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Author
2014-07-04
Add boilerplate code for H.264.
stefan@webrtc.org
2014-06-05
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
wu@webrtc.org
2014-05-30
Add a Reset() method to AudioFrame.
andrew@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
2014-05-19
Add interface to propagate audio capture timestamp to the renderer.
wu@webrtc.org
2014-05-13
The webrtc::AudioFrame struct contains a variable energy_. Since the energy i...
henrik.lundin@webrtc.org
2014-04-15
Propagate capture ntp timestamp from rtp to renderer.
wu@webrtc.org
2014-03-24
VoE changes to allow forwarding of packets from VoE to ViE BWE.
solenberg@webrtc.org
2014-03-06
Help to land 7969005 on behalf of solenberg. The review and try is done in 79...
wu@webrtc.org
2013-12-16
Revert r5294 to re-roll r5293.
pbos@webrtc.org
2013-12-15
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
turaj@webrtc.org
2013-12-13
Auto instantiate RBE depending on whether AST or TOF is available in incoming...
solenberg@webrtc.org
2013-11-11
Remove ".." from include_dirs in build/common.
pbos@webrtc.org
2013-10-25
Use clang-format -style=chromium to correct the format in webrtc/modules/inte...
xians@webrtc.org
2013-10-25
Added a "interleaved_" flag to webrtc::AudioFrame.
xians@webrtc.org
2013-09-06
Adds support for combining RTX and FEC/RED.
stefan@webrtc.org
2013-08-15
Update talk to 50918584.
wu@webrtc.org
2013-08-05
Switch C++-style C headers with their C equivalents.
pbos@webrtc.org
2013-07-16
Revert r4301
tnakamura@webrtc.org
2013-07-05
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
stefan@webrtc.org
2013-07-03
Proper spacing for end-of-namespace comments.
pbos@webrtc.org
2013-05-29
Breaking out RTP header parsing from the RTP module.
stefan@webrtc.org
2013-05-16
Add handling of the absolute send time header extension to the rtp_rtcp module.
solenberg@webrtc.org
2013-04-11
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
stefan@webrtc.org
2013-04-10
Removing remaining WebRtc_Word32 not in typedefs.h
pbos@webrtc.org
2013-03-12
Removed redundant VP8 width/height and made sure the generic width/height is ...
stefan@webrtc.org
2013-02-19
Rename webrtc::StatsObserver to webrtc::CallStatsObserver
fischman@webrtc.org
2013-02-01
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEs...
stefan@webrtc.org
2013-01-22
Replace AudioFrame's operator= with CopyFrom().
andrew@webrtc.org
2012-11-23
Remove operator overloading from RTPFragmentationHeader.
andrew@webrtc.org
2012-10-22
Move src/ -> webrtc/
andrew@webrtc.org