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webrtc
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src
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webrtc
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c7470676faa4d5ed25cf1fc83d2542d887604007
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tools
881c38a
Adding webrtc_video_streaming test
by houssainy@google.com
· 10 years ago
5eaf95b
Fix printing of error stack in rtcbot when a test fails via test.fail().
by houssainy@google.com
· 11 years ago
c529f50
Bot Browser files moved to /bot/browser/
by houssainy@google.com
· 11 years ago
f1b1a3b
Adding the ability to test on Chrome for Android.
by houssainy@google.com
· 11 years ago
de4cf01
- Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices.
by houssainy@google.com
· 11 years ago
c06f92d
Moving the api.js and bot.js to /rtcbot/bot/ to be shared between
by houssainy@google.com
· 11 years ago
49d6220
Change gflags and gmock includes to be full paths.
by kjellander@webrtc.org
· 11 years ago
5191730
Partial revert of r7014 (Android APK refactor)
by kjellander@webrtc.org
· 11 years ago
3dbd813
RTCBot is a framework that allows to write tests where logic runs on a single
by andresp@webrtc.org
· 11 years ago
b9d6b2b
Android APK tests built from a normal WebRTC checkout.
by kjellander@webrtc.org
· 11 years ago
00dffd7
Pass GYP DEPTH variable to isolate.
by kjellander@webrtc.org
· 11 years ago
f006e8d
Add kjellander@webrtc.org as OWNER for *.isolate
by kjellander@webrtc.org
· 11 years ago
00d9c49
Android: cleanup gtest_target_type conditions.
by henrike@webrtc.org
· 11 years ago
774b3d3
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
by henrike@webrtc.org
· 11 years ago
0a9ed7c
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
by mcasas@webrtc.org
· 11 years ago
28b7c07
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
by henrike@webrtc.org
· 11 years ago
0b8a1c4
Add webrtc field trials API.
by andresp@webrtc.org
· 11 years ago
ba47616
Replace scoped_array<T> with scoped_ptr<T[]>.
by andrew@webrtc.org
· 11 years ago
98f8320
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
by fischman@webrtc.org
· 11 years ago
5b3c956
Fix loopback test for case where no constraint is given.
by andresp@webrtc.org
· 11 years ago
ccef356
Add ability to control peer connection constraints for the loopback test.
by andresp@webrtc.org
· 11 years ago
1fb05fc
Allow loopback tests to do TURN when served from webrtc.googlecode.com.
by andresp@webrtc.org
· 11 years ago
d073362
Tool to establish a loopback call via apprtc turn server.
by andresp@webrtc.org
· 11 years ago
8b04780
Add support for YUV4MPEG file reading to tools files. (Minor fix).
by mcasas@webrtc.org
· 11 years ago
280ab2a
Add support for YUV4MPEG file reading to tools files.
by mcasas@webrtc.org
· 11 years ago
2c358e2
Now printing less output from compare_videos.py.
by phoglund@webrtc.org
· 11 years ago
ca63ad9
Roll chromium_revision 232627:238260
by kjellander@webrtc.org
· 11 years ago
d95137e
Made video quality toolchain more configurable.
by phoglund@webrtc.org
· 11 years ago
b4db9c3
Remove update_resources.py as it's no longer used.
by kjellander@webrtc.org
· 11 years ago
f8a1798
Make video quality analysis unittests print to log instead of stdout.
by kjellander@webrtc.org
· 11 years ago
2714c79
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
by sprang@webrtc.org
· 11 years ago
50edafc
Move audio_e2e_harness into include_tests==1 condition.
by kjellander@webrtc.org
· 11 years ago
b9586f0
Add audio_e2e_test target to tools.gyp
by kjellander@webrtc.org
· 11 years ago
8e70108
Reorganize GYP targets to make webrtc.gyp more usable.
by kjellander@webrtc.org
· 11 years ago
bf1da46
Add APK and isolate target for video_engine_tests
by kjellander@webrtc.org
· 11 years ago
22a2893
Fix include of isolate.gypi
by kjellander@webrtc.org
· 11 years ago
369da50
Workaround issue with stdin on Windows.
by kjellander@webrtc.org
· 11 years ago
053d45a
Update sampling rate and number of channels of NetEq4 if decoder is changed.
by turaj@webrtc.org
· 11 years ago
cdc5e6a
Remove include_dirs from tools.
by pbos@webrtc.org
· 11 years ago
dadb2a1
Fix format string in video_quality_analysis.cc.
by pbos@webrtc.org
· 12 years ago
3524ade
Convert printing in video quality tests to Chromium's perf format.
by kjellander@webrtc.org
· 12 years ago
8c6633c
Add isolate configuration for Android for all tests.
by kjellander@webrtc.org
· 12 years ago
3540c82
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 12 years ago
e155918
Revert 4547 "Isolate GYP target and .isolate files for tests"
by kjellander@webrtc.org
· 12 years ago
298bbdb
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 12 years ago
3f45c2e
Switch C++-style C headers with their C equivalents.
by pbos@webrtc.org
· 12 years ago
acb00f5
Adds all unittests to android NDK-APK framework.
by henrike@webrtc.org
· 12 years ago
08a3b0d
Fix some chromium-style warnings in webrtc/tools/
by pbos@webrtc.org
· 12 years ago
3081f6d
Improved error messages when binaries are missing. Also stderr = stdout now.
by phoglund@webrtc.org
· 12 years ago
e25e28f
Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests.
by henrike@webrtc.org
· 12 years ago
00d566e
Revert 4298 "Makes it possible to find files used by some unit t..."
by pbos@webrtc.org
· 12 years ago
222efdc
Makes it possible to find files used by some unit tests when running them as Chrome native tests.
by henrike@webrtc.org
· 12 years ago
8a5cb95
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
· 12 years ago
c10fc53
Fixed bad parameter passing in compare_videos.py
by phoglund@webrtc.org
· 12 years ago
cff5c03
Include files from webrtc/.. paths in tools/
by pbos@webrtc.org
· 12 years ago
5221d1c
Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.
by andrew@webrtc.org
· 12 years ago
de93f2c
Moved command line parsing to internal tools and moved back the mic volume thingie.
by phoglund@webrtc.org
· 12 years ago
fefc490
Moved force_volume_max to its own gyp file to avoid a circular dependency.
by phoglund@webrtc.org
· 12 years ago
a4f0d20
Wrote a small portable tool for forcing the mic volume to 100%.
by phoglund@webrtc.org
· 12 years ago
dc2b152
Add script for comparing video quality
by kjellander@webrtc.org
· 12 years ago
9f4ae03
Fix frame_editing_unittest reference file handling.
by kjellander@webrtc.org
· 12 years ago
7b6cbc9
Refactor barcode decoder to use Zxing's C++ version
by kjellander@webrtc.org
· 12 years ago
1e7f77a
Fixing/disabling Windows x64 warnings
by kjellander@webrtc.org
· 12 years ago
f6a2efa
Fix Win64 warnings
by kjellander@webrtc.org
· 12 years ago
f47223d
Fix frame_editing_unittest.cc
by kjellander@webrtc.org
· 12 years ago
0a7fc8d
Remove <(library) from gyp file.
by wjia@webrtc.org
· 12 years ago
f8b1da2
Added possibility to repeat frames. Also added unittest for that feature.
by brykt@google.com
· 12 years ago
bd11d95
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
557655f
Fixed bug that caused frame_cutter_unittest to fail when built with MVS2008.
by brykt@google.com
· 12 years ago
ad2a55a
Use <(webrtc_root) to point to webrtc files in tools.gyp.
by andrew@webrtc.org
· 12 years ago
d9b18e9
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
b015cbe
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago