external/webrtc.git
35 hours ago kjellander... PRESUBMIT: Add linux_msan to default trybots. master
40 hours ago buildbot@webrtc.org(Auto)update libjingle 78430441-> 78445452
44 hours ago buildbot@webrtc.org(Auto)update libjingle 78427027-> 78430441
45 hours ago perkj@webrtc.orgAdd HD support to Android if we detect a hardware video...
2 days ago houssainy@google.comAdding the subtool rtcBot report visualizer
2 days ago pbos@webrtc.orgMove min transmit bitrate to VideoEncoderConfig.
2 days ago pthatcher@webrtc.orgpatch from issue 25469004
2 days ago buildbot@webrtc.org(Auto)update libjingle 78381351-> 78389679
2 days ago buildbot@webrtc.org(Auto)update libjingle 78344087-> 78381351
2 days ago aluebs@webrtc.orgBreak out WebRtcNs_ComputeDdUpdate function in ns_core
2 days ago aluebs@webrtc.orgBreak out WebRtcNs_UpdateNoise function in ns_core
2 days ago aluebs@webrtc.orgBreak out FFT function in ns_core
2 days ago aluebs@webrtc.orgBreak out ComputeSnr function in ns_core
2 days ago houssainy@google.comAdding three video conference bots test
2 days ago houssainy@google.comAdding file from test.webrtc.org domain to be downloaded
2 days ago asapersson... Add macros and APIs for webrtc histograms.
2 days ago buildbot@webrtc.org(Auto)update libjingle 78296920-> 78342456
2 days ago kjellander... Download full Chromium checkouts by default
2 days ago stefan@webrtc.orgAdds support for sending first set of packets at increa...
3 days ago houssainy@google.comUsing the Unused turn configuration in two way test
3 days ago pbos@webrtc.orgLet video_loopback use internal VCM capturers.
3 days ago andrew@webrtc.orgAdd a memcheck exclusion for EndToEndTest.CanSwitchToUs...
3 days ago buildbot@webrtc.org(Auto)update libjingle 78273470-> 78296920
3 days ago glaznev@webrtc.orgMerging Henrik's and Peter's changes for AppRTCDemo
3 days ago houssainy@google.comNOTE: This code review based on the running issue:
3 days ago houssainy@google.comAdding Two way video and audio streaming test to RtcBot
3 days ago houssainy@google.comHTTPS Server used instead of HTTP for loading the bots...
3 days ago buildbot@webrtc.org(Auto)update libjingle 78262388-> 78262615
3 days ago pbos@webrtc.orgRemove some disabled tests in WebRtcVideoEngine2.
3 days ago kjellander... Suppress libyuv uninitialized read in CopyRow_AVX
3 days ago pbos@webrtc.orgMake ReconfigureVideoEncoder use current bitrate.
3 days ago kjellander... Tighten up MSan blacklist.txt owners.
3 days ago pbos@webrtc.orgDisable TestVp8Impl.BaseUnitTest on MSan.
3 days ago stefan@webrtc.orgFor FIR packet, payload length is zero, so SendToNetwor...
4 days ago kjellander... Roll chromium_revision de13cf4..28d1981 (299488:300483)
4 days ago aluebs@webrtc.orgBreak out WebRtcNs_Windowing function in ns_core
4 days ago aluebs@webrtc.orgBreak out WebRtcNs_Energy function in ns_core
4 days ago aluebs@webrtc.orgBreak out WebRtcNs_IFFT function in ns_core
4 days ago buildbot@webrtc.org(Auto)update libjingle 78193292-> 78199328
4 days ago guoweis@webrtc.orgFix local address leakage when IceTransportsType is...
4 days ago aluebs@webrtc.orgBreak out WebRtcNs_UpdateBuffer function in ns_core
4 days ago buildbot@webrtc.org(Auto)update libjingle 78106439-> 78193292
4 days ago henrik.lundin... Implement AudioEncoderPcmU/A classes and convert AudioD...
5 days ago bjornv@webrtc.orgaudio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSH...
5 days ago henrik.lundin... Fix for glitches in ACM when switching desired output...
5 days ago glaznev@webrtc.orgAvoid using EGLContext class for Android 4.1 and below.
5 days ago bjornv@webrtc.orgcommon_audio: Replaced invalid operand in min_max_opera...
5 days ago pbos@webrtc.orgSet up start bitrate in WebRtcVideoEngine2.
6 days ago pbos@webrtc.orgMake avg_{psnr,ssim}_threshold_ const.
6 days ago bjornv@webrtc.orgaudio_coding/codecs/isac/main: Replaced macro WEBRTC_SP...
6 days ago bjornv@webrtc.orgaudio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W3...
8 days ago henrike@webrtc.orgReverts r7459 "Create a copy of talk/xmpp and talk...
8 days ago buildbot@webrtc.org(Auto)update libjingle 77953038-> 77970462
8 days ago henrike@webrtc.orgRevert cls (original cl + fixes) 7422-7424 "Add VP9...
8 days ago glaznev@webrtc.orgCleaning up Android AppRTCDemo.
8 days ago houssainy@google.comMoving creating TURN configration to the host machine...
8 days ago glaznev@webrtc.orgQuery Android device orientation on every camera frame...
8 days ago henrike@webrtc.orgrtc_unittest: copied gtest excludes from libjingle_p2p_...
9 days ago houssainy@google.comTest names changed from e.g) testOneWayVideo/chrome...
9 days ago henrik.lundin... Add encoded_timestamp to AudioEncoder base class
9 days ago henrik.lundin... New interface class AudioEncoder
9 days ago stefan@webrtc.orgDisable a bunch of Nat and Ice tests when running under...
10 days ago andresp@webrtc.orgImprove rtcbot to load all test files at start and...
10 days ago asapersson... Add ability to include a larger time span (in addition...
10 days ago henrike@webrtc.orgCreate a copy of talk/xmpp and talk/p2p under webrtc...
10 days ago houssainy@google.comSelecting bot_type changed to be specified in the test...
10 days ago pbos@webrtc.orgFix data races in ThreadTest.ThreeThreadsInvoke.
10 days ago bjornv@webrtc.orgaudio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32...
10 days ago bjornv@webrtc.orgaudio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_...
10 days ago bjornv@webrtc.orgaudio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W...
11 days ago henrik.lundin... Extend AcmSwitchingOutputFrequencyOldApi with more...
11 days ago kjellander... Roll chromium_revision 2d714fa..de13cf4 (298667:299488)
11 days ago bjornv@webrtc.orgcommon_audio: Removed version API from signal_processing
11 days ago buildbot@webrtc.org(Auto)update libjingle 77701902-> 77709729
11 days ago buildbot@webrtc.org(Auto)update libjingle 77689511-> 77696841
11 days ago pbos@webrtc.orgRemove unused (no-op) VideoOptions.
11 days ago henrike@webrtc.orglibjingle: use _stricmp instead of deprecated stricmp.
11 days ago pbos@webrtc.orgRemove -1 from Call::Config::start_bitrate_bps.
11 days ago stefan@webrtc.orgAdd periodic logging of received RTP headers and estima...
11 days ago henrik.lundin... New ACM test to trigger audio glitch when switching...
11 days ago stefan@webrtc.orgAdd a packet loss full stack test to the new API.
11 days ago kwiberg@webrtc.orgWorkarounds for a bug in VS2013.3 linker when PGO is...
12 days ago pbos@webrtc.orgWire up external encoders.
12 days ago buildbot@webrtc.org(Auto)update libjingle 77554188-> 77629208
12 days ago marpan@webrtc.orgMove exlusion of VP9 integration tests for DrMemory
12 days ago aluebs@webrtc.orgAdjust speech probability in NS when echo
12 days ago henrike@webrtc.orgRemoves xmllite from talk/xmllite since webrtc/xmllite...
12 days ago marpan@webrtc.orgDisable VP9 integration tests on DrMemory.
12 days ago bjornv@webrtc.orgcommon_audio: Removed macro WEBRTC_SPL_RSHIFT_W16
12 days ago kwiberg@webrtc.orgiSAC tests: Type buffers as uint8_t[] to avoid casts
12 days ago bjornv@webrtc.orgaudio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16...
12 days ago kwiberg@webrtc.orgWebRtcIsac_Decode et al.: Type encoded data as uint8...
12 days ago kwiberg@webrtc.orgWebRtcIsac_UpdateBwEstimate et al.: Type byte streams...
12 days ago kwiberg@webrtc.orgSome WebRtcIsac_* and WebRtcIsacfix_* functions: type...
13 days ago buildbot@webrtc.org(Auto)update libjingle 77414393-> 77554188
13 days ago braveyao@webrtc.orgMerge the supporting to UYVY on Linux video capture...
13 days ago braveyao@webrtc.orgRelease _inputSendPin & _outputCapturePin before _captu...
2014-10-10 henrike@webrtc.orgRe-enable ThreadCheckerDeathTest.MethodNotAllowedOnDiff...
2014-10-10 marpan@webrtc.orgDisable SendsAndReceivesVP9 test for now.
2014-10-10 marpan@webrtc.orgAdjust/increase rate control thresold for a vp9 test.
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