external/webrtc.git
64 min ago houssainy@google.comAdding three video conference bots test master
2 hours ago houssainy@google.comAdding file from test.webrtc.org domain to be downloaded
4 hours ago asapersson... Add macros and APIs for webrtc histograms.
5 hours ago buildbot@webrtc.org(Auto)update libjingle 78296920-> 78342456
5 hours ago kjellander... Download full Chromium checkouts by default
5 hours ago stefan@webrtc.orgAdds support for sending first set of packets at increa...
9 hours ago houssainy@google.comUsing the Unused turn configuration in two way test
9 hours ago pbos@webrtc.orgLet video_loopback use internal VCM capturers.
12 hours ago andrew@webrtc.orgAdd a memcheck exclusion for EndToEndTest.CanSwitchToUs...
19 hours ago buildbot@webrtc.org(Auto)update libjingle 78273470-> 78296920
24 hours ago glaznev@webrtc.orgMerging Henrik's and Peter's changes for AppRTCDemo
24 hours ago houssainy@google.comNOTE: This code review based on the running issue:
24 hours ago houssainy@google.comAdding Two way video and audio streaming test to RtcBot
25 hours ago houssainy@google.comHTTPS Server used instead of HTTP for loading the bots...
26 hours ago buildbot@webrtc.org(Auto)update libjingle 78262388-> 78262615
26 hours ago pbos@webrtc.orgRemove some disabled tests in WebRtcVideoEngine2.
27 hours ago kjellander... Suppress libyuv uninitialized read in CopyRow_AVX
29 hours ago pbos@webrtc.orgMake ReconfigureVideoEncoder use current bitrate.
30 hours ago kjellander... Tighten up MSan blacklist.txt owners.
31 hours ago pbos@webrtc.orgDisable TestVp8Impl.BaseUnitTest on MSan.
32 hours ago stefan@webrtc.orgFor FIR packet, payload length is zero, so SendToNetwor...
35 hours ago kjellander... Roll chromium_revision de13cf4..28d1981 (299488:300483)
43 hours ago aluebs@webrtc.orgBreak out WebRtcNs_Windowing function in ns_core
43 hours ago aluebs@webrtc.orgBreak out WebRtcNs_Energy function in ns_core
44 hours ago aluebs@webrtc.orgBreak out WebRtcNs_IFFT function in ns_core
45 hours ago buildbot@webrtc.org(Auto)update libjingle 78193292-> 78199328
45 hours ago guoweis@webrtc.orgFix local address leakage when IceTransportsType is...
45 hours ago aluebs@webrtc.orgBreak out WebRtcNs_UpdateBuffer function in ns_core
46 hours ago buildbot@webrtc.org(Auto)update libjingle 78106439-> 78193292
2 days ago henrik.lundin... Implement AudioEncoderPcmU/A classes and convert AudioD...
2 days ago bjornv@webrtc.orgaudio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSH...
2 days ago henrik.lundin... Fix for glitches in ACM when switching desired output...
2 days ago glaznev@webrtc.orgAvoid using EGLContext class for Android 4.1 and below.
3 days ago bjornv@webrtc.orgcommon_audio: Replaced invalid operand in min_max_opera...
3 days ago pbos@webrtc.orgSet up start bitrate in WebRtcVideoEngine2.
3 days ago pbos@webrtc.orgMake avg_{psnr,ssim}_threshold_ const.
3 days ago bjornv@webrtc.orgaudio_coding/codecs/isac/main: Replaced macro WEBRTC_SP...
3 days ago bjornv@webrtc.orgaudio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W3...
5 days ago henrike@webrtc.orgReverts r7459 "Create a copy of talk/xmpp and talk...
5 days ago buildbot@webrtc.org(Auto)update libjingle 77953038-> 77970462
5 days ago henrike@webrtc.orgRevert cls (original cl + fixes) 7422-7424 "Add VP9...
6 days ago glaznev@webrtc.orgCleaning up Android AppRTCDemo.
6 days ago houssainy@google.comMoving creating TURN configration to the host machine...
6 days ago glaznev@webrtc.orgQuery Android device orientation on every camera frame...
6 days ago henrike@webrtc.orgrtc_unittest: copied gtest excludes from libjingle_p2p_...
6 days ago houssainy@google.comTest names changed from e.g) testOneWayVideo/chrome...
6 days ago henrik.lundin... Add encoded_timestamp to AudioEncoder base class
7 days ago henrik.lundin... New interface class AudioEncoder
7 days ago stefan@webrtc.orgDisable a bunch of Nat and Ice tests when running under...
7 days ago andresp@webrtc.orgImprove rtcbot to load all test files at start and...
7 days ago asapersson... Add ability to include a larger time span (in addition...
8 days ago henrike@webrtc.orgCreate a copy of talk/xmpp and talk/p2p under webrtc...
8 days ago houssainy@google.comSelecting bot_type changed to be specified in the test...
8 days ago pbos@webrtc.orgFix data races in ThreadTest.ThreeThreadsInvoke.
8 days ago bjornv@webrtc.orgaudio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32...
8 days ago bjornv@webrtc.orgaudio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_...
8 days ago bjornv@webrtc.orgaudio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W...
8 days ago henrik.lundin... Extend AcmSwitchingOutputFrequencyOldApi with more...
8 days ago kjellander... Roll chromium_revision 2d714fa..de13cf4 (298667:299488)
8 days ago bjornv@webrtc.orgcommon_audio: Removed version API from signal_processing
8 days ago buildbot@webrtc.org(Auto)update libjingle 77701902-> 77709729
8 days ago buildbot@webrtc.org(Auto)update libjingle 77689511-> 77696841
8 days ago pbos@webrtc.orgRemove unused (no-op) VideoOptions.
9 days ago henrike@webrtc.orglibjingle: use _stricmp instead of deprecated stricmp.
9 days ago pbos@webrtc.orgRemove -1 from Call::Config::start_bitrate_bps.
9 days ago stefan@webrtc.orgAdd periodic logging of received RTP headers and estima...
9 days ago henrik.lundin... New ACM test to trigger audio glitch when switching...
9 days ago stefan@webrtc.orgAdd a packet loss full stack test to the new API.
9 days ago kwiberg@webrtc.orgWorkarounds for a bug in VS2013.3 linker when PGO is...
9 days ago pbos@webrtc.orgWire up external encoders.
9 days ago buildbot@webrtc.org(Auto)update libjingle 77554188-> 77629208
9 days ago marpan@webrtc.orgMove exlusion of VP9 integration tests for DrMemory
9 days ago aluebs@webrtc.orgAdjust speech probability in NS when echo
9 days ago henrike@webrtc.orgRemoves xmllite from talk/xmllite since webrtc/xmllite...
10 days ago marpan@webrtc.orgDisable VP9 integration tests on DrMemory.
10 days ago bjornv@webrtc.orgcommon_audio: Removed macro WEBRTC_SPL_RSHIFT_W16
10 days ago kwiberg@webrtc.orgiSAC tests: Type buffers as uint8_t[] to avoid casts
10 days ago bjornv@webrtc.orgaudio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16...
10 days ago kwiberg@webrtc.orgWebRtcIsac_Decode et al.: Type encoded data as uint8...
10 days ago kwiberg@webrtc.orgWebRtcIsac_UpdateBwEstimate et al.: Type byte streams...
10 days ago kwiberg@webrtc.orgSome WebRtcIsac_* and WebRtcIsacfix_* functions: type...
10 days ago buildbot@webrtc.org(Auto)update libjingle 77414393-> 77554188
10 days ago braveyao@webrtc.orgMerge the supporting to UYVY on Linux video capture...
10 days ago braveyao@webrtc.orgRelease _inputSendPin & _outputCapturePin before _captu...
12 days ago henrike@webrtc.orgRe-enable ThreadCheckerDeathTest.MethodNotAllowedOnDiff...
12 days ago marpan@webrtc.orgDisable SendsAndReceivesVP9 test for now.
12 days ago marpan@webrtc.orgAdjust/increase rate control thresold for a vp9 test.
13 days ago marpan@webrtc.orgAdd VP9 codec to VCM and vie_auto_test.
13 days ago xians@webrtc.orgMark all virtual overrides in the hierarchy of Transpor...
13 days ago kjellander... Cleanup scripts and suppressions for TSan v1
13 days ago pbos@webrtc.orgRemove talk_base from suppressions.
13 days ago xians@webrtc.orgReland 28629004: adding new AEC dump start interface...
13 days ago kjellander... Workaround deps2git issue with inline Python in DEPS.
13 days ago henrike@webrtc.orgRe-enable allmost all base tests.
13 days ago henrike@webrtc.orgRe-enables a bunch of base unittests part II.
13 days ago glaznev@webrtc.orgChange setting VP8 codec specific info values by HW...
2014-10-09 henrike@webrtc.orgbase/thread_unittest: wrap test was setting current...
2014-10-09 henrike@webrtc.orgMake pbos and kjellander only owners of tsan2 suppressions.
2014-10-09 henrik.lundin... Fix comments in common_types.h
2014-10-09 pbos@webrtc.orgIncrease timeout for AsyncWriteTest.TestWrite.
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