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a3f30143b7065d08721df7ea9b4dc311cb013567
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pbos@webrtc.org a3f30143b7 Default constructor for RtcpAppHandler. 
Whenever this test (RtcpApplicationDefinedPacketsCanBeSentAndReceived) fails
because it's being run on a slower system (such as one running under valgrind),
valgrind reports a lot of undefined-value errors. Initializing the data
makes sure that, while the EXPECT_EQs trigger, they don't cause any errors in
valgrind.

BUG=
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1822004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4363 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-17 14:25:45 +00:00
..
build
Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle.
2013-07-16 16:37:22 +00:00
common_audio
Include files from webrtc/.. paths in signal_processing/.
2013-07-16 13:32:03 +00:00
common_video
Proper spacing for end-of-namespace comments.
2013-07-03 15:12:26 +00:00
modules
clean up incomplete revert in r4357
2013-07-16 21:52:59 +00:00
system_wrappers
Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle.
2013-07-16 16:37:22 +00:00
test
Add root_path_android.cc to webrtc/test/Android.mk.
2013-07-09 15:24:16 +00:00
tools
Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests.
2013-07-08 14:55:23 +00:00
video_engine
Revert r4301
2013-07-16 19:25:04 +00:00
voice_engine
Default constructor for RtcpAppHandler.
2013-07-17 14:25:45 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
common_types.h
Replace ExtraCodecOptions with new Config class that supports multiple settings at once.
2013-05-14 08:02:25 +00:00
common_unittest.cc
Fix unnamed-type-template-args warnings on clang.
2013-06-27 10:18:09 +00:00
common.h
Added a config class to ease passing a set of options across webrtc.
2013-05-13 08:06:36 +00:00
engine_configurations.h
Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.
2013-06-19 17:03:47 +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
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
Disable clang C++11 warnings to permit OVERRIDE keyword.
2013-05-08 08:34:34 +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.