Log - a0ce9fa2a65a693ca9a6ee4920fb41d5cdd92e3b - platform/external/webrtc

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
0b435ba A few fixes to avoid crash in HW codec on device orientation change. by glaznev@webrtc.org · 10 years ago
143ffa4 Update iOS video capture to use non-deprecated APIs. by tkchin@webrtc.org · 10 years ago
83af77b Revert maximum video codec resolution on Android back to 720p again. by glaznev@webrtc.org · 10 years ago
933d88a (Auto)update libjingle 75818332-> 75837294 by buildbot@webrtc.org · 10 years ago
c3091a6 Remove the 'webrtc_test_video_render_dependencies' target. by pbos@webrtc.org · 10 years ago
42731bd Avoid writing a double/float to a string to avoid a crash. by jiayl@webrtc.org · 10 years ago
ba737cb Do not require synchronization access on the thread if called from rtc::Thread::WrapCurrent. by jiayl@webrtc.org · 10 years ago
6116062 Trying to fix Chrome FYI bots. by andresp@webrtc.org · 10 years ago
e94f83a Cleanup .gclient_entries to avoid sync problems. by kjellander@webrtc.org · 10 years ago
205c15a Adds asan suppresions for rtc_unittests by henrike@webrtc.org · 10 years ago
6cd6ba8 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
c7134f8 Fix proper deps in BUILD.gn files. This should make Chrome GN bots happy. by andresp@webrtc.org · 10 years ago
fda2c2e Add Analyze API to NS by aluebs@webrtc.org · 10 years ago
ab071da Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
369a637 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
3b67f8e Enable HW video decoding on Qualcomm devices. by glaznev@webrtc.org · 10 years ago
d91608d The 2x2 black frame on windows when the shared window is minimized caused an assert from vp8 and may lead to memroy corruption. by jiayl@webrtc.org · 10 years ago
5422e72 Modifying NetEqExternalDecoderTest by henrik.lundin@webrtc.org · 10 years ago
4a5061f talk/p2p/base: removed unused variable "port_" by henrike@webrtc.org · 10 years ago
5a098c5 Refactor VP8 de-packetizer. by stefan@webrtc.org · 10 years ago
3bd5603 Revert "Disable video_capture_tests for Android." (revision 7023). by andresp@webrtc.org · 10 years ago
a74eda1 Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
85ef770 Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
ab990ae Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
6a9b155 (Auto)update libjingle 75683337-> 75695882 by buildbot@webrtc.org · 10 years ago
e387cc0 webrtc/overrides: add OWNERS-file. by henrike@webrtc.org · 10 years ago
dc8dcb4 Narrower include for constructormagic.h in Chromium. by pbos@webrtc.org · 10 years ago
eb43264 Remove linux_memcheck from default trybots. by kjellander@webrtc.org · 10 years ago
a59c501 Java VideoRenderer class may be backed by two different native by glaznev@webrtc.org · 10 years ago
40c2aa3 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
f8bff76 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
7351d4d Add a gyp target for producing a voice engine merged library. by andrew@webrtc.org · 10 years ago
a6cefca gn: Fix cflags usage by pbos@webrtc.org · 10 years ago
cddd17c Recreate VideoStreams when setting resolution. by pbos@webrtc.org · 10 years ago
88e85ad Add pbos@webrtc.org (myself) to talk/media/webrtc/. by pbos@webrtc.org · 10 years ago
dae612e Mark all virtual overrides in the hierarchies of UdpTransportData and by henrikg@webrtc.org · 10 years ago
80132e4 (Auto)update libjingle 75610402-> 75610402 by buildbot@webrtc.org · 10 years ago
699c46a rtc_unittest: prevent execution of broken tests. by henrike@webrtc.org · 10 years ago
4436020 Fix GN for rtc_base_approved target. by kjellander@webrtc.org · 10 years ago
178015d memcheck: suppressions didn't map over directly when moving base from talk to webrtc (part of the suppression that is not related to the signature differed). Fixed suppressions accordingly. by henrike@webrtc.org · 10 years ago
595b23c Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..." by kjellander@webrtc.org · 10 years ago
c75f607 audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2 by bjornv@webrtc.org · 10 years ago
6ae5a6d Add a target for the approved subset of rtc_base. by andrew@webrtc.org · 10 years ago
b3cbeb3 Fix memory leak in webrtc::MouseCursorMonitorMac by sergeyu@chromium.org · 10 years ago