Log - 38c121c484e12f677c2cb6afb882cd024bd469c1 - platform/external/webrtc

38c121c Minor modifications to test::RtpFileReader by henrik.lundin@webrtc.org · 10 years ago
1795c35 Add default implementation of Add/RemoveObserver. by pbos@webrtc.org · 10 years ago
65e56db audio_processing/aecm: Added help function for calculating log of energy by bjornv@webrtc.org · 10 years ago
23ec837 audio_processing: Removed usage of macro WEBRTC_SPL_MUL by bjornv@webrtc.org · 10 years ago
750423c audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with << by bjornv@webrtc.org · 10 years ago
8cad943 Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" by kjellander@webrtc.org · 10 years ago
02cd306 Update isolate.gypi files + link to isolate_driver.py by kjellander@webrtc.org · 10 years ago
359d720 Allow Android apps to set video renderer scaling type. by glaznev@webrtc.org · 10 years ago
7dfb7fa Reland disallowing blocking calls on the worker thread. by jiayl@webrtc.org · 10 years ago
ea6c12e Set thread scheduling parameters inside the new thread. by henrike@webrtc.org · 10 years ago
6266240 Disable flaky tests: by asapersson@webrtc.org · 10 years ago
e794c36 Fix parallel test execution for tools, testsupport and metrics tests. by kjellander@webrtc.org · 10 years ago
d711181 audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with << by bjornv@webrtc.org · 10 years ago
7c15510 common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32 by bjornv@webrtc.org · 10 years ago
24f62e1 Adding getStats function to the exposed PeerConnection in RtcBot by houssainy@google.com · 10 years ago
730d270 Remove callback from RtpDepacketizer::Parse(). by pbos@webrtc.org · 10 years ago
f21ea91 GN: Add common configs to all targets. by kjellander@webrtc.org · 10 years ago
34f2a9e Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 10 years ago
3a10d2f Roll chromium_revision deaf2f7e..c264a056 (295079:297113) by kjellander@webrtc.org · 10 years ago
6c6680a Cleanup .gclient.bot_entries to avoid sync problems on bots. by kjellander@webrtc.org · 10 years ago
3902054 Roll chromium_revision 6455c69..deaf2f7 (293954:295079) by kjellander@webrtc.org · 10 years ago
bebc75e Fix the duplicated candidate problem when using multiple STUN servers. by jiayl@webrtc.org · 10 years ago
0a256ac Getting orientation is not working properly. VideoCaptureImpl::RotationFromDegrees returns -1 in case fails not 0. So we need to change the if statement. by braveyao@webrtc.org · 10 years ago
5d0071f Build one of NSS or BoringSSL but not both. by pthatcher@webrtc.org · 10 years ago
a21d071 Reverting part of by thorcarpenter@google.com · 10 years ago
1fd362c Do not assert for blocking call allowed in Thread::Join. by jiayl@webrtc.org · 10 years ago
384d05f Remove the different block lengths in ns_core by aluebs@webrtc.org · 10 years ago
5088377 Revert 7297 "Remove the different block lengths in ns_core" by aluebs@webrtc.org · 10 years ago
ca110b8 Mark virtual overrides of ViENetwork and VoENetwork as such. by henrikg@webrtc.org · 10 years ago
8b2e50c Revert 7302 "Roll chromium revision: 6455c69:2687a76" by marpan@webrtc.org · 10 years ago
bfacaab Add accessors for array of channel pointers in AudioBuffer. They are by claguna@google.com · 10 years ago
b38959e Roll chromium revision: 6455c69:2687a76 by marpan@webrtc.org · 10 years ago
f1d751c Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. by jiayl@webrtc.org · 10 years ago
0530511 Explicitly initialize SSL for tests. by pbos@webrtc.org · 10 years ago
61e811f Bump to version 39 by tnakamura@webrtc.org · 10 years ago
60fbd65 Removing error triggered for disabling FEC on non-opus by minyue@webrtc.org · 10 years ago
5f39657 Remove the different block lengths in ns_core by aluebs@webrtc.org · 10 years ago
741711a Revert r7049/r7123, which added unnecessary "u"s to "return 0"s. by henrik.lundin@webrtc.org · 10 years ago
3156699 Fix typo from RtpPacketizerH264. by pbos@webrtc.org · 10 years ago
37e1846 Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293). by andresp@webrtc.org · 10 years ago
fe1eafb Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. by jiayl@webrtc.org · 10 years ago
30be827 Enable render downmixing to mono in AudioProcessing. by andrew@webrtc.org · 10 years ago
e1bba60 Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac by jiayl@webrtc.org · 10 years ago
3987b6d Fix a problem in Thread::Send. by jiayl@webrtc.org · 10 years ago
a0ce9fa Call NS AnalyzeCaptureAudio before AEC by aluebs@webrtc.org · 10 years ago
70e2d11 Reduce jitter delay for low fps streams. Enabled by finch flag. by sprang@webrtc.org · 10 years ago
275dac2 Moved the filter calculation from analyze to process in ns_core by aluebs@webrtc.org · 10 years ago
634c926 audioproc: Now also writes to output file in simulation mode by bjornv@webrtc.org · 10 years ago
7ee24a7 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 10 years ago
d60d79a Thread annotation of rtc::CriticalSection. by pbos@webrtc.org · 10 years ago
38344ed Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
8166fae Change Android video renderer to maintain video aspect by glaznev@webrtc.org · 10 years ago
90668b1 Switch HW video decoder to output byte buffers if video by glaznev@webrtc.org · 10 years ago
1b7dcc1 (Auto)update libjingle 76169599-> 76176062 by buildbot@webrtc.org · 10 years ago
94ff92c Use VPX_IMG_FMT_*/VPX_PLANE_* defines by johannkoenig@google.com · 10 years ago
2c1bcea Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
3987f10 Revert "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
bf7b9e0 Remove DTMF status methods from Voice Engine by henrik.lundin@webrtc.org · 10 years ago
e34a2e7 Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175) by kjellander@webrtc.org · 10 years ago
faf2410 gn: Hide modules/video_capture:video_capture_internal_impl behind an arg by pbos@webrtc.org · 10 years ago
0e6e4d2 Reland "Converting five tests to use new AudioCoding interface" (r7258) by henrik.lundin@webrtc.org · 10 years ago
4f6f22f Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface" by andresp@webrtc.org · 10 years ago
ea29787 audio_processing/agc: Solved building with AGC_DEBUG + few style changes by bjornv@webrtc.org · 10 years ago
0a2087a Skeleton for registering external encoders/decoders. by pbos@webrtc.org · 10 years ago
c569a49 Unit tests for SSLAdapter by tkchin@webrtc.org · 10 years ago
dc0b37d modules_unittests: Turned on ApmTest.Process test for Android by bjornv@webrtc.org · 10 years ago
a3c4d4d Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." by andrew@webrtc.org · 10 years ago
8c5740b WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 10 years ago
83f95ba Remove engine-level SetOptions. by pbos@webrtc.org · 10 years ago
99e404c Revert "Converting five tests to use new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
35850ff Adding test file path as argument of the rtcBot run command's arguments. by houssainy@google.com · 10 years ago
64a2f10 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 10 years ago
07ca949 Adding webrtc_video_streaming test by houssainy@google.com · 10 years ago
c570761 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 10 years ago
cfe0735 Convert AcmReceiverTest to new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
eb1de5c Converting five tests to use new AudioCoding interface by henrik.lundin@webrtc.org · 10 years ago
bdfdc96 Clang-format ns_core by aluebs@webrtc.org · 10 years ago
759982d Set number of temporal layers for VideoSendStream. by pbos@webrtc.org · 10 years ago
6121715 Ensure that NetEq recovers after a large timestamp jump by henrik.lundin@webrtc.org · 10 years ago
8877287 Disabled several rtc_unittests so the tests can be turned on in the waterfall by henrike@webrtc.org · 10 years ago
97ed393 Reapply 23529005 after fixing the build break issue (Chromium:582133002) by guoweis@webrtc.org · 10 years ago
ed5ca1f (Auto)update libjingle 75925673-> 75926712 by buildbot@webrtc.org · 10 years ago
c98f217 (Auto)update libjingle 75924589-> 75925673 by buildbot@webrtc.org · 10 years ago
0c9fe72 (Auto)update libjingle 75922684-> 75924589 by buildbot@webrtc.org · 10 years ago
ebf2757 Fix HW video decoder crash on some Android KK devices. by glaznev@webrtc.org · 10 years ago
c1eebfa Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc. by thorcarpenter@google.com · 10 years ago
e658124 Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
fbf3bfe Separate between Analyze and Process in NS by aluebs@webrtc.org · 10 years ago
9570560 Additional disabled tests in rtc_unittests. by kjellander@webrtc.org · 10 years ago
34ac776 Additional disabled tests in rtc_unittests. by kjellander@webrtc.org · 10 years ago
fded02c base: disabled several base tests on Mac so that rtc_unittests can be turned back on by henrike@webrtc.org · 10 years ago
bbe0a85 Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
0268611 Re-enable missing android tests disabled due to issue 3770. by andresp@webrtc.org · 10 years ago
2036a7b Clean directx_sdk_path as it is already defined in base/common.gypi by andresp@webrtc.org · 10 years ago
5ca6008 Creating a test helper class TimestampJumpRtpGenerator by henrik.lundin@webrtc.org · 10 years ago
6e5c784 (Auto)update libjingle 75875619-> 75878731 by buildbot@webrtc.org · 10 years ago
b5a5c44 (Auto)update libjingle 75865376-> 75875619 by buildbot@webrtc.org · 10 years ago
d7acf11 (Auto)update libjingle 75854833-> 75865376 by buildbot@webrtc.org · 10 years ago
ccb3e3f (Auto)update libjingle 75854418-> 75854833 by buildbot@webrtc.org · 10 years ago
dcc1f04 (Auto)update libjingle 75852725-> 75853560 by buildbot@webrtc.org · 10 years ago