webrtc_m130

History

sergeyu@chromium.org a0cd9182aa Add desktop_capture directory for screen and window capturers.

The screen captures will be moved from chromium to WebRTC to make it easy
to share this code with other projects. This CL adds a new directory where
the current screen capturer code will be moved to.

Review URL: https://webrtc-codereview.appspot.com/1297005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3866 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-17 18:21:42 +00:00

build

Use ninja with merge_libs.py on Windows and clean up.

2013-02-26 02:47:50 +00:00

common_audio

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

common_video

WebRtc_Word32 -> int32_t in common_video.

2013-04-09 13:36:08 +00:00

modules

Add desktop_capture directory for screen and window capturers.

2013-04-17 18:21:42 +00:00

system_wrappers

Make win_support_condition_variables_primitive global to aligned with |library|

2013-04-15 18:06:42 +00:00

test

Revert "Updating test file contents to emmastjernloef"

2013-04-17 14:59:28 +00:00

tools

Fix frame_editing_unittest reference file handling.

2013-03-11 15:30:33 +00:00

video_engine

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

voice_engine

Fix for "RTP dynamic payload type 100 is reserved"

2013-04-17 07:34:25 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

common_types.h

Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.

2013-03-28 09:14:36 +00:00

engine_configurations.h

Remove WEBRTC_*_ENGINE_NETWORK_API use

2013-04-04 19:51:42 +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

Replace legacy G_CONST with const.

2013-04-10 18:06:57 +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.