Log - 27cb3017f5f7037dcd577b853673015a15ca4c29 - platform/external/webrtc

27cb301 Updated version number to 3.20 by elham@webrtc.org · 13 years ago
bc9a959 Generalized suppression for Trace::Add by phoglund@webrtc.org · 13 years ago
acc54b4 Added perf expectations and corrected existing tests to remove spaces from series names. by phoglund@webrtc.org · 13 years ago
c38eef8 Reformatted RTPReceiver. by phoglund@webrtc.org · 13 years ago
df3a15f Removed spaces from full stack test labels, consolidated graphs by phoglund@webrtc.org · 13 years ago
1ea4b50 Refactor receiver.h/.cc. by stefan@webrtc.org · 13 years ago
1926d33 Change Sleep() comment in test fixture. by andrew@webrtc.org · 13 years ago
bcb7174 .gitignore: Add *.mk, created as part of ChromiumOS build by andrew@webrtc.org · 13 years ago
f545cf8 Addressing webrtc issue 1237, http://code.google.com/p/webrtc/issues/detail?id=1237. by kma@webrtc.org · 13 years ago
91d8933 Dashboard LKGR parsing builder names by kjellander@webrtc.org · 13 years ago
6f62836 Reverting two mixing test patches: seems to introduce a persistent problem for win voe_auto_test (wrapping problem?) by phoglund@webrtc.org · 13 years ago
5c8d9d3 Reformatted tick_util. by phoglund@webrtc.org · 13 years ago
daabfd2 Reformatted trace* files. by phoglund@webrtc.org · 13 years ago
201d4b6 Fix implicit conversion error in mixing test. by andrew@webrtc.org · 13 years ago
b2b628d Further relax thresholds in mixing test. by andrew@webrtc.org · 13 years ago
00c7c43 Replace voice engine utility functions with system wrapper variants. by andrew@webrtc.org · 13 years ago
943770b Fixed various problems with the reformat script: by phoglund@webrtc.org · 13 years ago
ec9c942 Reformatted thread and static_instance. by phoglund@webrtc.org · 13 years ago
a19d04e Coverity now uses Visual Studio 2010 project file by kjellander@webrtc.org · 13 years ago
1b6da28 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. by pwestin@webrtc.org · 13 years ago
f556890 Added possibility to repeat frames. Also added unittest for that feature. by brykt@google.com · 13 years ago
d73527c Changed assert to log. by mflodman@webrtc.org · 13 years ago
d0d4149 Adding AUDIO application as default for Opus stereo by tina.legrand@webrtc.org · 13 years ago
ad0ed58 Fixed a missed initialization (found by valgrind FYI bot). by phoglund@webrtc.org · 13 years ago
ac77084 Roll opus to 172355 and delete opus_demo from webrtc opus by leozwang@webrtc.org · 13 years ago
6bc5d4d Reformatted sort. by phoglund@webrtc.org · 13 years ago
1960219 Make protection method, filename and resolution configurable for FullStackTest. by stefan@webrtc.org · 13 years ago
4275ab1 Implement NetEq duration estimation for Opus. by tina.legrand@webrtc.org · 13 years ago
515ef24 Clean up variable after it gets deleted by leozwang@webrtc.org · 13 years ago
e239bf0 Making I420VideoFrame ref-counted by mikhal@webrtc.org · 13 years ago
b13dfbf Making barcode tools work on Windows + fixes. by kjellander@webrtc.org · 13 years ago
0b18fb3 vie auto test: Adding a constructor for NetworkParameters by mikhal@webrtc.org · 13 years ago
622c8bd ViE autotest: Adding loss models to the external transport by mikhal@webrtc.org · 13 years ago
6e0ce73 Reformatted map classes. by phoglund@webrtc.org · 13 years ago
61f39a3 Fixed bad header name. by phoglund@webrtc.org · 13 years ago
07bf43c Replaced the _audio parameter with a strategy. by phoglund@webrtc.org · 13 years ago
59ad541 Reformatted rw_lock classes. by phoglund@webrtc.org · 13 years ago
eaebeb3 Without specifying the input files the offsets will not automatically be regenerated when building for different architectures. That is very risky as it will cause crashes rather than build errors. by stefan@webrtc.org · 13 years ago
10abe25 Make audioproc output files be written to output dir by default. by kjellander@webrtc.org · 13 years ago
3c37354 Initialize 3 variables which are preventing VS2012 from building. by fbarchard@google.com · 13 years ago
4c32439 Roll libyuv to r520. Includes security fix to mark stack as not executable. by fbarchard@google.com · 13 years ago
ad6845f Updated version number to 3.19 by elham@webrtc.org · 13 years ago
c5fcb08 Update trace_event.h to match the one in Chromium by hclam@chromium.org · 13 years ago
dec09ee libyuv r515 ports matrix effects to Neon by fbarchard@google.com · 13 years ago
4aee6b6 Added API to get receive side video delay. by mflodman@webrtc.org · 13 years ago
1c75918 Disabled flaky test. by phoglund@webrtc.org · 13 years ago
7659d91 Decoupled video rtp receiver from rtp receiver. by phoglund@webrtc.org · 13 years ago
52d981f Reformatted list classes. by phoglund@webrtc.org · 13 years ago
3251939 Remove latency excl network and add render time diff stats. by stefan@webrtc.org · 13 years ago
b8ba4d8 Add number of inserted samples to NetEq statistics. by roosa@google.com · 13 years ago
c454fab Reformatting ACM. All changes are bit-exact in this CL. by turaj@webrtc.org · 13 years ago
ddebc17 Fix for buffer overflow, WebRTC issue 1196 by elham@webrtc.org · 13 years ago
96dc627 vpm unit test: Diasble frame dropping in tests by mikhal@webrtc.org · 13 years ago
4493db5 vpm: removing unnecessary memcpy by mikhal@webrtc.org · 13 years ago
7acb65a Added jitter to fake network pipe. by mflodman@webrtc.org · 13 years ago
91c91df Track the actual render time rather than the decode time. by stefan@webrtc.org · 13 years ago
e19b078 Changed so that frame_cutter takes and argument where one can specify in which interval the frames should be deleted between the first frame to cut and the last frame to cut. This can for example be used to decrease the frame rate. by brykt@google.com · 13 years ago
0240e8e Wider TSAN suppression for issue 300 by kjellander@webrtc.org · 13 years ago
92bb417 Decoupled RTP audio processor from RTP receiver. by phoglund@webrtc.org · 13 years ago
5b689ef Will now only require near-perfect PSNR and SSIM. by phoglund@webrtc.org · 13 years ago
86464ea ISAC_main_inst initialized to NULL to avoid potentially garbage pointer passed to WebRtcIsacfix_EncoderInit by fbarchard@google.com · 13 years ago
a8544ea Vp8 tests: Removing legacy unused tests and reorganization of existing ones. by mikhal@webrtc.org · 13 years ago
7877b0f Added noexecstack markers for assembly files (webrtc issue 1172). by kma@webrtc.org · 13 years ago
fa5b6bf Optimized WebRtcIsacfix_Spec2Time() for iSAC-Fix in ARM Neon processor. Speed doubled. by kma@webrtc.org · 13 years ago
1b60ceb Add GetAudioFrame API to VoiceEngine. by roosa@google.com · 13 years ago
b718619 Expose NetEq playout mode off through VoiceEngine. by roosa@google.com · 13 years ago
0870f02 Add API to retreive last received RTP timestamp to VoiceEngine. by roosa@google.com · 13 years ago
d8aeb30 Revert 3269 by andrew@webrtc.org · 13 years ago
735a6ce Will now only require near-perfect PSNR and SSIM. by phoglund@webrtc.org · 13 years ago
740be44 Reformatted file_* classes. by phoglund@webrtc.org · 13 years ago
4e16f25 Remove atomicops.h from WebRTC by hclam@chromium.org · 13 years ago
9f0fc97 Rolllibvpx to 7a09f6b89268 by marpan@webrtc.org · 13 years ago
770a01e Fix build by including trace_event_internal in webrtc namespace by hclam@chromium.org · 13 years ago
f222a00 Use TRACE_EVENT to track time spent in VP8 encoding by hclam@chromium.org · 13 years ago
d2bcde2 Suppressing TSan warnings for system_wrappers_unittests by kjellander@webrtc.org · 13 years ago
ad7efa6 Port Chromium's trace_event.h to WebKit and add by hclam@chromium.org · 13 years ago
02d9df4 Updated webrtc_resources_revision to 11, for adding two test files for APM and iSAC. by kma@webrtc.org · 13 years ago
71258c5 Add a third full stack test and support for random jitter in ext transport. by stefan@webrtc.org · 13 years ago
eaf7cf2 Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced. by mflodman@webrtc.org · 13 years ago
f98ffc6 Removing default trybot names by kjellander@webrtc.org · 13 years ago
42259e7 VoE Changes to enable dual_streaming. by turaj@webrtc.org · 13 years ago
36965b1 Bug fix for iSAC fixed-point. The bug was the result of changes in iSAC floating-point to add 48 kHz extension. by turaj@webrtc.org · 13 years ago
55edaec Revert r3254 due to bot failure on android. by marpan@webrtc.org · 13 years ago
1f3476d Roll libvpx to 000c8414b510. by marpan@webrtc.org · 13 years ago
5bbe069 Reformatted event* classes. by phoglund@webrtc.org · 13 years ago
3bb42ef Made e2e audio quality test write its results to perf. by phoglund@webrtc.org · 13 years ago
72feb0b Not to enum NOTPRESENT audio devices with CoreAudio on Win by braveyao@webrtc.org · 13 years ago
8e49b02 Add more audio codec information into codec list by leozwang@webrtc.org · 13 years ago
451aa5d Adding vp8 sequence coder: simple command line encode and decode. by mikhal@webrtc.org · 13 years ago
3a5a8a8 Properly zero out unmixed frames. by andrew@webrtc.org · 13 years ago
0e73950 Added buildbot benchmarking in iSAC and APM into Android platform build. by kma@webrtc.org · 13 years ago
b968213 vp8 test: Updating creation of enc/dec by mikhal@webrtc.org · 13 years ago
251f64e Updating vp8 test structure by mikhal@webrtc.org · 13 years ago
60d25f9 Updating Vp8 unit tests - Initiating the switch to gtest-based tests, and adding a stride test. by mikhal@webrtc.org · 13 years ago
75f8c78 Fixing path to ptypes.txt in NetEqRTPplay by henrik.lundin@webrtc.org · 13 years ago
df94329 Use different cpufeatures library when building with chrome. by wjia@webrtc.org · 13 years ago
81cffd1 Port Chromium's atomicops to WebRTC by hclam@chromium.org · 13 years ago
63a243a Replace the last occurrence of .s with .h by leozwang@webrtc.org · 13 years ago
96bcac8 Expose Set and Get Recording/Playout sample rate apis by leozwang@webrtc.org · 13 years ago
f4e070e Added auto-call feature to WebRTCDemo. by fischman@webrtc.org · 13 years ago