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Author
2014-02-26
Update makefiles after merge of Chromium at 253634
Android Chromium Automerger
2014-02-25
Update makefiles after merge of Chromium at 253112
Android Chromium Automerger
2014-02-17
Reset estimate if no frame has been seen for a certain time (to avoid large j...
asapersson@webrtc.org
2014-02-14
Adding a critical section missing in r5543.
stefan@webrtc.org
2014-02-13
Increase overuse and normal use thresholds for Mac.
asapersson@webrtc.org
2014-02-13
Fixes a race when writing to send_padding_.
stefan@webrtc.org
2014-02-12
Set pacing bitrates in SetEncoder.
pbos@webrtc.org
2014-02-11
Remove ViE external encryption API.
solenberg@webrtc.org
2014-02-10
Add stats of incoming frame delays for debugging bandwidth estimation.
jiayl@webrtc.org
2014-01-29
Connect webrtc::Config to WrappingBitrateEstimator
henrik.lundin@webrtc.org
2014-01-29
Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."
mallinath@webrtc.org
2014-01-27
Revert 5421 "Fix deadlock on register/unregister observer while ..."
mallinath@webrtc.org
2014-01-23
Fix deadlock on register/unregister observer while there is a an going callback.
andresp@webrtc.org
2014-01-23
Add callbacks for receive channel RTP statistics
sprang@webrtc.org
2014-01-20
Add configuration and test for extended RTCP reference time reports to new vi...
asapersson@webrtc.org
2014-01-14
Roll Chromium 238260 -> 243863
wjia@webrtc.org
2014-01-13
Updated Webrtc version to 3.49
elham@webrtc.org
2014-01-13
Removes usage of ListWrapper from several files.
henrike@webrtc.org
2014-01-07
Wire up statistics in video send stream of new video engine api
sprang@webrtc.org
2013-12-20
Add thread_annotations for clang targets.
andresp@webrtc.org
2013-12-20
If the configured start bitrate is higher than the configures max
mflodman@webrtc.org
2013-12-20
Race condition in ViECapturer::RegisterObserver
sprang@webrtc.org
2013-12-19
Update WebRTC to version 3.48
tnakamura@webrtc.org
2013-12-19
Add callbacks for receive channel RTCP statistics.
sprang@webrtc.org
2013-12-18
Integrate fake_network_pipe into direct_transport.
stefan@webrtc.org
2013-12-18
Remove media_file from VideoEngine dependencies.
pbos@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
2013-12-13
Callback for send bitrate estimates - new roll
sprang@webrtc.org
2013-12-13
Make sure channels in the same call are in the same channel group.
mflodman@webrtc.org
2013-12-13
Making RemoteRateControl::min_configured_bit_rate_ configurable
henrik.lundin@webrtc.org
2013-12-13
Update talk to 58127566 together with
wu@webrtc.org
2013-12-12
Revert 5274 "Update talk to 58113193 together with https://webrt..."
wu@webrtc.org
2013-12-12
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5...
wu@webrtc.org
2013-12-12
Complete rewrite of demo application.
henrike@webrtc.org
2013-12-12
Remove overloaded CpuOveruseMeasure function.
asapersson@webrtc.org
2013-12-11
Add SwapFrame() to VideoSendStreamInput.
pbos@webrtc.org
2013-12-11
Revert 5259 "Callback for send bitrate estimates"
sprang@webrtc.org
2013-12-11
Roll chromium_revision 232627:238260
kjellander@webrtc.org
2013-12-11
Callback for send bitrate estimates
sprang@webrtc.org
2013-12-06
Fraction lost statistics not being reported
sprang@webrtc.org
2013-12-05
Add callbacks for send channel rtp statistics
sprang@webrtc.org
2013-12-05
Add API to query video engine for the send-side delay.
stefan@webrtc.org
2013-12-05
Fixing the android build
henrik.lundin@webrtc.org
2013-12-05
Remove default implementations for SuspendBelowMinBitrate
henrik.lundin@webrtc.org
2013-12-05
Fix bug where fraction_lost is always set to 0 when getting received RTCP sta...
stefan@webrtc.org
2013-12-05
Add callbacks for send channel rtcp statistics
sprang@webrtc.org
2013-12-04
Remove the long disabled WEBRTC_SVNREVISION define.
andrew@webrtc.org
2013-12-04
Removing DropDeltaAfterKey functionality which is unused.
andresp@webrtc.org
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