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3fa87f782e0a11f1ce24e56a61f0b926e99670dc
platform-external-webrtc/webrtc
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zijiehe 3fa87f782e Use FallbackDesktopCapturerWrapper in ScreenCapturerWinMagnifier 
This is a trivial change to remove duplicate logic, i.e. fallback capturer, from
ScreenCapturerWinMagnifier.

BUG=webrtc:7215

Review-Url: https://codereview.webrtc.org/2704943002
Cr-Commit-Position: refs/heads/master@{#16781}
2017-02-22 21:47:00 +00:00
..
api
Adding RTCErrorOr class to be used by ORTC APIs.
2017-02-17 07:31:33 +00:00
audio
Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.
2017-02-21 14:28:10 +00:00
base
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
call
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
common_audio
Fix for left shift of potentially negative value.
2017-02-09 11:05:59 +00:00
common_video
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
examples
Remove non-ARC code from the codebase.
2017-02-22 08:42:11 +00:00
logging
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
media
Fix issue where video scaling gets stuck at low resolution
2017-02-22 17:30:27 +00:00
modules
Use FallbackDesktopCapturerWrapper in ScreenCapturerWinMagnifier
2017-02-22 21:47:00 +00:00
p2p
Rewrite of sigslot that avoids vtables.
2017-02-19 22:12:24 +00:00
pc
Add QP for libvpx VP8 decoder.
2017-02-20 14:43:58 +00:00
sdk
Move YuvConverter from Android API to src.
2017-02-21 13:29:59 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
test
Reland of Add optional visualization file writers to VideoProcessor tests. (patchset #1 id:1 of https://codereview.webrtc.org/2708103002/ )
2017-02-22 09:26:59 +00:00
tools
Add ana config to event log visualiser
2017-02-22 15:33:27 +00:00
video
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
voice_engine
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
.gitignore
…
BUILD.gn
Build plot_videoprocessor_integrationtest by default.
2017-02-21 07:56:39 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +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
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Move iOS tests to XCTest from gtest.
2017-02-14 12:58:56 +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.