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rtp_rtcp
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rtp_sender_unittest.cc
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Commit message (
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Author
2014-11-05
Wire up bandwidth stats to the new API and webrtcvideoengine2.
stefan@webrtc.org
2014-10-10
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
xians@webrtc.org
2014-09-03
Setting marker bit on DTMF correctly
stefan@webrtc.org
2014-07-15
Make RTCP sender report send media bytes.
pbos@webrtc.org
2014-07-11
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
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
2014-07-08
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module...
andresp@webrtc.org
2014-07-08
Some refactoring inside rtp_rtcp/.
pbos@webrtc.org
2014-06-17
Add round-robin selection of send stream to pad on.
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
2014-03-06
Help to land 7969005 on behalf of solenberg. The review and try is done in 79...
wu@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
Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks
sprang@webrtc.org
2013-12-05
Add callbacks for send channel rtp 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
2013-11-15
Increment RTP timestamps for padding packets
henrik.lundin@webrtc.org
2013-11-13
Fix for RTX in combination with pacing.
stefan@webrtc.org
2013-09-06
Adds support for combining RTX and FEC/RED.
stefan@webrtc.org
2013-06-19
Fixes some pacer/padding issues found while testing.
stefan@webrtc.org
2013-06-17
Wire up pacer-based padding.
stefan@webrtc.org
2013-06-03
Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disab...
wu@webrtc.org
2013-05-29
Include files from webrtc/.. paths in rtp_rtcp/
pbos@webrtc.org
2013-05-29
Breaking out RTP header parsing from the RTP module.
stefan@webrtc.org
2013-05-23
Fix regression where retransmission bitrate is no longer estimated.
stefan@webrtc.org
2013-05-20
Fix assertions in rtp_header_extension.h caused by not handling the AudioLeve...
solenberg@webrtc.org
2013-05-16
Add handling of the absolute send time header extension to the rtp_rtcp module.
solenberg@webrtc.org
2013-05-07
Fix clang errors in non-GYP_DEFINES=clang=1 build
pbos@webrtc.org
2013-04-27
Enable Nack pacing.
pwestin@webrtc.org
2013-04-08
WebRtc_Word32 => int32_t for rtp_rtcp/
pbos@webrtc.org
2013-03-18
Generic video-codec support.
pbos@webrtc.org
2013-01-21
Move video_coding to new Clock interface and remove fake clock implementation...
stefan@webrtc.org
2013-01-17
Break out RtpClock to system_wrappers and make it more generic.
stefan@webrtc.org
2012-11-13
Enable paced sender.
pwestin@webrtc.org
2012-11-07
pre-factor cleanup pre-work.
pwestin@webrtc.org
2012-10-22
Move src/ -> webrtc/
andrew@webrtc.org