Log - 3c0aae17f0e3a70fe90ecc6835926b66a3de18fb - platform/external/webrtc

3c0aae1 Change gflags and gmock includes to be full paths. by kjellander@webrtc.org · 10 years ago
51bb33c ACMOpus: Remove useless member variable fec_enabled_ by kwiberg@webrtc.org · 10 years ago
7825b1a Add support for multi-channel DTMF tone generation by henrik.lundin@webrtc.org · 10 years ago
bcb6bcf Remove HybridVideoEngine. by pbos@webrtc.org · 10 years ago
9d45393 Change return value for number of discarded packets to be int. by asapersson@webrtc.org · 10 years ago
01581da Fix audio/video sync when FEC is enabled. by stefan@webrtc.org · 10 years ago
bfd7a8c Fix compile errors on webrtc/base. by andresp@webrtc.org · 10 years ago
0229cba Remove ambiguous call to MakeCheckOpString. by andresp@webrtc.org · 10 years ago
95c2458 * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files. by thorcarpenter@google.com · 10 years ago
9328f39 cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error by fbarchard@google.com · 10 years ago
5b83af4 Fix leak of NSAutoreleasePool. by tkchin@webrtc.org · 10 years ago
609f987 (Auto)update libjingle 74696326-> 74723281 by buildbot@webrtc.org · 10 years ago
1b8b4c4 Revert 7041 " Audio codecs to include webrtc/typedefs.h" by henrike@webrtc.org · 10 years ago
fa4535b (Auto)update libjingle 74694022-> 74696326 by buildbot@webrtc.org · 10 years ago
26c0c41 Network up/down signaling in Call. by pbos@webrtc.org · 10 years ago
ebee401 Remove flake in SendsLowerResolutionOnSmallerFrames. by pbos@webrtc.org · 10 years ago
c4175b9 Set resolution based on incoming VideoFrames. by pbos@webrtc.org · 10 years ago
9730d3a Audio codecs to include webrtc/typedefs.h by andresp@webrtc.org · 10 years ago
0372b93 Partial revert of r7014 (Android APK refactor) by kjellander@webrtc.org · 10 years ago
bac0726 Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes by aluebs@webrtc.org · 10 years ago
adee8f9 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 10 years ago
0a214ff Setting marker bit on DTMF correctly by stefan@webrtc.org · 10 years ago
74cf916 Fix issues in audioproc for float aecdumps by aluebs@webrtc.org · 10 years ago
48f2568 audio_processing/nsx: Bug fix that could cause divide by zero by bjornv@webrtc.org · 10 years ago
d944a68 Suppressing VideoAdapterTest.AdaptResolutionWide and VideoAdapterTest.AdaptResolutionNarrow on DrMemory by minyue@webrtc.org · 10 years ago
72e4485 (Auto)update libjingle 74628537-> 74648573 by buildbot@webrtc.org · 10 years ago
9075048 Remove deprecated RTCVideoRenderer constructor. by tkchin@webrtc.org · 10 years ago
34a6764 Remove the checks.h dependence on logging.h in a standalone build. by andrew@webrtc.org · 10 years ago
8e24d87 Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator from both the audio playback thread and the network thread without locking. by stefan@webrtc.org · 10 years ago
9f34128 Remove WebRtcVideoEngine::default_codec_format(). by pbos@webrtc.org · 10 years ago
0365514 Remove files from talk/PRESUBMIT.py. by pbos@webrtc.org · 10 years ago
d72a759 Create a copy of talk/xmllite under webrtc/xmllite. by henrike@webrtc.org · 10 years ago
6f729e8 Disable video_engine_tests and webrtc_perf_tests on Android. by kjellander@webrtc.org · 10 years ago
ee0fb18 Divide-by-zero problem in NetEq's Normal::Process fixed by henrik.lundin@webrtc.org · 10 years ago
94da203 Remove retired android_apk[_rel] trybots from PRESUBMIT.py by kjellander@webrtc.org · 10 years ago
324b72d Disable video_capture_tests for Android. by kjellander@webrtc.org · 10 years ago
e281f7f GN: Update webrtc/base to recent GYP changes. by kjellander@webrtc.org · 10 years ago
468516c RTCBot is a framework that allows to write tests where logic runs on a single by andresp@webrtc.org · 10 years ago
561a9ec Update checkedeps.py rules in DEPS. by kjellander@webrtc.org · 10 years ago
76a4257 Remove build_with_chromium==1 conditions for Android by kjellander@webrtc.org · 10 years ago
841f58f Unpacking aecdumps generates wav files by aluebs@webrtc.org · 10 years ago
c3f42f3 Fix audio_decoder_unittests.isolate by kjellander@webrtc.org · 10 years ago
8dbeb5b Adding more codecs to the AcmSenderBitExactness by henrik.lundin@webrtc.org · 10 years ago
7e86049 Roll chromium_revision 681cc8e..f0a439d (r292217:r292861) by kjellander@webrtc.org · 10 years ago
3bd4156 Android APK tests built from a normal WebRTC checkout. by kjellander@webrtc.org · 10 years ago
c4870bb GN: Audio device module by kjellander@webrtc.org · 10 years ago
524b8f7 GN: Implement voice engine, common audio, audio coding and audio processing by kjellander@webrtc.org · 10 years ago
1b9a188 GN: Fix webrtc/video/BUILD.gn for Chromium build. by kjellander@webrtc.org · 10 years ago
a22485e MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 10 years ago
af7fdfc Add LTO support for Android Chromium. by andrew@webrtc.org · 10 years ago
f554d75 Allow same src and dst in InputAudioFile::DuplicateInterleaved by henrik.lundin@webrtc.org · 10 years ago
44010f3 win: Replace custom assert() macro with regular assert.h by thakis@chromium.org · 10 years ago
bc3f333 Add jiayl to talk OWNERS. by jiayl@webrtc.org · 10 years ago
e21cc9a When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated. by jiayl@webrtc.org · 10 years ago
b0dc3d7 Precompile out our standalone CHECK macros in a Chromium build. by andrew@webrtc.org · 10 years ago
a5b7869 Add CHECK and friends from Chromium. by andrew@webrtc.org · 10 years ago
11c6bde Specify an ECDH group for ECDHE. by jiayl@webrtc.org · 10 years ago
55e9da1 Add talk owners to migrated talk folders by henrike@webrtc.org · 10 years ago
4431fd6 Add 60 fps video support by niklas.enbom@webrtc.org · 10 years ago
788f058 GN: Implement video_engine, video_capture and video_render. by kjellander@webrtc.org · 10 years ago
df9fef6 common_audio: Removed macro WEBRTC_SPL_DIV by bjornv@webrtc.org · 10 years ago
1f8a237 (Auto)update libjingle 74235596-> 74297316 by buildbot@webrtc.org · 10 years ago
59a1b1b Fix the different samples per channel in aecdump by aluebs@webrtc.org · 10 years ago
deaece6 Disable VideoAdapterTest.BlackOutput on DrMemory. by pbos@webrtc.org · 10 years ago
f8723d6 Add unit tests to rtcp_receiver_test. by asapersson@webrtc.org · 10 years ago
2dbb47a Roll chromium_revision b1748b:681cc8 by marpan@webrtc.org · 10 years ago
956f281 Re-enable all VideoAdapterTests on DrMemory. by pbos@webrtc.org · 10 years ago
75c3ec1 Fix data races during VideoAdapterTest tear-down. by pbos@webrtc.org · 10 years ago
573a1ee (Auto)update libjingle 74202294-> 74230205 by buildbot@webrtc.org · 10 years ago
18584fc Move end of namespace inside #ifdef by henrik.lundin@webrtc.org · 10 years ago
c3c2911 Expose setPayloadType on the rtp_sender. Thus allowing other users of this module by andresp@webrtc.org · 10 years ago
00f11f5 - Make local constant non-static. - Remove spammy log line. by solenberg@webrtc.org · 10 years ago
66a3582 Create a copy of talk/sound under webrtc/sound. by henrike@webrtc.org · 10 years ago
7087857 implement handling ALTERNATE-SERVER response from turn protocol as by guoweis@webrtc.org · 10 years ago
dc926a0 Avoid syncing unnecessary Chromium deps for WebRTC. by kjellander@webrtc.org · 10 years ago
3533bfc (Auto)update libjingle 74132319-> 74133664 by buildbot@webrtc.org · 10 years ago
4470d78 (Auto)update libjingle 74128148-> 74132319 by buildbot@webrtc.org · 10 years ago
b623c5c Disable EndToEndTest.RestartingSendStreamPreservesRtpState in video_engine_tests because it is flaky by aluebs@webrtc.org · 10 years ago
f21ac1f Fix Win64 compile of videoadapter_unittest.cc. by pbos@webrtc.org · 10 years ago
c9b3f77 Fix data races in VideoAdapterTest. by pbos@webrtc.org · 10 years ago
8940ce7 Updating svn:ignore entries by kjellander@webrtc.org · 10 years ago
b648b9d Remove test constructor in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
4f71e22 Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV by bjornv@webrtc.org · 10 years ago
1de0cc4 common_audio: Re-enable WebRtcSpl_AddSatW32() and WebRtcSpl_SubSatW32() optimizations on armv7 by bjornv@webrtc.org · 10 years ago
047a46f Remove Android.mk build files. by pbos@webrtc.org · 10 years ago
b96ea2a Remove former team members from OWNERS and WATCHLISTS by kjellander@webrtc.org · 10 years ago
204cd56 (Auto)update libjingle 74064646-> 74072040 by buildbot@webrtc.org · 10 years ago
e9bfed0 Move constant so it is not stripped out for TSAN bots. by kjellander@webrtc.org · 10 years ago
857130f (Auto)update libjingle 74039473-> 74044292 by buildbot@webrtc.org · 10 years ago
79ad37e Update root OWNERS file by kjellander@webrtc.org · 10 years ago
6556a59 As expected, r6569 (https://code.google.com/p/webrtc/source/detail?r=6965) caused memcheck bots to complain. Adding expections for that, in line with outher peerconnection tests. by solenberg@webrtc.org · 10 years ago
c239234 Roll chromium_revision 289723:291647 by kjellander@webrtc.org · 10 years ago
42ee5b5 GN: Disable Chromium clang plugins for standalone build. by kjellander@webrtc.org · 10 years ago
b4c7b09 (Auto)update libjingle 73927775-> 74032598 by buildbot@webrtc.org · 10 years ago
926707b Refactoring common_audio: Replace trivial multiplication macro by bjornv@webrtc.org · 10 years ago
d32c438 Re-landing r6961 by bjornv@webrtc.org · 10 years ago
4a616be Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..." by bjornv@webrtc.org · 10 years ago
4f01017 common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 by bjornv@webrtc.org · 10 years ago
6e71d17 Refactoring common_audio/signal_processing: Replaces trivial macros by bjornv@webrtc.org · 10 years ago
584cd8d Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion) by kwiberg@webrtc.org · 10 years ago