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zijiehe 9d1ea5cc22 Clear DesktopFrame in DxgiFrame to avoid legacy image 
Once the buffer returned by Windows is not newly allocated, it may contain
legacy images from previous capturing attempts. This usually is not a problem,
as implementations other than ScreenCapturerWinDirectx paint each pixel on the
frame. But due to the one capturer per monitor design of
ScreenCapturerWinDirectx, part of the frame may not be covered by any
DxgiOutputDuplicator, and cause the legacy image to be shown.
So a very simple fix is to clear the DesktopFrame in DxgiFrame.

BUG=708766

Review-Url: https://codereview.webrtc.org/2827983007
Cr-Commit-Position: refs/heads/master@{#17847}
2017-04-24 18:50:05 +00:00
..
api
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
audio
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +00:00
base
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
call
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +00:00
common_audio
Enable GN check for webrtc/base
2017-04-19 15:37:36 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Espresso test for loopback video quality testing
2017-04-24 11:15:13 +00:00
logging
Delay based logging.
2017-04-11 08:50:23 +00:00
media
Add content type extension to capabilities
2017-04-24 12:12:35 +00:00
modules
Clear DesktopFrame in DxgiFrame to avoid legacy image
2017-04-24 18:50:05 +00:00
ortc
Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests
2017-04-18 22:49:09 +00:00
p2p
Fixing SignalSentPacket for TCP connections.
2017-04-21 21:22:23 +00:00
pc
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
sdk
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
test
Revert of Removing test deps in webrtc/test/fuzzers (patchset #1 id:1 of https://codereview.webrtc.org/2840523003/ )
2017-04-24 18:31:17 +00:00
tools
Add network tester client [android] to be able to test mobile networks in terms of packet size and sending rates.
2017-04-20 13:56:27 +00:00
video
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +00:00
voice_engine
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
2017-04-07 07:59:12 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
config.cc
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_decoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_encoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_frame.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.