external/webrtc.git
119 min ago andrew@webrtc.orgChanged mips_arch_variant variable value corresponding... master
2 hours ago xians@webrtc.orgRevert 7337 "Reland 28629004: adding new AEC dump start...
2 hours ago xians@webrtc.orgRevert 7338 "Fixed the android build by making the...
2 hours ago houssainy@google.comCollecting stats every fixed time in webrtc_video_strea...
2 hours ago andrew@webrtc.orgMinor code change to fix some warnings in MIPS build.
2 hours ago xians@webrtc.orgFixed the android build by making the interface pure...
3 hours ago xians@webrtc.orgReland 28629004: adding new AEC dump start interface...
3 hours ago henrike@webrtc.orgAdds isolate for rtc_unittests and moves sound's unitte...
4 hours ago xians@webrtc.orgRevert 7334 "adding new AEC dump start interface for...
4 hours ago xians@webrtc.orgadding new AEC dump start interface for chrome.
6 hours ago henrik.lundin... Minor modifications to test::RtpFileReader
8 hours ago pbos@webrtc.orgAdd default implementation of Add/RemoveObserver.
8 hours ago bjornv@webrtc.orgaudio_processing/aecm: Added help function for calculat...
8 hours ago bjornv@webrtc.orgaudio_processing: Removed usage of macro WEBRTC_SPL_MUL
8 hours ago bjornv@webrtc.orgaudio_processing: Replaced trivial macro WEBRTC_SPL_LSH...
9 hours ago kjellander... Revert 7327 "Update isolate.gypi files + link to isolat...
9 hours ago kjellander... Update isolate.gypi files + link to isolate_driver.py
18 hours ago glaznev@webrtc.orgAllow Android apps to set video renderer scaling type.
19 hours ago jiayl@webrtc.orgReland disallowing blocking calls on the worker thread.
23 hours ago henrike@webrtc.orgSet thread scheduling parameters inside the new thread.
27 hours ago asapersson... Disable flaky tests:
30 hours ago kjellander... Fix parallel test execution for tools, testsupport...
30 hours ago bjornv@webrtc.orgaudio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16...
32 hours ago bjornv@webrtc.orgcommon_audio refactoring: Removed macro WEBRTC_SPL_LSHI...
32 hours ago houssainy@google.comAdding getStats function to the exposed PeerConnection...
33 hours ago pbos@webrtc.orgRemove callback from RtpDepacketizer::Parse().
2 days ago kjellander... GN: Add common configs to all targets.
2 days ago pbos@webrtc.orgInitialize SSL in unittest_main.cc.
2 days ago kjellander... Roll chromium_revision deaf2f7e..c264a056 (295079:297113)
2 days ago kjellander... Cleanup .gclient.bot_entries to avoid sync problems...
2 days ago kjellander... Roll chromium_revision 6455c69..deaf2f7 (293954:295079)
3 days ago jiayl@webrtc.orgFix the duplicated candidate problem when using multipl...
3 days ago braveyao@webrtc.orgGetting orientation is not working properly. VideoCaptu...
3 days ago pthatcher@webrtc.orgBuild one of NSS or BoringSSL but not both.
4 days ago thorcarpenter... Reverting part of
4 days ago jiayl@webrtc.orgDo not assert for blocking call allowed in Thread:...
4 days ago aluebs@webrtc.orgRemove the different block lengths in ns_core
4 days ago aluebs@webrtc.orgRevert 7297 "Remove the different block lengths in...
4 days ago henrikg@webrtc.orgMark virtual overrides of ViENetwork and VoENetwork...
4 days ago marpan@webrtc.orgRevert 7302 "Roll chromium revision: 6455c69:2687a76"
4 days ago claguna@google.comAdd accessors for array of channel pointers in AudioBuf...
4 days ago marpan@webrtc.orgRoll chromium revision: 6455c69:2687a76
5 days ago jiayl@webrtc.orgCall SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
5 days ago pbos@webrtc.orgExplicitly initialize SSL for tests.
5 days ago tnakamura@webrtc.orgBump to version 39
5 days ago minyue@webrtc.orgRemoving error triggered for disabling FEC on non-opus
5 days ago aluebs@webrtc.orgRemove the different block lengths in ns_core
5 days ago henrik.lundin... Revert r7049/r7123, which added unnecessary "u"s to...
5 days ago pbos@webrtc.orgFix typo from RtpPacketizerH264.
5 days ago andresp@webrtc.orgRevert "Call SSL_shutdown in OpenSSLStreamAdapter:...
5 days ago jiayl@webrtc.orgCall SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
5 days ago andrew@webrtc.orgEnable render downmixing to mono in AudioProcessing.
6 days ago jiayl@webrtc.orgAdd missing DesktopConfigurationMonitor Unlock in webrt...
6 days ago jiayl@webrtc.orgFix a problem in Thread::Send.
6 days ago aluebs@webrtc.orgCall NS AnalyzeCaptureAudio before AEC
6 days ago sprang@webrtc.orgReduce jitter delay for low fps streams.
6 days ago aluebs@webrtc.orgMoved the filter calculation from analyze to process...
6 days ago bjornv@webrtc.orgaudioproc: Now also writes to output file in simulation...
6 days ago kwiberg@webrtc.orgWebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encode...
6 days ago pbos@webrtc.orgThread annotation of rtc::CriticalSection.
6 days ago pbos@webrtc.orgMove thread_annotations.h to webrtc/base/.
6 days ago glaznev@webrtc.orgChange Android video renderer to maintain video aspect
6 days ago glaznev@webrtc.orgSwitch HW video decoder to output byte buffers if video
7 days ago buildbot@webrtc.org(Auto)update libjingle 76169599-> 76176062
7 days ago johannkoenig... Use VPX_IMG_FMT_*/VPX_PLANE_* defines
7 days ago guoweis@webrtc.orgEnable ipv6 by default for webrtc under a Finch experiment.
7 days ago henrik.lundin... Revert "Remove DTMF status methods from Voice Engine...
7 days ago henrik.lundin... Remove DTMF status methods from Voice Engine
7 days ago kjellander... Revert "Set minimum SDK level to 10.7 for Mac and iOS...
7 days ago pbos@webrtc.orggn: Hide modules/video_capture:video_capture_internal_i...
7 days ago henrik.lundin... Reland "Converting five tests to use new AudioCoding...
7 days ago andresp@webrtc.orgReland (rev 7259) "Convert AcmReceiverTest to new Audio...
7 days ago bjornv@webrtc.orgaudio_processing/agc: Solved building with AGC_DEBUG...
7 days ago pbos@webrtc.orgSkeleton for registering external encoders/decoders.
7 days ago tkchin@webrtc.orgUnit tests for SSLAdapter
7 days ago bjornv@webrtc.orgmodules_unittests: Turned on ApmTest.Process test for...
7 days ago andrew@webrtc.orgRevert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode...
7 days ago kwiberg@webrtc.orgWebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encode...
8 days ago pbos@webrtc.orgRemove engine-level SetOptions.
8 days ago andresp@webrtc.orgRevert "Converting five tests to use new AudioCoding...
8 days ago houssainy@google.comAdding test file path as argument of the rtcBot run...
8 days ago henrik.lundin... Remove Get/SetNetEQPlayoutMode APIs
8 days ago houssainy@google.comAdding webrtc_video_streaming test
8 days ago andresp@webrtc.orgRevert "Convert AcmReceiverTest to new AudioCoding...
8 days ago henrik.lundin... Convert AcmReceiverTest to new AudioCoding interface
8 days ago henrik.lundin... Converting five tests to use new AudioCoding interface
8 days ago aluebs@webrtc.orgClang-format ns_core
8 days ago pbos@webrtc.orgSet number of temporal layers for VideoSendStream.
8 days ago henrik.lundin... Ensure that NetEq recovers after a large timestamp...
8 days ago henrike@webrtc.orgDisabled several rtc_unittests so the tests can be...
10 days ago guoweis@webrtc.orgReapply 23529005 after fixing the build break issue...
10 days ago buildbot@webrtc.org(Auto)update libjingle 75925673-> 75926712
10 days ago buildbot@webrtc.org(Auto)update libjingle 75924589-> 75925673
10 days ago buildbot@webrtc.org(Auto)update libjingle 75922684-> 75924589
10 days ago glaznev@webrtc.orgFix HW video decoder crash on some Android KK devices.
10 days ago thorcarpenter... Fix the libjingle_media_unittest failure in Windows...
11 days ago glaznev@webrtc.orgFixing compilation failure in peerconnection_jni.cc...
11 days ago aluebs@webrtc.orgSeparate between Analyze and Process in NS
11 days ago kjellander... Additional disabled tests in rtc_unittests.
11 days ago kjellander... Additional disabled tests in rtc_unittests.
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