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rtp_rtcp
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Author
2014-07-17
Fix issue where padding is sent before media with undefined timestamps if not...
stefan@webrtc.org
2014-07-16
Remove old padding path in RTPSender.
pbos@webrtc.org
2014-07-15
Fix breakage introduced by r6691.
pbos@webrtc.org
2014-07-15
Make RTCP sender report send media bytes.
pbos@webrtc.org
2014-07-11
Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .
tommi@webrtc.org
2014-07-11
Remove always-true expression.
tommi@webrtc.org
2014-07-11
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
tommi@webrtc.org
2014-07-11
Thread annotate RTCPSender.
pbos@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
Count total bytes sent in RTPSender::Bytes().
pbos@webrtc.org
2014-07-10
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module i...
andresp@webrtc.org
2014-07-10
Revert "Remove remains of WEBRTC_NO_STL." (rev 6641).
andresp@webrtc.org
2014-07-09
Remove remains of WEBRTC_NO_STL.
andresp@webrtc.org
2014-07-08
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module...
andresp@webrtc.org
2014-07-08
Some refactoring inside rtp_rtcp/.
pbos@webrtc.org
2014-07-07
Preserve RTP states for restarted VideoSendStreams.
pbos@webrtc.org
2014-07-04
Add boilerplate code for H.264.
stefan@webrtc.org
2014-07-04
Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.
stefan@webrtc.org
2014-07-03
Fix memcheck error in r6594.
marpan@webrtc.org
2014-07-03
Fix for FEC decoding with sequence number wrap-around.
marpan@webrtc.org
2014-06-26
Fixes a bug causing NACKs to be dropped excessively at the send-side.
stefan@webrtc.org
2014-06-25
fix after r6472 in rtp_sender, comparison between signed and unsigned integer...
henrike@webrtc.org
2014-06-25
Add RTCP packet types to packet builder:
asapersson@webrtc.org
2014-06-23
GN: Add BUILD.gn files + kjellander to OWNERS
kjellander@webrtc.org
2014-06-17
Add round-robin selection of send stream to pad on.
stefan@webrtc.org
2014-06-16
Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.
asapersson@webrtc.org
2014-06-05
Have RTX be enabled by setting an RTX payload type instead of by setting an R...
stefan@webrtc.org
2014-06-04
Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE.
andresp@webrtc.org
2014-05-30
Fix bug where RTP headers in the packet history were replaced with the RTX wr...
stefan@webrtc.org
2014-05-21
Switch to using base/constructormagic.h and remove system_wrappers/interface/...
henrike@webrtc.org
2014-05-21
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
mcasas@webrtc.org
2014-05-20
Switch to using base/constructormagic.h and remove system_wrappers/interface/...
henrike@webrtc.org
2014-05-20
Add NACK and RPSI packet types to RTCP packet builder.
asapersson@webrtc.org
2014-05-14
Move the capture ntp computing code to ntp_calculator so that later it can be...
wu@webrtc.org
2014-04-25
Replace scoped_array<T> with scoped_ptr<T[]>.
andrew@webrtc.org
2014-04-24
Calculate local/remote clock delta and capture ntp timestamp in receiver's ti...
wu@webrtc.org
2014-04-24
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
wu@webrtc.org
2014-04-15
Replace flooding logs in rtp_sender.cc with a comment.
andrew@webrtc.org
2014-04-15
Check if a header extension is registered before updating it and fail silentl...
stefan@webrtc.org
2014-04-14
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
fischman@webrtc.org
2014-04-08
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
andresp@webrtc.org
2014-03-26
Protect write of send_target_bitrate.
andresp@webrtc.org
2014-03-25
Make RTPHeaderParser skip over unknown RTP header extensions rather than bail...
solenberg@webrtc.org
2014-03-25
Fix race condition in RTPSEnder.
sprang@webrtc.org
2014-03-24
Have changes to REMB trigger RTCP to be sent immediately.
stefan@webrtc.org
2014-03-24
VoE changes to allow forwarding of packets from VoE to ViE BWE.
solenberg@webrtc.org
2014-03-24
Add AIMD option to BWE API.
stefan@webrtc.org
2014-03-19
Properly account for retransmitted packets when not using the pacer.
stefan@webrtc.org
2014-03-19
Fixes RTX related bugs.
stefan@webrtc.org
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