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AgeCommit message (Expand)Author
2014-07-11Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.tommi@webrtc.org
2014-07-04Add boilerplate code for H.264.stefan@webrtc.org
2014-06-05Fix the chain that propagates the audio frame's rtp and ntp timestamp including:wu@webrtc.org
2014-05-30Add a Reset() method to AudioFrame.andrew@webrtc.org
2014-05-21Switch to using base/constructormagic.h and remove system_wrappers/interface/...henrike@webrtc.org
2014-05-21Revert 6202 "Switch to using base/constructormagic.h and remove ..."mcasas@webrtc.org
2014-05-20Switch to using base/constructormagic.h and remove system_wrappers/interface/...henrike@webrtc.org
2014-05-19Add interface to propagate audio capture timestamp to the renderer.wu@webrtc.org
2014-05-13The webrtc::AudioFrame struct contains a variable energy_. Since the energy i...henrik.lundin@webrtc.org
2014-04-15Propagate capture ntp timestamp from rtp to renderer.wu@webrtc.org
2014-03-24VoE changes to allow forwarding of packets from VoE to ViE BWE.solenberg@webrtc.org
2014-03-06Help to land 7969005 on behalf of solenberg. The review and try is done in 79...wu@webrtc.org
2013-12-16Revert r5294 to re-roll r5293.pbos@webrtc.org
2013-12-15Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."turaj@webrtc.org
2013-12-13Auto instantiate RBE depending on whether AST or TOF is available in incoming...solenberg@webrtc.org
2013-11-11Remove ".." from include_dirs in build/common.pbos@webrtc.org
2013-10-25Use clang-format -style=chromium to correct the format in webrtc/modules/inte...xians@webrtc.org
2013-10-25Added a "interleaved_" flag to webrtc::AudioFrame.xians@webrtc.org
2013-09-06Adds support for combining RTX and FEC/RED.stefan@webrtc.org
2013-08-15Update talk to 50918584.wu@webrtc.org
2013-08-05Switch C++-style C headers with their C equivalents.pbos@webrtc.org
2013-07-16Revert r4301tnakamura@webrtc.org
2013-07-05Breaking out receive-stats, rtp-payload-registry and rtp-receiver from thestefan@webrtc.org
2013-07-03Proper spacing for end-of-namespace comments.pbos@webrtc.org
2013-05-29Breaking out RTP header parsing from the RTP module.stefan@webrtc.org
2013-05-16Add handling of the absolute send time header extension to the rtp_rtcp module.solenberg@webrtc.org
2013-04-11Fix issues with incorrect wrap checks when having big buffers and high bitrate.stefan@webrtc.org
2013-04-10Removing remaining WebRtc_Word32 not in typedefs.hpbos@webrtc.org
2013-03-12Removed redundant VP8 width/height and made sure the generic width/height is ...stefan@webrtc.org
2013-02-19Rename webrtc::StatsObserver to webrtc::CallStatsObserverfischman@webrtc.org
2013-02-01Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEs...stefan@webrtc.org
2013-01-22Replace AudioFrame's operator= with CopyFrom().andrew@webrtc.org
2012-11-23Remove operator overloading from RTPFragmentationHeader.andrew@webrtc.org
2012-10-22Move src/ -> webrtc/andrew@webrtc.org
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