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2014-03-20
Use codec width/height as the encoded_image width/height.
wu@webrtc.org
2014-03-20
Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets
henrik.lundin@webrtc.org
2014-03-20
Add ability to configure cpu overuse options via an API.
asapersson@webrtc.org
2014-03-20
Prevent playout delay wrap-around in VoiceEngine
henrik.lundin@webrtc.org
2014-03-20
Removes error printout in voe_cmd_test which was caused by attempts to transm...
henrika@webrtc.org
2014-03-20
Extend perf tests to perform rampup on single stream.
andresp@webrtc.org
2014-03-20
Adjust the captured window rect when the window is maximized.
jiayl@webrtc.org
2014-03-19
Properly account for retransmitted packets when not using the pacer.
stefan@webrtc.org
2014-03-19
Fixes RTX related bugs.
stefan@webrtc.org
2014-03-19
Disabling SendsSetSimulcastSsrcs.
pbos@webrtc.org
2014-03-19
Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets"
henrik.lundin@webrtc.org
2014-03-19
Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets
henrik.lundin@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
2014-03-19
Fix a deadlock in ViEEncoder::DeliverFrame.
wuchengli@chromium.org
2014-03-18
Adds a method to WindowCapturer to bring a window to the front.
jiayl@webrtc.org
2014-03-18
Adding thread annotations to NetEq4
henrik.lundin@webrtc.org
2014-03-18
Add #include <cstdlib> for std::abs.
pbos@webrtc.org
2014-03-18
Resolves TSan v2 warnings in voe_auto_test.
henrika@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
Small refactor on send_side_bandwidth_estimation.
andresp@webrtc.org
2014-03-17
Refactor rampup tests:
andresp@webrtc.org
2014-03-17
Tool to establish a loopback call via apprtc turn server.
andresp@webrtc.org
2014-03-14
References to includes in third_party should be relative, not absolute.
sprang@webrtc.org
2014-03-14
Add support for YUV4MPEG file reading to tools files. (Minor fix).
mcasas@webrtc.org
2014-03-14
Add support for YUV4MPEG file reading to tools files.
mcasas@webrtc.org
2014-03-14
Fix a bug where network freeze during CNG causes delay
henrik.lundin@webrtc.org
2014-03-14
Remove legacy weirdness in Merge::Downsample
henrik.lundin@webrtc.org
2014-03-13
Stopping network threads before tearing down test
henrik.lundin@webrtc.org
2014-03-13
Race condition in RTPSender
sprang@webrtc.org
2014-03-13
Add max delay to trace based filters and enhances drop tail queues with delay...
stefan@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
2014-03-13
Disable all protobuf dependent targets when enable_protobuf=0.
andrew@webrtc.org
2014-03-12
Enable VS2013 for Windows compilation by default.
kjellander@webrtc.org
2014-03-12
Remove platform-specific code from new-API tests.
pbos@webrtc.org
2014-03-12
Implement a test for an old corner-case in NetEq
henrik.lundin@webrtc.org
2014-03-12
Developing NetEqImpl unit tests
henrik.lundin@webrtc.org
2014-03-11
Disable TestOpusNewACM on Android.
andrew@webrtc.org
2014-03-11
Revert "Routing SuspendChange to VideoSendStream::Stats"
henrik.lundin@webrtc.org
2014-03-11
Reorder includes in audio_processing_impl_unittest.
andrew@webrtc.org
2014-03-11
Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instea...
braveyao@webrtc.org
2014-03-11
Routing SuspendChange to VideoSendStream::Stats
henrik.lundin@webrtc.org
2014-03-10
Classes and tests for audio an classifier. The class can be used to classify ...
jan.skoglund@webrtc.org
2014-03-10
Add tests and modify tools for new float deinterleaved interface.
andrew@webrtc.org
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