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pbos@webrtc.org 34f2a9ea72 Initialize SSL in unittest_main.cc. 
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-28 11:36:45 +00:00
..
base
Initialize SSL in unittest_main.cc.
2014-09-28 11:36:45 +00:00
build
Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175)
2014-09-23 12:43:14 +00:00
common_audio
gn: Fix cflags usage
2014-09-16 17:57:02 +00:00
common_video
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
2014-09-17 09:02:25 +00:00
examples
Split video engine android initialization into each internal module initialization.
2014-09-17 11:44:51 +00:00
libjingle/xmllite
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
modules
Getting orientation is not working properly. VideoCaptureImpl::RotationFromDegrees returns -1 in case fails not 0. So we need to change the if statement.
2014-09-26 22:50:06 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
sound
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
system_wrappers
Move thread_annotations.h to webrtc/base/.
2014-09-24 06:05:00 +00:00
test
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
tools
Adding test file path as argument of the rtcBot run command's arguments.
2014-09-22 15:24:56 +00:00
video
Move thread_annotations.h to webrtc/base/.
2014-09-24 06:05:00 +00:00
video_engine
Mark virtual overrides of ViENetwork and VoENetwork as such.
2014-09-26 11:09:08 +00:00
voice_engine
Mark virtual overrides of ViENetwork and VoENetwork as such.
2014-09-26 11:09:08 +00:00
.gitignore
…
BUILD.gn
Fix the RTC+Chromium GN build.
2014-09-09 19:15:33 +00:00
call.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
common_types.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
…
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +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
…
LICENSE_THIRD_PARTY
…
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
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
…
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
…
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_encoder.h
Expose VP8/H264 defaults through video_encoder.h.
2014-09-18 12:42:28 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_frame.h
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
2014-09-17 09:02:25 +00:00
video_receive_stream.h
Change return value for number of discarded packets to be int.
2014-09-04 07:07:44 +00:00
video_renderer.h
…
video_send_stream.h
Config struct for VideoEncoder.
2014-09-19 12:30:25 +00:00
webrtc_examples.gyp
Roll chromium_revision 6455c69..deaf2f7 (293954:295079)
2014-09-27 18:10:30 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Remove the 'webrtc_test_video_render_dependencies' target.
2014-09-18 17:22:18 +00:00
webrtc.gyp
Split video_render_module implementation into default and internal implementation.
2014-09-18 08:58:15 +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.