg@webrtc.org
2013-05-23
Adds integration test for RTX and fixes bugs found.
stefan@webrtc.org
2013-05-20
Add functions to ViE API to enable/disable the absolute send time header exte...
solenberg@webrtc.org
2013-05-17
Include files from webrtc/.. paths in video_engine/
pbos@webrtc.org
2013-05-16
Remove SetOverUseDetectorOptions and cleaned ViESharedData.
mflodman@webrtc.org
2013-05-14
Adding a factory to remote bitrate estimator and allow it to be set via config.
andresp@webrtc.org
2013-05-09
Fix compile errors in ViE with latest clang.
andrew@webrtc.org
2013-04-23
Revert "Add a default RTT to CallStats and use different values for buffered/...
stefan@webrtc.org
2013-04-23
Add a default RTT to CallStats and use different values for buffered/real-tim...
stefan@webrtc.org
2013-04-12
Adding a payload type for RTX.
mflodman@webrtc.org
2013-04-09
WebRtc_Word32 -> int32_t in video_engine/
pbos@webrtc.org
2013-02-15
Adding a receive side API for buffering mode.
mikhal@webrtc.org
2013-02-10
Updates to send side streaming mode:
mikhal@webrtc.org
2013-02-06
Don't report an error for GetEstimatedReceiveBandwidth if there is no valid
mflodman@webrtc.org
2013-02-01
Adding a send side API for streaming
mikhal@webrtc.org
2012-10-25
Revert the revert in r2988 since that wasn't the issue.
mflodman@webrtc.org
2012-10-24
Reverse Merged r2884 & r2888 from trunk.
vikasmarwaha@webrtc.org
2012-10-22
Move src/ -> webrtc/
andrew@webrtc.org