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rtp_rtcp
Age
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Author
2014-11-04
Reworked paced sender queue
sprang@webrtc.org
2014-10-29
Add stats for duplicate sent and received NACK requests.
asapersson@webrtc.org
2014-10-22
For FIR packet, payload length is zero, so SendToNetwork function is failing.
stefan@webrtc.org
2014-10-14
Add periodic logging of received RTP headers and estimated clock offsets for ...
stefan@webrtc.org
2014-10-10
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
xians@webrtc.org
2014-10-09
Estimating NTP time with a given RTT.
minyue@webrtc.org
2014-09-29
Remove callback from RtpDepacketizer::Parse().
pbos@webrtc.org
2014-09-28
GN: Add common configs to all targets.
kjellander@webrtc.org
2014-09-25
Fix typo from RtpPacketizerH264.
pbos@webrtc.org
2014-09-24
Move thread_annotations.h to webrtc/base/.
pbos@webrtc.org
2014-09-17
Refactor VP8 de-packetizer.
stefan@webrtc.org
2014-09-16
Mark all virtual overrides in the hierarchies of UdpTransportData and
henrikg@webrtc.org
2014-09-12
Initialize restored_packet in nack_rtx_unittest.cc.
pbos@webrtc.org
2014-09-12
Make RTPSender/RTPReceiver generic.
pbos@webrtc.org
2014-09-12
Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
stefan@webrtc.org
2014-09-11
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
henrik.lundin@webrtc.org
2014-09-08
Ignore FEC packet in stats, if it is first packet on ssrc.
sprang@webrtc.org
2014-09-03
Setting marker bit on DTMF correctly
stefan@webrtc.org
2014-08-28
Add unit tests to rtcp_receiver_test.
asapersson@webrtc.org
2014-08-27
Expose setPayloadType on the rtp_sender. Thus allowing other users of this mo...
andresp@webrtc.org
2014-08-26
Remove Android.mk build files.
pbos@webrtc.org
2014-08-26
Remove former team members from OWNERS and WATCHLISTS
kjellander@webrtc.org
2014-08-25
GN: Disable Chromium clang plugins for standalone build.
kjellander@webrtc.org
2014-08-14
Small refactor on ViE to remove redudant conditions and long ifdefs.
andresp@webrtc.org
2014-08-14
Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
stefan@webrtc.org
2014-08-14
Fix TimeToSendPadding return to be 0 if no padding bytes are sent.
andresp@webrtc.org
2014-08-13
Fix STAP-A bug where we might overflow the packet buffer due to not accountin...
stefan@webrtc.org
2014-08-12
Make a int64 constant use ULL suffix so it wont get truncated.
fbarchard@google.com
2014-08-07
Remove the old H264 code now that a new H.264 packetizer has been implemented.
stefan@webrtc.org
2014-08-07
Fix single nalu packetization bug.
stefan@webrtc.org
2014-08-06
Change the way we reference enumerators in H.264 packetization code to be sta...
stefan@webrtc.org
2014-08-04
Fix for retransmission. Base layer packets were not retransmitted.
asapersson@webrtc.org
2014-07-31
Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804.
stefan@webrtc.org
2014-07-31
Add H.264 packetization.
stefan@webrtc.org
2014-07-29
Integrate rtcp packet class to rtcp receiver tests.
asapersson@webrtc.org
2014-07-24
Make sure padding is sent on the first sending RTP module.
mflodman@webrtc.org
2014-07-22
Remove remains of WEBRTC_NO_STL.
andresp@webrtc.org
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
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