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platform-external-webrtc/webrtc
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andrew@webrtc.org c3e5d3422b Add a logging_no_op.cc when enable_tracing==0. 
This should hopefully fix static initializer warnings when rolling webrtc
in Chromium.

TEST=logging_unittest succeeds with enable_tracing==1 and fails appropriately with enable_tracing==0.

Review URL: https://webrtc-codereview.appspot.com/939026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3159 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 19:30:59 +00:00
..
build
Add a logging_no_op.cc when enable_tracing==0.
2012-11-23 19:30:59 +00:00
common_audio
Porting ARM optimization from Android to ios.
2012-11-17 00:22:46 +00:00
common_video
Adding an aligned stride test to LibYuv
2012-11-20 23:52:50 +00:00
modules
Remove operator overloading from RTPFragmentationHeader.
2012-11-23 19:17:23 +00:00
system_wrappers
Add a logging_no_op.cc when enable_tracing==0.
2012-11-23 19:30:59 +00:00
test
Add Chromium's perf_test to testsupport.
2012-11-20 00:20:20 +00:00
tools
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
video_engine
Fixes (or at least reduces) the flakiness in the full stack test by making sure the different frame monitors are registered and deregistered in the right order. Also makes sure only local preview frames which are actually transmitted are rendered by moving the local preview rendering to an effect filter.
2012-11-23 11:19:29 +00:00
voice_engine
Add a kTraceTerseInfo level for non-verbose logging.
2012-11-20 07:34:45 +00:00
common_types.h
Add a kTraceTerseInfo level for non-verbose logging.
2012-11-20 07:34:45 +00:00
engine_configurations.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
typedefs.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +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.