webrtc_m130

History

henrike@webrtc.org 0df5b8dfa6 Revert 4372 "Makes webrtc and libjingle build from the same gyp-..."

> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches.
> 
> TESTED=git try
> BUG=1932
> R=andrew@webrtc.org, fischman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1804004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1835004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4373 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-07-18 18:58:29 +00:00

build

AppRTCDemo: build fixes for iOS build in webrtc

2013-07-18 02:02:07 +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

AppRTCDemo: build fixes for iOS build in webrtc

2013-07-18 02:02:07 +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.