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Commit message (
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Author
2014-06-11
Add APIs to enable padding with redundant payloads.
stefan@webrtc.org
2014-06-06
Make VideoSendStream/VideoReceiveStream configs const.
pbos@webrtc.org
2014-06-05
Adding back platform specific renderer to video loopback test.
mflodman@webrtc.org
2014-06-05
Have RTX be enabled by setting an RTX payload type instead of by setting an R...
stefan@webrtc.org
2014-05-29
Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avo...
wu@webrtc.org
2014-05-26
First incoming packet was not accounted for in receive stats. Changed call or...
asapersson@webrtc.org
2014-05-23
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
pbos@webrtc.org
2014-05-23
Revert "Remove VideoSendStreamInput::PutFrame."
pbos@webrtc.org
2014-05-23
Remove VideoSendStreamInput::PutFrame.
pbos@webrtc.org
2014-05-20
Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky.
stefan@webrtc.org
2014-05-15
Fix Win VideoSendStream::...::ToString() compiles.
pbos@webrtc.org
2014-05-15
Add ToString() to VideoSendStream::Config.
pbos@webrtc.org
2014-05-14
Add DeliveryStatus enum to DeliverPacket().
pbos@webrtc.org
2014-05-14
Re-enable the BitrateEstimatorTest cases for the Call API.
solenberg@webrtc.org
2014-05-13
Move gflags usage to video_loopback.
pbos@webrtc.org
2014-05-07
Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can b...
wu@webrtc.org
2014-05-05
* Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter.
wu@webrtc.org
2014-04-28
Add thread annotations to Call API.
pbos@webrtc.org
2014-04-28
Disable flaky CaptureNtpTimeWithNetworkJitter.
pbos@webrtc.org
2014-04-28
Disabling flaky CanReceiveFec.
pbos@webrtc.org
2014-04-24
Calculate local/remote clock delta and capture ntp timestamp in receiver's ti...
wu@webrtc.org
2014-04-24
Remove TraceCallback use from Call.
pbos@webrtc.org
2014-04-24
Rename Start/Stop in Video{Send,Receive}Streams.
pbos@webrtc.org
2014-04-22
Let A/V sync test use default AudioCoding module
henrik.lundin@webrtc.org
2014-04-14
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
fischman@webrtc.org
2014-04-10
Disable UsesTraceCallback
pbos@webrtc.org
2014-04-08
Implement FEC support in VideoReceiveStream.
pbos@webrtc.org
2014-04-08
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
andresp@webrtc.org
2014-04-07
Clean up traces and logs in RemoteBitrateEstimator.
stefan@webrtc.org
2014-03-26
Re-submit: rev5775
andresp@webrtc.org
2014-03-25
Revert 5775 "Modify bitrate controller to update bitrate based o..."
andrew@webrtc.org
2014-03-25
Change sprintf format string from %zu to %i
henrik.lundin@webrtc.org
2014-03-25
Modify bitrate controller to update bitrate based on process call and not
andresp@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
2014-03-20
Extend perf tests to perform rampup on single stream.
andresp@webrtc.org
2014-03-19
Disabling SendsSetSimulcastSsrcs.
pbos@webrtc.org
2014-03-19
Disable flaky CanSwitchToUseAllSsrcs.
pbos@webrtc.org
2014-03-19
Simplify pacer interface.
pbos@webrtc.org
2014-03-19
Remove internal codecs from VideoSendStream.
pbos@webrtc.org
2014-03-18
Re-comitting r5711: "Fixing a flaky test in video_engine_tests"
henrik.lundin@webrtc.org
2014-03-18
Revert 5711 "Fixing a flaky test in video_engine_tests"
turaj@webrtc.org
2014-03-17
Fixing a flaky test in video_engine_tests
henrik.lundin@webrtc.org
2014-03-17
Refactor rampup tests:
andresp@webrtc.org
2014-03-13
Stopping network threads before tearing down test
henrik.lundin@webrtc.org
2014-03-13
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
henrik.lundin@webrtc.org
2014-03-13
Implement minimum transmit bitrate.
pbos@webrtc.org
2014-03-13
Enable all RampUpTest.UpDownUp* tests
henrik.lundin@webrtc.org
2014-03-13
Replace labs with std::abs.
pbos@webrtc.org
2014-03-12
Remove platform-specific code from new-API tests.
pbos@webrtc.org
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