Log - 86ce46d4ffec5d1556888309abaadeeab32744d1 - platform/external/webrtc

86ce46d Cleaned up and completed current dashboard milestone. by phoglund@webrtc.org · 13 years ago
c80d9d9 Removed default cases causing clang errors, -Wcovered-switch-default. by mflodman@webrtc.org · 13 years ago
5eeaa38 Improved readability of tests in master.cfg and enabling some tests by kjellander@webrtc.org · 13 years ago
4942832 Fix "may be used uninitialized" warning. by andrew@webrtc.org · 13 years ago
b783a55 Unit test for forward_error_correction. by marpan@webrtc.org · 13 years ago
307c1ff Fix for issue #254: windows crash of test_fec. by marpan@webrtc.org · 13 years ago
dde977e AudioFrame payload shouldn't be mutable. by andrew@webrtc.org · 13 years ago
ce0a6ff Restoring previous vie_auto_test.gypi structure due to problems on Mac by kjellander@webrtc.org · 13 years ago
918a8bf External transport is modified to never drop packets from the first frame. by kjellander@webrtc.org · 13 years ago
6838334 Fix for warning in GCC 4.6 by henrik.lundin@webrtc.org · 13 years ago
82e1c8d Fix for issue 253 by henrik.lundin@webrtc.org · 13 years ago
fdf21c8 Removed dead version code. by pwestin@webrtc.org · 13 years ago
4ea57e5 Changed VP8 to follow the style guide a little bit more. by pwestin@webrtc.org · 13 years ago
9b3474a Disable the unused API interfaces for VoE chromium build. by xians@webrtc.org · 13 years ago
07b45a5 Added API for getting the send-side estimated bandwidth. by stefan@webrtc.org · 13 years ago
ac7e89f Correct and update LICENSE by leozwang@webrtc.org · 13 years ago
de66b91 In voice engine, added member audioFrame to classes AudioCodingModuleImpl and VoEBaseImpl, by kma@webrtc.org · 13 years ago
2b87891 Implemented build status tracking. by phoglund@webrtc.org · 13 years ago
fede80c Updated test web page info for PeerConnection v2. by henrikg@webrtc.org · 13 years ago
d4f0a0e Refactored the dashboard in order to add new functionality and added some new functionality. by phoglund@webrtc.org · 13 years ago
7fe219f Add some additional checks for corrupt payload. by andrew@webrtc.org · 13 years ago
1f238fd Update libsrtp revision in DEPS to chrome/deps/../libsrtp@119742. by mallinath@webrtc.org · 13 years ago
727a0a0 Fixed a bug in assembly code in aecm_core.c (hasn't caused a problem yet). by kma@webrtc.org · 13 years ago
d8f58a4 Cross platform build fix for SSIM (part 2) by frkoenig@google.com · 13 years ago
26e8a58 VAD refactor: Create() and Free() by bjornv@webrtc.org · 13 years ago
dd478e2 Fix for warning in GCC 4.6 by henrik.lundin@webrtc.org · 13 years ago
79af734 This patch fixes the converity warnings in voice engine. by xians@webrtc.org · 13 years ago
91c6308 Fix potential VCMReceiver crash. by stefan@webrtc.org · 13 years ago
2919e95 Resolves Coverty issue #10347. by henrika@webrtc.org · 13 years ago
cdba1a8 test_fec: Reduce execution time of test, and use testsupport/fileutils.h for path of randomSeedLog file. by marpan@webrtc.org · 13 years ago
293d22b Add a new macro for bit-exact audioproc tests. by andrew@webrtc.org · 13 years ago
72fe244 Fix neteq watchlist. by andrew@webrtc.org · 13 years ago
4065403 Use pointer-based CriticalSectionScoped(). by andrew@webrtc.org · 13 years ago
7ca9925 Addding myself to video codec watch list. by pwestin@webrtc.org · 13 years ago
89a1000 A minor change in function WebRtcNetEQ_PacketBufferFindLowestTimestamp for by kma@webrtc.org · 13 years ago
7627843 Added NULL check in external transport test code. by mflodman@webrtc.org · 13 years ago
5dad00b Coverty fix: FEC unintended signed extension and resource leaks. by pwestin@webrtc.org · 13 years ago
d3b22c9 Resolved X11 shared memiory leak. by mflodman@webrtc.org · 13 years ago
0c6f931 Removed versions in module/audio_processing and common_audio/vad. by bjornv@webrtc.org · 13 years ago
2fd1e1e Add unittests for ReceiverFec. by stefan@webrtc.org · 13 years ago
31ba407 Enabled GCC 4.6 bot. by phoglund@webrtc.org · 13 years ago
04cf69a Coverty: cleanup CheckCSRC. by pwestin@webrtc.org · 13 years ago
2f77409 Fixed C errors from GCC 4.6. by phoglund@webrtc.org · 13 years ago
1f99280 Fixed frame scaler bugs. by mflodman@webrtc.org · 13 years ago
048eb7c Finished rewriting the audio processing test. by phoglund@webrtc.org · 13 years ago
832adeb Removed MapWrapper from ViEFrameProviderBase. by mflodman@webrtc.org · 13 years ago
194a93a Adding ViE NULL checks. by mflodman@webrtc.org · 13 years ago
cbe1de9 This CL solves three remaining Coverity warnings. by tina.legrand@webrtc.org · 13 years ago
4bcd177 New libsrtp roll @ 119285 by mallinath@webrtc.org · 13 years ago
53df136 Add upload and commit checks to a common function. by andrew@webrtc.org · 13 years ago
a8c568f Fix external codec erase in destructor. by mallinath@webrtc.org · 13 years ago
d1a860b Fixed GCC 4.6 errors (mostly 'unused variable' errors and incorrect usage of EXPECT_EQ with booleans. by phoglund@webrtc.org · 13 years ago
42ae41e Fix enumeral comparison error. by andrew@webrtc.org · 13 years ago
b9d7d93 Rename interface/ to include/ in audio_processing. by andrew@webrtc.org · 13 years ago
24bd58e Properly count anonymous mixing participants. by andrew@webrtc.org · 13 years ago
96bc9c4 Enable audioproc_unittest on Windows. by andrew@webrtc.org · 13 years ago
7adda5c Roll libsrtp 115467:118928. by andrew@webrtc.org · 13 years ago
dcf0064 Fix typo in a comment by henrik.lundin@webrtc.org · 13 years ago
4679652 Implemented a fix for Issue 88. by henrik.lundin@webrtc.org · 13 years ago
9b0a820 Fixed double erase in ViEChannelManager channel map. by mflodman@webrtc.org · 13 years ago
b11424b Remove ViEShared inheritance for interface impl. by mflodman@webrtc.org · 13 years ago
f4b77fd VAD refactor: Mode changed to "int". by bjornv@webrtc.org · 13 years ago
2a4dcd7 VAD refactor: WebRtcVad_InitCore(). by bjornv@webrtc.org · 13 years ago
567b99b Coverity report: fixes an issue where the returnvalue of a function is not checked. by henrike@webrtc.org · 13 years ago
f5d8c3b Fix audioproc_unittest on Windows. by andrew@webrtc.org · 13 years ago
76fa8c9 disable failing win tests. by ivinnichenko@webrtc.org · 13 years ago
d224306 enable video_processing_unittests for Win by ivinnichenko@webrtc.org · 13 years ago
2fc722c Enable building library and test app by leozwang@webrtc.org · 13 years ago
24f1c90 Enable audioproc_unittest on more platforms. by andrew@webrtc.org · 13 years ago
f6bb77a Cleaning up all use of RTP_PAYLOAD_NAME_SIZE and RTCP_CNAME_SIZE also fixed the char handing in trace. by pwestin@webrtc.org · 13 years ago
218db3d Iterator was invalid while removing entries from codec db maps. by mallinath@webrtc.org · 13 years ago
9e332ab Make sure we check the return value from shmat(). by stefan@webrtc.org · 13 years ago
b73c3d1 Bugfix android build. Review URL: https://webrtc-codereview.appspot.com/374003 by pwestin@webrtc.org · 13 years ago
96c39d1 Completed implementation of oauth in coverage scripts. by phoglund@webrtc.org · 13 years ago
28a5cb2 Bugfix receive side only packet loss estimate with NACK. by pwestin@webrtc.org · 13 years ago
40d3c08 Changed max number of vie channels to 32. by perkj@webrtc.org · 13 years ago
52c9d47 Android, Chrome, cleanup, etc. by ivinnichenko@webrtc.org · 13 years ago
ba09cf1 Correcting uninitialized members. by mflodman@webrtc.org · 13 years ago
a5a5cbb Switched from WebRTC wrappers to stl in ChannelManager. by mflodman@webrtc.org · 13 years ago
eeaf3d1 Merge /branches/3.2:r1380 to /trunk by andrew@webrtc.org · 13 years ago
6cf529d Changed REMB return value to int instead of bool. by mflodman@webrtc.org · 13 years ago
d3a0c1c Merge /branches/3.2:r1378 to /trunk by andrew@webrtc.org · 13 years ago
4bc24c4 Optimized function WebRtcSpl_FilterARFastQ12 for ARM platform. by kma@webrtc.org · 13 years ago
6da8eeb Removing an assert for a case that can occur by punyabrata@webrtc.org · 13 years ago
f5cacdc Fix line aligement Review URL: https://webrtc-codereview.appspot.com/373002 by leozwang@webrtc.org · 13 years ago
2442de1 Clean up PRESUBMIT.py, and enable license check. by andrew@webrtc.org · 13 years ago
f9cd693 Enable vp8 and videoengine on android by leozwang@webrtc.org · 13 years ago
a45d05a Add brighten.cc to makefile by leozwang@webrtc.org · 13 years ago
376be6c Fix compilation error Review URL: https://webrtc-codereview.appspot.com/358005 by leozwang@webrtc.org · 13 years ago
b30f0ed Bugfix buffer usage out of scope. by pwestin@webrtc.org · 13 years ago
12dbc23 Rewrote volume test. by phoglund@webrtc.org · 13 years ago
175fecd Fix clang build error. by stefan@webrtc.org · 13 years ago
8fe03af Refactor to use std::list in the video rtp play tools. by stefan@webrtc.org · 13 years ago
152c34c VAD-refactor. Changed to int as return value for WebRtcVad_set_mode(). by bjornv@webrtc.org · 13 years ago
3b57ee0 Rewrote DTMF test. by phoglund@webrtc.org · 13 years ago
31627fe Add vie_remb.cc to makefile by leozwang@webrtc.org · 13 years ago
e2ed5ba Enable audioproc_unittest on all platforms. by andrew@webrtc.org · 13 years ago
2638577 Add an argument in ANDROID_NOT_SUPPORT macro by leozwang@webrtc.org · 13 years ago
f27916a Remove use of MapWrapper in video_coding. by stefan@webrtc.org · 13 years ago
d798953 NetEqRTPplay modification by henrik.lundin@webrtc.org · 13 years ago