Log - f68cc0b0c3ce88c434afb8f55787b67b76c66ec6 - platform/external/webrtc

f68cc0b (Auto)update libjingle 77554188-> 77629208 by buildbot@webrtc.org · 10 years ago
82e6fa5 Move exlusion of VP9 integration tests for DrMemory by marpan@webrtc.org · 10 years ago
b6af428 Adjust speech probability in NS when echo by aluebs@webrtc.org · 10 years ago
1e6a5dd Removes xmllite from talk/xmllite since webrtc/xmllite is used instead. by henrike@webrtc.org · 10 years ago
8bee130 Disable VP9 integration tests on DrMemory. by marpan@webrtc.org · 10 years ago
bc1a457 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16 by bjornv@webrtc.org · 10 years ago
a3722b6 iSAC tests: Type buffers as uint8_t[] to avoid casts by kwiberg@webrtc.org · 10 years ago
d4fe824 audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> by bjornv@webrtc.org · 10 years ago
396a5e0 WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[] by kwiberg@webrtc.org · 10 years ago
3f7f899 WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16 by kwiberg@webrtc.org · 10 years ago
1172988 Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[] by kwiberg@webrtc.org · 10 years ago
3c16d8b (Auto)update libjingle 77414393-> 77554188 by buildbot@webrtc.org · 10 years ago
c502df5 Merge the supporting to UYVY on Linux video capture in crbug/410202 to webrtc standalone. by braveyao@webrtc.org · 10 years ago
651c05e Release _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since they should depend on the filters. by braveyao@webrtc.org · 10 years ago
7f7b0a1 Re-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed when enabling other base tests). by henrike@webrtc.org · 10 years ago
4ddbbed Disable SendsAndReceivesVP9 test for now. by marpan@webrtc.org · 10 years ago
c87b747 Adjust/increase rate control thresold for a vp9 test. by marpan@webrtc.org · 10 years ago
573c78e Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
3cefbc9 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. by xians@webrtc.org · 10 years ago
afede83 Cleanup scripts and suppressions for TSan v1 by kjellander@webrtc.org · 10 years ago
fae6bc4 Remove talk_base from suppressions. by pbos@webrtc.org · 10 years ago
e46bc77 Reland 28629004: adding new AEC dump start interface for chrome. by xians@webrtc.org · 10 years ago
c5593ef Workaround deps2git issue with inline Python in DEPS. by kjellander@webrtc.org · 10 years ago
c732a3e Re-enable allmost all base tests. by henrike@webrtc.org · 10 years ago
4a73519 Re-enables a bunch of base unittests part II. by henrike@webrtc.org · 10 years ago
dae40dc Change setting VP8 codec specific info values by HW VP8 encoder by glaznev@webrtc.org · 10 years ago
e30dab7 base/thread_unittest: wrap test was setting current thread to NULL. by henrike@webrtc.org · 10 years ago
17f8ddd Make pbos and kjellander only owners of tsan2 suppressions. by henrike@webrtc.org · 10 years ago
8768f16 Fix comments in common_types.h by henrik.lundin@webrtc.org · 10 years ago
3ff788c Increase timeout for AsyncWriteTest.TestWrite. by pbos@webrtc.org · 10 years ago
4bd2db9 Opus wrapper: Use const for inputs and uint8[] for byte streams by kwiberg@webrtc.org · 10 years ago
1bada48 Make DEPS find check_root_dir.py in legacy checkouts. by kjellander@webrtc.org · 10 years ago
2c0cdbc Estimating NTP time with a given RTT. by minyue@webrtc.org · 10 years ago
c803907 Removing useless packets when inserting them (NetEq) by minyue@webrtc.org · 10 years ago
0b0ac82 Remove root_dir variable from DEPS + enforce rename. by kjellander@webrtc.org · 10 years ago
3ea35fd common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16 by bjornv@webrtc.org · 10 years ago
127ca3f Disable TestDTLSConnectWithSmallMtu on all platforms. by pbos@webrtc.org · 10 years ago
0001adc Use openmax_dl on all ARM (v7 or higher) platforms. by andrew@webrtc.org · 10 years ago
95bacfe Remove bad waiting code from video decoder release function. by glaznev@webrtc.org · 10 years ago
97abeee (Auto)update libjingle 77263371-> 77296420 by buildbot@webrtc.org · 10 years ago
536eb98 Re-enables a bunch of base unittests. by henrike@webrtc.org · 10 years ago
9ea5396 Roll chromium_revision fc668e2..2d714fa (298195:298667) by andrew@webrtc.org · 10 years ago
4165f7a Add a variable for deciding when to use openmax_dl. by andrew@webrtc.org · 10 years ago
f71785c audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> by bjornv@webrtc.org · 10 years ago
575d126 Protect send_/recv_streams_ in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
9c6dc46 CHECK/DCHECK: Explicitly state whether the condition can have side effects by kwiberg@webrtc.org · 10 years ago
5e3d7c7 Change name of a NetEq internal member variable by henrik.lundin@webrtc.org · 10 years ago
742922b Make the media content send only if offerToReceive is false while local streams exist. by jiayl@webrtc.org · 10 years ago
d6bda09 Initialize sctp_paddrparams in OpenSctpSocket(). by pbos@webrtc.org · 10 years ago
27e5898 Explicitly unpoison FDs for MSan. by pbos@webrtc.org · 10 years ago
46ffc70 Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder. by glaznev@webrtc.org · 10 years ago
963b979 Remove potential deadlock in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
a9e363e Roll chromium_revision c264a05..fc668e2 (297113:298195) by kjellander@webrtc.org · 10 years ago
77d5a57 Revert "Only configure the SSL library in one place." by pbos@webrtc.org · 10 years ago
6ed1cf4 Isolate: Remove use of --ignore_broken_items by kjellander@webrtc.org · 10 years ago
9103953 Fix neteq_rtpplay so that empty SSRC is valid by henrik.lundin@webrtc.org · 10 years ago
7cbc4f9 Set NetEq playout mode through the Config struct by henrik.lundin@webrtc.org · 10 years ago
8b65d51 Add an SSRC filter to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
532ed43 Prevent reading outside iSAC bitstream, if the stream is corrupted. by turaj@webrtc.org · 10 years ago
8234fa6 Only configure the SSL library in one place. by henrike@webrtc.org · 10 years ago
2fe5893 Mac: adds missing _DEBUG flag to mac debug builds. by henrike@webrtc.org · 10 years ago
528fc65 Fixing build issue with L-sdk by henrike@webrtc.org · 10 years ago
9a742b4 talk: removes empty directories base and sound. by henrike@webrtc.org · 10 years ago
5d3e7ac Check on the existence of report directory by houssainy@google.com · 10 years ago
42684be Wire up CPU adaptation in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
31b75ea Moves xmllite's unittests to rtc_unittest. by henrike@webrtc.org · 10 years ago
25cc745 Switch to SW video decoder on Android after getting 2 or more by glaznev@webrtc.org · 10 years ago
4b133da Let RtpFileSource use RtpFileReader by henrik.lundin@webrtc.org · 10 years ago
348eac6 audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >> by bjornv@webrtc.org · 10 years ago
5fa8c45 Remove mouse cursor capturer from the ScreenCapturer interface by sergeyu@chromium.org · 10 years ago
6138f0f Revert "Remove mouse cursor capturer from the ScreenCapturer interface" by sergeyu@chromium.org · 10 years ago
1fced0f Remove mouse cursor capturer from the ScreenCapturer interface by sergeyu@chromium.org · 10 years ago
76819d3 Add error trap for XFixesGetCursorImage() by sergeyu@chromium.org · 10 years ago
325cff0 Import LappedTransform and friends. by andrew@webrtc.org · 10 years ago
593c3a0 rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. by henrike@webrtc.org · 10 years ago
4530b2c Revert 7355 "Fix parallelization in libjingle_p2p_unittest." by henrike@webrtc.org · 10 years ago
36b0c1a Adds PRESUBMIT.py dispensation for depending on rtc_base. by henrike@webrtc.org · 10 years ago
fd29205 Fix parallelization in libjingle_p2p_unittest. by pbos@webrtc.org · 10 years ago
c86e45d Fix parallelizability in modules_tests. by pbos@webrtc.org · 10 years ago
4cebd84 Reland "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
4e4fe4f Add support for MSan by kjellander@webrtc.org · 10 years ago
afefed5 Update checkdeps.py rules in DEPS by kjellander@webrtc.org · 10 years ago
83fe69d Added presubmit protecting against inclusion of rtc_base, while allowing rtc_base_approved. by henrike@webrtc.org · 10 years ago
3037bc3 GN: Add common configs to tools and test. by kjellander@webrtc.org · 10 years ago
b8caf6a GN: Enable libvpx, add link target and convert some test targets by kjellander@webrtc.org · 10 years ago
d05756f Changed mips_arch_variant variable value corresponding to Chromium code changes. by andrew@webrtc.org · 10 years ago
79a7148 Revert 7337 "Reland 28629004: adding new AEC dump start interfac..." by xians@webrtc.org · 10 years ago
7aad5e5 Revert 7338 "Fixed the android build by making the interface pur..." by xians@webrtc.org · 10 years ago
d0bb586 Collecting stats every fixed time in webrtc_video_streaming.js test by houssainy@google.com · 10 years ago
db75a66 Minor code change to fix some warnings in MIPS build. by andrew@webrtc.org · 10 years ago
90d1979 Fixed the android build by making the interface pure virtual. by xians@webrtc.org · 10 years ago
14092e0 Reland 28629004: adding new AEC dump start interface for chrome by xians@webrtc.org · 10 years ago
792d1a0 Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest. by henrike@webrtc.org · 10 years ago
8752061 Revert 7334 "adding new AEC dump start interface for chrome." by xians@webrtc.org · 10 years ago
2e417d6 adding new AEC dump start interface for chrome. by xians@webrtc.org · 10 years ago
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