webrtc_m130

History

pbos@webrtc.org e4b6064f8e Replace legacy G_CONST with const.

BUG=1608

Review URL: https://webrtc-codereview.appspot.com/1310005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3814 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-10 18:06:57 +00:00

build

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

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

common_audio

Replace legacy G_CONST with const.

2013-04-10 18:06:57 +00:00

common_video

WebRtc_Word32 -> int32_t in common_video.

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

modules

Replace legacy G_CONST with const.

2013-04-10 18:06:57 +00:00

system_wrappers

Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps."

2013-04-09 14:56:29 +00:00

test

WebRtc_Word32 -> int32_t in test/

2013-04-09 11:10:21 +00:00

tools

Fix frame_editing_unittest reference file handling.

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

video_engine

More trace events

2013-04-09 19:54:10 +00:00

voice_engine

WebRtc_Word32 -> int32_t in voice_engine/

2013-04-09 10:09:10 +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.