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kjellander@webrtc.org c3e097cdc5 Add Android test runner script for WebRTC. 
The Android test execution toolchain scripts in Chromium
has been causing headaches for us several times. Mostly
because they're tailored at running Chrome tests only.

Wrapping their script in our own avoids the pain of
upstreaming new test names to Chromium and rolling them
in to get them running on our bots.

TESTED=Ran a test on a local device using:
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests  --verbose --isolate-file-path webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate --release
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7794 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 09:57:08 +00:00
..
base
Use c++11 features in webrtc/base/network.cc as a test to see if we can use them.
2014-12-01 23:18:27 +00:00
build
Add Android test runner script for WebRTC.
2014-12-03 09:57:08 +00:00
common_audio
Remove -flax-vector-conversions flag for ARM NEON building.
2014-12-02 19:36:14 +00:00
common_video
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
examples
Update Android projects to API level 21.
2014-10-31 23:26:10 +00:00
libjingle
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
modules
Adjust parameter in videoprocessor_integration_test for vp9.
2014-12-03 02:00:12 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
p2p
TurnPort should ignore STUN binding reponses when using shared socket.
2014-12-03 02:17:07 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Add framerate for complete received frames to histogram stats:
2014-11-28 10:17:13 +00:00
test
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
tools
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_engine
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
voice_engine
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
call.h
Implement settable min/start/max bitrates in Call.
2014-11-25 14:03:34 +00:00
common_types.h
Add receive bitrates to histogram stats:
2014-11-27 07:38:56 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Log formatting fix for VideoEncoderConfig.
2014-11-06 09:35:08 +00:00
config.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Put send-side bwe probing under finch experiment.
2014-11-10 13:55:16 +00:00
webrtc.gyp
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.