Log - 68a95e15db40860be332f453b6e3ae6b44a327ca - fp2-dev/platform/external/chromium_org/third_party/webrtc

b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
3841668 Fix loopback test for case where no constraint is given. by andresp@webrtc.org · 10 years ago
ea1b72d Add ability to control peer connection constraints for the loopback test. by andresp@webrtc.org · 10 years ago
4c6d59a Allow loopback tests to do TURN when served from webrtc.googlecode.com. by andresp@webrtc.org · 10 years ago
3bf1f38 Tool to establish a loopback call via apprtc turn server. by andresp@webrtc.org · 10 years ago
340b16e Add support for YUV4MPEG file reading to tools files. (Minor fix). by mcasas@webrtc.org · 10 years ago
bc5c7bc Add support for YUV4MPEG file reading to tools files. by mcasas@webrtc.org · 10 years ago
859b462 Now printing less output from compare_videos.py. by phoglund@webrtc.org · 11 years ago
894dab9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
5892ce5 Made video quality toolchain more configurable. by phoglund@webrtc.org · 11 years ago
e0df4d7 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
e3709a8 Make video quality analysis unittests print to log instead of stdout. by kjellander@webrtc.org · 11 years ago
4043e7e Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
05ee189 Move audio_e2e_harness into include_tests==1 condition. by kjellander@webrtc.org · 11 years ago
8ee76b9 Add audio_e2e_test target to tools.gyp by kjellander@webrtc.org · 11 years ago
0b7aefe Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
9670be6 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
3de1b22 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
712d30f Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
37ae275 Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
1a58624 Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
4c94668 Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 11 years ago
b1b278e Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 11 years ago
fd9b155 Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
d09ee87 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
ac38916 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
12e3ee7 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
50ff6a5 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
9d939ee Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
9bf2b46 Fix some chromium-style warnings in webrtc/tools/ by pbos@webrtc.org · 11 years ago
263411c Improved error messages when binaries are missing. Also stderr = stdout now. by phoglund@webrtc.org · 11 years ago
0642536 Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
ad63306 Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
c82b35c Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
0c9b40d Moves tools/update.py to trunk/webrtc/tools and updates it so that it no longer pulls any information from the DEPS file. by henrike@webrtc.org · 11 years ago
9f56b60 Fixed bad parameter passing in compare_videos.py by phoglund@webrtc.org · 11 years ago
7ddad3e Include files from webrtc/.. paths in tools/ by pbos@webrtc.org · 11 years ago
251c209 Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
3f3dcd1 Moved command line parsing to internal tools and moved back the mic volume thingie. by phoglund@webrtc.org · 11 years ago
1d76489 Moved force_volume_max to its own gyp file to avoid a circular dependency. by phoglund@webrtc.org · 11 years ago
f20975f Wrote a small portable tool for forcing the mic volume to 100%. by phoglund@webrtc.org · 11 years ago
6bbd8b1 Add script for comparing video quality by kjellander@webrtc.org · 11 years ago
38a5679 Fix frame_editing_unittest reference file handling. by kjellander@webrtc.org · 11 years ago
46672bb Refactor barcode decoder to use Zxing's C++ version by kjellander@webrtc.org · 11 years ago
cc895d1 Fixing/disabling Windows x64 warnings by kjellander@webrtc.org · 11 years ago
294f055 Fix Win64 warnings by kjellander@webrtc.org · 12 years ago
6caf203 Fix frame_editing_unittest.cc by kjellander@webrtc.org · 12 years ago
3f6593a Remove <(library) from gyp file. by wjia@webrtc.org · 12 years ago
17e37d4 Added possibility to repeat frames. Also added unittest for that feature. by brykt@google.com · 12 years ago
eeb8b97 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 · 12 years ago
e3c4549 Fixed bug that caused frame_cutter_unittest to fail when built with MVS2008. by brykt@google.com · 12 years ago
10dff40 Use <(webrtc_root) to point to webrtc files in tools.gyp. by andrew@webrtc.org · 12 years ago
5e6a94a Tool for editing of yuv-files. Specify a path to the clip that should be edited, the height and width of the clip, one set of frames that should be removed from the clip, and a path to where the result should be written. There is a executable created that make use of the library where the functionality is implemented. There is also a unittest added for the library. by brykt@google.com · 12 years ago
a7b57da Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago