Log - 54536bb6d4073c35524579e6be58888213398af4 - platform/external/webrtc

54536bb Refactoring of the TMMBRSet class, giving it a reasonably tight interface. by hta@webrtc.org · 12 years ago
3bc38c8 Fix color enhancement test. by pwestin@webrtc.org · 12 years ago
209442a Missing file Review URL: https://webrtc-codereview.appspot.com/556005 by pwestin@webrtc.org · 12 years ago
e9727cd Fixed some memory leaks. by pwestin@webrtc.org · 12 years ago
63a5098 Rename AudioFrame members. by andrew@webrtc.org · 12 years ago
7fdb909 Reformat and add more debug info into ViESurfaceRenderer by leozwang@webrtc.org · 12 years ago
b286bfb VAD refactoring: Replaced hard coded array sizes with enum. by bjornv@webrtc.org · 12 years ago
404843e Timeout tests for TMMBR by hta@webrtc.org · 12 years ago
3c0df7d Fixed a build break: I'd forgotten to remove a DELETE statement. by hta@webrtc.org · 12 years ago
47059b5 Adds unit tests for RTCP receiver, focusing on TMMBR handling. by hta@webrtc.org · 12 years ago
1e1dd17 Disabling PTY for mv command on Android bot. by kjellander@webrtc.org · 12 years ago
719dba7 Further cleaned up voe_standard_test. by phoglund@webrtc.org · 12 years ago
dbb7f91 The ChromeBloat bot will now also be ignored for LKGRs. by phoglund@webrtc.org · 12 years ago
efecc18 libyuv updates for better code generation on OSX clang for scale by fbarchard@google.com · 12 years ago
ecac9b7 Add tests for downmixing and no processing. by andrew@webrtc.org · 12 years ago
63ea5ef Regenerate jni files and bring audio alive by leozwang@webrtc.org · 12 years ago
d5548f5 Disable clang Chrome plugins on all platforms. by andrew@webrtc.org · 12 years ago
fed1894 Roll Chromium 132375:134666 and libyuv 216:254. by andrew@webrtc.org · 12 years ago
85b089a Fix ConvertI420ToRGB565 bug by leozwang@webrtc.org · 12 years ago
e7ac5fd Minor changes to remove dead code in opensl es by leozwang@webrtc.org · 12 years ago
65a4e4e Minor refactoring: RTCPReceiver::BoundingSet by hta@webrtc.org · 12 years ago
890aa0d Disabling HTTP status push on Build master since we're not using it by kjellander@webrtc.org · 12 years ago
be0ac63 Overriding tgrid URL for master web status by kjellander@webrtc.org · 12 years ago
c2d9852 untabify by hta@webrtc.org · 12 years ago
9d54cd1 TMMBN sender test passes, fixed receiver flag bug by hta@webrtc.org · 12 years ago
c6c4ffc Android trybots + fixing web status config by kjellander@webrtc.org · 12 years ago
5c0c18d Fix coverity issues in ACM. by andrew@webrtc.org · 12 years ago
a88cb6f Add HighPassFilter and StereoChannelSwapping tests. by andrew@webrtc.org · 12 years ago
2d02232 VPM: fix to coverity issues 10255-10258 (unintended sign extension). by marpan@webrtc.org · 12 years ago
ca08c41 Replacing virtual camera on linux video bot: adaptings tests accordingly. by phoglund@webrtc.org · 12 years ago
2191a47 Update ARM specific libvpx config files. by stefan@webrtc.org · 12 years ago
b1313aa Fix missing h file change. by pwestin@webrtc.org · 12 years ago
49888ce Breaking out send side bitrate controll cont. by pwestin@webrtc.org · 12 years ago
e611619 Fixing the header file path in gypi file. by mallinath@webrtc.org · 12 years ago
9c4f6a5 Add an AudioFrameOperations unittest. by andrew@webrtc.org · 12 years ago
e49d908 Fix how we were using TbInterfaces and disallow operator=() and the copy constructor. by tommi@webrtc.org · 12 years ago
a990e12 * Change the reference counting implementation for VoE to be per object and by tommi@webrtc.org · 12 years ago
497fb4f Fixed vie_auto_test on mac so it will exit when the test completes instead of hanging like it used to. by phoglund@webrtc.org · 12 years ago
bc1b43b Refactoring of audio_coding_module_impl by tina.legrand@webrtc.org · 12 years ago
a6ecd1e Refactoring one of the ACM tests: TestStereo, to follow the style guide. by tina.legrand@webrtc.org · 12 years ago
6133113 Reduced the time from check-in to build significantly. by phoglund@webrtc.org · 12 years ago
1868780 Disabled UnremovedSocketsGetCollectedAtManagerDeletion in UdpSocketManager unittest. by mflodman@webrtc.org · 12 years ago
1c7bfe0 Fail silently when swapping mono. by andrew@webrtc.org · 12 years ago
da12dde Upgrade libvpx to dba0538. by stefan@webrtc.org · 12 years ago
ad92989 Tests for udp_socket_manager. by hta@webrtc.org · 12 years ago
5288481 Removed dependency which has moved into tools/DEPS. by phoglund@webrtc.org · 12 years ago
6a65cfb Enabled the new PyAuto test on the build bot. by phoglund@webrtc.org · 12 years ago
d18dd6d Made OnPacketLossStatisticsUpdate function virtual (needed for ViCE). by asapersson@webrtc.org · 12 years ago
02d7174 Add API to swap stereo channels. by andrew@webrtc.org · 12 years ago
369166a Add API for disabling the high pass filter. by andrew@webrtc.org · 12 years ago
48a5df6 Embed svn revision number into code by leozwang@webrtc.org · 12 years ago
b28b43a6 Adding alwaysUseLatest parameter for GClient sync. by kjellander@webrtc.org · 12 years ago
5f49dba Hi Magnus, I added some of the changes that you suggested before. Let me know what you think. by elham@webrtc.org · 12 years ago
7401a1f Updating Chrome excludes to use the same as Chrome buildbots. by kjellander@webrtc.org · 12 years ago
4e423b3 Handle master/slave timestamp wrap. by andrew@webrtc.org · 12 years ago
55e4fff LCOV fix in addition to r2095. by kjellander@webrtc.org · 12 years ago
99ac3f7 Fixed trunacated trace problem in WebRTC. http://b.corp.google.com/issue?id=5607856 by vikasmarwaha@webrtc.org · 12 years ago
f3794f8 Fixed normal LCOV case by kjellander@webrtc.org · 12 years ago
ddab60b Wire up pading. Review URL: https://webrtc-codereview.appspot.com/509002 by pwestin@webrtc.org · 12 years ago
11654c2 VAD refactoring: Local variable name changes by bjornv@webrtc.org · 12 years ago
5284d6e Minor change to trigger REMB packets in RTCP RR if there is no sending channel. by mflodman@webrtc.org · 12 years ago
bf9f469 Lifetime management for UdpSocketManager by hta@webrtc.org · 12 years ago
92591ad Fixes link issues in google3 (change by tomasl). by asapersson@webrtc.org · 12 years ago
83ed0a4 Try to resend next packet in nack list even if previous packet is not found. by asapersson@webrtc.org · 12 years ago
f6cd33d Implemented bloat calculation. This will measure the binary size of Chrome+WebRTC components each weekend. by phoglund@webrtc.org · 12 years ago
39946f1 Skipping code coverage HTML generation on failed builds. by kjellander@webrtc.org · 12 years ago
fcbbe1f Removed unused callback code from video coding test. by pwestin@webrtc.org · 12 years ago
a2cd732 Fix for calling OnNetworkChanged too often. by pwestin@webrtc.org · 12 years ago
88ad06b VCM: Added condition(s) for setting FEC protection factor to zero at low bitrates. by marpan@webrtc.org · 12 years ago
63a34f4 Fix in SendPadData. Do not increase sequence number if packet is "empty" and not sent. by asapersson@webrtc.org · 12 years ago
bb77000 Added a virtual destructor to get the test to compile on all platforms. by phoglund@webrtc.org · 12 years ago
b73f01e Removed some obviously dead stuff from voe_auto_test. by phoglund@webrtc.org · 12 years ago
bbd6b56 Memory leak fix: Deleting a factory by hta@webrtc.org · 12 years ago
bcde776 Changed Delay Estimator create call by bjornv@webrtc.org · 12 years ago
0abe535 Refactored udp_transport to take socket manager as dependency injection by hta@webrtc.org · 12 years ago
4e645ee Improved error message when capture device is missing. by kjellander@webrtc.org · 12 years ago
b61f1fa Reset slave when switching to a stereo codec. by andrew@webrtc.org · 12 years ago
00a8dbb Change Watchlist Review URL: https://webrtc-codereview.appspot.com/503001 by leozwang@webrtc.org · 12 years ago
f4c80fc Switch the other android build over to android_posix.cc and not android_linux.cc. by tommi@webrtc.org · 12 years ago
9018c9f Fix androidndk build take two. by tommi@webrtc.org · 12 years ago
3db5cb7 Fix AndroidNDK build. by tommi@webrtc.org · 12 years ago
3d48b09 Fix android build. by tommi@webrtc.org · 12 years ago
0cac8be Fixing e-mail notification for buildbot master by kjellander@webrtc.org · 12 years ago
e84373c Atomic32Wrapper -> Atomic32 by tommi@webrtc.org · 12 years ago
1cd1162 Break out of send side bandwidth estimation and controll. by pwestin@webrtc.org · 12 years ago
b2bd1e0 Bugfix too many initialize for RTP module by pwestin@webrtc.org · 12 years ago
76643d7 Enabling fastbuild in GYP define. by kjellander@webrtc.org · 12 years ago
a768970 Parse out ssrcs in REMB message (needed for ViCE) . by asapersson@webrtc.org · 12 years ago
faa0ab8 NetEQ stereo sync by tina.legrand@webrtc.org · 12 years ago
22082e7 Enable iSAC_FIX on android by leozwang@webrtc.org · 12 years ago
16f6bb3 Fix a minor compilation error on android by leozwang@webrtc.org · 12 years ago
efd01fd Removing unused code from QMVideoSettingsCallback. by marpan@webrtc.org · 12 years ago
82d85ae All errors are now printed to stderr instead of stdout. by kjellander@webrtc.org · 12 years ago
4ade550 Delay Estimator Unit tests by bjornv@webrtc.org · 12 years ago
2e72976 New _CreateBinaryDelayEstimator() and removed _history_size() by bjornv@webrtc.org · 12 years ago
180f83f File name change to follow style by bjornv@webrtc.org · 12 years ago
1bc98bc Remove erroneous error trace. by andrew@webrtc.org · 12 years ago
7ab5149 Remove usage of Atomic32Wrapper from a few places. by tommi@webrtc.org · 12 years ago
52c0fec Added UDP socket factory function to UdpTransportImpl constructor by hta@webrtc.org · 12 years ago
c3eb178 Will now filter chrome-triggered builds. by phoglund@webrtc.org · 12 years ago