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2014-04-09
Update makefiles after merge of Chromium at 262754
Android Chromium Automerger
2014-04-07
Update makefiles after merge of Chromium at 262110
Android Chromium Automerger
2014-04-04
Update makefiles after merge of Chromium at 261622
Android Chromium Automerger
2014-04-01
Update makefiles after merge of Chromium at 260927
Android Chromium Automerger
2014-03-26
Re-submit: rev5775
andresp@webrtc.org
2014-03-26
Add API to allow deducting bitrate from incoming estimates before the capacit...
solenberg@webrtc.org
2014-03-25
Revert 5775 "Modify bitrate controller to update bitrate based o..."
andrew@webrtc.org
2014-03-25
Removing VideoCodecDerived and moving methods inside VideoCodec.
mallinath@webrtc.org
2014-03-25
Updated WebRTC version to 3.51
elham@webrtc.org
2014-03-25
Modify bitrate controller to update bitrate based on process call and not
andresp@webrtc.org
2014-03-25
Adding API for setting bandwidth estimation configurations.
stefan@webrtc.org
2014-03-24
Add configuration for ability to use the encode usage measure for triggering ...
asapersson@webrtc.org
2014-03-24
Implement ViE forwarding to RBE of packets for BWE coming in through the ViEN...
solenberg@webrtc.org
2014-03-24
Have changes to REMB trigger RTCP to be sent immediately.
stefan@webrtc.org
2014-03-24
VoE changes to allow forwarding of packets from VoE to ViE BWE.
solenberg@webrtc.org
2014-03-24
Add AIMD option to BWE API.
stefan@webrtc.org
2014-03-21
Refactor in BitrateController module.
andresp@webrtc.org
2014-03-21
Adding operator== and != methods for CodecInst and VideoCodec structures.
mallinath@webrtc.org
2014-03-20
Add ability to configure cpu overuse options via an API.
asapersson@webrtc.org
2014-03-19
Fixes RTX related bugs.
stefan@webrtc.org
2014-03-19
Simplify pacer interface.
pbos@webrtc.org
2014-03-19
Fix a deadlock in ViEEncoder::DeliverFrame.
wuchengli@chromium.org
2014-03-13
Implement minimum transmit bitrate.
pbos@webrtc.org
2014-03-07
Avoid crash in ViEEncoder::DeRegisterExternalEncoder().
fischman@webrtc.org
2014-03-06
Adding a new ramp-up-down-up test
henrik.lundin@webrtc.org
2014-02-27
Adds APIs for reporting pacer queuing delay.
jiayl@webrtc.org
2014-02-24
Fix to get total number of sent and received rtcp packets.
asapersson@webrtc.org
2014-02-20
Updated WebRTC version to 3.50
elham@webrtc.org
2014-02-20
Modified overuse detection thresholds.
asapersson@webrtc.org
2014-02-19
Add RTCP packet type counter (for getting statistics such as sent/received NA...
asapersson@webrtc.org
2014-02-18
Remove external encryption API for VoE.
solenberg@webrtc.org
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
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