webrtc_m130

History

sergeyu@chromium.org 9f282403f2 WindowCapturer implementation for Linux.

Window enumeration is based on the code used by hangouts plugin
(see libjingle/talk/base/linuxwindowpicker.cc). XServerPixelBuffer
is used to capture windows. It had to be refactored to support window
capturing (previously it worked only for the whole screen).

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1741004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4605 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-08-23 18:22:12 +00:00

build

Isolate GYP target and .isolate files for tests

2013-08-22 07:57:00 +00:00

common_audio

Isolate GYP target and .isolate files for tests

2013-08-22 07:57:00 +00:00

common_video

Isolate GYP target and .isolate files for tests

2013-08-22 07:57:00 +00:00

modules

WindowCapturer implementation for Linux.

2013-08-23 18:22:12 +00:00

system_wrappers

Isolate GYP target and .isolate files for tests

2013-08-22 07:57:00 +00:00

test

Isolate GYP target and .isolate files for tests

2013-08-22 07:57:00 +00:00

tools

Isolate GYP target and .isolate files for tests

2013-08-22 07:57:00 +00:00

video_engine

Remove newapi:: namespace for typenames without overlap.

2013-08-23 09:19:30 +00:00

voice_engine

Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests).

2013-08-23 16:16:03 +00:00

.gitignore

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

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

common_types.h

webrtc/common_types.h: Document bitrate fields' units.

2013-07-23 18:32:10 +00:00

common_unittest.cc

Fix unnamed-type-template-args warnings on clang.

2013-06-27 10:18:09 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +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

PRESUBMIT.py

Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.

2013-07-22 22:32:50 +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

webrtc.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. Also disables 64 bit Mac builds for libjingle

2013-07-23 18:15:11 +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.