external/webrtc.git
7 hours ago buildbot@webrtc.org(Auto)update libjingle 78273470-> 78296920 master
11 hours ago glaznev@webrtc.orgMerging Henrik's and Peter's changes for AppRTCDemo
11 hours ago houssainy@google.comNOTE: This code review based on the running issue:
11 hours ago houssainy@google.comAdding Two way video and audio streaming test to RtcBot
12 hours ago houssainy@google.comHTTPS Server used instead of HTTP for loading the bots...
13 hours ago buildbot@webrtc.org(Auto)update libjingle 78262388-> 78262615
13 hours ago pbos@webrtc.orgRemove some disabled tests in WebRtcVideoEngine2.
15 hours ago kjellander... Suppress libyuv uninitialized read in CopyRow_AVX
16 hours ago pbos@webrtc.orgMake ReconfigureVideoEncoder use current bitrate.
17 hours ago kjellander... Tighten up MSan blacklist.txt owners.
18 hours ago pbos@webrtc.orgDisable TestVp8Impl.BaseUnitTest on MSan.
19 hours ago stefan@webrtc.orgFor FIR packet, payload length is zero, so SendToNetwor...
22 hours ago kjellander... Roll chromium_revision de13cf4..28d1981 (299488:300483)
30 hours ago aluebs@webrtc.orgBreak out WebRtcNs_Windowing function in ns_core
30 hours ago aluebs@webrtc.orgBreak out WebRtcNs_Energy function in ns_core
31 hours ago aluebs@webrtc.orgBreak out WebRtcNs_IFFT function in ns_core
32 hours ago buildbot@webrtc.org(Auto)update libjingle 78193292-> 78199328
32 hours ago guoweis@webrtc.orgFix local address leakage when IceTransportsType is...
32 hours ago aluebs@webrtc.orgBreak out WebRtcNs_UpdateBuffer function in ns_core
33 hours ago buildbot@webrtc.org(Auto)update libjingle 78106439-> 78193292
40 hours ago henrik.lundin... Implement AudioEncoderPcmU/A classes and convert AudioD...
45 hours ago bjornv@webrtc.orgaudio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSH...
46 hours 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.
2 days ago bjornv@webrtc.orgcommon_audio: Replaced invalid operand in min_max_opera...
2 days ago pbos@webrtc.orgSet up start bitrate in WebRtcVideoEngine2.
2 days ago pbos@webrtc.orgMake avg_{psnr,ssim}_threshold_ const.
2 days ago bjornv@webrtc.orgaudio_coding/codecs/isac/main: Replaced macro WEBRTC_SP...
2 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...
5 days ago glaznev@webrtc.orgCleaning up Android AppRTCDemo.
5 days ago houssainy@google.comMoving creating TURN configration to the host machine...
5 days ago glaznev@webrtc.orgQuery Android device orientation on every camera frame...
5 days ago henrike@webrtc.orgrtc_unittest: copied gtest excludes from libjingle_p2p_...
5 days ago houssainy@google.comTest names changed from e.g) testOneWayVideo/chrome...
6 days ago henrik.lundin... Add encoded_timestamp to AudioEncoder base class
6 days ago henrik.lundin... New interface class AudioEncoder
6 days ago stefan@webrtc.orgDisable a bunch of Nat and Ice tests when running under...
6 days ago andresp@webrtc.orgImprove rtcbot to load all test files at start and...
6 days ago asapersson... Add ability to include a larger time span (in addition...
7 days ago henrike@webrtc.orgCreate a copy of talk/xmpp and talk/p2p under webrtc...
7 days ago houssainy@google.comSelecting bot_type changed to be specified in the test...
7 days ago pbos@webrtc.orgFix data races in ThreadTest.ThreeThreadsInvoke.
7 days ago bjornv@webrtc.orgaudio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32...
7 days ago bjornv@webrtc.orgaudio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_...
7 days ago bjornv@webrtc.orgaudio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W...
7 days ago henrik.lundin... Extend AcmSwitchingOutputFrequencyOldApi with more...
7 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.
8 days ago henrike@webrtc.orglibjingle: use _stricmp instead of deprecated stricmp.
8 days ago pbos@webrtc.orgRemove -1 from Call::Config::start_bitrate_bps.
8 days ago stefan@webrtc.orgAdd periodic logging of received RTP headers and estima...
8 days ago henrik.lundin... New ACM test to trigger audio glitch when switching...
8 days ago stefan@webrtc.orgAdd a packet loss full stack test to the new API.
8 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...
9 days ago marpan@webrtc.orgDisable VP9 integration tests on DrMemory.
9 days ago bjornv@webrtc.orgcommon_audio: Removed macro WEBRTC_SPL_RSHIFT_W16
9 days ago kwiberg@webrtc.orgiSAC tests: Type buffers as uint8_t[] to avoid casts
9 days ago bjornv@webrtc.orgaudio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16...
9 days ago kwiberg@webrtc.orgWebRtcIsac_Decode et al.: Type encoded data as uint8...
9 days ago kwiberg@webrtc.orgWebRtcIsac_UpdateBwEstimate et al.: Type byte streams...
9 days ago kwiberg@webrtc.orgSome WebRtcIsac_* and WebRtcIsacfix_* functions: type...
9 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.
12 days ago marpan@webrtc.orgAdd VP9 codec to VCM and vie_auto_test.
12 days ago xians@webrtc.orgMark all virtual overrides in the hierarchy of Transpor...
12 days ago kjellander... Cleanup scripts and suppressions for TSan v1
12 days ago pbos@webrtc.orgRemove talk_base from suppressions.
12 days ago xians@webrtc.orgReland 28629004: adding new AEC dump start interface...
12 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...
13 days ago henrike@webrtc.orgbase/thread_unittest: wrap test was setting current...
13 days ago henrike@webrtc.orgMake pbos and kjellander only owners of tsan2 suppressions.
13 days ago henrik.lundin... Fix comments in common_types.h
13 days ago pbos@webrtc.orgIncrease timeout for AsyncWriteTest.TestWrite.
13 days ago kwiberg@webrtc.orgOpus wrapper: Use const for inputs and uint8[] for...
13 days ago kjellander... Make DEPS find check_root_dir.py in legacy checkouts.
13 days ago minyue@webrtc.orgEstimating NTP time with a given RTT.
13 days ago minyue@webrtc.orgRemoving useless packets when inserting them (NetEq)
13 days ago kjellander... Remove root_dir variable from DEPS + enforce rename.
13 days ago bjornv@webrtc.orgcommon_audio: Removed macro WEBRTC_SPL_LSHIFT_W16
13 days ago pbos@webrtc.orgDisable TestDTLSConnectWithSmallMtu on all platforms.
2014-10-09 andrew@webrtc.orgUse openmax_dl on all ARM (v7 or higher) platforms.
2014-10-09 glaznev@webrtc.orgRemove bad waiting code from video decoder release...
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