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path: root/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
AgeCommit message (Expand)Author
2015-12-22rtcp::Nack packet moved into own file and got Parse functiondanilchap
2015-12-15[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLsdanilchap
2015-12-10lint build/include errors fixed in rtp_rtcp moduledanilchap
2015-12-10Prevent RTCP SR to be sent with bogus timestamp.mflodman
2015-11-04modules: more interface -> include renamesHenrik Kjellander
2015-10-28system_wrappers: rename interface -> includeHenrik Kjellander
2015-10-02Add a PacketOptions struct to webrtc::Transport.stefan
2015-10-02Unify newapi::RtcpMode and RTCPMethod.pbos
2015-09-28Unify Transport and newapi::Transport interfaces.pbos
2015-09-21Make sure rtp_rtcp module doesn't directly reference anything in the pacer mo...sprang
2015-09-17Remove channel ids from various interfaces.Peter Boström
2015-06-08Add support for fragmentation in RtcpPacket.Erik Språng
2015-05-11Refactor RTCP senderErik Språng
2015-03-04Add CVO support to RTP sender side.guoweis@webrtc.org
2015-03-04Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macrokjellander@webrtc.org
2015-02-26Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going awaykwiberg@webrtc.org
2015-02-23Move SetTargetSendBitrates logic from default module to payload router.mflodman@webrtc.org
2015-02-19Add RtcpPacketTypeCounter stats to new API.pbos@webrtc.org
2015-01-29Allow rtp packet history to dynamically expand in size.sprang@webrtc.org
2015-01-22Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.asapersson@webrtc.org
2015-01-13Only return Rtx mode in RTXSendStatus().pbos@webrtc.org
2015-01-12Use int64_t more consistently for times, in particular for RTT values.pkasting@chromium.org
2015-01-09Revert 8028 "Support associated payload type when registering Rt..."andrew@webrtc.org
2015-01-09Support associated payload type when registering Rtx payload type.pbos@webrtc.org
2014-12-19Rtp-Rtcp sender cleanup.pbos@webrtc.org
2014-12-16Add field to counters for when first rtp/rtcp packet is sent/received.asapersson@webrtc.org
2014-12-08Change type of nack_last_time_sent_full_ from uint32_t to int64_t.asapersson@webrtc.org
2014-11-20Use size_t more consistently for packet/payload lengths.pkasting@chromium.org
2014-10-29Add stats for duplicate sent and received NACK requests.asapersson@webrtc.org
2014-10-10Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. xians@webrtc.org
2014-08-04Fix for retransmission. Base layer packets were not retransmitted.asapersson@webrtc.org
2014-07-24Make sure padding is sent on the first sending RTP module.mflodman@webrtc.org
2014-07-17Fix issue where padding is sent before media with undefined timestamps if not...stefan@webrtc.org
2014-07-08Some refactoring inside rtp_rtcp/.pbos@webrtc.org
2014-06-17Add round-robin selection of send stream to pad on.stefan@webrtc.org
2014-06-05Have RTX be enabled by setting an RTX payload type instead of by setting an R...stefan@webrtc.org
2014-03-24VoE changes to allow forwarding of packets from VoE to ViE BWE.solenberg@webrtc.org
2014-02-19Add RTCP packet type counter (for getting statistics such as sent/received NA...asapersson@webrtc.org
2014-01-24Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtc...asapersson@webrtc.org
2013-12-16Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."asapersson@webrtc.org
2013-12-13Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."asapersson@webrtc.org
2013-12-05Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.asapersson@webrtc.org
2013-11-21Added API for enabling/disabling RTCP Receiver Reference Time extension.asapersson@webrtc.org
2013-11-20Add possibility to get the last processed RTT from the call stats class (to b...asapersson@webrtc.org
2013-10-31Propagate estimated RTT from receivers to rtt observer.asapersson@webrtc.org
2013-10-23Fix for incorrect RTT estimation. A too low RTT value could be estimated.asapersson@webrtc.org
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