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jiayl@webrtc.org f0fc72f70e Call PrintWindow for the first time of capturing to capture the window frames correctly. 
This will fix artifacts on the captured window frames, especially for cmd, which
sometimes leaks glimpss of other window's content.

BUG=
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5616 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-27 16:43:12 +00:00
..
build
Rename merged webrtc lib to libwebrtc_merged.a.
2014-02-11 18:22:29 +00:00
common_audio
Clean up CPU detection defines in SincResampler a little.
2014-02-27 04:12:34 +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
Call PrintWindow for the first time of capturing to capture the window frames correctly.
2014-02-27 16:43:12 +00:00
system_wrappers
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
2014-02-20 21:08:36 +00:00
test
Add SetConfig method to FakeNetworkPipe and to DirectTransport
2014-02-26 13:34:52 +00:00
tools
Now printing less output from compare_videos.py.
2014-01-07 17:59:30 +00:00
video
Incorrect overhead calculation when using FEC + RTP extension headers.
2014-02-18 08:40:33 +00:00
video_engine
Fix to get total number of sent and received rtcp packets.
2014-02-24 09:27:38 +00:00
voice_engine
Removes VoERTP_RTCP::InsertExtraRTPPacket.
2014-02-19 08:58:08 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Add configuration for cpu overuse detection to video send stream.
2014-01-31 10:05:07 +00:00
common_types.h
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
2014-02-19 11:59:02 +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
Missing include in experiments.h
2014-02-25 09:17:43 +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
Clean up CPU detection defines in SincResampler a little.
2014-02-27 04:12:34 +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
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +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.