Log - refs/tags/FP2-open-17.01.0 - fp2-dev/platform/external/chromium_org/third_party/webrtc

58b5140 Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
8d08a85 Clean directx_sdk_path as it is already defined in base/common.gypi by andresp@webrtc.org · 10 years ago
89fe3ca Remove the 'webrtc_test_video_render_dependencies' target. by pbos@webrtc.org · 10 years ago
6b97015 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
0ab271b Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
36e363e Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
1c65545 Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
8f32b79 Mark all virtual overrides in the hierarchies of UdpTransportData and by henrikg@webrtc.org · 10 years ago
61576f5 Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h." by henrikg@webrtc.org · 10 years ago
e6fd232 Expose VideoEncoders with webrtc/video_encoder.h. by pbos@webrtc.org · 10 years ago
c00c63f Drop buildbot_tests.py script by kjellander@webrtc.org · 10 years ago
d01c491 Fix audio/video sync when FEC is enabled. by stefan@webrtc.org · 10 years ago
5191730 Partial revert of r7014 (Android APK refactor) by kjellander@webrtc.org · 10 years ago
79426b9 Disable video_engine_tests and webrtc_perf_tests on Android. by kjellander@webrtc.org · 10 years ago
606c1cd Remove build_with_chromium==1 conditions for Android by kjellander@webrtc.org · 10 years ago
b9d6b2b Android APK tests built from a normal WebRTC checkout. by kjellander@webrtc.org · 10 years ago
0e2b7ec Remove Android.mk build files. by pbos@webrtc.org · 10 years ago
7a2cfc5 Remove former team members from OWNERS and WATCHLISTS by kjellander@webrtc.org · 10 years ago
22c283b Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 10 years ago
ac772a4 RTP video playback tool using Call APIs. by pbos@webrtc.org · 10 years ago
6c3f505 Fix crashing fake network pipe tests. by stefan@webrtc.org · 10 years ago
617e272 Add end-to-end H.264 packetization test. by stefan@webrtc.org · 10 years ago
4068313 Add simulation of network effects to video_loopback tool. by stefan@webrtc.org · 10 years ago
408fa71 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. by andresp@webrtc.org · 10 years ago
54f889f Support VP8 encoder settings in VideoSendStream. by pbos@webrtc.org · 10 years ago
79b66f4 Add full stack test cases with a fake network pipe. by stefan@webrtc.org · 10 years ago
6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
7f0b309 Remove GetDefaultConfigs() from Call. by pbos@webrtc.org · 10 years ago
9417f66 Add pbos@webrtc.org as owner for webrtc/test/. by pbos@webrtc.org · 10 years ago
2d4a80c Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
88b558f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
8ade059 Removing W3C conformance tests after move to web-platform-tests. by phoglund@webrtc.org · 10 years ago
eb67a6b Refactor Call-based tests. by pbos@webrtc.org · 10 years ago
7eec1dd Add RTCP packet types to packet builder: by asapersson@webrtc.org · 10 years ago
38a2d46 Updated W3C getusermedia tests to the latest version of the spec. by phoglund@webrtc.org · 10 years ago
847dfa5 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 10 years ago
e82b71d Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 10 years ago
00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
f006e8d Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
f03a4a6 Updated conformance tests and w3c-ified them. by phoglund@webrtc.org · 10 years ago
bdfcddf Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
81f8df9 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
59a001f Adding back platform specific renderer to video loopback test. by mflodman@webrtc.org · 10 years ago
00d9c49 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 10 years ago
774b3d3 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
0a9ed7c Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
28b7c07 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
b5b8648 Add NACK and RPSI packet types to RTCP packet builder. by asapersson@webrtc.org · 10 years ago
22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
bd49ac2 Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} by andresp@webrtc.org · 10 years ago
bc57e0f Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
0b8a1c4 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
11de507 Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
151f6f2 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 10 years ago
c476e64 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
093fc0b Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
1ed7008 Remove use of tmpnam. by kjellander@webrtc.org · 10 years ago
98f8320 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
e2a7a77 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
3f83f9c Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
ee86b90 Remove platform-specific code from new-API tests. by pbos@webrtc.org · 10 years ago
f9747a8 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 10 years ago
9900e37 Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 10 years ago
b1a7102 Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 10 years ago
663ba07 Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 10 years ago
a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
3e4cdec Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 10 years ago
c71929d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
083049f Removes usage of ListWrapper from several files. by henrike@webrtc.org · 10 years ago
aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 10 years ago
a07c56f Remove metrics_unittests by kjellander@webrtc.org · 10 years ago
efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 10 years ago
12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 10 years ago
984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 10 years ago
c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 10 years ago
ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 10 years ago
b589c65 Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 10 years ago
e388f9e Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 10 years ago
532b8f7 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered. by sprang@webrtc.org · 10 years ago
d05597a Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 10 years ago
47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
4b50db1 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
66f4394 Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
2e98d45 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
e1e050e Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
3009c81 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
09f84e5 Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
e028410 Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
4985c7b Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
b4db9c3 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
626d764 Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
9398252 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
8e70108 Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago