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jiayl@webrtc.org 7ee0c16edd Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow. 
No behavior change for now since Chromium has not been updated to call SetExcludedWindow.

BUG=2789
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5792 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-26 15:57:43 +00:00
..
build
Enable VS2013 for Windows compilation by default.
2014-03-12 14:41:58 +00:00
common_audio
Add fir_filter to common_audio
2014-03-24 10:16:11 +00:00
common_video
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
2014-02-20 21:08:36 +00:00
examples
Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.
2014-01-24 18:42:12 +00:00
modules
Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow.
2014-03-26 15:57:43 +00:00
system_wrappers
Remove std:: prefixes from C functions in webrtc/.
2014-03-07 15:23:34 +00:00
test
VoE changes to allow forwarding of packets from VoE to ViE BWE.
2014-03-24 10:38:25 +00:00
tools
Tool to establish a loopback call via apprtc turn server.
2014-03-17 14:23:22 +00:00
video
Revert 5775 "Modify bitrate controller to update bitrate based o..."
2014-03-25 19:42:39 +00:00
video_engine
Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams.
2014-03-26 14:32:47 +00:00
voice_engine
Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out.
2014-03-25 19:57:07 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Remove internal codecs from VideoSendStream.
2014-03-19 08:43:57 +00:00
common_types.h
Removing VideoCodecDerived and moving methods inside VideoCodec.
2014-03-25 18:44:58 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
engine_configurations.h
Remove external encryption API for VoE.
2014-02-18 11:27:22 +00:00
experiments.h
Adding API for setting bandwidth estimation configurations.
2014-03-25 10:37:31 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +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
supplement.gypi
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
2014-01-21 15:54:56 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Moves WEBRTC_POSIX define from header file to gyp-settings.
2014-03-07 15:30:21 +00:00
video_engine_tests.isolate
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
video_receive_stream.h
Make VideoReceiveStream::GetStats() const.
2014-02-07 15:32:45 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Simplify pacer interface.
2014-03-19 10:59:52 +00:00
webrtc_examples.gyp
Android example apps: fixes issue where useful failure information was suppressed.
2014-01-21 19:03:51 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Adding NetEq performance test to webrtc_perf_tests
2014-01-10 08:24:04 +00:00
webrtc.gyp
Integrate fake_network_pipe into direct_transport.
2013-12-18 20:28:25 +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.