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Author
2014-06-10
Add kjellander@webrtc.org as OWNER for *.isolate
kjellander@webrtc.org
2014-06-09
Updated conformance tests and w3c-ified them.
phoglund@webrtc.org
2014-06-06
Make VideoSendStream/VideoReceiveStream configs const.
pbos@webrtc.org
2014-06-05
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
wu@webrtc.org
2014-06-05
Adding back platform specific renderer to video loopback test.
mflodman@webrtc.org
2014-06-04
Android: cleanup gtest_target_type conditions.
henrike@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
2014-05-20
Add NACK and RPSI packet types to RTCP packet builder.
asapersson@webrtc.org
2014-05-19
Add interface to propagate audio capture timestamp to the renderer.
wu@webrtc.org
2014-05-16
Wire up --force_fieldtrials for vie_auto_test and for test targets linking wi...
andresp@webrtc.org
2014-05-14
Add DeliveryStatus enum to DeliverPacket().
pbos@webrtc.org
2014-05-14
Add webrtc field trials API.
andresp@webrtc.org
2014-05-13
Move gflags usage to video_loopback.
pbos@webrtc.org
2014-04-29
Added include of assert.h for files calling assert but missing the include.
henrike@webrtc.org
2014-04-28
Add thread annotations to Call API.
pbos@webrtc.org
2014-04-25
Replace scoped_array<T> with scoped_ptr<T[]>.
andrew@webrtc.org
2014-04-24
Calculate local/remote clock delta and capture ntp timestamp in receiver's ti...
wu@webrtc.org
2014-04-16
Remove use of tmpnam.
kjellander@webrtc.org
2014-04-14
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
fischman@webrtc.org
2014-03-24
VoE changes to allow forwarding of packets from VoE to ViE BWE.
solenberg@webrtc.org
2014-03-19
Remove internal codecs from VideoSendStream.
pbos@webrtc.org
2014-03-13
Implement minimum transmit bitrate.
pbos@webrtc.org
2014-03-12
Remove platform-specific code from new-API tests.
pbos@webrtc.org
2014-03-01
Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed.
fischman@webrtc.org
2014-02-26
Add SetConfig method to FakeNetworkPipe and to DirectTransport
henrik.lundin@webrtc.org
2014-02-23
Disable libjingle_peerconnection_java_unittest
kjellander@webrtc.org
2014-02-21
Add RTCP packet class.
asapersson@webrtc.org
2014-02-18
Remove external encryption API for VoE.
solenberg@webrtc.org
2014-02-18
Incorrect overhead calculation when using FEC + RTP extension headers.
sprang@webrtc.org
2014-01-24
Wire up RTX in VideoReceiveStream.
pbos@webrtc.org
2014-01-13
Removes usage of ListWrapper from several files.
henrike@webrtc.org
2013-12-18
Integrate fake_network_pipe into direct_transport.
stefan@webrtc.org
2013-12-18
Remove metrics_unittests
kjellander@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
2013-12-11
Add SwapFrame() to VideoSendStreamInput.
pbos@webrtc.org
2013-12-11
Roll chromium_revision 232627:238260
kjellander@webrtc.org
2013-12-10
Improve VideoSendStreamTest::MaxPacketSize
sprang@webrtc.org
2013-12-06
Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..."
andrew@webrtc.org
2013-12-05
Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to...
sprang@webrtc.org
2013-12-05
Move implementation files out of the webrtc/ root.
pbos@webrtc.org
2013-12-04
Adds support for sending redundant payloads over RTX.
stefan@webrtc.org
2013-12-03
Set local SSRC for VideoReceiveStream.
pbos@webrtc.org
2013-11-28
Set up SSRCs correctly after switching codec.
pbos@webrtc.org
2013-11-26
Implement and test EncodedImageCallback in new ViE API.
sprang@webrtc.org
2013-11-21
Add -Wnon-virtual-dtor warning for C++ code.
pbos@webrtc.org
2013-11-20
Rename newapi::Transport::SendRTP()->SendRtp().
pbos@webrtc.org
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